From msc at freeswitch.org Sun Jul 1 01:03:13 2012 From: msc at freeswitch.org (Michael Collins) Date: Sat, 30 Jun 2012 14:03:13 -0700 Subject: [Freeswitch-users] how to get the content of sip header ? In-Reply-To: <1341061052.23871.YahooMailNeo@web120106.mail.ne1.yahoo.com> References: <1341061052.23871.YahooMailNeo@web120106.mail.ne1.yahoo.com> Message-ID: See if you have a bunch of sip_xxx channel variables. Send the call to the info app and see what you've got. -MC On Sat, Jun 30, 2012 at 5:57 AM, Samira Mh wrote: > hi, > how to get the content of sip header in lua? > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120630/838d07e9/attachment-0001.html From nbhatti at gmail.com Sun Jul 1 01:56:58 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Sun, 1 Jul 2012 00:56:58 +0300 Subject: [Freeswitch-users] Strange issue setting variables for mod_xml_cdr Message-ID: I have gone nuts trying to figure out what is going on. Using xml_cdr to post CDR(s) to webserver, The Bridge is done outside the lua script, and all vars are set in lua and exported. .... session:setVariable("import", "gateway") session:setVariable("import", "admin_rate_group") session:setVariable("import", "sell_rate") session:setVariable("import", "cost_rate") .... The problem is that if I set sell_rate and cost_rate both variables together, who ever is in the end disapears from the XML POST. If I change their order, for example set cost_rate first and then sell_rate, then sell_rate works fine and cost_rate goes off. Looks like some sort of variable overlapping. I am HEAD with latest/latest. freeswitch.consoleLog(FREESWITCH_CONSOLE_LOG_LEVEL, "\n cost_rate ".. cost_rate .." \n"); freeswitch.consoleLog(FREESWITCH_CONSOLE_LOG_LEVEL, "\n sell_rate ".. sell_rate .." \n"); This works fine. There is data in these two variables. I am setting up these session variables along with other variables for the bridge. log-b-leg is false in the config for xml_cdr. http://pastebin.freeswitch.org/19408 is the complete PB for the call log. http://pastebin.freeswitch.org/19409 is the complete XML CDR debug. cdr_csv shows the same "Bhatti",*"2.10000","0.19280"*,"415591","9198","default","2012-06-28 03:32:29","2012-06-28 03:32:29","2012-06-28 03:32:32","3","3","NORMAL_CLEARING","be292014-c0b8-11e1-ae74-2bc60749f960","","","","" "Bhatti",*"2.10000",""*,"415591","9198","default","2012-06-28 03:32:29","2012-06-28 03:32:29","2012-06-28 03:32:32","3","3","NORMAL_CLEARING","be2782fe-c0b8-11e1-ae6f-2bc60749f960","be292014-c0b8-11e1-ae74-2bc60749f960","","","" Sometimes it works sometimes it wont. All the variables are seen in the call log. Am I doing something wrong or we hit some sort of a bug? Thank. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120701/daf518c7/attachment.html From gabe at gundy.org Sun Jul 1 06:26:35 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Sat, 30 Jun 2012 20:26:35 -0600 Subject: [Freeswitch-users] separate channel playback In-Reply-To: <20120614080415.50590@gmx.com> References: <20120614080415.50590@gmx.com> Message-ID: On Thu, Jun 14, 2012 at 2:04 AM, Mi Ke wrote: > Is it possible to select a particular channel (left or right) when I play > stereo audio file to the call leg or conference? Seems like there might be a better way of accomplishing what you're trying to do. What's the use case? Gabe From saami_mh at ymail.com Sun Jul 1 07:38:18 2012 From: saami_mh at ymail.com (Samira Mh) Date: Sat, 30 Jun 2012 20:38:18 -0700 (PDT) Subject: [Freeswitch-users] how to get the content of sip header ? In-Reply-To: References: <1341061052.23871.YahooMailNeo@web120106.mail.ne1.yahoo.com> Message-ID: <1341113898.92837.YahooMailNeo@web120104.mail.ne1.yahoo.com> is it possible to explain in details? ,i have review the wiki on subject but don't understand clearly, i am new on freeswitch, thanks ... ________________________________ From: Michael Collins To: FreeSWITCH Users Help Sent: Sunday, July 1, 2012 1:33 AM Subject: Re: [Freeswitch-users] how to get the content of sip header ? See if you have a bunch of sip_xxx channel variables. Send the call to the info app and see what you've got. -MC On Sat, Jun 30, 2012 at 5:57 AM, Samira Mh wrote: hi, >how to get the content of sip header ?in lua? > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120630/82564f04/attachment.html From peter.olsson at visionutveckling.se Sun Jul 1 11:03:41 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 1 Jul 2012 07:03:41 +0000 Subject: [Freeswitch-users] how to get the content of sip header ? In-Reply-To: <1341113898.92837.YahooMailNeo@web120104.mail.ne1.yahoo.com> References: <1341061052.23871.YahooMailNeo@web120106.mail.ne1.yahoo.com> , <1341113898.92837.YahooMailNeo@web120104.mail.ne1.yahoo.com> Message-ID: Use session:getVariable("var") to read whatever channel variable you need. As Michael said, the SIP headers are stored like sip_header_name. For instance, to read the header Diversion:, use the variable sip_diversion. Michael also mentions the info app, which is a good way to dump all channel variables for a channel, so you know exactly what you have available. To use this, just execute the app info in the dialplan. /Peter 1 jul 2012 kl. 05:47 skrev "Samira Mh" >: is it possible to explain in details? ,i have review the wiki on subject but don't understand clearly, i am new on freeswitch, thanks ... ________________________________ From: Michael Collins > To: FreeSWITCH Users Help > Sent: Sunday, July 1, 2012 1:33 AM Subject: Re: [Freeswitch-users] how to get the content of sip header ? See if you have a bunch of sip_xxx channel variables. Send the call to the info app and see what you've got. -MC On Sat, Jun 30, 2012 at 5:57 AM, Samira Mh > wrote: hi, how to get the content of sip header in lua? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fefc4ee32761372387519! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fefc4ee32761372387519! From gabe at gundy.org Sun Jul 1 11:24:24 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Sun, 1 Jul 2012 01:24:24 -0600 Subject: [Freeswitch-users] Relatively new FreeSwitch user, calls from outside not terminated In-Reply-To: <244FC5DA-2A80-4CA7-9CDD-8C7AF29B2109@cool.de> References: <244FC5DA-2A80-4CA7-9CDD-8C7AF29B2109@cool.de> Message-ID: Post your SIP traffic to the paste bin. Gabe > As soon as the remote party hangs up, siptrace and logfile look as they probably should. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120701/ed5d6b0d/attachment.html From jack.nikolas at ymail.com Sun Jul 1 10:51:23 2012 From: jack.nikolas at ymail.com (Jack Nikolas) Date: Sun, 1 Jul 2012 07:51:23 +0100 (BST) Subject: [Freeswitch-users] Unregister sip user via fs_cli Message-ID: <1341125483.61235.YahooMailNeo@web171501.mail.ir2.yahoo.com> Hi Guys, i am new on freeswitch, please let me know is there any command to? unregister or remove the user who is registering now? using fs_cli ? thanks for help.. ----------------- Regards Jack -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120701/83e503ac/attachment-0001.html From richard.t.ngo at gmail.com Sun Jul 1 11:47:49 2012 From: richard.t.ngo at gmail.com (Richard Ngo) Date: Sun, 1 Jul 2012 02:47:49 -0500 Subject: [Freeswitch-users] Help with google voice Message-ID: hi all, I followed the instruction on http://wiki.freeswitch.org/wiki/Google_Voicejust to test out google voice. I can receive calls if its coming from another gmail account but not using the phone number from the google voice. I also can't call out, my log is on pastebin link below http://pastebin.com/ejb5D7Lk Can anyone please help? Thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120701/1299507a/attachment-0001.html From avi at avimarcus.net Sun Jul 1 12:42:42 2012 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 1 Jul 2012 11:42:42 +0300 Subject: [Freeswitch-users] Unregister sip user via fs_cli In-Reply-To: <1341125483.61235.YahooMailNeo@web171501.mail.ir2.yahoo.com> References: <1341125483.61235.YahooMailNeo@web171501.mail.ir2.yahoo.com> Message-ID: http://wiki.freeswitch.org/wiki/Sofia#Flushing_Inbound_Registrations Just remember if their username/password or ACL is still valid, they can simply re-register. Or are you referring to being flooded with bogus registration attempts? -Avi On Sun, Jul 1, 2012 at 9:51 AM, Jack Nikolas wrote: > Hi Guys, > i am new on freeswitch, > please let me know is there any command to unregister or remove the user > who is registering now using fs_cli ? > thanks for help.. > > ----------------- > Regards Jack > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120701/10d2be1c/attachment.html From nbhatti at gmail.com Sun Jul 1 14:44:42 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Sun, 1 Jul 2012 13:44:42 +0300 Subject: [Freeswitch-users] Strange issue setting variables for mod_xml_cdr In-Reply-To: References: Message-ID: I figured it out. import works with a comma separated list of variables. So instead of defining them each line, line of comma separated works. On Sun, Jul 1, 2012 at 12:56 AM, Muhammad Naseer Bhatti wrote: > > I have gone nuts trying to figure out what is going on. Using xml_cdr to > post CDR(s) to webserver, The Bridge is done outside the lua script, and > all vars are set in lua and exported. > > .... > session:setVariable("import", "gateway") > session:setVariable("import", "admin_rate_group") > session:setVariable("import", "sell_rate") > session:setVariable("import", "cost_rate") > .... > > The problem is that if I set sell_rate and cost_rate both variables > together, who ever is in the end disapears from the XML POST. If I change > their order, for example set cost_rate first and then sell_rate, then > sell_rate works fine and cost_rate goes off. Looks like some sort of > variable overlapping. I am HEAD with latest/latest. > > freeswitch.consoleLog(FREESWITCH_CONSOLE_LOG_LEVEL, "\n cost_rate ".. > cost_rate .." \n"); > freeswitch.consoleLog(FREESWITCH_CONSOLE_LOG_LEVEL, "\n sell_rate ".. > sell_rate .." \n"); > > This works fine. There is data in these two variables. I am setting up > these session variables along with other variables for the bridge. > log-b-leg is false in the config for xml_cdr. > > http://pastebin.freeswitch.org/19408 is the complete PB for the call log. > http://pastebin.freeswitch.org/19409 is the complete XML CDR debug. > > cdr_csv shows the same > > "Bhatti",*"2.10000","0.19280"*,"415591","9198","default","2012-06-28 > 03:32:29","2012-06-28 03:32:29","2012-06-28 > 03:32:32","3","3","NORMAL_CLEARING","be292014-c0b8-11e1-ae74-2bc60749f960","","","","" > > "Bhatti",*"2.10000",""*,"415591","9198","default","2012-06-28 > 03:32:29","2012-06-28 03:32:29","2012-06-28 > 03:32:32","3","3","NORMAL_CLEARING","be2782fe-c0b8-11e1-ae6f-2bc60749f960","be292014-c0b8-11e1-ae74-2bc60749f960","","","" > > Sometimes it works sometimes it wont. All the variables are seen in the > call log. Am I doing something wrong or we hit some sort of a bug? > > > Thank. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120701/8d159476/attachment.html From nbhatti at gmail.com Sun Jul 1 19:33:57 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Sun, 1 Jul 2012 18:33:57 +0300 Subject: [Freeswitch-users] limit getting reset after call state change Message-ID: I am using execute_on_originate to impose limit for outgoing gateway. Limit is set to 1 here, the call is established and right after call state changes, The limit is dropped. Is this normal? 2012-06-28 10:26:35.522664 [NOTICE] switch_channel.c:926 New Channel sofia/vBilling/9198 [97924522-c0f2-11e1-b7cf-a5c0d2d3774a] 2012-06-28 10:26:35.522664 [DEBUG] mod_sofia.c:4775 (sofia/vBilling/9198) State Change CS_NEW -> CS_INIT 2012-06-28 10:26:35.522664 [DEBUG] switch_core_session.c:1229 Send signal sofia/vBilling/9198 [BREAK] EXECUTE sofia/vBilling/9198 limit(hash 10.211.55.5 10.211.55.5 1 !CALL_REJECTED) *2012-06-28 10:26:35.522664 [INFO] switch_limit.c:126 incr called: 10.211.55.5_10.211.55.5 max:1, interval:0* *2012-06-28 10:26:35.522664 [INFO] mod_hash.c:202 Usage for 10.211.55.5_10.211.55.5 is now 1/1* 2012-06-28 10:26:35.522664 [DEBUG] switch_core_state_machine.c:385 (sofia/vBilling/9198) Running State Change CS_INIT 2012-06-28 10:26:35.522664 [DEBUG] switch_core_state_machine.c:424 (sofia/vBilling/9198) State INIT 2012-06-28 10:26:35.522664 [DEBUG] mod_sofia.c:85 sofia/vBilling/9198 SOFIA INIT 2012-06-28 10:26:35.522664 [DEBUG] sofia_glue.c:2602 Local SDP: v=0 o=- 3550145379 3550145379 IN IP4 192.168.40.141 s=pjmedia c=IN IP4 192.168.40.141 t=0 0 a=X-nat:0 m=audio 4014 RTP/AVP 103 102 104 109 3 0 8 9 101 a=rtpmap:103 speex/16000 a=rtpmap:102 speex/8000 a=rtpmap:104 speex/32000 a=rtpmap:109 iLBC/8000 a=fmtp:109 mode=30 a=rtpmap:3 GSM/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtcp:4015 IN IP4 192.168.40.141 2012-06-28 10:26:35.522664 [DEBUG] mod_sofia.c:125 (sofia/vBilling/9198) State Change CS_INIT -> CS_ROUTING *2012-06-28 10:26:35.522664 [INFO] mod_hash.c:304 Usage for 10.211.55.5_10.211.55.5 is now 0* 2012-06-28 10:26:35.522664 [DEBUG] switch_core_session.c:1229 Send signal sofia/vBilling/9198 [BREAK] 2012-06-28 10:26:35.522664 [DEBUG] switch_core_state_machine.c:424 (sofia/vBilling/9198) State INIT going to sleep 2012-06-28 10:26:35.522664 [DEBUG] switch_core_state_machine.c:385 (sofia/vBilling/9198) Running State Change CS_ROUTING 2012-06-28 10:26:35.522664 [DEBUG] switch_channel.c:1919 (sofia/vBilling/9198) Callstate Change DOWN -> RINGING 2012-06-28 10:26:35.522664 [DEBUG] switch_core_state_machine.c:433 (sofia/vBilling/9198) State ROUTING -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120701/b9cd47a6/attachment.html From admin at blindi.net Sun Jul 1 22:07:00 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Sun, 1 Jul 2012 20:07:00 +0200 (CEST) Subject: [Freeswitch-users] I missing channel_variables in eavesdrop In-Reply-To: References: Message-ID: Hi guys, I missing the follwing channelvariables for better handling in eavesdrop: eavesdrop_current_uuid to store the current uuid. eavesdrop_current_key to defined a key. eavesdrop_current_action for diffent actionactions: for example: press 4 to bridge the current channel. Press 5 to mute the channel, and so on. eavesdrop_current_status=uuid_name status of the channel, true of false You can handle the features modular with channelvariable. This solution is very flexiblity. you can split your own keys to different contexs for different permissions. thanks --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From jeff at jefflenk.com Mon Jul 2 00:12:06 2012 From: jeff at jefflenk.com (Jeff Lenk) Date: Sun, 1 Jul 2012 13:12:06 -0700 (PDT) Subject: [Freeswitch-users] Help with google voice In-Reply-To: References: Message-ID: <1341173526708-7580436.post@n2.nabble.com> did you try removing the definition for video from your codec list. gv doest support that. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Help-with-google-voice-tp7580430p7580436.html Sent from the freeswitch-users mailing list archive at Nabble.com. From avi at avimarcus.net Mon Jul 2 01:15:31 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 2 Jul 2012 00:15:31 +0300 Subject: [Freeswitch-users] Leap second & FS using more CPU Message-ID: I was wondering why FS which usually uses only 8-10% CPU at for my 20 active calls was suddenly using 60%. I was pointed to the leap-second being a possible cause. So, as per: http://marc.info/?l=linux-kernel&m=134113577921904 I ran: $ date -s "`date`" and the CPU usage went immediately back to the norm. Not sure if it glitched the audio when doing so, though... This is just a PSA and YMMV. -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120702/1adfae83/attachment-0001.html From robin.gilks at taitradio.com Mon Jul 2 01:43:37 2012 From: robin.gilks at taitradio.com (Robin Gilks) Date: Mon, 2 Jul 2012 09:43:37 +1200 Subject: [Freeswitch-users] Simple SPA8800 FXO connection to FreeSwitch In-Reply-To: References: Message-ID: Spot on!! I'd compiled mod_distributor but not loaded it (two lots of modules.conf!!). I'm getting all the right responses from the cli with the distributor command now, just got to sort out the gateway definitions. Many thanks - I'm moving on now :) On Sat, Jun 30, 2012 at 3:20 AM, Michael Collins wrote: > Go to fs_cli and type: > > distributor cisco_fxo > > If you get an error then you know your distributor is not set up correctly. > If you get "fxolineX" then your dialplan call to distributor is not correct. > Most likely it's the former, and most likely mod_distributor is not compiled > and/or installed. > > In FS source: > Edit modules.conf and uncomment the line with "mod_distributor" > Do "make mod_distributor-install" > > In fs_cli do: > load mod_distributor > > Then go to conf/autoload_configs/modules.conf.xml > Uncomment the line w/ mod_distributor > That will make mod_distributor load each time FS starts. > > -MC > > On Thu, Jun 28, 2012 at 9:22 PM, Robin Gilks > wrote: >> >> Greetings >> >> I'm a noob with FS (and VoIP generally!!) and am really struggling >> with getting 2 way comms between a PABX and a soft-phone using FS and >> an SPA8800. >> >> Calls from the PABX are OK, the FXO ports have registered with FS and >> when I dial the FXO extension I get the dial tone, dial the number for >> the soft-phone and it all works fine. >> >> The problem is connections *TO* the PABX. The SPA8800 AFAIK has a >> separate SIP port for each of its FXO ports. My understanding of how >> it should work is as follows: >> >> I have to have 4 gateways defined in a file in the >> sip-profiles/externals directory, each with a different "sip-port" >> line. I'm testing with 2 to start with: >> >> # cat cicsco_fxo.xml >> >> ? >> ? ? >> ? ? >> ? ? >> ? ? >> ? ? >> ? >> ? >> ? ? >> ? ? >> ? ? >> ? ? >> ? ? >> ? >> >> I have to use mod_distributor to select one of the gateways that >> define the SIP ports. Thats is configured in the autoload_configs >> directory. >> >> # cat distributor.conf.xml >> >> ? >> ? ? >> ? ? ? >> ? ? ? >> ? ? >> ? >> >> >> Finally, I use a dialplan to access this distribution of gateways when >> a 4 digit number starting with 8 is dialed from the soft-phone. >> >> # cat 33_ciscofox.xml >> >> ? >> ? ? >> ? ? ?> data="sofia/gateway/${distributor(cisco_fxo)}/$1"/> >> ? ? >> ? >> >> >> Result expected - it all works. >> Result I get is: >> >> freeswitch at PBX.localdomain> 2012-06-29 04:13:44.287982 [NOTICE] >> switch_channel.c:926 New Channel sofia/internal/1002 at 172.16.164.33 >> [d0fc47e6-c1a0-11e1-881d-f70f560259d2] >> 2012-06-29 04:13:44.287982 [INFO] mod_dialplan_xml.c:485 Processing >> 1002 <1002>->8569 in context default >> 2012-06-29 04:13:44.298584 [ERR] mod_sofia.c:4492 Invalid Gateway '' >> 2012-06-29 04:13:44.298584 [NOTICE] mod_sofia.c:4889 Close Channel N/A >> [CS_NEW] >> 2012-06-29 04:13:44.298584 [NOTICE] switch_ivr_originate.c:2535 Cannot >> create outgoing channel of type [sofia] cause: [INVALID_GATEWAY] >> 2012-06-29 04:13:44.298584 [INFO] mod_dptools.c:2956 Originate Failed. >> ?Cause: INVALID_GATEWAY >> 2012-06-29 04:13:44.298584 [NOTICE] mod_dptools.c:3076 Hangup >> sofia/internal/1002 at 172.16.164.33 [CS_EXECUTE] [INVALID_GATEWAY] >> 2012-06-29 04:13:44.298584 [NOTICE] switch_core_session.c:1447 Session >> 11 (sofia/internal/1002 at 172.16.164.33) Ended >> 2012-06-29 04:13:44.298584 [NOTICE] switch_core_session.c:1449 Close >> Channel sofia/internal/1002 at 172.16.164.33 [CS_DESTROY] >> >> So what is invalid about the gateway? >> >> Cheers >> >> -- >> Robin Gilks >> >> -- > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Robin Gilks -- ------------------------------ This email, including any attachments, is only for the intended recipient. 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The recipient relies upon its own procedures and assumes all risk of use and of opening any attachments. ------------------------------ From fiorix at gmail.com Mon Jul 2 06:09:38 2012 From: fiorix at gmail.com (Alexandre Fiori) Date: Sun, 1 Jul 2012 22:09:38 -0400 Subject: [Freeswitch-users] tls ca setup Message-ID: Bria 3 suddenly stopped working on my mac, reporting this: All accounts failed to enable Account: test could not be enabled. Problem at server, error 503. Try again later. Nothing shows up on fs_cli, but tcpdump shows traffic. Changing sofia to loglevel 9 gives me this: tport_wakeup_pri(0x164dbd0): events IN tport_alloc_secondary(0x164dbd0): new secondary tport 0x7f39b9acce80 tport_tls_accept(0x7f39b9acce80): new connection from tls/x.x.x.x:33351/sips tls_connect(0x7f39b9acce80): events NEGOTIATING tls_connect(0x7f39b9acce80): events NEGOTIATING tls_connect(0x7f39b9acce80): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) tport_close(0x7f39b9acce80): tls/x.x.x.x:33351/sips This is not a happy Canada day, where's my phone? It turns out the self-signed root CA generated by `gentls_cert setup` has expired. How I figured it out? First, on the server: # openssl x509 -noout -in /opt/freeswitch/conf/ssl/CA/cacert.pem -dates notBefore=Jun 2 01:44:26 2012 GMT notAfter=Jul 1 01:44:26 2012 GMT Second, because I opened https://my-server:5061 on Safari and got a "This certificate is not valid (expired root)". It seems the script is missing `-days`, here: http://git.freeswitch.org/git/freeswitch/tree/scripts/gentls_cert.in#n83 Manually adding it fixed the problem. -- Ship, ahoy! Hast seen the White Whale? - Cap'n Ahab -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120701/e1b5070a/attachment.html From bdfoster at endigotech.com Mon Jul 2 07:32:33 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sun, 1 Jul 2012 23:32:33 -0400 Subject: [Freeswitch-users] Leap second & FS using more CPU In-Reply-To: References: Message-ID: Thanks for the notice. I ended up doing a rolling restart of all my servers because of that, so did not end up losing calls but took forever. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 1, 2012 5:16 PM, "Avi Marcus" wrote: > I was wondering why FS which usually uses only 8-10% CPU at for my 20 > active calls was suddenly using 60%. > I was pointed to the leap-second being a possible cause. > > So, as per: http://marc.info/?l=linux-kernel&m=134113577921904 > I ran: $ date -s "`date`" > and the CPU usage went immediately back to the norm. Not sure if it > glitched the audio when doing so, though... > > This is just a PSA and YMMV. > > -Avi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120701/6532cd2e/attachment.html From peter.olsson at visionutveckling.se Mon Jul 2 09:40:12 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 2 Jul 2012 05:40:12 +0000 Subject: [Freeswitch-users] Leap second & FS using more CPU In-Reply-To: References: Message-ID: <6D7A6148-D70A-4321-B788-3C22C4FBAAF2@visionutveckling.se> Avi, thanks for the info. There is also a Jira for this, so I guess we can close that one again, or maybe let it stay open for a couple of days, so it's easier for people to find. http://jira.freeswitch.org/browse/FS-4372 About glitches in audio - since FS uses a monotonic timer it shouldn't be a problem. /Peter 1 jul 2012 kl. 23:23 skrev "Avi Marcus" >: I was wondering why FS which usually uses only 8-10% CPU at for my 20 active calls was suddenly using 60%. I was pointed to the leap-second being a possible cause. So, as per: http://marc.info/?l=linux-kernel&m=134113577921904 I ran: $ date -s "`date`" and the CPU usage went immediately back to the norm. Not sure if it glitched the audio when doing so, though... This is just a PSA and YMMV. -Avi !DSPAM:4ff0bc8d32765450411589! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff0bc8d32765450411589! From joohny at mail.ru Mon Jul 2 11:54:34 2012 From: joohny at mail.ru (=?UTF-8?B?0JXQstCz0LXQvdC40Lk=?=) Date: Mon, 02 Jul 2012 11:54:34 +0400 Subject: [Freeswitch-users] =?utf-8?q?How_about_a_USER_FORUM_and_kill_off_?= =?utf-8?q?the_mail_list=3F?= In-Reply-To: References: Message-ID: <1341215674.681203045@f170.mail.ru> Hi, friends. I decided to make forum by myself. So if someone wants to join - I'll be happy :) I made it in both languages - English and Russian(my language). First of all I plan to find out what structures and threads are needed in forum. Who are interested - welcome to http://freeswitchforum.com Best regards, Evginey. ? ????????? ???????. FreeSWITCH ????? - http://freeswitchforum.com FreeSWITCH user-forum - http://freeswitchforum.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120702/85137d51/attachment.html From manieq at wp.eu Mon Jul 2 17:55:59 2012 From: manieq at wp.eu (Mariusz Czulada) Date: Mon, 02 Jul 2012 15:55:59 +0200 Subject: [Freeswitch-users] RTP not sent while waiting for DTMF in "read" or "play_and_get_digits" Message-ID: <4ff1a86f06b581.44105516@wp.pl> Hi all, I faced a problem with applications mentioned in the subject. Both of them, after playing a prompt message, start to intercept DTMF codes from the endpoint for a specified time. During this period _no_ _RTP_ packets are sent towards calling side, which causes (in my case - ALU IMS) C-BGF to diagnose end of media in channel, which in turn causes P-CSCF to terminate a call. My question: is it possible not to eforce FSW to send 'silence' RTP while waiting for user DTMF actions? TIA, Mariusz From manieq at wp.eu Mon Jul 2 18:12:40 2012 From: manieq at wp.eu (Mariusz Czulada) Date: Mon, 02 Jul 2012 16:12:40 +0200 Subject: [Freeswitch-users] Odp: RTP not sent while waiting for DTMF in "read" or "play_and_get_digits" In-Reply-To: <4ff1a86f06b581.44105516@wp.pl> References: <4ff1a86f06b581.44105516@wp.pl> Message-ID: <4ff1ac584c6021.10163067@wp.pl> Of course, "not" question is: is it possible to eforce FSW to send 'silence' RTP while waiting for user DTMF actions? Mariusz Dnia 2-07-2012 o godz. 15:55 Mariusz Czulada napisa?(a): > Hi all, > > I faced a problem with applications mentioned in the subject. Both of > them, after playing a prompt message, start to intercept DTMF codes from > the endpoint for a specified time. During this period _no_ _RTP_ packets > are sent towards calling side, which causes (in my case - ALU IMS) C-BGF > to diagnose end of media in channel, which in turn causes P-CSCF to > terminate a call. > > My question: is it possible not to eforce FSW to send 'silence' RTP > while waiting for user DTMF actions? > > TIA, > > Mariusz > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From peter.olsson at visionutveckling.se Mon Jul 2 18:25:46 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 2 Jul 2012 14:25:46 +0000 Subject: [Freeswitch-users] RTP not sent while waiting for DTMF in "read" or "play_and_get_digits" Message-ID: <1FFF97C269757C458224B7C895F35F1512AC2F@cantor.std.visionutv.se> This should probably be enough; http://wiki.freeswitch.org/wiki/Variable_send_silence_when_idle Set it to 1400 Also, there is http://wiki.freeswitch.org/wiki/Variable_record_waste_resources, if you want to send RTP during record. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Mariusz Czulada Skickat: den 2 juli 2012 15:56 Till: FreeSWITCH Users ?mne: [Freeswitch-users] RTP not sent while waiting for DTMF in "read" or "play_and_get_digits" Hi all, I faced a problem with applications mentioned in the subject. Both of them, after playing a prompt message, start to intercept DTMF codes from the endpoint for a specified time. During this period _no_ _RTP_ packets are sent towards calling side, which causes (in my case - ALU IMS) C-BGF to diagnose end of media in channel, which in turn causes P-CSCF to terminate a call. My question: is it possible not to eforce FSW to send 'silence' RTP while waiting for user DTMF actions? TIA, Mariusz _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff1a79f32767637819317! From mitch.capper at gmail.com Mon Jul 2 18:26:45 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Mon, 2 Jul 2012 07:26:45 -0700 Subject: [Freeswitch-users] tls ca setup In-Reply-To: References: Message-ID: Thanks will get a patch on JIRA for it. ~Mitch On Sun, Jul 1, 2012 at 7:09 PM, Alexandre Fiori wrote: > > Bria 3 suddenly stopped working on my mac, reporting this: > > ? All accounts failed to enable > > ? Account: test could not be enabled. > ? Problem at server, error 503. Try again later. > > Nothing shows up on fs_cli, but tcpdump shows traffic. Changing sofia to > loglevel 9 gives me this: > > ? tport_wakeup_pri(0x164dbd0): events IN > ? tport_alloc_secondary(0x164dbd0): new secondary tport 0x7f39b9acce80 > ? tport_tls_accept(0x7f39b9acce80): new connection from > tls/x.x.x.x:33351/sips > ? tls_connect(0x7f39b9acce80): events NEGOTIATING > ? tls_connect(0x7f39b9acce80): events NEGOTIATING > ? tls_connect(0x7f39b9acce80): TLS setup failed > (error:00000001:lib(0):func(0):reason(1)) > ? tport_close(0x7f39b9acce80): tls/x.x.x.x:33351/sips > > This is not a happy Canada day, where's my phone? It turns out the > self-signed root CA generated by `gentls_cert setup` has expired. > How I figured it out? First, on the server: > > ? # openssl x509 -noout -in /opt/freeswitch/conf/ssl/CA/cacert.pem -dates > ? notBefore=Jun ?2 01:44:26 2012 GMT > ? notAfter=Jul ?1 01:44:26 2012 GMT > > Second, because I opened https://my-server:5061 on Safari and got a "This > certificate is not valid (expired root)". > > It seems the script is missing `-days`, > here:?http://git.freeswitch.org/git/freeswitch/tree/scripts/gentls_cert.in#n83 > Manually adding it fixed the problem. > > > -- > Ship, ahoy! Hast seen the White Whale? > ??- Cap'n Ahab > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mytemike72 at gmail.com Mon Jul 2 18:27:24 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Mon, 2 Jul 2012 16:27:24 +0200 Subject: [Freeswitch-users] (false detected and) generated dtmf when bridging calls using uuid_bridge on any leg wich uses start_dtmf Message-ID: Hi, Trying again... I have a serious issue with randomly generated dtmf's on bridged calls on where i have a provider wich does not support rfc2833 and I have to use start_dtmf in the dialplan to make dtmf's work. My setup is: A> 1 incomming call, using start_dtmf to activate inband dtmf detection. This makes dtmf work correctly in my (Lua) IVR scripts and respond to digits. (not all the time, but that might be a different issue?) B> This call is at some point in IVR bridged directly from the Lua script to new call on the same switch (no start_dtmf). This script activates a Lua scrpt from the dialplan which does some plays. (so caller A can hear this- as they are bridged) (there is no 'start_dtmf' on this incomming call in the dialplan) C> The Lua script B then triggers a an inbound ESL connection which is generating a completely new session using an api call "originate ... &park()" When C answers the phone, C is bridged to B using api call "uuid_bridge C B" after that B is instructed to end the Lua script to make the bridge work (the bridge works only after the Lua scripts ended) from my ESL connection I have two listeners, one on A and one on C to listen for (dtmf) events. So, there are two bridges, A<->B and B<->C, becuase B is a new incomming session this al works perfectly, and I can do asynchronous stuff from my ESL server and I do need to leave my orignial Lua script A (which is required) The problem: When the incomming call A is using inband detection (start_dtmf in the dialplan) I get a lot (almost always) spontaneous generated dtmf events on the A legg. I am 1000% sure they are not pressed. My ESL listener on A detects them and is showing them in my log. It mostly 'detects' generaly unused dtmfs like 'A' and 'D'. but sometimes also just 'normal' digits. (like '7' in my pastebin) The behaviour leads to hearable dtmf's to C, which of course find it very anoying to hear beeps. They are not heard on the A leg. I can easily reproduce same behaviour on an rfc2833 incomming call and forcing the C leg on inband detection using an "execute_on_answer=start_dtmf" in my originate call. Which is the same setup as my case, but just switched the leggs. Th exact same behaviour will occur but only the dtmf's are detected/generated on the C leg (called user) instead of the A leg (calling user). I have added switchlogs on pastebin: http://pastebin.com/WwW3eDgK for this particulair situation. Please ANY help would be appreciated! I was thinking using for example 'stop_generate_dtmf', to at least make FS stop generating RTP dtmf when (falsely) detecting them. But that doesn't solve the root cause of course, and I have no idea what the consequence might be as that function is not very well documented... Thanks for any help, Regards, Mike. From richard at klingler.net Mon Jul 2 14:50:26 2012 From: richard at klingler.net (Richard Klingler) Date: Mon, 2 Jul 2012 12:50:26 +0200 Subject: [Freeswitch-users] Disable reinvite Message-ID: <20120702125026817891.9461489f@klingler.net> Hello Just switched this weekend from asterisk to freeswitch on freebsd 9.0. And so far I was able to implement basic dialplan for inbound/outbound calls (o; What I'm trying to find is a similar configuration option from asterisk sip.conf where individual user agents can have SIP reinvite disbaled with "canreinvite=no". Fomr the Freeswitch documentation it states that all RTP streams are passed through by default, but I still see SIP clients trying to send UDP streams to my box directly. Also setting "" in sofia.conf.xml doesn't solve this problem. I just want the UDP streams frmo outside are only allowed from my SIP trunk provider. thanx in advance richard From h.maghsoudy at gmail.com Mon Jul 2 17:23:03 2012 From: h.maghsoudy at gmail.com (Hanie Maghsoudy) Date: Mon, 2 Jul 2012 17:53:03 +0430 Subject: [Freeswitch-users] Load Balancing Message-ID: Dear all, I tried OpenSIPs for load balancing. I works just fine, but as it is mentioned here,OpenSIPs sends requests from itself. How about Ultramonkey? Does it change the source too, or it just manipulate destination of registration packets? Thanks, Hanie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120702/b7e9c0a6/attachment.html From ssinyagin at yahoo.com Mon Jul 2 12:45:54 2012 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Mon, 2 Jul 2012 01:45:54 -0700 (PDT) Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail list? In-Reply-To: <1341215674.681203045@f170.mail.ru> References: <1341215674.681203045@f170.mail.ru> Message-ID: <1341218754.83357.YahooMailNeo@web39301.mail.mud.yahoo.com> I would propose concentrating on one language only, probably Russian. There are already many English resourcesavailable, including this mailing list, so why another one? >________________________________ > From: ??????? >To: FreeSWITCH Users Help >Sent: Monday, July 2, 2012 9:54 AM >Subject: Re: [Freeswitch-users] How about a USER FORUM and kill off the mail list? > > >Hi, friends. >I decided to make forum by myself. So if someone wants to join - I'll be happy :) >I made it in both languages - English and Russian(my language). First of all I plan to find out what structures and threads are needed in forum. Who are interested - welcome to http://freeswitchforum.com > > > >Best regards, Evginey. >? ????????? ???????. >FreeSWITCH ????? - http://freeswitchforum.com >FreeSWITCH user-forum - http://freeswitchforum.com > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120702/40e40d47/attachment.html From msc at freeswitch.org Mon Jul 2 18:43:08 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 2 Jul 2012 07:43:08 -0700 Subject: [Freeswitch-users] how to get the content of sip header ? In-Reply-To: References: <1341061052.23871.YahooMailNeo@web120106.mail.ne1.yahoo.com> <1341113898.92837.YahooMailNeo@web120104.mail.ne1.yahoo.com> Message-ID: On Sun, Jul 1, 2012 at 12:03 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Use session:getVariable("var") to read whatever channel variable you need. > As Michael said, the SIP headers are stored like sip_header_name. For > instance, to read the header Diversion:, use the variable sip_diversion. > > Michael also mentions the info app, which is a good way to dump all > channel variables for a channel, so you know exactly what you have > available. To use this, just execute the app info in the dialplan. > Apologies for my brief reply last week - time was short. The quickest way to see what the info dump does is to dial x9192 in the default dialplan and watch the console. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120702/e3b2da13/attachment-0001.html From nasida at live.ru Mon Jul 2 18:57:18 2012 From: nasida at live.ru (Yuriy Nasida) Date: Mon, 2 Jul 2012 18:57:18 +0400 Subject: [Freeswitch-users] Disable reinvite In-Reply-To: <20120702125026817891.9461489f@klingler.net> References: <20120702125026817891.9461489f@klingler.net> Message-ID: You have to use bypass_media for it. http://wiki.freeswitch.org/wiki/Bypass_Media > Date: Mon, 2 Jul 2012 12:50:26 +0200 > From: richard at klingler.net > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] Disable reinvite > > Hello > > Just switched this weekend from asterisk to freeswitch on freebsd 9.0. > And so far I was able to implement basic dialplan for inbound/outbound calls (o; > > What I'm trying to find is a similar configuration option from asterisk sip.conf > where individual user agents can have SIP reinvite disbaled with "canreinvite=no". > > Fomr the Freeswitch documentation it states that all RTP streams are passed through > by default, but I still see SIP clients trying to send UDP streams to my box directly. > > Also setting "" in sofia.conf.xml > doesn't solve this problem. I just want the UDP streams frmo outside are only allowed > from my SIP trunk provider. > > > thanx in advance > richard > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120702/03efeb29/attachment.html From asaad2 at gmail.com Mon Jul 2 19:22:23 2012 From: asaad2 at gmail.com (BookBag) Date: Mon, 2 Jul 2012 10:22:23 -0500 Subject: [Freeswitch-users] Playback of multiple wavs in a round-robin scenario Message-ID: Hello all, I have been searching day and night for a way on how to do this. I have a customer who would like like random ads stored as wav's to be played once an incoming call comes in. For example, this customer has 5 ads each one in its own wav file. If an incoming call comes in, one of those wav's gets played. When a second call comes in, another wav gets selected and gets played. Anybody can help point me in the right direction. Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120702/1ecfaa87/attachment.html From msc at freeswitch.org Mon Jul 2 19:31:45 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 2 Jul 2012 08:31:45 -0700 Subject: [Freeswitch-users] Playback of multiple wavs in a round-robin scenario In-Reply-To: References: Message-ID: Is this part of an IVR? -MC On Mon, Jul 2, 2012 at 8:22 AM, BookBag wrote: > Hello all, I have been searching day and night for a way on how to do > this. I have a customer who would like like random ads stored as wav's to > be played once an incoming call comes in. For example, this customer has 5 > ads each one in its own wav file. If an incoming call comes in, one of > those wav's gets played. When a second call comes in, another wav gets > selected and gets played. Anybody can help point me in the right direction. > > > Thank you. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120702/f078b616/attachment.html From avi at avimarcus.net Mon Jul 2 20:14:35 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 2 Jul 2012 19:14:35 +0300 Subject: [Freeswitch-users] Playback of multiple wavs in a round-robin scenario In-Reply-To: References: Message-ID: Do you need a strict round robin list, or is a: "pick a random one where everything has the same chance of being called" -- if so, http://wiki.freeswitch.org/wiki/Mod_distributor might work for you, setting everything to an equal weight. That page can probably be cleaned up to display it's usefulness... -Avi On Mon, Jul 2, 2012 at 6:31 PM, Michael Collins wrote: > Is this part of an IVR? > -MC > > > On Mon, Jul 2, 2012 at 8:22 AM, BookBag wrote: > >> Hello all, I have been searching day and night for a way on how to do >> this. I have a customer who would like like random ads stored as wav's to >> be played once an incoming call comes in. For example, this customer has 5 >> ads each one in its own wav file. If an incoming call comes in, one of >> those wav's gets played. When a second call comes in, another wav gets >> selected and gets played. Anybody can help point me in the right direction. >> >> >> Thank you. >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120702/98722f22/attachment.html From richard at klingler.net Mon Jul 2 20:40:52 2012 From: richard at klingler.net (Richard Klingler) Date: Mon, 2 Jul 2012 18:40:52 +0200 Subject: [Freeswitch-users] stacksize is too large: run ulimit -s 240 Message-ID: <20120702184052972277.8fd9bcd5@klingler.net> Good evening Seems asterisk is not the only one haviong problems on FreeBSD (o; Starting freeswitch from the shell without "-nc" brings it up fine and operates. But when staring with "-nc" or via the rc.d script it complains about stacksize being too big: Error: stacksize 524288 is too large: run ulimit -s 240 from your shell before starting the application. auto-adjusting stack size for optimal performance... 79103 Backgrounding. It states though it runs then in the background, but it just crashes without any logs. Anyone got freeswitch 1.0.7 running on amd64 FreeBSD 9.0? thanx in advance richard From asaad2 at gmail.com Mon Jul 2 20:45:41 2012 From: asaad2 at gmail.com (BookBag) Date: Mon, 2 Jul 2012 11:45:41 -0500 Subject: [Freeswitch-users] Playback of multiple wavs in a round-robin scenario In-Reply-To: References: Message-ID: no its not part of an ivr On Mon, Jul 2, 2012 at 10:31 AM, Michael Collins wrote: > Is this part of an IVR? > -MC > > > On Mon, Jul 2, 2012 at 8:22 AM, BookBag wrote: > >> Hello all, I have been searching day and night for a way on how to do >> this. I have a customer who would like like random ads stored as wav's to >> be played once an incoming call comes in. For example, this customer has 5 >> ads each one in its own wav file. If an incoming call comes in, one of >> those wav's gets played. When a second call comes in, another wav gets >> selected and gets played. Anybody can help point me in the right direction. >> >> >> Thank you. >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120702/6117a141/attachment-0001.html From saami_mh at ymail.com Mon Jul 2 20:47:23 2012 From: saami_mh at ymail.com (Samira Mh) Date: Mon, 2 Jul 2012 09:47:23 -0700 (PDT) Subject: [Freeswitch-users] how to solve the error Rejected by acl "domains". Falling back to Digest auth? Message-ID: <1341247643.41122.YahooMailNeo@web120105.mail.ne1.yahoo.com> hi guys, please let me know paste myconfigurations as follow: 1- vim /usr/local/freeswitch/conf/directory/default/v_212263612400.xml ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 2-vim /usr/local/freeswitch/conf/autoload_configs/acl.conf.xml ? ?? ? ? ? ? ? ? ?? ? ??? ? ? *max-registrations-per-extension ? =1? when issue the following ?command the user is registered so another user with the extension '2122636124' couldn't register simultaneously : sofia status profile internal ?reg Now what is problem? every time i want to dial some extensions that is configured in my dialplan the following erroe occure: 2012-07-02 21:02:12.050074 [WARNING] sofia_reg.c:1471 SIP auth challenge (REGISTER) on sofia profile 'internal_private' for [212263612400 at 192.168.10.70] from ip 192.168.18.120 2012-07-02 21:02:14.710049 [DEBUG] sofia.c:7904 IP 192.168.18.120 Rejected by acl "domains". Falling back to Digest auth. 2012-07-02 21:02:14.710049 [WARNING] sofia_reg.c:1471 SIP auth challenge (INVITE) on sofia profile 'internal_private' for [00989191949637 at 192.168.10.70] from ip 192.168.18.120 2012-07-02 21:02:14.710049 [DEBUG] sofia.c:7904 IP 192.168.18.120 Rejected by acl "domains". Falling back to Digest auth. 2012-07-02 21:02:14.710049 [WARNING] sofia_reg.c:2607 SIP auth failure (REGISTER) due to reaching max allowed registrations. ?Count: 1 2012-07-02 21:02:14.710049 [WARNING] sofia_reg.c:1416 SIP auth failure (INVITE) on sofia profile 'internal_private' for [00989191949637 at 192.168.10.70] from ip 192.168.18.120 so if i remove the line ?? ?from?/usr/local/freeswitch/conf/directory/default/v_212263612400.xml or from within?/usr/local/freeswitch/conf/sip_profiles/internal.xml(it is posible to defined either? internal.xml? or?/usr/local/freeswitch/conf/directory/default/v_212263612400.xml) the problem solved --with the same settings on?vim /usr/local/freeswitch/conf/autoload_configs/acl.conf.xml--?but??simultaneously? registeration per extension couldn't worked properly ,,, plz help, what is problem on my settings? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120702/c7a4be91/attachment.html From asaad2 at gmail.com Mon Jul 2 20:48:10 2012 From: asaad2 at gmail.com (BookBag) Date: Mon, 2 Jul 2012 11:48:10 -0500 Subject: [Freeswitch-users] Playback of multiple wavs in a round-robin scenario In-Reply-To: References: Message-ID: thanks avi. I can see some really good ways I can use this On Mon, Jul 2, 2012 at 11:14 AM, Avi Marcus wrote: > Do you need a strict round robin list, or is a: "pick a random one where > everything has the same chance of being called" -- if so, > http://wiki.freeswitch.org/wiki/Mod_distributor might work for you, > setting everything to an equal weight. > That page can probably be cleaned up to display it's usefulness... > > -Avi > > > On Mon, Jul 2, 2012 at 6:31 PM, Michael Collins wrote: > >> Is this part of an IVR? >> -MC >> >> >> On Mon, Jul 2, 2012 at 8:22 AM, BookBag wrote: >> >>> Hello all, I have been searching day and night for a way on how to do >>> this. I have a customer who would like like random ads stored as wav's to >>> be played once an incoming call comes in. For example, this customer has 5 >>> ads each one in its own wav file. If an incoming call comes in, one of >>> those wav's gets played. When a second call comes in, another wav gets >>> selected and gets played. Anybody can help point me in the right direction. >>> >>> >>> Thank you. >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120702/d87a846d/attachment.html From peter.olsson at visionutveckling.se Mon Jul 2 20:51:16 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 2 Jul 2012 16:51:16 +0000 Subject: [Freeswitch-users] stacksize is too large: run ulimit -s 240 In-Reply-To: <20120702184052972277.8fd9bcd5@klingler.net> References: <20120702184052972277.8fd9bcd5@klingler.net> Message-ID: <1FFF97C269757C458224B7C895F35F1512AFDE@cantor.std.visionutv.se> Already in Jira, please check out FS-4374 /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Richard Klingler [richard at klingler.net] Skickat: den 2 juli 2012 18:40 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] stacksize is too large: run ulimit -s 240 Good evening Seems asterisk is not the only one haviong problems on FreeBSD (o; Starting freeswitch from the shell without "-nc" brings it up fine and operates. But when staring with "-nc" or via the rc.d script it complains about stacksize being too big: Error: stacksize 524288 is too large: run ulimit -s 240 from your shell before starting the application. auto-adjusting stack size for optimal performance... 79103 Backgrounding. It states though it runs then in the background, but it just crashes without any logs. Anyone got freeswitch 1.0.7 running on amd64 FreeBSD 9.0? thanx in advance richard _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff1cd7c32761386117162! From peter.olsson at visionutveckling.se Mon Jul 2 21:05:59 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 2 Jul 2012 17:05:59 +0000 Subject: [Freeswitch-users] how to solve the error Rejected by acl "domains". Falling back to Digest auth? In-Reply-To: <1341247643.41122.YahooMailNeo@web120105.mail.ne1.yahoo.com> References: <1341247643.41122.YahooMailNeo@web120105.mail.ne1.yahoo.com> Message-ID: <0BC3D6F5-9FC8-4E53-85AD-F2A83B6B26A5@visionutveckling.se> It seems the phone tries to register, even though it is already registered according to FS. So if you really want to use this method, I think you must do further debugging on the phone... As mentioned before on this list, I don't think this is a good approach, and since noone had even heard about this variable before, I'm guessing it's not widely used. However, in this particular example, FS is just doing exactly what it has been told to do, only to allow a registration if it doesn't exist already. /Peter 2 jul 2012 kl. 18:53 skrev "Samira Mh" >: hi guys, please let me know paste myconfigurations as follow: 1- vim /usr/local/freeswitch/conf/directory/default/v_212263612400.xml 2-vim /usr/local/freeswitch/conf/autoload_configs/acl.conf.xml *max-registrations-per-extension =1 when issue the following command the user is registered so another user with the extension '2122636124' couldn't register simultaneously : sofia status profile internal reg Now what is problem? every time i want to dial some extensions that is configured in my dialplan the following erroe occure: 2012-07-02 21:02:12.050074 [WARNING] sofia_reg.c:1471 SIP auth challenge (REGISTER) on sofia profile 'internal_private' for [212263612400 at 192.168.10.70] from ip 192.168.18.120 2012-07-02 21:02:14.710049 [DEBUG] sofia.c:7904 IP 192.168.18.120 Rejected by acl "domains". Falling back to Digest auth. 2012-07-02 21:02:14.710049 [WARNING] sofia_reg.c:1471 SIP auth challenge (INVITE) on sofia profile 'internal_private' for [00989191949637 at 192.168.10.70] from ip 192.168.18.120 2012-07-02 21:02:14.710049 [DEBUG] sofia.c:7904 IP 192.168.18.120 Rejected by acl "domains". Falling back to Digest auth. 2012-07-02 21:02:14.710049 [WARNING] sofia_reg.c:2607 SIP auth failure (REGISTER) due to reaching max allowed registrations. Count: 1 2012-07-02 21:02:14.710049 [WARNING] sofia_reg.c:1416 SIP auth failure (INVITE) on sofia profile 'internal_private' for [00989191949637 at 192.168.10.70] from ip 192.168.18.120 so if i remove the line from /usr/local/freeswitch/conf/directory/default/v_212263612400.xml or from within /usr/local/freeswitch/conf/sip_profiles/internal.xml(it is posible to defined either internal.xml or /usr/local/freeswitch/conf/directory/default/v_212263612400.xml) the problem solved --with the same settings on vim /usr/local/freeswitch/conf/autoload_configs/acl.conf.xml-- but simultaneously registeration per extension couldn't worked properly ,,, plz help, what is problem on my settings? !DSPAM:4ff1cecc32766478410542! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff1cecc32766478410542! From anthony.minessale at gmail.com Mon Jul 2 21:41:01 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Jul 2012 12:41:01 -0500 Subject: [Freeswitch-users] (false detected and) generated dtmf when bridging calls using uuid_bridge on any leg wich uses start_dtmf In-Reply-To: References: Message-ID: You can try the app: 'spandsp_start_dtmf' as an alternative it works better over VoIP. Typically you do not want to do inband dtmf over voip if you can avoid it. You should look for a better provider who can actually work right instead of spending too many engineering cycles trying to work around their shortcomings. On Mon, Jul 2, 2012 at 9:27 AM, Michael Lutz wrote: > Hi, > > Trying again... > I have a serious issue with randomly generated dtmf's on bridged calls > on where i have a provider wich does not support rfc2833 and I have to > use start_dtmf in the dialplan to make dtmf's work. > > My setup is: > A> 1 incomming call, using start_dtmf to activate inband dtmf detection. > This makes dtmf work correctly in my (Lua) IVR scripts and respond to > digits. (not all the time, but that might be a different issue?) > > B> This call is at some point in IVR bridged directly from the Lua > script to new call on the same switch (no start_dtmf). > This script activates a Lua scrpt from the dialplan which does some > plays. (so caller A can hear this- as they are bridged) (there is no > 'start_dtmf' on this incomming call in the dialplan) > > C> The Lua script B then triggers a an inbound ESL connection which is > generating a completely new session using an api call "originate ... > &park()" > When C answers the phone, C is bridged to B using api call > "uuid_bridge C B" after that B is instructed to end the Lua script to > make the bridge work (the bridge works only after the Lua scripts > ended) > > from my ESL connection I have two listeners, one on A and one on C to > listen for (dtmf) events. > > So, there are two bridges, A<->B and B<->C, becuase B is a new > incomming session this al works perfectly, and I can do asynchronous > stuff from my ESL server and I do need to leave my orignial Lua script > A (which is required) > > > The problem: > When the incomming call A is using inband detection (start_dtmf in the > dialplan) I get a lot (almost always) spontaneous generated dtmf > events on the A legg. I am 1000% sure they are not pressed. > My ESL listener on A detects them and is showing them in my log. It > mostly 'detects' generaly unused dtmfs like 'A' and 'D'. but sometimes > also just 'normal' digits. (like '7' in my pastebin) > > The behaviour leads to hearable dtmf's to C, which of course find it > very anoying to hear beeps. They are not heard on the A leg. > > > I can easily reproduce same behaviour on an rfc2833 incomming call and > forcing the C leg on inband detection using an > "execute_on_answer=start_dtmf" in my originate call. Which is the same > setup as my case, but just switched the leggs. > Th exact same behaviour will occur but only the dtmf's are > detected/generated on the C leg (called user) instead of the A leg > (calling user). > > I have added switchlogs on pastebin: http://pastebin.com/WwW3eDgK for > this particulair situation. > > > Please ANY help would be appreciated! > > > I was thinking using for example 'stop_generate_dtmf', to at least > make FS stop generating RTP dtmf when (falsely) detecting them. But > that doesn't solve the root cause of course, and I have no idea what > the consequence might be as that function is not very well > documented... > > > Thanks for any help, > > Regards, > Mike. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From marketing at cluecon.com Mon Jul 2 21:52:58 2012 From: marketing at cluecon.com (Michael Collins) Date: Mon, 2 Jul 2012 10:52:58 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: Welcome to July! We hope the leap second didn't crash any of your servers. Some folks were not so fortunate . On last week's conference call we had an open discussion on topics chosen by members of the community. One topic that got a fair amount of traction was how to address NAT when FreeSWITCH has a public IP address and the clients themselves are behind NAT. Dave Kompel shared his strategy of using "aggressive-nat-detection" on the SIP profile and then optionally setting "NDLB-connectile-dysfunction" for the directory entry. We also discussed other topics like IPv6 and using FreeSWITCH in the cloud with Amazon. The audio is available here . Darren Schreiber's company, 2600hz, made a very interesting announcementregarding "Kazoo BETA." This new cloud-based PBX solution uses a lot of open source software under the hood, including FreeSWITCH. Stay tuned for a more in-depth discussion with Darren about this new platform. On the ClueCon front we are happy to report that many have taken advantage of the fact that they can get 2^4 tickets (i.e. 16 tickets) for the drawings at ClueCon 2012. The bits will shift at the end of the day tomorrow, July 3, so don't delay! You can register quickly and easily hereon the ClueCon site. After July 3rd registrants will receive 2^3 or 8 tickets. Don't delay - we are giving away a total of ten 7" Android tablets among many other prizes supplied by our sponsors . Only 35 days until ClueCon 2012! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0702 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120702/4008baf3/attachment.html From ssinyagin at yahoo.com Mon Jul 2 23:05:10 2012 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Mon, 2 Jul 2012 12:05:10 -0700 (PDT) Subject: [Freeswitch-users] Playback of multiple wavs in a round-robin scenario In-Reply-To: References: Message-ID: <1341255910.24027.YahooMailNeo@web39302.mail.mud.yahoo.com> you can pass the call control to a Lua or Javascript script, and then calculate the file number and execute the playback. The easiest is to take the current time in seconds, take a modulo by the total number of your WAV files, and then play the corresponding file. Should be a quite trivial exercise for anyone having basic programming skills :) >________________________________ > From: BookBag >To: freeswitch-users at lists.freeswitch.org >Sent: Monday, July 2, 2012 5:22 PM >Subject: [Freeswitch-users] Playback of multiple wavs in a round-robin scenario > > >Hello all, I have been searching day and night for a way on how to do this. I have a customer who would like like random ads stored as wav's? to be played once an incoming call comes in. For example, this customer has 5 ads each one in its own wav file. If an incoming call comes in, one of those wav's gets played. When a second call comes in, another wav gets selected and gets played. Anybody can help point me in the right direction. > > >Thank you. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120702/117f0162/attachment.html From ben at langfeld.co.uk Tue Jul 3 01:29:02 2012 From: ben at langfeld.co.uk (Ben Langfeld) Date: Mon, 2 Jul 2012 22:29:02 +0100 Subject: [Freeswitch-users] Inbound EventSocket anomoly Message-ID: Gents, I was hoping someone would be able to clear up what appears to be an obvious omission from IES. If I execute the playback application on a channel via IES, I get an immediate response to indicate it was queued, and I then have to wait for either PLAYBACK_STOP or CHANNEL_EXECUTE_COMPLETE. The problem is, the response contains nothing to filter events by. The only link I have are the 'Application' and 'Application-Data' fields on the event, and there is no guarantee those are unique to an individual application execution. I would expect the immediate response to return a UUID which would be present on the related events. Am I missing something, or is this actually a bug? Example: https://gist.github.com/40f079ae2a079abe6393 Regards, Ben Langfeld From anthony.minessale at gmail.com Tue Jul 3 02:01:46 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 2 Jul 2012 17:01:46 -0500 Subject: [Freeswitch-users] Inbound EventSocket anomoly In-Reply-To: References: Message-ID: just something nobody has needed. I added some stuff to tree have a look at latest git. On Mon, Jul 2, 2012 at 4:29 PM, Ben Langfeld wrote: > Gents, > > I was hoping someone would be able to clear up what appears to be an > obvious omission from IES. > > If I execute the playback application on a channel via IES, I get an > immediate response to indicate it was queued, and I then have to wait > for either PLAYBACK_STOP or CHANNEL_EXECUTE_COMPLETE. The problem is, > the response contains nothing to filter events by. The only link I > have are the 'Application' and 'Application-Data' fields on the event, > and there is no guarantee those are unique to an individual > application execution. I would expect the immediate response to return > a UUID which would be present on the related events. > > Am I missing something, or is this actually a bug? > > Example: https://gist.github.com/40f079ae2a079abe6393 > > Regards, > Ben Langfeld > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From marketing at cluecon.com Tue Jul 3 03:46:49 2012 From: marketing at cluecon.com (Michael Collins) Date: Mon, 2 Jul 2012 16:46:49 -0700 Subject: [Freeswitch-users] Reminder: Last chance for 16 tickets before the bits shift! Message-ID: Just a friendly reminder: July 3rd is the last day to get all 16 tickets in the great ClueCon giveway. Be sure to get registeredby the end of the day on Tuesday, July 3rd to maximize your chance to win big! Looking forward to seeing you all in Chicago next month. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120702/840c814b/attachment-0001.html From saami_mh at ymail.com Tue Jul 3 07:48:33 2012 From: saami_mh at ymail.com (Samira Mh) Date: Mon, 2 Jul 2012 20:48:33 -0700 (PDT) Subject: [Freeswitch-users] how to solve the error Rejected by acl "domains". Falling back to Digest auth? In-Reply-To: <0BC3D6F5-9FC8-4E53-85AD-F2A83B6B26A5@visionutveckling.se> References: <1341247643.41122.YahooMailNeo@web120105.mail.ne1.yahoo.com> <0BC3D6F5-9FC8-4E53-85AD-F2A83B6B26A5@visionutveckling.se> Message-ID: <1341287313.55682.YahooMailNeo@web120104.mail.ne1.yahoo.com> hi Peter, this feature(max-registrations-per-extension)?is embeded in the sofia_reg.c as follow : ? ............ if (max_registrations_perext > 0 && (sip && sip->sip_contact && (sip->sip_contact->m_expires == NULL || atol(sip->sip_contact->m_expires) > 0))) { ? ? ? ? ? ? ? ? /* if expires is null still process */ ? ? ? ? ? ? ? ? /* expires == 0 means the phone is going to unregiser, so don't count against max */ ? ? ? ? ? ? ? ? uint32_t count = 0; ? ? ? ? ? ? ? ? call_id = sip->sip_call_id->i_id; ? ? ? ? ? ? ? ? switch_assert(call_id); ? ? ? ? ? ? ? ? sql = switch_mprintf("select count(sip_user) from sip_registrations where sip_user='%q' AND call_id <> '%q'", username, call_id); ? ? ? ? ? ? ? ? switch_assert(sql != NULL); ? ? ? ? ? ? ? ? sofia_glue_execute_sql_callback(profile, NULL, sql, sofia_reg_regcount_callback, &count); ? ? ? ? ? ? ? ? free(sql); ? ? ? ? ? ? ? ? if (count + 1> max_registrations_perext) { ? ? ? ? ? ? ? ? ? ? ? ? ret = AUTH_FORBIDDEN; ? ? ? ? ? ? ? ? ? ? ? ? if (sofia_test_pflag(profile, PFLAG_LOG_AUTH_FAIL)) { ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? "SIP auth failure (REGISTER) due to reaching max allowed registrations. ?Count: %d\n", count); ? ? ? ? ? ? ? ? ? ? ? ? } ? ? ? ? ? ? ? ? ? ? ? ? goto end; ? ? ? ? ? ? ? ? } ? ? ? ? } ............. ________________________________ From: Peter Olsson To: FreeSWITCH Users Help Sent: Monday, July 2, 2012 9:35 PM Subject: Re: [Freeswitch-users] how to solve the error Rejected by acl "domains". Falling back to Digest auth? It seems the phone tries to register, even though it is already registered according to FS. So if you really want to use this method, I think you must do further debugging on the phone... As mentioned before on this list, I don't think this is a good approach, and since noone had even heard about this variable before, I'm guessing it's not widely used. However, in this particular example, FS is just doing exactly what it has been told to do, only to allow a registration if it doesn't exist already. /Peter 2 jul 2012 kl. 18:53 skrev "Samira Mh" >: hi guys, please let me know paste myconfigurations as follow: 1- vim /usr/local/freeswitch/conf/directory/default/v_212263612400.xml ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 2-vim /usr/local/freeswitch/conf/autoload_configs/acl.conf.xml ? ? ? ? ? ? ? ? ? ? *max-registrations-per-extension? =1 when issue the following? command the user is registered so another user with the extension '2122636124' couldn't register simultaneously : sofia status profile internal? reg Now what is problem? every time i want to dial some extensions that is configured in my dialplan the following erroe occure: 2012-07-02 21:02:12.050074 [WARNING] sofia_reg.c:1471 SIP auth challenge (REGISTER) on sofia profile 'internal_private' for [212263612400 at 192.168.10.70] from ip 192.168.18.120 2012-07-02 21:02:14.710049 [DEBUG] sofia.c:7904 IP 192.168.18.120 Rejected by acl "domains". Falling back to Digest auth. 2012-07-02 21:02:14.710049 [WARNING] sofia_reg.c:1471 SIP auth challenge (INVITE) on sofia profile 'internal_private' for [00989191949637 at 192.168.10.70] from ip 192.168.18.120 2012-07-02 21:02:14.710049 [DEBUG] sofia.c:7904 IP 192.168.18.120 Rejected by acl "domains". Falling back to Digest auth. 2012-07-02 21:02:14.710049 [WARNING] sofia_reg.c:2607 SIP auth failure (REGISTER) due to reaching max allowed registrations.? Count: 1 2012-07-02 21:02:14.710049 [WARNING] sofia_reg.c:1416 SIP auth failure (INVITE) on sofia profile 'internal_private' for [00989191949637 at 192.168.10.70] from ip 192.168.18.120 so if i remove the line? ? from /usr/local/freeswitch/conf/directory/default/v_212263612400.xml or from within /usr/local/freeswitch/conf/sip_profiles/internal.xml(it is posible to defined either internal.xml? or /usr/local/freeswitch/conf/directory/default/v_212263612400.xml) the problem solved --with the same settings on vim /usr/local/freeswitch/conf/autoload_configs/acl.conf.xml-- but? simultaneously? registeration per extension couldn't worked properly ,,, plz help, what is problem on my settings? !DSPAM:4ff1cecc32766478410542! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff1cecc32766478410542! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120702/9bbf4288/attachment.html From h.maghsoudy at gmail.com Tue Jul 3 08:39:13 2012 From: h.maghsoudy at gmail.com (Hanie Maghsoudy) Date: Tue, 3 Jul 2012 09:09:13 +0430 Subject: [Freeswitch-users] Load Balancing In-Reply-To: References: Message-ID: I think I got it! it's explained herein packet forwarding section. Changing the source won't happen. On Mon, Jul 2, 2012 at 5:53 PM, Hanie Maghsoudy wrote: > Dear all, > > I tried OpenSIPs for load balancing. I works just fine, but as it is > mentioned here,OpenSIPs sends requests from itself. > How about Ultramonkey? Does it change the source too, or it just > manipulate destination of registration packets? > > Thanks, > Hanie > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/a82dd359/attachment.html From avi at avimarcus.net Tue Jul 3 09:04:35 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 3 Jul 2012 08:04:35 +0300 Subject: [Freeswitch-users] Multiple 183 SDP Change Codecs - allowed? Message-ID: Are multiple 183s from the endpoint that changes the SDP allowed? I'm under the impression this is broken, similar to http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp ... which is why when the codec changes, FS freaks out and cancels the call because of codec negotiation error. Does that NDLB flag allow this, too? First has: audio 10116 RTP/AVP *0* 101 13 Second has: audio 49020 RTP/AVP *8* 13 101 PCAP: http://ge.tt/7MpyBwJ Can someone point me to the specific RFC so I can tell the supplier to fix it? And just curious.. what would make this allowed? A re-INVITE..? -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/0900297f/attachment-0001.html From richard.t.ngo at gmail.com Tue Jul 3 06:54:52 2012 From: richard.t.ngo at gmail.com (Richard Ngo) Date: Mon, 2 Jul 2012 21:54:52 -0500 Subject: [Freeswitch-users] Help with google voice In-Reply-To: <1341173526708-7580436.post@n2.nabble.com> References: <1341173526708-7580436.post@n2.nabble.com> Message-ID: wow man you are my hero. Just in case someone else in the future faces the same issue, they need to remove h264 from conf/autoload_configs/dingaling.conf.xml and just leave PCMU. Hope they add this on their guide. Again, thanks bro On Sun, Jul 1, 2012 at 3:12 PM, Jeff Lenk wrote: > did you try removing the definition for video from your codec list. gv > doest > support that. > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Help-with-google-voice-tp7580430p7580436.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120702/a06d1618/attachment.html From peter.olsson at visionutveckling.se Tue Jul 3 09:28:29 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 3 Jul 2012 05:28:29 +0000 Subject: [Freeswitch-users] how to solve the error Rejected by acl "domains". Falling back to Digest auth? In-Reply-To: <1341287313.55682.YahooMailNeo@web120104.mail.ne1.yahoo.com> References: <1341247643.41122.YahooMailNeo@web120105.mail.ne1.yahoo.com> <0BC3D6F5-9FC8-4E53-85AD-F2A83B6B26A5@visionutveckling.se>, <1341287313.55682.YahooMailNeo@web120104.mail.ne1.yahoo.com> Message-ID: <1FFF97C269757C458224B7C895F35F1512B642@cantor.std.visionutv.se> Yes, I know. That code does exactly what you want. It's the feature itself that is more questionable... Personally I don't believe this will ever work, since there are so many possibilities for timing issues for the registration. As MC said earlier on this discussion, you need to rethink if this is something you really want. When it comes to security this will not, in any way, increase the security for you. If you're unlucky it might do the opposite, leaving the phone registered to the "theif", and leave the "real" user blocked from registering. Keep your accounts secure, and this shouldn't be needed. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Samira Mh [saami_mh at ymail.com] Skickat: den 3 juli 2012 05:48 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] how to solve the error Rejected by acl "domains". Falling back to Digest auth? hi Peter, this feature(max-registrations-per-extension) is embeded in the sofia_reg.c as follow : ............ if (max_registrations_perext > 0 && (sip && sip->sip_contact && (sip->sip_contact->m_expires == NULL || atol(sip->sip_contact->m_expires) > 0))) { /* if expires is null still process */ /* expires == 0 means the phone is going to unregiser, so don't count against max */ uint32_t count = 0; call_id = sip->sip_call_id->i_id; switch_assert(call_id); sql = switch_mprintf("select count(sip_user) from sip_registrations where sip_user='%q' AND call_id <> '%q'", username, call_id); switch_assert(sql != NULL); sofia_glue_execute_sql_callback(profile, NULL, sql, sofia_reg_regcount_callback, &count); free(sql); if (count + 1> max_registrations_perext) { ret = AUTH_FORBIDDEN; if (sofia_test_pflag(profile, PFLAG_LOG_AUTH_FAIL)) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "SIP auth failure (REGISTER) due to reaching max allowed registrations. Count: %d\n", count); } goto end; } } ............. ________________________________ From: Peter Olsson To: FreeSWITCH Users Help Sent: Monday, July 2, 2012 9:35 PM Subject: Re: [Freeswitch-users] how to solve the error Rejected by acl "domains". Falling back to Digest auth? It seems the phone tries to register, even though it is already registered according to FS. So if you really want to use this method, I think you must do further debugging on the phone... As mentioned before on this list, I don't think this is a good approach, and since noone had even heard about this variable before, I'm guessing it's not widely used. However, in this particular example, FS is just doing exactly what it has been told to do, only to allow a registration if it doesn't exist already. /Peter 2 jul 2012 kl. 18:53 skrev "Samira Mh" >>: hi guys, please let me know paste myconfigurations as follow: 1- vim /usr/local/freeswitch/conf/directory/default/v_212263612400.xml 2-vim /usr/local/freeswitch/conf/autoload_configs/acl.conf.xml *max-registrations-per-extension =1 when issue the following command the user is registered so another user with the extension '2122636124' couldn't register simultaneously : sofia status profile internal reg Now what is problem? every time i want to dial some extensions that is configured in my dialplan the following erroe occure: 2012-07-02 21:02:12.050074 [WARNING] sofia_reg.c:1471 SIP auth challenge (REGISTER) on sofia profile 'internal_private' for [212263612400 at 192.168.10.70] from ip 192.168.18.120 2012-07-02 21:02:14.710049 [DEBUG] sofia.c:7904 IP 192.168.18.120 Rejected by acl "domains". Falling back to Digest auth. 2012-07-02 21:02:14.710049 [WARNING] sofia_reg.c:1471 SIP auth challenge (INVITE) on sofia profile 'internal_private' for [00989191949637 at 192.168.10.70] from ip 192.168.18.120 2012-07-02 21:02:14.710049 [DEBUG] sofia.c:7904 IP 192.168.18.120 Rejected by acl "domains". Falling back to Digest auth. 2012-07-02 21:02:14.710049 [WARNING] sofia_reg.c:2607 SIP auth failure (REGISTER) due to reaching max allowed registrations. Count: 1 2012-07-02 21:02:14.710049 [WARNING] sofia_reg.c:1416 SIP auth failure (INVITE) on sofia profile 'internal_private' for [00989191949637 at 192.168.10.70] from ip 192.168.18.120 so if i remove the line from /usr/local/freeswitch/conf/directory/default/v_212263612400.xml or from within /usr/local/freeswitch/conf/sip_profiles/internal.xml(it is posible to defined either internal.xml or /usr/local/freeswitch/conf/directory/default/v_212263612400.xml) the problem solved --with the same settings on vim /usr/local/freeswitch/conf/autoload_configs/acl.conf.xml-- but simultaneously registeration per extension couldn't worked properly ,,, plz help, what is problem on my settings? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff1cecc32766478410542! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff26ae632762054019490! From peter.olsson at visionutveckling.se Tue Jul 3 09:34:37 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 3 Jul 2012 05:34:37 +0000 Subject: [Freeswitch-users] Inbound EventSocket anomoly In-Reply-To: References: Message-ID: <1FFF97C269757C458224B7C895F35F1512B653@cantor.std.visionutv.se> I saw Tony added a new patch for you. However, even before this it was actually possible. For playback specific features, you could use playback local vars that will be sent back on PLAYBACK_START and PLAYBACK_STOP. More info here; http://wiki.freeswitch.org/wiki/Mod_playback#Example_for_specific_playback_variables For a more generic approach for CHANNEL_EXECUTE_COMPLETE, it is also possible to use scoped variables, as mentioned here; http://wiki.freeswitch.org/wiki/Channel_Variables#Scoped_Variables The difference with these approaches is that you (on the ESL client side) can decide a unique value. Tonys patch cretes a UUID for you, and uses that on the CHANNEL_EXECUTE_COMPLETE event. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Ben Langfeld [ben at langfeld.co.uk] Skickat: den 2 juli 2012 23:29 Till: FreeSWITCH Users Help Cc: Ben Klang ?mne: [Freeswitch-users] Inbound EventSocket anomoly Gents, I was hoping someone would be able to clear up what appears to be an obvious omission from IES. If I execute the playback application on a channel via IES, I get an immediate response to indicate it was queued, and I then have to wait for either PLAYBACK_STOP or CHANNEL_EXECUTE_COMPLETE. The problem is, the response contains nothing to filter events by. The only link I have are the 'Application' and 'Application-Data' fields on the event, and there is no guarantee those are unique to an individual application execution. I would expect the immediate response to return a UUID which would be present on the related events. Am I missing something, or is this actually a bug? Example: https://gist.github.com/40f079ae2a079abe6393 Regards, Ben Langfeld _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff2113932762747716525! From mel0torme at gmail.com Tue Jul 3 09:38:52 2012 From: mel0torme at gmail.com (Tom C) Date: Mon, 2 Jul 2012 22:38:52 -0700 Subject: [Freeswitch-users] How to add profiler flag when making just one module Message-ID: I'd like to build one module (mod_fifo) with the -pg compiler flag. But my inexperience with gcc makefiles is making it difficult. :-) Does anyone have any info for doing profiling in a FreeSwitch module? The only way I can get it to even compile with the profiling flag is to run ./configure to set the flag for ALL modules, then make mod_fifo, then re-run configure to remove the flag. My code: $> export MOD_CFLAGS="-pg" $> export CFLAGS="-pg" $> ./configure $> make mod_fifo $> # undo the changes so I don't screw up my entire build. $> export MOD_CFLAGS= $> export CFLAGS= $> ../configure Unfortunately, ./configure takes about 40 minutes to run on my poor little pogoplug, and then "make mod_fifo" feels the need to rebuild a bunch of libraries with the -pg flag (which I don't want), which adds another 15 minutes. In all, my script takes 105 minutes to run. After all this, I can see the -pg flag in use when doing "make mod_fifo", and -pg shows up in the relink_command in mod_fifo.la. But it doesn't create the profiling data file, so apparently the -pg didn't actually work. I'm stumped. :-) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120702/1307f3a6/attachment.html From saami_mh at ymail.com Tue Jul 3 09:47:38 2012 From: saami_mh at ymail.com (Samira Mh) Date: Mon, 2 Jul 2012 22:47:38 -0700 (PDT) Subject: [Freeswitch-users] how to solve the error Rejected by acl "domains". Falling back to Digest auth? In-Reply-To: <1FFF97C269757C458224B7C895F35F1512B642@cantor.std.visionutv.se> References: <1341247643.41122.YahooMailNeo@web120105.mail.ne1.yahoo.com> <0BC3D6F5-9FC8-4E53-85AD-F2A83B6B26A5@visionutveckling.se>, <1341287313.55682.YahooMailNeo@web120104.mail.ne1.yahoo.com> <1FFF97C269757C458224B7C895F35F1512B642@cantor.std.visionutv.se> Message-ID: <1341294458.85684.YahooMailNeo@web120104.mail.ne1.yahoo.com> i have to limit the count of registerations because that feature is??exactly?what my manager want !:( so i must to implement it correctly ... ________________________________ From: Peter Olsson To: FreeSWITCH Users Help Sent: Tuesday, July 3, 2012 9:58 AM Subject: Re: [Freeswitch-users] how to solve the error Rejected by acl "domains". Falling back to Digest auth? Yes, I know. That code does exactly what you want. It's the feature itself that is more questionable... Personally I don't believe this will ever work, since there are so many possibilities for timing issues for the registration. As MC said earlier on this discussion, you need to rethink if this is something you really want. When it comes to security this will not, in any way, increase the security for you. If you're unlucky it might do the opposite, leaving the phone registered to the "theif", and leave the "real" user blocked from registering. Keep your accounts secure, and this shouldn't be needed. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Samira Mh [saami_mh at ymail.com] Skickat: den 3 juli 2012 05:48 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] how to solve the error Rejected by acl "domains". Falling back to Digest auth? hi Peter, this feature(max-registrations-per-extension) is embeded in the sofia_reg.c as follow : ............ if (max_registrations_perext > 0 && (sip && sip->sip_contact && (sip->sip_contact->m_expires == NULL || atol(sip->sip_contact->m_expires) > 0))) { ? ? ? ? ? ? ? ? /* if expires is null still process */ ? ? ? ? ? ? ? ? /* expires == 0 means the phone is going to unregiser, so don't count against max */ ? ? ? ? ? ? ? ? uint32_t count = 0; ? ? ? ? ? ? ? ? call_id = sip->sip_call_id->i_id; ? ? ? ? ? ? ? ? switch_assert(call_id); ? ? ? ? ? ? ? ? sql = switch_mprintf("select count(sip_user) from sip_registrations where sip_user='%q' AND call_id <> '%q'", username, call_id); ? ? ? ? ? ? ? ? switch_assert(sql != NULL); ? ? ? ? ? ? ? ? sofia_glue_execute_sql_callback(profile, NULL, sql, sofia_reg_regcount_callback, &count); ? ? ? ? ? ? ? ? free(sql); ? ? ? ? ? ? ? ? if (count + 1> max_registrations_perext) { ? ? ? ? ? ? ? ? ? ? ? ? ret = AUTH_FORBIDDEN; ? ? ? ? ? ? ? ? ? ? ? ? if (sofia_test_pflag(profile, PFLAG_LOG_AUTH_FAIL)) { ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? "SIP auth failure (REGISTER) due to reaching max allowed registrations.? Count: %d\n", count); ? ? ? ? ? ? ? ? ? ? ? ? } ? ? ? ? ? ? ? ? ? ? ? ? goto end; ? ? ? ? ? ? ? ? } ? ? ? ? } ............. ________________________________ From: Peter Olsson To: FreeSWITCH Users Help Sent: Monday, July 2, 2012 9:35 PM Subject: Re: [Freeswitch-users] how to solve the error Rejected by acl "domains". Falling back to Digest auth? It seems the phone tries to register, even though it is already registered according to FS. So if you really want to use this method, I think you must do further debugging on the phone... As mentioned before on this list, I don't think this is a good approach, and since noone had even heard about this variable before, I'm guessing it's not widely used. However, in this particular example, FS is just doing exactly what it has been told to do, only to allow a registration if it doesn't exist already. /Peter 2 jul 2012 kl. 18:53 skrev "Samira Mh" >>: hi guys, please let me know paste myconfigurations as follow: 1- vim /usr/local/freeswitch/conf/directory/default/v_212263612400.xml ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? 2-vim /usr/local/freeswitch/conf/autoload_configs/acl.conf.xml ? ? ? ? ? ? ? ? ? ? *max-registrations-per-extension? =1 when issue the following? command the user is registered so another user with the extension '2122636124' couldn't register simultaneously : sofia status profile internal? reg Now what is problem? every time i want to dial some extensions that is configured in my dialplan the following erroe occure: 2012-07-02 21:02:12.050074 [WARNING] sofia_reg.c:1471 SIP auth challenge (REGISTER) on sofia profile 'internal_private' for [212263612400 at 192.168.10.70] from ip 192.168.18.120 2012-07-02 21:02:14.710049 [DEBUG] sofia.c:7904 IP 192.168.18.120 Rejected by acl "domains". Falling back to Digest auth. 2012-07-02 21:02:14.710049 [WARNING] sofia_reg.c:1471 SIP auth challenge (INVITE) on sofia profile 'internal_private' for [00989191949637 at 192.168.10.70] from ip 192.168.18.120 2012-07-02 21:02:14.710049 [DEBUG] sofia.c:7904 IP 192.168.18.120 Rejected by acl "domains". Falling back to Digest auth. 2012-07-02 21:02:14.710049 [WARNING] sofia_reg.c:2607 SIP auth failure (REGISTER) due to reaching max allowed registrations.? Count: 1 2012-07-02 21:02:14.710049 [WARNING] sofia_reg.c:1416 SIP auth failure (INVITE) on sofia profile 'internal_private' for [00989191949637 at 192.168.10.70] from ip 192.168.18.120 so if i remove the line? ? from /usr/local/freeswitch/conf/directory/default/v_212263612400.xml or from within /usr/local/freeswitch/conf/sip_profiles/internal.xml(it is posible to defined either internal.xml? or /usr/local/freeswitch/conf/directory/default/v_212263612400.xml) the problem solved --with the same settings on vim /usr/local/freeswitch/conf/autoload_configs/acl.conf.xml-- but? simultaneously? registeration per extension couldn't worked properly ,,, plz help, what is problem on my settings? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff1cecc32766478410542! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff26ae632762054019490! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120702/17c7c1c2/attachment-0001.html From edwin.hoexum at office.ziggo.nl Tue Jul 3 13:34:20 2012 From: edwin.hoexum at office.ziggo.nl (Hoexum, Edwin) Date: Tue, 3 Jul 2012 11:34:20 +0200 Subject: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a In-Reply-To: References: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern> <4FED4DA1.10304@belrosbank.by> Message-ID: <2C18241C6768CF469C7E432A25327AD51352973F9E@MAIL-WPV01.office.intern> Thanks for your response. I think the box is good the OS ubuntu 64bit is ok 10.04 (little bit old but still). Interrupts are also fine traffic is on eth1 ? Can load balancing the RTP/SIP over more eth interfaces give better performance. ? The key point is that the box is not very busy but I introduce Delay in the RTP more than 100ms on 20% of the packets I am very impressed with this box flexibily and feature rich. If I can upscale the RTP sessions and can prove that it is the hardware/Os that is causing the limits I can look for a solution for that. root at mnd-rc0001-srvot06:~# cat /proc/interrupts CPU0 CPU1 CPU2 CPU3 CPU4 CPU5 CPU6 CPU7 CPU8 CPU9 CPU10 CPU11 0: 69683 0 0 0 0 0 0 0 0 0 0 0 IO-APIC-edge timer 1: 2 0 0 0 0 0 0 0 0 0 0 0 IO-APIC-edge i8042 8: 1 0 0 0 0 0 0 0 0 0 0 0 IO-APIC-edge rtc0 9: 0 0 0 0 0 0 0 0 0 0 0 0 IO-APIC-fasteoi acpi 12: 4 0 0 0 0 0 0 0 0 0 0 0 IO-APIC-edge i8042 17: 117 0 0 0 0 0 0 0 0 0 0 0 IO-APIC-fasteoi ata_piix, uhci_hcd:usb6, hpilo 20: 0 0 0 0 0 0 0 0 0 0 0 0 IO-APIC-fasteoi ehci_hcd:usb1, uhci_hcd:usb2 22: 0 0 0 0 0 0 0 0 0 0 0 0 IO-APIC-fasteoi uhci_hcd:usb4 23: 0 0 0 0 0 0 0 0 0 0 0 0 IO-APIC-fasteoi uhci_hcd:usb3, uhci_hcd:usb5 63: 7959851 0 0 0 0 0 0 0 0 0 0 0 PCI-MSI-edge cciss0 65: 209078530 0 564128112 27706399 791204941 23 9322 6045075 1592 0 0 268 PCI-MSI-edge eth0-0 66: 137960307 0 1582289 0 4940 298816564 10884351 10426 0 186127 0 0 PCI-MSI-edge eth0-1 67: 915 0 9646 0 394 2 73497 1854844 490614 0 576 0 PCI-MSI-edge eth0-2 68: 874899 0 794424 0 0 0 323647 269982 0 0 18 6493 PCI-MSI-edge eth0-3 69: 2669 0 422 42 6590154 3406669 0 0 0 30 0 903 PCI-MSI-edge eth0-4 70: 41932 0 0 0 0 0 495253 30756 407 0 95 1201810 PCI-MSI-edge eth0-5 71: 560 0 0 0 0 0 1010 0 124 1880079 832733 0 PCI-MSI-edge eth0-6 72: 464 0 0 0 0 0 0 0 0 0 1237460 590278 PCI-MSI-edge eth0-7 90: 47515 342579 29 0 0 26 10993393 636 229587566 309472596 0 0 PCI-MSI-edge eth1-0 91: 53030059 0 75345905 17924308 0 0 50523045 122821 99843488 0 0 0 PCI-MSI-edge eth1-1 92: 76891342 6123 0 0 0 3731477 26657193 381158 168532115 0 0 0 PCI-MSI-edge eth1-2 93: 91 0 0 0 6292 110184701 71884833 344882089 0 0 0 0 PCI-MSI-edge eth1-3 94: 1854321 117636 32951731 15344357 0 3837296 7813757 6339 48898569 5782626 81065749 100168154 PCI-MSI-edge eth1-4 95: 2190993 122282228 26207097 23270728 11389030 178851 4200707 677260 7872638 478713281 0 0 PCI-MSI-edge eth1-5 96: 81 0 7073 0 503040924 0 148032258 0 0 0 0 0 PCI-MSI-edge eth1-6 97: 34 0 0 0 0 0 120430131 0 0 0 0 0 PCI-MSI-edge eth1-7 99: 21 0 0 0 0 357998 0 27 0 78 0 252009 PCI-MSI-edge eth2-0 100: 105 0 0 218420433 0 222 0 111 4866 3996 104 203 PCI-MSI-edge eth2-1 101: 23 0 0 0 0 0 458 0 0 0 0 0 PCI-MSI-edge eth2-2 102: 25 0 0 0 0 0 3 47 278 0 0 0 PCI-MSI-edge eth2-3 103: 32 0 0 0 0 0 9 39 2 80 246248 357997 PCI-MSI-edge eth2-4 104: 22 0 0 0 0 0 110 16 1647 3929 0 0 PCI-MSI-edge eth2-5 105: 76464 0 218499029 0 0 2212 111 4877 116 0 213 4372 PCI-MSI-edge eth2-6 106: 22124779 0 0 0 0 33 692 8912306 29 0 636 375 PCI-MSI-edge eth2-7 NMI: 0 0 0 0 0 0 0 0 0 0 0 0 Non-maskable interrupts LOC: 531441723 604557984 442098088 191760494 355469815 385314458 454471543 516991693 445322058 189023248 295447258 396084299 Local timer interrupts SPU: 0 0 0 0 0 0 0 0 0 0 0 0 Spurious interrupts PMI: 0 0 0 0 0 0 0 0 0 0 0 0 Performance monitoring interrupts PND: 0 0 0 0 0 0 0 0 0 0 0 0 Performance pending work RES: 16982423 17383275 11711037 7496750 10565637 8566169 11456190 14451915 8547242 4777805 7432008 6263465 Rescheduling interrupts CAL: 86 121 94 120 118 115 109 117 115 121 118 120 Function call interrupts TLB: 5069457 4983872 4744031 4253779 4813250 4745450 4611803 4793054 4597800 4393898 4440386 4381963 TLB shootdowns TRM: 0 0 0 0 0 0 0 0 0 0 0 0 Thermal event interrupts THR: 0 0 0 0 0 0 0 0 0 0 0 0 Threshold APIC interrupts MCE: 0 0 0 0 0 0 0 0 0 0 0 0 Machine check exceptions MCP: 69655 69655 69655 69655 69655 69655 69655 69655 69655 69655 69655 69655 Machine check polls ERR: 11 MIS: 0 mvg, Edwin Edwin Hoexum | Senior Systeeem Specialist | Voice Management _________________________________________________________________ Dr. van Deenweg 120, Zwolle | Postbus 9501 | 9703 LM Groningen | www.ziggo.nl Tel: +31887172699 | Mobiel: +31655148679| e-mailadres: Edwin.hoexum at office.ziggo.nl Werk Locaties: Maandag en Woensdag locatie Utrecht Dinsdag, Donderdag en Vrijdag locatie Zwolle -----Oorspronkelijk bericht----- Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Anthony Minessale Verzonden: vrijdag 29 juni 2012 19:07 Aan: FreeSWITCH Users Help Onderwerp: Re: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a you could do with more than 2 cpu, the more cores the better you will do. For similar prices you could find a 8 or 12 core box. The 3 most important things in a FS server are, as many CPU cores as possible, as much ram as possible, the best possible CPU with shared bus to RAM and CPU. 2012/6/29 Eugene Shcherbatyuk : > Session limit is configurable. Have a look at switch.conf.xml file. > > > On 28/06/12 21:41, Hoexum, Edwin wrote: >> >> I am trying to do some load testing on a freeswitch for transcoding from >> SILK to G711a. The question is: Is the limit for RTP sessions >> limited to CPU power or are there also limits on the freeswitch >> application and can I tune freeswitch to do the job for 1500 sessions or >> more. > > > ========================================================= > > ?????? ????????? ? ????? ???????? (<>) ???????? ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? ?????????????. > > ========================================================= > > This message and any attachments (the "message") are confidential, intended solely for the addressees, and may contain legally privileged information. Any unauthorized use or dissemination is prohibited. E-mails are susceptible to alteration. Neither JSC "BELROSBANK" nor any of its subsidiaries or affiliates shall be liable for the message if altered, changed or falsified. > > ========================================================= > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org This message is confidential and may be privileged. Any review, retransmission, dissemination or other use of, taking any action with reference to this information by persons other then the intended recipient is prohibited. If you receive this message in error, please notify the sender by reply e-mail and delete this message from all computers. Please note that e-mails are susceptible to change. The sender will not accept liability for the improper or incomplete transmission of the information contained in this message. Spaar het milieu door deze e-mail niet te printen/Please consider the environment before printing this email. From saami_mh at ymail.com Tue Jul 3 14:16:05 2012 From: saami_mh at ymail.com (Samira Mh) Date: Tue, 3 Jul 2012 03:16:05 -0700 (PDT) Subject: [Freeswitch-users] how to get the content of sip header ? In-Reply-To: References: <1341061052.23871.YahooMailNeo@web120106.mail.ne1.yahoo.com> , <1341113898.92837.YahooMailNeo@web120104.mail.ne1.yahoo.com> Message-ID: <1341310565.53286.YahooMailNeo@web120102.mail.ne1.yahoo.com> for example i want to read the ?value of ?sip-header '?CSeq' ? in lua(that is configure for mod_xml in /usr/loca/freeswitch/conf/autoload/lia.xml.conf), must to issue as follow: local var = session:getVariable("sip_CSeq") but the error occure: ?attempt to index global 'session' (a nil value) ?REGISTER sip:192.168.10.89 SIP/2.0 ? ?Via: SIP/2.0/UDP 192.168.18.120:33648;branch=z9hG4bK-d8754z-7e634b5c7a239f6b-1---d8754z-;rport ? ?Max-Forwards: 70 ? ?Contact: ? ?To: ? ?From: ;tag=695adb0d ? ?Call-ID: MGUzNTFkZWVlOTcwODBmOWViNGY4MjM3MzE2NTQzMTM. ? ?CSeq: 1 REGISTER ? ?Expires: 5454 ? ?Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO ? ?Supported: timer ? ?User-Agent: eyeBeam release 1102q stamp 51814 ? ?Content-Length: 0 ________________________________ From: Peter Olsson To: FreeSWITCH Users Help Sent: Sunday, July 1, 2012 11:33 AM Subject: Re: [Freeswitch-users] how to get the content of sip header ? Use session:getVariable("var") to read whatever channel variable you need. As Michael said, the SIP headers are stored like sip_header_name. For instance, to read the header Diversion:, use the variable sip_diversion. Michael also mentions the info app, which is a good way to dump all channel variables for a channel, so you know exactly what you have available. To use this, just execute the app info in the dialplan. /Peter 1 jul 2012 kl. 05:47 skrev "Samira Mh" >: is it possible to explain in details? ,i have review the wiki on subject but don't understand clearly, i am new on freeswitch, thanks ... ________________________________ From: Michael Collins > To: FreeSWITCH Users Help > Sent: Sunday, July 1, 2012 1:33 AM Subject: Re: [Freeswitch-users] how to get the content of sip header ? See if you have a bunch of sip_xxx channel variables. Send the call to the info app and see what you've got. -MC On Sat, Jun 30, 2012 at 5:57 AM, Samira Mh > wrote: hi, how to get the content of sip header? in lua? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fefc4ee32761372387519! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fefc4ee32761372387519! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120703/7c0ea16f/attachment-0001.html From akostenko at broadvox.com Tue Jul 3 14:10:02 2012 From: akostenko at broadvox.com (Alexandr Kostenko) Date: Tue, 3 Jul 2012 13:10:02 +0300 Subject: [Freeswitch-users] Freeswitch console log and file log In-Reply-To: <5935829.173.1341309872664.JavaMail.a66a@a66a> Message-ID: <8989605.233.1341310197992.JavaMail.a66a@a66a> Hi, Guys. Can you explain me more understandable about logs in freeswitch. Can I for example decrease console log level (or fully disable) and increase file log level? Also can full log level causing freeswitch crush ? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/a55a84a3/attachment.html From mayamatakeshi at gmail.com Tue Jul 3 14:25:12 2012 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Tue, 3 Jul 2012 19:25:12 +0900 Subject: [Freeswitch-users] how to get the content of sip header ? In-Reply-To: <1341310565.53286.YahooMailNeo@web120102.mail.ne1.yahoo.com> References: <1341061052.23871.YahooMailNeo@web120106.mail.ne1.yahoo.com> <1341113898.92837.YahooMailNeo@web120104.mail.ne1.yahoo.com> <1341310565.53286.YahooMailNeo@web120102.mail.ne1.yahoo.com> Message-ID: On Tue, Jul 3, 2012 at 7:16 PM, Samira Mh wrote: > for example i want to read the value of sip-header ' CSeq' in lua(that > is configure for mod_xml in > /usr/loca/freeswitch/conf/autoload/lia.xml.conf), must to issue as follow: > local var = session:getVariable("sip_CSeq") > but the error occure: > attempt to index global 'session' (a nil value) > Lua syntax error. It should be: local var = session.getVariable("sip_CSeq") regards, Takeshi. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/696cdb51/attachment.html From anton.jugatsu at gmail.com Tue Jul 3 14:28:32 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Tue, 3 Jul 2012 14:28:32 +0400 Subject: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a In-Reply-To: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern> References: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern> Message-ID: Can you share please what sysctl params you tweaked? 2012/6/28 Hoexum, Edwin > Gents,**** > > ** ** > > I am trying to do some load testing on a freeswitch for transcoding from > SILK to G711a. The question is: Is the limit for RTP sessions limited > to CPU power or are there also limits on the freeswitch application and can > I tune freeswitch to do the job for 1500 sessions or more. **** > > ** ** > > At the moment freeswitch is the best transcoding device for SILK to G711a > that I know of. Also compared to the big companies who deliver transcoding > devices. **** > > ** ** > > SETUP:**** > > ** ** > > 3 X G7 HP DL 380 with 16 G mem 6 core 12 CPU?s**** > > ** ** > > 1 for SIPP UAC (1Gb interface for media and Signaling)**** > > 1 for Freeswitch (trancoding function SILK to G711a) 1Gb interface for > media and Signaling**** > > 1 for SIPP UAS (1Gb interface for media and Signaling)**** > > ** ** > > First tuning OS (with help from the wiki and some LINUX sysctl tuning)**** > > ** ** > > ** ** > > SIPP UAC <-> Freeswitch**** > > ** ** > > ** ** > > root at mnd-rc0001-srvot02:~# iperf -c 172.23.125.46 -u -d -b 400m**** > > ------------------------------------------------------------**** > > Server listening on UDP port 5001**** > > Receiving 1470 byte datagrams**** > > UDP buffer size: 64.0 KByte (default)**** > > ------------------------------------------------------------**** > > ------------------------------------------------------------**** > > Client connecting to 172.23.125.46, UDP port 5001**** > > Sending 1470 byte datagrams**** > > UDP buffer size: 64.0 KByte (default)**** > > ------------------------------------------------------------**** > > [ 4] local 172.23.125.42 port 46443 connected with 172.23.125.46 port 5001 > **** > > [ 3] local 172.23.125.42 port 5001 connected with 172.23.125.46 port 39259 > **** > > [ ID] Interval Transfer Bandwidth**** > > [ 4] 0.0-10.0 sec 483 MBytes 405 Mbits/sec**** > > [ 4] Sent 344796 datagrams**** > > [ 3] 0.0-10.0 sec 483 MBytes 406 Mbits/sec 0.001 ms 0/344828 > (0%)**** > > [ 3] 0.0-10.0 sec 1 datagrams received out-of-order**** > > [ 4] Server Report:**** > > [ 4] 0.0-10.0 sec 483 MBytes 405 Mbits/sec 0.004 ms 261/344795 > (0.076%)**** > > [ 4] 0.0-10.0 sec 1 datagrams received out-of-order**** > > root at mnd-rc0001-srvot02:~# **** > > ** ** > > ** ** > > ** ** > > Linux mnd-rc0001-srvot06 2.6.32-28-vserver #55~ppa1-Ubuntu SMP Fri Feb 4 > 21:25:09 UTC 2011 x86_64 GNU/Linux**** > > Ubuntu 10.04 LTS**** > > ** ** > > Welcome to Ubuntu!**** > > * Documentation: https://help.ubuntu.com/**** > > Last login: Thu Jun 21 15:01:35 2012 from 172.21.88.65**** > > root at mnd-rc0001-srvot06:~# iperf -s -u -i1**** > > ------------------------------------------------------------**** > > Server listening on UDP port 5001**** > > Receiving 1470 byte datagrams**** > > UDP buffer size: 64.0 KByte (default)**** > > ------------------------------------------------------------**** > > ------------------------------------------------------------**** > > Client connecting to 172.23.125.42, UDP port 5001**** > > Sending 1470 byte datagrams**** > > UDP buffer size: 64.0 KByte (default)**** > > ------------------------------------------------------------**** > > [ 5] local 172.23.125.46 port 39259 connected with 172.23.125.42 port 5001 > **** > > [ 3] local 172.23.125.46 port 5001 connected with 172.23.125.42 port 46443 > **** > > [ ID] Interval Transfer Bandwidth**** > > [ 5] 0.0- 1.0 sec 48.3 MBytes 406 Mbits/sec**** > > [ 3] 0.0- 1.0 sec 48.4 MBytes 406 Mbits/sec 0.004 ms 262/34754 > (0.75%)**** > > [ 5] 1.0- 2.0 sec 48.3 MBytes 406 Mbits/sec**** > > [ 3] 1.0- 2.0 sec 48.3 MBytes 405 Mbits/sec 0.005 ms 0/34476 (0%) > **** > > [ 5] 2.0- 3.0 sec 48.3 MBytes 406 Mbits/sec**** > > [ 3] 2.0- 3.0 sec 48.3 MBytes 405 Mbits/sec 0.005 ms 0/34480 (0%) > **** > > [ 5] 3.0- 4.0 sec 48.3 MBytes 406 Mbits/sec**** > > [ 3] 3.0- 4.0 sec 48.3 MBytes 406 Mbits/sec 0.004 ms 0/34483 (0%) > **** > > [ 5] 4.0- 5.0 sec 48.3 MBytes 406 Mbits/sec**** > > [ 3] 4.0- 5.0 sec 48.3 MBytes 406 Mbits/sec 0.005 ms 0/34482 (0%) > **** > > [ 5] 5.0- 6.0 sec 48.3 MBytes 406 Mbits/sec**** > > [ 3] 5.0- 6.0 sec 48.3 MBytes 405 Mbits/sec 0.004 ms 0/34474 (0%) > **** > > [ 5] 6.0- 7.0 sec 48.3 MBytes 406 Mbits/sec**** > > [ 3] 6.0- 7.0 sec 48.3 MBytes 405 Mbits/sec 0.005 ms 0/34477 (0%) > **** > > [ 5] 7.0- 8.0 sec 48.3 MBytes 406 Mbits/sec**** > > [ 3] 7.0- 8.0 sec 48.3 MBytes 405 Mbits/sec 0.004 ms 0/34480 (0%) > **** > > [ 5] 8.0- 9.0 sec 48.3 MBytes 406 Mbits/sec**** > > [ 3] 8.0- 9.0 sec 48.3 MBytes 406 Mbits/sec 0.005 ms 0/34482 (0%) > **** > > [ 5] 9.0-10.0 sec 48.3 MBytes 406 Mbits/sec**** > > [ 5] 0.0-10.0 sec 483 MBytes 406 Mbits/sec**** > > [ 5] Sent 344829 datagrams**** > > [ 3] 0.0-10.0 sec 483 MBytes 405 Mbits/sec 0.004 ms 261/344795 > (0.076%)**** > > [ 3] 0.0-10.0 sec 1 datagrams received out-of-order**** > > [ 5] Server Report:**** > > [ 5] 0.0-10.0 sec 483 MBytes 406 Mbits/sec 0.000 ms 0/344828 > (0%)**** > > [ 5] 0.0-10.0 sec 1 datagrams received out-of-order**** > > ** ** > > ** ** > > UDP traffic looks fine.**** > > ** ** > > SIPP (UAC) inject SILK RTP (PCAP ) and send this to Freeswitch. Freeswitch > send this G711a to the SIPP( UAS).**** > > ** ** > > I doing some testing and until 700 sessions it look fine (Packed loss > <0.1% and delay <40ms)**** > > ** ** > > If I do more sessions 10% of packet have no packet loss but delay of more > the 100ms.**** > > ** ** > > ** ** > > SIPP uac side:**** > > ** ** > > ------------------------------ Scenario Screen -------- [1-9]: Change > Screen --**** > > Call-rate(length) Port Total-time Total-calls Remote-host**** > > 50.0(0 ms)/1.000s 5060 230.53 s 2000 172.23.125.12:5060 > (UDP)**** > > ** ** > > Call limit reached (-m 2000), 0.000 s period 0 ms scheduler resolution* > *** > > 0 calls (limit 4500) Peak was 2000 calls, after 40 s** > ** > > 0 Running, 1654 Paused, 0 Woken up**** > > 0 dead call msg (discarded) 6459 out-of-call msg > (discarded) **** > > 1 open sockets ** ** > > ** ** > > Messages Retrans Timeout > Unexpected-Msg**** > > Pause [ 2000ms] 2000 0 ** > ** > > INVITE ----------> 2000 0 0 ** > ** > > 100 <---------- 2000 0 0 0 ** > ** > > 183 <---------- 0 0 0 0 ** > ** > > 400 <---------- 0 0 0 0 ** > ** > > 401 <---------- 0 0 0 0 ** > ** > > 403 <---------- 0 0 0 0 ** > ** > > 404 <---------- 0 0 0 0 ** > ** > > 405 <---------- 0 0 0 0 ** > ** > > 406 <---------- 0 0 0 0 ** > ** > > 408 <---------- 0 0 0 0 ** > ** > > 415 <---------- 0 0 0 0 ** > ** > > 433 <---------- 0 0 0 0 ** > ** > > 480 <---------- 0 0 0 0 ** > ** > > 481 <---------- 0 0 0 0 ** > ** > > 484 <---------- 0 0 0 0 ** > ** > > 485 <---------- 0 0 0 0 ** > ** > > 486 <---------- 0 0 0 0 ** > ** > > 487 <---------- 0 0 0 0 ** > ** > > 488 <---------- 0 0 0 0 ** > ** > > 489 <---------- 0 0 0 0 ** > ** > > 500 <---------- 0 0 0 0 ** > ** > > 502 <---------- 0 0 0 0 ** > ** > > 503 <---------- 0 0 0 0 ** > ** > > 180 <---------- 2000 0 0 0 ** > ** > > 200 <---------- E-RTD1 2000 0 0 0 ** > ** > > ACK ----------> 2000 0 ** > ** > > Pause [ 500ms] 2000 0 ** > ** > > [ NOP ] **** > > BYE <---------- 2000 0 0 0 ** > ** > > 200 ----------> 2000 0 ** > ** > > ------------------------------ Test Terminated > --------------------------------**** > > ** ** > > ** ** > > ----------------------------- Statistics Screen ------- [1-9]: Change > Screen --**** > > Start Time | 2012-06-28 20:17:17:312 > 1340907437.312927 **** > > Last Reset Time | 2012-06-28 20:21:07:857 > 1340907667.857562 **** > > Current Time | 2012-06-28 20:21:07:857 > 1340907667.857769 **** > > > -------------------------+---------------------------+-------------------------- > **** > > Counter Name | Periodic value | Cumulative value*** > * > > > -------------------------+---------------------------+-------------------------- > **** > > Elapsed Time | 00:00:00:000 | > 00:03:50:544 **** > > Call Rate | 0.000 cps | 8.675 > cps **** > > > -------------------------+---------------------------+-------------------------- > **** > > Incoming call created | 0 | > 0 **** > > OutGoing call created | 0 | > 2000 **** > > Total Call created | | > 2000 **** > > Current Call | 0 > | **** > > ** ** > > ** ** > > ** ** > > SIPP UAS side:**** > > ** ** > > ------------------------------ Scenario Screen -------- [1-9]: Change > Screen --**** > > Port Total-time Total-calls Transport**** > > 5060 65.10 s 2000 UDP**** > > ** ** > > 0 new calls during 1.001 s period 1 ms scheduler resolution**** > > 2000 calls Peak was 2000 calls, after 44 s** > ** > > 0 Running, 2002 Paused, 4 Woken up**** > > 0 dead call msg (discarded) ** ** > > 3 open sockets ** ** > > 2021 Total echo RTP pckts 1st stream 5.327 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 2000 0 0 0 *** > * > > <---------- 100 2000 0 ** > ** > > [ 500ms] Pause 2000 0 ** > ** > > <---------- 180 2000 0 ** > ** > > <---------- 200 2000 0 0 ** > ** > > ----------> ACK 2000 0 0 0 ** > ** > > [ 3:08] Pause 2000 0 ** > ** > > <---------- BYE 0 0 0 ** > ** > > ----------> 200 0 0 0 0 ** > ** > > ------------------------------ Sipp Server Mode > -------------------------------**** > > ** ** > > ** ** > > > ############################################################################## > **** > > ** ** > > MPSTAT ?A 5 on freeswitch:**** > > ** ** > > ** ** > > 8:29:50 CPU %usr %nice %sys %iowait %irq %soft %steal > %guest %idle**** > > 18:29:55 all 24.84 0.00 25.63 0.00 0.02 7.46 > 0.00 0.00 42.05**** > > 18:29:55 0 43.52 0.00 26.72 0.00 0.00 4.05 0.00 > 0.00 25.71**** > > 18:29:55 1 27.88 0.00 39.18 0.00 0.00 3.12 > 0.00 0.00 29.82**** > > 18:29:55 2 27.70 0.00 33.40 0.00 0.00 4.17 > 0.00 0.00 34.72**** > > 18:29:55 3 25.57 0.00 35.98 0.00 0.00 3.03 > 0.00 0.00 35.42**** > > 18:29:55 4 27.36 0.00 27.56 0.00 0.00 2.56 > 0.00 0.00 42.52**** > > 18:29:55 5 25.45 0.00 24.44 0.00 0.00 2.22 > 0.00 0.00 47.88**** > > 18:29:55 6 24.77 0.00 31.28 0.00 0.00 2.98 > 0.00 0.00 40.97**** > > 18:29:55 7 23.05 0.00 23.81 0.00 0.00 2.67 > 0.00 0.00 50.48**** > > 18:29:55 8 22.22 0.00 23.03 0.00 0.20 2.02 > 0.00 0.00 52.53**** > > 18:29:55 9 4.84 0.00 5.71 0.00 0.00 62.86 > 0.00 0.00 26.59**** > > 18:29:55 10 23.36 0.00 17.01 0.00 0.00 2.46 > 0.00 0.00 57.17**** > > 18:29:55 11 20.21 0.00 15.11 0.00 0.00 2.98 > 0.00 0.00 61.70**** > > ** ** > > 18:29:50 CPU intr/s**** > > 18:29:55 all 30433.80**** > > 18:29:55 0 0.00**** > > 18:29:55 1 0.00**** > > 18:29:55 2 0.00**** > > 18:29:55 3 0.00**** > > 18:29:55 4 0.00**** > > 18:29:55 5 0.00**** > > 18:29:55 6 0.00**** > > 18:29:55 7 0.00**** > > 18:29:55 8 0.00**** > > 18:29:55 9 0.00**** > > 18:29:55 10 0.00**** > > 18:29:55 11 0.00**** > > ** ** > > 18:29:50 CPU 0/s 1/s 8/s**** > > 18:29:55 0 0.00 0.00 0.00**** > > 18:29:55 1 0.00 0.00 0.00**** > > 18:29:55 2 0.00 0.00 0.00**** > > 18:29:55 3 0.00 0.00 0.00**** > > 18:29:55 4 0.00 0.00 0.00**** > > 18:29:55 5 0.00 0.00 0.00**** > > 18:29:55 6 0.00 0.00 0.00**** > > 18:29:55 7 0.00 0.00 0.00**** > > 18:29:55 8 0.00 0.00 0.00**** > > 18:29:55 9 0.00 0.00 0.00**** > > 18:29:55 10 0.00 0.00 0.00**** > > 18:29:55 11 0.00 0.00 0.00**** > > ** ** > > I tested it with G711a to G711a and got the same result. **** > > ** ** > > I tested also some RTP parameters like ** ** > > ** ** > > or false**** > > or false**** > > ** ** > > The lot of the module are not activated like CDR stuff.**** > > ** ** > > *mvg,* > > *Edwin* > > ** ** > > ------------------------------ > This message is confidential and may be privileged. Any review, > retransmission, dissemination or other use of, taking any action with > reference to this information by persons other then the intended recipient > is prohibited. If you receive this message in error, please notify the > sender by reply e-mail and delete this message from all computers. Please > note that e-mails are susceptible to change. The sender will not accept > liability for the improper or incomplete transmission of the information > contained in this message. > > Spaar het milieu door deze e-mail niet te printen/Please consider the > environment before printing this email. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120703/b1507e3d/attachment-0001.html From peter.olsson at visionutveckling.se Tue Jul 3 14:32:20 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 3 Jul 2012 10:32:20 +0000 Subject: [Freeswitch-users] how to get the content of sip header ? Message-ID: <1FFF97C269757C458224B7C895F35F1512BF36@cantor.std.visionutv.se> I'm not sure if that specific header i available, but that is not the problem here. It seems you have not started the luascript from the dialplan, and because of this the session variable is not set. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh Skickat: den 3 juli 2012 12:16 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] how to get the content of sip header ? for example i want to read the value of sip-header ' CSeq' in lua(that is configure for mod_xml in /usr/loca/freeswitch/conf/autoload/lia.xml.conf), must to issue as follow: local var = session:getVariable("sip_CSeq") but the error occure: attempt to index global 'session' (a nil value) REGISTER sip:192.168.10.89 SIP/2.0 Via: SIP/2.0/UDP 192.168.18.120:33648;branch=z9hG4bK-d8754z-7e634b5c7a239f6b-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: ;tag=695adb0d Call-ID: MGUzNTFkZWVlOTcwODBmOWViNGY4MjM3MzE2NTQzMTM. CSeq: 1 REGISTER Expires: 5454 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: timer User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ________________________________ From: Peter Olsson > To: FreeSWITCH Users Help > Sent: Sunday, July 1, 2012 11:33 AM Subject: Re: [Freeswitch-users] how to get the content of sip header ? Use session:getVariable("var") to read whatever channel variable you need. As Michael said, the SIP headers are stored like sip_header_name. For instance, to read the header Diversion:, use the variable sip_diversion. Michael also mentions the info app, which is a good way to dump all channel variables for a channel, so you know exactly what you have available. To use this, just execute the app info in the dialplan. /Peter 1 jul 2012 kl. 05:47 skrev "Samira Mh" >>: is it possible to explain in details? ,i have review the wiki on subject but don't understand clearly, i am new on freeswitch, thanks ... ________________________________ From: Michael Collins >> To: FreeSWITCH Users Help >> Sent: Sunday, July 1, 2012 1:33 AM Subject: Re: [Freeswitch-users] how to get the content of sip header ? See if you have a bunch of sip_xxx channel variables. Send the call to the info app and see what you've got. -MC On Sat, Jun 30, 2012 at 5:57 AM, Samira Mh >> wrote: hi, how to get the content of sip header in lua? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fefc4ee32761372387519! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff2c56532769990114684! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/3a623638/attachment.html From peter.olsson at visionutveckling.se Tue Jul 3 14:38:24 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 3 Jul 2012 10:38:24 +0000 Subject: [Freeswitch-users] how to get the content of sip header ? Message-ID: <1FFF97C269757C458224B7C895F35F1512BF5B@cantor.std.visionutv.se> Hmm... I've never used Lua, but that means that all documentation for mod_lua on the wiki is wrong? Sounds strange to me... /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r mayamatakeshi Skickat: den 3 juli 2012 12:25 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] how to get the content of sip header ? On Tue, Jul 3, 2012 at 7:16 PM, Samira Mh > wrote: for example i want to read the value of sip-header ' CSeq' in lua(that is configure for mod_xml in /usr/loca/freeswitch/conf/autoload/lia.xml.conf), must to issue as follow: local var = session:getVariable("sip_CSeq") but the error occure: attempt to index global 'session' (a nil value) Lua syntax error. It should be: local var = session.getVariable("sip_CSeq") regards, Takeshi. !DSPAM:4ff2c7f532762054221329! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/ff3a2f0b/attachment-0001.html From peter.olsson at visionutveckling.se Tue Jul 3 14:44:19 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 3 Jul 2012 10:44:19 +0000 Subject: [Freeswitch-users] how to get the content of sip header ? Message-ID: <1FFF97C269757C458224B7C895F35F1512BF71@cantor.std.visionutv.se> Is this some kind of xml-generating script for register bindings (as in here: http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration)? If that's the case, you have no session object, and you will have to follow the information on that wiki page instead. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh Skickat: den 3 juli 2012 12:16 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] how to get the content of sip header ? for example i want to read the value of sip-header ' CSeq' in lua(that is configure for mod_xml in /usr/loca/freeswitch/conf/autoload/lia.xml.conf), must to issue as follow: local var = session:getVariable("sip_CSeq") but the error occure: attempt to index global 'session' (a nil value) REGISTER sip:192.168.10.89 SIP/2.0 Via: SIP/2.0/UDP 192.168.18.120:33648;branch=z9hG4bK-d8754z-7e634b5c7a239f6b-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: ;tag=695adb0d Call-ID: MGUzNTFkZWVlOTcwODBmOWViNGY4MjM3MzE2NTQzMTM. CSeq: 1 REGISTER Expires: 5454 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: timer User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ________________________________ From: Peter Olsson > To: FreeSWITCH Users Help > Sent: Sunday, July 1, 2012 11:33 AM Subject: Re: [Freeswitch-users] how to get the content of sip header ? Use session:getVariable("var") to read whatever channel variable you need. As Michael said, the SIP headers are stored like sip_header_name. For instance, to read the header Diversion:, use the variable sip_diversion. Michael also mentions the info app, which is a good way to dump all channel variables for a channel, so you know exactly what you have available. To use this, just execute the app info in the dialplan. /Peter 1 jul 2012 kl. 05:47 skrev "Samira Mh" >>: is it possible to explain in details? ,i have review the wiki on subject but don't understand clearly, i am new on freeswitch, thanks ... ________________________________ From: Michael Collins >> To: FreeSWITCH Users Help >> Sent: Sunday, July 1, 2012 1:33 AM Subject: Re: [Freeswitch-users] how to get the content of sip header ? See if you have a bunch of sip_xxx channel variables. Send the call to the info app and see what you've got. -MC On Sat, Jun 30, 2012 at 5:57 AM, Samira Mh >> wrote: hi, how to get the content of sip header in lua? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fefc4ee32761372387519! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff2c56532769990114684! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/d398edeb/attachment.html From mayamatakeshi at gmail.com Tue Jul 3 14:50:32 2012 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Tue, 3 Jul 2012 19:50:32 +0900 Subject: [Freeswitch-users] how to get the content of sip header ? In-Reply-To: <1FFF97C269757C458224B7C895F35F1512BF5B@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1512BF5B@cantor.std.visionutv.se> Message-ID: A yes, sorry. I don't use lua with FS that much either. But i got it from here: http://wiki.freeswitch.org/wiki/Session_getVariable var caller_id =session.getVariable("caller_id_number") I can see that page is the one that is wrong (will correct it now). On Tue, Jul 3, 2012 at 7:38 PM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Hmm... I?ve never used Lua, but that means that all documentation for > mod_lua on the wiki is wrong? Sounds strange to me...**** > > ** ** > > /Peter**** > > ** ** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *mayamatakeshi > *Skickat:* den 3 juli 2012 12:25 > > *Till:* FreeSWITCH Users Help > *?mne:* Re: [Freeswitch-users] how to get the content of sip header ?**** > > ** ** > > ** ** > > On Tue, Jul 3, 2012 at 7:16 PM, Samira Mh wrote:**** > > for example i want to read the value of sip-header ' CSeq' in lua(that > is configure for mod_xml in > /usr/loca/freeswitch/conf/autoload/lia.xml.conf), must to issue as follow: > **** > > local var = session:getVariable("sip_CSeq")**** > > but the error occure:**** > > attempt to index global 'session' (a nil value)**** > > > Lua syntax error. It should be:**** > > local var = session.getVariable("sip_CSeq") > > regards, > Takeshi. > !DSPAM:4ff2c7f532762054221329! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120703/29c1ab9e/attachment-0001.html From ben at langfeld.co.uk Tue Jul 3 14:52:05 2012 From: ben at langfeld.co.uk (Ben Langfeld) Date: Tue, 3 Jul 2012 11:52:05 +0100 Subject: [Freeswitch-users] Inbound EventSocket anomoly In-Reply-To: <1FFF97C269757C458224B7C895F35F1512B653@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1512B653@cantor.std.visionutv.se> Message-ID: Thanks gents, and especially Anthony for that blazingly fast response. I will give both options a try, and probably include references in the IES docs. The concept of scoped variables is hidden fairly effectively. Regards, Ben Langfeld On 3 July 2012 06:34, Peter Olsson wrote: > I saw Tony added a new patch for you. However, even before this it was actually possible. For playback specific features, you could use playback local vars that will be sent back on PLAYBACK_START and PLAYBACK_STOP. > > More info here; > http://wiki.freeswitch.org/wiki/Mod_playback#Example_for_specific_playback_variables > > For a more generic approach for CHANNEL_EXECUTE_COMPLETE, it is also possible to use scoped variables, as mentioned here; > http://wiki.freeswitch.org/wiki/Channel_Variables#Scoped_Variables > > The difference with these approaches is that you (on the ESL client side) can decide a unique value. Tonys patch cretes a UUID for you, and uses that on the CHANNEL_EXECUTE_COMPLETE event. > > /Peter > > ________________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Ben Langfeld [ben at langfeld.co.uk] > Skickat: den 2 juli 2012 23:29 > Till: FreeSWITCH Users Help > Cc: Ben Klang > ?mne: [Freeswitch-users] Inbound EventSocket anomoly > > Gents, > > I was hoping someone would be able to clear up what appears to be an > obvious omission from IES. > > If I execute the playback application on a channel via IES, I get an > immediate response to indicate it was queued, and I then have to wait > for either PLAYBACK_STOP or CHANNEL_EXECUTE_COMPLETE. The problem is, > the response contains nothing to filter events by. The only link I > have are the 'Application' and 'Application-Data' fields on the event, > and there is no guarantee those are unique to an individual > application execution. I would expect the immediate response to return > a UUID which would be present on the related events. > > Am I missing something, or is this actually a bug? > > Example: https://gist.github.com/40f079ae2a079abe6393 > > Regards, > Ben Langfeld > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4ff2113932762747716525! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From saami_mh at ymail.com Tue Jul 3 15:05:50 2012 From: saami_mh at ymail.com (Samira Mh) Date: Tue, 3 Jul 2012 04:05:50 -0700 (PDT) Subject: [Freeswitch-users] how to get the content of sip header ? In-Reply-To: <1FFF97C269757C458224B7C895F35F1512BF71@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1512BF71@cantor.std.visionutv.se> Message-ID: <1341313550.3730.YahooMailNeo@web120103.mail.ne1.yahoo.com> thanks Peter, yes , i have followed the page :??http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration my configurations as follow: in register.lua script , i have authenticared and registered users usig lua script instead of using in-memory method, ?so the sip header ?initiatiate in lua script ?as follow: (i don't now why ?CSeq/contact/, etc left empty)... recv 551 bytes from udp/[192.168.18.120]:57916 at 11:02:40.742843: ? ?------------------------------------------------------------------------ ? ?REGISTER sip:192.168.10.89 SIP/2.0 ? ?Via: SIP/2.0/UDP 192.168.18.120:57916;branch=z9hG4bK-d8754z-7d027150e23ed159-1---d8754z-;rport ? ?Max-Forwards: 70 ? ?Contact: ? ?To: ? ?From: ;tag=0e70ba56 ? ?Call-ID: NjU3NzZhZjYwNGUxNWZmOTZmN2VjYWI0NTM5NDEyYjE. ? ?CSeq: 1 REGISTER ? ?Expires: 5454 ? ?Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO ? ?Supported: timer ? ?User-Agent: eyeBeam release 1102q stamp 51814 ? ?Content-Length: 0 ? ?------------------------------------------------------------------------ send 695 bytes to udp/[192.168.18.120]:57916 at 11:02:40.743818: ? ?------------------------------------------------------------------------ ? ?SIP/2.0 401 Unauthorized ? ?Via: SIP/2.0/UDP 192.168.18.120:57916;branch=z9hG4bK-d8754z-7d027150e23ed159-1---d8754z-;rport=57916 ? ?From: ;tag=0e70ba56 ? ?To: ;tag=KQcaD6Ba9e2te ? ?Call-ID: NjU3NzZhZjYwNGUxNWZmOTZmN2VjYWI0NTM5NDEyYjE. ? ?CSeq: 1 REGISTER ? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120622T231506Z~76fae0cec0+unclean~20120625T042938Z ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ? ?Supported: timer, precondition, path, replaces ? ?WWW-Authenticate: Digest realm="192.168.10.89", nonce="9b803030-c4fe-11e1-ad8a-099620eb6996", algorithm=MD5, qop="auth" ? ?Content-Length: 0 ? ?------------------------------------------------------------------------ recv 802 bytes from udp/[192.168.18.120]:57916 at 11:02:40.947421: ? ?------------------------------------------------------------------------ ? ?REGISTER sip:192.168.10.89 SIP/2.0 ? ?Via: SIP/2.0/UDP 192.168.18.120:57916;branch=z9hG4bK-d8754z-8717e047bb6d4d76-1---d8754z-;rport ? ?Max-Forwards: 70 ? ?Contact: ? ?To: ? ?From: ;tag=0e70ba56 ? ?Call-ID: NjU3NzZhZjYwNGUxNWZmOTZmN2VjYWI0NTM5NDEyYjE. ? ?CSeq: 2 REGISTER ? ?Expires: 5454 ? ?Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO ? ?Supported: timer ? ?User-Agent: eyeBeam release 1102q stamp 51814 ? ?Authorization: Digest username="1000",realm="192.168.10.89",nonce="9b803030-c4fe-11e1-ad8a-099620eb6996",uri="sip:192.168.10.89",response="2044cb3f7f932d83402d6a38b26cd5e6",cnonce="2290ee958a69ed51c8bb638ab4341cef",nc=00000001,qop=auth,algorithm=MD5 ? ?Content-Length: 0 ? ?------------------------------------------------------------------------ 2012-07-03 15:32:40.940760 [NOTICE] switch_cpp.cpp:1227 Debug from gen_dir_user_xml.lua, provided params: 'Event-Name: REQUEST_PARAMS Core-UUID: 83d1034e-c4f2-11e1-ad58-099620eb6996 FreeSWITCH-Hostname: PBX FreeSWITCH-Switchname: PBX FreeSWITCH-IPv4: 192.168.10.89 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-07-03%2015%3A32%3A40 Event-Date-GMT: Tue,%2003%20Jul%202012%2011%3A02%3A40%20GMT Event-Date-Timestamp: 1341313360940760 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_parse_auth Event-Calling-Line-Number: 2364 Event-Sequence: 3304 action: sip_auth sip_profile: internal sip_user_agent: eyeBeam%20release%201102q%20stamp%2051814 sip_auth_username: 1000 sip_auth_realm: 192.168.10.89 sip_auth_nonce: 9b803030-c4fe-11e1-ad8a-099620eb6996 sip_auth_uri: sip%3A192.168.10.89 sip_contact_user: 1000 sip_contact_host: 192.168.18.120 sip_to_user: 1000 sip_to_host: 192.168.10.89 sip_from_user: 1000 sip_from_host: 192.168.10.89 sip_request_host: 192.168.10.89 sip_auth_qop: auth sip_auth_cnonce: 2290ee958a69ed51c8bb638ab4341cef sip_auth_nc: 00000001 sip_auth_response: 2044cb3f7f932d83402d6a38b26cd5e6 sip_auth_method: REGISTER key: id user: 1000 domain: 192.168.10.89 ip: 192.168.18.120 ' 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 profile ?internal 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 useragent ?eyeBeam release 1102q stamp 51814 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 username ?1000 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 ?req_auth_realm 192.168.10.89 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 contact_user ?1000 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 contact_host ?192.168.18.120 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 _to_host ?192.168.10.89 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 from_host ?192.168.10.89 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 request_host ?192.168.10.89 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 req_domain ?192.168.10.89 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 req_ip ? 192.168.18.120 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 ?
? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?
send 683 bytes to udp/[192.168.18.120]:57916 at 11:02:40.955909: ? ?------------------------------------------------------------------------ ? ?SIP/2.0 200 OK ? ?Via: SIP/2.0/UDP 192.168.18.120:57916;branch=z9hG4bK-d8754z-8717e047bb6d4d76-1---d8754z-;rport=57916 ? ?From: ;tag=0e70ba56 ? ?To: ;tag=m052e1vD6QrDa ? ?Call-ID: NjU3NzZhZjYwNGUxNWZmOTZmN2VjYWI0NTM5NDEyYjE. ? ?CSeq: 2 REGISTER ? ?Contact: ;expires=5454 ? ?Date: Tue, 03 Jul 2012 11:02:40 GMT ? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120622T231506Z~76fae0cec0+unclean~20120625T042938Z ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ? ?Supported: timer, precondition, path, replaces ? ?Content-Length: 0 ? ?------------------------------------------------------------------------ recv 554 bytes from udp/[192.168.18.120]:57916 at 11:02:41.060557: ? ?------------------------------------------------------------------------ ? ?SUBSCRIBE sip:1000 at 192.168.10.89 SIP/2.0 ? ?Via: SIP/2.0/UDP 192.168.18.120:57916;branch=z9hG4bK-d8754z-ac23c509701fe314-1---d8754z-;rport ? ?Max-Forwards: 70 ? ?Contact: ? ?To: ? ?From: ;tag=96753574 ? ?Call-ID: Yzg3Y2M4OGNmNDQ3ODhhMjU4OGVhZjE5NDYyOGI4OWE. ? ?CSeq: 1 SUBSCRIBE ? ?Expires: 300 ? ?Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO ? ?Supported: timer ? ?User-Agent: eyeBeam release 1102q stamp 51814 ? ?Event: message-summary ? ?Content-Length: 0 ? ?------------------------------------------------------------------------ send 805 bytes to udp/[192.168.18.120]:57916 at 11:02:41.061900: ? ?------------------------------------------------------------------------ ? ?SIP/2.0 202 Accepted ? ?Via: SIP/2.0/UDP 192.168.18.120:57916;branch=z9hG4bK-d8754z-ac23c509701fe314-1---d8754z-;rport=57916 ? ?From: ;tag=96753574 ? ?To: ;tag=PbULiimW0TlP ? ?Call-ID: Yzg3Y2M4OGNmNDQ3ODhhMjU4OGVhZjE5NDYyOGI4OWE. ? ?CSeq: 1 SUBSCRIBE ? ?Contact: ? ?Expires: 300 ? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120622T231506Z~76fae0cec0+unclean~20120625T042938Z ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ? ?Supported: timer, precondition, path, replaces ? ?Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer ? ?Subscription-State: active;expires=300 ? ?Content-Length: 0 ? ?------------------------------------------------------------------------ send 1030 bytes to udp/[192.168.18.120]:57916 at 11:02:41.064419: ? ?------------------------------------------------------------------------ ? ?NOTIFY sip:1000 at 192.168.18.120:57916;rinstance=12e37149747f3c4b SIP/2.0 ? ?Via: SIP/2.0/UDP 192.168.10.89;rport;branch=z9hG4bK494QZ0eB01Z4D ? ?Route: ;rinstance=12e37149747f3c4b ? ?Max-Forwards: 70 ? ?From: ;tag=pjrmjQym094jH ? ?To: ? ?Call-ID: 730f6d3e-3fa1-1230-cbb5-005056945e65 ? ?CSeq: 30320616 NOTIFY ? ?Contact: ? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120622T231506Z~76fae0cec0+unclean~20120625T042938Z ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ? ?Supported: timer, precondition, path, replaces ? ?Event: message-summary ? ?Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer ? ?Subscription-State: terminated;reason=noresource ? ?Content-Type: application/simple-message-summary ? ?Content-Length: 65 ? ?Messages-Waiting: no ? ?Message-Account: sip:1000 at 192.168.10.89 ? ?------------------------------------------------------------------------ send 1005 bytes to udp/[192.168.18.120]:57916 at 11:02:41.164620: ? ?------------------------------------------------------------------------ ? ?NOTIFY sip:1000 at 192.168.18.120:57916 SIP/2.0 ? ?Via: SIP/2.0/UDP 192.168.10.89;rport;branch=z9hG4bK5jyg1UZeXapQS ? ?Route: ;transport=udp ? ?Max-Forwards: 70 ? ?From: ;tag=PbULiimW0TlP ? ?To: ;tag=96753574 ? ?Call-ID: Yzg3Y2M4OGNmNDQ3ODhhMjU4OGVhZjE5NDYyOGI4OWE. ? ?CSeq: 286150130 NOTIFY ? ?Contact: ? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120622T231506Z~76fae0cec0+unclean~20120625T042938Z ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ? ?Supported: timer, precondition, path, replaces ? ?Event: message-summary ? ?Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer ? ?Subscription-State: active;expires=300 ? ?Content-Type: application/simple-message-summary ? ?Content-Length: 65 ? ?Messages-Waiting: no ? ?Message-Account: sip:1000 at 192.168.10.89 ? ?------------------------------------------------------------------------ recv 355 bytes from udp/[192.168.18.120]:57916 at 11:02:41.166525: ? ?------------------------------------------------------------------------ ? ?SIP/2.0 200 OK ? ?Via: SIP/2.0/UDP 192.168.10.89;rport=5060;branch=z9hG4bK494QZ0eB01Z4D ? ?Contact: ? ?To: ;tag=2e2a6254 ? ?From: ;tag=pjrmjQym094jH ? ?Call-ID: 730f6d3e-3fa1-1230-cbb5-005056945e65 ? ?CSeq: 30320616 NOTIFY ? ?User-Agent: eyeBeam release 1102q stamp 51814 ? ?Content-Length: 0 ? ?------------------------------------------------------------------------ recv 368 bytes from udp/[192.168.18.120]:57916 at 11:02:41.267410: ? ?------------------------------------------------------------------------ ? ?SIP/2.0 200 OK ? ?Via: SIP/2.0/UDP 192.168.10.89;rport=5060;branch=z9hG4bK5jyg1UZeXapQS ? ?Contact: ? ?To: ;tag=96753574 ? ?From: ;tag=PbULiimW0TlP ? ?Call-ID: Yzg3Y2M4OGNmNDQ3ODhhMjU4OGVhZjE5NDYyOGI4OWE. ? ?CSeq: 286150130 NOTIFY ? ?User-Agent: eyeBeam release 1102q stamp 51814 ? ?Content-Length: 0 ________________________________ From: Peter Olsson To: 'FreeSWITCH Users Help' Sent: Tuesday, July 3, 2012 3:14 PM Subject: Re: [Freeswitch-users] how to get the content of sip header ? Is this some kind of xml-generating script for register bindings (as in here: http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration)? If that?s the case, you have no session object, and you will have to follow the information on that wiki page instead. ? /Peter ? Fr?n:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh Skickat: den 3 juli 2012 12:16 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] how to get the content of sip header ? ? for example i want to read the ?value of ?sip-header '?CSeq' ? in lua(that is configure for mod_xml in /usr/loca/freeswitch/conf/autoload/lia.xml.conf), must to issue as follow: local var = session:getVariable("sip_CSeq") but the error occure: ?attempt to index global 'session' (a nil value) ? ? ? ? ?REGISTER sip:192.168.10.89 SIP/2.0 ? ?Via: SIP/2.0/UDP 192.168.18.120:33648;branch=z9hG4bK-d8754z-7e634b5c7a239f6b-1---d8754z-;rport ? ?Max-Forwards: 70 ? ?Contact: ? ?To: ? ?From: ;tag=695adb0d ? ?Call-ID: MGUzNTFkZWVlOTcwODBmOWViNGY4MjM3MzE2NTQzMTM. ? ?CSeq: 1 REGISTER ? ?Expires: 5454 ? ?Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO ? ?Supported: timer ? ?User-Agent: eyeBeam release 1102q stamp 51814 ? ?Content-Length: 0 ? ? ________________________________ From:Peter Olsson To: FreeSWITCH Users Help Sent: Sunday, July 1, 2012 11:33 AM Subject: Re: [Freeswitch-users] how to get the content of sip header ? Use session:getVariable("var") to read whatever channel variable you need. As Michael said, the SIP headers are stored like sip_header_name. For instance, to read the header Diversion:, use the variable sip_diversion. Michael also mentions the info app, which is a good way to dump all channel variables for a channel, so you know exactly what you have available. To use this, just execute the app info in the dialplan. /Peter 1 jul 2012 kl. 05:47 skrev "Samira Mh" >: is it possible to explain in details? ,i have review the wiki on subject but don't understand clearly, i am new on freeswitch, thanks ... ________________________________ From: Michael Collins > To: FreeSWITCH Users Help > Sent: Sunday, July 1, 2012 1:33 AM Subject: Re: [Freeswitch-users] how to get the content of sip header ? See if you have a bunch of sip_xxx channel variables. Send the call to the info app and see what you've got. -MC On Sat, Jun 30, 2012 at 5:57 AM, Samira Mh > wrote: hi, how to get the content of sip header? in lua? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fefc4ee32761372387519! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff2c56532769990114684! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120703/f27b1826/attachment-0001.html From edwin.hoexum at office.ziggo.nl Tue Jul 3 15:05:59 2012 From: edwin.hoexum at office.ziggo.nl (Hoexum, Edwin) Date: Tue, 3 Jul 2012 13:05:59 +0200 Subject: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a In-Reply-To: <4FED4DA1.10304@belrosbank.by> References: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern> <4FED4DA1.10304@belrosbank.by> Message-ID: <2C18241C6768CF469C7E432A25327AD51352974004@MAIL-WPV01.office.intern> I have this parameters. Is there some other parameters where I can root at mnd-rc0001-srvot06:/usr/local/freeswitch/conf/autoload_configs# vim switch.conf.xml mvg, Edwin Edwin Hoexum | Senior Systeeem Specialist | Voice Management _________________________________________________________________ Dr. van Deenweg 120, Zwolle | Postbus 9501 | 9703 LM Groningen | www.ziggo.nl Tel: +31887172699 | Mobiel: +31655148679| e-mailadres: Edwin.hoexum at office.ziggo.nl Werk Locaties: Maandag en Woensdag locatie Utrecht Dinsdag, Donderdag en Vrijdag locatie Zwolle -----Oorspronkelijk bericht----- Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens Eugene Shcherbatyuk Verzonden: vrijdag 29 juni 2012 8:39 Aan: FreeSWITCH Users Help Onderwerp: Re: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a Session limit is configurable. Have a look at switch.conf.xml file. On 28/06/12 21:41, Hoexum, Edwin wrote: > > I am trying to do some load testing on a freeswitch for transcoding from > SILK to G711a. The question is: Is the limit for RTP sessions > limited to CPU power or are there also limits on the freeswitch > application and can I tune freeswitch to do the job for 1500 sessions or > more. ========================================================= ?????? ????????? ? ????? ???????? (???????????) ???????? ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? ?????????????. ========================================================= This message and any attachments (the ?message?) are confidential, intended solely for the addressees, and may contain legally privileged information. Any unauthorized use or dissemination is prohibited. E-mails are susceptible to alteration. Neither JSC ?BELROSBANK? nor any of its subsidiaries or affiliates shall be liable for the message if altered, changed or falsified. ========================================================= _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org This message is confidential and may be privileged. Any review, retransmission, dissemination or other use of, taking any action with reference to this information by persons other then the intended recipient is prohibited. If you receive this message in error, please notify the sender by reply e-mail and delete this message from all computers. Please note that e-mails are susceptible to change. The sender will not accept liability for the improper or incomplete transmission of the information contained in this message. Spaar het milieu door deze e-mail niet te printen/Please consider the environment before printing this email. From saami_mh at ymail.com Tue Jul 3 15:12:31 2012 From: saami_mh at ymail.com (Samira Mh) Date: Tue, 3 Jul 2012 04:12:31 -0700 (PDT) Subject: [Freeswitch-users] how to get the content of sip header ? In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1512BF5B@cantor.std.visionutv.se> Message-ID: <1341313951.3420.YahooMailNeo@web120102.mail.ne1.yahoo.com> notice: JavaScript: var caller_id = session.getVariable("caller_id_number") that is not wrong !! ________________________________ From: mayamatakeshi To: FreeSWITCH Users Help Sent: Tuesday, July 3, 2012 3:20 PM Subject: Re: [Freeswitch-users] how to get the content of sip header ? A yes, sorry. I don't use lua with FS that much either. But i got it from here: http://wiki.freeswitch.org/wiki/Session_getVariable var caller_id =session.getVariable("caller_id_number")I can see that page is the one that is wrong (will correct it now). On Tue, Jul 3, 2012 at 7:38 PM, Peter Olsson wrote: Hmm... I?ve never used Lua, but that means that all documentation for mod_lua on the wiki is wrong? Sounds strange to me... >? >/Peter >? >Fr?n:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r mayamatakeshi >Skickat: den 3 juli 2012 12:25 > >Till: FreeSWITCH Users Help >?mne: Re: [Freeswitch-users] how to get the content of sip header ? >? >? >On Tue, Jul 3, 2012 at 7:16 PM, Samira Mh wrote: >for example i want to read the ?value of ?sip-header '?CSeq' ? in lua(that is configure for mod_xml in /usr/loca/freeswitch/conf/autoload/lia.xml.conf), must to issue as follow: >local var = session:getVariable("sip_CSeq") >but the error occure: >?attempt to index global 'session' (a nil value) > >Lua syntax error. It should be: >local var = session.getVariable("sip_CSeq") > >regards, >Takeshi. >!DSPAM:4ff2c7f532762054221329! >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >Join Us At ClueCon - Aug 7-9, 2012 > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120703/b976cdba/attachment-0001.html From saami_mh at ymail.com Tue Jul 3 15:24:29 2012 From: saami_mh at ymail.com (Samira Mh) Date: Tue, 3 Jul 2012 04:24:29 -0700 (PDT) Subject: [Freeswitch-users] Fw: how to get the content of sip header ? In-Reply-To: <1341313550.3730.YahooMailNeo@web120103.mail.ne1.yahoo.com> References: <1FFF97C269757C458224B7C895F35F1512BF71@cantor.std.visionutv.se> <1341313550.3730.YahooMailNeo@web120103.mail.ne1.yahoo.com> Message-ID: <1341314669.23239.YahooMailNeo@web120106.mail.ne1.yahoo.com> thanks Peter, yes , i have followed the page :??http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration my configurations as follow: in register.lua script , i have authenticared and registered users usig lua script instead of using in-memory method, ?so the sip header ?initiatiate in lua script ?as follow: (i don't now why ?CSeq/contact/, etc left empty)... recv 551 bytes from udp/[192.168.18.120]:57916 at 11:02:40.742843: ? ?------------------------------------------------------------------------ ? ?REGISTER sip:192.168.10.89 SIP/2.0 ? ?Via: SIP/2.0/UDP 192.168.18.120:57916;branch=z9hG4bK-d8754z-7d027150e23ed159-1---d8754z-;rport ? ?Max-Forwards: 70 ? ?Contact: ? ?To: ? ?From: ;tag=0e70ba56 ? ?Call-ID: NjU3NzZhZjYwNGUxNWZmOTZmN2VjYWI0NTM5NDEyYjE. ? ?CSeq: 1 REGISTER ? ?Expires: 5454 ? ?Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO ? ?Supported: timer ? ?User-Agent: eyeBeam release 1102q stamp 51814 ? ?Content-Length: 0 ? ?------------------------------------------------------------------------ send 695 bytes to udp/[192.168.18.120]:57916 at 11:02:40.743818: ? ?------------------------------------------------------------------------ ? ?SIP/2.0 401 Unauthorized ? ?Via: SIP/2.0/UDP 192.168.18.120:57916;branch=z9hG4bK-d8754z-7d027150e23ed159-1---d8754z-;rport=57916 ? ?From: ;tag=0e70ba56 ? ?To: ;tag=KQcaD6Ba9e2te ? ?Call-ID: NjU3NzZhZjYwNGUxNWZmOTZmN2VjYWI0NTM5NDEyYjE. ? ?CSeq: 1 REGISTER ? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120622T231506Z~76fae0cec0+unclean~20120625T042938Z ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ? ?Supported: timer, precondition, path, replaces ? ?WWW-Authenticate: Digest realm="192.168.10.89", nonce="9b803030-c4fe-11e1-ad8a-099620eb6996", algorithm=MD5, qop="auth" ? ?Content-Length: 0 ? ?------------------------------------------------------------------------ recv 802 bytes from udp/[192.168.18.120]:57916 at 11:02:40.947421: ? ?------------------------------------------------------------------------ ? ?REGISTER sip:192.168.10.89 SIP/2.0 ? ?Via: SIP/2.0/UDP 192.168.18.120:57916;branch=z9hG4bK-d8754z-8717e047bb6d4d76-1---d8754z-;rport ? ?Max-Forwards: 70 ? ?Contact: ? ?To: ? ?From: ;tag=0e70ba56 ? ?Call-ID: NjU3NzZhZjYwNGUxNWZmOTZmN2VjYWI0NTM5NDEyYjE. ? ?CSeq: 2 REGISTER ? ?Expires: 5454 ? ?Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO ? ?Supported: timer ? ?User-Agent: eyeBeam release 1102q stamp 51814 ? ?Authorization: Digest username="1000",realm="192.168.10.89",nonce="9b803030-c4fe-11e1-ad8a-099620eb6996",uri="sip:192.168.10.89",response="2044cb3f7f932d83402d6a38b26cd5e6",cnonce="2290ee958a69ed51c8bb638ab4341cef",nc=00000001,qop=auth,algorithm=MD5 ? ?Content-Length: 0 ? ?------------------------------------------------------------------------ 2012-07-03 15:32:40.940760 [NOTICE] switch_cpp.cpp:1227 Debug from gen_dir_user_xml.lua, provided params: 'Event-Name: REQUEST_PARAMS Core-UUID: 83d1034e-c4f2-11e1-ad58-099620eb6996 FreeSWITCH-Hostname: PBX FreeSWITCH-Switchname: PBX FreeSWITCH-IPv4: 192.168.10.89 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-07-03%2015%3A32%3A40 Event-Date-GMT: Tue,%2003%20Jul%202012%2011%3A02%3A40%20GMT Event-Date-Timestamp: 1341313360940760 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_parse_auth Event-Calling-Line-Number: 2364 Event-Sequence: 3304 action: sip_auth sip_profile: internal sip_user_agent: eyeBeam%20release%201102q%20stamp%2051814 sip_auth_username: 1000 sip_auth_realm: 192.168.10.89 sip_auth_nonce: 9b803030-c4fe-11e1-ad8a-099620eb6996 sip_auth_uri: sip%3A192.168.10.89 sip_contact_user: 1000 sip_contact_host: 192.168.18.120 sip_to_user: 1000 sip_to_host: 192.168.10.89 sip_from_user: 1000 sip_from_host: 192.168.10.89 sip_request_host: 192.168.10.89 sip_auth_qop: auth sip_auth_cnonce: 2290ee958a69ed51c8bb638ab4341cef sip_auth_nc: 00000001 sip_auth_response: 2044cb3f7f932d83402d6a38b26cd5e6 sip_auth_method: REGISTER key: id user: 1000 domain: 192.168.10.89 ip: 192.168.18.120 ' 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 profile ?internal 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 useragent ?eyeBeam release 1102q stamp 51814 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 username ?1000 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 ?req_auth_realm 192.168.10.89 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 contact_user ?1000 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 contact_host ?192.168.18.120 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 _to_host ?192.168.10.89 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 from_host ?192.168.10.89 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 request_host ?192.168.10.89 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 req_domain ?192.168.10.89 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 req_ip ? 192.168.18.120 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 ?
? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?
send 683 bytes to udp/[192.168.18.120]:57916 at 11:02:40.955909: ? ?------------------------------------------------------------------------ ? ?SIP/2.0 200 OK ? ?Via: SIP/2.0/UDP 192.168.18.120:57916;branch=z9hG4bK-d8754z-8717e047bb6d4d76-1---d8754z-;rport=57916 ? ?From: ;tag=0e70ba56 ? ?To: ;tag=m052e1vD6QrDa ? ?Call-ID: NjU3NzZhZjYwNGUxNWZmOTZmN2VjYWI0NTM5NDEyYjE. ? ?CSeq: 2 REGISTER ? ?Contact: ;expires=5454 ? ?Date: Tue, 03 Jul 2012 11:02:40 GMT ? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120622T231506Z~76fae0cec0+unclean~20120625T042938Z ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ? ?Supported: timer, precondition, path, replaces ? ?Content-Length: 0 ? ?------------------------------------------------------------------------ recv 554 bytes from udp/[192.168.18.120]:57916 at 11:02:41.060557: ? ?------------------------------------------------------------------------ ? ?SUBSCRIBE sip:1000 at 192.168.10.89 SIP/2.0 ? ?Via: SIP/2.0/UDP 192.168.18.120:57916;branch=z9hG4bK-d8754z-ac23c509701fe314-1---d8754z-;rport ? ?Max-Forwards: 70 ? ?Contact: ? ?To: ? ?From: ;tag=96753574 ? ?Call-ID: Yzg3Y2M4OGNmNDQ3ODhhMjU4OGVhZjE5NDYyOGI4OWE. ? ?CSeq: 1 SUBSCRIBE ? ?Expires: 300 ? ?Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO ? ?Supported: timer ? ?User-Agent: eyeBeam release 1102q stamp 51814 ? ?Event: message-summary ? ?Content-Length: 0 ? ?------------------------------------------------------------------------ send 805 bytes to udp/[192.168.18.120]:57916 at 11:02:41.061900: ? ?------------------------------------------------------------------------ ? ?SIP/2.0 202 Accepted ? ?Via: SIP/2.0/UDP 192.168.18.120:57916;branch=z9hG4bK-d8754z-ac23c509701fe314-1---d8754z-;rport=57916 ? ?From: ;tag=96753574 ? ?To: ;tag=PbULiimW0TlP ? ?Call-ID: Yzg3Y2M4OGNmNDQ3ODhhMjU4OGVhZjE5NDYyOGI4OWE. ? ?CSeq: 1 SUBSCRIBE ? ?Contact: ? ?Expires: 300 ? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120622T231506Z~76fae0cec0+unclean~20120625T042938Z ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ? ?Supported: timer, precondition, path, replaces ? ?Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer ? ?Subscription-State: active;expires=300 ? ?Content-Length: 0 ? ?------------------------------------------------------------------------ send 1030 bytes to udp/[192.168.18.120]:57916 at 11:02:41.064419: ? ?------------------------------------------------------------------------ ? ?NOTIFY sip:1000 at 192.168.18.120:57916;rinstance=12e37149747f3c4b SIP/2.0 ? ?Via: SIP/2.0/UDP 192.168.10.89;rport;branch=z9hG4bK494QZ0eB01Z4D ? ?Route: ;rinstance=12e37149747f3c4b ? ?Max-Forwards: 70 ? ?From: ;tag=pjrmjQym094jH ? ?To: ? ?Call-ID: 730f6d3e-3fa1-1230-cbb5-005056945e65 ? ?CSeq: 30320616 NOTIFY ? ?Contact: ? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120622T231506Z~76fae0cec0+unclean~20120625T042938Z ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ? ?Supported: timer, precondition, path, replaces ? ?Event: message-summary ? ?Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer ? ?Subscription-State: terminated;reason=noresource ? ?Content-Type: application/simple-message-summary ? ?Content-Length: 65 ? ?Messages-Waiting: no ? ?Message-Account: sip:1000 at 192.168.10.89 ? ?------------------------------------------------------------------------ send 1005 bytes to udp/[192.168.18.120]:57916 at 11:02:41.164620: ? ?------------------------------------------------------------------------ ? ?NOTIFY sip:1000 at 192.168.18.120:57916 SIP/2.0 ? ?Via: SIP/2.0/UDP 192.168.10.89;rport;branch=z9hG4bK5jyg1UZeXapQS ? ?Route: ;transport=udp ? ?Max-Forwards: 70 ? ?From: ;tag=PbULiimW0TlP ? ?To: ;tag=96753574 ? ?Call-ID: Yzg3Y2M4OGNmNDQ3ODhhMjU4OGVhZjE5NDYyOGI4OWE. ? ?CSeq: 286150130 NOTIFY ? ?Contact: ? ?User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120622T231506Z~76fae0cec0+unclean~20120625T042938Z ? ?Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE ? ?Supported: timer, precondition, path, replaces ? ?Event: message-summary ? ?Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer ? ?Subscription-State: active;expires=300 ? ?Content-Type: application/simple-message-summary ? ?Content-Length: 65 ? ?Messages-Waiting: no ? ?Message-Account: sip:1000 at 192.168.10.89 ? ?------------------------------------------------------------------------ recv 355 bytes from udp/[192.168.18.120]:57916 at 11:02:41.166525: ? ?------------------------------------------------------------------------ ? ?SIP/2.0 200 OK ? ?Via: SIP/2.0/UDP 192.168.10.89;rport=5060;branch=z9hG4bK494QZ0eB01Z4D ? ?Contact: ? ?To: ;tag=2e2a6254 ? ?From: ;tag=pjrmjQym094jH ? ?Call-ID: 730f6d3e-3fa1-1230-cbb5-005056945e65 ? ?CSeq: 30320616 NOTIFY ? ?User-Agent: eyeBeam release 1102q stamp 51814 ? ?Content-Length: 0 ? ?------------------------------------------------------------------------ recv 368 bytes from udp/[192.168.18.120]:57916 at 11:02:41.267410: ? ?------------------------------------------------------------------------ ? ?SIP/2.0 200 OK ? ?Via: SIP/2.0/UDP 192.168.10.89;rport=5060;branch=z9hG4bK5jyg1UZeXapQS ? ?Contact: ? ?To: ;tag=96753574 ? ?From: ;tag=PbULiimW0TlP ? ?Call-ID: Yzg3Y2M4OGNmNDQ3ODhhMjU4OGVhZjE5NDYyOGI4OWE. ? ?CSeq: 286150130 NOTIFY ? ?User-Agent: eyeBeam release 1102q stamp 51814 ? ?Content-Length: 0 ________________________________ From: Peter Olsson To: 'FreeSWITCH Users Help' Sent: Tuesday, July 3, 2012 3:14 PM Subject: Re: [Freeswitch-users] how to get the content of sip header ? Is this some kind of xml-generating script for register bindings (as in here: http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration)? If that?s the case, you have no session object, and you will have to follow the information on that wiki page instead. ? /Peter ? Fr?n:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh Skickat: den 3 juli 2012 12:16 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] how to get the content of sip header ? ? for example i want to read the ?value of ?sip-header '?CSeq' ? in lua(that is configure for mod_xml in /usr/loca/freeswitch/conf/autoload/lia.xml.conf), must to issue as follow: local var = session:getVariable("sip_CSeq") but the error occure: ?attempt to index global 'session' (a nil value) ? ? ? ? ?REGISTER sip:192.168.10.89 SIP/2.0 ? ?Via: SIP/2.0/UDP 192.168.18.120:33648;branch=z9hG4bK-d8754z-7e634b5c7a239f6b-1---d8754z-;rport ? ?Max-Forwards: 70 ? ?Contact: ? ?To: ? ?From: ;tag=695adb0d ? ?Call-ID: MGUzNTFkZWVlOTcwODBmOWViNGY4MjM3MzE2NTQzMTM. ? ?CSeq: 1 REGISTER ? ?Expires: 5454 ? ?Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO ? ?Supported: timer ? ?User-Agent: eyeBeam release 1102q stamp 51814 ? ?Content-Length: 0 ? ? ________________________________ From:Peter Olsson To: FreeSWITCH Users Help Sent: Sunday, July 1, 2012 11:33 AM Subject: Re: [Freeswitch-users] how to get the content of sip header ? Use session:getVariable("var") to read whatever channel variable you need. As Michael said, the SIP headers are stored like sip_header_name. For instance, to read the header Diversion:, use the variable sip_diversion. Michael also mentions the info app, which is a good way to dump all channel variables for a channel, so you know exactly what you have available. To use this, just execute the app info in the dialplan. /Peter 1 jul 2012 kl. 05:47 skrev "Samira Mh" >: is it possible to explain in details? ,i have review the wiki on subject but don't understand clearly, i am new on freeswitch, thanks ... ________________________________ From: Michael Collins > To: FreeSWITCH Users Help > Sent: Sunday, July 1, 2012 1:33 AM Subject: Re: [Freeswitch-users] how to get the content of sip header ? See if you have a bunch of sip_xxx channel variables. Send the call to the info app and see what you've got. -MC On Sat, Jun 30, 2012 at 5:57 AM, Samira Mh > wrote: hi, how to get the content of sip header? in lua? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4fefc4ee32761372387519! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff2c56532769990114684! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120703/0f82d7ab/attachment-0001.html From peter.olsson at visionutveckling.se Tue Jul 3 15:56:39 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 3 Jul 2012 11:56:39 +0000 Subject: [Freeswitch-users] how to solve the error Rejected by acl "domains". Falling back to Digest auth? Message-ID: <1FFF97C269757C458224B7C895F35F1512C1D9@cantor.std.visionutv.se> Ok, then your manager is wrong :) Actually, I don't know enough to move this discussion forward, maybe anyone else can help out. However, for the specific problem we're talking about here, I can't see anything else than it's the software in the phone that causes problems. But I'm not really sure that the solution you want is even possible to do - according to me I don't think it is, at least not in a 100% working way. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh Skickat: den 3 juli 2012 07:48 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] how to solve the error Rejected by acl "domains". Falling back to Digest auth? i have to limit the count of registerations because that feature is exactly what my manager want !:( so i must to implement it correctly ... ________________________________ From: Peter Olsson > To: FreeSWITCH Users Help > Sent: Tuesday, July 3, 2012 9:58 AM Subject: Re: [Freeswitch-users] how to solve the error Rejected by acl "domains". Falling back to Digest auth? Yes, I know. That code does exactly what you want. It's the feature itself that is more questionable... Personally I don't believe this will ever work, since there are so many possibilities for timing issues for the registration. As MC said earlier on this discussion, you need to rethink if this is something you really want. When it comes to security this will not, in any way, increase the security for you. If you're unlucky it might do the opposite, leaving the phone registered to the "theif", and leave the "real" user blocked from registering. Keep your accounts secure, and this shouldn't be needed. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Samira Mh [saami_mh at ymail.com] Skickat: den 3 juli 2012 05:48 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] how to solve the error Rejected by acl "domains". Falling back to Digest auth? hi Peter, this feature(max-registrations-per-extension) is embeded in the sofia_reg.c as follow : ............ if (max_registrations_perext > 0 && (sip && sip->sip_contact && (sip->sip_contact->m_expires == NULL || atol(sip->sip_contact->m_expires) > 0))) { /* if expires is null still process */ /* expires == 0 means the phone is going to unregiser, so don't count against max */ uint32_t count = 0; call_id = sip->sip_call_id->i_id; switch_assert(call_id); sql = switch_mprintf("select count(sip_user) from sip_registrations where sip_user='%q' AND call_id <> '%q'", username, call_id); switch_assert(sql != NULL); sofia_glue_execute_sql_callback(profile, NULL, sql, sofia_reg_regcount_callback, &count); free(sql); if (count + 1> max_registrations_perext) { ret = AUTH_FORBIDDEN; if (sofia_test_pflag(profile, PFLAG_LOG_AUTH_FAIL)) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_WARNING, "SIP auth failure (REGISTER) due to reaching max allowed registrations. Count: %d\n", count); } goto end; } } ............. ________________________________ From: Peter Olsson > To: FreeSWITCH Users Help > Sent: Monday, July 2, 2012 9:35 PM Subject: Re: [Freeswitch-users] how to solve the error Rejected by acl "domains". Falling back to Digest auth? It seems the phone tries to register, even though it is already registered according to FS. So if you really want to use this method, I think you must do further debugging on the phone... As mentioned before on this list, I don't think this is a good approach, and since noone had even heard about this variable before, I'm guessing it's not widely used. However, in this particular example, FS is just doing exactly what it has been told to do, only to allow a registration if it doesn't exist already. /Peter 2 jul 2012 kl. 18:53 skrev "Samira Mh" >>>>: hi guys, please let me know paste myconfigurations as follow: 1- vim /usr/local/freeswitch/conf/directory/default/v_212263612400.xml 2-vim /usr/local/freeswitch/conf/autoload_configs/acl.conf.xml *max-registrations-per-extension =1 when issue the following command the user is registered so another user with the extension '2122636124' couldn't register simultaneously : sofia status profile internal reg Now what is problem? every time i want to dial some extensions that is configured in my dialplan the following erroe occure: 2012-07-02 21:02:12.050074 [WARNING] sofia_reg.c:1471 SIP auth challenge (REGISTER) on sofia profile 'internal_private' for [212263612400 at 192.168.10.70] from ip 192.168.18.120 2012-07-02 21:02:14.710049 [DEBUG] sofia.c:7904 IP 192.168.18.120 Rejected by acl "domains". Falling back to Digest auth. 2012-07-02 21:02:14.710049 [WARNING] sofia_reg.c:1471 SIP auth challenge (INVITE) on sofia profile 'internal_private' for [00989191949637 at 192.168.10.70] from ip 192.168.18.120 2012-07-02 21:02:14.710049 [DEBUG] sofia.c:7904 IP 192.168.18.120 Rejected by acl "domains". Falling back to Digest auth. 2012-07-02 21:02:14.710049 [WARNING] sofia_reg.c:2607 SIP auth failure (REGISTER) due to reaching max allowed registrations. Count: 1 2012-07-02 21:02:14.710049 [WARNING] sofia_reg.c:1416 SIP auth failure (INVITE) on sofia profile 'internal_private' for [00989191949637 at 192.168.10.70] from ip 192.168.18.120 so if i remove the line from /usr/local/freeswitch/conf/directory/default/v_212263612400.xml or from within /usr/local/freeswitch/conf/sip_profiles/internal.xml(it is posible to defined either internal.xml or /usr/local/freeswitch/conf/directory/default/v_212263612400.xml) the problem solved --with the same settings on vim /usr/local/freeswitch/conf/autoload_configs/acl.conf.xml-- but simultaneously registeration per extension couldn't worked properly ,,, plz help, what is problem on my settings? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org>>> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff26ae632762054019490! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff2867f32765777574751! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/2b2ffae8/attachment-0001.html From peter.olsson at visionutveckling.se Tue Jul 3 15:58:38 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 3 Jul 2012 11:58:38 +0000 Subject: [Freeswitch-users] Fw: how to get the content of sip header ? Message-ID: <1FFF97C269757C458224B7C895F35F1512C213@cantor.std.visionutv.se> Not all headers are available, and some have been parsed for you. For instance Contact header is named sip_contact_user and sip_contact_host /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh Skickat: den 3 juli 2012 13:24 Till: Free SWITCH Users Help ?mne: [Freeswitch-users] Fw: how to get the content of sip header ? thanks Peter, yes , i have followed the page : http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration my configurations as follow: in register.lua script , i have authenticared and registered users usig lua script instead of using in-memory method, so the sip header initiatiate in lua script as follow: (i don't now why CSeq/contact/, etc left empty)... recv 551 bytes from udp/[192.168.18.120]:57916 at 11:02:40.742843: ------------------------------------------------------------------------ REGISTER sip:192.168.10.89 SIP/2.0 Via: SIP/2.0/UDP 192.168.18.120:57916;branch=z9hG4bK-d8754z-7d027150e23ed159-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: ;tag=0e70ba56 Call-ID: NjU3NzZhZjYwNGUxNWZmOTZmN2VjYWI0NTM5NDEyYjE. CSeq: 1 REGISTER Expires: 5454 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: timer User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ------------------------------------------------------------------------ send 695 bytes to udp/[192.168.18.120]:57916 at 11:02:40.743818: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.18.120:57916;branch=z9hG4bK-d8754z-7d027150e23ed159-1---d8754z-;rport=57916 From: ;tag=0e70ba56 To: ;tag=KQcaD6Ba9e2te Call-ID: NjU3NzZhZjYwNGUxNWZmOTZmN2VjYWI0NTM5NDEyYjE. CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120622T231506Z~76fae0cec0+unclean~20120625T042938Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="192.168.10.89", nonce="9b803030-c4fe-11e1-ad8a-099620eb6996", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ recv 802 bytes from udp/[192.168.18.120]:57916 at 11:02:40.947421: ------------------------------------------------------------------------ REGISTER sip:192.168.10.89 SIP/2.0 Via: SIP/2.0/UDP 192.168.18.120:57916;branch=z9hG4bK-d8754z-8717e047bb6d4d76-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: ;tag=0e70ba56 Call-ID: NjU3NzZhZjYwNGUxNWZmOTZmN2VjYWI0NTM5NDEyYjE. CSeq: 2 REGISTER Expires: 5454 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: timer User-Agent: eyeBeam release 1102q stamp 51814 Authorization: Digest username="1000",realm="192.168.10.89",nonce="9b803030-c4fe-11e1-ad8a-099620eb6996",uri="sip:192.168.10.89",response="2044cb3f7f932d83402d6a38b26cd5e6",cnonce="2290ee958a69ed51c8bb638ab4341cef",nc=00000001,qop=auth,algorithm=MD5 Content-Length: 0 ------------------------------------------------------------------------ 2012-07-03 15:32:40.940760 [NOTICE] switch_cpp.cpp:1227 Debug from gen_dir_user_xml.lua, provided params: 'Event-Name: REQUEST_PARAMS Core-UUID: 83d1034e-c4f2-11e1-ad58-099620eb6996 FreeSWITCH-Hostname: PBX FreeSWITCH-Switchname: PBX FreeSWITCH-IPv4: 192.168.10.89 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-07-03%2015%3A32%3A40 Event-Date-GMT: Tue,%2003%20Jul%202012%2011%3A02%3A40%20GMT Event-Date-Timestamp: 1341313360940760 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_parse_auth Event-Calling-Line-Number: 2364 Event-Sequence: 3304 action: sip_auth sip_profile: internal sip_user_agent: eyeBeam%20release%201102q%20stamp%2051814 sip_auth_username: 1000 sip_auth_realm: 192.168.10.89 sip_auth_nonce: 9b803030-c4fe-11e1-ad8a-099620eb6996 sip_auth_uri: sip%3A192.168.10.89 sip_contact_user: 1000 sip_contact_host: 192.168.18.120 sip_to_user: 1000 sip_to_host: 192.168.10.89 sip_from_user: 1000 sip_from_host: 192.168.10.89 sip_request_host: 192.168.10.89 sip_auth_qop: auth sip_auth_cnonce: 2290ee958a69ed51c8bb638ab4341cef sip_auth_nc: 00000001 sip_auth_response: 2044cb3f7f932d83402d6a38b26cd5e6 sip_auth_method: REGISTER key: id user: 1000 domain: 192.168.10.89 ip: 192.168.18.120 ' 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 profile internal 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 useragent eyeBeam release 1102q stamp 51814 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 username 1000 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 req_auth_realm 192.168.10.89 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 contact_user 1000 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 contact_host 192.168.18.120 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 _to_host 192.168.10.89 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 from_host 192.168.10.89 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 request_host 192.168.10.89 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 req_domain 192.168.10.89 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227 req_ip 192.168.18.120 2012-07-03 15:32:40.940760 [INFO] switch_cpp.cpp:1227
send 683 bytes to udp/[192.168.18.120]:57916 at 11:02:40.955909: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.18.120:57916;branch=z9hG4bK-d8754z-8717e047bb6d4d76-1---d8754z-;rport=57916 From: ;tag=0e70ba56 To: ;tag=m052e1vD6QrDa Call-ID: NjU3NzZhZjYwNGUxNWZmOTZmN2VjYWI0NTM5NDEyYjE. CSeq: 2 REGISTER Contact: ;expires=5454 Date: Tue, 03 Jul 2012 11:02:40 GMT User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120622T231506Z~76fae0cec0+unclean~20120625T042938Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 554 bytes from udp/[192.168.18.120]:57916 at 11:02:41.060557: ------------------------------------------------------------------------ SUBSCRIBE sip:1000 at 192.168.10.89 SIP/2.0 Via: SIP/2.0/UDP 192.168.18.120:57916;branch=z9hG4bK-d8754z-ac23c509701fe314-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: ;tag=96753574 Call-ID: Yzg3Y2M4OGNmNDQ3ODhhMjU4OGVhZjE5NDYyOGI4OWE. CSeq: 1 SUBSCRIBE Expires: 300 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: timer User-Agent: eyeBeam release 1102q stamp 51814 Event: message-summary Content-Length: 0 ------------------------------------------------------------------------ send 805 bytes to udp/[192.168.18.120]:57916 at 11:02:41.061900: ------------------------------------------------------------------------ SIP/2.0 202 Accepted Via: SIP/2.0/UDP 192.168.18.120:57916;branch=z9hG4bK-d8754z-ac23c509701fe314-1---d8754z-;rport=57916 From: ;tag=96753574 To: ;tag=PbULiimW0TlP Call-ID: Yzg3Y2M4OGNmNDQ3ODhhMjU4OGVhZjE5NDYyOGI4OWE. CSeq: 1 SUBSCRIBE Contact: Expires: 300 User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120622T231506Z~76fae0cec0+unclean~20120625T042938Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=300 Content-Length: 0 ------------------------------------------------------------------------ send 1030 bytes to udp/[192.168.18.120]:57916 at 11:02:41.064419: ------------------------------------------------------------------------ NOTIFY sip:1000 at 192.168.18.120:57916;rinstance=12e37149747f3c4b SIP/2.0 Via: SIP/2.0/UDP 192.168.10.89;rport;branch=z9hG4bK494QZ0eB01Z4D Route: ;rinstance=12e37149747f3c4b Max-Forwards: 70 From: ;tag=pjrmjQym094jH To: Call-ID: 730f6d3e-3fa1-1230-cbb5-005056945e65 CSeq: 30320616 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120622T231506Z~76fae0cec0+unclean~20120625T042938Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=noresource Content-Type: application/simple-message-summary Content-Length: 65 Messages-Waiting: no Message-Account: sip:1000 at 192.168.10.89 ------------------------------------------------------------------------ send 1005 bytes to udp/[192.168.18.120]:57916 at 11:02:41.164620: ------------------------------------------------------------------------ NOTIFY sip:1000 at 192.168.18.120:57916 SIP/2.0 Via: SIP/2.0/UDP 192.168.10.89;rport;branch=z9hG4bK5jyg1UZeXapQS Route: ;transport=udp Max-Forwards: 70 From: ;tag=PbULiimW0TlP To: ;tag=96753574 Call-ID: Yzg3Y2M4OGNmNDQ3ODhhMjU4OGVhZjE5NDYyOGI4OWE. CSeq: 286150130 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120622T231506Z~76fae0cec0+unclean~20120625T042938Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: active;expires=300 Content-Type: application/simple-message-summary Content-Length: 65 Messages-Waiting: no Message-Account: sip:1000 at 192.168.10.89 ------------------------------------------------------------------------ recv 355 bytes from udp/[192.168.18.120]:57916 at 11:02:41.166525: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.89;rport=5060;branch=z9hG4bK494QZ0eB01Z4D Contact: To: ;tag=2e2a6254 From: ;tag=pjrmjQym094jH Call-ID: 730f6d3e-3fa1-1230-cbb5-005056945e65 CSeq: 30320616 NOTIFY User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ------------------------------------------------------------------------ recv 368 bytes from udp/[192.168.18.120]:57916 at 11:02:41.267410: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.10.89;rport=5060;branch=z9hG4bK5jyg1UZeXapQS Contact: To: ;tag=96753574 From: ;tag=PbULiimW0TlP Call-ID: Yzg3Y2M4OGNmNDQ3ODhhMjU4OGVhZjE5NDYyOGI4OWE. CSeq: 286150130 NOTIFY User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ________________________________ From: Peter Olsson To: 'FreeSWITCH Users Help' Sent: Tuesday, July 3, 2012 3:14 PM Subject: Re: [Freeswitch-users] how to get the content of sip header ? Is this some kind of xml-generating script for register bindings (as in here: http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration)? If that?s the case, you have no session object, and you will have to follow the information on that wiki page instead. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Samira Mh Skickat: den 3 juli 2012 12:16 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] how to get the content of sip header ? for example i want to read the value of sip-header ' CSeq' in lua(that is configure for mod_xml in /usr/loca/freeswitch/conf/autoload/lia.xml.conf), must to issue as follow: local var = session:getVariable("sip_CSeq") but the error occure: attempt to index global 'session' (a nil value) REGISTER sip:192.168.10.89 SIP/2.0 Via: SIP/2.0/UDP 192.168.18.120:33648;branch=z9hG4bK-d8754z-7e634b5c7a239f6b-1---d8754z-;rport Max-Forwards: 70 Contact: To: From: ;tag=695adb0d Call-ID: MGUzNTFkZWVlOTcwODBmOWViNGY4MjM3MzE2NTQzMTM. CSeq: 1 REGISTER Expires: 5454 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Supported: timer User-Agent: eyeBeam release 1102q stamp 51814 Content-Length: 0 ________________________________ From: Peter Olsson > To: FreeSWITCH Users Help > Sent: Sunday, July 1, 2012 11:33 AM Subject: Re: [Freeswitch-users] how to get the content of sip header ? Use session:getVariable("var") to read whatever channel variable you need. As Michael said, the SIP headers are stored like sip_header_name. For instance, to read the header Diversion:, use the variable sip_diversion. Michael also mentions the info app, which is a good way to dump all channel variables for a channel, so you know exactly what you have available. To use this, just execute the app info in the dialplan. /Peter 1 jul 2012 kl. 05:47 skrev "Samira Mh" >>: is it possible to explain in details? ,i have review the wiki on subject but don't understand clearly, i am new on freeswitch, thanks ... ________________________________ From: Michael Collins >> To: FreeSWITCH Users Help >> Sent: Sunday, July 1, 2012 1:33 AM Subject: Re: [Freeswitch-users] how to get the content of sip header ? See if you have a bunch of sip_xxx channel variables. Send the call to the info app and see what you've got. -MC On Sat, Jun 30, 2012 at 5:57 AM, Samira Mh >> wrote: hi, how to get the content of sip header in lua? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff2c56532769990114684! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff2d5bc32762911420139! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/46b0731c/attachment-0001.html From kris at kriskinc.com Tue Jul 3 17:26:48 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 3 Jul 2012 09:26:48 -0400 Subject: [Freeswitch-users] Multiple 183 SDP Change Codecs - allowed? In-Reply-To: References: Message-ID: Avi, Multiple 18x responses that change the SDP are allowed. I can't find the specific document text now but as a random guy on the internet (for whatever that's worth) I'm certain it is allowed. A re-INVITE can't be sent from either side until the dialog has been established (200+ACK). In a case where the UAC (caller) would like to update the session before it is established method UPDATE must be used. On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus wrote: > Are multiple 183s from the endpoint that changes the SDP allowed? I'm under > the impression this is broken, similar to > http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp > ... which is why when the codec changes, FS freaks out and cancels the call > because of codec negotiation error. > Does that NDLB flag allow this, too? > > First has: audio 10116 RTP/AVP 0 101 13 > Second has: audio 49020 RTP/AVP 8 13 101 > PCAP: http://ge.tt/7MpyBwJ > > Can someone point me to the specific RFC so I can tell the supplier to fix > it? > And just curious.. what would make this allowed? A re-INVITE..? > > -Avi > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 From avi at avimarcus.net Tue Jul 3 17:38:09 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 3 Jul 2012 16:38:09 +0300 Subject: [Freeswitch-users] Multiple 183 SDP Change Codecs - allowed? In-Reply-To: References: Message-ID: If so.. which I think is not.. is this a bug then in FS? http://pastebin.freeswitch.org/19423 You see from the first SDP that PCMU and PCMA were both options. Then for the next SDP FreeSWITCH was only considering PCMU which was the one that got chosen by the first SDP. Voxbeam claims it's not their fault.. is it? -Avi On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner wrote: > Avi, > > Multiple 18x responses that change the SDP are allowed. I can't > find the specific document text now but as a random guy on the > internet (for whatever that's worth) I'm certain it is allowed. > > A re-INVITE can't be sent from either side until the dialog has been > established (200+ACK). In a case where the UAC (caller) would like to > update the session before it is established method UPDATE must be > used. > > On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus wrote: > > Are multiple 183s from the endpoint that changes the SDP allowed? I'm > under > > the impression this is broken, similar to > > http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp > > ... which is why when the codec changes, FS freaks out and cancels the > call > > because of codec negotiation error. > > Does that NDLB flag allow this, too? > > > > First has: audio 10116 RTP/AVP 0 101 13 > > Second has: audio 49020 RTP/AVP 8 13 101 > > PCAP: http://ge.tt/7MpyBwJ > > > > Can someone point me to the specific RFC so I can tell the supplier to > fix > > it? > > And just curious.. what would make this allowed? A re-INVITE..? > > > > -Avi > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120703/8b1604e1/attachment.html From joohny at mail.ru Tue Jul 3 17:43:25 2012 From: joohny at mail.ru (=?UTF-8?B?0JXQstCz0LXQvdC40Lk=?=) Date: Tue, 03 Jul 2012 17:43:25 +0400 Subject: [Freeswitch-users] =?utf-8?q?How_about_a_USER_FORUM_and_kill_off_?= =?utf-8?q?the_mail_list=3F?= Message-ID: <1341323005.525923853@f90.mail.ru> Hi, It will be a forum in Russian, first off all. Also I can help in English too (a little). If somebody can help for English users too, it would be very nice. Mailing list is a good thing, but a forum is much better to me. All those who think this way - welcome to freeswitchforum.com. If this mail topic appears from time to time - so, many people are interested in it. I try to give an opportunity and not a holy war. Evginey ? ????????? ???????. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/56619ec2/attachment.html From jeff at jefflenk.com Tue Jul 3 17:50:38 2012 From: jeff at jefflenk.com (Jeff Lenk) Date: Tue, 3 Jul 2012 06:50:38 -0700 (PDT) Subject: [Freeswitch-users] Help with google voice In-Reply-To: References: <1341173526708-7580436.post@n2.nabble.com> Message-ID: <1341323438586-7580493.post@n2.nabble.com> Please have a look at the wiki page and verify I represented your information correctly and in the proper location. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Help-with-google-voice-tp7580430p7580493.html Sent from the freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Tue Jul 3 18:39:10 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 03 Jul 2012 09:39:10 -0500 Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail list? In-Reply-To: <1341323005.525923853@f90.mail.ru> Message-ID: You cant use FreeSWITCHForum.com as a domain name without violating FreeSWITCH Trademarks... Now you are putting the project maintainers in an awkward position as you can not keep your trademarks protected unless you defend them... On 7/3/12 8:43 AM, "???????" wrote: > Hi, > > It will be a forum in Russian, first off all. Also I can help in English too > (a little). If somebody can help for English users too, it would be very nice. > Mailing list is a good thing, but a forum is much better to me. All those who > think this way - welcome to freeswitchforum.com. > If this mail topic appears from time to time - so, many people are interested > in it. I try to give an opportunity and not a holy war. > > Evginey > ? ????????? ???????. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120703/ab60d46c/attachment.html From steveayre at gmail.com Tue Jul 3 18:46:19 2012 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 3 Jul 2012 15:46:19 +0100 Subject: [Freeswitch-users] Multiple 183 SDP Change Codecs - allowed? In-Reply-To: References: Message-ID: Looking at lines 75-78 you have PCMA enabled on FS as a codec option but not PCMU. When it goes to change to the PCMU codec it looks through the enabled codecs but doesn't find PCMU so hangs up the call with INCOMPATIBLE_DESTINATION. You need to add both PCMA and PCMU in your codec preferences. -Steve On 3 July 2012 14:38, Avi Marcus wrote: > If so.. which I think is not.. is this a bug then in FS? > > http://pastebin.freeswitch.org/19423 > You see from the first SDP that PCMU and PCMA were both options. > Then for the next SDP FreeSWITCH was only considering PCMU which was the > one that got chosen by the first SDP. > > Voxbeam claims it's not their fault.. is it? > > -Avi > > > > On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner wrote: > >> Avi, >> >> Multiple 18x responses that change the SDP are allowed. I can't >> find the specific document text now but as a random guy on the >> internet (for whatever that's worth) I'm certain it is allowed. >> >> A re-INVITE can't be sent from either side until the dialog has been >> established (200+ACK). In a case where the UAC (caller) would like to >> update the session before it is established method UPDATE must be >> used. >> >> On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus wrote: >> > Are multiple 183s from the endpoint that changes the SDP allowed? I'm >> under >> > the impression this is broken, similar to >> > http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp >> > ... which is why when the codec changes, FS freaks out and cancels the >> call >> > because of codec negotiation error. >> > Does that NDLB flag allow this, too? >> > >> > First has: audio 10116 RTP/AVP 0 101 13 >> > Second has: audio 49020 RTP/AVP 8 13 101 >> > PCAP: http://ge.tt/7MpyBwJ >> > >> > Can someone point me to the specific RFC so I can tell the supplier to >> fix >> > it? >> > And just curious.. what would make this allowed? A re-INVITE..? >> > >> > -Avi >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120703/5dd33246/attachment-0001.html From steveayre at gmail.com Tue Jul 3 18:50:10 2012 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 3 Jul 2012 15:50:10 +0100 Subject: [Freeswitch-users] Multiple 183 SDP Change Codecs - allowed? In-Reply-To: References: Message-ID: Actually rereading the trace, it looks like you have both PCMU & PCMA enabled on lines 39-40. AviMarcus, Can you confirm your codec preferences to be sure? If so, it appears to only be comparing the codec in the 2nd SDP to the codec that's been selected for the call, not to all the codecs enabled on the profile. That does sound like a bug, unless there's some parameter set that's intentionally causing that behaviour... -Steve On 3 July 2012 15:46, Steven Ayre wrote: > Looking at lines 75-78 you have PCMA enabled on FS as a codec option but > not PCMU. > > When it goes to change to the PCMU codec it looks through the enabled > codecs but doesn't find PCMU so hangs up the call with > INCOMPATIBLE_DESTINATION. > > You need to add both PCMA and PCMU in your codec preferences. > > -Steve > > > > > On 3 July 2012 14:38, Avi Marcus wrote: > >> If so.. which I think is not.. is this a bug then in FS? >> >> http://pastebin.freeswitch.org/19423 >> You see from the first SDP that PCMU and PCMA were both options. >> Then for the next SDP FreeSWITCH was only considering PCMU which was the >> one that got chosen by the first SDP. >> >> Voxbeam claims it's not their fault.. is it? >> >> -Avi >> >> >> >> On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner wrote: >> >>> Avi, >>> >>> Multiple 18x responses that change the SDP are allowed. I can't >>> find the specific document text now but as a random guy on the >>> internet (for whatever that's worth) I'm certain it is allowed. >>> >>> A re-INVITE can't be sent from either side until the dialog has been >>> established (200+ACK). In a case where the UAC (caller) would like to >>> update the session before it is established method UPDATE must be >>> used. >>> >>> On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus wrote: >>> > Are multiple 183s from the endpoint that changes the SDP allowed? I'm >>> under >>> > the impression this is broken, similar to >>> > http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp >>> > ... which is why when the codec changes, FS freaks out and cancels the >>> call >>> > because of codec negotiation error. >>> > Does that NDLB flag allow this, too? >>> > >>> > First has: audio 10116 RTP/AVP 0 101 13 >>> > Second has: audio 49020 RTP/AVP 8 13 101 >>> > PCAP: http://ge.tt/7MpyBwJ >>> > >>> > Can someone point me to the specific RFC so I can tell the supplier to >>> fix >>> > it? >>> > And just curious.. what would make this allowed? A re-INVITE..? >>> > >>> > -Avi >>> > >>> > >>> _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > Join Us At ClueCon - Aug 7-9, 2012 >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at 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 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20120703/26aaf2ab/attachment.html From avi at avimarcus.net Tue Jul 3 19:02:06 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 3 Jul 2012 18:02:06 +0300 Subject: [Freeswitch-users] Multiple 183 SDP Change Codecs - allowed? In-Reply-To: References: Message-ID: Indeed both are enabled... check the PCAP from the original message: http://ge.tt/7MpyBwJ It shows I'm offering in the original invite: audio 28380 RTP/AVP *8 0* 9 101 13 I remember this similar issue with 18x have a different codec than a 200 and Anthony said it was against the RFC but http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp would allow that broken behavior. I'm wondering if this is the same thing if so what RFC to point the vendor to. Would there be a potential negative to turning on this NDLB setting? -Avi On Tue, Jul 3, 2012 at 5:50 PM, Steven Ayre wrote: > Actually rereading the trace, it looks like you have both PCMU & PCMA > enabled on lines 39-40. > AviMarcus, Can you confirm your codec preferences to be sure? > > If so, it appears to only be comparing the codec in the 2nd SDP to the > codec that's been selected for the call, not to all the codecs enabled on > the profile. That does sound like a bug, unless there's some parameter set > that's intentionally causing that behaviour... > > -Steve > > > > > On 3 July 2012 15:46, Steven Ayre wrote: > >> Looking at lines 75-78 you have PCMA enabled on FS as a codec option but >> not PCMU. >> >> When it goes to change to the PCMU codec it looks through the enabled >> codecs but doesn't find PCMU so hangs up the call with >> INCOMPATIBLE_DESTINATION. >> >> You need to add both PCMA and PCMU in your codec preferences. >> >> -Steve >> >> >> >> >> On 3 July 2012 14:38, Avi Marcus wrote: >> >>> If so.. which I think is not.. is this a bug then in FS? >>> >>> http://pastebin.freeswitch.org/19423 >>> You see from the first SDP that PCMU and PCMA were both options. >>> Then for the next SDP FreeSWITCH was only considering PCMU which was the >>> one that got chosen by the first SDP. >>> >>> Voxbeam claims it's not their fault.. is it? >>> >>> -Avi >>> >>> >>> >>> On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner wrote: >>> >>>> Avi, >>>> >>>> Multiple 18x responses that change the SDP are allowed. I can't >>>> find the specific document text now but as a random guy on the >>>> internet (for whatever that's worth) I'm certain it is allowed. >>>> >>>> A re-INVITE can't be sent from either side until the dialog has been >>>> established (200+ACK). In a case where the UAC (caller) would like to >>>> update the session before it is established method UPDATE must be >>>> used. >>>> >>>> On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus wrote: >>>> > Are multiple 183s from the endpoint that changes the SDP allowed? I'm >>>> under >>>> > the impression this is broken, similar to >>>> > http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp >>>> > ... which is why when the codec changes, FS freaks out and cancels >>>> the call >>>> > because of codec negotiation error. >>>> > Does that NDLB flag allow this, too? >>>> > >>>> > First has: audio 10116 RTP/AVP 0 101 13 >>>> > Second has: audio 49020 RTP/AVP 8 13 101 >>>> > PCAP: http://ge.tt/7MpyBwJ >>>> > >>>> > Can someone point me to the specific RFC so I can tell the supplier >>>> to fix >>>> > it? >>>> > And just curious.. what would make this allowed? A re-INVITE..? >>>> > >>>> > -Avi >>>> > >>>> > >>>> _________________________________________________________________________ >>>> > Professional FreeSWITCH Consulting Services: >>>> > consulting at freeswitch.org >>>> > http://www.freeswitchsolutions.com >>>> > >>>> > >>>> > >>>> > >>>> > Official FreeSWITCH Sites >>>> > http://www.freeswitch.org >>>> > http://wiki.freeswitch.org >>>> > http://www.cluecon.com >>>> > >>>> > Join Us At ClueCon - Aug 7-9, 2012 >>>> > >>>> > FreeSWITCH-users mailing list >>>> > FreeSWITCH-users at 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 >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120703/2c9b167b/attachment-0001.html From kris at kriskinc.com Tue Jul 3 19:22:34 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 3 Jul 2012 11:22:34 -0400 Subject: [Freeswitch-users] Multiple 183 SDP Change Codecs - allowed? In-Reply-To: References: Message-ID: That's pretty nasty of voxbeam to do... Considering they return the first 183 less than one second after the initial INVITE I'd say it's very likely they're providing false ringback and/or FAS (false answer supervision). Meanwhile the actual ringback comes over two seconds later with PCMA. That's shady and if I were you I'd switch carriers just on principle. However, as I said before they're not doing anything that's not allowed by the various specifications although FAS is considered by many as fraud. What are your codec settings like? Perhaps FS is rejecting PCMA because you're not allowing it? On Tue, Jul 3, 2012 at 9:38 AM, Avi Marcus wrote: > If so.. which I think is not.. is this a bug then in FS? > > http://pastebin.freeswitch.org/19423 > You see from the first SDP that PCMU and PCMA were both options. > Then for the next SDP FreeSWITCH was only considering PCMU which was the one > that got chosen by the first SDP. > > Voxbeam claims it's not their fault.. is it? > > -Avi > > > > On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner > wrote: >> >> Avi, >> >> Multiple 18x responses that change the SDP are allowed. I can't >> find the specific document text now but as a random guy on the >> internet (for whatever that's worth) I'm certain it is allowed. >> >> A re-INVITE can't be sent from either side until the dialog has been >> established (200+ACK). In a case where the UAC (caller) would like to >> update the session before it is established method UPDATE must be >> used. >> >> On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus wrote: >> > Are multiple 183s from the endpoint that changes the SDP allowed? I'm >> > under >> > the impression this is broken, similar to >> > http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp >> > ... which is why when the codec changes, FS freaks out and cancels the >> > call >> > because of codec negotiation error. >> > Does that NDLB flag allow this, too? >> > >> > First has: audio 10116 RTP/AVP 0 101 13 >> > Second has: audio 49020 RTP/AVP 8 13 101 >> > PCAP: http://ge.tt/7MpyBwJ >> > >> > Can someone point me to the specific RFC so I can tell the supplier to >> > fix >> > it? >> > And just curious.. what would make this allowed? A re-INVITE..? >> > >> > -Avi >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 From kris at kriskinc.com Tue Jul 3 19:50:45 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 3 Jul 2012 11:50:45 -0400 Subject: [Freeswitch-users] Multiple 183 SDP Change Codecs - allowed? In-Reply-To: References: Message-ID: Sorry, missed the pcap in the original message. I'd try the NDLB setting and see what happens. Keep in mind what I said about the carrier and FAS... While it's not technically non-compliant behavior it reeks of either incompetence or malice. Generally speaking (in the SDP offer/answer model) as long as the remote end's answer contains codecs in your offer it is valid answer (even if there are several of them). What happens if you remove either PCMA or PCMU from your INVITE (offer)? On Tue, Jul 3, 2012 at 11:02 AM, Avi Marcus wrote: > Indeed both are enabled... check the PCAP from the original message: > http://ge.tt/7MpyBwJ > It shows I'm offering in the original invite: audio 28380 RTP/AVP 8 0 9 101 > 13 > > I remember this similar issue with 18x have a different codec than a 200 and > Anthony said it was against the RFC but > http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp would allow that > broken behavior. > I'm wondering if this is the same thing if so what RFC to point the vendor > to. > > Would there be a potential negative to turning on this NDLB setting? > > -Avi > > > > On Tue, Jul 3, 2012 at 5:50 PM, Steven Ayre wrote: >> >> Actually rereading the trace, it looks like you have both PCMU & PCMA >> enabled on lines 39-40. >> AviMarcus, Can you confirm your codec preferences to be sure? >> >> If so, it appears to only be comparing the codec in the 2nd SDP to the >> codec that's been selected for the call, not to all the codecs enabled on >> the profile. That does sound like a bug, unless there's some parameter set >> that's intentionally causing that behaviour... >> >> -Steve >> >> >> >> >> On 3 July 2012 15:46, Steven Ayre wrote: >>> >>> Looking at lines 75-78 you have PCMA enabled on FS as a codec option but >>> not PCMU. >>> >>> When it goes to change to the PCMU codec it looks through the enabled >>> codecs but doesn't find PCMU so hangs up the call with >>> INCOMPATIBLE_DESTINATION. >>> >>> You need to add both PCMA and PCMU in your codec preferences. >>> >>> -Steve >>> >>> >>> >>> >>> On 3 July 2012 14:38, Avi Marcus wrote: >>>> >>>> If so.. which I think is not.. is this a bug then in FS? >>>> >>>> http://pastebin.freeswitch.org/19423 >>>> You see from the first SDP that PCMU and PCMA were both options. >>>> Then for the next SDP FreeSWITCH was only considering PCMU which was the >>>> one that got chosen by the first SDP. >>>> >>>> Voxbeam claims it's not their fault.. is it? >>>> >>>> -Avi >>>> >>>> >>>> >>>> On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner >>>> wrote: >>>>> >>>>> Avi, >>>>> >>>>> Multiple 18x responses that change the SDP are allowed. I can't >>>>> find the specific document text now but as a random guy on the >>>>> internet (for whatever that's worth) I'm certain it is allowed. >>>>> >>>>> A re-INVITE can't be sent from either side until the dialog has been >>>>> established (200+ACK). In a case where the UAC (caller) would like to >>>>> update the session before it is established method UPDATE must be >>>>> used. >>>>> >>>>> On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus wrote: >>>>> > Are multiple 183s from the endpoint that changes the SDP allowed? I'm >>>>> > under >>>>> > the impression this is broken, similar to >>>>> > http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp >>>>> > ... which is why when the codec changes, FS freaks out and cancels >>>>> > the call >>>>> > because of codec negotiation error. >>>>> > Does that NDLB flag allow this, too? >>>>> > >>>>> > First has: audio 10116 RTP/AVP 0 101 13 >>>>> > Second has: audio 49020 RTP/AVP 8 13 101 >>>>> > PCAP: http://ge.tt/7MpyBwJ >>>>> > >>>>> > Can someone point me to the specific RFC so I can tell the supplier >>>>> > to fix >>>>> > it? >>>>> > And just curious.. what would make this allowed? A re-INVITE..? >>>>> > >>>>> > -Avi >>>>> > >>>>> > >>>>> > _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://wiki.freeswitch.org >>>>> > http://www.cluecon.com >>>>> > >>>>> > Join Us At ClueCon - Aug 7-9, 2012 >>>>> > >>>>> > FreeSWITCH-users mailing list >>>>> > FreeSWITCH-users at 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 >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> Join Us At ClueCon - Aug 7-9, 2012 >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 From avi at avimarcus.net Tue Jul 3 19:56:30 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 3 Jul 2012 18:56:30 +0300 Subject: [Freeswitch-users] Multiple 183 SDP Change Codecs - allowed? In-Reply-To: References: Message-ID: They said the 1st carrier rejected it so the second 183 is from a second carrier.. but yes, the first carrier shouldn't be sending a 183... Anyway -- if you *please* look at the pcap from the first email, the log I posted a link to later, or the past two emails -- you'll see FS originally offers both PCMU and PCMA in the initial invite. I think we need Anthony to chime in here... he knows the RFC stuff! -Avi On Tue, Jul 3, 2012 at 6:22 PM, Kristian Kielhofner wrote: > That's pretty nasty of voxbeam to do... Considering they return the > first 183 less than one second after the initial INVITE I'd say it's > very likely they're providing false ringback and/or FAS (false answer > supervision). Meanwhile the actual ringback comes over two seconds > later with PCMA. That's shady and if I were you I'd switch carriers > just on principle. > > However, as I said before they're not doing anything that's not > allowed by the various specifications although FAS is considered by > many as fraud. > > What are your codec settings like? Perhaps FS is rejecting PCMA > because you're not allowing it? > > On Tue, Jul 3, 2012 at 9:38 AM, Avi Marcus wrote: > > If so.. which I think is not.. is this a bug then in FS? > > > > http://pastebin.freeswitch.org/19423 > > You see from the first SDP that PCMU and PCMA were both options. > > Then for the next SDP FreeSWITCH was only considering PCMU which was the > one > > that got chosen by the first SDP. > > > > Voxbeam claims it's not their fault.. is it? > > > > -Avi > > > > > > > > On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner > > wrote: > >> > >> Avi, > >> > >> Multiple 18x responses that change the SDP are allowed. I can't > >> find the specific document text now but as a random guy on the > >> internet (for whatever that's worth) I'm certain it is allowed. > >> > >> A re-INVITE can't be sent from either side until the dialog has been > >> established (200+ACK). In a case where the UAC (caller) would like to > >> update the session before it is established method UPDATE must be > >> used. > >> > >> On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus wrote: > >> > Are multiple 183s from the endpoint that changes the SDP allowed? I'm > >> > under > >> > the impression this is broken, similar to > >> > http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp > >> > ... which is why when the codec changes, FS freaks out and cancels the > >> > call > >> > because of codec negotiation error. > >> > Does that NDLB flag allow this, too? > >> > > >> > First has: audio 10116 RTP/AVP 0 101 13 > >> > Second has: audio 49020 RTP/AVP 8 13 101 > >> > PCAP: http://ge.tt/7MpyBwJ > >> > > >> > Can someone point me to the specific RFC so I can tell the supplier to > >> > fix > >> > it? > >> > And just curious.. what would make this allowed? A re-INVITE..? > >> > > >> > -Avi > >> > > >> > > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > Join Us At ClueCon - Aug 7-9, 2012 > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at 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 > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120703/683d3273/attachment-0001.html From admin at blindi.net Tue Jul 3 19:58:33 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Tue, 3 Jul 2012 17:58:33 +0200 (CEST) Subject: [Freeswitch-users] question: lua how can i create a postcastlist? In-Reply-To: References: Message-ID: Hi guys, I have many potcast mp3 files in a directory: /home/postcast the files beginning with: postcast_year_mon_day_hour_minutes_.mp3 I like to create a playlist in lua: 1. read all files, and make a messagecount. 2. generate a menu to use from phone: press 1 for the next postcast. 2, To repeate the postcast 3. to go to the prev potcast 4 for the fist potcast 5 for the last potcast 6 to hear the date an time for these. I like to sort, firt in first out. I don.t find a command in lua: to read the filelist, to generate a filecount. and to generate a navigation keys: forward, backward, repeate and say the time and date of the current file. Can your help please? nice thanks to all. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From Rob.Moore at Aeriandi.com Tue Jul 3 21:05:44 2012 From: Rob.Moore at Aeriandi.com (Rob Moore) Date: Tue, 3 Jul 2012 17:05:44 +0000 Subject: [Freeswitch-users] Monitoring Success of SIP Registration Message-ID: <49C5FCA19A8A114493EBAACA42FE5899105B0A1A@1AERDCEXCHMBX1.AER.AERCO.local> Hi Everyone, Recently we have had some problems with our Freeswitch boxes failing to respond to registration requests. A restart of FS always clears this up, and I'm looking else where for a long term solutions to this problem. However this does leave me with another situation while I look for the permanent solution. Its normally during out of hours that our freeswitch boxes decide they do not want to respond to registrations and I would like to setup some rolling registrations every 2 minutes that the result of can be reported back on to allow for some Alerting to take place. This could be from another Freeswitch server (maybe using a gateway, or some API commands run in a lua script) or a small application of some sort hosted on our monitoring platform. I even considered using SIP-P but everything I find seems to be rather heavy weight for what needs to be something a simple as a few lines of code in a lua script that returns a result. So I thought I 'd open it up to the users group to see if you had any suggestions on a light weight method I could use to generate these registration request. Thanks Rob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/2a974244/attachment.html From toddb at toddbailey.net Tue Jul 3 21:13:58 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Tue, 03 Jul 2012 10:13:58 -0700 Subject: [Freeswitch-users] Fs compatible phones? Message-ID: <1341335638.5953.23.camel@mythtv> Hi All, New to this tech and a bit confused on what to buy/features etc. I'm looking for a basic desk phone, perhaps with a few speed dial and maybe gee-wiz buttons. I found this one with a decent price 3Com NBX 1102 but the ad states Requires R4.X system software or greater. Not sure what this is nor do I need to? If I buy "any" digital phone that has a rj-45 ethernet jack does that automatically mean it's compatible? other models in my price/feature range Cisco CP-7940G Unified IP Phone I was looking at a few grand stream models but the reviews lead me to move on thanks From anthony.minessale at gmail.com Tue Jul 3 21:14:33 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Jul 2012 12:14:33 -0500 Subject: [Freeswitch-users] Multiple 183 SDP Change Codecs - allowed? In-Reply-To: References: Message-ID: This is another case of what works vs what the rules say and how they should be interpreted. It for sure is not a bug because there are ways to prove its optional. I could be wrong, and like kris an operating from the top of my head but I am pretty sure: If you have 100rel on (which we don't by default because of issues in the sofia stack) then once you answer with a sdp, you must not send anything other than that exact sdp on subsequent responses. The RFC mandates you ignore any sdp that differes from the last. if you DONT have 100rel enabled, they you are allowed to completely ignore the 1xx + sdp messages. So either way we could claim not to support changing the codec in the fashion described but on top of any rules we simply do not do it by default because there are a ton of broken devices who unintentionally send a series of packets that cause us to break. NDLB-allow-nondup-sdp will unlock this to the best we support it. However once we have chosen a codec we will only support that one codec in this mode iirc. You need a new offer and renegotiate-codec-on-reinvite to change the codecs. On Tue, Jul 3, 2012 at 10:22 AM, Kristian Kielhofner wrote: > That's pretty nasty of voxbeam to do... Considering they return the > first 183 less than one second after the initial INVITE I'd say it's > very likely they're providing false ringback and/or FAS (false answer > supervision). Meanwhile the actual ringback comes over two seconds > later with PCMA. That's shady and if I were you I'd switch carriers > just on principle. > > However, as I said before they're not doing anything that's not > allowed by the various specifications although FAS is considered by > many as fraud. > > What are your codec settings like? Perhaps FS is rejecting PCMA > because you're not allowing it? > > On Tue, Jul 3, 2012 at 9:38 AM, Avi Marcus wrote: >> If so.. which I think is not.. is this a bug then in FS? >> >> http://pastebin.freeswitch.org/19423 >> You see from the first SDP that PCMU and PCMA were both options. >> Then for the next SDP FreeSWITCH was only considering PCMU which was the one >> that got chosen by the first SDP. >> >> Voxbeam claims it's not their fault.. is it? >> >> -Avi >> >> >> >> On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner >> wrote: >>> >>> Avi, >>> >>> Multiple 18x responses that change the SDP are allowed. I can't >>> find the specific document text now but as a random guy on the >>> internet (for whatever that's worth) I'm certain it is allowed. >>> >>> A re-INVITE can't be sent from either side until the dialog has been >>> established (200+ACK). In a case where the UAC (caller) would like to >>> update the session before it is established method UPDATE must be >>> used. >>> >>> On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus wrote: >>> > Are multiple 183s from the endpoint that changes the SDP allowed? I'm >>> > under >>> > the impression this is broken, similar to >>> > http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp >>> > ... which is why when the codec changes, FS freaks out and cancels the >>> > call >>> > because of codec negotiation error. >>> > Does that NDLB flag allow this, too? >>> > >>> > First has: audio 10116 RTP/AVP 0 101 13 >>> > Second has: audio 49020 RTP/AVP 8 13 101 >>> > PCAP: http://ge.tt/7MpyBwJ >>> > >>> > Can someone point me to the specific RFC so I can tell the supplier to >>> > fix >>> > it? >>> > And just curious.. what would make this allowed? A re-INVITE..? >>> > >>> > -Avi >>> > >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > Join Us At ClueCon - Aug 7-9, 2012 >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at 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 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From anthony.minessale at gmail.com Tue Jul 3 21:15:15 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Jul 2012 12:15:15 -0500 Subject: [Freeswitch-users] Multiple 183 SDP Change Codecs - allowed? In-Reply-To: References: Message-ID: if its a sip forked dial. We do not support that. On Tue, Jul 3, 2012 at 10:56 AM, Avi Marcus wrote: > They said the 1st carrier rejected it so the second 183 is from a second > carrier.. but yes, the first carrier shouldn't be sending a 183... > Anyway -- if you please look at the pcap from the first email, the log I > posted a link to later, or the past two emails -- you'll see FS originally > offers both PCMU and PCMA in the initial invite. > > I think we need Anthony to chime in here... he knows the RFC stuff! > > -Avi > > > > On Tue, Jul 3, 2012 at 6:22 PM, Kristian Kielhofner > wrote: >> >> That's pretty nasty of voxbeam to do... Considering they return the >> first 183 less than one second after the initial INVITE I'd say it's >> very likely they're providing false ringback and/or FAS (false answer >> supervision). Meanwhile the actual ringback comes over two seconds >> later with PCMA. That's shady and if I were you I'd switch carriers >> just on principle. >> >> However, as I said before they're not doing anything that's not >> allowed by the various specifications although FAS is considered by >> many as fraud. >> >> What are your codec settings like? Perhaps FS is rejecting PCMA >> because you're not allowing it? >> >> On Tue, Jul 3, 2012 at 9:38 AM, Avi Marcus wrote: >> > If so.. which I think is not.. is this a bug then in FS? >> > >> > http://pastebin.freeswitch.org/19423 >> > You see from the first SDP that PCMU and PCMA were both options. >> > Then for the next SDP FreeSWITCH was only considering PCMU which was the >> > one >> > that got chosen by the first SDP. >> > >> > Voxbeam claims it's not their fault.. is it? >> > >> > -Avi >> > >> > >> > >> > On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner >> > wrote: >> >> >> >> Avi, >> >> >> >> Multiple 18x responses that change the SDP are allowed. I can't >> >> find the specific document text now but as a random guy on the >> >> internet (for whatever that's worth) I'm certain it is allowed. >> >> >> >> A re-INVITE can't be sent from either side until the dialog has been >> >> established (200+ACK). In a case where the UAC (caller) would like to >> >> update the session before it is established method UPDATE must be >> >> used. >> >> >> >> On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus wrote: >> >> > Are multiple 183s from the endpoint that changes the SDP allowed? I'm >> >> > under >> >> > the impression this is broken, similar to >> >> > http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp >> >> > ... which is why when the codec changes, FS freaks out and cancels >> >> > the >> >> > call >> >> > because of codec negotiation error. >> >> > Does that NDLB flag allow this, too? >> >> > >> >> > First has: audio 10116 RTP/AVP 0 101 13 >> >> > Second has: audio 49020 RTP/AVP 8 13 101 >> >> > PCAP: http://ge.tt/7MpyBwJ >> >> > >> >> > Can someone point me to the specific RFC so I can tell the supplier >> >> > to >> >> > fix >> >> > it? >> >> > And just curious.. what would make this allowed? A re-INVITE..? >> >> > >> >> > -Avi >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ >> >> > Professional FreeSWITCH Consulting Services: >> >> > consulting at freeswitch.org >> >> > http://www.freeswitchsolutions.com >> >> > >> >> > >> >> > >> >> > >> >> > Official FreeSWITCH Sites >> >> > http://www.freeswitch.org >> >> > http://wiki.freeswitch.org >> >> > http://www.cluecon.com >> >> > >> >> > Join Us At ClueCon - Aug 7-9, 2012 >> >> > >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at 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 >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From paul at cupis.co.uk Tue Jul 3 21:42:38 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Tue, 03 Jul 2012 18:42:38 +0100 Subject: [Freeswitch-users] Monitoring Success of SIP Registration In-Reply-To: <49C5FCA19A8A114493EBAACA42FE5899105B0A1A@1AERDCEXCHMBX1.AER.AERCO.local> References: <49C5FCA19A8A114493EBAACA42FE5899105B0A1A@1AERDCEXCHMBX1.AER.AERCO.local> Message-ID: <4FF32F0E.60302@cupis.co.uk> On 03/07/12 18:05, Rob Moore wrote: > However this does leave me with another situation while I look for the > permanent solution. Its normally during out of hours that our freeswitch > boxes decide they do not want to respond to registrations and I would > like to setup some rolling registrations every 2 minutes that the result > of can be reported back on to allow for some Alerting to take place. ... > So I thought I ?d open it up to the users group to see if you had any > suggestions on a light weight method I could use to generate these > registration request. I'd suggest looking at tools like sipsak and possibly the Nagios plugin check_sip for this sort of functionality. http://sipsak.org/ http://exchange.nagios.org/directory/Plugins/Network-Protocols/*-VoIP/SIP/check_sip-sipsak/details Regards, From ben122uk at gmail.com Tue Jul 3 16:03:28 2012 From: ben122uk at gmail.com (Ben ben_naylor1942@hotmail.com) Date: Tue, 3 Jul 2012 13:03:28 +0100 Subject: [Freeswitch-users] ZRTP call setup Message-ID: Hi Freeswitch community! Could someone advise on how you have the a-leg as a non-ZRTP client, and the b-leg as a ZRTP client, with trusted main-in-the-middle? I have set up trusted man-in-the-middle with the server, which works well when making an outbound call from the ZRTP client to the standard SIP client. The INVITE from the ZRTP client includes the zrtp hash in the SDP info, which I'm presuming highlights to freeswitch that the call will be using ZRTP for this leg. The Freeswitch then decrypts the ZRTP ready for the b-leg. Everyone's happy. However, when going the other way, the INVITE from the non-ZRTP device does not include zrtp hash. So the INVITE passed to the ZRTP client does not include a ZRTP hash, which then causes the ZRTP client to fail. I have tried getting around this by adding the following in the dialplan - But the freeswitch doesn't create the zrtp hash in the INVITE. Is there something I can do in Freeswitch to say "all calls to this phone must use ZRTP (hash), whether the original INVITE uses ZRTP or not"? Thanks in advance for the help! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/db36bd15/attachment.html From ben122uk at gmail.com Tue Jul 3 16:18:29 2012 From: ben122uk at gmail.com (Ben ben_naylor1942@hotmail.com) Date: Tue, 3 Jul 2012 13:18:29 +0100 Subject: [Freeswitch-users] ZRTP call setup In-Reply-To: References: Message-ID: Hi All Please ignore, I had misconfigured something in the dialplan! Thanks. On Tue, Jul 3, 2012 at 1:03 PM, Ben wrote: > Hi Freeswitch community! > > Could someone advise on how you have the a-leg as a non-ZRTP client, and > the b-leg as a ZRTP client, with trusted main-in-the-middle? > > I have set up trusted man-in-the-middle with the server, which works well > when making an outbound call from the ZRTP client to the standard SIP > client. The INVITE from the ZRTP client includes the zrtp hash in the SDP > info, which I'm presuming highlights to freeswitch that the call will be > using ZRTP for this leg. The Freeswitch then decrypts the ZRTP ready for > the b-leg. Everyone's happy. > > However, when going the other way, the INVITE from the non-ZRTP device > does not include zrtp hash. So the INVITE passed to the ZRTP client does > not include a ZRTP hash, which then causes the ZRTP client to fail. I have > tried getting around this by adding the following in the dialplan - > > > > But the freeswitch doesn't create the zrtp hash in the INVITE. > > Is there something I can do in Freeswitch to say "all calls to this phone > must use ZRTP (hash), whether the original INVITE uses ZRTP or not"? > > Thanks in advance for the help! > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/531af7ed/attachment.html From leonardo at daitangroup.com Tue Jul 3 21:09:40 2012 From: leonardo at daitangroup.com (Leonardo) Date: Tue, 3 Jul 2012 14:09:40 -0300 Subject: [Freeswitch-users] How to keep one leg call in FreeSwitch In-Reply-To: References: <9438D04074E0DE45A49CD7609982127246742C4D@CORP-MAIL-002.edge.local> <63B00DD1DA6A364E9F64A3A0BD2FE7B6BA9B72@BY2PRD0710MB390.namprd07.prod.outlook.com> <1FFF97C269757C458224B7C895F35F15106C51@cantor.std.visionutv.se> <9438D04074E0DE45A49CD7609982127246742D17@CORP-MAIL-002.edge.local> <4FF1FCC0.8040107@daitangroup.com> Message-ID: <4FF32754.1090208@daitangroup.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/6e672740/attachment.html From leonardo at daitangroup.com Tue Jul 3 21:23:07 2012 From: leonardo at daitangroup.com (Leonardo) Date: Tue, 3 Jul 2012 14:23:07 -0300 Subject: [Freeswitch-users] How to keep one leg call in FreeSwitch In-Reply-To: References: <9438D04074E0DE45A49CD7609982127246742C4D@CORP-MAIL-002.edge.local> <63B00DD1DA6A364E9F64A3A0BD2FE7B6BA9B72@BY2PRD0710MB390.namprd07.prod.outlook.com> <1FFF97C269757C458224B7C895F35F15106C51@cantor.std.visionutv.se> <9438D04074E0DE45A49CD7609982127246742D17@CORP-MAIL-002.edge.local> <4FF1FCC0.8040107@daitangroup.com> Message-ID: <4FF32A7B.8020309@daitangroup.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/77871511/attachment.html From steveayre at gmail.com Tue Jul 3 21:50:49 2012 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 3 Jul 2012 18:50:49 +0100 Subject: [Freeswitch-users] Multiple 183 SDP Change Codecs - allowed? In-Reply-To: References: Message-ID: 18x messages are supposed to make the call unreroutable since they indicate the call has been successful on that route. -Steve On 3 July 2012 16:56, Avi Marcus wrote: > They said the 1st carrier rejected it so the second 183 is from a second > carrier.. but yes, the first carrier shouldn't be sending a 183... > Anyway -- if you *please* look at the pcap from the first email, the log > I posted a link to later, or the past two emails -- you'll see FS > originally offers both PCMU and PCMA in the initial invite. > > I think we need Anthony to chime in here... he knows the RFC stuff! > > -Avi > > > > On Tue, Jul 3, 2012 at 6:22 PM, Kristian Kielhofner wrote: > >> That's pretty nasty of voxbeam to do... Considering they return the >> first 183 less than one second after the initial INVITE I'd say it's >> very likely they're providing false ringback and/or FAS (false answer >> supervision). Meanwhile the actual ringback comes over two seconds >> later with PCMA. That's shady and if I were you I'd switch carriers >> just on principle. >> >> However, as I said before they're not doing anything that's not >> allowed by the various specifications although FAS is considered by >> many as fraud. >> >> What are your codec settings like? Perhaps FS is rejecting PCMA >> because you're not allowing it? >> >> On Tue, Jul 3, 2012 at 9:38 AM, Avi Marcus wrote: >> > If so.. which I think is not.. is this a bug then in FS? >> > >> > http://pastebin.freeswitch.org/19423 >> > You see from the first SDP that PCMU and PCMA were both options. >> > Then for the next SDP FreeSWITCH was only considering PCMU which was >> the one >> > that got chosen by the first SDP. >> > >> > Voxbeam claims it's not their fault.. is it? >> > >> > -Avi >> > >> > >> > >> > On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner >> > wrote: >> >> >> >> Avi, >> >> >> >> Multiple 18x responses that change the SDP are allowed. I can't >> >> find the specific document text now but as a random guy on the >> >> internet (for whatever that's worth) I'm certain it is allowed. >> >> >> >> A re-INVITE can't be sent from either side until the dialog has been >> >> established (200+ACK). In a case where the UAC (caller) would like to >> >> update the session before it is established method UPDATE must be >> >> used. >> >> >> >> On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus wrote: >> >> > Are multiple 183s from the endpoint that changes the SDP allowed? I'm >> >> > under >> >> > the impression this is broken, similar to >> >> > http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp >> >> > ... which is why when the codec changes, FS freaks out and cancels >> the >> >> > call >> >> > because of codec negotiation error. >> >> > Does that NDLB flag allow this, too? >> >> > >> >> > First has: audio 10116 RTP/AVP 0 101 13 >> >> > Second has: audio 49020 RTP/AVP 8 13 101 >> >> > PCAP: http://ge.tt/7MpyBwJ >> >> > >> >> > Can someone point me to the specific RFC so I can tell the supplier >> to >> >> > fix >> >> > it? >> >> > And just curious.. what would make this allowed? A re-INVITE..? >> >> > >> >> > -Avi >> >> > >> >> > >> >> > >> _________________________________________________________________________ >> >> > Professional FreeSWITCH Consulting Services: >> >> > consulting at freeswitch.org >> >> > http://www.freeswitchsolutions.com >> >> > >> >> > >> >> > >> >> > >> >> > Official FreeSWITCH Sites >> >> > http://www.freeswitch.org >> >> > http://wiki.freeswitch.org >> >> > http://www.cluecon.com >> >> > >> >> > Join Us At ClueCon - Aug 7-9, 2012 >> >> > >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at 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 >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at 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 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120703/ad49eb36/attachment-0001.html From kris at kriskinc.com Tue Jul 3 22:09:48 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 3 Jul 2012 14:09:48 -0400 Subject: [Freeswitch-users] Multiple 183 SDP Change Codecs - allowed? In-Reply-To: References: Message-ID: I see the original offer from FreeSWITCH. I was asking what the carrier's SBC does if you configure FreeSWITCH to only offer PCMU or PCMA. If you only offer one codec they absolutely shouldn't switch codecs in between 183s... On Tue, Jul 3, 2012 at 11:56 AM, Avi Marcus wrote: > They said the 1st carrier rejected it so the second 183 is from a second > carrier.. but yes, the first carrier shouldn't be sending a 183... > Anyway -- if you please look at the pcap from the first email, the log I > posted a link to later, or the past two emails -- you'll see FS originally > offers both PCMU and PCMA in the initial invite. > > I think we need Anthony to chime in here... he knows the RFC stuff! > > -Avi > -- Kristian Kielhofner From ben at langfeld.co.uk Tue Jul 3 22:15:12 2012 From: ben at langfeld.co.uk (Ben Langfeld) Date: Tue, 3 Jul 2012 19:15:12 +0100 Subject: [Freeswitch-users] How to keep one leg call in FreeSwitch In-Reply-To: <4FF32754.1090208@daitangroup.com> References: <9438D04074E0DE45A49CD7609982127246742C4D@CORP-MAIL-002.edge.local> <63B00DD1DA6A364E9F64A3A0BD2FE7B6BA9B72@BY2PRD0710MB390.namprd07.prod.outlook.com> <1FFF97C269757C458224B7C895F35F15106C51@cantor.std.visionutv.se> <9438D04074E0DE45A49CD7609982127246742D17@CORP-MAIL-002.edge.local> <4FF1FCC0.8040107@daitangroup.com> <4FF32754.1090208@daitangroup.com> Message-ID: You simply need to park the call: Regards, Ben Langfeld On 3 July 2012 18:09, Leonardo wrote: > Hi. > I'm newbie in FreeSwitch and I'm trying to develop an application using > mod_socket interface. > First, I configured the FreeSwitch to answer any call coming from my > gateway: > > ??? (dialplan/public.xml) > ??? > ??????? > ??????????? > ??????????????? > ??????????? > ??????? > ??? > > I can make a call from my SIP phone in FreeSwitch to PSTN number that is > routed back to the FreeSwitch and the incoming call is? answered (due to > dialplan above). The problem is that the call is answered and right after > hungup. > Is there a way to tell to the FreeSwitch to keep this incoming call active? > I would like to be able to manipulate this call via event_socket (play > audio, detect digits or even do nothing) > > > Thanks in advance, > Leo > > -- > > Leonardo N. S. Pereira, Software Engineer > T:+55.19.3112-1200 ext. 1283|F:+55.19.3207-1437 > DaitanGroup|www.daitangroup.com|Highly Reliable Outsourcing. Value Added > Services Worldwide. > Privileged and confidential. If this message has been received in error, > please notify sender and delete it immediately. > Conte?do confidencial. Se esta mensagem foi recebida por engano, favor > avisar o remetente e apag?-la imediatamente. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jrichey at itltd.net Tue Jul 3 22:22:22 2012 From: jrichey at itltd.net (JRichey) Date: Tue, 3 Jul 2012 11:22:22 -0700 Subject: [Freeswitch-users] How to keep one leg call in FreeSwitch Message-ID: <6ECAF1527329364583AB525CF34ABF950B31A68F@ms.kallback.com> Add this after the answer. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of Leonardo Sent: Tuesday, July 03, 2012 10:23 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] How to keep one leg call in FreeSwitch Hi. I'm newbie in FreeSwitch and I'm trying to develop an application using mod_socket interface. First, I configured the FreeSwitch to answer any call coming from my gateway: (dialplan/public.xml) I can make a call from my SIP phone in FreeSwitch to PSTN number that is routed back to the FreeSwitch and the incoming call is answered (due to dialplan above). The problem is that the call is answered and right after hungup. Is there a way to tell to the FreeSwitch to keep this incoming call active? I would like to be able to manipulate this call via event_socket (play audio, detect digits or even do nothing) Thanks in advance, Leo -- Leonardo N. S. Pereira, Software Engineer T:+55.19.3112-1200 ext. 1283|F:+55.19.3207-1437 DaitanGroup| www.daitangroup.com|Highly Reliable Outsourcing. Value Added Services Worldwide. Privileged and confidential. If this message has been received in error, please notify sender and delete it immediately. Conte?do confidencial. Se esta mensagem foi recebida por engano, favor avisar o remetente e apag?-la imediatamente. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/826002d7/attachment.html From roger.castaldo at gmail.com Tue Jul 3 22:22:27 2012 From: roger.castaldo at gmail.com (Roger Castaldo) Date: Tue, 3 Jul 2012 14:22:27 -0400 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: <1341335638.5953.23.camel@mythtv> References: <1341335638.5953.23.camel@mythtv> Message-ID: I have had good success with snom phones and they are solid for the price On Jul 3, 2012 12:36 PM, "Todd Bailey" wrote: > Hi All, > > New to this tech and a bit confused on what to buy/features etc. > > I'm looking for a basic desk phone, perhaps with a few speed dial and > maybe gee-wiz buttons. I found this one with a decent price 3Com NBX > 1102 but the ad states Requires R4.X system software or greater. Not > sure what this is nor do I need to? > > If I buy "any" digital phone that has a rj-45 ethernet jack does that > automatically mean it's compatible? > > > other models in my price/feature range > > Cisco CP-7940G Unified IP Phone > > I was looking at a few grand stream models but the reviews lead me to > move on > > > thanks > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120703/10735d49/attachment.html From Hector.Geraldino at ipsoft.com Tue Jul 3 22:31:46 2012 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Tue, 3 Jul 2012 14:31:46 -0400 Subject: [Freeswitch-users] How to keep one leg call in FreeSwitch In-Reply-To: <4FF32A7B.8020309@daitangroup.com> References: <9438D04074E0DE45A49CD7609982127246742C4D@CORP-MAIL-002.edge.local> <63B00DD1DA6A364E9F64A3A0BD2FE7B6BA9B72@BY2PRD0710MB390.namprd07.prod.outlook.com> <1FFF97C269757C458224B7C895F35F15106C51@cantor.std.visionutv.se> <9438D04074E0DE45A49CD7609982127246742D17@CORP-MAIL-002.edge.local> <4FF1FCC0.8040107@daitangroup.com> <4FF32A7B.8020309@daitangroup.com> Message-ID: <6A6B4C284AD15042B429EB9D904544AD022F4EC330@NY1-EXMB-01.ip-soft.net> Use ESL in outbound mode. For this, your application should be listening in a port (e.g. 8040) so you can add a line to your dialplan that passes the control of the incoming call to your application. @see http://wiki.freeswitch.org/wiki/Event_Socket_Outbound http://wiki.freeswitch.org/wiki/Event_Socket#Outbound From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Leonardo Sent: Tuesday, July 03, 2012 1:23 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] How to keep one leg call in FreeSwitch Hi. I'm newbie in FreeSwitch and I'm trying to develop an application using mod_socket interface. First, I configured the FreeSwitch to answer any call coming from my gateway: (dialplan/public.xml) I can make a call from my SIP phone in FreeSwitch to PSTN number that is routed back to the FreeSwitch and the incoming call is answered (due to dialplan above). The problem is that the call is answered and right after hungup. Is there a way to tell to the FreeSwitch to keep this incoming call active? I would like to be able to manipulate this call via event_socket (play audio, detect digits or even do nothing) Thanks in advance, Leo -- Leonardo N. S. Pereira, Software Engineer T:+55.19.3112-1200 ext. 1283|F:+55.19.3207-1437 DaitanGroup|www.daitangroup.com|Highly Reliable Outsourcing. Value Added Services Worldwide. Privileged and confidential. If this message has been received in error, please notify sender and delete it immediately. Conte?do confidencial. Se esta mensagem foi recebida por engano, favor avisar o remetente e apag?-la imediatamente. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/8f33692b/attachment-0001.html From kris at kriskinc.com Tue Jul 3 22:48:57 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 3 Jul 2012 14:48:57 -0400 Subject: [Freeswitch-users] Multiple 183 SDP Change Codecs - allowed? In-Reply-To: References: Message-ID: If we only we had absolutes that were that simple... The default setting in FreeSWITCH when forking (parallel or serial) is to consider the first bridge (leg) that returns media (of any sort) as successful. This FreeSWITCH behavior can be changed. Many other implementations behave differently. In SIP (especially with proxies) this isn't an absolute and several other scenarios are still valid. A leg can return 183 w/SDP and then a 4xx/5xx/6xx, which should then defer to another leg (or pass the error, or branch further, etc). Various scenarios are perfectly valid per the specs. It's up to the implementor/application to decide which behavior is preferred. On Tue, Jul 3, 2012 at 1:50 PM, Steven Ayre wrote: > 18x messages are supposed to make the call unreroutable since they indicate > the call has been successful on that route. > > -Steve > > > > On 3 July 2012 16:56, Avi Marcus wrote: >> >> They said the 1st carrier rejected it so the second 183 is from a second >> carrier.. but yes, the first carrier shouldn't be sending a 183... >> Anyway -- if you please look at the pcap from the first email, the log I >> posted a link to later, or the past two emails -- you'll see FS originally >> offers both PCMU and PCMA in the initial invite. >> >> I think we need Anthony to chime in here... he knows the RFC stuff! >> >> -Avi >> >> >> >> On Tue, Jul 3, 2012 at 6:22 PM, Kristian Kielhofner >> wrote: >>> >>> That's pretty nasty of voxbeam to do... Considering they return the >>> first 183 less than one second after the initial INVITE I'd say it's >>> very likely they're providing false ringback and/or FAS (false answer >>> supervision). Meanwhile the actual ringback comes over two seconds >>> later with PCMA. That's shady and if I were you I'd switch carriers >>> just on principle. >>> >>> However, as I said before they're not doing anything that's not >>> allowed by the various specifications although FAS is considered by >>> many as fraud. >>> >>> What are your codec settings like? Perhaps FS is rejecting PCMA >>> because you're not allowing it? >>> >>> On Tue, Jul 3, 2012 at 9:38 AM, Avi Marcus wrote: >>> > If so.. which I think is not.. is this a bug then in FS? >>> > >>> > http://pastebin.freeswitch.org/19423 >>> > You see from the first SDP that PCMU and PCMA were both options. >>> > Then for the next SDP FreeSWITCH was only considering PCMU which was >>> > the one >>> > that got chosen by the first SDP. >>> > >>> > Voxbeam claims it's not their fault.. is it? >>> > >>> > -Avi >>> > >>> > >>> > >>> > On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner >>> > wrote: >>> >> >>> >> Avi, >>> >> >>> >> Multiple 18x responses that change the SDP are allowed. I can't >>> >> find the specific document text now but as a random guy on the >>> >> internet (for whatever that's worth) I'm certain it is allowed. >>> >> >>> >> A re-INVITE can't be sent from either side until the dialog has been >>> >> established (200+ACK). In a case where the UAC (caller) would like to >>> >> update the session before it is established method UPDATE must be >>> >> used. >>> >> >>> >> On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus wrote: >>> >> > Are multiple 183s from the endpoint that changes the SDP allowed? >>> >> > I'm >>> >> > under >>> >> > the impression this is broken, similar to >>> >> > http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp >>> >> > ... which is why when the codec changes, FS freaks out and cancels >>> >> > the >>> >> > call >>> >> > because of codec negotiation error. >>> >> > Does that NDLB flag allow this, too? >>> >> > >>> >> > First has: audio 10116 RTP/AVP 0 101 13 >>> >> > Second has: audio 49020 RTP/AVP 8 13 101 >>> >> > PCAP: http://ge.tt/7MpyBwJ >>> >> > >>> >> > Can someone point me to the specific RFC so I can tell the supplier >>> >> > to >>> >> > fix >>> >> > it? >>> >> > And just curious.. what would make this allowed? A re-INVITE..? >>> >> > >>> >> > -Avi >>> >> > >>> >> > >>> >> > >>> >> > _________________________________________________________________________ >>> >> > Professional FreeSWITCH Consulting Services: >>> >> > consulting at freeswitch.org >>> >> > http://www.freeswitchsolutions.com >>> >> > >>> >> > >>> >> > >>> >> > >>> >> > Official FreeSWITCH Sites >>> >> > http://www.freeswitch.org >>> >> > http://wiki.freeswitch.org >>> >> > http://www.cluecon.com >>> >> > >>> >> > Join Us At ClueCon - Aug 7-9, 2012 >>> >> > >>> >> > FreeSWITCH-users mailing list >>> >> > FreeSWITCH-users at 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 >>> >> >>> >> >>> >> _________________________________________________________________________ >>> >> Professional FreeSWITCH Consulting Services: >>> >> consulting at freeswitch.org >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> >>> >> >>> >> >>> >> Official FreeSWITCH Sites >>> >> http://www.freeswitch.org >>> >> http://wiki.freeswitch.org >>> >> http://www.cluecon.com >>> >> >>> >> Join Us At ClueCon - Aug 7-9, 2012 >>> >> >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > >>> > >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > Join Us At ClueCon - Aug 7-9, 2012 >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at 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 >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 From grcamauer at gmail.com Tue Jul 3 23:20:13 2012 From: grcamauer at gmail.com (Guillermo Ruiz Camauer) Date: Tue, 3 Jul 2012 16:20:13 -0300 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: References: <1341335638.5953.23.camel@mythtv> Message-ID: <-7284538021435011907@unknownmsgid> I second the Snom phone recommendation. I have S pm 320, 360 and 370 models and they all work quite well. Guillermo Sent from my iPhone On 03/07/2012, at 15:37, Roger Castaldo wrote: I have had good success with snom phones and they are solid for the price On Jul 3, 2012 12:36 PM, "Todd Bailey" wrote: > Hi All, > > New to this tech and a bit confused on what to buy/features etc. > > I'm looking for a basic desk phone, perhaps with a few speed dial and > maybe gee-wiz buttons. I found this one with a decent price 3Com NBX > 1102 but the ad states Requires R4.X system software or greater. Not > sure what this is nor do I need to? > > If I buy "any" digital phone that has a rj-45 ethernet jack does that > automatically mean it's compatible? > > > other models in my price/feature range > > Cisco CP-7940G Unified IP Phone > > I was looking at a few grand stream models but the reviews lead me to > move on > > > thanks > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120703/334ba351/attachment.html From avi at avimarcus.net Tue Jul 3 23:21:55 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 3 Jul 2012 22:21:55 +0300 Subject: [Freeswitch-users] Multiple 183 SDP Change Codecs - allowed? In-Reply-To: References: Message-ID: So with NDLB-allow-nondup-sdp does that mean when the first SDP chooses PCMA then the second SDP with PCMU comes in - FS will still have that option available? (http://pastebin.freeswitch.org/19423 shows that currently it got lost after the 1st negotiation) This was a parallel dial - the remote opensips apparently was trying various gateways, though.. -Avi On Tue, Jul 3, 2012 at 9:48 PM, Kristian Kielhofner wrote: > If we only we had absolutes that were that simple... > > The default setting in FreeSWITCH when forking (parallel or serial) is > to consider the first bridge (leg) that returns media (of any sort) as > successful. This FreeSWITCH behavior can be changed. Many other > implementations behave differently. > > In SIP (especially with proxies) this isn't an absolute and several > other scenarios are still valid. A leg can return 183 w/SDP and then > a 4xx/5xx/6xx, which should then defer to another leg (or pass the > error, or branch further, etc). Various scenarios are perfectly valid > per the specs. It's up to the implementor/application to decide which > behavior is preferred. > > On Tue, Jul 3, 2012 at 1:50 PM, Steven Ayre wrote: > > 18x messages are supposed to make the call unreroutable since they > indicate > > the call has been successful on that route. > > > > -Steve > > > > > > > > On 3 July 2012 16:56, Avi Marcus wrote: > >> > >> They said the 1st carrier rejected it so the second 183 is from a second > >> carrier.. but yes, the first carrier shouldn't be sending a 183... > >> Anyway -- if you please look at the pcap from the first email, the log I > >> posted a link to later, or the past two emails -- you'll see FS > originally > >> offers both PCMU and PCMA in the initial invite. > >> > >> I think we need Anthony to chime in here... he knows the RFC stuff! > >> > >> -Avi > >> > >> > >> > >> On Tue, Jul 3, 2012 at 6:22 PM, Kristian Kielhofner > >> wrote: > >>> > >>> That's pretty nasty of voxbeam to do... Considering they return the > >>> first 183 less than one second after the initial INVITE I'd say it's > >>> very likely they're providing false ringback and/or FAS (false answer > >>> supervision). Meanwhile the actual ringback comes over two seconds > >>> later with PCMA. That's shady and if I were you I'd switch carriers > >>> just on principle. > >>> > >>> However, as I said before they're not doing anything that's not > >>> allowed by the various specifications although FAS is considered by > >>> many as fraud. > >>> > >>> What are your codec settings like? Perhaps FS is rejecting PCMA > >>> because you're not allowing it? > >>> > >>> On Tue, Jul 3, 2012 at 9:38 AM, Avi Marcus wrote: > >>> > If so.. which I think is not.. is this a bug then in FS? > >>> > > >>> > http://pastebin.freeswitch.org/19423 > >>> > You see from the first SDP that PCMU and PCMA were both options. > >>> > Then for the next SDP FreeSWITCH was only considering PCMU which was > >>> > the one > >>> > that got chosen by the first SDP. > >>> > > >>> > Voxbeam claims it's not their fault.. is it? > >>> > > >>> > -Avi > >>> > > >>> > > >>> > > >>> > On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner < > kris at kriskinc.com> > >>> > wrote: > >>> >> > >>> >> Avi, > >>> >> > >>> >> Multiple 18x responses that change the SDP are allowed. I can't > >>> >> find the specific document text now but as a random guy on the > >>> >> internet (for whatever that's worth) I'm certain it is allowed. > >>> >> > >>> >> A re-INVITE can't be sent from either side until the dialog has > been > >>> >> established (200+ACK). In a case where the UAC (caller) would like > to > >>> >> update the session before it is established method UPDATE must be > >>> >> used. > >>> >> > >>> >> On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus > wrote: > >>> >> > Are multiple 183s from the endpoint that changes the SDP allowed? > >>> >> > I'm > >>> >> > under > >>> >> > the impression this is broken, similar to > >>> >> > http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp > >>> >> > ... which is why when the codec changes, FS freaks out and cancels > >>> >> > the > >>> >> > call > >>> >> > because of codec negotiation error. > >>> >> > Does that NDLB flag allow this, too? > >>> >> > > >>> >> > First has: audio 10116 RTP/AVP 0 101 13 > >>> >> > Second has: audio 49020 RTP/AVP 8 13 101 > >>> >> > PCAP: http://ge.tt/7MpyBwJ > >>> >> > > >>> >> > Can someone point me to the specific RFC so I can tell the > supplier > >>> >> > to > >>> >> > fix > >>> >> > it? > >>> >> > And just curious.. what would make this allowed? A re-INVITE..? > >>> >> > > >>> >> > -Avi > >>> >> > > >>> >> > > >>> >> > > >>> >> > > _________________________________________________________________________ > >>> >> > Professional FreeSWITCH Consulting Services: > >>> >> > consulting at freeswitch.org > >>> >> > http://www.freeswitchsolutions.com > >>> >> > > >>> >> > > >>> >> > > >>> >> > > >>> >> > Official FreeSWITCH Sites > >>> >> > http://www.freeswitch.org > >>> >> > http://wiki.freeswitch.org > >>> >> > http://www.cluecon.com > >>> >> > > >>> >> > Join Us At ClueCon - Aug 7-9, 2012 > >>> >> > > >>> >> > FreeSWITCH-users mailing list > >>> >> > FreeSWITCH-users at 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 > >>> >> > >>> >> > >>> >> > _________________________________________________________________________ > >>> >> Professional FreeSWITCH Consulting Services: > >>> >> consulting at freeswitch.org > >>> >> http://www.freeswitchsolutions.com > >>> >> > >>> >> > >>> >> > >>> >> > >>> >> Official FreeSWITCH Sites > >>> >> http://www.freeswitch.org > >>> >> http://wiki.freeswitch.org > >>> >> http://www.cluecon.com > >>> >> > >>> >> Join Us At ClueCon - Aug 7-9, 2012 > >>> >> > >>> >> FreeSWITCH-users mailing list > >>> >> FreeSWITCH-users at lists.freeswitch.org > >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> >> > >>> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> >> http://www.freeswitch.org > >>> > > >>> > > >>> > > >>> > > >>> > > _________________________________________________________________________ > >>> > Professional FreeSWITCH Consulting Services: > >>> > consulting at freeswitch.org > >>> > http://www.freeswitchsolutions.com > >>> > > >>> > > >>> > > >>> > > >>> > Official FreeSWITCH Sites > >>> > http://www.freeswitch.org > >>> > http://wiki.freeswitch.org > >>> > http://www.cluecon.com > >>> > > >>> > Join Us At ClueCon - Aug 7-9, 2012 > >>> > > >>> > FreeSWITCH-users mailing list > >>> > FreeSWITCH-users at 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 > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> Join Us At ClueCon - Aug 7-9, 2012 > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120703/607aa2e9/attachment-0001.html From steveayre at gmail.com Tue Jul 3 23:36:13 2012 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 3 Jul 2012 20:36:13 +0100 Subject: [Freeswitch-users] FreeSWITCH, NTP daemon and clock drift Message-ID: Anthony, There's previously been a discussion captured on http://wiki.freeswitch.org/wiki/Clock regarding the internal clock time FS keeps. Currently FreeSWITCH uses the monotonic clock and ignores system time. On machines where the time is very unreliable, NTP isn't in use and the sysadmin is changing the time manually or by ntpdate the system time can make large jumps - and I can indeed see the reasoning that having FS use its own clock on these systems is more reliable as it prevents billing time being lost because of a large clock jump. However, on systems correctly running ntpd that have clock drift the system time will constantly be being corrected and won't experience large jumps and in these cases the system time will be more accurate than FreeSWITCH's internal time. On such systems although FreeSWITCH might not be losing billing seconds from clock jumps (which shouldn't normally happen) the clock drift will mean that there is also that clock drift present in FreeSWITCH's CDRs - and these could conceivably cause billing disputes where a customer's CDRs do not have the same times because their times are more accurate than FS's due to that clock drift. This is unlikely to be large enough to affect billed minutes much, but could put a large number of calls on the wrong billing rate (eg if FS's internal time in the CDRs identifies calls as on a peak rate when the customer believes it is on an offpeak rate). On a badly drifting system over a month on a large volume of calls that could become a noticeable discrepancy. Obviously there is the sync_clock option, but it seems you shouldn't need to run that frequently (and would that cause any sideeffects?) Running it infrequently would cause the same time-jumping behaviour FS's internal time was designed to avoid, and sync_clock_when_idle isn't possible on busier systems. On systems running ntpd perhaps it would be worth adding an opt-in option that lets FreeSWITCH use the system time and rely on ntpd (but keeping the present behaviour as the default)? Your thoughts please... Warm regards, -Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/13ca1f84/attachment.html From steveayre at gmail.com Tue Jul 3 23:39:06 2012 From: steveayre at gmail.com (Steven Ayre) Date: Tue, 3 Jul 2012 20:39:06 +0100 Subject: [Freeswitch-users] Multiple 183 SDP Change Codecs - allowed? In-Reply-To: References: Message-ID: True, that is simplistic. I should have said where the 18x isn't ignored... And yes it's one of those situations where everyone's idea of what's right may differ from everyone else's. -Steve On 3 July 2012 19:48, Kristian Kielhofner wrote: > If we only we had absolutes that were that simple... > > The default setting in FreeSWITCH when forking (parallel or serial) is > to consider the first bridge (leg) that returns media (of any sort) as > successful. This FreeSWITCH behavior can be changed. Many other > implementations behave differently. > > In SIP (especially with proxies) this isn't an absolute and several > other scenarios are still valid. A leg can return 183 w/SDP and then > a 4xx/5xx/6xx, which should then defer to another leg (or pass the > error, or branch further, etc). Various scenarios are perfectly valid > per the specs. It's up to the implementor/application to decide which > behavior is preferred. > > On Tue, Jul 3, 2012 at 1:50 PM, Steven Ayre wrote: > > 18x messages are supposed to make the call unreroutable since they > indicate > > the call has been successful on that route. > > > > -Steve > > > > > > > > On 3 July 2012 16:56, Avi Marcus wrote: > >> > >> They said the 1st carrier rejected it so the second 183 is from a second > >> carrier.. but yes, the first carrier shouldn't be sending a 183... > >> Anyway -- if you please look at the pcap from the first email, the log I > >> posted a link to later, or the past two emails -- you'll see FS > originally > >> offers both PCMU and PCMA in the initial invite. > >> > >> I think we need Anthony to chime in here... he knows the RFC stuff! > >> > >> -Avi > >> > >> > >> > >> On Tue, Jul 3, 2012 at 6:22 PM, Kristian Kielhofner > >> wrote: > >>> > >>> That's pretty nasty of voxbeam to do... Considering they return the > >>> first 183 less than one second after the initial INVITE I'd say it's > >>> very likely they're providing false ringback and/or FAS (false answer > >>> supervision). Meanwhile the actual ringback comes over two seconds > >>> later with PCMA. That's shady and if I were you I'd switch carriers > >>> just on principle. > >>> > >>> However, as I said before they're not doing anything that's not > >>> allowed by the various specifications although FAS is considered by > >>> many as fraud. > >>> > >>> What are your codec settings like? Perhaps FS is rejecting PCMA > >>> because you're not allowing it? > >>> > >>> On Tue, Jul 3, 2012 at 9:38 AM, Avi Marcus wrote: > >>> > If so.. which I think is not.. is this a bug then in FS? > >>> > > >>> > http://pastebin.freeswitch.org/19423 > >>> > You see from the first SDP that PCMU and PCMA were both options. > >>> > Then for the next SDP FreeSWITCH was only considering PCMU which was > >>> > the one > >>> > that got chosen by the first SDP. > >>> > > >>> > Voxbeam claims it's not their fault.. is it? > >>> > > >>> > -Avi > >>> > > >>> > > >>> > > >>> > On Tue, Jul 3, 2012 at 4:26 PM, Kristian Kielhofner < > kris at kriskinc.com> > >>> > wrote: > >>> >> > >>> >> Avi, > >>> >> > >>> >> Multiple 18x responses that change the SDP are allowed. I can't > >>> >> find the specific document text now but as a random guy on the > >>> >> internet (for whatever that's worth) I'm certain it is allowed. > >>> >> > >>> >> A re-INVITE can't be sent from either side until the dialog has > been > >>> >> established (200+ACK). In a case where the UAC (caller) would like > to > >>> >> update the session before it is established method UPDATE must be > >>> >> used. > >>> >> > >>> >> On Tue, Jul 3, 2012 at 1:04 AM, Avi Marcus > wrote: > >>> >> > Are multiple 183s from the endpoint that changes the SDP allowed? > >>> >> > I'm > >>> >> > under > >>> >> > the impression this is broken, similar to > >>> >> > http://wiki.freeswitch.org/wiki/NDLB#NDLB-allow-nondup-sdp > >>> >> > ... which is why when the codec changes, FS freaks out and cancels > >>> >> > the > >>> >> > call > >>> >> > because of codec negotiation error. > >>> >> > Does that NDLB flag allow this, too? > >>> >> > > >>> >> > First has: audio 10116 RTP/AVP 0 101 13 > >>> >> > Second has: audio 49020 RTP/AVP 8 13 101 > >>> >> > PCAP: http://ge.tt/7MpyBwJ > >>> >> > > >>> >> > Can someone point me to the specific RFC so I can tell the > supplier > >>> >> > to > >>> >> > fix > >>> >> > it? > >>> >> > And just curious.. what would make this allowed? A re-INVITE..? > >>> >> > > >>> >> > -Avi > >>> >> > > >>> >> > > >>> >> > > >>> >> > > _________________________________________________________________________ > >>> >> > Professional FreeSWITCH Consulting Services: > >>> >> > consulting at freeswitch.org > >>> >> > http://www.freeswitchsolutions.com > >>> >> > > >>> >> > > >>> >> > > >>> >> > > >>> >> > Official FreeSWITCH Sites > >>> >> > http://www.freeswitch.org > >>> >> > http://wiki.freeswitch.org > >>> >> > http://www.cluecon.com > >>> >> > > >>> >> > Join Us At ClueCon - Aug 7-9, 2012 > >>> >> > > >>> >> > FreeSWITCH-users mailing list > >>> >> > FreeSWITCH-users at 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 > >>> >> > >>> >> > >>> >> > _________________________________________________________________________ > >>> >> Professional FreeSWITCH Consulting Services: > >>> >> consulting at freeswitch.org > >>> >> http://www.freeswitchsolutions.com > >>> >> > >>> >> > >>> >> > >>> >> > >>> >> Official FreeSWITCH Sites > >>> >> http://www.freeswitch.org > >>> >> http://wiki.freeswitch.org > >>> >> http://www.cluecon.com > >>> >> > >>> >> Join Us At ClueCon - Aug 7-9, 2012 > >>> >> > >>> >> FreeSWITCH-users mailing list > >>> >> FreeSWITCH-users at lists.freeswitch.org > >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> >> > >>> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> >> http://www.freeswitch.org > >>> > > >>> > > >>> > > >>> > > >>> > > _________________________________________________________________________ > >>> > Professional FreeSWITCH Consulting Services: > >>> > consulting at freeswitch.org > >>> > http://www.freeswitchsolutions.com > >>> > > >>> > > >>> > > >>> > > >>> > Official FreeSWITCH Sites > >>> > http://www.freeswitch.org > >>> > http://wiki.freeswitch.org > >>> > http://www.cluecon.com > >>> > > >>> > Join Us At ClueCon - Aug 7-9, 2012 > >>> > > >>> > FreeSWITCH-users mailing list > >>> > FreeSWITCH-users at 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 > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> Join Us At ClueCon - Aug 7-9, 2012 > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120703/48eaebce/attachment-0001.html From toddb at toddbailey.net Tue Jul 3 23:48:14 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Tue, 03 Jul 2012 12:48:14 -0700 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: References: <1341335638.5953.23.camel@mythtv> Message-ID: <1341344894.5953.27.camel@mythtv> Thanks for the info but well beyond my price range. The 3com is cheap and probably due to being tied to the specific pbx package. The Cisco has good reviews but the display isn't back lit, making use in a dark room problematic. The used GrandStream models are affordable but from what I read suffer from poor audio. I rapidly running out of options here, any one else care to jump in? On Tue, 2012-07-03 at 14:22 -0400, Roger Castaldo wrote: > I have had good success with snom phones and they are solid for the > price > > On Jul 3, 2012 12:36 PM, "Todd Bailey" wrote: > Hi All, > > New to this tech and a bit confused on what to buy/features > etc. > > I'm looking for a basic desk phone, perhaps with a few speed > dial and > maybe gee-wiz buttons. I found this one with a decent price > 3Com NBX > 1102 but the ad states Requires R4.X system software or > greater. Not > sure what this is nor do I need to? > > If I buy "any" digital phone that has a rj-45 ethernet jack > does that > automatically mean it's compatible? > > > other models in my price/feature range > > Cisco CP-7940G Unified IP Phone > > I was looking at a few grand stream models but the reviews > lead me to > move on > > > thanks > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ifoundthetao at gmail.com Tue Jul 3 23:53:43 2012 From: ifoundthetao at gmail.com (Timothy Bolton) Date: Tue, 03 Jul 2012 14:53:43 -0500 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: <1341335638.5953.23.camel@mythtv> References: <1341335638.5953.23.camel@mythtv> Message-ID: <4FF34DC7.5000907@gmail.com> I've had good luck with Nortel. 'We who cut mere stones must always be envisioning cathedrals.' Quarry Worker's Creed On 7/3/2012 12:13 PM, Todd Bailey wrote: > Hi All, > > New to this tech and a bit confused on what to buy/features etc. > > I'm looking for a basic desk phone, perhaps with a few speed dial and > maybe gee-wiz buttons. I found this one with a decent price 3Com NBX > 1102 but the ad states Requires R4.X system software or greater. Not > sure what this is nor do I need to? > > If I buy "any" digital phone that has a rj-45 ethernet jack does that > automatically mean it's compatible? > > > other models in my price/feature range > > Cisco CP-7940G Unified IP Phone > > I was looking at a few grand stream models but the reviews lead me to > move on > > > thanks > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jmesquita at freeswitch.org Wed Jul 4 00:12:14 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Tue, 3 Jul 2012 17:12:14 -0300 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: <1341344894.5953.27.camel@mythtv> References: <1341335638.5953.23.camel@mythtv> <1341344894.5953.27.camel@mythtv> Message-ID: <9C4CF5C595E7490ABC21D45FCE92A95B@freeswitch.org> Aastra an Polys are good phones as well. -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Tuesday, July 3, 2012 at 4:48 PM, Todd Bailey wrote: > Thanks for the info but well beyond my price range. > The 3com is cheap and probably due to being tied to the specific pbx > package. > > The Cisco has good reviews but the display isn't back lit, making use in > a dark room problematic. > > The used GrandStream models are affordable but from what I read suffer > from poor audio. > > I rapidly running out of options here, any one else care to jump in? > > > > On Tue, 2012-07-03 at 14:22 -0400, Roger Castaldo wrote: > > I have had good success with snom phones and they are solid for the > > price > > > > On Jul 3, 2012 12:36 PM, "Todd Bailey" wrote: > > Hi All, > > > > New to this tech and a bit confused on what to buy/features > > etc. > > > > I'm looking for a basic desk phone, perhaps with a few speed > > dial and > > maybe gee-wiz buttons. I found this one with a decent price > > 3Com NBX > > 1102 but the ad states Requires R4.X system software or > > greater. Not > > sure what this is nor do I need to? > > > > If I buy "any" digital phone that has a rj-45 ethernet jack > > does that > > automatically mean it's compatible? > > > > > > other models in my price/feature range > > > > Cisco CP-7940G Unified IP Phone > > > > I was looking at a few grand stream models but the reviews > > lead me to > > move on > > > > > > thanks > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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/20120703/9037a3d7/attachment.html From toddb at toddbailey.net Wed Jul 4 00:37:14 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Tue, 03 Jul 2012 13:37:14 -0700 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: <1341344894.5953.27.camel@mythtv> References: <1341335638.5953.23.camel@mythtv> <1341344894.5953.27.camel@mythtv> Message-ID: <1341347834.5953.29.camel@mythtv> My question is evolving into not who but what. It it safe to say any phone marketed as a 'IP Phone' will work or are there gotchas that I need to know about ? On Tue, 2012-07-03 at 12:48 -0700, Todd Bailey wrote: > Thanks for the info but well beyond my price range. > The 3com is cheap and probably due to being tied to the specific pbx > package. > > The Cisco has good reviews but the display isn't back lit, making use in > a dark room problematic. > > The used GrandStream models are affordable but from what I read suffer > from poor audio. > > I rapidly running out of options here, any one else care to jump in? > > > > On Tue, 2012-07-03 at 14:22 -0400, Roger Castaldo wrote: > > I have had good success with snom phones and they are solid for the > > price > > > > On Jul 3, 2012 12:36 PM, "Todd Bailey" wrote: > > Hi All, > > > > New to this tech and a bit confused on what to buy/features > > etc. > > > > I'm looking for a basic desk phone, perhaps with a few speed > > dial and > > maybe gee-wiz buttons. I found this one with a decent price > > 3Com NBX > > 1102 but the ad states Requires R4.X system software or > > greater. Not > > sure what this is nor do I need to? > > > > If I buy "any" digital phone that has a rj-45 ethernet jack > > does that > > automatically mean it's compatible? > > > > > > other models in my price/feature range > > > > Cisco CP-7940G Unified IP Phone > > > > I was looking at a few grand stream models but the reviews > > lead me to > > move on > > > > > > thanks > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Jul 4 00:53:04 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jul 2012 13:53:04 -0700 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: <1341347834.5953.29.camel@mythtv> References: <1341335638.5953.23.camel@mythtv> <1341344894.5953.27.camel@mythtv> <1341347834.5953.29.camel@mythtv> Message-ID: On Tue, Jul 3, 2012 at 1:37 PM, Todd Bailey wrote: > My question is evolving into not who but what. > > It it safe to say any phone marketed as a 'IP Phone' will work or are > there gotchas that I need to know about ? > There are ALWAYS gotchas with every manufacturer. The problem is that there is not a comprehensive list of every gotcha for every phone maker because that's just too much work to keep up with. (Things change, firmwares get updated, etc.) If you want a really inexpensive phone that "just works" then you might want to check out the Yealink T26. Alternatively, the Snom 300 is cheap, but it has a really small display (although it's backlit, so at least you have that.) Personally, I would stay away from the Cisco 79xx phones unless you really, really enjoy drama. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/940896f5/attachment-0001.html From sdevoy at bizfocused.com Wed Jul 4 01:01:37 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Tue, 3 Jul 2012 17:01:37 -0400 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: <1341347834.5953.29.camel@mythtv> References: <1341335638.5953.23.camel@mythtv> <1341344894.5953.27.camel@mythtv> <1341347834.5953.29.camel@mythtv> Message-ID: <047301cd595f$095960b0$1c0c2210$@bizfocused.com> Todd, I think the general rule you are looking for is "Any Internet phone that uses the SIP protocol should work." That rules out the 3COM NBX phones. I have used Cisco 504Gs rather extensively. Once we got the NAT settings down, they have been great. They do have backlighting, but are rather costly at about $175 - $200. I am watching replies to your question for an answer from someone who says something "I use ______ and they are cheap, have 4 lines and work great!" Hope that helps a little. Sean -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Todd Bailey Sent: Tuesday, July 03, 2012 4:37 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Fs compatible phones? My question is evolving into not who but what. It it safe to say any phone marketed as a 'IP Phone' will work or are there gotchas that I need to know about ? On Tue, 2012-07-03 at 12:48 -0700, Todd Bailey wrote: > Thanks for the info but well beyond my price range. > The 3com is cheap and probably due to being tied to the specific pbx > package. > > The Cisco has good reviews but the display isn't back lit, making use > in a dark room problematic. > > The used GrandStream models are affordable but from what I read suffer > from poor audio. > > I rapidly running out of options here, any one else care to jump in? > > > > On Tue, 2012-07-03 at 14:22 -0400, Roger Castaldo wrote: > > I have had good success with snom phones and they are solid for the > > price > > > > On Jul 3, 2012 12:36 PM, "Todd Bailey" wrote: > > Hi All, > > > > New to this tech and a bit confused on what to buy/features > > etc. > > > > I'm looking for a basic desk phone, perhaps with a few speed > > dial and > > maybe gee-wiz buttons. I found this one with a decent price > > 3Com NBX > > 1102 but the ad states Requires R4.X system software or > > greater. Not > > sure what this is nor do I need to? > > > > If I buy "any" digital phone that has a rj-45 ethernet jack > > does that > > automatically mean it's compatible? > > > > > > other models in my price/feature range > > > > Cisco CP-7940G Unified IP Phone > > > > I was looking at a few grand stream models but the reviews > > lead me to > > move on > > > > > > thanks > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > ____________________________________________________________________ > > _____ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u > > sers > > http://www.freeswitch.org > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users 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 > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jmesquita at freeswitch.org Wed Jul 4 01:04:17 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Tue, 3 Jul 2012 18:04:17 -0300 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: <1341347834.5953.29.camel@mythtv> References: <1341335638.5953.23.camel@mythtv> <1341344894.5953.27.camel@mythtv> <1341347834.5953.29.camel@mythtv> Message-ID: <5B728E5A44624EE0A788DD4668BC3569@freeswitch.org> Unfortunately this is not an easy question to reply. You have 4 options of non proprietary IP Phones nowadays (at least). MGCP (Skinny), H.323, IAX2 and SIP. SIP is the most widely used protocol nowadays and pretty much any SIP IP Phone available nowadays on the market will work with FreeSWITCH. If you are looking for more advanced stuff, then you're in trouble and you have to know your stuff to make a informed decision. SIP is a not just a simple standard but rather a miryad of RFCs, each one suggesting a new feature with and it's implementations depends on the manufacturer understanding of each. Yes, interop is a btich, but at least there is no such thing as a proprietary digital phone nowadays, uh? Hope that helps. -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Tuesday, July 3, 2012 at 5:37 PM, Todd Bailey wrote: > My question is evolving into not who but what. > > It it safe to say any phone marketed as a 'IP Phone' will work or are > there gotchas that I need to know about ? > > > > On Tue, 2012-07-03 at 12:48 -0700, Todd Bailey wrote: > > Thanks for the info but well beyond my price range. > > The 3com is cheap and probably due to being tied to the specific pbx > > package. > > > > The Cisco has good reviews but the display isn't back lit, making use in > > a dark room problematic. > > > > The used GrandStream models are affordable but from what I read suffer > > from poor audio. > > > > I rapidly running out of options here, any one else care to jump in? > > > > > > > > On Tue, 2012-07-03 at 14:22 -0400, Roger Castaldo wrote: > > > I have had good success with snom phones and they are solid for the > > > price > > > > > > On Jul 3, 2012 12:36 PM, "Todd Bailey" wrote: > > > Hi All, > > > > > > New to this tech and a bit confused on what to buy/features > > > etc. > > > > > > I'm looking for a basic desk phone, perhaps with a few speed > > > dial and > > > maybe gee-wiz buttons. I found this one with a decent price > > > 3Com NBX > > > 1102 but the ad states Requires R4.X system software or > > > greater. Not > > > sure what this is nor do I need to? > > > > > > If I buy "any" digital phone that has a rj-45 ethernet jack > > > does that > > > automatically mean it's compatible? > > > > > > > > > other models in my price/feature range > > > > > > Cisco CP-7940G Unified IP Phone > > > > > > I was looking at a few grand stream models but the reviews > > > lead me to > > > move on > > > > > > > > > thanks > > > > > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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/20120703/9663b821/attachment.html From marketing at cluecon.com Wed Jul 4 01:19:57 2012 From: marketing at cluecon.com (Michael Collins) Date: Tue, 3 Jul 2012 14:19:57 -0700 Subject: [Freeswitch-users] ClueCon 2012 - Last Chance for 16 Entries Message-ID: Hello all! I am happy to report that a number of people have taken advantage of this opportunity and have registered in time to get their 16 entries. Remember that you can register any time on July 3rd and still qualify for the extra entries. The bits will shift at midnight Central time so don't delay! Only 34 more days and we'll all be in Chicago. See you then! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/d493e14b/attachment-0001.html From msc at freeswitch.org Wed Jul 4 01:48:02 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jul 2012 14:48:02 -0700 Subject: [Freeswitch-users] question: lua how can i create a postcastlist? In-Reply-To: References: Message-ID: You'll have a challenge here because natively Lua has no directory commands. I have never tried reading in the results of a system call into a Lua table, but it might be possible. I'd ask on #Lua or check Lua docs to see if such a thing is even possible. Once you have the files in a table then you could do the next/prev/first/last/repeat thing. -MC On Tue, Jul 3, 2012 at 8:58 AM, Thomas Hoellriegel wrote: > Hi guys, > I have many potcast mp3 files in a directory: > /home/postcast > the files beginning with: > postcast_year_mon_day_hour_**minutes_.mp3 > I like to create a playlist in lua: > 1. read all files, and make a messagecount. > 2. generate a menu to use from phone: > > press 1 for the next postcast. > 2, To repeate the postcast > 3. to go to the prev potcast > 4 for the fist potcast > 5 for the last potcast > 6 to hear the date an time for these. > > I like to sort, firt in first out. > > I don.t find a command in lua: > to read the filelist, > to generate a filecount. > and to generate a navigation keys: > forward, backward, repeate and say the time and date of the current file. > > Can your help please? > > nice thanks to all. > > > > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120703/fe825d51/attachment.html From msc at freeswitch.org Wed Jul 4 02:01:02 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jul 2012 15:01:02 -0700 Subject: [Freeswitch-users] How about a USER FORUM and kill off the mail list? In-Reply-To: References: <1341323005.525923853@f90.mail.ru> Message-ID: On Tue, Jul 3, 2012 at 7:39 AM, Ken Rice wrote: > You cant use FreeSWITCHForum.com as a domain name without violating > FreeSWITCH Trademarks... Now you are putting the project maintainers in an > awkward position as you can not keep your trademarks protected unless you > defend them... > This is a slippery slope. If the target server is in a country whose laws differ from the country where the trademark was established then it's a huge mess. However, the safe thing to do is to call it the "Unofficial FreeSWITCH Forum" and have a disclaimer on the site itself and say that "FreeSWITCH is a registered trademark of Anthony Minessale. FreeSWITCHForum.com is not in any way endorsed by, or affiliated with, the FreeSWITCH project." As far as "defending" the trademark - that's also a slippery slope. Trademarks are a form of IP that are designed to protect consumers, not trademark holders. A trademark holder must only "defend" if there is some sort of activity that may cause "confusion" or "dilution" or something similar. Simply creating a website with "FreeSWITCH" in the domain name is not an act of trademark infringement, even in the US. However, if the title on the web page said "Official FreeSWITCH Forum" then that would absolutely be an act of trademark infringement because people would be "confused" - that is, people would think that the site was sponsored/endorsed by the trademark holder when in fact it is not. Also, deceptive domain names could be infringing, like www.officialfreeswitchforum.com. I recommend that the OP politely ask Anthony if he has an reservations about the OP running a site called "FreeSWITCH Forum". -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/7251fc72/attachment.html From msc at freeswitch.org Wed Jul 4 02:09:58 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jul 2012 15:09:58 -0700 Subject: [Freeswitch-users] how to solve the error Rejected by acl "domains". Falling back to Digest auth? In-Reply-To: <1341294458.85684.YahooMailNeo@web120104.mail.ne1.yahoo.com> References: <1341247643.41122.YahooMailNeo@web120105.mail.ne1.yahoo.com> <0BC3D6F5-9FC8-4E53-85AD-F2A83B6B26A5@visionutveckling.se> <1341287313.55682.YahooMailNeo@web120104.mail.ne1.yahoo.com> <1FFF97C269757C458224B7C895F35F1512B642@cantor.std.visionutv.se> <1341294458.85684.YahooMailNeo@web120104.mail.ne1.yahoo.com> Message-ID: On Mon, Jul 2, 2012 at 10:47 PM, Samira Mh wrote: > i have to limit the count of registerations because that feature is > exactly what my manager want !:( > so i must to implement it correctly ... > > Two questions: *WHY* does your manager want this feature? What is the ultimate problem that is being addressed? It may be that he or she made the decision with limited or incorrect information. If we know what the ultimate goal is with all this then we might be able to help you with a solution that is even better than what your manager had hoped to see implemented. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/3e10d97f/attachment.html From anthony.minessale at gmail.com Wed Jul 4 02:16:16 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 3 Jul 2012 17:16:16 -0500 Subject: [Freeswitch-users] FreeSWITCH, NTP daemon and clock drift In-Reply-To: References: Message-ID: My first instinct is to say: don't use boxes with unreliable timing in them since if they can't keep time, the precision for audio could be even worse. But, Its not an unreasonable request give this param a try in latest git: enable-use-system-time in switch.conf.xml the status uptime should still be right but the timestamps etc will follow system time. use at your own risk On Tue, Jul 3, 2012 at 2:36 PM, Steven Ayre wrote: > Anthony, > > There's previously been a discussion captured on > http://wiki.freeswitch.org/wiki/Clock regarding the internal clock time FS > keeps. > > Currently FreeSWITCH uses the monotonic clock and ignores system time. On > machines where the time is very unreliable, NTP isn't in use and the > sysadmin is changing the time manually or by ntpdate the system time can > make large jumps - and I can indeed see the reasoning that having FS use its > own clock on these systems is more reliable as it prevents billing time > being lost because of a large clock jump. > > However, on systems correctly running ntpd that have clock drift the system > time will constantly be being corrected and won't experience large jumps and > in these cases the system time will be more accurate than FreeSWITCH's > internal time. > > On such systems although FreeSWITCH might not be losing billing seconds from > clock jumps (which shouldn't normally happen) the clock drift will mean that > there is also that clock drift present in FreeSWITCH's CDRs - and these > could conceivably cause billing disputes where a customer's CDRs do not have > the same times because their times are more accurate than FS's due to that > clock drift. This is unlikely to be large enough to affect billed minutes > much, but could put a large number of calls on the wrong billing rate (eg if > FS's internal time in the CDRs identifies calls as on a peak rate when the > customer believes it is on an offpeak rate). On a badly drifting system over > a month on a large volume of calls that could become a noticeable > discrepancy. > > Obviously there is the sync_clock option, but it seems you shouldn't need to > run that frequently (and would that cause any sideeffects?) Running it > infrequently would cause the same time-jumping behaviour FS's internal time > was designed to avoid, and sync_clock_when_idle isn't possible on busier > systems. > > On systems running ntpd perhaps it would be worth adding an opt-in option > that lets FreeSWITCH use the system time and rely on ntpd (but keeping the > present behaviour as the default)? > > Your thoughts please... > > Warm regards, > -Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From msc at freeswitch.org Wed Jul 4 02:17:30 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 3 Jul 2012 15:17:30 -0700 Subject: [Freeswitch-users] Freeswitch console log and file log In-Reply-To: <8989605.233.1341310197992.JavaMail.a66a@a66a> References: <5935829.173.1341309872664.JavaMail.a66a@a66a> <8989605.233.1341310197992.JavaMail.a66a@a66a> Message-ID: On Tue, Jul 3, 2012 at 3:10 AM, Alexandr Kostenko wrote: > Hi, Guys. > Can you explain me more understandable about logs in freeswitch. > Can I for example decrease console log level (or fully disable) and > increase file log level? > Also can full log level causing freeswitch crush ? > Thanks > > You can have console logging at a different level than the log file. Look in conf/autoload_configs/console.conf.xml and logfile.conf.xml and you'll see that you have a fair number of options. In console.conf.xml there's a lot of explanatory information about mappings and how to make it so that only certain items have logging turned on at specific log levels. Note that what's listed in console.conf.xml also applies to logfile.conf.xml. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/e8ea1034/attachment-0001.html From dujinfang at gmail.com Wed Jul 4 05:43:19 2012 From: dujinfang at gmail.com (Seven Du) Date: Wed, 4 Jul 2012 09:43:19 +0800 Subject: [Freeswitch-users] conference param video-floor-only Message-ID: Hi, What is video-floor-only trying to do in conference? I assume it tries to freeze the floor holder ( and his video) as long as he has video, but in my testing it's not the case, I set the param in the default profile and it seems doesn't make any difference. I tried the following change it looks more clear to me and seems works as what I expected. - if ((!floor_holder || (imember->score_iir > SCORE_IIR_SPEAKING_MAX && (floor_holder->score_iir < SCORE_IIR_SPEAKING_MIN))) && - (!switch_test_flag(conference, CFLAG_VID_FLOOR) || switch_channel_test_flag(channel, CF_VIDEO))) { - floor_holder = imember; + if (switch_channel_test_flag(channel, CF_VIDEO)) { + if (!floor_holder) { + floor_holder = imember; + } else if ((imember->score_iir > SCORE_IIR_SPEAKING_MAX) && (floor_holder->score_iir < SCORE_IIR_SPEAKING_MIN) && + (!switch_test_flag(conference, CFLAG_VID_FLOOR))) { + floor_holder = imember; + } } Another thing I found is that in the normal case(without floor-only set), the video changes according to the volume of the speaker fine, but if the floor press 0 to be muted, there's no chance of other members can get the floor, is there a problem? Thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120704/d7e98705/attachment.html From toddb at toddbailey.net Wed Jul 4 08:06:16 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Tue, 03 Jul 2012 21:06:16 -0700 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: <1341335638.5953.23.camel@mythtv> References: <1341335638.5953.23.camel@mythtv> Message-ID: <1341374776.5343.6.camel@mythtv> Thanks for all the good advice, but sadly out of my current price range. I'm finding several phones sub $50 range listed on ebay, A few so far and this list will grow, Toshiba IP5122-SD Cisco CP-6921 Cisco 7940 Cisco SPA-942 Fanvil FV6020 Avaya 4412D+ Soyo 7 G668 Polycom Soundpoint IP501 Any Opinions? On Tue, 2012-07-03 at 10:13 -0700, Todd Bailey wrote: > Hi All, > > New to this tech and a bit confused on what to buy/features etc. > > I'm looking for a basic desk phone, perhaps with a few speed dial and > maybe gee-wiz buttons. I found this one with a decent price 3Com NBX > 1102 but the ad states Requires R4.X system software or greater. Not > sure what this is nor do I need to? > > If I buy "any" digital phone that has a rj-45 ethernet jack does that > automatically mean it's compatible? > > > other models in my price/feature range > > Cisco CP-7940G Unified IP Phone > > I was looking at a few grand stream models but the reviews lead me to > move on > > > thanks > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From yehavi.bourvine at gmail.com Wed Jul 4 08:25:24 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Wed, 4 Jul 2012 07:25:24 +0300 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: <1341374776.5343.6.camel@mythtv> References: <1341335638.5953.23.camel@mythtv> <1341374776.5343.6.camel@mythtv> Message-ID: Hi, Polycoms are the best (from my experience) but the most expensive ones. Polycom-501 is an old model, needs special cable with power supply (no PoE, unless you buy a special cable); some of my 501 are already broken, so I won't buy it these days. You can also look for AudioCodes-320 or AudioCodes-310. I have a few dosens of the first one, and the users are quite happy. Note that they do not support SLA yet, and BLF will be supported in the next software release (I have a beta version of it). Regards, __Yehavi: 2012/7/4 Todd Bailey > Thanks for all the good advice, but sadly out of my current price range. > > I'm finding several phones sub $50 range listed on ebay, A few so far > and this list will grow, > > Toshiba IP5122-SD > Cisco CP-6921 > Cisco 7940 > Cisco SPA-942 > Fanvil FV6020 > Avaya 4412D+ > Soyo 7 G668 > Polycom Soundpoint IP501 > > Any Opinions? > > On Tue, 2012-07-03 at 10:13 -0700, Todd Bailey wrote: > > Hi All, > > > > New to this tech and a bit confused on what to buy/features etc. > > > > I'm looking for a basic desk phone, perhaps with a few speed dial and > > maybe gee-wiz buttons. I found this one with a decent price 3Com NBX > > 1102 but the ad states Requires R4.X system software or greater. Not > > sure what this is nor do I need to? > > > > If I buy "any" digital phone that has a rj-45 ethernet jack does that > > automatically mean it's compatible? > > > > > > other models in my price/feature range > > > > Cisco CP-7940G Unified IP Phone > > > > I was looking at a few grand stream models but the reviews lead me to > > move on > > > > > > thanks > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120704/30ddefc2/attachment.html From william.king at quentustech.com Wed Jul 4 08:32:48 2012 From: william.king at quentustech.com (William King) Date: Tue, 03 Jul 2012 21:32:48 -0700 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: <1341374776.5343.6.camel@mythtv> References: <1341335638.5953.23.camel@mythtv> <1341374776.5343.6.camel@mythtv> Message-ID: <4FF3C770.2020006@quentustech.com> I just finished doing interop with Yealink and if you are looking for a very low cost phone then the T22 new can be had for ~$70-90. http://www.amazon.com/Yealink-SIP-T22P-Professional-Phone-Lines/dp/B002D10BYO/ William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 07/03/2012 09:06 PM, Todd Bailey wrote: > Thanks for all the good advice, but sadly out of my current price range. > > I'm finding several phones sub $50 range listed on ebay, A few so far > and this list will grow, > > Toshiba IP5122-SD > Cisco CP-6921 > Cisco 7940 > Cisco SPA-942 > Fanvil FV6020 > Avaya 4412D+ > Soyo 7 G668 > Polycom Soundpoint IP501 > > Any Opinions? > > On Tue, 2012-07-03 at 10:13 -0700, Todd Bailey wrote: >> Hi All, >> >> New to this tech and a bit confused on what to buy/features etc. >> >> I'm looking for a basic desk phone, perhaps with a few speed dial and >> maybe gee-wiz buttons. I found this one with a decent price 3Com NBX >> 1102 but the ad states Requires R4.X system software or greater. Not >> sure what this is nor do I need to? >> >> If I buy "any" digital phone that has a rj-45 ethernet jack does that >> automatically mean it's compatible? >> >> >> other models in my price/feature range >> >> Cisco CP-7940G Unified IP Phone >> >> I was looking at a few grand stream models but the reviews lead me to >> move on >> >> >> thanks >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From toddb at toddbailey.net Wed Jul 4 10:01:13 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Tue, 03 Jul 2012 23:01:13 -0700 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: <1341335638.5953.23.camel@mythtv> References: <1341335638.5953.23.camel@mythtv> Message-ID: <1341381673.5343.9.camel@mythtv> I'm going to purchase a Cisco 7960 from ebay for under $50 unless someone can tell me this is a bad idea. I watched several Utube videos where is was working on Asterisk. With the volumns of documentation and how to videos I think this is the better choice, On Tue, 2012-07-03 at 10:13 -0700, Todd Bailey wrote: > Hi All, > > New to this tech and a bit confused on what to buy/features etc. > > I'm looking for a basic desk phone, perhaps with a few speed dial and > maybe gee-wiz buttons. I found this one with a decent price 3Com NBX > 1102 but the ad states Requires R4.X system software or greater. Not > sure what this is nor do I need to? > > If I buy "any" digital phone that has a rj-45 ethernet jack does that > automatically mean it's compatible? > > > other models in my price/feature range > > Cisco CP-7940G Unified IP Phone > > I was looking at a few grand stream models but the reviews lead me to > move on > > > thanks > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From saami_mh at ymail.com Wed Jul 4 10:27:14 2012 From: saami_mh at ymail.com (Samira Mh) Date: Tue, 3 Jul 2012 23:27:14 -0700 (PDT) Subject: [Freeswitch-users] what is different between user who is registered or not registered using event? Message-ID: <1341383234.47450.YahooMailNeo@web120104.mail.ne1.yahoo.com> as link below: http://wiki.freeswitch.org/wiki/Mod_lua/Serving_Configuration? when registering (REGISTER) (register to sip phone) 2012-07-04 10:52:24.480734 [NOTICE] switch_cpp.cpp:1227 Debug from gen_dir_user_xml.lua, provided params: 'Event-Name: REQUEST_PARAMS Core-UUID: dc37b240-c596-11e1-b315-0daf7608d9db FreeSWITCH-Hostname: PBX FreeSWITCH-Switchname: PBX FreeSWITCH-IPv4: 192.168.10.89 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-07-04%2010%3A52%3A24 Event-Date-GMT: Wed,%2004%20Jul%202012%2006%3A22%3A24%20GMT Event-Date-Timestamp: 1341382944480734 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_parse_auth Event-Calling-Line-Number: 2364 Event-Sequence: 1233 action: sip_auth sip_profile: internal sip_user_agent: eyeBeam%20release%201102q%20stamp%2051814 sip_auth_username: 1000 sip_auth_realm: 192.168.10.89 sip_auth_nonce: 9e86a118-c5a0-11e1-b324-0daf7608d9db sip_auth_uri: sip%3A192.168.10.89 sip_contact_user: 1000 sip_contact_host: 192.168.18.120 sip_to_user: 1000 sip_to_host: 192.168.10.89 sip_from_user: 1000 sip_from_host: 192.168.10.89 sip_request_host: 192.168.10.89 sip_auth_qop: auth sip_auth_cnonce: ccbce7a4be40e6660e1a39350b5610b9 sip_auth_nc: 00000001 sip_auth_response: b7c175e2b6680b43300a36aa3c50356a sip_auth_method: REGISTER key: id user: 1000 domain: 192.168.10.89 ip: 192.168.18.120 and when user want to logout(unregistered from sip phone) 2012-07-04 10:52:53.440720 [NOTICE] switch_cpp.cpp:1227 Debug from gen_dir_user_xml.lua, provided params: 'Event-Name: REQUEST_PARAMS Core-UUID: dc37b240-c596-11e1-b315-0daf7608d9db FreeSWITCH-Hostname: PBX FreeSWITCH-Switchname: PBX FreeSWITCH-IPv4: 192.168.10.89 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-07-04%2010%3A52%3A53 Event-Date-GMT: Wed,%2004%20Jul%202012%2006%3A22%3A53%20GMT Event-Date-Timestamp: 1341382973440720 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_parse_auth Event-Calling-Line-Number: 2364 Event-Sequence: 1254 action: sip_auth sip_profile: internal sip_user_agent: eyeBeam%20release%201102q%20stamp%2051814 sip_auth_username: 1000 sip_auth_realm: 192.168.10.89 sip_auth_nonce: 9e86a118-c5a0-11e1-b324-0daf7608d9db sip_auth_uri: sip%3A192.168.10.89 sip_contact_user: 1000 sip_contact_host: 192.168.18.120 sip_to_user: 1000 sip_to_host: 192.168.10.89 sip_from_user: 1000 sip_from_host: 192.168.10.89 sip_request_host: 192.168.10.89 sip_auth_qop: auth sip_auth_cnonce: 6096bb96b4316b2a47bf52299b9156f5 sip_auth_nc: 00000002 sip_auth_response: 7bdb41c435cb7f46bc44b52fce5c7034 sip_auth_method: REGISTER key: id user: 1000 domain: 192.168.10.89 ip: 192.168.18.120 in both the event?sip_auth_method is REGISTERED and every header parameter are the same, but in the second one the user is not registered now ! how to find out the different between them ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120703/8a0b3adb/attachment-0001.html From ssinyagin at yahoo.com Wed Jul 4 10:34:12 2012 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Tue, 3 Jul 2012 23:34:12 -0700 (PDT) Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: <047301cd595f$095960b0$1c0c2210$@bizfocused.com> References: <1341335638.5953.23.camel@mythtv> <1341344894.5953.27.camel@mythtv> <1341347834.5953.29.camel@mythtv> <047301cd595f$095960b0$1c0c2210$@bizfocused.com> Message-ID: <1341383652.89194.YahooMailNeo@web39302.mail.mud.yahoo.com> I'm using Gigaset 610IP, they are not too expensive, and work great >________________________________ > From: Sean Devoy >To: 'FreeSWITCH Users Help' >Sent: Tuesday, July 3, 2012 11:01 PM >Subject: Re: [Freeswitch-users] Fs compatible phones? > >Todd, > >I think the general rule you are looking for is "Any Internet phone that >uses the SIP protocol should work."? That rules out the 3COM NBX phones. > >I have used Cisco 504Gs rather extensively.? Once we got the NAT settings >down, they have been great.? They do have backlighting, but are rather >costly at about $175 - $200. > >I am watching replies to your question for an answer from someone who says >something "I use ______ and they are cheap, have 4 lines and work great!" > >Hope that helps a little. > >Sean >-----Original Message----- >From: freeswitch-users-bounces at lists.freeswitch.org >[mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Todd >Bailey >Sent: Tuesday, July 03, 2012 4:37 PM >To: FreeSWITCH Users Help >Subject: Re: [Freeswitch-users] Fs compatible phones? > >My question is evolving into not who but what. > >It it safe to say any phone marketed as a 'IP Phone' will work or are there >gotchas that I need to know about ? > > > >On Tue, 2012-07-03 at 12:48 -0700, Todd Bailey wrote: >> Thanks for the info but well beyond my price range. >> The 3com is cheap and probably due to being tied to the specific pbx >> package. >> >> The Cisco has good reviews but the display isn't back lit, making use >> in a dark room problematic. >> >> The used GrandStream models are affordable but from what I read suffer >> from poor audio. >> >> I rapidly running out of options here, any one else care to jump in? >> >> >> >> On Tue, 2012-07-03 at 14:22 -0400, Roger Castaldo wrote: >> > I have had good success with snom phones and they are solid for the >> > price >> > >> > On Jul 3, 2012 12:36 PM, "Todd Bailey" wrote: >> >? ? ? ? Hi All, >> >? ? ? ? >> >? ? ? ? New to this tech and a bit confused on what to buy/features >> >? ? ? ? etc. >> >? ? ? ? >> >? ? ? ? I'm looking for a basic desk phone, perhaps with a few speed >> >? ? ? ? dial and >> >? ? ? ? maybe gee-wiz buttons. I found this one with a decent price >> >? ? ? ? 3Com NBX >> >? ? ? ? 1102 but the ad states Requires R4.X system software or >> >? ? ? ? greater.? Not >> >? ? ? ? sure what this is nor do I need to? >> >? ? ? ? >> >? ? ? ? If I buy "any" digital phone that has a rj-45 ethernet jack >> >? ? ? ? ? does that >> >? ? ? ? automatically mean it's compatible? >> >? ? ? ? >> >? ? ? ? >> >? ? ? ? other models in my price/feature range >> >? ? ? ? >> >? ? ? ? ? ? ? ? Cisco CP-7940G Unified IP Phone >> >? ? ? ? >> >? ? ? ? I was looking at a few grand stream models but the reviews >> >? ? ? ? lead me to >> >? ? ? ? move on >> >? ? ? ? >> >? ? ? ? >> >? ? ? ? thanks >> >? ? ? ? >> >? ? ? ? >> >? ? ? ? >> >? ? ? ? >> > >_________________________________________________________________________ >> >? ? ? ? Professional FreeSWITCH Consulting Services: >> >? ? ? ? consulting at freeswitch.org >> >? ? ? ? http://www.freeswitchsolutions.com >> >? ? ? ? >> >? ? ? ? >> >? ? ? ? >> >? ? ? ? >> >? ? ? ? Official FreeSWITCH Sites >> >? ? ? ? http://www.freeswitch.org >> >? ? ? ? http://wiki.freeswitch.org >> >? ? ? ? http://www.cluecon.com >> >? ? ? ? >> >? ? ? ? Join Us At ClueCon - Aug 7-9, 2012 >> >? ? ? ? >> >? ? ? ? FreeSWITCH-users mailing list >> >? ? ? ? FreeSWITCH-users at lists.freeswitch.org >> >? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >? ? ? ? http://www.freeswitch.org >> > ____________________________________________________________________ >> > _____ Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u >> > sers >> > http://www.freeswitch.org >> >> >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users 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 >> > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >Join Us At ClueCon - Aug 7-9, 2012 > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >Join Us At ClueCon - Aug 7-9, 2012 > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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/20120703/7161c4c3/attachment.html From ssinyagin at yahoo.com Wed Jul 4 10:41:41 2012 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Tue, 3 Jul 2012 23:41:41 -0700 (PDT) Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: <1341374776.5343.6.camel@mythtv> References: <1341335638.5953.23.camel@mythtv> <1341374776.5343.6.camel@mythtv> Message-ID: <1341384101.16847.YahooMailNeo@web39303.mail.mud.yahoo.com> maybe a softphone on user's PC would be a better option? There are some free options available. there's another topic that you didn't touch yet: automatic provisioning. If you have a few dozens of phone, you don't want to configure every one of them individually from the handset. >________________________________ > From: Todd Bailey >To: FreeSWITCH Users Help >Sent: Wednesday, July 4, 2012 6:06 AM >Subject: Re: [Freeswitch-users] Fs compatible phones? > >Thanks for all the good advice, but sadly out of my current price range. > >I'm finding several phones sub $50 range listed on ebay, A few so far >and this list will grow, > >Toshiba IP5122-SD >Cisco CP-6921 >Cisco 7940 >Cisco SPA-942 >Fanvil FV6020 >Avaya 4412D+ >Soyo 7? G668 >Polycom Soundpoint IP501 > >Any Opinions? > >On Tue, 2012-07-03 at 10:13 -0700, Todd Bailey wrote: >> Hi All, >> >> New to this tech and a bit confused on what to buy/features etc. >> >> I'm looking for a basic desk phone, perhaps with a few speed dial and >> maybe gee-wiz buttons. I found this one with a decent price 3Com NBX >> 1102 but the ad states Requires R4.X system software or greater.? Not >> sure what this is nor do I need to? >> >> If I buy "any" digital phone that has a rj-45 ethernet jack? does that >> automatically mean it's compatible? >> >> >> other models in my price/feature range >> >> ??? Cisco CP-7940G Unified IP Phone >> >> I was looking at a few grand stream models but the reviews lead me to >> move on >> >> >> thanks >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >Join Us At ClueCon - Aug 7-9, 2012 > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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/20120703/ca5f2a76/attachment-0001.html From casteven at gmail.com Wed Jul 4 11:00:36 2012 From: casteven at gmail.com (Campbell Steven) Date: Wed, 4 Jul 2012 19:00:36 +1200 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: <1341381673.5343.9.camel@mythtv> References: <1341335638.5953.23.camel@mythtv> <1341381673.5343.9.camel@mythtv> Message-ID: I would strongly suggest the old Cisco is a bad choice, I have alot of uptime with those handsets on alternative platforms, I say alternative as really they are just designed to work with Call Manager. If you had to pick from the bunch you suggested I would probably pick the SPA-942, not that it's a great phone but it's the best of that average lot. The reasons I would pick it: -Full web UI that you can configure pretty much everything without having to resort to provisioning straight up -BLF support (useful to have and the Cisco 7960 doesn't, under SIP anyway) -Ability to centrally provision if you so desire in the future -Can register 4 SIP accounts -Can do Broadsoft SLA -802.3af PoE (unlike the 7960 which is pre-standard) Normally I'd be pushing Polycom as well, but the 501 was getting old when the 942 brand new... Campbell On 4 July 2012 18:01, Todd Bailey wrote: > I'm going to purchase a Cisco 7960 from ebay for under $50 unless > someone can tell me this is a bad idea. I watched several Utube videos > where is was working on Asterisk. With the volumns of documentation and > how to videos I think this is the better choice, > > > On Tue, 2012-07-03 at 10:13 -0700, Todd Bailey wrote: >> Hi All, >> >> New to this tech and a bit confused on what to buy/features etc. >> >> I'm looking for a basic desk phone, perhaps with a few speed dial and >> maybe gee-wiz buttons. I found this one with a decent price 3Com NBX >> 1102 but the ad states Requires R4.X system software or greater. Not >> sure what this is nor do I need to? >> >> If I buy "any" digital phone that has a rj-45 ethernet jack does that >> automatically mean it's compatible? >> >> >> other models in my price/feature range >> >> Cisco CP-7940G Unified IP Phone >> >> I was looking at a few grand stream models but the reviews lead me to >> move on >> >> >> thanks >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ben at langfeld.co.uk Wed Jul 4 12:51:25 2012 From: ben at langfeld.co.uk (Ben Langfeld) Date: Wed, 4 Jul 2012 09:51:25 +0100 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: References: <1341335638.5953.23.camel@mythtv> <1341381673.5343.9.camel@mythtv> Message-ID: I'd second the recommendation to stay away from Cisco's junk. Regards, Ben Langfeld On 4 July 2012 08:00, Campbell Steven wrote: > I would strongly suggest the old Cisco is a bad choice, I have alot of > uptime with those handsets on alternative platforms, I say alternative > as really they are just designed to work with Call Manager. If you had > to pick from the bunch you suggested I would probably pick the > SPA-942, not that it's a great phone but it's the best of that average > lot. The reasons I would pick it: > > -Full web UI that you can configure pretty much everything without > having to resort to provisioning straight up > -BLF support (useful to have and the Cisco 7960 doesn't, under SIP anyway) > -Ability to centrally provision if you so desire in the future > -Can register 4 SIP accounts > -Can do Broadsoft SLA > -802.3af PoE (unlike the 7960 which is pre-standard) > > Normally I'd be pushing Polycom as well, but the 501 was getting old > when the 942 brand new... > > Campbell > > On 4 July 2012 18:01, Todd Bailey wrote: >> I'm going to purchase a Cisco 7960 from ebay for under $50 unless >> someone can tell me this is a bad idea. I watched several Utube videos >> where is was working on Asterisk. With the volumns of documentation and >> how to videos I think this is the better choice, >> >> >> On Tue, 2012-07-03 at 10:13 -0700, Todd Bailey wrote: >>> Hi All, >>> >>> New to this tech and a bit confused on what to buy/features etc. >>> >>> I'm looking for a basic desk phone, perhaps with a few speed dial and >>> maybe gee-wiz buttons. I found this one with a decent price 3Com NBX >>> 1102 but the ad states Requires R4.X system software or greater. Not >>> sure what this is nor do I need to? >>> >>> If I buy "any" digital phone that has a rj-45 ethernet jack does that >>> automatically mean it's compatible? >>> >>> >>> other models in my price/feature range >>> >>> Cisco CP-7940G Unified IP Phone >>> >>> I was looking at a few grand stream models but the reviews lead me to >>> move on >>> >>> >>> thanks >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nbhatti at gmail.com Wed Jul 4 13:06:41 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 4 Jul 2012 12:06:41 +0300 Subject: [Freeswitch-users] Load Balancing In-Reply-To: References: Message-ID: We did this little trick by adding an additional channel variable which holds the actual IP address forwarded by OpenSIPS. You can later play around with this variable to do post processing etc. -B On Tue, Jul 3, 2012 at 7:39 AM, Hanie Maghsoudy wrote: > I think I got it! it's explained herein packet forwarding section. Changing the source won't happen. > > On Mon, Jul 2, 2012 at 5:53 PM, Hanie Maghsoudy wrote: > >> Dear all, >> >> I tried OpenSIPs for load balancing. I works just fine, but as it is >> mentioned here,OpenSIPs sends requests from itself. >> How about Ultramonkey? Does it change the source too, or it just >> manipulate destination of registration packets? >> >> Thanks, >> Hanie >> >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120704/b90173ef/attachment.html From freeswitch at top-nachhilfe.de Wed Jul 4 13:27:03 2012 From: freeswitch at top-nachhilfe.de (freeswitchtest) Date: Wed, 04 Jul 2012 11:27:03 +0200 Subject: [Freeswitch-users] freeswitch - openfire - chat In-Reply-To: References: <1341335638.5953.23.camel@mythtv> <1341381673.5343.9.camel@mythtv> Message-ID: <4FF40C67.4040203@top-nachhilfe.de> Hello, perhaps someone can give me an advice: I have written an chat with openfire (built with strophe) that works fine FS is connected to the openfire as an external component - this works correct. Now I want freeswitch to send an message to the (an) chatroom every time a call is coming in or at other special events How could I realize it ? Thank you very much for any answer Greetings Robert From h.maghsoudy at gmail.com Wed Jul 4 13:53:50 2012 From: h.maghsoudy at gmail.com (Hanie Maghsoudy) Date: Wed, 4 Jul 2012 14:23:50 +0430 Subject: [Freeswitch-users] Load Balancing In-Reply-To: References: Message-ID: Thanks Muhammad, that would be a good idea. However, I'm trying direct routing mode of IPVS for load balancing along with Heartbeat for HA. Could anyone of you guys suggest a better way to implement Load balancing and HA for FreeSwitch? Thanks On Wed, Jul 4, 2012 at 1:36 PM, Muhammad Naseer Bhatti wrote: > > We did this little trick by adding an additional channel variable which > holds the actual IP address forwarded by OpenSIPS. You can later play > around with this variable to do post processing etc. > > -B > > On Tue, Jul 3, 2012 at 7:39 AM, Hanie Maghsoudy wrote: > >> I think I got it! it's explained herein packet forwarding section. Changing the source won't happen. >> >> On Mon, Jul 2, 2012 at 5:53 PM, Hanie Maghsoudy wrote: >> >>> Dear all, >>> >>> I tried OpenSIPs for load balancing. I works just fine, but as it is >>> mentioned here,OpenSIPs sends requests from itself. >>> How about Ultramonkey? Does it change the source too, or it just >>> manipulate destination of registration packets? >>> >>> Thanks, >>> Hanie >>> >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120704/bea595ff/attachment-0001.html From jayesh.voip at gmail.com Wed Jul 4 14:55:07 2012 From: jayesh.voip at gmail.com (Jayesh Nambiar) Date: Wed, 4 Jul 2012 16:25:07 +0530 Subject: [Freeswitch-users] Call recording quality problems when both legs have different codecs Message-ID: Hi, I am using mod_callcenter and have enabled recording_template in my queues for all my calls to be recorded. Now there is a problem in quality of recorded file when freeswitch answers the call in one codec and connects with the agent on a different codec. For eg: The incoming call comes in and get connected with G711u codec. The caller is listening to MoH while waiting in queue. The agents have G722, G711u both enabled on their phones and the sofia has outbound-codec-prefs as G722,PCMU. So when the agent answers, the call is connected with G722 codec. So in this case, the recorded file generated plays as if the call is getting fast-forwarded. The audio in the call is very fast. Whereas if I change the codec of Agent's phone to G711u only the recorded file quality is perfect. I don't think it is related to callcenter module anywhere but since I am using it in callcenter I have given this example. I think its more related to the record function. I am running latest GIT version and using mod_shout for recording calls in mp3 format. I had the same problems when recording in wav format also. Any help regarding this is really appreciated. I have attached two files; one which has PCMu on first leg and 722 on other and the second one has PCMu on both legs. Thanks in advance. --- Jayesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120704/635c067f/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: same_codec_on_both_leg.mp3 Type: audio/mpeg Size: 30096 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120704/635c067f/attachment-0002.mp3 -------------- next part -------------- A non-text attachment was scrubbed... Name: different_codecs_on_both_legs.mp3 Type: audio/mpeg Size: 19728 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120704/635c067f/attachment-0003.mp3 From peter.olsson at visionutveckling.se Wed Jul 4 17:18:15 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 4 Jul 2012 13:18:15 +0000 Subject: [Freeswitch-users] FreeSWITCH, NTP daemon and clock drift Message-ID: <1FFF97C269757C458224B7C895F35F1512DDDB@cantor.std.visionutv.se> Tony, I tested this on Windows, and found a few glitches (some are probably there for Linux as well). Could you please check out FS-4387 for a proposed patch. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Anthony Minessale Skickat: den 4 juli 2012 00:16 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] FreeSWITCH, NTP daemon and clock drift My first instinct is to say: don't use boxes with unreliable timing in them since if they can't keep time, the precision for audio could be even worse. But, Its not an unreasonable request give this param a try in latest git: enable-use-system-time in switch.conf.xml the status uptime should still be right but the timestamps etc will follow system time. use at your own risk On Tue, Jul 3, 2012 at 2:36 PM, Steven Ayre wrote: > Anthony, > > There's previously been a discussion captured on > http://wiki.freeswitch.org/wiki/Clock regarding the internal clock > time FS keeps. > > Currently FreeSWITCH uses the monotonic clock and ignores system time. > On machines where the time is very unreliable, NTP isn't in use and > the sysadmin is changing the time manually or by ntpdate the system > time can make large jumps - and I can indeed see the reasoning that > having FS use its own clock on these systems is more reliable as it > prevents billing time being lost because of a large clock jump. > > However, on systems correctly running ntpd that have clock drift the > system time will constantly be being corrected and won't experience > large jumps and in these cases the system time will be more accurate > than FreeSWITCH's internal time. > > On such systems although FreeSWITCH might not be losing billing > seconds from clock jumps (which shouldn't normally happen) the clock > drift will mean that there is also that clock drift present in > FreeSWITCH's CDRs - and these could conceivably cause billing disputes > where a customer's CDRs do not have the same times because their times > are more accurate than FS's due to that clock drift. This is unlikely > to be large enough to affect billed minutes much, but could put a > large number of calls on the wrong billing rate (eg if FS's internal > time in the CDRs identifies calls as on a peak rate when the customer > believes it is on an offpeak rate). On a badly drifting system over a > month on a large volume of calls that could become a noticeable discrepancy. > > Obviously there is the sync_clock option, but it seems you shouldn't > need to run that frequently (and would that cause any sideeffects?) > Running it infrequently would cause the same time-jumping behaviour > FS's internal time was designed to avoid, and sync_clock_when_idle > isn't possible on busier systems. > > On systems running ntpd perhaps it would be worth adding an opt-in > option that lets FreeSWITCH use the system time and rely on ntpd (but > keeping the present behaviour as the default)? > > Your thoughts please... > > Warm regards, > -Steve > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff36ed932765090247789! From adam at vrs.pl Wed Jul 4 11:13:08 2012 From: adam at vrs.pl (Adam Obuchowski) Date: Wed, 4 Jul 2012 09:13:08 +0200 Subject: [Freeswitch-users] Registration from behind NAT doesn't work Message-ID: Hi Im having problem on registering some sip devices that are behind NAT and dont support STUN. Please have a look at: ftp://v2.vrs.pl/pub/test.pcap Freeswitch replies to public IP which means it knows its NATed device, but why not to port that request comes from ? I found that this can be forced by I have added this line to extension file under section, but it brings no results. Im new to freeswitch so if you need some action from my side, thread me as newbie. Kind regards, -- Adam Obuchowski -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120704/ccf29992/attachment.html From akostenko at broadvox.com Wed Jul 4 11:48:08 2012 From: akostenko at broadvox.com (Alexandr Kostenko) Date: Wed, 4 Jul 2012 10:48:08 +0300 Subject: [Freeswitch-users] Freeswitch console log and file log In-Reply-To: Message-ID: <11778415.459.1341388085338.JavaMail.a66a@a66a> Thanks a lot, I will follow your advice. ----- Original Message ----- From: "Michael Collins" To: "FreeSWITCH Users Help" Sent: Wednesday, July 4, 2012 1:17:30 AM Subject: Re: [Freeswitch-users] Freeswitch console log and file log On Tue, Jul 3, 2012 at 3:10 AM, Alexandr Kostenko < akostenko at broadvox.com > wrote: Hi, Guys. Can you explain me more understandable about logs in freeswitch. Can I for example decrease console log level (or fully disable) and increase file log level? Also can full log level causing freeswitch crush ? Thanks You can have console logging at a different level than the log file. Look in conf/autoload_configs/console.conf.xml and logfile.conf.xml and you'll see that you have a fair number of options. In console.conf.xml there's a lot of explanatory information about mappings and how to make it so that only certain items have logging turned on at specific log levels. Note that what's listed in console.conf.xml also applies to logfile.conf.xml. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120704/dff0c44b/attachment.html From jack.nikolas at ymail.com Wed Jul 4 12:51:02 2012 From: jack.nikolas at ymail.com (Jack Nikolas) Date: Wed, 4 Jul 2012 09:51:02 +0100 (BST) Subject: [Freeswitch-users] how to capture packets of siptrace? Message-ID: <1341391862.55681.YahooMailNeo@web171501.mail.ir2.yahoo.com> hi all, in conf.back/sip_profiles/internal.xml? configuration i have enabled two options as below: but how can i capture packets when user is want to register sip client or making calls or logout or transfer, etc ? i want the following content ? recv 529 bytes from udp/[85.15.7.6]:23944 at 08:48:58.358101: ?? ?? SUBSCRIBE sip:XXXXXX ?? Via: SIP/2.0/UDP XXXX:23944;branch=z9hG4bK-d8754z-7d25245fc51dd824-1---d8754z-;rport ?? Max-Forwards: 70 ?? Contact: XXXX ?? To: XXXXX ?? From: XXXXX ?? Call-ID: ZjE3ZGMyNTBhODI5NTU0Y2RmZmMwYTY4ODQ2NDJjNzQ. ?? CSeq: 4 SUBSCRIBE ?? Expires: 0 ?? User-Agent: eyeBeam release 1102q stamp 51814 ?? Event: message-summary ?? Content-Length: 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120704/07e16f67/attachment.html From khuenm at vega.com.vn Wed Jul 4 15:04:06 2012 From: khuenm at vega.com.vn (Khue Nguyen Minh) Date: Wed, 4 Jul 2012 18:04:06 +0700 Subject: [Freeswitch-users] how to play mp4 file Message-ID: Hi all, I want play mp4 file in freeswitch. Please guide me how I can play it? Brs, Khue Nguyen. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120704/45ed7972/attachment.html From dujinfang at gmail.com Wed Jul 4 19:32:53 2012 From: dujinfang at gmail.com (Seven Du) Date: Wed, 4 Jul 2012 23:32:53 +0800 Subject: [Freeswitch-users] how to play mp4 file In-Reply-To: References: Message-ID: check mod_mp4 also I'm working on mod_vlc which can play .mp4 (and possiblely any video VLC can) in my lab now with libx264 On Wednesday, July 4, 2012 at 7:04 PM, Khue Nguyen Minh wrote: > Hi all, > > I want play mp4 file in freeswitch. Please guide me how I can play it? > > Brs, > Khue Nguyen. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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/20120704/fb7f2af4/attachment-0001.html From abaci64 at gmail.com Wed Jul 4 20:25:49 2012 From: abaci64 at gmail.com (Abaci) Date: Wed, 04 Jul 2012 12:25:49 -0400 Subject: [Freeswitch-users] question: lua how can i create a postcastlist? In-Reply-To: References: Message-ID: <4FF46E8D.9040804@gmail.com> you can use LuaFileSystem http://keplerproject.github.com/luafilesystem/ On 7/3/2012 5:48 PM, Michael Collins wrote: > You'll have a challenge here because natively Lua has no directory > commands. I have never tried reading in the results of a system call > into a Lua table, but it might be possible. I'd ask on #Lua or check > Lua docs to see if such a thing is even possible. Once you have the > files in a table then you could do the next/prev/first/last/repeat thing. > > -MC > > On Tue, Jul 3, 2012 at 8:58 AM, Thomas Hoellriegel > wrote: > > Hi guys, > I have many potcast mp3 files in a directory: > /home/postcast > the files beginning with: > postcast_year_mon_day_hour_minutes_.mp3 > I like to create a playlist in lua: > 1. read all files, and make a messagecount. > 2. generate a menu to use from phone: > > press 1 for the next postcast. > 2, To repeate the postcast > 3. to go to the prev potcast > 4 for the fist potcast > 5 for the last potcast > 6 to hear the date an time for these. > > I like to sort, firt in first out. > > I don.t find a command in lua: > to read the filelist, > to generate a filecount. > and to generate a navigation keys: > forward, backward, repeate and say the time and date of the > current file. > > Can your help please? > > nice thanks to all. > > > > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120704/ab02bd50/attachment.html From toddb at toddbailey.net Wed Jul 4 20:42:06 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Wed, 04 Jul 2012 09:42:06 -0700 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: References: <1341335638.5953.23.camel@mythtv> <1341381673.5343.9.camel@mythtv> Message-ID: <1341420126.5343.19.camel@mythtv> OK, No Cisco phones even though the feature set on the 7960 is rather impressive. Using a pc for a phone is an option and the 3cx does pretty much every thing I'm looking for. But, I'd like to avoid having a pc/laptop running a phone app 24x7 as it's a bit hard on the power bill. However is there a way to run twinkle or linphone on the FS server? So far I get error messages stating that the ports are already in use. Meanwhile I'll refocus on polycom ip phones off ebay. On Wed, 2012-07-04 at 19:00 +1200, Campbell Steven wrote: > I would strongly suggest the old Cisco is a bad choice, I have alot of > uptime with those handsets on alternative platforms, I say alternative > as really they are just designed to work with Call Manager. If you had > to pick from the bunch you suggested I would probably pick the > SPA-942, not that it's a great phone but it's the best of that average > lot. The reasons I would pick it: > > -Full web UI that you can configure pretty much everything without > having to resort to provisioning straight up > -BLF support (useful to have and the Cisco 7960 doesn't, under SIP anyway) > -Ability to centrally provision if you so desire in the future > -Can register 4 SIP accounts > -Can do Broadsoft SLA > -802.3af PoE (unlike the 7960 which is pre-standard) > > Normally I'd be pushing Polycom as well, but the 501 was getting old > when the 942 brand new... > > Campbell > > On 4 July 2012 18:01, Todd Bailey wrote: > > I'm going to purchase a Cisco 7960 from ebay for under $50 unless > > someone can tell me this is a bad idea. I watched several Utube videos > > where is was working on Asterisk. With the volumns of documentation and > > how to videos I think this is the better choice, > > > > > > On Tue, 2012-07-03 at 10:13 -0700, Todd Bailey wrote: > >> Hi All, > >> > >> New to this tech and a bit confused on what to buy/features etc. > >> > >> I'm looking for a basic desk phone, perhaps with a few speed dial and > >> maybe gee-wiz buttons. I found this one with a decent price 3Com NBX > >> 1102 but the ad states Requires R4.X system software or greater. Not > >> sure what this is nor do I need to? > >> > >> If I buy "any" digital phone that has a rj-45 ethernet jack does that > >> automatically mean it's compatible? > >> > >> > >> other models in my price/feature range > >> > >> Cisco CP-7940G Unified IP Phone > >> > >> I was looking at a few grand stream models but the reviews lead me to > >> move on > >> > >> > >> thanks > >> > >> > >> > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From toddb at toddbailey.net Wed Jul 4 21:26:09 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Wed, 04 Jul 2012 10:26:09 -0700 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: <1341335638.5953.23.camel@mythtv> References: <1341335638.5953.23.camel@mythtv> Message-ID: <1341422769.5343.20.camel@mythtv> OK, then, how about a Polycom IP 501? On Tue, 2012-07-03 at 10:13 -0700, Todd Bailey wrote: > Hi All, > > New to this tech and a bit confused on what to buy/features etc. > > I'm looking for a basic desk phone, perhaps with a few speed dial and > maybe gee-wiz buttons. I found this one with a decent price 3Com NBX > 1102 but the ad states Requires R4.X system software or greater. Not > sure what this is nor do I need to? > > If I buy "any" digital phone that has a rj-45 ethernet jack does that > automatically mean it's compatible? > > > other models in my price/feature range > > Cisco CP-7940G Unified IP Phone > > I was looking at a few grand stream models but the reviews lead me to > move on > > > thanks > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From ssinyagin at yahoo.com Wed Jul 4 22:32:01 2012 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Wed, 4 Jul 2012 11:32:01 -0700 (PDT) Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: <1341420126.5343.19.camel@mythtv> References: <1341335638.5953.23.camel@mythtv> <1341381673.5343.9.camel@mythtv> <1341420126.5343.19.camel@mythtv> Message-ID: <1341426721.32921.YahooMailNeo@web39302.mail.mud.yahoo.com> you can also run a SIP client on an Android device, or even on an Apple device :) Also there are SIP-to-Analog adapters: for example, Cisco SPA2102 costs around $60, and it supports 2 analog phones. the cheapest hardware SIP phone that I can find (Central Europe) is FRITZ!Fon MT-D, which is about $80 a piece. >________________________________ > From: Todd Bailey >To: freeswitch-users at lists.freeswitch.org >Sent: Wednesday, July 4, 2012 6:42 PM >Subject: Re: [Freeswitch-users] Fs compatible phones? > >OK, No Cisco phones even though the feature set on the 7960 is rather >impressive. > >Using a pc for a phone is an option and the 3cx does pretty much every >thing I'm looking for.? But, I'd like to avoid having a pc/laptop >running a phone app 24x7 as it's a bit hard on the power bill. >However is there a way to run twinkle or linphone on the FS server?? So >far I get error messages stating that the ports are already in use.? > >Meanwhile I'll refocus on polycom ip phones off ebay. > > > >On Wed, 2012-07-04 at 19:00 +1200, Campbell Steven wrote: >> I would strongly suggest the old Cisco is a bad choice, I have alot of >> uptime with those handsets on alternative platforms, I say alternative >> as really they are just designed to work with Call Manager. If you had >> to pick from the bunch you suggested I would probably pick the >> SPA-942, not that it's a great phone but it's the best of that average >> lot. The reasons I would pick it: >> >> -Full web UI that you can configure pretty much everything without >> having to resort to provisioning straight up >> -BLF support (useful to have and the Cisco 7960 doesn't, under SIP anyway) >> -Ability to centrally provision if you so desire in the future >> -Can register 4 SIP accounts >> -Can do Broadsoft SLA >> -802.3af PoE (unlike the 7960 which is pre-standard) >> >> Normally I'd be pushing Polycom as well, but the 501 was getting old >> when the 942 brand new... >> >> Campbell >> >> On 4 July 2012 18:01, Todd Bailey wrote: >> > I'm going to purchase a Cisco 7960 from ebay for under $50 unless >> > someone can tell me this is a bad idea.? I watched several Utube videos >> > where is was working on Asterisk. With the volumns of documentation and >> > how to videos I think this is the better choice, >> > >> > >> > On Tue, 2012-07-03 at 10:13 -0700, Todd Bailey wrote: >> >> Hi All, >> >> >> >> New to this tech and a bit confused on what to buy/features etc. >> >> >> >> I'm looking for a basic desk phone, perhaps with a few speed dial and >> >> maybe gee-wiz buttons. I found this one with a decent price 3Com NBX >> >> 1102 but the ad states Requires R4.X system software or greater.? Not >> >> sure what this is nor do I need to? >> >> >> >> If I buy "any" digital phone that has a rj-45 ethernet jack? does that >> >> automatically mean it's compatible? >> >> >> >> >> >> other models in my price/feature range >> >> >> >>? ? ? Cisco CP-7940G Unified IP Phone >> >> >> >> I was looking at a few grand stream models but the reviews lead me to >> >> move on >> >> >> >> >> >> thanks >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >Join Us At ClueCon - Aug 7-9, 2012 > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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/20120704/e3a25ba0/attachment-0001.html From engelster at gmail.com Wed Jul 4 22:32:11 2012 From: engelster at gmail.com (Der Engel) Date: Wed, 4 Jul 2012 13:32:11 -0500 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: <1341422769.5343.20.camel@mythtv> References: <1341335638.5953.23.camel@mythtv> <1341422769.5343.20.camel@mythtv> Message-ID: For most supported I guess you should stick to Polycom since is what cudatel supports most and what devs work most with. On Wed, Jul 4, 2012 at 12:26 PM, Todd Bailey wrote: > OK, then, how about a Polycom IP 501? > > > > On Tue, 2012-07-03 at 10:13 -0700, Todd Bailey wrote: >> Hi All, >> >> New to this tech and a bit confused on what to buy/features etc. >> >> I'm looking for a basic desk phone, perhaps with a few speed dial and >> maybe gee-wiz buttons. I found this one with a decent price 3Com NBX >> 1102 but the ad states Requires R4.X system software or greater. ? Not >> sure what this is nor do I need to? >> >> If I buy "any" digital phone that has a rj-45 ethernet jack ?does that >> automatically mean it's compatible? >> >> >> other models in my price/feature range >> >> ? ? ? Cisco CP-7940G Unified IP Phone >> >> I was looking at a few grand stream models but the reviews lead me to >> move on >> >> >> thanks >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bdfoster at endigotech.com Wed Jul 4 22:44:49 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 4 Jul 2012 14:44:49 -0400 Subject: [Freeswitch-users] how to capture packets of siptrace? In-Reply-To: <1341391862.55681.YahooMailNeo@web171501.mail.ir2.yahoo.com> References: <1341391862.55681.YahooMailNeo@web171501.mail.ir2.yahoo.com> Message-ID: Either do this from fs_cli: sofia profile siptrace on ...or do a tcpdump and look at it in wireshark. Second option is more comprehensive and "looks pretty". Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 4, 2012 9:37 AM, "Jack Nikolas" wrote: > hi all, > in conf.back/sip_profiles/internal.xml configuration > i have enabled two options as below: > > > > but how can i capture packets when user is want to register sip client or > making calls or logout or transfer, etc ? > i want the following content > ? > recv 529 bytes from udp/[85.15.7.6]:23944 at 08:48:58.358101: > > SUBSCRIBE sip:XXXXXX > Via: SIP/2.0/UDP > XXXX:23944;branch=z9hG4bK-d8754z-7d25245fc51dd824-1---d8754z-;rport > Max-Forwards: 70 > Contact: XXXX > To: XXXXX > From: XXXXX > Call-ID: ZjE3ZGMyNTBhODI5NTU0Y2RmZmMwYTY4ODQ2NDJjNzQ. > CSeq: 4 SUBSCRIBE > Expires: 0 > User-Agent: eyeBeam release 1102q stamp 51814 > Event: message-summary > Content-Length: 0 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120704/df4c9e9b/attachment.html From bdfoster at endigotech.com Wed Jul 4 22:57:30 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 4 Jul 2012 14:57:30 -0400 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: References: <1341335638.5953.23.camel@mythtv> <1341422769.5343.20.camel@mythtv> Message-ID: I don't really know the cudatel product, id imagine they have some stuff that they use to interface with the UC stuff on the Polycoms. However, that doesn't make those phones superior to others as much of that stuff is outside FreeSWITCH. SPA's are nice, stay away from the cisco 79xx. They aren't worth the trouble (as others have stated). Polycoms are nice but expensive, Snoms are good, Yealink is ok, etc. I use Polycom 335's and SPA 921's, I like the polycoms better but it comes at a cost. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 4, 2012 2:49 PM, "Der Engel" wrote: > For most supported I guess you should stick to Polycom since is what > cudatel supports most and what devs work most with. > > On Wed, Jul 4, 2012 at 12:26 PM, Todd Bailey wrote: > > OK, then, how about a Polycom IP 501? > > > > > > > > On Tue, 2012-07-03 at 10:13 -0700, Todd Bailey wrote: > >> Hi All, > >> > >> New to this tech and a bit confused on what to buy/features etc. > >> > >> I'm looking for a basic desk phone, perhaps with a few speed dial and > >> maybe gee-wiz buttons. I found this one with a decent price 3Com NBX > >> 1102 but the ad states Requires R4.X system software or greater. Not > >> sure what this is nor do I need to? > >> > >> If I buy "any" digital phone that has a rj-45 ethernet jack does that > >> automatically mean it's compatible? > >> > >> > >> other models in my price/feature range > >> > >> Cisco CP-7940G Unified IP Phone > >> > >> I was looking at a few grand stream models but the reviews lead me to > >> move on > >> > >> > >> thanks > >> > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120704/b916d815/attachment.html From toddb at toddbailey.net Wed Jul 4 23:16:16 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Wed, 04 Jul 2012 12:16:16 -0700 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: References: <1341335638.5953.23.camel@mythtv> <1341422769.5343.20.camel@mythtv> Message-ID: <1341429376.25744.4.camel@mythtv> OK Polycom it is. My limited research indicates the ip 501 is probably going to be the best you can get in my budget range. Several on Ebay for $50 or less. I have 3cx on my android, it it works nearly as well as the pc based apps, excepts it takes over the entire screen when open. Still any app like this would have to be running 24x7, other wise no phone ring and inbound calls go to vm. On Wed, 2012-07-04 at 14:57 -0400, Brian Foster wrote: > I don't really know the cudatel product, id imagine they have some > stuff that they use to interface with the UC stuff on the Polycoms. > However, that doesn't make those phones superior to others as much of > that stuff is outside FreeSWITCH. > > SPA's are nice, stay away from the cisco 79xx. They aren't worth the > trouble (as others have stated). Polycoms are nice but expensive, > Snoms are good, Yealink is ok, etc. I use Polycom 335's and SPA 921's, > I like the polycoms better but it comes at a cost. > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > > On Jul 4, 2012 2:49 PM, "Der Engel" wrote: > For most supported I guess you should stick to Polycom since > is what > cudatel supports most and what devs work most with. > > On Wed, Jul 4, 2012 at 12:26 PM, Todd Bailey > wrote: > > OK, then, how about a Polycom IP 501? > > > > > > > > On Tue, 2012-07-03 at 10:13 -0700, Todd Bailey wrote: > >> Hi All, > >> > >> New to this tech and a bit confused on what to buy/features > etc. > >> > >> I'm looking for a basic desk phone, perhaps with a few > speed dial and > >> maybe gee-wiz buttons. I found this one with a decent price > 3Com NBX > >> 1102 but the ad states Requires R4.X system software or > greater. Not > >> sure what this is nor do I need to? > >> > >> If I buy "any" digital phone that has a rj-45 ethernet > jack does that > >> automatically mean it's compatible? > >> > >> > >> other models in my price/feature range > >> > >> Cisco CP-7940G Unified IP Phone > >> > >> I was looking at a few grand stream models but the reviews > lead me to > >> move on > >> > >> > >> thanks > >> > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From itamar at ispbrasil.com.br Wed Jul 4 23:23:02 2012 From: itamar at ispbrasil.com.br (Itamar Reis Peixoto) Date: Wed, 4 Jul 2012 16:23:02 -0300 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: References: <1341335638.5953.23.camel@mythtv> <1341422769.5343.20.camel@mythtv> Message-ID: On Wed, Jul 4, 2012 at 3:57 PM, Brian Foster wrote: > I don't really know the cudatel product, id imagine they have some stuff > that they use to interface with the UC stuff on the Polycoms. However, that > doesn't make those phones superior to others as much of that stuff is > outside FreeSWITCH. > > SPA's are nice, stay away from the cisco 79xx. They aren't worth the trouble > (as others have stated). Polycoms are nice but expensive, Snoms are good, > Yealink is ok, etc. I use Polycom 335's and SPA 921's, I like the polycoms > better but it comes at a cost. > > Brian Foster > Endigo Computer LLC > how about android phone with wireless ? -- ------------ Itamar Reis Peixoto msn, google talk: itamar at ispbrasil.com.br +55 11 4063 5033 (FIXO SP) +55 34 9158 9329 (TIM) +55 34 8806 3989 (OI) +55 34 3221 8599 (FIXO MG) From itamar at ispbrasil.com.br Wed Jul 4 23:34:49 2012 From: itamar at ispbrasil.com.br (Itamar Reis Peixoto) Date: Wed, 4 Jul 2012 16:34:49 -0300 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: <1341429376.25744.4.camel@mythtv> References: <1341335638.5953.23.camel@mythtv> <1341422769.5343.20.camel@mythtv> <1341429376.25744.4.camel@mythtv> Message-ID: On Wed, Jul 4, 2012 at 4:16 PM, Todd Bailey wrote: > > I have 3cx on my android, it it works nearly as well as the pc based > apps, excepts it takes over the entire screen when open. > > Still any app like this would have to be running 24x7, other wise no > phone ring and inbound calls go to vm. I am using android internal client settings -> call settings -> accounts ------------ Itamar Reis Peixoto msn, google talk: itamar at ispbrasil.com.br +55 11 4063 5033 (FIXO SP) +55 34 9158 9329 (TIM) +55 34 8806 3989 (OI) +55 34 3221 8599 (FIXO MG) From bdfoster at endigotech.com Wed Jul 4 23:35:57 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 4 Jul 2012 15:35:57 -0400 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: References: <1341335638.5953.23.camel@mythtv> <1341422769.5343.20.camel@mythtv> Message-ID: When you introduce wireless there can be some audio issues (depending on what and how much that network is used for). There are ways of getting around that but it can be out of the users control and is outside the scope of this discussion. Softphones can be good, but they again are more trouble that they are worth. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 4, 2012 3:31 PM, "Itamar Reis Peixoto" wrote: > On Wed, Jul 4, 2012 at 3:57 PM, Brian Foster > wrote: > > I don't really know the cudatel product, id imagine they have some stuff > > that they use to interface with the UC stuff on the Polycoms. However, > that > > doesn't make those phones superior to others as much of that stuff is > > outside FreeSWITCH. > > > > SPA's are nice, stay away from the cisco 79xx. They aren't worth the > trouble > > (as others have stated). Polycoms are nice but expensive, Snoms are good, > > Yealink is ok, etc. I use Polycom 335's and SPA 921's, I like the > polycoms > > better but it comes at a cost. > > > > Brian Foster > > Endigo Computer LLC > > > > > how about android phone with wireless ? > > > > > > -- > ------------ > > Itamar Reis Peixoto > msn, google talk: itamar at ispbrasil.com.br > +55 11 4063 5033 (FIXO SP) > +55 34 9158 9329 (TIM) > +55 34 8806 3989 (OI) > +55 34 3221 8599 (FIXO MG) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120704/31e3af1e/attachment.html From jmesquita at freeswitch.org Thu Jul 5 00:02:52 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Wed, 4 Jul 2012 17:02:52 -0300 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: References: <1341335638.5953.23.camel@mythtv> <1341422769.5343.20.camel@mythtv> Message-ID: That is indeed my personal experience. Specially if you live in a country where home assembled machines are popular on the corporate environment. What that generates is that the audio interfaces are so crappy that they become a headache. Most ppl I've seen end up buying good headsets that are worth pretty much the same as what a simple IP Phone would cost. -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Wednesday, July 4, 2012 at 4:35 PM, Brian Foster wrote: > When you introduce wireless there can be some audio issues (depending on what and how much that network is used for). There are ways of getting around that but it can be out of the users control and is outside the scope of this discussion. > Softphones can be good, but they again are more trouble that they are worth. > Brian Foster > Endigo Computer LLC > Sent from a mobile device. > On Jul 4, 2012 3:31 PM, "Itamar Reis Peixoto" wrote: > > On Wed, Jul 4, 2012 at 3:57 PM, Brian Foster wrote: > > > I don't really know the cudatel product, id imagine they have some stuff > > > that they use to interface with the UC stuff on the Polycoms. However, that > > > doesn't make those phones superior to others as much of that stuff is > > > outside FreeSWITCH. > > > > > > SPA's are nice, stay away from the cisco 79xx. They aren't worth the trouble > > > (as others have stated). Polycoms are nice but expensive, Snoms are good, > > > Yealink is ok, etc. I use Polycom 335's and SPA 921's, I like the polycoms > > > better but it comes at a cost. > > > > > > Brian Foster > > > Endigo Computer LLC > > > > > > > > > how about android phone with wireless ? > > > > > > > > > > > > -- > > ------------ > > > > Itamar Reis Peixoto > > msn, google talk: itamar at ispbrasil.com.br (mailto:itamar at ispbrasil.com.br) > > +55 11 4063 5033 (tel:%2B55%2011%204063%205033) (FIXO SP) > > +55 34 9158 9329 (tel:%2B55%2034%209158%209329) (TIM) > > +55 34 8806 3989 (tel:%2B55%2034%208806%203989) (OI) > > +55 34 3221 8599 (tel:%2B55%2034%203221%208599) (FIXO MG) > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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/20120704/77c49aac/attachment.html From avi at avimarcus.net Thu Jul 5 00:49:00 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 4 Jul 2012 23:49:00 +0300 Subject: [Freeswitch-users] Fs compatible phones? In-Reply-To: <1341426721.32921.YahooMailNeo@web39302.mail.mud.yahoo.com> References: <1341335638.5953.23.camel@mythtv> <1341381673.5343.9.camel@mythtv> <1341420126.5343.19.camel@mythtv> <1341426721.32921.YahooMailNeo@web39302.mail.mud.yahoo.com> Message-ID: On Wed, Jul 4, 2012 at 9:32 PM, Stanislav Sinyagin wrote: > Also there are SIP-to-Analog adapters: for example, Cisco SPA2102 costs > around $60, and it supports 2 analog phones. > > > I'm a fan of the spa-2102. It won't let you do any fancy SIP stuff... no presence, no built in transfer or the like, no big caller ID screen. But they seem to be pretty reliable. It even seems to handle NAT pretty well. Provisioning is pretty easy and fast. Just my experience. It's an ATA not a sip phone, though... -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120704/0615e1b2/attachment-0001.html From neo.cheema at gmail.com Thu Jul 5 02:06:59 2012 From: neo.cheema at gmail.com (Neo Cheema) Date: Thu, 5 Jul 2012 03:36:59 +0530 Subject: [Freeswitch-users] Error in compilation after git pull Message-ID: Hi All, I did a git pull just now. After make clean, the compile failed. The latest git commit log that I can see is: commit 04bd463d12bf4c0116bec3cee537749b2040ed40 Author: Michael S Collins Date: Wed Jul 4 12:09:32 2012 -0700 The relevant error is: making all mod_spandsp Creating mod_spandsp_la-mod_spandsp.lo quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../../../../src/include -I../../../../libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT mod_spandsp_la-mod_spandsp.lo -MD -MP -MF .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o .libs/mod_spandsp_la-mod_spandsp.o In file included from /usr/local/src/freeswitch/libs/spandsp/src/spandsp/expose.h:96, from /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:135, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:33: error: expected specifier-qualifier-list before ?ademco_contactid_report_func_t? cc1: warnings being treated as errors /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:48: error: struct has no members make[4]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1 make[3]: *** [mod_spandsp-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Am I missing something? I was able to compile a few weeks ago. Regards From gavin.henry at gmail.com Thu Jul 5 03:00:58 2012 From: gavin.henry at gmail.com (Gavin Henry) Date: Thu, 5 Jul 2012 00:00:58 +0100 Subject: [Freeswitch-users] XMPP Presence? Message-ID: Hi all, We're running our own XMPP chat server and would like to tie this in to FreeSWITCH. Has anyone tested presence via XMPP? The wiki is very light on this and would be happy to update. Thanks. From avi at avimarcus.net Thu Jul 5 03:03:35 2012 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 5 Jul 2012 02:03:35 +0300 Subject: [Freeswitch-users] Error in compilation after git pull In-Reply-To: References: Message-ID: I think they're messing with mod_spandsp... Please post your compilation issue on http://jira.freeswitch.org to ensure it gets followed up on. Thanks, -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120705/e5034b1e/attachment.html From bob.mccarthy at experient.com Thu Jul 5 03:09:07 2012 From: bob.mccarthy at experient.com (Bob McCarthy) Date: Wed, 4 Jul 2012 17:09:07 -0600 Subject: [Freeswitch-users] Error in compilation after git pull In-Reply-To: References: Message-ID: <05e401cd5a3a$04f54bc0$0edfe340$@mccarthy@experient.com> Had the same problem -> make spandsp-reconf make install -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Neo Cheema Sent: Wednesday, July 04, 2012 4:07 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Error in compilation after git pull Hi All, I did a git pull just now. After make clean, the compile failed. The latest git commit log that I can see is: commit 04bd463d12bf4c0116bec3cee537749b2040ed40 Author: Michael S Collins Date: Wed Jul 4 12:09:32 2012 -0700 The relevant error is: making all mod_spandsp Creating mod_spandsp_la-mod_spandsp.lo quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../../../../src/include -I../../../../libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT mod_spandsp_la-mod_spandsp.lo -MD -MP -MF .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o .libs/mod_spandsp_la-mod_spandsp.o In file included from /usr/local/src/freeswitch/libs/spandsp/src/spandsp/expose.h:96, from /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:135, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid. h:33: error: expected specifier-qualifier-list before 'ademco_contactid_report_func_t' cc1: warnings being treated as errors /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid. h:48: error: struct has no members make[4]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1 make[3]: *** [mod_spandsp-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Am I missing something? I was able to compile a few weeks ago. Regards _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From gabe at gundy.org Thu Jul 5 06:31:15 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 4 Jul 2012 20:31:15 -0600 Subject: [Freeswitch-users] how to capture packets of siptrace? In-Reply-To: <1341391862.55681.YahooMailNeo@web171501.mail.ir2.yahoo.com> References: <1341391862.55681.YahooMailNeo@web171501.mail.ir2.yahoo.com> Message-ID: On Wed, Jul 4, 2012 at 2:51 AM, Jack Nikolas wrote: > how can i capture packets when user is want to register sip client or making > calls or logout or transfer, etc ? This is generally how it's done: http://wiki.freeswitch.org/wiki/Packet_Capture Gabe From fiorix at gmail.com Thu Jul 5 03:10:34 2012 From: fiorix at gmail.com (Alexandre Fiori) Date: Wed, 4 Jul 2012 19:10:34 -0400 Subject: [Freeswitch-users] Error in compilation after git pull In-Reply-To: References: Message-ID: <52EE080B-2A72-43A0-9188-937859296534@gmail.com> last night I had trouble compiling spandsp and spidermonkey on osx On 2012-07-04, at 7:03 PM, Avi Marcus wrote: > I think they're messing with mod_spandsp... > > Please post your compilation issue on http://jira.freeswitch.org to ensure it gets followed up on. > Thanks, > -Avi > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Ship, ahoy! Hast seen the White Whale? - Cap'n Ahab -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120704/26ac5e51/attachment-0001.html From robin.gilks at taitradio.com Thu Jul 5 05:18:08 2012 From: robin.gilks at taitradio.com (Robin Gilks) Date: Thu, 5 Jul 2012 13:18:08 +1200 Subject: [Freeswitch-users] Crash on incoming E1 call Message-ID: Current running from commit 04bd463d12bf4c0116bec3cee537749b2040ed40 This occurs on all incoming calls from a Digium Wildcard TE210P with a zaptel E1 interface with libpri. I've tried updating to dahdi-linux-complete-2.6.1+2.6.1.tar.gz but failed miserably - couldn't even get the E1 link itself to come up :( I've captured a backtrace of the event on the zaptel drivers, hopefully useful to someone. gdb /usr/local/freeswitch_TR/bin/freeswitch -x ${HOME}/gdbcommands $ cat gdbcommands handle SIGPIPE nostop noprint handle SIG33 nostop noprint set logging on set pagination off set breakpoint pending on run thread apply all bt full set logging off Attached is the output of the backtrace. -- Robin Gilks -- ------------------------------ This email, including any attachments, is only for the intended recipient. It is subject to copyright, is confidential and may be the subject of legal or other privilege, none of which is waived or lost by reason of this transmission. If you are not an intended recipient, you may not use, disseminate, distribute or reproduce such email, any attachments, or any part thereof. If you have received a message in error, please notify the sender immediately and erase all copies of the message and any attachments. Unfortunately, we cannot warrant that the email has not been altered or corrupted during transmission nor can we guarantee that any email or any attachments are free from computer viruses or other conditions which may damage or interfere with recipient data, hardware or software. The recipient relies upon its own procedures and assumes all risk of use and of opening any attachments. ------------------------------ -------------- next part -------------- Function "qFatal" not defined. Breakpoint 1 (qFatal) pending. [Thread debugging using libthread_db enabled] Program exited normally. [Thread debugging using libthread_db enabled] Executing new program: /usr/local/freeswitch_TR/bin/freeswitch [Thread debugging using libthread_db enabled] [New Thread 0xb7f8eb90 (LWP 3171)] [New Thread 0xb7f52b90 (LWP 3172)] [New Thread 0xb7e51b90 (LWP 3173)] [New Thread 0xb7e15b90 (LWP 3174)] [Thread 0xb7e15b90 (LWP 3174) exited] [New Thread 0xb7dd9b90 (LWP 3175)] [Thread 0xb7dd9b90 (LWP 3175) exited] [New Thread 0xb7dd9b90 (LWP 3176)] [New Thread 0xb7e15b90 (LWP 3177)] [New Thread 0xb7cd7b90 (LWP 3178)] [New Thread 0xb7c9bb90 (LWP 3179)] [Thread 0xb7c9bb90 (LWP 3179) exited] [New Thread 0xb7c9bb90 (LWP 3180)] [New Thread 0xb7acdb90 (LWP 3181)] [New Thread 0xb7a91b90 (LWP 3182)] [New Thread 0xb7a19b90 (LWP 3184)] [New Thread 0xb7a55b90 (LWP 3183)] [New Thread 0xb79ddb90 (LWP 3185)] [New Thread 0xb796fb90 (LWP 3186)] [New Thread 0xb7933b90 (LWP 3187)] [New Thread 0xb78c5b90 (LWP 3188)] [New Thread 0xb7889b90 (LWP 3189)] [New Thread 0xb781bb90 (LWP 3190)] [New Thread 0xb77dfb90 (LWP 3191)] [New Thread 0xb77a3b90 (LWP 3192)] [New Thread 0xb7735b90 (LWP 3193)] [New Thread 0xb76f9b90 (LWP 3194)] [New Thread 0xb76bdb90 (LWP 3195)] [New Thread 0xb7681b90 (LWP 3196)] [Thread 0xb76f9b90 (LWP 3194) exited] [New Thread 0xb76f9b90 (LWP 3197)] [Thread 0xb7681b90 (LWP 3196) exited] [Thread 0xb76bdb90 (LWP 3195) exited] [Thread 0xb76f9b90 (LWP 3197) exited] [New Thread 0xb76f9b90 (LWP 3198)] [New Thread 0xb7681b90 (LWP 3199)] [New Thread 0xb76bdb90 (LWP 3200)] [New Thread 0xb7645b90 (LWP 3201)] [New Thread 0xb7609b90 (LWP 3202)] [New Thread 0xb75cdb90 (LWP 3203)] [New Thread 0xb7591b90 (LWP 3204)] [Thread 0xb75cdb90 (LWP 3203) exited] [Thread 0xb7591b90 (LWP 3204) exited] [New Thread 0xb7591b90 (LWP 3207)] [Thread 0xb7591b90 (LWP 3207) exited] [New Thread 0xb7591b90 (LWP 3208)] [New Thread 0xb75cdb90 (LWP 3209)] [Thread 0xb7591b90 (LWP 3208) exited] [Thread 0xb75cdb90 (LWP 3209) exited] [New Thread 0xb75cdb90 (LWP 3210)] [New Thread 0xb7591b90 (LWP 3211)] [Thread 0xb7591b90 (LWP 3211) exited] [New Thread 0xb7591b90 (LWP 3229)] [Thread 0xb7591b90 (LWP 3229) exited] [New Thread 0xb7591b90 (LWP 3249)] [Thread 0xb7591b90 (LWP 3249) exited] [New Thread 0xb7591b90 (LWP 3250)] [New Thread 0xb7555b90 (LWP 3251)] [New Thread 0xb73ffb90 (LWP 3252)] [Thread 0xb73ffb90 (LWP 3252) exited] [Thread 0xb7555b90 (LWP 3251) exited] [Thread 0xb7591b90 (LWP 3250) exited] [New Thread 0xb7591b90 (LWP 3256)] [Thread 0xb7591b90 (LWP 3256) exited] [New Thread 0xb7591b90 (LWP 3257)] [New Thread 0xb7555b90 (LWP 3258)] [Thread 0xb7555b90 (LWP 3258) exited] [Thread 0xb7591b90 (LWP 3257) exited] [New Thread 0xb7591b90 (LWP 3259)] [Thread 0xb7591b90 (LWP 3259) exited] [New Thread 0xb7591b90 (LWP 3260)] [New Thread 0xb7555b90 (LWP 3261)] Program received signal SIGSEGV, Segmentation fault. [Switching to Thread 0xb7591b90 (LWP 3260)] 0x0061ecfe in teletone_dtmf_get (dtmf_detect_state=0xb7b24d24, buf=0xb758e4a0 "0", dur=0x50) at libs/libteletone/src/libteletone_detect.c:466 466 *dur = dtmf_detect_state->dur; Thread 51 (Thread 0xb7555b90 (LWP 3261)): #0 0x002c19f0 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/libpthread.so.0 No symbol table info available. #1 0x00637daa in apr_thread_cond_wait (cond=0x8120040, mutex=0x8120010) at locks/unix/thread_cond.c:68 rv = -4 #2 0x00571644 in switch_thread_cond_wait (cond=0x8120040, mutex=0x8120010) at src/switch_apr.c:374 No locals. #3 0x00616c6e in timer_next (timer=0xb7be2574) at src/switch_time.c:676 private_info = 0xb7be6cf8 cond_index = 30 #4 0x00588ffc in switch_core_timer_next (timer=0xf95) at src/switch_core_timer.c:74 __func__ = "switch_core_timer_next" #5 0x005ca0e8 in rtp_common_read (rtp_session=0xb7bd2320, payload_type=0xb7bce37c "", flags=0xb7bce390, io_flags=0) at src/switch_rtp.c:3193 do_cng = 0 read_pretriggered = 1 session = 0xb7bc6300 channel = 0xb7bcba38 bytes = 172 rtcp_bytes = 0 status = SWITCH_STATUS_SUCCESS poll_status = SWITCH_STATUS_SUCCESS rtcp_status = 3081764880 rtcp_poll_status = check = 0 ret = sleep_mss = 30000 poll_loop = 0 fdr = 0 rtcp_fdr = 0 hot_socket = 0 read_loops = 0 __func__ = "rtp_common_read" #6 0x005cb663 in switch_rtp_zerocopy_read_frame (rtp_session=0xb7bd2320, frame=0xb7bce354, io_flags=0) at src/switch_rtp.c:3875 bytes = #7 0x00755815 in sofia_read_frame (session=0xb7bc6300, frame=0xb7555198, flags=0, stream_id=0) at mod_sofia.c:1069 status = SWITCH_STATUS_GENERR channel = 0xb7bcba38 sanity = rtcp_frame = {report_count = 6301, packet_type = 87, ssrc = 3082605288, ntp_msw = 7452332, ntp_lsw = 3075812680, timestamp = 5745299, packet_count = 3082605288, octect_count = 7452332, nb_reports = 3075812680, reports = {{ssrc = 5870656, fraction = 88 'X', lost = 1, highest_sequence_number_received = 2274, jitter = 5864607, lsr = 0, dlsr = 0}, {ssrc = 1623859, fraction = 172 '\254', lost = 0, highest_sequence_number_received = 3082576640, jitter = 3075812856, lsr = 6270647, dlsr = 3082598968}, {ssrc = 63, fraction = 1 '\001', lost = 7435646, highest_sequence_number_received = 0, jitter = 3075814268, lsr = 1452897, dlsr = 3075812732}, {ssrc = 3075812840, fraction = 2 '\002', lost = 3082598968, highest_sequence_number_received = 7327234, jitter = 34, lsr = 0, dlsr = 0}, {ssrc = 0, fraction = 0 '\000', lost = 0, highest_sequence_number_received = 0, jitter = 0, lsr = 0, dlsr = 0}}} __PRETTY_FUNCTION__ = "sofia_read_frame" __func__ = "sofia_read_frame" #8 0x005a47bd in switch_core_session_read_frame (session=0xb7bc6300, frame=0xb7555198, flags=0, stream_id=0) at src/switch_core_io.c:191 ptr = status = SWITCH_STATUS_SUCCESS need_codec = perfect = do_bugs = 0 do_resample = 0 is_cng = 0 codec_impl = {codec_type = 3082608520, ianacode = 172 '\254', iananame = 0xb7553f38 "h?U\267P\220Y", fmtp = 0x71b6ac "\270\063\036", samples_per_second = 3082576640, actual_samples_per_second = 0, bits_per_second = -1219150024, microseconds_per_packet = 5701956, samples_per_packet = 3082608472, decoded_bytes_per_packet = 3075817308, encoded_bytes_per_packet = 3075817272, number_of_channels = 172 '\254', codec_frames_per_packet = -1212390656, init = 0x71b6ac , encode = 0xb7553f68, decode = 0x599050 , destroy = 0xb7bcdf58, codec_id = 3075817308, impl_id = 3075817320, next = 0x578440} flag = 0 i = __PRETTY_FUNCTION__ = "switch_core_session_read_frame" __func__ = "switch_core_session_read_frame" #9 0x005d04ab in audio_bridge_thread (thread=, obj=) at src/switch_ivr_bridge.c:482 status = event = 0x6f90cb data = 0xb7be7170 stream_id = 0 pre_b = 0 ans_a = 1 ans_b = 1 originator = 0 input_callback = 0 msg = {from = 0x0, message_id = SWITCH_MESSAGE_REDIRECT_AUDIO, numeric_arg = 0, string_arg = 0x0, string_arg_size = 0, pointer_arg = 0x0, pointer_arg_size = 0, numeric_reply = 0, string_reply = 0x0, string_reply_size = 0, pointer_reply = 0x0, pointer_reply_size = 0, flags = 0, _file = 0x0, _func = 0x0, _line = 0, string_array_arg = {0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0}, delivery_time = 0} user_data = 0x0 chan_a = 0xb7bcba38 chan_b = 0x80fd9d8 read_frame = 0x0 session_a = 0xb7bc6300 session_b = 0x80f82a0 read_frame_count = 3 app_name = app_arg = hook_var = silence_codec = {codec_interface = 0x0, implementation = 0x0, fmtp_in = 0x0, fmtp_out = 0x0, flags = 0, memory_pool = 0x0, private_info = 0x0, agreed_pt = 0 '\000', mutex = 0x0, next = 0x0} silence_frame = {codec = 0x0, source = 0x0, packet = 0x0, packetlen = 0, extra_data = 0x0, data = 0x0, datalen = 0, buflen = 0, samples = 0, rate = 0, payload = 0 '\000', timestamp = 0, seq = 0, ssrc = 0, m = SWITCH_FALSE, flags = 0} silence_data = {0 } silence_var = 0x0 silence_val = 0 bypass_media_after_bridge = 0 bridge_answer_timeout = sent_update = 0 answer_limit = 0 exec_app = 0x0 exec_data = 0x0 vid_thread = 0x0 vh = {session_a = 0x0, session_b = 0x0, up = 0} vid_launch = 0 __func__ = "audio_bridge_thread" __PRETTY_FUNCTION__ = "audio_bridge_thread" #10 0x005d1d91 in audio_bridge_on_exchange_media (session=0xb7bc6300) at src/switch_ivr_bridge.c:645 channel = 0xb7bcba38 bd = 0xb7be7170 state = var = __func__ = "audio_bridge_on_exchange_media" #11 0x005a0af5 in switch_core_session_run (session=0xb7bc6300) at src/switch_core_state_machine.c:443 global_proceed = proceed = 1 ptr = rstatus = state = CS_EXCHANGE_MEDIA endstate = endpoint_interface = driver_state_handler = 0x8ca6e0 application_state_handler = new_loops = 60000 __PRETTY_FUNCTION__ = "switch_core_session_run" __func__ = "switch_core_session_run" #12 0x0059e6ca in switch_core_session_thread (thread=0xb7bcf020, obj=0xb7bc6300) at src/switch_core_session.c:1412 session = 0xb7bc6300 event = event_str = 0x0 val = __func__ = "switch_core_session_thread" __PRETTY_FUNCTION__ = "switch_core_session_thread" #13 0x0063e0f6 in dummy_worker (opaque=0xb7bcf020) at threadproc/unix/thread.c:138 No locals. #14 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #15 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 50 (Thread 0xb7591b90 (LWP 3260)): #0 0x0061ecfe in teletone_dtmf_get (dtmf_detect_state=0xb7b24d24, buf=0xb758e4a0 "0", dur=0x50) at libs/libteletone/src/libteletone_detect.c:466 No locals. #1 0x009329ca in zap_channel_read (zchan=0xb7b24cc8, data=0x827310c, datalen=0xb758e5d8) at src/zap_io.c:2435 sln_buf = "\000z \372\000\212\000\212\200\340\000\067\000?\200\022\200\347\300\367\000'\000\061\300\364\000\236\000\222\200\353\000f\000z\000J\000\311\000\206\000\232\210\377\000R\000R\200\025\000\337\000\333P\374\300\017\020\375\000\337\000\337\240\006\000\065\000;\300\r\000\321\000\276\000?\023\000\063\000'\350\000\200\346\200\347\240\372\240\a`\005\210\376\220\374\020\002\340\005p\003\b\377H\376\210\377H\377\210\377H\377\210\377\310\377\310\377\310\377\310\377\310\377\310\377\370\377\370\377\370\377\070\000\370\377\070\000\070\000\070\000\070\000\070\000\070\000\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377\370\377", '\000' sln = 0xb758e160 slen = 160 digit_str = "0", '\000' status = ZAP_SUCCESS codec_func = max = 4096 i = __PRETTY_FUNCTION__ = "zap_channel_read" __func__ = "zap_channel_read" #2 0x003488a5 in channel_read_frame (session=0x80f82a0, frame=0xb7590968, flags=0, stream_id=0) at /home/taitadmin/freeswitch/src/mod/../../libs/openzap/mod_openzap/mod_openzap.c:707 channel = 0x80fd9d8 len = 160 wflags = dtmf = '\000' status = total_to = 240 chunk = 40 do_break = 0 __PRETTY_FUNCTION__ = "channel_read_frame" __func__ = "channel_read_frame" #3 0x005a47bd in switch_core_session_read_frame (session=0x80f82a0, frame=0xb7590968, flags=0, stream_id=0) at src/switch_core_io.c:191 ptr = status = SWITCH_STATUS_SUCCESS need_codec = perfect = do_bugs = 0 do_resample = 0 is_cng = 0 codec_impl = {codec_type = 135618968, ianacode = 172 '\254', iananame = 0xb758f708 "8\367X\267P\220Y", fmtp = 0x71b6ac "\270\063\036", samples_per_second = 135234208, actual_samples_per_second = 1, bits_per_second = -1218906360, microseconds_per_packet = 5701956, samples_per_packet = 135618920, decoded_bytes_per_packet = 3076060972, encoded_bytes_per_packet = 3076060936, number_of_channels = 172 '\254', codec_frames_per_packet = 135234208, init = 0x71b6ac , encode = 0xb758f738, decode = 0x599050 , destroy = 0x8156168, codec_id = 3076060972, impl_id = 3076060984, next = 0x578440} flag = 0 i = __PRETTY_FUNCTION__ = "switch_core_session_read_frame" __func__ = "switch_core_session_read_frame" #4 0x005d04ab in audio_bridge_thread (thread=, obj=) at src/switch_ivr_bridge.c:482 status = event = 0x6e8ce4 data = 0xb7be7328 stream_id = 0 pre_b = 0 ans_a = 1 ans_b = 1 originator = 1 input_callback = 0 msg = {from = 0x0, message_id = SWITCH_MESSAGE_REDIRECT_AUDIO, numeric_arg = 0, string_arg = 0x0, string_arg_size = 0, pointer_arg = 0x0, pointer_arg_size = 0, numeric_reply = 0, string_reply = 0x0, string_reply_size = 0, pointer_reply = 0x0, pointer_reply_size = 0, flags = 0, _file = 0x0, _func = 0x0, _line = 0, string_array_arg = {0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0}, delivery_time = 0} user_data = 0x0 chan_a = 0x80fd9d8 chan_b = 0xb7bcba38 read_frame = 0x0 session_a = 0x80f82a0 session_b = 0xb7bc6300 read_frame_count = 8 app_name = app_arg = hook_var = silence_codec = {codec_interface = 0x0, implementation = 0x0, fmtp_in = 0x0, fmtp_out = 0x0, flags = 0, memory_pool = 0x0, private_info = 0x0, agreed_pt = 0 '\000', mutex = 0x0, next = 0x0} silence_frame = {codec = 0x0, source = 0x0, packet = 0x0, packetlen = 0, extra_data = 0x0, data = 0x0, datalen = 0, buflen = 0, samples = 0, rate = 0, payload = 0 '\000', timestamp = 0, seq = 0, ssrc = 0, m = SWITCH_FALSE, flags = 0} silence_data = {0 } silence_var = 0x1
silence_val = 0 bypass_media_after_bridge = 0 bridge_answer_timeout = sent_update = 1 answer_limit = 0 exec_app = 0x0 exec_data = 0x0 vid_thread = 0x0 vh = {session_a = 0x0, session_b = 0x0, up = 0} vid_launch = 0 __func__ = "audio_bridge_thread" __PRETTY_FUNCTION__ = "audio_bridge_thread" #5 0x005d39b7 in switch_ivr_multi_threaded_bridge (session=0x80f82a0, peer_session=0xb7bc6300, input_callback=0, session_data=0x0, peer_session_data=0x0) at src/switch_ivr_bridge.c:1360 app = data = a_leg = 0x0 caller_channel = 0x80fd9d8 peer_channel = 0xb7bcba38 status = 25 state = event = 0x0 br = 1 var = cause = msg = {from = 0x6f8db0 "src/switch_ivr_bridge.c", message_id = SWITCH_MESSAGE_INDICATE_BRIDGE, numeric_arg = 0, string_arg = 0xb7be7468 "12532daa-c63c-11e1-86f5-af2d94b5a6fd", string_arg_size = 0, pointer_arg = 0x0, pointer_arg_size = 0, numeric_reply = 0, string_reply = 0x0, string_reply_size = 0, pointer_reply = 0x0, pointer_reply_size = 0, flags = 0, _file = 0x0, _func = 0x0, _line = 0, string_array_arg = {0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0}, delivery_time = 0} __func__ = "switch_ivr_multi_threaded_bridge" #6 0x004426c1 in audio_bridge_function (session=0x80f82a0, data=0xb7b96fb8 "sofia/gateway/fxoline3/8569") at /home/taitadmin/freeswitch/src/mod/applications/mod_dptools/mod_dptools.c:3120 channel = peer_channel = 0xb7bcba38 a_key = 0x0 ok = func = 0 b_key = 0x0 caller_channel = 0x80fd9d8 peer_session = 0xb7bc6300 transfer_on_fail = 0x0 tof_data = tof_array = {0x0, 0x0, 0x0, 0x0} v_campon = v_campon_retries = v_campon_sleep = 0x400
v_campon_timeout = 0xb7b774c0 "sofia/gateway/fxoline3/8569" v_campon_fallback_exten = 0xb7590b08 "" cause = SWITCH_CAUSE_SUCCESS campon_retries = 7452332 campon_timeout = 28 campon_sleep = fail = thread_started = -1218901128 stake = {session = 0x0, running = 0, do_xfer = 0, moh = 0x0} moh = 0x1
thread = 0x0 thd_attr = 0x0 camp_data = 0xb7b03e00 "CANCEL" status = SWITCH_STATUS_SUCCESS camp_loops = -1212718032 __func__ = "audio_bridge_function" #7 0x0059d51b in switch_core_session_exec (session=0x80f82a0, application_interface=0xb7b80dc8, arg=0x8275728 "sofia/gateway/${distributor(cisco_fxo)}/8569") at src/switch_core_session.c:2332 log = lp = event = 0x0 var = channel = 0x80fd9d8 expanded = 0xb7b96fb8 "sofia/gateway/fxoline3/8569" app = 0x4443b8 "bridge" app_uuid_var = msg = {from = 0x6f02ed "src/switch_core_session.c", message_id = SWITCH_MESSAGE_INDICATE_APPLICATION_EXEC, numeric_arg = 0, string_arg = 0x0, string_arg_size = 0, pointer_arg = 0x0, pointer_arg_size = 0, numeric_reply = 0, string_reply = 0x0, string_reply_size = 0, pointer_reply = 0x0, pointer_reply_size = 0, flags = 0, _file = 0x0, _func = 0x0, _line = 0, string_array_arg = {0x4443b8 "bridge", 0xb7b96fb8 "sofia/gateway/fxoline3/8569", 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0, 0x0}, delivery_time = 0} delim = scope = 0 uuid_str = "125325f8-c63c-11e1-86f2-af2d94b5a6fd\000{Z5\000\020\000\000\000\\F{\231\000\000\000\000@\021(\000 \000\000\000\030\000\000\000@\021(\000\370\020Y\267\001\000\000\000\033\000\000\000\002\000\001\001?\025\b \000\000\000\300\004(\000\\\031o\000 o\000\000\000\000\000\\F{\231\n\304\004\000\230\020Y\267\276\060\031\000\020\215\024\000\364\377'\000\230.\037\b\364\377'\000@\021(\000\000\000\000\000\270\020Y\267\276\060\031\000\020\215\024\000\254\266q\000\310\020Y\267\254\266q\000\n\000\000\000\301\204n\000\350\020Y\267\221\342d\000\000\000\000\000\301\204n\000e\000\000\000\254\266q\000\020?\267\000\000\000\000\370\020Y\267+Ed\000`\177\267\267 W'\b\a\000\000\000\305\037W\000\020\215\024\000\254\266q\000("... app_uuid = 0xb7591003 "125325f8-c63c-11e1-86f2-af2d94b5a6fd" __PRETTY_FUNCTION__ = "switch_core_session_exec" __func__ = "switch_core_session_exec" #8 0x0059dc7e in switch_core_session_execute_application_get_flags (session=0x80f82a0, app=0x8275720 "bridge", arg=, flags=0x0) at src/switch_core_session.c:2207 application_interface = 0xb7b80dc8 status = SWITCH_STATUS_SUCCESS __func__ = "switch_core_session_execute_application_get_flags" #9 0x005a147c in switch_core_session_run (session=0x80f82a0) at src/switch_core_state_machine.c:217 global_proceed = proceed = ptr = rstatus = state = endstate = endpoint_interface = driver_state_handler = 0x356360 application_state_handler = new_loops = 60000 __PRETTY_FUNCTION__ = "switch_core_session_run" __func__ = "switch_core_session_run" #10 0x0059e6ca in switch_core_session_thread (thread=0x82754b8, obj=0x80f82a0) at src/switch_core_session.c:1412 session = 0x80f82a0 event = event_str = 0x0 val = __func__ = "switch_core_session_thread" __PRETTY_FUNCTION__ = "switch_core_session_thread" #11 0x0063e0f6 in dummy_worker (opaque=0x82754b8) at threadproc/unix/thread.c:138 No locals. #12 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #13 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 39 (Thread 0xb75cdb90 (LWP 3210)): #0 0x001f34b7 in select () from /lib/libc.so.6 No symbol table info available. #1 0x0063fc79 in apr_sleep (t=100000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 58000} #2 0x0061416f in do_sleep (t=100000) at src/switch_time.c:171 ts = {tv_sec = -1218653288, tv_nsec = 0} #3 0x0047798a in vm_event_thread_run (thread=0x822bd28, obj=0x0) at /home/taitadmin/freeswitch/src/mod/applications/mod_voicemail/mod_voicemail.c:3886 pop = 0xb7b78a20 __func__ = "vm_event_thread_run" #4 0x0063e0f6 in dummy_worker (opaque=0x822bd28) at threadproc/unix/thread.c:138 No locals. #5 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #6 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 33 (Thread 0xb7609b90 (LWP 3202)): #0 0x002c4781 in read () from /lib/libpthread.so.0 No symbol table info available. #1 0x006b6cd3 in read_char (el=0xb7b971d8, cp=0xb760934b "\n\254\266q") at read.c:294 num_read = -1212583088 tried = 0 #2 0x006b67bc in el_getc (el=0xb7b971d8, cp=0xb760934b "\n\254\266q") at read.c:362 num_read = -1218407605 ma = 0xb7b97478 #3 0x006b68ff in read_getcmd (el=0xb7b971d8, nread=0xb7609398) at read.c:241 No locals. #4 el_gets (el=0xb7b971d8, nread=0xb7609398) at read.c:497 cmdnum = 1 '\001' num = 1 ch = 10 '\n' #5 0x005845ab in console_thread (thread=0xb7bb2a40, obj=0xb7bb28e8) at src/switch_console.c:1026 arg = 1 count = 1 line = pool = 0xb7bb28e8 __func__ = "console_thread" __PRETTY_FUNCTION__ = "console_thread" #6 0x0063e0f6 in dummy_worker (opaque=0xb7bb2a40) at threadproc/unix/thread.c:138 No locals. #7 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #8 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 32 (Thread 0xb7645b90 (LWP 3201)): #0 0x002c19f0 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/libpthread.so.0 No symbol table info available. #1 0x00637daa in apr_thread_cond_wait (cond=0xb7ba77a8, mutex=0xb7ba7778) at locks/unix/thread_cond.c:68 rv = -4 #2 0x0062e99c in apr_queue_pop (queue=0xb7ba7748, data=0xb7645398) at misc/apr_queue.c:276 rv = 0 #3 0x005701f4 in switch_queue_pop (queue=0xb7ba7748, data=0xb7645398) at src/switch_apr.c:1040 No locals. #4 0x005b8649 in chat_thread_run (thread=0xb7bae738, obj=0xb7ba7748) at src/switch_loadable_module.c:625 pop = 0x5b85cb __func__ = "chat_thread_run" #5 0x0063e0f6 in dummy_worker (opaque=0xb7bae738) at threadproc/unix/thread.c:138 No locals. #6 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #7 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 31 (Thread 0xb76bdb90 (LWP 3200)): #0 0x002c19f0 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/libpthread.so.0 No symbol table info available. #1 0x00637daa in apr_thread_cond_wait (cond=0x811f000, mutex=0x811efd0) at locks/unix/thread_cond.c:68 rv = -4 #2 0x0062e99c in apr_queue_pop (queue=0x811efa0, data=0xb76bd398) at misc/apr_queue.c:276 rv = 0 #3 0x005701f4 in switch_queue_pop (queue=0x811efa0, data=0xb76bd398) at src/switch_apr.c:1040 No locals. #4 0x005b8649 in chat_thread_run (thread=0xb7ba7728, obj=0x811efa0) at src/switch_loadable_module.c:625 pop = 0x5b85cb __func__ = "chat_thread_run" #5 0x0063e0f6 in dummy_worker (opaque=0xb7ba7728) at threadproc/unix/thread.c:138 No locals. #6 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #7 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 30 (Thread 0xb7681b90 (LWP 3199)): #0 0x002c499e in accept () from /lib/libpthread.so.0 No symbol table info available. #1 0x0063ce7d in apr_socket_accept (new=0xb768133c, sock=0x8291220, connection_context=0x82930a0) at network_io/unix/sockets.c:187 No locals. #2 0x00570c1b in switch_socket_accept (new_sock=0xb768133c, sock=0x8291220, pool=0x82930a0) at src/switch_apr.c:700 No locals. #3 0x0032c65f in mod_event_socket_runtime () at /home/taitadmin/freeswitch/src/mod/event_handlers/mod_event_socket/mod_event_socket.c:2829 pool = 0x8291098 listener_pool = 0x82930a0 rv = sa = 0x8291168 inbound_socket = 0x8293170 listener = 0x0 x = errs = 0 __func__ = "mod_event_socket_runtime" #4 0x005b6d8a in switch_loadable_module_exec (thread=0x811ef50, obj=0x811ed40) at src/switch_loadable_module.c:98 status = 4294966784 module = 0xb7b0ac08 __PRETTY_FUNCTION__ = "switch_loadable_module_exec" __func__ = "switch_loadable_module_exec" #5 0x0063e0f6 in dummy_worker (opaque=0x811ef50) at threadproc/unix/thread.c:138 No locals. #6 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #7 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 29 (Thread 0xb76f9b90 (LWP 3198)): #0 0x001f34b7 in select () from /lib/libc.so.6 No symbol table info available. #1 0x0063fc79 in apr_sleep (t=9562) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 9000} #2 0x0061416f in do_sleep (t=9562) at src/switch_time.c:171 ts = {tv_sec = 822547974, tv_nsec = -1217424712} #3 0x0061674c in softtimer_runtime () at src/switch_time.c:939 timediff = 4294966782 too_late = 20000000 current_ms = 1490 x = tick = 29 ts = last = 1341449712812517 fwd_errs = 0 rev_errs = 0 time_sync = 4 __func__ = "softtimer_runtime" #4 0x005b6d8a in switch_loadable_module_exec (thread=0x811ece8, obj=0x811ea48) at src/switch_loadable_module.c:98 status = 4294966782 module = 0x811ff30 __PRETTY_FUNCTION__ = "switch_loadable_module_exec" __func__ = "switch_loadable_module_exec" #5 0x0063e0f6 in dummy_worker (opaque=0x811ece8) at threadproc/unix/thread.c:138 No locals. #6 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #7 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 24 (Thread 0xb7735b90 (LWP 3193)): #0 0x002c19f0 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/libpthread.so.0 No symbol table info available. #1 0x00637daa in apr_thread_cond_wait (cond=0x82400f8, mutex=0x82400c8) at locks/unix/thread_cond.c:68 rv = -4 #2 0x00571644 in switch_thread_cond_wait (cond=0x82400f8, mutex=0x82400c8) at src/switch_apr.c:374 No locals. #3 0x00ed1752 in timer_thread_run (thread=0x8240208, obj=0x0) at mod_spandsp_fax.c:207 timer = {interval = 20, flags = 1, samples = 160, samplecount = 160, timer_interface = 0x811fe98, memory_pool = 0x82477e8, private_info = 0x8247878, diff = 0, tick = 0} pvt = 0x0 __func__ = "timer_thread_run" #4 0x0063e0f6 in dummy_worker (opaque=0x8240208) at threadproc/unix/thread.c:138 No locals. #5 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #6 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 23 (Thread 0xb77a3b90 (LWP 3192)): #0 0x001f34b7 in select () from /lib/libc.so.6 No symbol table info available. #1 0x0063fc79 in apr_sleep (t=1000000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 590000} #2 0x0061416f in do_sleep (t=1000000) at src/switch_time.c:171 ts = {tv_sec = 7452332, tv_nsec = -1216728328} #3 0x0044ef68 in node_thread_run (thread=0x820b5e8, obj=0x820aef0) at /home/taitadmin/freeswitch/src/mod/applications/mod_fifo/mod_fifo.c:1901 ppl_waiting = 0 consumer_total = 0 idle_consumers = 0 node = 0x0 last = 0x820efb8 cur_priority = 1 __func__ = "node_thread_run" #4 0x0063e0f6 in dummy_worker (opaque=0x820b5e8) at threadproc/unix/thread.c:138 No locals. #5 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #6 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 22 (Thread 0xb77dfb90 (LWP 3191)): #0 0x001f34b7 in select () from /lib/libc.so.6 No symbol table info available. #1 0x0063fc79 in apr_sleep (t=100000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 93000} #2 0x0061416f in do_sleep (t=100000) at src/switch_time.c:171 ts = {tv_sec = -1216482408, tv_nsec = 0} #3 0x007ab2c1 in sofia_presence_event_thread_run (thread=0x81352c8, obj=0x0) at sofia_presence.c:1395 count = 0 pop = 0xb7b8cee0 __func__ = "sofia_presence_event_thread_run" #4 0x0063e0f6 in dummy_worker (opaque=0x81352c8) at threadproc/unix/thread.c:138 No locals. #5 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #6 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 21 (Thread 0xb781bb90 (LWP 3190)): #0 0x001f34b7 in select () from /lib/libc.so.6 No symbol table info available. #1 0x00382a41 in lpwrap_one_loop (spri=0x812d098) at src/ozmod/ozmod_libpri/lpwrap_pri.c:233 rfds = {fds_bits = {0, 0, 2048, 0 }} efds = {fds_bits = {0, 0, 2048, 0 }} now = {tv_sec = 0, tv_usec = 70000} next = event = handler = sel = __func__ = "lpwrap_one_loop" #2 0x00382b6f in lpwrap_run_pri (spri=0x812d098) at src/ozmod/ozmod_libpri/lpwrap_pri.c:274 ret = 0 __func__ = "lpwrap_run_pri" #3 0x003812e1 in zap_libpri_run (me=0x81299c0, obj=0xb7b208e0) at src/ozmod/ozmod_libpri/ozmod_libpri.c:1059 isdn_data = 0x812d050 i = 31 x = 1 down = 0 got_d = 1 __func__ = "zap_libpri_run" #4 0x00937b4a in thread_launch (args=0x81299c0) at src/zap_threadmutex.c:90 exit_val = #5 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #6 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 20 (Thread 0xb7889b90 (LWP 3189)): #0 0x002c1d08 in pthread_cond_timedwait@@GLIBC_2.3.2 () from /lib/libpthread.so.0 No symbol table info available. #1 0x00637d59 in apr_thread_cond_timedwait (cond=0x8151490, mutex=0x8151460, timeout=1000000) at locks/unix/thread_cond.c:89 rv = then = 1341449713192582 abstime = {tv_sec = 1341449713, tv_nsec = 192582000} #2 0x0062e896 in apr_queue_pop_timeout (queue=0x8151430, data=0xb7889394, timeout=1000000) at misc/apr_queue.c:339 rv = 0 #3 0x005701c2 in switch_queue_pop_timeout (queue=0x8151430, data=0xb7889394, timeout=4294967292) at src/switch_apr.c:1045 No locals. #4 0x007716ca in sofia_profile_worker_thread_run (thread=0x8151410, obj=0x8150588) at sofia.c:1795 sleepy_time = ireg_loops = 2 gateway_loops = 0 pop = 0x81f1d58 sql_len = 32768 sqlbuf = 0x816fbe8 "delete from sip_subscriptions where call_id='72a10054-40de-1230-1f82-bc305be6cb52';\n" sql = 0x0 last_commit = 1341449712182571 last_check = 1341449712182571 len = 0 statements = 0 __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" __func__ = "sofia_profile_worker_thread_run" #5 0x0063e0f6 in dummy_worker (opaque=0x8151410) at threadproc/unix/thread.c:138 No locals. #6 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #7 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 19 (Thread 0xb78c5b90 (LWP 3188)): #0 0x001fb17c in epoll_wait () from /lib/libc.so.6 No symbol table info available. #1 0x0085ca30 in su_epoll_port_wait_events (self=0x81646f8, tout=1000) at su_epoll_port.c:495 j = n = 135719856 events = index = version = 3 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #2 0x0086257d in su_base_port_run (self=0x81646f8) at su_base_port.c:349 tout = 1000 tout2 = 0 __PRETTY_FUNCTION__ = "su_base_port_run" #3 0x00869189 in su_port_run (self=0x8161bb8) at su_port.h:326 No locals. #4 su_root_run (self=0x8161bb8) at su_root.c:819 __PRETTY_FUNCTION__ = "su_root_run" #5 0x0085f257 in su_pthread_port_clone_main (varg=0xb7a54f9c) at su_pthread_port.c:334 arg = task = {{sut_port = 0x81646f8, sut_root = 0x8161bb8}} zap = 0 #6 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #7 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 18 (Thread 0xb7933b90 (LWP 3187)): #0 0x002c1d08 in pthread_cond_timedwait@@GLIBC_2.3.2 () from /lib/libpthread.so.0 No symbol table info available. #1 0x00637d59 in apr_thread_cond_timedwait (cond=0x813d440, mutex=0x813d410, timeout=1000000) at locks/unix/thread_cond.c:89 rv = then = 1341449713539554 abstime = {tv_sec = 1341449713, tv_nsec = 539554000} #2 0x0062e896 in apr_queue_pop_timeout (queue=0x813d3e0, data=0xb7933394, timeout=1000000) at misc/apr_queue.c:339 rv = 0 #3 0x005701c2 in switch_queue_pop_timeout (queue=0x813d3e0, data=0xb7933394, timeout=4294967292) at src/switch_apr.c:1045 No locals. #4 0x007716ca in sofia_profile_worker_thread_run (thread=0x813d3c0, obj=0x813c5d0) at sofia.c:1795 sleepy_time = ireg_loops = 0 gateway_loops = 0 pop = 0x0 sql_len = 32768 sqlbuf = 0x8165108 "" sql = 0x0 last_commit = 1341449711532619 last_check = 1341449711532619 len = 0 statements = 0 __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" __func__ = "sofia_profile_worker_thread_run" #5 0x0063e0f6 in dummy_worker (opaque=0x813d3c0) at threadproc/unix/thread.c:138 No locals. #6 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #7 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 17 (Thread 0xb796fb90 (LWP 3186)): #0 0x001fb17c in epoll_wait () from /lib/libc.so.6 No symbol table info available. #1 0x0085ca30 in su_epoll_port_wait_events (self=0x814ffe0, tout=1000) at su_epoll_port.c:495 j = n = 135590504 events = index = version = 3 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #2 0x0086257d in su_base_port_run (self=0x814ffe0) at su_base_port.c:349 tout = 1000 tout2 = 0 __PRETTY_FUNCTION__ = "su_base_port_run" #3 0x00869189 in su_port_run (self=0x812de38) at su_port.h:326 No locals. #4 su_root_run (self=0x812de38) at su_root.c:819 __PRETTY_FUNCTION__ = "su_root_run" #5 0x0085f257 in su_pthread_port_clone_main (varg=0xb7a90f9c) at su_pthread_port.c:334 arg = task = {{sut_port = 0x814ffe0, sut_root = 0x812de38}} zap = 0 #6 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #7 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 16 (Thread 0xb79ddb90 (LWP 3185)): #0 0x002c1d08 in pthread_cond_timedwait@@GLIBC_2.3.2 () from /lib/libpthread.so.0 No symbol table info available. #1 0x00637d59 in apr_thread_cond_timedwait (cond=0x8139c18, mutex=0x8139be8, timeout=1000000) at locks/unix/thread_cond.c:89 rv = then = 1341449713533559 abstime = {tv_sec = 1341449713, tv_nsec = 533559000} #2 0x0062e896 in apr_queue_pop_timeout (queue=0x8139bb8, data=0xb79dd394, timeout=1000000) at misc/apr_queue.c:339 rv = 0 #3 0x005701c2 in switch_queue_pop_timeout (queue=0x8139bb8, data=0xb79dd394, timeout=4294967292) at src/switch_apr.c:1045 No locals. #4 0x007716ca in sofia_profile_worker_thread_run (thread=0x8139b98, obj=0x8138490) at sofia.c:1795 sleepy_time = ireg_loops = 29 gateway_loops = 0 pop = 0x0 sql_len = 32768 sqlbuf = 0x81594f0 "" sql = 0x0 last_commit = 1341449712532555 last_check = 1341449712532555 len = 0 statements = 0 __PRETTY_FUNCTION__ = "sofia_profile_worker_thread_run" __func__ = "sofia_profile_worker_thread_run" #5 0x0063e0f6 in dummy_worker (opaque=0x8139b98) at threadproc/unix/thread.c:138 No locals. #6 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #7 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 15 (Thread 0xb7a55b90 (LWP 3183)): #0 0x001fb17c in epoll_wait () from /lib/libc.so.6 No symbol table info available. #1 0x0085ca30 in su_epoll_port_wait_events (self=0xb7b0e920, tout=1000) at su_epoll_port.c:495 j = n = 0 events = index = version = 1 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #2 0x00862374 in su_base_port_step (self=0xb7b0e920, tout=1000) at su_base_port.c:467 now = __PRETTY_FUNCTION__ = "su_base_port_step" #3 0x00869540 in su_port_step (self=0xb7b0ed78, tout=1000) at su_port.h:340 No locals. #4 su_root_step (self=0xb7b0ed78, tout=1000) at su_root.c:858 __PRETTY_FUNCTION__ = "su_root_step" #5 0x0077bcb1 in sofia_profile_thread_run (thread=0x8151320, obj=0x8150588) at sofia.c:2223 profile = node = 0x89bb91 s_event = 0x0 sanity = worker_thread = 0x8151410 st = SWITCH_STATUS_SUCCESS __func__ = "sofia_profile_thread_run" __PRETTY_FUNCTION__ = "sofia_profile_thread_run" #6 0x0063e0f6 in dummy_worker (opaque=0x8151320) at threadproc/unix/thread.c:138 No locals. #7 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #8 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 14 (Thread 0xb7a19b90 (LWP 3184)): #0 0x001fb17c in epoll_wait () from /lib/libc.so.6 No symbol table info available. #1 0x0085ca30 in su_epoll_port_wait_events (self=0x8150068, tout=1000) at su_epoll_port.c:495 j = n = 135625752 events = index = version = 3 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #2 0x0086257d in su_base_port_run (self=0x8150068) at su_base_port.c:349 tout = 1000 tout2 = 0 __PRETTY_FUNCTION__ = "su_base_port_run" #3 0x00869189 in su_port_run (self=0x81566b0) at su_port.h:326 No locals. #4 su_root_run (self=0x81566b0) at su_root.c:819 __PRETTY_FUNCTION__ = "su_root_run" #5 0x0085f257 in su_pthread_port_clone_main (varg=0xb7accf9c) at su_pthread_port.c:334 arg = task = {{sut_port = 0x8150068, sut_root = 0x81566b0}} zap = 0 #6 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #7 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 13 (Thread 0xb7a91b90 (LWP 3182)): #0 0x001fb17c in epoll_wait () from /lib/libc.so.6 No symbol table info available. #1 0x0085ca30 in su_epoll_port_wait_events (self=0x8142bb8, tout=1000) at su_epoll_port.c:495 j = n = 0 events = index = version = 1 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #2 0x00862374 in su_base_port_step (self=0x8142bb8, tout=1000) at su_base_port.c:467 now = __PRETTY_FUNCTION__ = "su_base_port_step" #3 0x00869540 in su_port_step (self=0x8145f58, tout=1000) at su_port.h:340 No locals. #4 su_root_step (self=0x8145f58, tout=1000) at su_root.c:858 __PRETTY_FUNCTION__ = "su_root_step" #5 0x0077bcb1 in sofia_profile_thread_run (thread=0x813d2e0, obj=0x813c5d0) at sofia.c:2223 profile = node = 0x89bb91 s_event = 0x0 sanity = worker_thread = 0x813d3c0 st = SWITCH_STATUS_SUCCESS __func__ = "sofia_profile_thread_run" __PRETTY_FUNCTION__ = "sofia_profile_thread_run" #6 0x0063e0f6 in dummy_worker (opaque=0x813d2e0) at threadproc/unix/thread.c:138 No locals. #7 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #8 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 12 (Thread 0xb7acdb90 (LWP 3181)): #0 0x001fb17c in epoll_wait () from /lib/libc.so.6 No symbol table info available. #1 0x0085ca30 in su_epoll_port_wait_events (self=0x813c408, tout=1000) at su_epoll_port.c:495 j = n = 0 events = index = version = 1 __PRETTY_FUNCTION__ = "su_epoll_port_wait_events" #2 0x00862374 in su_base_port_step (self=0x813c408, tout=1000) at su_base_port.c:467 now = __PRETTY_FUNCTION__ = "su_base_port_step" #3 0x00869540 in su_port_step (self=0x812d818, tout=1000) at su_port.h:340 No locals. #4 su_root_step (self=0x812d818, tout=1000) at su_root.c:858 __PRETTY_FUNCTION__ = "su_root_step" #5 0x0077bcb1 in sofia_profile_thread_run (thread=0x8139128, obj=0x8138490) at sofia.c:2223 profile = node = 0x89bb91 s_event = 0x0 sanity = worker_thread = 0x8139b98 st = SWITCH_STATUS_SUCCESS __func__ = "sofia_profile_thread_run" __PRETTY_FUNCTION__ = "sofia_profile_thread_run" #6 0x0063e0f6 in dummy_worker (opaque=0x8139128) at threadproc/unix/thread.c:138 No locals. #7 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #8 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 11 (Thread 0xb7c9bb90 (LWP 3180)): #0 0x002c19f0 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/libpthread.so.0 No symbol table info available. #1 0x00637daa in apr_thread_cond_wait (cond=0xb7afedd0, mutex=0xb7afeda0) at locks/unix/thread_cond.c:68 rv = -4 #2 0x0062e99c in apr_queue_pop (queue=0xb7afed70, data=0xb7c9b398) at misc/apr_queue.c:276 rv = 0 #3 0x005701f4 in switch_queue_pop (queue=0xb7afed70, data=0xb7c9b398) at src/switch_apr.c:1040 No locals. #4 0x0078a04f in sofia_msg_thread_run (thread=0x8134bd0, obj=0xb7afed70) at sofia.c:1386 pop = 0xb7b8b538 my_id = __func__ = "sofia_msg_thread_run" #5 0x0063e0f6 in dummy_worker (opaque=0x8134bd0) at threadproc/unix/thread.c:138 No locals. #6 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #7 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 9 (Thread 0xb7cd7b90 (LWP 3178)): #0 0x001f34b7 in select () from /lib/libc.so.6 No symbol table info available. #1 0x0063fc79 in apr_sleep (t=500000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 327000} #2 0x0061416f in do_sleep (t=500000) at src/switch_time.c:171 ts = {tv_sec = 135612016, tv_nsec = 5959643} #3 0x005af344 in switch_scheduler_task_thread (thread=0x8117e70, obj=0x0) at src/switch_scheduler.c:171 __func__ = "switch_scheduler_task_thread" #4 0x0063e0f6 in dummy_worker (opaque=0x8117e70) at threadproc/unix/thread.c:138 No locals. #5 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #6 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 8 (Thread 0xb7e15b90 (LWP 3177)): #0 0x001f34b7 in select () from /lib/libc.so.6 No symbol table info available. #1 0x0063fc79 in apr_sleep (t=1000000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 824000} #2 0x0061416f in do_sleep (t=1000000) at src/switch_time.c:171 ts = {tv_sec = 136186440, tv_nsec = 7474448} #3 0x0059130f in switch_core_sql_db_thread (thread=0xb7d39b08, obj=0x0) at src/switch_core_sqldb.c:979 sec = 10 reg_sec = 10 #4 0x0063e0f6 in dummy_worker (opaque=0xb7d39b08) at threadproc/unix/thread.c:138 No locals. #5 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #6 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 7 (Thread 0xb7dd9b90 (LWP 3176)): #0 0x001f34b7 in select () from /lib/libc.so.6 No symbol table info available. #1 0x0063fc79 in apr_sleep (t=200000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 60000} #2 0x0061416f in do_sleep (t=200000) at src/switch_time.c:171 ts = {tv_sec = 137175627, tv_nsec = 0} #3 0x005972c5 in switch_core_sql_thread (thread=0xb7d39ae8, obj=0x0) at src/switch_core_sqldb.c:1133 pop = 0x0 iterations = 3 trans = 1 '\001' len = 517 sql_len = 105306 sqlbuf = 0x82d20a0 "" sql = save_sql = 0x0 lc = wrote = 1 do_sleep = 1 auto_pause = 0 __PRETTY_FUNCTION__ = "switch_core_sql_thread" __func__ = "switch_core_sql_thread" #4 0x0063e0f6 in dummy_worker (opaque=0xb7d39ae8) at threadproc/unix/thread.c:138 No locals. #5 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #6 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 4 (Thread 0xb7e51b90 (LWP 3173)): #0 0x002c19f0 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/libpthread.so.0 No symbol table info available. #1 0x00637daa in apr_thread_cond_wait (cond=0x8065640, mutex=0x8065610) at locks/unix/thread_cond.c:68 rv = -4 #2 0x0062e99c in apr_queue_pop (queue=0x80655e0, data=0xb7e51398) at misc/apr_queue.c:276 rv = 0 #3 0x005701f4 in switch_queue_pop (queue=0x80655e0, data=0xb7e51398) at src/switch_apr.c:1040 No locals. #4 0x006088e0 in log_thread (t=0xb7eb3ae0, obj=0x0) at src/switch_log.c:294 pop = 0x0 node = 0x0 binding = 0x0 __func__ = "log_thread" #5 0x0063e0f6 in dummy_worker (opaque=0xb7eb3ae0) at threadproc/unix/thread.c:138 No locals. #6 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #7 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 3 (Thread 0xb7f52b90 (LWP 3172)): #0 0x002c19f0 in pthread_cond_wait@@GLIBC_2.3.2 () from /lib/libpthread.so.0 No symbol table info available. #1 0x00637daa in apr_thread_cond_wait (cond=0x8060258, mutex=0x8060228) at locks/unix/thread_cond.c:68 rv = -4 #2 0x0062e99c in apr_queue_pop (queue=0x80601f8, data=0xb7f52398) at misc/apr_queue.c:276 rv = 0 #3 0x005701f4 in switch_queue_pop (queue=0x80601f8, data=0xb7f52398) at src/switch_apr.c:1040 No locals. #4 0x005c058f in switch_event_dispatch_thread (thread=0x8065440, obj=0x80601f8) at src/switch_event.c:264 pop = 0x0 event = 0x0 my_id = 0 __func__ = "switch_event_dispatch_thread" #5 0x0063e0f6 in dummy_worker (opaque=0x8065440) at threadproc/unix/thread.c:138 No locals. #6 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #7 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 2 (Thread 0xb7f8eb90 (LWP 3171)): #0 0x001f34b7 in select () from /lib/libc.so.6 No symbol table info available. #1 0x0063fc79 in apr_sleep (t=1000000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 411000} #2 0x0061416f in do_sleep (t=1000000) at src/switch_time.c:171 ts = {tv_sec = 0, tv_nsec = -1208425608} #3 0x0058bfc6 in pool_thread (thread=0xb7fbfe38, obj=0x0) at src/switch_core_memory.c:555 No locals. #4 0x0063e0f6 in dummy_worker (opaque=0xb7fbfe38) at threadproc/unix/thread.c:138 No locals. #5 0x002bd6e1 in start_thread () from /lib/libpthread.so.0 No symbol table info available. #6 0x001faaee in clone () from /lib/libc.so.6 No symbol table info available. Thread 1 (Thread 0xb7ff39c0 (LWP 3168)): #0 0x001f34b7 in select () from /lib/libc.so.6 No symbol table info available. #1 0x0063fc79 in apr_sleep (t=1000000) at time/unix/time.c:246 tv = {tv_sec = 0, tv_usec = 637000} #2 0x0061416f in do_sleep (t=1000000) at src/switch_time.c:171 ts = {tv_sec = 245760, tv_nsec = -1213202416} #3 0x00582cc7 in switch_console_loop () at src/switch_console.c:1144 arg = 1 thread = 0xb7bb2a40 thd_attr = 0xb7bb2978 pool = 0xb7bb28e8 __func__ = "switch_console_loop" __PRETTY_FUNCTION__ = "switch_console_loop" #4 0x005a94c8 in switch_core_runtime_loop (bg=0) at src/switch_core.c:870 No locals. #5 0x0804acb9 in main (argc=1, argv=0xbfffeb24) at src/switch.c:936 pid_path = "/usr/local/freeswitch_TR/run/freeswitch.pid", '\000' pid_buffer = "3168", '\000' old_pid_buffer = '\000' pid_len = 4 old_pid_len = 1385573 err = 0x6f7fcc "Success" nf = 0 runas_user = 0x0 runas_group = 0x0 nc = 0 pid = x = opts = opts_str = '\000' local_argv = {0xbfffec24 "/usr/local/freeswitch_TR/bin/freeswitch", 0x0 } local_argc = 1 arg_argv = {0x0 } alt_dirs = 0 log_set = 0 run_set = 0 do_kill = 0 known_opt = 0 priority = 0 do_wait = 0 flags = 68225 ret = destroy_status = fd = 0x8054548 pool = 0x8054508 rlp = {rlim_cur = 245760, rlim_max = 245760} waste = 0 From anton.jugatsu at gmail.com Thu Jul 5 09:30:56 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Thu, 5 Jul 2012 09:30:56 +0400 Subject: [Freeswitch-users] Crash on incoming E1 call In-Reply-To: References: Message-ID: Post this output to backtracker. 2012/7/5 Robin Gilks > Current running from commit 04bd463d12bf4c0116bec3cee537749b2040ed40 > > This occurs on all incoming calls from a Digium Wildcard TE210P with a > zaptel E1 interface with libpri. > > I've tried updating to dahdi-linux-complete-2.6.1+2.6.1.tar.gz but > failed miserably - couldn't even get the E1 link itself to come up :( > > I've captured a backtrace of the event on the zaptel drivers, > hopefully useful to someone. > > gdb /usr/local/freeswitch_TR/bin/freeswitch -x ${HOME}/gdbcommands > > > $ cat gdbcommands > handle SIGPIPE nostop noprint > handle SIG33 nostop noprint > set logging on > set pagination off > set breakpoint pending on > run > thread apply all bt full > set logging off > > Attached is the output of the backtrace. > > > > -- > Robin Gilks > > -- > > ------------------------------ > This email, including any attachments, is only for the intended recipient. > It is subject to copyright, is confidential and may be the subject of legal > or other privilege, none of which is waived or lost by reason of this > transmission. > If you are not an intended recipient, you may not use, disseminate, > distribute or reproduce such email, any attachments, or any part thereof. > If you have received a message in error, please notify the sender > immediately and erase all copies of the message and any attachments. > Unfortunately, we cannot warrant that the email has not been altered or > corrupted during transmission nor can we guarantee that any email or any > attachments are free from computer viruses or other conditions which may > damage or interfere with recipient data, hardware or software. The > recipient relies upon its own procedures and assumes all risk of use and of > opening any attachments. > ------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120705/51aa66c0/attachment.html From neo.cheema at gmail.com Thu Jul 5 09:52:38 2012 From: neo.cheema at gmail.com (Neo Cheema) Date: Thu, 5 Jul 2012 11:22:38 +0530 Subject: [Freeswitch-users] Crash on incoming E1 call In-Reply-To: References: Message-ID: Facing exactly the same issue. But in my case, freeswitch crashes at startup itself while configuring spandsp. My stack is Feetdm E1 with libpri. On Thu, Jul 5, 2012 at 11:00 AM, Anton Kvashenkin wrote: > Post this output to backtracker. > > 2012/7/5 Robin Gilks >> >> Current running from commit 04bd463d12bf4c0116bec3cee537749b2040ed40 >> >> This occurs on all incoming calls from a Digium Wildcard TE210P with a >> zaptel E1 interface with libpri. >> >> I've tried updating to dahdi-linux-complete-2.6.1+2.6.1.tar.gz but >> failed miserably - couldn't even get the E1 link itself to come up :( >> >> I've captured a backtrace of the event on the zaptel drivers, >> hopefully useful to someone. >> >> gdb /usr/local/freeswitch_TR/bin/freeswitch -x ${HOME}/gdbcommands >> >> >> $ cat gdbcommands >> handle SIGPIPE nostop noprint >> handle SIG33 nostop noprint >> set logging on >> set pagination off >> set breakpoint pending on >> run >> thread apply all bt full >> set logging off >> >> Attached is the output of the backtrace. >> >> >> >> -- >> Robin Gilks >> >> -- >> >> ------------------------------ >> This email, including any attachments, is only for the intended recipient. >> It is subject to copyright, is confidential and may be the subject of >> legal >> or other privilege, none of which is waived or lost by reason of this >> transmission. >> If you are not an intended recipient, you may not use, disseminate, >> distribute or reproduce such email, any attachments, or any part thereof. >> If you have received a message in error, please notify the sender >> immediately and erase all copies of the message and any attachments. >> Unfortunately, we cannot warrant that the email has not been altered or >> corrupted during transmission nor can we guarantee that any email or any >> attachments are free from computer viruses or other conditions which may >> damage or interfere with recipient data, hardware or software. The >> recipient relies upon its own procedures and assumes all risk of use and >> of >> opening any attachments. >> ------------------------------ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From avi at avimarcus.net Thu Jul 5 11:08:24 2012 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 5 Jul 2012 10:08:24 +0300 Subject: [Freeswitch-users] Link to JIRA from the RSS feed in google reader? (greasemonkey help) Message-ID: Hi - I try to follow the commits to GIT - I use the RSS feed jira provides and follow in google reader. it's at http://fisheye.freeswitch.org/changelog/freeswitch.git?view=all&max=30&RSS=true Anyway - I try to see the actual issue to get context, but for example: FS-4381 isn't clickable... Can someone whip together a quick greasemonkey script (or patch jira!) to replace (FS-\d{4}) with a link to http://jira.freeswitch.org/browse/$1 I tried for about a half hour but couldn't even replace "FS" within a google reader window.... no clue what I'm doing. Thanks! -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120705/1482da9f/attachment.html From peter.olsson at visionutveckling.se Thu Jul 5 11:08:57 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 5 Jul 2012 07:08:57 +0000 Subject: [Freeswitch-users] Crash on incoming E1 call Message-ID: <1FFF97C269757C458224B7C895F35F1512E8EE@cantor.std.visionutv.se> Please report to Jira, with all needed information http://wiki.freeswitch.org/wiki/Reporting_Bugs /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Neo Cheema Skickat: den 5 juli 2012 07:53 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Crash on incoming E1 call Facing exactly the same issue. But in my case, freeswitch crashes at startup itself while configuring spandsp. My stack is Feetdm E1 with libpri. On Thu, Jul 5, 2012 at 11:00 AM, Anton Kvashenkin wrote: > Post this output to backtracker. > > 2012/7/5 Robin Gilks >> >> Current running from commit 04bd463d12bf4c0116bec3cee537749b2040ed40 >> >> This occurs on all incoming calls from a Digium Wildcard TE210P with >> a zaptel E1 interface with libpri. >> >> I've tried updating to dahdi-linux-complete-2.6.1+2.6.1.tar.gz but >> failed miserably - couldn't even get the E1 link itself to come up :( >> >> I've captured a backtrace of the event on the zaptel drivers, >> hopefully useful to someone. >> >> gdb /usr/local/freeswitch_TR/bin/freeswitch -x ${HOME}/gdbcommands >> >> >> $ cat gdbcommands >> handle SIGPIPE nostop noprint >> handle SIG33 nostop noprint >> set logging on >> set pagination off >> set breakpoint pending on >> run >> thread apply all bt full >> set logging off >> >> Attached is the output of the backtrace. >> >> >> >> -- >> Robin Gilks >> >> -- >> >> ------------------------------ >> This email, including any attachments, is only for the intended recipient. >> It is subject to copyright, is confidential and may be the subject of >> legal or other privilege, none of which is waived or lost by reason >> of this transmission. >> If you are not an intended recipient, you may not use, disseminate, >> distribute or reproduce such email, any attachments, or any part thereof. >> If you have received a message in error, please notify the sender >> immediately and erase all copies of the message and any attachments. >> Unfortunately, we cannot warrant that the email has not been altered >> or corrupted during transmission nor can we guarantee that any email >> or any attachments are free from computer viruses or other conditions >> which may damage or interfere with recipient data, hardware or >> software. The recipient relies upon its own procedures and assumes >> all risk of use and of opening any attachments. >> ------------------------------ >> >> _____________________________________________________________________ >> ____ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> ers >> http://www.freeswitch.org >> > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users 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 > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff5389332761306118382! From royj at yandex.ru Thu Jul 5 12:57:11 2012 From: royj at yandex.ru (royj) Date: Thu, 5 Jul 2012 12:57:11 +0400 Subject: [Freeswitch-users] twin events MESSAGE (mod_gsmopen, sms_incoming) Message-ID: <20120705125711.b2bfb0d7.royj@yandex.ru> Hi all I want to use the mod_event_socket inbound mode to handle incoming SMS, there's a chatplan (http://pastebin.freeswitch.com/19448), and with this simple script (http://pastebin.freeswitch.com/19451) for every SMS I see two identical events (Event-Sequence with the same numbers as the other corresponding Headers do not differ from each other), is this normal? -- Regards, royj From dnotivol at gmail.com Thu Jul 5 13:36:21 2012 From: dnotivol at gmail.com (David Notivol) Date: Thu, 5 Jul 2012 11:36:21 +0200 Subject: [Freeswitch-users] Wrong date/time in cli and events Message-ID: Hello all, I've seen FreeSwitch is reporting wrong dates/times in the events (Event-Date-*, Caller-Channel-*-Time). Checking the current date in the cli, it says the wrong date as well, although the system date of the machine is fine. freeswitch at internal> strftime 1970-01-05 13:42:15 freeswitch at internal> strftime_tz 1970-01-05 freeswitch at internal> strepoch 412943 [root at hermes-test1 freeswitch]# date Thu Jul 5 05:28:26 EDT 2012 Do you have any clue of why this is happening and how could I solve it? Thanks. -- Regards, David Notivol -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120705/ceca8031/attachment-0001.html From peter.olsson at visionutveckling.se Thu Jul 5 14:05:51 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 5 Jul 2012 10:05:51 +0000 Subject: [Freeswitch-users] Wrong date/time in cli and events Message-ID: <1FFF97C269757C458224B7C895F35F1512EACF@cantor.std.visionutv.se> Update to latest git, it was fixed yesterday. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r David Notivol Skickat: den 5 juli 2012 11:36 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] Wrong date/time in cli and events Hello all, I've seen FreeSwitch is reporting wrong dates/times in the events (Event-Date-*, Caller-Channel-*-Time). Checking the current date in the cli, it says the wrong date as well, although the system date of the machine is fine. freeswitch at internal> strftime 1970-01-05 13:42:15 freeswitch at internal> strftime_tz 1970-01-05 freeswitch at internal> strepoch 412943 [root at hermes-test1 freeswitch]# date Thu Jul 5 05:28:26 EDT 2012 Do you have any clue of why this is happening and how could I solve it? Thanks. -- Regards, David Notivol !DSPAM:4ff55f0732761568247525! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120705/56786a40/attachment.html From amusingparsnip at me.com Thu Jul 5 14:52:16 2012 From: amusingparsnip at me.com (AP) Date: Thu, 05 Jul 2012 11:52:16 +0100 Subject: [Freeswitch-users] Outbound ending after 30secs Message-ID: Ok, so I have calls cutting off after 30 seconds on one type of handset but not another. Looking at ngrep, two things that jump out at me: 1. The working handset is sending ACK responses to FreeSWITCH's SDP messages, but the other one isn't. Presumably this is why FS is issuing BYE and dropping the call after 30 seconds, but there are no settings on the phone to this affect (Siemens Gigaset), and I'm not sure whether it's related to point 2 below: 2. The one which is cutting off is using UDP, and the working one TCP. TBH I can't see why it's using UDP when the SIP profile etc is set to use TCP!? Can anyone give me some pointers / see anything I'm missing? U 2012/07/03 15:40:18.370502 (FS_PRIVATE_IP):5060 -> (HANDSET_IP):5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP (HANDSET_IP):5060;branch=z9hG4bKeb201ee255084c9790fb39182842931b;rport=5060. From: ;tag=2284500536. To: ;tag=graNreK1vcrXg. Call-ID: 3301553880@(HANDSET_HOSTNAME). CSeq: 3 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120620T194320Z~a0a9efcf02+unclean~20120621T135422Z. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 245. Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1341299187 1341299188 IN IP4 (FS_PRIVATE_IP). s=FreeSWITCH. c=IN IP4 (FS_PRIVATE_IP). t=0 0. m=audio 27202 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. # T 2012/07/03 15:40:19.841678 (ITSP_IP):5060 -> (FS_PRIVATE_IP):35664 [AP] OPTIONS sip:gw+MyGateway(FS_PUBLIC_IP):36304;tport=tcp;transport=tcp;gw=MyGateway SIP/2.0. Via: SIP/2.0/TCP (ITSP_IP);branch=z9hG4bKaae6e0SZ07p9c. Max-Forwards: 70. From: ;tag=Fp3vKpcZBN4UF. To: . Call-ID: d8f67e63-3fbf-1230-2cb8-001b78596f84. CSeq: 31241096 OPTIONS. Contact: . User-Agent: my-itsp.com. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Length: 0. . # T 2012/07/03 15:40:19.842080 (FS_PRIVATE_IP):35664 -> (ITSP_IP):5060 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP (ITSP_IP);branch=z9hG4bKaae6e0SZ07p9c;rport=5060. From: ;tag=Fp3vKpcZBN4UF. To: ;tag=Q1y1N4BZU22pe. Call-ID: d8f67e63-3fbf-1230-2cb8-001b78596f84. CSeq: 31241096 OPTIONS. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120620T194320Z~a0a9efcf02+unclean~20120621T135422Z. Accept: application/sdp. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, refer. Content-Length: 0. . ## U 2012/07/03 15:40:22.370503 (FS_PRIVATE_IP):5060 -> (HANDSET_IP):5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP (HANDSET_IP):5060;branch=z9hG4bKeb201ee255084c9790fb39182842931b;rport=5060. From: ;tag=2284500536. To: ;tag=graNreK1vcrXg. Call-ID: 3301553880@(HANDSET_HOSTNAME). CSeq: 3 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120620T194320Z~a0a9efcf02+unclean~20120621T135422Z. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 245. Remote-Party-ID: "Outbound Call" ;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1341299187 1341299188 IN IP4 (FS_PRIVATE_IP). s=FreeSWITCH. c=IN IP4 (FS_PRIVATE_IP). t=0 0. m=audio 27202 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. # U 2012/07/03 15:40:22.863571 (FS_PRIVATE_IP):5060 -> (HANDSET_IP):5060 BYE sip:1002@(HANDSET_IP):5060 SIP/2.0. Via: SIP/2.0/UDP (FS_PUBLIC_IP);rport;branch=z9hG4bKp90ccpcU3jUaN. Max-Forwards: 70. From: ;tag=graNreK1vcrXg. To: ;tag=2284500536. Call-ID: 3301553880@(HANDSET_HOSTNAME). CSeq: 30327147 BYE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120620T194320Z~a0a9efcf02+unclean~20120621T135422Z. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. Reason: SIP;cause=408;text="ACK Timeout". Content-Length: 0. . (FS_PRIVATE_IP) From muelbuesch at as-infodienste.de Thu Jul 5 15:33:57 2012 From: muelbuesch at as-infodienste.de (=?ISO-8859-15?Q?Marcus_M=FClb=FCsch?=) Date: Thu, 05 Jul 2012 13:33:57 +0200 Subject: [Freeswitch-users] Unable to set effective_caller_id_number when bridging using Openzap Message-ID: <4FF57BA5.7090000@as-infodienste.de> Hello all, yes, for hardware reasons (very old Sangoma card with very old firmware) I have to use OpenZAP instead of freetTDM. So, when setting the effective_caller_id that value isn't used. See that part of my dialplan here: ---snip--- ---snip--- That configuration works with SIP, however. The line is capable to accept a change of callerid; that was tested with asterisk. Here is the relevant part of my openzap.conf.xml ---snip--- ---snip--- I am at my wits end and have no more idea what I'm doing wrong. Any idea is appreciated (but don't expect a reply or thank you before Tuesday, sorry) Marcus From dnotivol at gmail.com Thu Jul 5 15:38:30 2012 From: dnotivol at gmail.com (David Notivol) Date: Thu, 5 Jul 2012 13:38:30 +0200 Subject: [Freeswitch-users] Wrong date/time in cli and events In-Reply-To: <1FFF97C269757C458224B7C895F35F1512EACF@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1512EACF@cantor.std.visionutv.se> Message-ID: Thanks Peter, that solved the problem. I was running yesterday's git head, so I didn't think it was fixed since then. Regards, David Notivol. 2012/7/5 Peter Olsson > Update to latest git, it was fixed yesterday.**** > > ** ** > > /Peter**** > > ** ** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *David Notivol > *Skickat:* den 5 juli 2012 11:36 > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* [Freeswitch-users] Wrong date/time in cli and events**** > > ** ** > > Hello all,**** > > ** ** > > I've seen FreeSwitch is reporting wrong dates/times in the events > (Event-Date-*, Caller-Channel-*-Time). Checking the current date in the > cli, it says the wrong date as well, although the system date of the > machine is fine.**** > > ** ** > > freeswitch at internal> strftime**** > > 1970-01-05 13:42:15**** > > freeswitch at internal> strftime_tz**** > > 1970-01-05**** > > freeswitch at internal> strepoch **** > > 412943**** > > ** ** > > [root at hermes-test1 freeswitch]# date**** > > Thu Jul 5 05:28:26 EDT 2012**** > > ** ** > > Do you have any clue of why this is happening and how could I solve it?*** > * > > Thanks.**** > > ** ** > > -- > Regards, > David Notivol**** > > !DSPAM:4ff55f0732761568247525! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120705/4d845dff/attachment.html From peter.olsson at visionutveckling.se Thu Jul 5 15:52:59 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 5 Jul 2012 11:52:59 +0000 Subject: [Freeswitch-users] Wrong date/time in cli and events Message-ID: <1FFF97C269757C458224B7C895F35F1512EBD2@cantor.std.visionutv.se> You just happened to be one of the few unlucky :) /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r David Notivol Skickat: den 5 juli 2012 13:39 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Wrong date/time in cli and events Thanks Peter, that solved the problem. I was running yesterday's git head, so I didn't think it was fixed since then. Regards, David Notivol. 2012/7/5 Peter Olsson > Update to latest git, it was fixed yesterday. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r David Notivol Skickat: den 5 juli 2012 11:36 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] Wrong date/time in cli and events Hello all, I've seen FreeSwitch is reporting wrong dates/times in the events (Event-Date-*, Caller-Channel-*-Time). Checking the current date in the cli, it says the wrong date as well, although the system date of the machine is fine. freeswitch at internal> strftime 1970-01-05 13:42:15 freeswitch at internal> strftime_tz 1970-01-05 freeswitch at internal> strepoch 412943 [root at hermes-test1 freeswitch]# date Thu Jul 5 05:28:26 EDT 2012 Do you have any clue of why this is happening and how could I solve it? Thanks. -- Regards, David Notivol _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff57bb032769674818857! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120705/891fca31/attachment-0001.html From hakkie42 at gmail.com Thu Jul 5 11:42:21 2012 From: hakkie42 at gmail.com (Jim) Date: Thu, 05 Jul 2012 09:42:21 +0200 Subject: [Freeswitch-users] Getting started wiki page & external gateway: do I understand correctly? Message-ID: <4FF5455D.4020004@gmail.com> Hi list, More or less beginner, home situation; trying to set up fs (again).. I've got 2 questions: 1. Posting here to make sure I understand before modifying the wiki. Could you please correct me if I'm wrong? Over here http://wiki.freeswitch.org/wiki/Getting_Started_Guide#External it says "The External (formerly "outbound") profile handles outbound registrations to a SIP provider." However, earlier on it also mentions you can let external devices (i.e. user phones in their own networks) register with that profile... so I'd change this to: "The External (formerly "outbound") profile also handles outbound registrations to a SIP provider." Then this: "The external profile allows anonymous calling, which is required as your provider will never authenticate with you to send you a call." Skimmed through the bridge book p78, Receiving calls, which seems to confirm external profile does not require authentication. Ok, fine. Then this: "In order to secure your FreeSWITCH it is wise to link your outbound profile to a dialplan context other than 'default', which in the default configuration is the where authenticated users are placed." Seems this advice mixes a default situation (default dialplan being sensitive) with conditional advice (your outbound profile which would be external in a default config). I would change outbound to external in order to lessen confusion: "In order to secure your FreeSWITCH it is wise to link your exgternal profile to a dialplan context other than 'default', which in the default configuration is where authenticated users are placed." ... although what is probably really meant is something like: "As mentioned, the profile used for outbound registrations allows anonymous, unauthenticated calling. By default, this profile is the external profile. In order to secure your FreeSWITCH, don't link this profile to a dialplan that allows dialing paid numbers or dialing users (who may be bothered/harrassed) without any further checking. Summary: in a default configuration: don't link your external profile to a 'default' dialplan." ... which is a mouthful. 2. Given the above, if I want to have external users in their own network behind NAT register to me, it would be best if I define an additional profile that does require SIP authentication, right? I can then use the external profile to register with SIP trunks etc. Thanks! -- Regards, jb From gmaruzz at gmail.com Thu Jul 5 18:01:34 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 5 Jul 2012 16:01:34 +0200 Subject: [Freeswitch-users] twin events MESSAGE (mod_gsmopen, sms_incoming) In-Reply-To: <20120705125711.b2bfb0d7.royj@yandex.ru> References: <20120705125711.b2bfb0d7.royj@yandex.ru> Message-ID: On Thu, Jul 5, 2012 at 10:57 AM, royj wrote: > > I want to use the mod_event_socket inbound mode to handle incoming SMS, > there's a chatplan (http://pastebin.freeswitch.com/19448), and with this > simple script (http://pastebin.freeswitch.com/19451) for every SMS I see > two identical events (Event-Sequence with the same numbers as the other > corresponding Headers do not differ from each other), is this normal? > > Yes, that's normal, is mod_sms that's duplicating messages. If you unload mod_sms you'll have just one message in ESL. -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120705/76904e78/attachment.html From dujinfang at gmail.com Thu Jul 5 18:50:10 2012 From: dujinfang at gmail.com (dujinfang) Date: Thu, 5 Jul 2012 22:50:10 +0800 Subject: [Freeswitch-users] twin events MESSAGE (mod_gsmopen, sms_incoming) In-Reply-To: References: <20120705125711.b2bfb0d7.royj@yandex.ru> Message-ID: <9C206E75-749E-4810-8C70-756615735117@gmail.com> I saw this too but I think if you don't fire or send in chatplan then the message should be only one? ???? iPhone ? 2012-7-5???10:01?Giovanni Maruzzelli ??? > On Thu, Jul 5, 2012 at 10:57 AM, royj wrote: > > I want to use the mod_event_socket inbound mode to handle incoming SMS, there's a chatplan (http://pastebin.freeswitch.com/19448), and with this simple script (http://pastebin.freeswitch.com/19451) for every SMS I see two identical events (Event-Sequence with the same numbers as the other corresponding Headers do not differ from each other), is this normal? > > > Yes, that's normal, is mod_sms that's duplicating messages. If you unload mod_sms you'll have just one message in ESL. > > -giovanni > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120705/23872c6e/attachment.html From gmaruzz at celliax.org Thu Jul 5 19:07:56 2012 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 5 Jul 2012 17:07:56 +0200 Subject: [Freeswitch-users] twin events MESSAGE (mod_gsmopen, sms_incoming) In-Reply-To: <9C206E75-749E-4810-8C70-756615735117@gmail.com> References: <20120705125711.b2bfb0d7.royj@yandex.ru> <9C206E75-749E-4810-8C70-756615735117@gmail.com> Message-ID: On Thu, Jul 5, 2012 at 4:50 PM, dujinfang wrote: > I saw this too but I think if you don't fire or send in chatplan then the > message should be only one? > > Ciao Seven! Yep, I believe the same (I just associate the chatplan with firing or sending, stupid me). Not tested, tough. -giovanni > ???? iPhone > > ? 2012-7-5???10:01?Giovanni Maruzzelli ??? > > On Thu, Jul 5, 2012 at 10:57 AM, royj wrote: > >> >> I want to use the mod_event_socket inbound mode to handle incoming SMS, >> there's a chatplan (http://pastebin.freeswitch.com/19448), and with this >> simple script (http://pastebin.freeswitch.com/19451) for every SMS I see >> two identical events (Event-Sequence with the same numbers as the other >> corresponding Headers do not differ from each other), is this normal? >> >> > Yes, that's normal, is mod_sms that's duplicating messages. If you unload > mod_sms you'll have just one message in ESL. > > -giovanni > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120705/aa6dd07c/attachment.html From curriegrad2004 at gmail.com Thu Jul 5 19:08:39 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Thu, 5 Jul 2012 08:08:39 -0700 Subject: [Freeswitch-users] Getting started wiki page & external gateway: do I understand correctly? In-Reply-To: <4FF5455D.4020004@gmail.com> References: <4FF5455D.4020004@gmail.com> Message-ID: Actually I'd say not to modify the wiki at all. It is valid as it stands. Typically most users who use this default diaplan will register their SIP provider's gateways on the external profile and all of their phones in the internal profile. Regarding the security part, well said over at that front. * users have been burned because of this. You can also set the default dialplan for that authenticated profile to be on public and set your users's user_context param to whatever context you want the user to be on. And the answer to your last question. Yes you sure as heck as can create a new sofia sip profile just for phones to register from the outside world. On Thu, Jul 5, 2012 at 12:42 AM, Jim wrote: > Hi list, > > More or less beginner, home situation; trying to set up fs (again).. > > I've got 2 questions: > > 1. Posting here to make sure I understand before modifying the wiki. > Could you please correct me if I'm wrong? > > Over here > http://wiki.freeswitch.org/wiki/Getting_Started_Guide#External > > it says > "The External (formerly "outbound") profile handles outbound > registrations to a SIP provider." > However, earlier on it also mentions you can let external devices (i.e. > user phones in their own networks) register with that profile... so I'd > change this to: > "The External (formerly "outbound") profile also handles outbound > registrations to a SIP provider." > > Then this: > "The external profile allows anonymous calling, which is required as > your provider will never authenticate with you to send you a call." > > Skimmed through the bridge book p78, Receiving calls, which seems to > confirm external profile does not require authentication. > Ok, fine. > > Then this: > "In order to secure your FreeSWITCH it is wise to link your outbound > profile to a dialplan context other than 'default', which in the default > configuration is the where authenticated users are placed." > Seems this advice mixes a default situation (default dialplan being > sensitive) with conditional advice (your outbound profile which would be > external in a default config). > > I would change outbound to external in order to lessen confusion: > "In order to secure your FreeSWITCH it is wise to link your exgternal > profile to a dialplan context other than 'default', which in the default > configuration is where authenticated users are placed." > > ... although what is probably really meant is something like: > "As mentioned, the profile used for outbound registrations allows > anonymous, unauthenticated calling. By default, this profile is the > external profile. In order to secure your FreeSWITCH, don't link this > profile to a dialplan that allows dialing paid numbers or dialing users > (who may be bothered/harrassed) without any further checking. > > Summary: in a default configuration: don't link your external profile to > a 'default' dialplan." > ... which is a mouthful. > > 2. Given the above, if I want to have external users in their own > network behind NAT register to me, it would be best if I define an > additional profile that does require SIP authentication, right? > > I can then use the external profile to register with SIP trunks etc. > > Thanks! > -- > Regards, > > jb > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Thu Jul 5 19:13:10 2012 From: gmaruzz at celliax.org (Giovanni Maruzzelli) Date: Thu, 5 Jul 2012 17:13:10 +0200 Subject: [Freeswitch-users] twin events MESSAGE (mod_gsmopen, sms_incoming) In-Reply-To: References: <20120705125711.b2bfb0d7.royj@yandex.ru> <9C206E75-749E-4810-8C70-756615735117@gmail.com> Message-ID: On Thu, Jul 5, 2012 at 5:07 PM, Giovanni Maruzzelli wrote: > On Thu, Jul 5, 2012 at 4:50 PM, dujinfang wrote: > >> I saw this too but I think if you don't fire or send in chatplan then the >> message should be only one? >> >> > Ciao Seven! > Yep, I believe the same (I just associate the chatplan with firing or > sending, stupid me). Not tested, tough. > > Confirmed, just tested. If you do not have send or fire in chatplan, no additional message will be on esl. -giovanni -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120705/d1e61221/attachment.html From neo.cheema at gmail.com Thu Jul 5 17:37:14 2012 From: neo.cheema at gmail.com (Neo Cheema) Date: Thu, 5 Jul 2012 19:07:14 +0530 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq Message-ID: Hi all, I was hoping to find a way to push CDR information into a queue. Zeromq comes out as an obvious choice because mod_event_zmq already exits. However, I can't find a way to configure this module to push CDR info. Have any of you guys tried it? Regards From Rob.Moore at Aeriandi.com Thu Jul 5 19:42:41 2012 From: Rob.Moore at Aeriandi.com (Rob Moore) Date: Thu, 5 Jul 2012 15:42:41 +0000 Subject: [Freeswitch-users] Monitoring Success of SIP Registration In-Reply-To: <4FF32F0E.60302@cupis.co.uk> References: <49C5FCA19A8A114493EBAACA42FE5899105B0A1A@1AERDCEXCHMBX1.AER.AERCO.local> <4FF32F0E.60302@cupis.co.uk> Message-ID: <49C5FCA19A8A114493EBAACA42FE5899105B11A5@1AERDCEXCHMBX1.AER.AERCO.local> Hi Paul I just picked up on your reply. Thanks very much, this does indeed seem to be exactly what I need. Thanks again Rob -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul Cupis Sent: 03 July 2012 18:43 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Monitoring Success of SIP Registration On 03/07/12 18:05, Rob Moore wrote: > However this does leave me with another situation while I look for the > permanent solution. Its normally during out of hours that our > freeswitch boxes decide they do not want to respond to registrations > and I would like to setup some rolling registrations every 2 minutes > that the result of can be reported back on to allow for some Alerting to take place. ... > So I thought I 'd open it up to the users group to see if you had any > suggestions on a light weight method I could use to generate these > registration request. I'd suggest looking at tools like sipsak and possibly the Nagios plugin check_sip for this sort of functionality. http://sipsak.org/ http://exchange.nagios.org/directory/Plugins/Network-Protocols/*-VoIP/SIP/check_sip-sipsak/details Regards, _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jmesquita at freeswitch.org Thu Jul 5 19:56:08 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Thu, 5 Jul 2012 12:56:08 -0300 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: References: Message-ID: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> I haven't actually looked at the code, but you have to look at the variables on CHANNEL_REPORTING for example. There you will see all the vars needed to produce CDR entries, but they won't be on XML format like xml_curl would create. Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Thursday, July 5, 2012 at 10:37 AM, Neo Cheema wrote: > Hi all, > > I was hoping to find a way to push CDR information into a queue. > Zeromq comes out as an obvious choice because mod_event_zmq already > exits. However, I can't find a way to configure this module to push > CDR info. Have any of you guys tried it? > > Regards > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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/20120705/7942a52c/attachment.html From dujinfang at gmail.com Thu Jul 5 20:30:00 2012 From: dujinfang at gmail.com (Seven Du) Date: Fri, 6 Jul 2012 00:30:00 +0800 Subject: [Freeswitch-users] conference auto dial no video Message-ID: <710B80A59BBD4FABB36D9E3D43CBF3CA@gmail.com> Hi, Is it possible to setup video when auto_call in conference? I see no video even I set absolute_codec_string INVITE sip:1000 at 10.64.14.206:42894;rinstance=c688826a567bedcb SIP/2.0 Via: SIP/2.0/UDP 10.64.14.206;rport;branch=z9hG4bKm16gttgFX1rrg Max-Forwards: 70 From: "FreeSWITCH" ;tag=Q49S334vvyBQS To: Call-ID: 9ded3052-e523-49b5-949e-ec18ab13e0af CSeq: 30415873 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120703T221514Z~b1ae97466d+unclean~20120704T022616Z Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 201 X-FS-Support: update_display,send_info Remote-Party-ID: "FreeSWITCH" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1341480903 1341480904 IN IP4 10.64.14.206 s=FreeSWITCH c=IN IP4 10.64.14.206 t=0 0 m=audio 22972 RTP/AVP 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 Thanks -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120706/77fc3150/attachment.html From mgg at giagnocavo.net Thu Jul 5 22:08:24 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 5 Jul 2012 18:08:24 +0000 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> Wouldn?t the problem with ZeroMQ be no reliable delivery of CDRs? I was under the impression that 0MQ was more for high-performance, and things like guaranteed delivery are left as an exercise to the user. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita Sent: Thursday, July 05, 2012 9:56 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Pushing CDR Information into Zeromq I haven't actually looked at the code, but you have to look at the variables on CHANNEL_REPORTING for example. There you will see all the vars needed to produce CDR entries, but they won't be on XML format like xml_curl would create. Regards, -- Jo?o Mesquita Sent with Sparrow On Thursday, July 5, 2012 at 10:37 AM, Neo Cheema wrote: Hi all, I was hoping to find a way to push CDR information into a queue. Zeromq comes out as an obvious choice because mod_event_zmq already exits. However, I can't find a way to configure this module to push CDR info. Have any of you guys tried it? Regards _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120705/3297a9a3/attachment-0001.html From all.eforums at gmail.com Fri Jul 6 00:27:53 2012 From: all.eforums at gmail.com (A E G) Date: Thu, 5 Jul 2012 16:27:53 -0400 Subject: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a In-Reply-To: References: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern> <4FED4DA1.10304@belrosbank.by> Message-ID: Thanks for the short/straight pointers re: what matters most for freeswitch performance. Are there any equivalent pearls of wisdom "best practices" re: CPU, RAM and Shared Memory/Bus that you or others can advise on, if this was to run in a virtual environment in a private cloud of Xen Server, KVM or vCloud etc? Probably it will depend on the underlying hardware or not? kernel tuning of the Linux guest as well as resource sharing? Any light you or anyone else can shed on this would be greatly appreciated Thanks in advance On Fri, Jun 29, 2012 at 1:07 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > you could do with more than 2 cpu, the more cores the better you will do. > For similar prices you could find a 8 or 12 core box. > > The 3 most important things in a FS server are, as many CPU cores as > possible, as much ram as possible, the best possible CPU with shared > bus to RAM and CPU. > > > > 2012/6/29 Eugene Shcherbatyuk : > > Session limit is configurable. Have a look at switch.conf.xml file. > > > > > > On 28/06/12 21:41, Hoexum, Edwin wrote: > >> > >> I am trying to do some load testing on a freeswitch for transcoding from > >> SILK to G711a. The question is: Is the limit for RTP sessions > >> limited to CPU power or are there also limits on the freeswitch > >> application and can I tune freeswitch to do the job for 1500 sessions or > >> more. > > > > > > ========================================================= > > > > ?????? ????????? ? ????? ???????? (<>) ???????? > ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? > ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? > ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. > ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? > "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? > ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? > ?????????????. > > > > ========================================================= > > > > This message and any attachments (the "message") are confidential, > intended solely for the addressees, and may contain legally privileged > information. Any unauthorized use or dissemination is prohibited. E-mails > are susceptible to alteration. Neither JSC "BELROSBANK" nor any of its > subsidiaries or affiliates shall be liable for the message if altered, > changed or falsified. > > > > ========================================================= > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120705/f50166f0/attachment.html From gavin.henry at gmail.com Fri Jul 6 00:48:05 2012 From: gavin.henry at gmail.com (Gavin Henry) Date: Thu, 5 Jul 2012 21:48:05 +0100 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> Message-ID: <9343F57A41054F8DB89BD4A0C17CFD7B@gmail.com> We sponsored this: https://metacpan.org/release/Message-Passing https://metacpan.org/release/Message-Passing-ZeroMQ https://metacpan.org/module/BOBTFISH/Message-Passing-Input-Freeswitch-0.002/lib/Message/Passing/Input/Freeswitch.pm https://metacpan.org/release/Message-Passing-Output-WebHooks We use it to pump events into zeromq as part of our API - http://www.surevoip.co.uk/api You can easily do cdrs. Gavin. -- Gavin Henry Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Thursday, 5 July 2012 at 19:08, Michael Giagnocavo wrote: > > Wouldn?t the problem with ZeroMQ be no reliable delivery of CDRs? I was under the impression that 0MQ was more for high-performance, and things like guaranteed delivery are left as an exercise to the user. > > > -Michael > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita > Sent: Thursday, July 05, 2012 9:56 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Pushing CDR Information into Zeromq > > > > > > I haven't actually looked at the code, but you have to look at the variables on CHANNEL_REPORTING for example. There you will see all the vars needed to produce CDR entries, but they won't be on XML format like xml_curl would create. > > > > > > > Regards, > > > > > > > > > -- > > > > Jo?o Mesquita > > > > Sent with Sparrow (http://www.sparrowmailapp.com/?sig) > > > > > > > > On Thursday, July 5, 2012 at 10:37 AM, Neo Cheema wrote: > > > > Hi all, > > > > > > > > > > > > > > > > I was hoping to find a way to push CDR information into a queue. > > > > > > > > Zeromq comes out as an obvious choice because mod_event_zmq already > > > > > > > > exits. However, I can't find a way to configure this module to push > > > > > > > > CDR info. Have any of you guys tried it? > > > > > > > > > > > > > > > > Regards > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > > > > Professional FreeSWITCH Consulting Services: > > > > > > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > > > > http://www.freeswitch.org > > > > > > > > http://wiki.freeswitch.org > > > > > > > > http://www.cluecon.com > > > > > > > > > > > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > > > > > > > > > > > FreeSWITCH-users mailing list > > > > > > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120705/926d839d/attachment-0001.html From jmesquita at freeswitch.org Fri Jul 6 01:08:33 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Thu, 5 Jul 2012 18:08:33 -0300 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> Message-ID: <537ACDFCE00243439E851C86C8C25E7A@freeswitch.org> Michael, You are partially correct because it depends on the socket type you use on ZeroMQ. Since the socket implemented on mod_zmq is a simple PUB/SUB, yes, messages would be discarded if they are not received by the other end. If we would use a XREQ/XREP then it would be queued until you are able to receive them. -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Thursday, July 5, 2012 at 3:08 PM, Michael Giagnocavo wrote: > > Wouldn?t the problem with ZeroMQ be no reliable delivery of CDRs? I was under the impression that 0MQ was more for high-performance, and things like guaranteed delivery are left as an exercise to the user. > > > -Michael > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita > Sent: Thursday, July 05, 2012 9:56 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Pushing CDR Information into Zeromq > > > > > > I haven't actually looked at the code, but you have to look at the variables on CHANNEL_REPORTING for example. There you will see all the vars needed to produce CDR entries, but they won't be on XML format like xml_curl would create. > > > > > > > Regards, > > > > > > > > > -- > > > > Jo?o Mesquita > > > > Sent with Sparrow (http://www.sparrowmailapp.com/?sig) > > > > > > > > On Thursday, July 5, 2012 at 10:37 AM, Neo Cheema wrote: > > > > Hi all, > > > > > > > > > > > > > > > > I was hoping to find a way to push CDR information into a queue. > > > > > > > > Zeromq comes out as an obvious choice because mod_event_zmq already > > > > > > > > exits. However, I can't find a way to configure this module to push > > > > > > > > CDR info. Have any of you guys tried it? > > > > > > > > > > > > > > > > Regards > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > > > > Professional FreeSWITCH Consulting Services: > > > > > > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > > > > http://www.freeswitch.org > > > > > > > > http://wiki.freeswitch.org > > > > > > > > http://www.cluecon.com > > > > > > > > > > > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > > > > > > > > > > > FreeSWITCH-users mailing list > > > > > > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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/20120705/f07d22a6/attachment.html From msc at freeswitch.org Fri Jul 6 01:17:25 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 5 Jul 2012 14:17:25 -0700 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: <9343F57A41054F8DB89BD4A0C17CFD7B@gmail.com> References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <9343F57A41054F8DB89BD4A0C17CFD7B@gmail.com> Message-ID: Woohoo Perl! http://www.bastichlabz.org/bastich/Strips/ba980225.gif -MC On Thu, Jul 5, 2012 at 1:48 PM, Gavin Henry wrote: > We sponsored this: > > https://metacpan.org/release/Message-Passing > https://metacpan.org/release/Message-Passing-ZeroMQ > > https://metacpan.org/module/BOBTFISH/Message-Passing-Input-Freeswitch-0.002/lib/Message/Passing/Input/Freeswitch.pm > https://metacpan.org/release/Message-Passing-Output-WebHooks > > We use it to pump events into zeromq as part of our API - > http://www.surevoip.co.uk/api > > You can easily do cdrs. > > Gavin. > > -- > Gavin Henry > Sent with Sparrow > > On Thursday, 5 July 2012 at 19:08, Michael Giagnocavo wrote: > > Wouldn?t the problem with ZeroMQ be no reliable delivery of CDRs? I was > under the impression that 0MQ was more for high-performance, and things > like guaranteed delivery are left as an exercise to the user.**** > > -Michael**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jo?o > Mesquita > *Sent:* Thursday, July 05, 2012 9:56 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Pushing CDR Information into Zeromq**** > > ** ** > > I haven't actually looked at the code, but you have to look at the > variables on CHANNEL_REPORTING for example. There you will see all the vars > needed to produce CDR entries, but they won't be on XML format like > xml_curl would create.**** > > ** ** > > Regards,**** > > ** ** > > -- **** > > Jo?o Mesquita**** > > Sent with Sparrow **** > > ** ** > > On Thursday, July 5, 2012 at 10:37 AM, Neo Cheema wrote:**** > > Hi all,**** > > ** ** > > I was hoping to find a way to push CDR information into a queue.**** > > Zeromq comes out as an obvious choice because mod_event_zmq already**** > > exits. However, I can't find a way to configure this module to push**** > > CDR info. Have any of you guys tried it?**** > > ** ** > > Regards**** > > ** ** > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120705/5034c3f1/attachment-0001.html From aam-003 at cox.net Fri Jul 6 01:04:33 2012 From: aam-003 at cox.net (aahndee) Date: Thu, 5 Jul 2012 21:04:33 +0000 (UTC) Subject: [Freeswitch-users] Error in compilation after git pull References: <33959.2329379686$1341443648@news.gmane.org> Message-ID: > Hi All, > > I did a git pull just now. After make clean, the compile failed. > > The latest git commit log that I can see is: > > commit 04bd463d12bf4c0116bec3cee537749b2040ed40 > Author: Michael S Collins > Date: Wed Jul 4 12:09:32 2012 -0700 > > The relevant error is: > > making all mod_spandsp > Creating mod_spandsp_la-mod_spandsp.lo > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. > -I../../../../src/include -I../../../../libs/xmlrpc-c > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g > -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic > -Wdeclaration-after-statement > -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff > -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT > mod_spandsp_la-mod_spandsp.lo -MD -MP -MF > .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o > .libs/mod_spandsp_la-mod_spandsp.o > In file included from > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/expose.h:96, > from > /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:135, > from mod_spandsp.h:50, > from mod_spandsp.c:39: > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid. > h:33: > error: expected specifier-qualifier-list before > 'ademco_contactid_report_func_t' > cc1: warnings being treated as errors > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid. > h:48: > error: struct has no members > make[4]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1 > make[3]: *** [mod_spandsp-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > Am I missing something? I was able to compile a few weeks ago. > > Regards It seems like there are two header files for ademco_contactid.h: /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h and /usr/local/src/freeswitch/libs/spandsp/src/spandsp/ademco_contactid.h 'ademco_contactid_report_func_t' is defined in the latter. I tried adding this second header file in /usr/local/src/freeswitch/libs/spandsp/src/spandsp/expose.h (before the private one) and managed to get a little further - now things fail with quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../../../../src/include -I../../../../libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DMACOSX -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT mod_spandsp_la-mod_spandsp_modem.lo -MD -MP -MF .deps/mod_spandsp_la -mod_spandsp_modem.Tpo -c mod_spandsp_modem.c -o mod_spandsp_la-mod_spandsp_modem.o >/dev/null 2>&1 make[5]: *** No rule to make target `config/ax_compiler_vendor.m4', needed by `Makefile.in'. Stop. make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la] Error 2 make[3]: *** [mod_spandsp-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 From avi at avimarcus.net Fri Jul 6 01:59:39 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 6 Jul 2012 00:59:39 +0300 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <9343F57A41054F8DB89BD4A0C17CFD7B@gmail.com> Message-ID: What about simply using xml_cdr or json_cdr and then process the queue of the files? Move them to a sub-directory once processed. That gives you disk persistence... -Avi On Fri, Jul 6, 2012 at 12:17 AM, Michael Collins wrote: > Woohoo Perl! > http://www.bastichlabz.org/bastich/Strips/ba980225.gif > > -MC > > > On Thu, Jul 5, 2012 at 1:48 PM, Gavin Henry wrote: > >> We sponsored this: >> >> https://metacpan.org/release/Message-Passing >> https://metacpan.org/release/Message-Passing-ZeroMQ >> >> https://metacpan.org/module/BOBTFISH/Message-Passing-Input-Freeswitch-0.002/lib/Message/Passing/Input/Freeswitch.pm >> https://metacpan.org/release/Message-Passing-Output-WebHooks >> >> We use it to pump events into zeromq as part of our API - >> http://www.surevoip.co.uk/api >> >> You can easily do cdrs. >> >> Gavin. >> >> -- >> Gavin Henry >> Sent with Sparrow >> >> On Thursday, 5 July 2012 at 19:08, Michael Giagnocavo wrote: >> >> Wouldn?t the problem with ZeroMQ be no reliable delivery of CDRs? I was >> under the impression that 0MQ was more for high-performance, and things >> like guaranteed delivery are left as an exercise to the user.**** >> >> -Michael**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jo?o >> Mesquita >> *Sent:* Thursday, July 05, 2012 9:56 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Pushing CDR Information into Zeromq**** >> >> ** ** >> >> I haven't actually looked at the code, but you have to look at the >> variables on CHANNEL_REPORTING for example. There you will see all the vars >> needed to produce CDR entries, but they won't be on XML format like >> xml_curl would create.**** >> >> ** ** >> >> Regards,**** >> >> ** ** >> >> -- **** >> >> Jo?o Mesquita**** >> >> Sent with Sparrow **** >> >> ** ** >> >> On Thursday, July 5, 2012 at 10:37 AM, Neo Cheema wrote:**** >> >> Hi all,**** >> >> ** ** >> >> I was hoping to find a way to push CDR information into a queue.**** >> >> Zeromq comes out as an obvious choice because mod_event_zmq already**** >> >> exits. However, I can't find a way to configure this module to push**** >> >> CDR info. Have any of you guys tried it?**** >> >> ** ** >> >> Regards**** >> >> ** ** >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120706/d5fbfe66/attachment.html From robin.gilks at taitradio.com Fri Jul 6 02:37:42 2012 From: robin.gilks at taitradio.com (Robin Gilks) Date: Fri, 6 Jul 2012 10:37:42 +1200 Subject: [Freeswitch-users] Crash on incoming E1 call In-Reply-To: <1FFF97C269757C458224B7C895F35F1512E8EE@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F1512E8EE@cantor.std.visionutv.se> Message-ID: Done - FS-4395 - Segfault on E1 incoming call On Thu, Jul 5, 2012 at 7:08 PM, Peter Olsson wrote: > Please report to Jira, with all needed information > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > /Peter > > -----Ursprungligt meddelande----- > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Neo Cheema > Skickat: den 5 juli 2012 07:53 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Crash on incoming E1 call > > Facing exactly the same issue. But in my case, freeswitch crashes at startup itself while configuring spandsp. > My stack is Feetdm E1 with libpri. > > On Thu, Jul 5, 2012 at 11:00 AM, Anton Kvashenkin wrote: >> Post this output to backtracker. >> >> 2012/7/5 Robin Gilks >>> >>> Current running from commit 04bd463d12bf4c0116bec3cee537749b2040ed40 >>> >>> This occurs on all incoming calls from a Digium Wildcard TE210P with >>> a zaptel E1 interface with libpri. >>> >>> I've tried updating to dahdi-linux-complete-2.6.1+2.6.1.tar.gz but >>> failed miserably - couldn't even get the E1 link itself to come up :( >>> >>> I've captured a backtrace of the event on the zaptel drivers, >>> hopefully useful to someone. >>> >>> gdb /usr/local/freeswitch_TR/bin/freeswitch -x ${HOME}/gdbcommands >>> >>> >>> $ cat gdbcommands >>> handle SIGPIPE nostop noprint >>> handle SIG33 nostop noprint >>> set logging on >>> set pagination off >>> set breakpoint pending on >>> run >>> thread apply all bt full >>> set logging off >>> >>> Attached is the output of the backtrace. >>> >>> >>> >>> -- >>> Robin Gilks >>> >>> -- >>> >>> ------------------------------ >>> This email, including any attachments, is only for the intended recipient. >>> It is subject to copyright, is confidential and may be the subject of >>> legal or other privilege, none of which is waived or lost by reason >>> of this transmission. >>> If you are not an intended recipient, you may not use, disseminate, >>> distribute or reproduce such email, any attachments, or any part thereof. >>> If you have received a message in error, please notify the sender >>> immediately and erase all copies of the message and any attachments. >>> Unfortunately, we cannot warrant that the email has not been altered >>> or corrupted during transmission nor can we guarantee that any email >>> or any attachments are free from computer viruses or other conditions >>> which may damage or interfere with recipient data, hardware or >>> software. The recipient relies upon its own procedures and assumes >>> all risk of use and of opening any attachments. >>> ------------------------------ >>> >>> _____________________________________________________________________ >>> ____ Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>> ers >>> http://www.freeswitch.org >>> >> >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users 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 >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4ff5389332761306118382! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Robin Gilks -- ------------------------------ This email, including any attachments, is only for the intended recipient. It is subject to copyright, is confidential and may be the subject of legal or other privilege, none of which is waived or lost by reason of this transmission. If you are not an intended recipient, you may not use, disseminate, distribute or reproduce such email, any attachments, or any part thereof. If you have received a message in error, please notify the sender immediately and erase all copies of the message and any attachments. Unfortunately, we cannot warrant that the email has not been altered or corrupted during transmission nor can we guarantee that any email or any attachments are free from computer viruses or other conditions which may damage or interfere with recipient data, hardware or software. The recipient relies upon its own procedures and assumes all risk of use and of opening any attachments. ------------------------------ From nbhatti at gmail.com Fri Jul 6 02:39:56 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Fri, 6 Jul 2012 01:39:56 +0300 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <9343F57A41054F8DB89BD4A0C17CFD7B@gmail.com> Message-ID: Michael, you changed your glasses ... Again? -- Sent from a mobile device On Jul 6, 2012 12:39 AM, "Michael Collins" wrote: > Woohoo Perl! > http://www.bastichlabz.org/bastich/Strips/ba980225.gif > > -MC > > On Thu, Jul 5, 2012 at 1:48 PM, Gavin Henry wrote: > >> We sponsored this: >> >> https://metacpan.org/release/Message-Passing >> https://metacpan.org/release/Message-Passing-ZeroMQ >> >> https://metacpan.org/module/BOBTFISH/Message-Passing-Input-Freeswitch-0.002/lib/Message/Passing/Input/Freeswitch.pm >> https://metacpan.org/release/Message-Passing-Output-WebHooks >> >> We use it to pump events into zeromq as part of our API - >> http://www.surevoip.co.uk/api >> >> You can easily do cdrs. >> >> Gavin. >> >> -- >> Gavin Henry >> Sent with Sparrow >> >> On Thursday, 5 July 2012 at 19:08, Michael Giagnocavo wrote: >> >> Wouldn?t the problem with ZeroMQ be no reliable delivery of CDRs? I was >> under the impression that 0MQ was more for high-performance, and things >> like guaranteed delivery are left as an exercise to the user.**** >> >> -Michael**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Jo?o >> Mesquita >> *Sent:* Thursday, July 05, 2012 9:56 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Pushing CDR Information into Zeromq**** >> >> ** ** >> >> I haven't actually looked at the code, but you have to look at the >> variables on CHANNEL_REPORTING for example. There you will see all the vars >> needed to produce CDR entries, but they won't be on XML format like >> xml_curl would create.**** >> >> ** ** >> >> Regards,**** >> >> ** ** >> >> -- **** >> >> Jo?o Mesquita**** >> >> Sent with Sparrow **** >> >> ** ** >> >> On Thursday, July 5, 2012 at 10:37 AM, Neo Cheema wrote:**** >> >> Hi all,**** >> >> ** ** >> >> I was hoping to find a way to push CDR information into a queue.**** >> >> Zeromq comes out as an obvious choice because mod_event_zmq already**** >> >> exits. However, I can't find a way to configure this module to push**** >> >> CDR info. Have any of you guys tried it?**** >> >> ** ** >> >> Regards**** >> >> ** ** >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120706/7ca53673/attachment.html From mario_fs at mgtech.com Fri Jul 6 02:42:54 2012 From: mario_fs at mgtech.com (Mario G) Date: Thu, 5 Jul 2012 15:42:54 -0700 Subject: [Freeswitch-users] Error in compilation after git pull In-Reply-To: References: <33959.2329379686$1341443648@news.gmane.org> Message-ID: <8692BE1E-F305-4157-A7BF-F237E442295A@mgtech.com> FYI I had the problem yesterday on osX but after I did a ./bootstrap and ./configure it was fine. Mario G On Jul 5, 2012, at 2:04 PM, aahndee wrote: >> Hi All, >> >> I did a git pull just now. After make clean, the compile failed. >> >> The latest git commit log that I can see is: >> >> commit 04bd463d12bf4c0116bec3cee537749b2040ed40 >> Author: Michael S Collins >> Date: Wed Jul 4 12:09:32 2012 -0700 >> >> The relevant error is: >> >> making all mod_spandsp >> Creating mod_spandsp_la-mod_spandsp.lo >> quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. >> -I../../../../src/include -I../../../../libs/xmlrpc-c >> -I/usr/local/src/freeswitch/libs/curl/include >> -I/usr/local/src/freeswitch/src/include >> -I/usr/local/src/freeswitch/src/include >> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC >> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >> -I/usr/local/src/freeswitch/libs/curl/include >> -I/usr/local/src/freeswitch/src/include >> -I/usr/local/src/freeswitch/src/include >> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror >> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g >> -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic >> -Wdeclaration-after-statement >> -I/usr/local/src/freeswitch/libs/spandsp/src >> -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff >> -I/usr/local/src/freeswitch/libs/spandsp/src >> -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT >> mod_spandsp_la-mod_spandsp.lo -MD -MP -MF >> .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o >> .libs/mod_spandsp_la-mod_spandsp.o >> In file included from >> /usr/local/src/freeswitch/libs/spandsp/src/spandsp/expose.h:96, >> from >> /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:135, >> from mod_spandsp.h:50, >> from mod_spandsp.c:39: >> /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid. >> h:33: >> error: expected specifier-qualifier-list before >> 'ademco_contactid_report_func_t' >> cc1: warnings being treated as errors >> /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid. >> h:48: >> error: struct has no members >> make[4]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1 >> make[3]: *** [mod_spandsp-all] Error 1 >> make[2]: *** [all-recursive] Error 1 >> make[1]: *** [all-recursive] Error 1 >> make: *** [all] Error 2 >> >> Am I missing something? I was able to compile a few weeks ago. >> >> Regards > > It seems like there are two header files for ademco_contactid.h: > > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h > > and > > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/ademco_contactid.h > > 'ademco_contactid_report_func_t' is defined in the latter. I tried adding this > second header file in > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/expose.h (before the private > one) and managed to get a little further - now things fail with > > quiet_libtool: > compile: > gcc -DHAVE_CONFIG_H -I. -I../../../../src/include -I../../../../libs/xmlrpc-c > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src > -fPIC > -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 > -DHAVE_VISIBILITY=1 > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src > -Werror > -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 > -DHAVE_VISIBILITY=1 > -g > -ggdb > -DMACOSX > -DHAVE_OPENSSL > -Wall > -std=c99 > -pedantic > -Wdeclaration-after-statement > -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff > -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff > -I. > -g > -O2 > -MT mod_spandsp_la-mod_spandsp_modem.lo > -MD > -MP > -MF > .deps/mod_spandsp_la > -mod_spandsp_modem.Tpo > -c > mod_spandsp_modem.c > -o > mod_spandsp_la-mod_spandsp_modem.o >> /dev/null 2>&1 > make[5]: *** No rule to make target `config/ax_compiler_vendor.m4', needed by > `Makefile.in'. Stop. > make[4]: *** [/usr/local/src/freeswitch/libs/spandsp/src/libspandsp.la] Error 2 > make[3]: *** [mod_spandsp-all] Error 1 > make[2]: *** [all-recursive] Error 1 > make[1]: *** [all-recursive] Error 1 > make: *** [all] Error 2 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Jul 6 04:32:45 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 5 Jul 2012 19:32:45 -0500 Subject: [Freeswitch-users] Crash on incoming E1 call In-Reply-To: References: <1FFF97C269757C458224B7C895F35F1512E8EE@cantor.std.visionutv.se> Message-ID: Someone assign this to moy On Jul 5, 2012 6:06 PM, "Robin Gilks" wrote: > Done - FS-4395 - Segfault on E1 incoming call > > On Thu, Jul 5, 2012 at 7:08 PM, Peter Olsson > wrote: > > Please report to Jira, with all needed information > > > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > > > /Peter > > > > -----Ursprungligt meddelande----- > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] F?r Neo Cheema > > Skickat: den 5 juli 2012 07:53 > > Till: FreeSWITCH Users Help > > ?mne: Re: [Freeswitch-users] Crash on incoming E1 call > > > > Facing exactly the same issue. But in my case, freeswitch crashes at > startup itself while configuring spandsp. > > My stack is Feetdm E1 with libpri. > > > > On Thu, Jul 5, 2012 at 11:00 AM, Anton Kvashenkin < > anton.jugatsu at gmail.com> wrote: > >> Post this output to backtracker. > >> > >> 2012/7/5 Robin Gilks > >>> > >>> Current running from commit 04bd463d12bf4c0116bec3cee537749b2040ed40 > >>> > >>> This occurs on all incoming calls from a Digium Wildcard TE210P with > >>> a zaptel E1 interface with libpri. > >>> > >>> I've tried updating to dahdi-linux-complete-2.6.1+2.6.1.tar.gz but > >>> failed miserably - couldn't even get the E1 link itself to come up :( > >>> > >>> I've captured a backtrace of the event on the zaptel drivers, > >>> hopefully useful to someone. > >>> > >>> gdb /usr/local/freeswitch_TR/bin/freeswitch -x ${HOME}/gdbcommands > >>> > >>> > >>> $ cat gdbcommands > >>> handle SIGPIPE nostop noprint > >>> handle SIG33 nostop noprint > >>> set logging on > >>> set pagination off > >>> set breakpoint pending on > >>> run > >>> thread apply all bt full > >>> set logging off > >>> > >>> Attached is the output of the backtrace. > >>> > >>> > >>> > >>> -- > >>> Robin Gilks > >>> > >>> -- > >>> > >>> ------------------------------ > >>> This email, including any attachments, is only for the intended > recipient. > >>> It is subject to copyright, is confidential and may be the subject of > >>> legal or other privilege, none of which is waived or lost by reason > >>> of this transmission. > >>> If you are not an intended recipient, you may not use, disseminate, > >>> distribute or reproduce such email, any attachments, or any part > thereof. > >>> If you have received a message in error, please notify the sender > >>> immediately and erase all copies of the message and any attachments. > >>> Unfortunately, we cannot warrant that the email has not been altered > >>> or corrupted during transmission nor can we guarantee that any email > >>> or any attachments are free from computer viruses or other conditions > >>> which may damage or interfere with recipient data, hardware or > >>> software. The recipient relies upon its own procedures and assumes > >>> all risk of use and of opening any attachments. > >>> ------------------------------ > >>> > >>> _____________________________________________________________________ > >>> ____ Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> Join Us At ClueCon - Aug 7-9, 2012 > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us > >>> ers > >>> http://www.freeswitch.org > >>> > >> > >> > >> ______________________________________________________________________ > >> ___ Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users 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 > >> > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > !DSPAM:4ff5389332761306118382! > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -- > Robin Gilks > > -- > > ------------------------------ > This email, including any attachments, is only for the intended recipient. > It is subject to copyright, is confidential and may be the subject of legal > or other privilege, none of which is waived or lost by reason of this > transmission. > If you are not an intended recipient, you may not use, disseminate, > distribute or reproduce such email, any attachments, or any part thereof. > If you have received a message in error, please notify the sender > immediately and erase all copies of the message and any attachments. > Unfortunately, we cannot warrant that the email has not been altered or > corrupted during transmission nor can we guarantee that any email or any > attachments are free from computer viruses or other conditions which may > damage or interfere with recipient data, hardware or software. The > recipient relies upon its own procedures and assumes all risk of use and of > opening any attachments. > ------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120705/0f2cd9e5/attachment-0001.html From jmesquita at freeswitch.org Fri Jul 6 05:06:20 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Thu, 5 Jul 2012 22:06:20 -0300 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <9343F57A41054F8DB89BD4A0C17CFD7B@gmail.com> Message-ID: <0B4DC9FFE198449888339EA7BA4E3E24@freeswitch.org> You don't seem to have used Zmq. It's FUN!!! :D Besides, with ZMQ you get fan in/fan out as well as other types of socket that can REALLY increase cdr processing power (distributed processing). Of course, I agree that having it as a module might not be the best solution tho, since zmq has some nasty asserts on its core code. -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Thursday, July 5, 2012 at 6:59 PM, Avi Marcus wrote: > What about simply using xml_cdr or json_cdr and then process the queue of the files? Move them to a sub-directory once processed. That gives you disk persistence... > -Avi > > > On Fri, Jul 6, 2012 at 12:17 AM, Michael Collins wrote: > > Woohoo Perl! > > http://www.bastichlabz.org/bastich/Strips/ba980225.gif > > > > -MC > > > > > > On Thu, Jul 5, 2012 at 1:48 PM, Gavin Henry wrote: > > > We sponsored this: > > > > > > https://metacpan.org/release/Message-Passing > > > https://metacpan.org/release/Message-Passing-ZeroMQ > > > https://metacpan.org/module/BOBTFISH/Message-Passing-Input-Freeswitch-0.002/lib/Message/Passing/Input/Freeswitch.pm > > > https://metacpan.org/release/Message-Passing-Output-WebHooks > > > > > > We use it to pump events into zeromq as part of our API - http://www.surevoip.co.uk/api > > > > > > You can easily do cdrs. > > > > > > Gavin. > > > > > > -- > > > Gavin Henry > > > Sent with Sparrow (http://www.sparrowmailapp.com/?sig) > > > > > > > > > On Thursday, 5 July 2012 at 19:08, Michael Giagnocavo wrote: > > > > > > > > > > > Wouldn?t the problem with ZeroMQ be no reliable delivery of CDRs? I was under the impression that 0MQ was more for high-performance, and things like guaranteed delivery are left as an exercise to the user. > > > > > > > > > > > > -Michael > > > > > > > > > > > > > > > > > > > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org (mailto:freeswitch-users-bounces at lists.freeswitch.org) [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita > > > > Sent: Thursday, July 05, 2012 9:56 AM > > > > To: FreeSWITCH Users Help > > > > Subject: Re: [Freeswitch-users] Pushing CDR Information into Zeromq > > > > > > > > > > > > > > > > > > > > > > > > I haven't actually looked at the code, but you have to look at the variables on CHANNEL_REPORTING for example. There you will see all the vars needed to produce CDR entries, but they won't be on XML format like xml_curl would create. > > > > > > > > > > > > > > > > > > > > > > > > > > > > Regards, > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > > > > > > > > > > > > > > > Jo?o Mesquita > > > > > > > > > > > > > > > > Sent with Sparrow (http://www.sparrowmailapp.com/?sig) > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > On Thursday, July 5, 2012 at 10:37 AM, Neo Cheema wrote: > > > > > > > > > > Hi all, > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > I was hoping to find a way to push CDR information into a queue. > > > > > > > > > > > > > > > > > > > > Zeromq comes out as an obvious choice because mod_event_zmq already > > > > > > > > > > > > > > > > > > > > exits. However, I can't find a way to configure this module to push > > > > > > > > > > > > > > > > > > > > CDR info. Have any of you guys tried it? > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Regards > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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/20120705/39cae1dd/attachment.html From jaybinks at gmail.com Fri Jul 6 07:03:16 2012 From: jaybinks at gmail.com (jay binks) Date: Fri, 6 Jul 2012 13:03:16 +1000 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: <0B4DC9FFE198449888339EA7BA4E3E24@freeswitch.org> References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <9343F57A41054F8DB89BD4A0C17CFD7B@gmail.com> <0B4DC9FFE198449888339EA7BA4E3E24@freeswitch.org> Message-ID: while we are talking about funky CDR stuff. Ive been planning to look at riak ( http://wiki.basho.com/ ) for bulk CDR storage ( storing data other than billing data ) but just not finding the time. anyone done something similar ? The idea is to store as much data in a distributed manner, so as to aid debugging. ( not so much for billing ) Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120706/bb9a2c78/attachment.html From achandra at telrouter.com Fri Jul 6 05:40:13 2012 From: achandra at telrouter.com (Akhil Chandra) Date: Thu, 5 Jul 2012 18:40:13 -0700 Subject: [Freeswitch-users] Problem with Polycom behind firewall with freeswitch Message-ID: <005001cd5b18$4bc6b9b0$e3542d10$@com> Hi all, I come from a asterisk background and having trouble keeping the firewall hole opened up for Polycom registering to freeswitch server. I am running freeswitch ( with bluebox UI) and trying to register a Polycom behind a router. I have exhausted all the options in sip_profiles directory for the interface I am using. The issue I am seeing is that the freeswitch sends the OPTIONS message to the internal IP ie ( 192.168.X.X) rather than the public IP, behind which the Polycom phones are. These phones register fine with asterisk in the stated scenario. I have tried changing these options in the xml file but still the OPTIONS message ( as seen in tcpdump) is sent to 192.168.X.X Any help appreciated !! .. Thanks in advance. Regards, Akhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120705/24518e9f/attachment-0001.html From achandra at telrouter.com Fri Jul 6 06:28:10 2012 From: achandra at telrouter.com (Akhil Chandra) Date: Thu, 5 Jul 2012 19:28:10 -0700 Subject: [Freeswitch-users] Problem with Polycom behind firewall with freeswitch Message-ID: <005501cd5b1e$fd87bbd0$f8973370$@com> Hi all, I come from a asterisk background and having trouble keeping the firewall hole opened up for Polycom registering to freeswitch server. I am running freeswitch ( with bluebox UI) and trying to register a Polycom behind a router. I have exhausted all the options in sip_profiles directory for the interface I am using. The issue I am seeing is that the freeswitch sends the OPTIONS message to the internal IP ie ( 192.168.X.X) rather than the public IP, behind which the Polycom phones are. These phones register fine with asterisk in the stated scenario. I have tried changing these options in the xml file but still the OPTIONS message ( as seen in tcpdump) is sent to 192.168.X.X Any help appreciated !! .. Thanks in advance. Regards, Akhil -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120705/4962b7b6/attachment.html From ksims.ml at gmail.com Fri Jul 6 06:07:39 2012 From: ksims.ml at gmail.com (KPS Maillinglist) Date: Thu, 5 Jul 2012 21:07:39 -0500 Subject: [Freeswitch-users] Transfer using lua no bleg present Message-ID: Hi All, I have a issue where when i transfer a call using lua. It tells me that "[WARNING] mod_dptools.c:976 No B-leg present." Below you will find some background info: @@@@@@@@@@ Log: http://pastebin.freeswitch.org/19458 @@@@@@@@@@ lua file relevant potions ########## dx_script.lua ###################### session:setAutoHangup(false) session:answer(); session:set_tts_parms("cepstral", "amy"); -- session:execute("", "") digits = session:read(4, 11, "tone_stream://%(10000,0,350,440)", 5000, "#") write_FS_LOG("Got dtmf: ", digits) --freeswitch.consoleLog("info", "Got dtmf: ".. digits .."\n"); --session:execute("read", "11 11 'tone_stream://%(10000,0,350,440)' digits 5000 #") session:execute("export","orig_transfer_digits=" .. digits) session:setVariable("orig_transfer_digits", digits) session:setVariable("orig_user_context",usr_context) session:execute("export","orig_user_context=" .. usr_context) atoit = string.sub(dst_number, 4) session:execute("transfer", "is_transfer-" .. atoit .. " XML FEATURES") write_FS_LOG("Script End Time: ", os.date("%Y-%m-%d %X")) ############### is_transfer_script.lua ###################### if string.find(dst_number, "-") ~= nil then dst_num_context_str = string.split(dst_number,"-") for dncs_index,dncs_str in pairs(dst_num_context_str) do -- do some stuff write_FS_LOG("DCNS Index", string.format("dcns_ind%s", dncs_index)) write_FS_LOG("DCNS VAR", string.format("%s", dncs_str)) session:setVariable("dcns_ind" .. dncs_index, dncs_str) orig_ucon = dncs_str end end -- orig_ucon = session:getVariable("dcns_ind2") -- orig_ucon = dcns_ind2 write_FS_LOG("orig_ucon", orig_ucon) if orig_ucon == nil then write_FS_ERRLOG("Missing data", "Missing user originating context") write_FS_LOG("Script End Time", os.date("%Y-%m-%d %X")) session:hangup() else write_FS_LOG("Recieved Context - Continuing...", orig_ucon) end if session:getVariable("orig_transfer_digits") == nil then write_FS_ERRLOG("Missing data", "session:getVariable(orig_transfer_digits)") write_FS_LOG("Script End Time", os.date("%Y-%m-%d %X")) session:hangup() else write_FS_LOG("Recieved digits - Continuing...", session:getVariable("orig_transfer_digits")) end -- session:execute("info","") -- session:execute("info","") -- orig_user_context = session:getVariable("orig_user_context") orig_transfer_digits = session:getVariable("orig_transfer_digits") write_FS_LOG("Script: ", "regsrv is_transfer_script.lua") write_FS_LOG("Script Start Time: ", os.date("%Y-%m-%d %X")) session:setAutoHangup(false) session:answer(); session:set_tts_parms("cepstral", "amy"); -- session:execute("", "") session:execute("transfer", "-bleg " .. orig_transfer_digits .. " XML " .. orig_ucon) write_FS_LOG("Script End Time: ", os.date("%Y-%m-%d %X")) @@@@@@@@@@ dialplan
-------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120705/90974c97/attachment.html From all.eforums at gmail.com Fri Jul 6 08:02:55 2012 From: all.eforums at gmail.com (A E G) Date: Fri, 6 Jul 2012 00:02:55 -0400 Subject: [Freeswitch-users] Load Balancing In-Reply-To: References: Message-ID: On Wed, Jul 4, 2012 at 5:53 AM, Hanie Maghsoudy wrote: > Thanks Muhammad, that would be a good idea. > However, I'm trying direct routing mode of IPVS for load balancing along > with Heartbeat for HA. > Could anyone of you guys suggest a better way to implement Load balancing > and HA for FreeSwitch? > > Thanks > > Not sure about HA, but the call distributor module: http://wiki.freeswitch.org/wiki/Mod_distributor used to work well for us ...haven't used it for a while now. Not sure if anything has changed in newer builds. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120706/2970ba8d/attachment.html From msc at freeswitch.org Fri Jul 6 08:28:41 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 5 Jul 2012 21:28:41 -0700 Subject: [Freeswitch-users] Transfer using lua no bleg present In-Reply-To: References: Message-ID: I honestly don't mean to be a smart alleck, but the warning "No B-leg present" is exactly what is wrong. This is a one-legged call into a Lua script, therefore there is only the A-leg. That means this line is incorrect: session:execute("transfer", "-bleg " .. orig_transfer_digits .. " XML " .. orig_ucon) The session that the Lua script is controlling is not bridged to another leg (which would be the B-leg) and so there is no reason to try and transfer the B-leg. Perhaps you could just do a simple transfer? -MC On Thu, Jul 5, 2012 at 7:07 PM, KPS Maillinglist wrote: > Hi All, > I have a issue where when i transfer a call using lua. It tells me that > "[WARNING] mod_dptools.c:976 No B-leg present." > > Below you will find some background info: > > @@@@@@@@@@ Log: > http://pastebin.freeswitch.org/19458 > > @@@@@@@@@@ lua file relevant potions > ########## dx_script.lua ###################### > session:setAutoHangup(false) > > session:answer(); > session:set_tts_parms("cepstral", "amy"); > > -- session:execute("", "") > > digits = session:read(4, 11, "tone_stream://%(10000,0,350,440)", 5000, "#") > write_FS_LOG("Got dtmf: ", digits) > > --freeswitch.consoleLog("info", "Got dtmf: ".. digits .."\n"); > --session:execute("read", "11 11 'tone_stream://%(10000,0,350,440)' digits > 5000 #") > session:execute("export","orig_transfer_digits=" .. digits) > session:setVariable("orig_transfer_digits", digits) > session:setVariable("orig_user_context",usr_context) > session:execute("export","orig_user_context=" .. usr_context) > atoit = string.sub(dst_number, 4) > session:execute("transfer", "is_transfer-" .. atoit .. " XML FEATURES") > > > write_FS_LOG("Script End Time: ", os.date("%Y-%m-%d %X")) > > ############### is_transfer_script.lua ###################### > > if string.find(dst_number, "-") ~= nil then > dst_num_context_str = string.split(dst_number,"-") > for dncs_index,dncs_str in pairs(dst_num_context_str) do > -- do some stuff > write_FS_LOG("DCNS Index", string.format("dcns_ind%s", dncs_index)) > write_FS_LOG("DCNS VAR", string.format("%s", dncs_str)) > session:setVariable("dcns_ind" .. dncs_index, dncs_str) > orig_ucon = dncs_str > end > end > -- orig_ucon = session:getVariable("dcns_ind2") > -- orig_ucon = dcns_ind2 > write_FS_LOG("orig_ucon", orig_ucon) > > if orig_ucon == nil then > write_FS_ERRLOG("Missing data", "Missing user originating context") > write_FS_LOG("Script End Time", os.date("%Y-%m-%d %X")) > session:hangup() > else > write_FS_LOG("Recieved Context - Continuing...", orig_ucon) > end > > if session:getVariable("orig_transfer_digits") == nil then > write_FS_ERRLOG("Missing data", > "session:getVariable(orig_transfer_digits)") > write_FS_LOG("Script End Time", os.date("%Y-%m-%d %X")) > session:hangup() > else > write_FS_LOG("Recieved digits - Continuing...", > session:getVariable("orig_transfer_digits")) > end > > -- session:execute("info","") > -- session:execute("info","") > > -- orig_user_context = session:getVariable("orig_user_context") > orig_transfer_digits = session:getVariable("orig_transfer_digits") > > write_FS_LOG("Script: ", "regsrv is_transfer_script.lua") > write_FS_LOG("Script Start Time: ", os.date("%Y-%m-%d %X")) > session:setAutoHangup(false) > > session:answer(); > session:set_tts_parms("cepstral", "amy"); > > -- session:execute("", "") > > session:execute("transfer", "-bleg " .. orig_transfer_digits .. " XML " .. > orig_ucon) > > > write_FS_LOG("Script End Time: ", os.date("%Y-%m-%d %X")) > > > @@@@@@@@@@ dialplan
name="dialplan" description="FreeSWITCH Dialplan"> name="FEATURES"> field="destination_number" expression="^DX-.*"> application="lua" data="/fs/scripts/users/dx_script.lua"/> > > > data="/fs/scripts/users/is_transfer_script.lua"/> application="eval" data="cancel transfer"/> >
> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120705/6165bbe2/attachment-0001.html From shaheryarkh at googlemail.com Fri Jul 6 09:00:55 2012 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Fri, 6 Jul 2012 10:00:55 +0500 Subject: [Freeswitch-users] Load Balancing In-Reply-To: References: Message-ID: You can also try DNS based fail-over and load balancing, works perfect for us. Thank you. On Fri, Jul 6, 2012 at 9:02 AM, A E G wrote: > > On Wed, Jul 4, 2012 at 5:53 AM, Hanie Maghsoudy wrote: > >> Thanks Muhammad, that would be a good idea. >> However, I'm trying direct routing mode of IPVS for load balancing along >> with Heartbeat for HA. >> Could anyone of you guys suggest a better way to implement Load balancing >> and HA for FreeSwitch? >> >> Thanks >> >> > Not sure about HA, but the call distributor module: > http://wiki.freeswitch.org/wiki/Mod_distributor used to work well for us > ...haven't used it for a while now. Not sure if anything has changed in > newer builds. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120706/14895b59/attachment.html From jack.nikolas at ymail.com Fri Jul 6 09:40:50 2012 From: jack.nikolas at ymail.com (Jack Nikolas) Date: Fri, 6 Jul 2012 06:40:50 +0100 (BST) Subject: [Freeswitch-users] how to prevent the changes of nonce ttl sip auth ? Message-ID: <1341553250.66528.YahooMailNeo@web171504.mail.ir2.yahoo.com> HI, i find out every time we want to register on sipclient the sip_nonce_auth? get the value,and every time we unregister the same sipclient the value of sip_nonce_auth was changed ! but is there any solutions to prevent of changing the value of sip_nonce_auth for same sipclient that registered and unregistered? because suppose we have 25 sip client that is registered on our server,now i want to check the sip client that is registered be the same that is unregistered so if the value of sip_nonce_auth for the sipcliet be the equal with the value of the same sipclient that is unregister ,the problem will solve, in the internal.xml we have it means every 60sec the value of? sip_nonce_auth for each sipclient was chnaged , and if the value of? nonce-ttl is less than the value of? parametere "register every ... " softphone, so the nonce-ttl was changed again. but i don't want change to check every sip client that is registered is the same that is unregistered i want when the status of sipclient was changed to 'unregistered' form 'registered' the value of 'sip_nonce_auth' be the same? to find out the sip client that is registered was the same that is unregistered ! is there any solutions? ( i know we can check using fs_cli with sofia status profile internal reg and sofia_count_reg ..) but i want to check with sip header ... for example when sipclient is registered 'Event-Name: REQUEST_PARAMS Core-UUID: 86fdb0e4-c6cd-11e1-a155-c382209fd02d FreeSWITCH-Hostname: PBX FreeSWITCH-Switchname: PBX FreeSWITCH-IPv4: 192.168.10.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-07-06%2009%3A42%3A44 Event-Date-GMT: Fri,%2006%20Jul%202012%2005%3A12%3A44%20GMT Event-Date-Timestamp: 1341551564411676 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_parse_auth Event-Calling-Line-Number: 2364 Event-Sequence: 5610 action: sip_auth sip_profile: internal sip_user_agent: eyeBeam%20release%201102q%20stamp%2051814 sip_auth_username: jack sip_auth_realm: 192.168.10.89 sip_auth_nonce: 37cb2234-c729-11e1-a170-c382209fd02d sip_auth_uri: sip%3A192.168.10.1 sip_contact_user: jack sip_contact_host: 192.168.12.1 sip_to_user: jack sip_to_host: 192.168.10.1 sip_from_user: jack sip_from_host: 192.168.10.1 sip_request_host: 192.168.10.1 sip_auth_qop: auth sip_auth_cnonce: c3f4cce1becf9f7fb6b20bafc27e2c68 sip_auth_nc: 00000001 sip_auth_response: 5489fc3afb45d7a826b865eef559baf5 sip_auth_method: REGISTER key: id user: jack domain: 192.168.10.1 ip: 192.168.12.1 when the same sip client is unregistered: 'Event-Name: REQUEST_PARAMS Core-UUID: 86fdb0e4-c6cd-11e1-a155-c382209fd02d FreeSWITCH-Hostname: PBX FreeSWITCH-Switchname: PBX FreeSWITCH-IPv4: 192.168.10.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-07-06%2009%3A49%3A56 Event-Date-GMT: Fri,%2006%20Jul%202012%2005%3A19%3A56%20GMT Event-Date-Timestamp: 1341551996651662 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_parse_auth Event-Calling-Line-Number: 2364 Event-Sequence: 5610 action: sip_auth sip_profile: internal sip_user_agent: eyeBeam%20release%201102q%20stamp%2051814 sip_auth_username:jack sip_auth_realm: 192.168.10.89 sip_auth_nonce:397d1b86-c72a-11e1-a171-c382209fd02d sip_auth_uri: sip%3A192.168.10.1 sip_contact_user:jack sip_contact_host: 192.168.12.1 sip_to_user: jack sip_to_host: 192.168.10.1 sip_from_user: jack sip_from_host: 192.168.10.1 sip_request_host: 192.168.10.1 sip_auth_qop: auth sip_auth_cnonce: c3f4cce1becf9f7fb6b20bafc27e2c68 sip_auth_nc: 00000001 sip_auth_response: 5489fc3afb45d7a826b865eef559baf5 sip_auth_method: REGISTER key: id user: jack domain: 192.168.10.1 ip: 192.168.12.1 as you see the sip_auth_method is the same when user is registered AND unregistered ... but how can i findout? that ? is there any method to change the 'sip_auth_method ' to another value when sipclient status was changed to 'unregistered'? thanks in advanced -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120706/a3c659c9/attachment.html From neo.cheema at gmail.com Fri Jul 6 11:14:08 2012 From: neo.cheema at gmail.com (Neo Cheema) Date: Fri, 6 Jul 2012 12:44:08 +0530 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <9343F57A41054F8DB89BD4A0C17CFD7B@gmail.com> <0B4DC9FFE198449888339EA7BA4E3E24@freeswitch.org> Message-ID: Jo?o, "Since the socket implemented on mod_zmq is a simple PUB/SUB, yes, messages would be discarded if they are not received by the other end." I thought it didn't matter if the 'worker' was up before the 'producer'/freeswitch started pushing in the messages. 0mq handles that part, if atleast one worker is connected. My main reason for looking towards 0mq was to bypass a webserver, which would be required for xml_cdr. On Fri, Jul 6, 2012 at 8:33 AM, jay binks wrote: > while we are talking about funky CDR stuff. > > Ive been planning to look at riak ( http://wiki.basho.com/ ) for bulk CDR > storage > ( storing data other than billing data ) but just not finding the time. > > anyone done something similar ? > The idea is to store as much data in a distributed manner, so as to aid > debugging. > ( not so much for billing ) > > Jay > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From avi at avimarcus.net Fri Jul 6 11:48:02 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 6 Jul 2012 10:48:02 +0300 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <9343F57A41054F8DB89BD4A0C17CFD7B@gmail.com> <0B4DC9FFE198449888339EA7BA4E3E24@freeswitch.org> Message-ID: xml_cdr by default writes to disk, no web server required. You want to skip the disk, I presume..? -Avi On Fri, Jul 6, 2012 at 10:14 AM, Neo Cheema wrote: > Jo?o, > > "Since the socket implemented on mod_zmq is a simple PUB/SUB, yes, > messages would be discarded if they are not received by the other > end." > > I thought it didn't matter if the 'worker' was up before the > 'producer'/freeswitch started pushing in the messages. 0mq handles > that part, if atleast one worker is connected. > > My main reason for looking towards 0mq was to bypass a webserver, > which would be required for xml_cdr. > > On Fri, Jul 6, 2012 at 8:33 AM, jay binks wrote: > > while we are talking about funky CDR stuff. > > > > Ive been planning to look at riak ( http://wiki.basho.com/ ) for bulk > CDR > > storage > > ( storing data other than billing data ) but just not finding the time. > > > > anyone done something similar ? > > The idea is to store as much data in a distributed manner, so as to aid > > debugging. > > ( not so much for billing ) > > > > Jay > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120706/a8b3bc6e/attachment-0001.html From neo.cheema at gmail.com Fri Jul 6 12:10:08 2012 From: neo.cheema at gmail.com (Neo Cheema) Date: Fri, 6 Jul 2012 13:40:08 +0530 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <9343F57A41054F8DB89BD4A0C17CFD7B@gmail.com> <0B4DC9FFE198449888339EA7BA4E3E24@freeswitch.org> Message-ID: Avi, What I really want is to provide near-real time updates of CDRs on a webpage, the webserver not necessarily being on the same machine as Freeswitch. I would hate to continuously scan DB on freeswitch machine for latest CDR info. In a related issue, I read that the only reliable info on "call length", billsec etc should be read from CDRs. I mean, I can't do a getvariable on session in say a Lua script, to read call length, billsec session variables. Infact I tried it and got null responses, even after session hangup. Is there a way to get "correct" values of these variables in the Lua script itself? It would make my life simpler, because I can then just push the cdr info + custom values into any queue or webserver, without waiting for the Freeswitch to do it for me. Any Ideas? On Fri, Jul 6, 2012 at 1:18 PM, Avi Marcus wrote: > xml_cdr by default writes to disk, no web server required. You want to skip > the disk, I presume..? > > -Avi > > > On Fri, Jul 6, 2012 at 10:14 AM, Neo Cheema wrote: >> >> Jo?o, >> >> "Since the socket implemented on mod_zmq is a simple PUB/SUB, yes, >> messages would be discarded if they are not received by the other >> end." >> >> I thought it didn't matter if the 'worker' was up before the >> 'producer'/freeswitch started pushing in the messages. 0mq handles >> that part, if atleast one worker is connected. >> >> My main reason for looking towards 0mq was to bypass a webserver, >> which would be required for xml_cdr. >> >> On Fri, Jul 6, 2012 at 8:33 AM, jay binks wrote: >> > while we are talking about funky CDR stuff. >> > >> > Ive been planning to look at riak ( http://wiki.basho.com/ ) for bulk >> > CDR >> > storage >> > ( storing data other than billing data ) but just not finding the time. >> > >> > anyone done something similar ? >> > The idea is to store as much data in a distributed manner, so as to aid >> > debugging. >> > ( not so much for billing ) >> > >> > Jay >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From royj at yandex.ru Fri Jul 6 12:12:34 2012 From: royj at yandex.ru (royj) Date: Fri, 6 Jul 2012 12:12:34 +0400 Subject: [Freeswitch-users] twin events MESSAGE (mod_gsmopen, sms_incoming) In-Reply-To: References: <20120705125711.b2bfb0d7.royj@yandex.ru> <9C206E75-749E-4810-8C70-756615735117@gmail.com> Message-ID: <20120706121234.87e31766.royj@yandex.ru> Thank you for your answer Really, in my case there is no need to use mod_sms and chatplan, I unload mod_sms and have only one event MESSAGE (why I decided that I need to use mod_sms?) On Thu, 5 Jul 2012 17:13:10 +0200 Giovanni Maruzzelli wrote: > On Thu, Jul 5, 2012 at 5:07 PM, Giovanni Maruzzelli wrote: > > > On Thu, Jul 5, 2012 at 4:50 PM, dujinfang wrote: > > > >> I saw this too but I think if you don't fire or send in chatplan then the > >> message should be only one? > >> > >> > > Ciao Seven! > > Yep, I believe the same (I just associate the chatplan with firing or > > sending, stupid me). Not tested, tough. > > > > > Confirmed, just tested. If you do not have send or fire in chatplan, no > additional message will be on esl. > > -giovanni > > > > -- > Sincerely, > > Giovanni Maruzzelli > Cell : +39-347-2665618 -- Regards, royj From avi at avimarcus.net Fri Jul 6 12:37:42 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 6 Jul 2012 11:37:42 +0300 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <9343F57A41054F8DB89BD4A0C17CFD7B@gmail.com> <0B4DC9FFE198449888339EA7BA4E3E24@freeswitch.org> Message-ID: I process CDRs on the same machine and then replicate the DB elsewhere. Or you could have a daemon processing the CDRs on the main machine and doing whatever with them. The benefit is that then you'll have file persistence of the CDRs until they actually get processed... -Avi On Fri, Jul 6, 2012 at 11:10 AM, Neo Cheema wrote: > Avi, > > What I really want is to provide near-real time updates of CDRs on a > webpage, the webserver not necessarily being on the same machine as > Freeswitch. I would hate to continuously scan DB on freeswitch machine > for latest CDR info. > > In a related issue, I read that the only reliable info on "call > length", billsec etc should be read from CDRs. > > I mean, I can't do a getvariable on session in say a Lua script, to > read call length, billsec session variables. Infact I tried it and got > null responses, even after session hangup. Is there a way to get > "correct" values of these variables in the Lua script itself? It would > make my life simpler, because I can then just push the cdr info + > custom values into any queue or webserver, without waiting for the > Freeswitch to do it for me. > > Any Ideas? > > > On Fri, Jul 6, 2012 at 1:18 PM, Avi Marcus wrote: > > xml_cdr by default writes to disk, no web server required. You want to > skip > > the disk, I presume..? > > > > -Avi > > > > > > On Fri, Jul 6, 2012 at 10:14 AM, Neo Cheema > wrote: > >> > >> Jo?o, > >> > >> "Since the socket implemented on mod_zmq is a simple PUB/SUB, yes, > >> messages would be discarded if they are not received by the other > >> end." > >> > >> I thought it didn't matter if the 'worker' was up before the > >> 'producer'/freeswitch started pushing in the messages. 0mq handles > >> that part, if atleast one worker is connected. > >> > >> My main reason for looking towards 0mq was to bypass a webserver, > >> which would be required for xml_cdr. > >> > >> On Fri, Jul 6, 2012 at 8:33 AM, jay binks wrote: > >> > while we are talking about funky CDR stuff. > >> > > >> > Ive been planning to look at riak ( http://wiki.basho.com/ ) for bulk > >> > CDR > >> > storage > >> > ( storing data other than billing data ) but just not finding the > time. > >> > > >> > anyone done something similar ? > >> > The idea is to store as much data in a distributed manner, so as to > aid > >> > debugging. > >> > ( not so much for billing ) > >> > > >> > Jay > >> > > >> > > >> > > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > Join Us At ClueCon - Aug 7-9, 2012 > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120706/fa1c0215/attachment.html From neo.cheema at gmail.com Fri Jul 6 13:15:42 2012 From: neo.cheema at gmail.com (Neo Cheema) Date: Fri, 6 Jul 2012 14:45:42 +0530 Subject: [Freeswitch-users] How to get hangup_time, billsec variables inside script Message-ID: Hi All, I'm using Lua script to handle calls. For logging purposes, I would want to get hangup_time, billsec info in the script. However, the code doesn't work as expected. local new_session = freeswitch.Session(call_to_create_new_session, session) if new_session:ready() then new_session:streamFile("/home/ubuntu/sounds/ward5_1.wav") new_session:hangup(); local created_time = new_session:getVariable('created_time') //Correct value received local answered_time = new_session:getVariable('answered_time') // Correct value receive local hangup_time = new_session:getVariable('hangup_time') // NULL received. Problem!! Is there anyway to get this info from Inside the script? I understand that I can get this info from xml_cdr etc, but is it possible to get these values inside the script itself? Regards From neo.cheema at gmail.com Fri Jul 6 13:42:01 2012 From: neo.cheema at gmail.com (Neo Cheema) Date: Fri, 6 Jul 2012 15:12:01 +0530 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <9343F57A41054F8DB89BD4A0C17CFD7B@gmail.com> <0B4DC9FFE198449888339EA7BA4E3E24@freeswitch.org> Message-ID: Avi, Yea. That would work as well. I just thought it would be a 'nice-to-have' feature. Guess I'll go ahead with your model. Regards On Fri, Jul 6, 2012 at 2:07 PM, Avi Marcus wrote: > I process CDRs on the same machine and then replicate the DB elsewhere. > Or you could have a daemon processing the CDRs on the main machine and doing > whatever with them. > The benefit is that then you'll have file persistence of the CDRs until they > actually get processed... > > -Avi > > > On Fri, Jul 6, 2012 at 11:10 AM, Neo Cheema wrote: >> >> Avi, >> >> What I really want is to provide near-real time updates of CDRs on a >> webpage, the webserver not necessarily being on the same machine as >> Freeswitch. I would hate to continuously scan DB on freeswitch machine >> for latest CDR info. >> >> In a related issue, I read that the only reliable info on "call >> length", billsec etc should be read from CDRs. >> >> I mean, I can't do a getvariable on session in say a Lua script, to >> read call length, billsec session variables. Infact I tried it and got >> null responses, even after session hangup. Is there a way to get >> "correct" values of these variables in the Lua script itself? It would >> make my life simpler, because I can then just push the cdr info + >> custom values into any queue or webserver, without waiting for the >> Freeswitch to do it for me. >> >> Any Ideas? >> >> >> On Fri, Jul 6, 2012 at 1:18 PM, Avi Marcus wrote: >> > xml_cdr by default writes to disk, no web server required. You want to >> > skip >> > the disk, I presume..? >> > >> > -Avi >> > >> > >> > On Fri, Jul 6, 2012 at 10:14 AM, Neo Cheema >> > wrote: >> >> >> >> Jo?o, >> >> >> >> "Since the socket implemented on mod_zmq is a simple PUB/SUB, yes, >> >> messages would be discarded if they are not received by the other >> >> end." >> >> >> >> I thought it didn't matter if the 'worker' was up before the >> >> 'producer'/freeswitch started pushing in the messages. 0mq handles >> >> that part, if atleast one worker is connected. >> >> >> >> My main reason for looking towards 0mq was to bypass a webserver, >> >> which would be required for xml_cdr. >> >> >> >> On Fri, Jul 6, 2012 at 8:33 AM, jay binks wrote: >> >> > while we are talking about funky CDR stuff. >> >> > >> >> > Ive been planning to look at riak ( http://wiki.basho.com/ ) for bulk >> >> > CDR >> >> > storage >> >> > ( storing data other than billing data ) but just not finding the >> >> > time. >> >> > >> >> > anyone done something similar ? >> >> > The idea is to store as much data in a distributed manner, so as to >> >> > aid >> >> > debugging. >> >> > ( not so much for billing ) >> >> > >> >> > Jay >> >> > >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ >> >> > Professional FreeSWITCH Consulting Services: >> >> > consulting at freeswitch.org >> >> > http://www.freeswitchsolutions.com >> >> > >> >> > >> >> > >> >> > >> >> > Official FreeSWITCH Sites >> >> > http://www.freeswitch.org >> >> > http://wiki.freeswitch.org >> >> > http://www.cluecon.com >> >> > >> >> > Join Us At ClueCon - Aug 7-9, 2012 >> >> > >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> > http://www.freeswitch.org >> >> > >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From avi at avimarcus.net Fri Jul 6 14:02:07 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 6 Jul 2012 13:02:07 +0300 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <9343F57A41054F8DB89BD4A0C17CFD7B@gmail.com> <0B4DC9FFE198449888339EA7BA4E3E24@freeswitch.org> Message-ID: I'm pretty sure there's a variable to have the CDR in the ESL events, but I can't find it. -Avi On Fri, Jul 6, 2012 at 12:42 PM, Neo Cheema wrote: > Avi, > Yea. That would work as well. I just thought it would be a > 'nice-to-have' feature. Guess I'll go ahead with your model. > > Regards > > On Fri, Jul 6, 2012 at 2:07 PM, Avi Marcus wrote: > > I process CDRs on the same machine and then replicate the DB elsewhere. > > Or you could have a daemon processing the CDRs on the main machine and > doing > > whatever with them. > > The benefit is that then you'll have file persistence of the CDRs until > they > > actually get processed... > > > > -Avi > > > > > > On Fri, Jul 6, 2012 at 11:10 AM, Neo Cheema > wrote: > >> > >> Avi, > >> > >> What I really want is to provide near-real time updates of CDRs on a > >> webpage, the webserver not necessarily being on the same machine as > >> Freeswitch. I would hate to continuously scan DB on freeswitch machine > >> for latest CDR info. > >> > >> In a related issue, I read that the only reliable info on "call > >> length", billsec etc should be read from CDRs. > >> > >> I mean, I can't do a getvariable on session in say a Lua script, to > >> read call length, billsec session variables. Infact I tried it and got > >> null responses, even after session hangup. Is there a way to get > >> "correct" values of these variables in the Lua script itself? It would > >> make my life simpler, because I can then just push the cdr info + > >> custom values into any queue or webserver, without waiting for the > >> Freeswitch to do it for me. > >> > >> Any Ideas? > >> > >> > >> On Fri, Jul 6, 2012 at 1:18 PM, Avi Marcus wrote: > >> > xml_cdr by default writes to disk, no web server required. You want to > >> > skip > >> > the disk, I presume..? > >> > > >> > -Avi > >> > > >> > > >> > On Fri, Jul 6, 2012 at 10:14 AM, Neo Cheema > >> > wrote: > >> >> > >> >> Jo?o, > >> >> > >> >> "Since the socket implemented on mod_zmq is a simple PUB/SUB, yes, > >> >> messages would be discarded if they are not received by the other > >> >> end." > >> >> > >> >> I thought it didn't matter if the 'worker' was up before the > >> >> 'producer'/freeswitch started pushing in the messages. 0mq handles > >> >> that part, if atleast one worker is connected. > >> >> > >> >> My main reason for looking towards 0mq was to bypass a webserver, > >> >> which would be required for xml_cdr. > >> >> > >> >> On Fri, Jul 6, 2012 at 8:33 AM, jay binks > wrote: > >> >> > while we are talking about funky CDR stuff. > >> >> > > >> >> > Ive been planning to look at riak ( http://wiki.basho.com/ ) for > bulk > >> >> > CDR > >> >> > storage > >> >> > ( storing data other than billing data ) but just not finding the > >> >> > time. > >> >> > > >> >> > anyone done something similar ? > >> >> > The idea is to store as much data in a distributed manner, so as to > >> >> > aid > >> >> > debugging. > >> >> > ( not so much for billing ) > >> >> > > >> >> > Jay > >> >> > > >> >> > > >> >> > > >> >> > > >> >> > > _________________________________________________________________________ > >> >> > Professional FreeSWITCH Consulting Services: > >> >> > consulting at freeswitch.org > >> >> > http://www.freeswitchsolutions.com > >> >> > > >> >> > > >> >> > > >> >> > > >> >> > Official FreeSWITCH Sites > >> >> > http://www.freeswitch.org > >> >> > http://wiki.freeswitch.org > >> >> > http://www.cluecon.com > >> >> > > >> >> > Join Us At ClueCon - Aug 7-9, 2012 > >> >> > > >> >> > FreeSWITCH-users mailing list > >> >> > FreeSWITCH-users at lists.freeswitch.org > >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > > >> >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> > http://www.freeswitch.org > >> >> > > >> >> > >> >> > >> >> > _________________________________________________________________________ > >> >> Professional FreeSWITCH Consulting Services: > >> >> consulting at freeswitch.org > >> >> http://www.freeswitchsolutions.com > >> >> > >> >> > >> >> > >> >> > >> >> Official FreeSWITCH Sites > >> >> http://www.freeswitch.org > >> >> http://wiki.freeswitch.org > >> >> http://www.cluecon.com > >> >> > >> >> Join Us At ClueCon - Aug 7-9, 2012 > >> >> > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> > > >> > > >> > > >> > > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > Join Us At ClueCon - Aug 7-9, 2012 > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120706/831b84a4/attachment-0001.html From richard at klingler.net Fri Jul 6 15:37:12 2012 From: richard at klingler.net (Richard Klingler) Date: Fri, 6 Jul 2012 13:37:12 +0200 Subject: [Freeswitch-users] Disable reinvite In-Reply-To: References: <20120702125026817891.9461489f@klingler.net> Message-ID: <20120706133712335045.de3036fd@klingler.net> Hmm... Doesn't help setting: Seems to be som eother problem... Internal phones are on network 10.0/16 and connect to FS via internal network... But as soon RTP traffic is flowing, RTP traffic goes out via internet gateway and back in into FS via its WAN IP. (Internet FW and FS WAN are one same /29 subnet). So somehow FS thinks the internal phones have a public IP and not RFC1918. On Mon, 2 Jul 2012 18:57:18 +0400, Yuriy Nasida wrote: > You have to use bypass_media for it. > > http://wiki.freeswitch.org/wiki/Bypass_Media > > > > >> Date: Mon, 2 Jul 2012 12:50:26 +0200 >> From: richard at klingler.net >> To: freeswitch-users at lists.freeswitch.org >> Subject: [Freeswitch-users] Disable reinvite >> >> Hello >> >> Just switched this weekend from asterisk to freeswitch on freebsd 9.0. >> And so far I was able to implement basic dialplan for > inbound/outbound calls (o; >> >> What I'm trying to find is a similar configuration option from > asterisk sip.conf >> where individual user agents can have SIP reinvite disbaled with > "canreinvite=no". >> >> Fomr the Freeswitch documentation it states that all RTP streams > are passed through >> by default, but I still see SIP clients trying to send UDP streams > to my box directly. >> >> Also setting "" in > sofia.conf.xml >> doesn't solve this problem. I just want the UDP streams frmo > outside are only allowed >> from my SIP trunk provider. >> >> >> thanx in advance >> richard >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From luis.daniel.lucio at gmail.com Fri Jul 6 15:55:10 2012 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Fri, 6 Jul 2012 06:55:10 -0500 Subject: [Freeswitch-users] How to send a tigger and monitoring Message-ID: Allo! Just Wondering if there is an easy way to do this: a) when we got an outgoing call that meets a regexp to send a tigger, like an email or a nagios push alert b) how to measure a counter for concurrent calls,and total calls per day. I'm in the willling to do a nagios pluging for this but i i dont know how to grab data. Regards, LD From peter.olsson at visionutveckling.se Fri Jul 6 16:11:13 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 6 Jul 2012 12:11:13 +0000 Subject: [Freeswitch-users] How to send a tigger and monitoring Message-ID: <1FFF97C269757C458224B7C895F35F15130F5F@cantor.std.visionutv.se> ESL is probably the best approach. Read more here; http://wiki.freeswitch.org/wiki/Esl http://wiki.freeswitch.org/wiki/Mod_event_socket /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Luis Daniel Lucio Quiroz Skickat: den 6 juli 2012 13:55 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] How to send a tigger and monitoring Allo! Just Wondering if there is an easy way to do this: a) when we got an outgoing call that meets a regexp to send a tigger, like an email or a nagios push alert b) how to measure a counter for concurrent calls,and total calls per day. I'm in the willling to do a nagios pluging for this but i i dont know how to grab data. Regards, LD _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff6d16a32761304020267! From neo.cheema at gmail.com Fri Jul 6 16:22:42 2012 From: neo.cheema at gmail.com (Neo Cheema) Date: Fri, 6 Jul 2012 17:52:42 +0530 Subject: [Freeswitch-users] How to send a tigger and monitoring In-Reply-To: References: Message-ID: Hello Luis, a) when we got an outgoing call that meets a regexp to send a tigger, like an email or a nagios push alert You can set an application in any language of your preference which checks the regex and sends an alert. Then chage your "outcall" dialplan, to call this application whenever an outbound call takes place. b) how to measure a counter for concurrent calls,and total calls per day. I'm in the willling to do a nagios pluging for this but i i dont know how to grab data. Use mod_cdv_csv to push data into a database. At the end of day, you can query the database. Hope it helped. On Fri, Jul 6, 2012 at 5:25 PM, Luis Daniel Lucio Quiroz wrote: > Allo! > > Just Wondering if there is an easy way to do this: > > a) when we got an outgoing call that meets a regexp to send a tigger, > like an email or a nagios push alert > b) how to measure a counter for concurrent calls,and total calls per > day. I'm in the willling to do a nagios pluging for this but i i dont > know how to grab data. > > Regards, > > LD > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From anita.hall at simmortel.com Fri Jul 6 17:06:23 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Fri, 6 Jul 2012 18:36:23 +0530 Subject: [Freeswitch-users] bridge: early media: call waiting Message-ID: Hi I have been using early media fail-over while bridging the call to multiple endpoints. For example, this is taken from the FreeSWITCH cookbook and works fine for user_busy and destination_out_of_order. {ignore_early_media=true,originate_continue_on_timeout=true,call_timout=60,monitor_early_media_fail=user_busy:2:480+620!destination_out_of_order:2:1776.7} But, it does not work when the telco sends a "call waiting" frequency. What should be the option for call waiting? TIA. regards, Anita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120706/d9ce4370/attachment.html From hakkie42 at gmail.com Fri Jul 6 12:15:06 2012 From: hakkie42 at gmail.com (Jim) Date: Fri, 06 Jul 2012 10:15:06 +0200 Subject: [Freeswitch-users] Getting started wiki page & external gateway: do I understand correctly? In-Reply-To: References: <4FF5455D.4020004@gmail.com> Message-ID: <4FF69E8A.9040406@gmail.com> Thanks - glad I understood. Comments inline. On 5-7-2012 17:08, curriegrad2004 wrote: > Actually I'd say not to modify the wiki at all. It is valid as it > stands. Well, that'll certainly take the least effort required ;) > Typically most users who use this default diaplan will > register their SIP provider's gateways on the external profile and all > of their phones in the internal profile. Got that.... personnally would like to see some more (even redundant) explanation of why things are done the way they are so people can understand the concepts more easily and leaving the fact that no phones are registered via the external profile more or less unspoken doesn't really help understanding IMO... but: if you're fine with it, not modifying the wiki doesn't take any effort, so I'll leave it ;) (Yes, I'm the kind of guy that is prepared to write multiple posts about one single word... and proud of it ;) > Regarding the security part, well said over at that front. * users > have been burned because of this. You can also set the default > dialplan for that authenticated profile to be on public and set your > users's user_context param to whatever context you want the user to be > on. Ok, got it. > And the answer to your last question. Yes you sure as heck as can > create a new sofia sip profile just for phones to register from the > outside world. Ok, thanks ;) Did that, now going on to create users and remove some more default configs. Next up: testing with LAN ATA, softphones and with double NAT: phone-------------------phone ... thinking I'm going to need NDLB-connectile-dysfunction there... ... but I'll certainly post again if I can't find my way out of trouble. Thanks for the reply! > On Thu, Jul 5, 2012 at 12:42 AM, Jim wrote: >> Hi list, >> >> More or less beginner, home situation; trying to set up fs (again).. >> >> I've got 2 questions: >> >> 1. Posting here to make sure I understand before modifying the wiki. >> Could you please correct me if I'm wrong? >> >> Over here >> http://wiki.freeswitch.org/wiki/Getting_Started_Guide#External >> >> it says >> "The External (formerly "outbound") profile handles outbound >> registrations to a SIP provider." >> However, earlier on it also mentions you can let external devices (i.e. >> user phones in their own networks) register with that profile... so I'd >> change this to: >> "The External (formerly "outbound") profile also handles outbound >> registrations to a SIP provider." >> >> Then this: >> "The external profile allows anonymous calling, which is required as >> your provider will never authenticate with you to send you a call." >> >> Skimmed through the bridge book p78, Receiving calls, which seems to >> confirm external profile does not require authentication. >> Ok, fine. >> >> Then this: >> "In order to secure your FreeSWITCH it is wise to link your outbound >> profile to a dialplan context other than 'default', which in the default >> configuration is the where authenticated users are placed." >> Seems this advice mixes a default situation (default dialplan being >> sensitive) with conditional advice (your outbound profile which would be >> external in a default config). >> >> I would change outbound to external in order to lessen confusion: >> "In order to secure your FreeSWITCH it is wise to link your exgternal >> profile to a dialplan context other than 'default', which in the default >> configuration is where authenticated users are placed." >> >> ... although what is probably really meant is something like: >> "As mentioned, the profile used for outbound registrations allows >> anonymous, unauthenticated calling. By default, this profile is the >> external profile. In order to secure your FreeSWITCH, don't link this >> profile to a dialplan that allows dialing paid numbers or dialing users >> (who may be bothered/harrassed) without any further checking. >> >> Summary: in a default configuration: don't link your external profile to >> a 'default' dialplan." >> ... which is a mouthful. >> >> 2. Given the above, if I want to have external users in their own >> network behind NAT register to me, it would be best if I define an >> additional profile that does require SIP authentication, right? >> >> I can then use the external profile to register with SIP trunks etc. From h.maghsoudy at gmail.com Fri Jul 6 20:03:55 2012 From: h.maghsoudy at gmail.com (Hanie Maghsoudy) Date: Fri, 6 Jul 2012 20:33:55 +0430 Subject: [Freeswitch-users] Load Balancing In-Reply-To: References: Message-ID: Thanks, But I think distributor module works for routing calls after FS receives them. I plan to have a cluster of FreeSwitches, so that I can balance the load of my incoming calls. On Fri, Jul 6, 2012 at 8:32 AM, A E G wrote: > > On Wed, Jul 4, 2012 at 5:53 AM, Hanie Maghsoudy wrote: > >> Thanks Muhammad, that would be a good idea. >> However, I'm trying direct routing mode of IPVS for load balancing along >> with Heartbeat for HA. >> Could anyone of you guys suggest a better way to implement Load balancing >> and HA for FreeSwitch? >> >> Thanks >> >> > Not sure about HA, but the call distributor module: > http://wiki.freeswitch.org/wiki/Mod_distributor used to work well for us > ...haven't used it for a while now. Not sure if anything has changed in > newer builds. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120706/407639bc/attachment-0001.html From h.maghsoudy at gmail.com Fri Jul 6 20:09:22 2012 From: h.maghsoudy at gmail.com (Hanie Maghsoudy) Date: Fri, 6 Jul 2012 20:39:22 +0430 Subject: [Freeswitch-users] Load Balancing In-Reply-To: References: Message-ID: DNS SRV is a good way too, but its selection is kind of static (round robin and wight) and doesn't care about the servers' load: http://wiki.freeswitch.org/wiki/Enterprise_deployment_DNS_SRV Thanks On Fri, Jul 6, 2012 at 9:30 AM, Muhammad Shahzad wrote: > You can also try DNS based fail-over and load balancing, works perfect for > us. > > Thank you. > > > On Fri, Jul 6, 2012 at 9:02 AM, A E G wrote: > >> >> On Wed, Jul 4, 2012 at 5:53 AM, Hanie Maghsoudy wrote: >> >>> Thanks Muhammad, that would be a good idea. >>> However, I'm trying direct routing mode of IPVS for load balancing along >>> with Heartbeat for HA. >>> Could anyone of you guys suggest a better way to implement Load >>> balancing and HA for FreeSwitch? >>> >>> Thanks >>> >>> >> Not sure about HA, but the call distributor module: >> http://wiki.freeswitch.org/wiki/Mod_distributor used to work well for us >> ...haven't used it for a while now. Not sure if anything has changed in >> newer builds. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120706/e9f587c3/attachment.html From adam at vrs.pl Fri Jul 6 20:04:04 2012 From: adam at vrs.pl (Adam Obuchowski) Date: Fri, 6 Jul 2012 18:04:04 +0200 Subject: [Freeswitch-users] Load Balancing In-Reply-To: References: Message-ID: Hello, What you mean by DNS based failover ? Short ttl on DNS A record may cause name resolution issues, isnt it ? How does it work ? Kind regards, -- Adam Obuchowski VoipSwitch support www.voipswitch.com 2012/7/6 Muhammad Shahzad > You can also try DNS based fail-over and load balancing, works perfect for > us. > > Thank you. > > > On Fri, Jul 6, 2012 at 9:02 AM, A E G wrote: > >> >> On Wed, Jul 4, 2012 at 5:53 AM, Hanie Maghsoudy wrote: >> >>> Thanks Muhammad, that would be a good idea. >>> However, I'm trying direct routing mode of IPVS for load balancing along >>> with Heartbeat for HA. >>> Could anyone of you guys suggest a better way to implement Load >>> balancing and HA for FreeSwitch? >>> >>> Thanks >>> >>> >> Not sure about HA, but the call distributor module: >> http://wiki.freeswitch.org/wiki/Mod_distributor used to work well for us >> ...haven't used it for a while now. Not sure if anything has changed in >> newer builds. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120706/08c172fa/attachment.html From anthony.minessale at gmail.com Fri Jul 6 20:54:15 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Jul 2012 11:54:15 -0500 Subject: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a In-Reply-To: References: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern> <4FED4DA1.10304@belrosbank.by> Message-ID: Like I said "Really Really Nice Motherboard", as many CPU as possible/affordable, and a good chunk of RAM. Motherboard is the most important, a cheap motherboard with a ton of cores is a waste. On Thu, Jul 5, 2012 at 3:27 PM, A E G wrote: > Thanks for the short/straight pointers re: what matters most for freeswitch > performance. > > Are there any equivalent pearls of wisdom "best practices" re: CPU, RAM and > Shared Memory/Bus that you or others can advise on, if this was to run in a > virtual environment in a private cloud of Xen Server, KVM or vCloud etc? > Probably it will depend on the underlying hardware or not? kernel tuning of > the Linux guest as well as resource sharing? > > Any light you or anyone else can shed on this would be greatly appreciated > > Thanks in advance > > On Fri, Jun 29, 2012 at 1:07 PM, Anthony Minessale > wrote: >> >> you could do with more than 2 cpu, the more cores the better you will do. >> For similar prices you could find a 8 or 12 core box. >> >> The 3 most important things in a FS server are, as many CPU cores as >> possible, as much ram as possible, the best possible CPU with shared >> bus to RAM and CPU. >> >> >> >> 2012/6/29 Eugene Shcherbatyuk : >> > Session limit is configurable. Have a look at switch.conf.xml file. >> > >> > >> > On 28/06/12 21:41, Hoexum, Edwin wrote: >> >> >> >> I am trying to do some load testing on a freeswitch for transcoding >> >> from >> >> SILK to G711a. The question is: Is the limit for RTP sessions >> >> limited to CPU power or are there also limits on the freeswitch >> >> application and can I tune freeswitch to do the job for 1500 sessions >> >> or >> >> more. >> > >> > >> > ========================================================= >> > >> > ?????? ????????? ? ????? ???????? (<>) ???????? >> > ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? >> > ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? >> > ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. >> > ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? >> > "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? >> > ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? >> > ?????????????. >> > >> > ========================================================= >> > >> > This message and any attachments (the "message") are confidential, >> > intended solely for the addressees, and may contain legally privileged >> > information. Any unauthorized use or dissemination is prohibited. E-mails >> > are susceptible to alteration. Neither JSC "BELROSBANK" nor any of its >> > subsidiaries or affiliates shall be liable for the message if altered, >> > changed or falsified. >> > >> > ========================================================= >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From jmesquita at freeswitch.org Fri Jul 6 21:06:04 2012 From: jmesquita at freeswitch.org (Jmesquita@freeswitch.org) Date: Fri, 6 Jul 2012 14:06:04 -0300 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <9343F57A41054F8DB89BD4A0C17CFD7B@gmail.com> <0B4DC9FFE198449888339EA7BA4E3E24@freeswitch.org> Message-ID: That is a common misconception. ZeroMQ is a socket on steroids, that's it. It does up to what it can do without messing with the protocol you are implementing. Pub/sub can handle message delivery reliability but it is not default on the pattern because you can easily flood memory if the worker is not available. Most of the protocols using zeromq implement a ready message that specifies when a certain worker is available and then start sending its messages as well as handle disk persistence if needed. Zeromq also implemented something they call high water mark on the socket and that is the amount of messages to be queued on memory before starting to discard. This is only used on durable sockets, which from what I understand is being deprecated in favor of leaving the task to the nderlying protocol. Hope that clears it. Regards, Jo?o Mesquita On 06/07/2012, at 04:14 a.m., Neo Cheema wrote: > Jo?o, > > "Since the socket implemented on mod_zmq is a simple PUB/SUB, yes, > messages would be discarded if they are not received by the other > end." > > I thought it didn't matter if the 'worker' was up before the > 'producer'/freeswitch started pushing in the messages. 0mq handles > that part, if atleast one worker is connected. > > My main reason for looking towards 0mq was to bypass a webserver, > which would be required for xml_cdr. > > On Fri, Jul 6, 2012 at 8:33 AM, jay binks wrote: >> while we are talking about funky CDR stuff. >> >> Ive been planning to look at riak ( http://wiki.basho.com/ ) for bulk CDR >> storage >> ( storing data other than billing data ) but just not finding the time. >> >> anyone done something similar ? >> The idea is to store as much data in a distributed manner, so as to aid >> debugging. >> ( not so much for billing ) >> >> Jay >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nasida at live.ru Fri Jul 6 21:18:58 2012 From: nasida at live.ru (Yuriy Nasida) Date: Fri, 6 Jul 2012 21:18:58 +0400 Subject: [Freeswitch-users] sensibility of voice level for Mod_pocketsphinx Message-ID: Hi guys, Whether somebody know how to lower sensibility of voice level for Mod_pocketsphinx. I play with "threshold" in pocketsphinx.conf.xml but I have not any effect.the Mod_pocketsphinx begins to generate event == "begin-speaking" when I don't say anything... Also i would like do NOT break the current playing of session:streamFile when events of Mod_pocketsphinx is being generated. Is it possible ? Also can somebody explain that this parameters means ? I understand approximately but I would like to know for certain. I have found nothing in wiki. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120706/49af8e4d/attachment.html From omortimer at gmail.com Fri Jul 6 21:29:04 2012 From: omortimer at gmail.com (Oz Mortimer) Date: Fri, 6 Jul 2012 18:29:04 +0100 Subject: [Freeswitch-users] Load Balancing In-Reply-To: References: Message-ID: <98EBE59F-CAB0-4BC9-95E9-C912DD0AA17C@gmail.com> Yes, srv records would work but as you said, they don't know the state of you installation, I.e. up, down or indifferent. Also, I have found that there are not many carriers that support srv and those that do lack the full implementation. If it was me, I'd put openser or something similar in front of your freeswitch boxes which will act as a router, load balancer. The is nothing wrong with using heartbeat and corosync but your public ip would need to be on the same subnet for it to work correctly. Hope this helps. Oz. Sent from my iPad On 6 Jul 2012, at 17:04, Adam Obuchowski wrote: > Hello, > What you mean by DNS based failover ? > Short ttl on DNS A record may cause name resolution issues, isnt it ? > How does it work ? > > Kind regards, > > -- > Adam Obuchowski > VoipSwitch support > www.voipswitch.com > > > > > 2012/7/6 Muhammad Shahzad > You can also try DNS based fail-over and load balancing, works perfect for us. > > Thank you. > > > On Fri, Jul 6, 2012 at 9:02 AM, A E G wrote: > > On Wed, Jul 4, 2012 at 5:53 AM, Hanie Maghsoudy wrote: > Thanks Muhammad, that would be a good idea. > However, I'm trying direct routing mode of IPVS for load balancing along with Heartbeat for HA. > Could anyone of you guys suggest a better way to implement Load balancing and HA for FreeSwitch? > > Thanks > > > Not sure about HA, but the call distributor module: http://wiki.freeswitch.org/wiki/Mod_distributor used to work well for us ...haven't used it for a while now. Not sure if anything has changed in newer builds. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120706/87bc9232/attachment.html From jalsot at gmail.com Fri Jul 6 22:16:50 2012 From: jalsot at gmail.com (Tamas Jalsovszky) Date: Fri, 6 Jul 2012 20:16:50 +0200 Subject: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a In-Reply-To: References: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern> <4FED4DA1.10304@belrosbank.by> Message-ID: Hello, It is an interesting thread :) How do we know which motherboards are "really really nice? Which parameters are the most important? Maybe a brand (eg. supermicro) or price range could help choosing? Br, Jalsot On Fri, Jul 6, 2012 at 6:54 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Like I said "Really Really Nice Motherboard", as many CPU as > possible/affordable, and a good chunk of RAM. > Motherboard is the most important, a cheap motherboard with a ton of > cores is a waste. > > > On Thu, Jul 5, 2012 at 3:27 PM, A E G wrote: > > Thanks for the short/straight pointers re: what matters most for > freeswitch > > performance. > > > > Are there any equivalent pearls of wisdom "best practices" re: CPU, RAM > and > > Shared Memory/Bus that you or others can advise on, if this was to run > in a > > virtual environment in a private cloud of Xen Server, KVM or vCloud etc? > > Probably it will depend on the underlying hardware or not? kernel tuning > of > > the Linux guest as well as resource sharing? > > > > Any light you or anyone else can shed on this would be greatly > appreciated > > > > Thanks in advance > > > > On Fri, Jun 29, 2012 at 1:07 PM, Anthony Minessale > > wrote: > >> > >> you could do with more than 2 cpu, the more cores the better you will > do. > >> For similar prices you could find a 8 or 12 core box. > >> > >> The 3 most important things in a FS server are, as many CPU cores as > >> possible, as much ram as possible, the best possible CPU with shared > >> bus to RAM and CPU. > >> > >> > >> > >> 2012/6/29 Eugene Shcherbatyuk : > >> > Session limit is configurable. Have a look at switch.conf.xml file. > >> > > >> > > >> > On 28/06/12 21:41, Hoexum, Edwin wrote: > >> >> > >> >> I am trying to do some load testing on a freeswitch for transcoding > >> >> from > >> >> SILK to G711a. The question is: Is the limit for RTP sessions > >> >> limited to CPU power or are there also limits on the freeswitch > >> >> application and can I tune freeswitch to do the job for 1500 sessions > >> >> or > >> >> more. > >> > > >> > > >> > ========================================================= > >> > > >> > ?????? ????????? ? ????? ???????? (<>) ???????? > >> > ?????????????????, ???????????????? ????????????? ??? ?????????, ? > ????? > >> > ????????? ????????, ?????????? ? ???????????? ? ?????????????????. > ????? > >> > ??????????????????? ????????????? ??? ??????????????? ????????? > ?????????. > >> > ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? > >> > "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? > ????? > >> > ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? > >> > ?????????????. > >> > > >> > ========================================================= > >> > > >> > This message and any attachments (the "message") are confidential, > >> > intended solely for the addressees, and may contain legally privileged > >> > information. Any unauthorized use or dissemination is prohibited. > E-mails > >> > are susceptible to alteration. Neither JSC "BELROSBANK" nor any of its > >> > subsidiaries or affiliates shall be liable for the message if altered, > >> > changed or falsified. > >> > > >> > ========================================================= > >> > > >> > > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > Join Us At ClueCon - Aug 7-9, 2012 > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> Twitter: http://twitter.com/FreeSWITCH_wire > >> > >> AIM: anthm > >> MSN:anthony_minessale at hotmail.com > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conference > >> sip:888 at conference.freeswitch.org > >> googletalk:conf+888 at conference.freeswitch.org > >> pstn:+19193869900 > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120706/13cb2cac/attachment-0001.html From anthony.minessale at gmail.com Fri Jul 6 23:10:45 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Jul 2012 14:10:45 -0500 Subject: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a In-Reply-To: References: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern> <4FED4DA1.10304@belrosbank.by> Message-ID: The ones that are not nice tend to be the ones who use a shared bus for all the resources so if you are using them all at once like FS does things are waiting for each other. like RAM, DISK, CPU have to take turns. The last gen of xeon are somewhat nice. They are basically the same architecture as i5/i7 but still xeon and you can have multiples of 6 at a time in one box. These CPU http://www.newegg.com/Product/Product.aspx?Item=N82E16819117256 and this barebones supermicro http://www.newegg.com/Product/Product.aspx?Item=N82E16816101314 is what I use on my dev box with 24 gigs of ram. On Fri, Jul 6, 2012 at 1:16 PM, Tamas Jalsovszky wrote: > Hello, > > It is an interesting thread :) > How do we know which motherboards are "really really nice? Which parameters > are the most important? > Maybe a brand (eg. supermicro) or price range could help choosing? > > Br, > Jalsot > > > On Fri, Jul 6, 2012 at 6:54 PM, Anthony Minessale > wrote: >> >> Like I said "Really Really Nice Motherboard", as many CPU as >> possible/affordable, and a good chunk of RAM. >> Motherboard is the most important, a cheap motherboard with a ton of >> cores is a waste. >> >> >> On Thu, Jul 5, 2012 at 3:27 PM, A E G wrote: >> > Thanks for the short/straight pointers re: what matters most for >> > freeswitch >> > performance. >> > >> > Are there any equivalent pearls of wisdom "best practices" re: CPU, RAM >> > and >> > Shared Memory/Bus that you or others can advise on, if this was to run >> > in a >> > virtual environment in a private cloud of Xen Server, KVM or vCloud etc? >> > Probably it will depend on the underlying hardware or not? kernel tuning >> > of >> > the Linux guest as well as resource sharing? >> > >> > Any light you or anyone else can shed on this would be greatly >> > appreciated >> > >> > Thanks in advance >> > >> > On Fri, Jun 29, 2012 at 1:07 PM, Anthony Minessale >> > wrote: >> >> >> >> you could do with more than 2 cpu, the more cores the better you will >> >> do. >> >> For similar prices you could find a 8 or 12 core box. >> >> >> >> The 3 most important things in a FS server are, as many CPU cores as >> >> possible, as much ram as possible, the best possible CPU with shared >> >> bus to RAM and CPU. >> >> >> >> >> >> >> >> 2012/6/29 Eugene Shcherbatyuk : >> >> > Session limit is configurable. Have a look at switch.conf.xml file. >> >> > >> >> > >> >> > On 28/06/12 21:41, Hoexum, Edwin wrote: >> >> >> >> >> >> I am trying to do some load testing on a freeswitch for transcoding >> >> >> from >> >> >> SILK to G711a. The question is: Is the limit for RTP sessions >> >> >> limited to CPU power or are there also limits on the freeswitch >> >> >> application and can I tune freeswitch to do the job for 1500 >> >> >> sessions >> >> >> or >> >> >> more. >> >> > >> >> > >> >> > ========================================================= >> >> > >> >> > ?????? ????????? ? ????? ???????? (<>) ???????? >> >> > ?????????????????, ???????????????? ????????????? ??? ?????????, ? >> >> > ????? >> >> > ????????? ????????, ?????????? ? ???????????? ? ?????????????????. >> >> > ????? >> >> > ??????????????????? ????????????? ??? ??????????????? ????????? >> >> > ?????????. >> >> > ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? >> >> > "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? >> >> > ????? >> >> > ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? >> >> > ?????????????. >> >> > >> >> > ========================================================= >> >> > >> >> > This message and any attachments (the "message") are confidential, >> >> > intended solely for the addressees, and may contain legally >> >> > privileged >> >> > information. Any unauthorized use or dissemination is prohibited. >> >> > E-mails >> >> > are susceptible to alteration. Neither JSC "BELROSBANK" nor any of >> >> > its >> >> > subsidiaries or affiliates shall be liable for the message if >> >> > altered, >> >> > changed or falsified. >> >> > >> >> > ========================================================= >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ >> >> > Professional FreeSWITCH Consulting Services: >> >> > consulting at freeswitch.org >> >> > http://www.freeswitchsolutions.com >> >> > >> >> > >> >> > >> >> > >> >> > Official FreeSWITCH Sites >> >> > http://www.freeswitch.org >> >> > http://wiki.freeswitch.org >> >> > http://www.cluecon.com >> >> > >> >> > Join Us At ClueCon - Aug 7-9, 2012 >> >> > >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> > http://www.freeswitch.org >> >> >> >> >> >> >> >> -- >> >> Anthony Minessale II >> >> >> >> FreeSWITCH http://www.freeswitch.org/ >> >> ClueCon http://www.cluecon.com/ >> >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> >> >> AIM: anthm >> >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> >> >> FreeSWITCH Developer Conference >> >> sip:888 at conference.freeswitch.org >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:+19193869900 >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From lazyvirus at gmx.com Sat Jul 7 00:26:35 2012 From: lazyvirus at gmx.com (Bzzz) Date: Fri, 6 Jul 2012 22:26:35 +0200 Subject: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a In-Reply-To: References: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern> <4FED4DA1.10304@belrosbank.by> Message-ID: <20120706222635.66d55798@anubis.defcon1> On Fri, 6 Jul 2012 20:16:50 +0200 Tamas Jalsovszky wrote: > How do we know which motherboards are "really really nice? Which > parameters are the most important? AM told you: cores NB + RAM + Server Mobo (not especially a top 1, but studied to pipe a maximum in parallel processing) > Maybe a brand (eg. supermicro) or price range could help choosing? Supermicro has recentered its mobo gammut more on intel cpus than AMD these last years, and their quality/price isn't, uh... very good. http://www.tyan.com/product_SKU_spec.aspx?ProductType=MB&pid=670&SKU=600000180 It hasn't a low cost but that doesn't matter very much for a professional use, and this one can go up to 64 cores & 512 GB RAM. If you're patient (and if FS work on them?), HP and some others as concocting multi-many-bi/quad-core-ARM-A7~A9 server mobos which will give you a tremendous number of cores & RAM with the benefit of (very) low power consumption. Jean-Yves -- children make stupidities in the dark =P stupidities in the dark make children... From itamar at ispbrasil.com.br Sat Jul 7 00:41:54 2012 From: itamar at ispbrasil.com.br (Itamar Reis Peixoto) Date: Fri, 6 Jul 2012 17:41:54 -0300 Subject: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a In-Reply-To: References: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern> <4FED4DA1.10304@belrosbank.by> Message-ID: On Fri, Jul 6, 2012 at 4:10 PM, Anthony Minessale wrote: > The ones that are not nice tend to be the ones who use a shared bus > for all the resources so if you are using them all at once like FS > does things are waiting for each other. like RAM, DISK, CPU have to > take turns. > > The last gen of xeon are somewhat nice. They are basically the same > architecture as i5/i7 but still xeon and you can have multiples of 6 > at a time in one box. > These CPU http://www.newegg.com/Product/Product.aspx?Item=N82E16819117256 > > and this barebones supermicro > http://www.newegg.com/Product/Product.aspx?Item=N82E16816101314 > > is what I use on my dev box with 24 gigs of ram. > I like this one http://www.newegg.com/Product/Product.aspx?Item=N82E16816110061 -- ------------ Itamar Reis Peixoto msn, google talk: itamar at ispbrasil.com.br +55 11 4063 5033 (FIXO SP) +55 34 9158 9329 (TIM) +55 34 8806 3989 (OI) +55 34 3221 8599 (FIXO MG) From shaheryarkh at googlemail.com Fri Jul 6 23:45:11 2012 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Sat, 7 Jul 2012 00:45:11 +0500 Subject: [Freeswitch-users] Load Balancing In-Reply-To: <98EBE59F-CAB0-4BC9-95E9-C912DD0AA17C@gmail.com> References: <98EBE59F-CAB0-4BC9-95E9-C912DD0AA17C@gmail.com> Message-ID: There are many ways to achieve load balance and fail-over in VOIP world. Each has its advantages and disadvantages, so the right solution really depends on your needs rather then your wants (needs vs wants). Please see attached research paper from Columbia University students, which discusses the topic in detail. Thank you. On Fri, Jul 6, 2012 at 10:29 PM, Oz Mortimer wrote: > Yes, srv records would work but as you said, they don't know the state of > you installation, I.e. up, down or indifferent. Also, I have found that > there are not many carriers that support srv and those that do lack the > full implementation. > > If it was me, I'd put openser or something similar in front of your > freeswitch boxes which will act as a router, load balancer. > > The is nothing wrong with using heartbeat and corosync but your public ip > would need to be on the same subnet for it to work correctly. > > Hope this helps. > Oz. > > Sent from my iPad > > On 6 Jul 2012, at 17:04, Adam Obuchowski wrote: > > Hello, > What you mean by DNS based failover ? > Short ttl on DNS A record may cause name resolution issues, isnt it ? > How does it work ? > > Kind regards, > > -- > Adam Obuchowski > VoipSwitch support > www.voipswitch.com > > > > > 2012/7/6 Muhammad Shahzad > >> You can also try DNS based fail-over and load balancing, works perfect >> for us. >> >> Thank you. >> >> >> On Fri, Jul 6, 2012 at 9:02 AM, A E G wrote: >> >>> >>> On Wed, Jul 4, 2012 at 5:53 AM, Hanie Maghsoudy wrote: >>> >>>> Thanks Muhammad, that would be a good idea. >>>> However, I'm trying direct routing mode of IPVS for load balancing >>>> along with Heartbeat for HA. >>>> Could anyone of you guys suggest a better way to implement Load >>>> balancing and HA for FreeSwitch? >>>> >>>> Thanks >>>> >>>> >>> Not sure about HA, but the call distributor module: >>> http://wiki.freeswitch.org/wiki/Mod_distributor used to work well for >>> us ...haven't used it for a while now. Not sure if anything has changed in >>> newer builds. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +92 334 422 40 88 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk at hotmail.com Email: shaheryarkh at googlemail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120707/17b791af/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: cucs-011-04.pdf Type: application/pdf Size: 245307 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120707/17b791af/attachment-0001.pdf From all.eforums at gmail.com Sat Jul 7 00:59:27 2012 From: all.eforums at gmail.com (A E G) Date: Fri, 6 Jul 2012 16:59:27 -0400 Subject: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a In-Reply-To: References: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern> <4FED4DA1.10304@belrosbank.by> Message-ID: Yup got that, and are you saying the same is applicable and true even if one was to try and run FreeSWITCH on a private cloud or in a virtual environment? assuming of course that those tips would then apply to the machine on which the Hypervisor will run. Just trying to get a handle on whether running FreeSWITCH to do something like wholesale or calling card traffic in a purely virtual environment is / has proven to work. You probably know the most in terms of all different environments people are running FS in, and if you (or they) have pointers specific to it being run in a virtual environment. Thanks so much On Fri, Jul 6, 2012 at 12:54 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Like I said "Really Really Nice Motherboard", as many CPU as > possible/affordable, and a good chunk of RAM. > Motherboard is the most important, a cheap motherboard with a ton of > cores is a waste. > > > On Thu, Jul 5, 2012 at 3:27 PM, A E G wrote: > > Thanks for the short/straight pointers re: what matters most for > freeswitch > > performance. > > > > Are there any equivalent pearls of wisdom "best practices" re: CPU, RAM > and > > Shared Memory/Bus that you or others can advise on, if this was to run > in a > > virtual environment in a private cloud of Xen Server, KVM or vCloud etc? > > Probably it will depend on the underlying hardware or not? kernel tuning > of > > the Linux guest as well as resource sharing? > > > > Any light you or anyone else can shed on this would be greatly > appreciated > > > > Thanks in advance > > > > On Fri, Jun 29, 2012 at 1:07 PM, Anthony Minessale > > wrote: > >> > >> you could do with more than 2 cpu, the more cores the better you will > do. > >> For similar prices you could find a 8 or 12 core box. > >> > >> The 3 most important things in a FS server are, as many CPU cores as > >> possible, as much ram as possible, the best possible CPU with shared > >> bus to RAM and CPU. > >> > >> > >> > >> 2012/6/29 Eugene Shcherbatyuk : > >> > Session limit is configurable. Have a look at switch.conf.xml file. > >> > > >> > > >> > On 28/06/12 21:41, Hoexum, Edwin wrote: > >> >> > >> >> I am trying to do some load testing on a freeswitch for transcoding > >> >> from > >> >> SILK to G711a. The question is: Is the limit for RTP sessions > >> >> limited to CPU power or are there also limits on the freeswitch > >> >> application and can I tune freeswitch to do the job for 1500 sessions > >> >> or > >> >> more. > >> > > >> > > >> > ========================================================= > >> > > >> > ?????? ????????? ? ????? ???????? (<>) ???????? > >> > ?????????????????, ???????????????? ????????????? ??? ?????????, ? > ????? > >> > ????????? ????????, ?????????? ? ???????????? ? ?????????????????. > ????? > >> > ??????????????????? ????????????? ??? ??????????????? ????????? > ?????????. > >> > ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? > >> > "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? > ????? > >> > ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? > >> > ?????????????. > >> > > >> > ========================================================= > >> > > >> > This message and any attachments (the "message") are confidential, > >> > intended solely for the addressees, and may contain legally privileged > >> > information. Any unauthorized use or dissemination is prohibited. > E-mails > >> > are susceptible to alteration. Neither JSC "BELROSBANK" nor any of its > >> > subsidiaries or affiliates shall be liable for the message if altered, > >> > changed or falsified. > >> > > >> > ========================================================= > >> > > >> > > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > Join Us At ClueCon - Aug 7-9, 2012 > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> Twitter: http://twitter.com/FreeSWITCH_wire > >> > >> AIM: anthm > >> MSN:anthony_minessale at hotmail.com > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conference > >> sip:888 at conference.freeswitch.org > >> googletalk:conf+888 at conference.freeswitch.org > >> pstn:+19193869900 > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120706/7e4f5d29/attachment.html From all.eforums at gmail.com Sat Jul 7 01:35:36 2012 From: all.eforums at gmail.com (A E G) Date: Fri, 6 Jul 2012 17:35:36 -0400 Subject: [Freeswitch-users] Load Balancing In-Reply-To: References: Message-ID: On Fri, Jul 6, 2012 at 12:03 PM, Hanie Maghsoudy wrote: > Thanks, But I think distributor module works for routing calls after FS > receives them. I plan to have a cluster of FreeSwitches, so that I can > balance the load of my incoming calls. > > Yes, but we had FS at the head-end of a cluster of FS boxes. No one said FS can only sit at the core ;) The other benefit you get with it is ALG type functionality (with some configuration). DNS isn't a good way to do this as it has no sense of context of what is being sent (SRV does though, but is not widely supported by many ISPs or domain name providers, unless you run your own DNS servers) and will blindly RR between IPs whether it is up or down or worse hung ...so if you have 3 machine farm, and one of them is hung, 1 in 3 calls will fail. Proper load-balancing should have context and awareness of the application as well the downstream server to which the call should or should not be sent to if it's not capable of servicing the call. > On Fri, Jul 6, 2012 at 8:32 AM, A E G wrote: > >> >> On Wed, Jul 4, 2012 at 5:53 AM, Hanie Maghsoudy wrote: >> >>> Thanks Muhammad, that would be a good idea. >>> However, I'm trying direct routing mode of IPVS for load balancing along >>> with Heartbeat for HA. >>> Could anyone of you guys suggest a better way to implement Load >>> balancing and HA for FreeSwitch? >>> >>> Thanks >>> >>> >> Not sure about HA, but the call distributor module: >> http://wiki.freeswitch.org/wiki/Mod_distributor used to work well for us >> ...haven't used it for a while now. Not sure if anything has changed in >> newer builds. >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120706/180806b2/attachment-0001.html From anthony.minessale at gmail.com Sat Jul 7 01:43:10 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Jul 2012 16:43:10 -0500 Subject: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a In-Reply-To: References: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern> <4FED4DA1.10304@belrosbank.by> Message-ID: I would be insane to endorse using it in a virtual world. The support expectations would consume the rest of my life. I try to make the code work well on real boxes and if it works on virtual ones then that's a bonus =D We have good luck with openVZ since its a single real kernel and virtual runtimes but either way we do not recommend it so its a use at your own risk kind of thing. On Fri, Jul 6, 2012 at 3:59 PM, A E G wrote: > Yup got that, and are you saying the same is applicable and true even if one > was to try and run FreeSWITCH on a private cloud or in a virtual > environment? assuming of course that those tips would then apply to the > machine on which the Hypervisor will run. > > Just trying to get a handle on whether running FreeSWITCH to do something > like wholesale or calling card traffic in a purely virtual environment is / > has proven to work. You probably know the most in terms of all different > environments people are running FS in, and if you (or they) have pointers > specific to it being run in a virtual environment. > > Thanks so much > > On Fri, Jul 6, 2012 at 12:54 PM, Anthony Minessale > wrote: >> >> Like I said "Really Really Nice Motherboard", as many CPU as >> possible/affordable, and a good chunk of RAM. >> Motherboard is the most important, a cheap motherboard with a ton of >> cores is a waste. >> >> >> On Thu, Jul 5, 2012 at 3:27 PM, A E G wrote: >> > Thanks for the short/straight pointers re: what matters most for >> > freeswitch >> > performance. >> > >> > Are there any equivalent pearls of wisdom "best practices" re: CPU, RAM >> > and >> > Shared Memory/Bus that you or others can advise on, if this was to run >> > in a >> > virtual environment in a private cloud of Xen Server, KVM or vCloud etc? >> > Probably it will depend on the underlying hardware or not? kernel tuning >> > of >> > the Linux guest as well as resource sharing? >> > >> > Any light you or anyone else can shed on this would be greatly >> > appreciated >> > >> > Thanks in advance >> > >> > On Fri, Jun 29, 2012 at 1:07 PM, Anthony Minessale >> > wrote: >> >> >> >> you could do with more than 2 cpu, the more cores the better you will >> >> do. >> >> For similar prices you could find a 8 or 12 core box. >> >> >> >> The 3 most important things in a FS server are, as many CPU cores as >> >> possible, as much ram as possible, the best possible CPU with shared >> >> bus to RAM and CPU. >> >> >> >> >> >> >> >> 2012/6/29 Eugene Shcherbatyuk : >> >> > Session limit is configurable. Have a look at switch.conf.xml file. >> >> > >> >> > >> >> > On 28/06/12 21:41, Hoexum, Edwin wrote: >> >> >> >> >> >> I am trying to do some load testing on a freeswitch for transcoding >> >> >> from >> >> >> SILK to G711a. The question is: Is the limit for RTP sessions >> >> >> limited to CPU power or are there also limits on the freeswitch >> >> >> application and can I tune freeswitch to do the job for 1500 >> >> >> sessions >> >> >> or >> >> >> more. >> >> > >> >> > >> >> > ========================================================= >> >> > >> >> > ?????? ????????? ? ????? ???????? (<>) ???????? >> >> > ?????????????????, ???????????????? ????????????? ??? ?????????, ? >> >> > ????? >> >> > ????????? ????????, ?????????? ? ???????????? ? ?????????????????. >> >> > ????? >> >> > ??????????????????? ????????????? ??? ??????????????? ????????? >> >> > ?????????. >> >> > ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? >> >> > "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? >> >> > ????? >> >> > ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? >> >> > ?????????????. >> >> > >> >> > ========================================================= >> >> > >> >> > This message and any attachments (the "message") are confidential, >> >> > intended solely for the addressees, and may contain legally >> >> > privileged >> >> > information. Any unauthorized use or dissemination is prohibited. >> >> > E-mails >> >> > are susceptible to alteration. Neither JSC "BELROSBANK" nor any of >> >> > its >> >> > subsidiaries or affiliates shall be liable for the message if >> >> > altered, >> >> > changed or falsified. >> >> > >> >> > ========================================================= >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ >> >> > Professional FreeSWITCH Consulting Services: >> >> > consulting at freeswitch.org >> >> > http://www.freeswitchsolutions.com >> >> > >> >> > >> >> > >> >> > >> >> > Official FreeSWITCH Sites >> >> > http://www.freeswitch.org >> >> > http://wiki.freeswitch.org >> >> > http://www.cluecon.com >> >> > >> >> > Join Us At ClueCon - Aug 7-9, 2012 >> >> > >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> > http://www.freeswitch.org >> >> >> >> >> >> >> >> -- >> >> Anthony Minessale II >> >> >> >> FreeSWITCH http://www.freeswitch.org/ >> >> ClueCon http://www.cluecon.com/ >> >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> >> >> AIM: anthm >> >> MSN:anthony_minessale at hotmail.com >> >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> >> IRC: irc.freenode.net #freeswitch >> >> >> >> FreeSWITCH Developer Conference >> >> sip:888 at conference.freeswitch.org >> >> googletalk:conf+888 at conference.freeswitch.org >> >> pstn:+19193869900 >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From hi-tecc at hotmail.com Sat Jul 7 02:07:52 2012 From: hi-tecc at hotmail.com (DP .) Date: Fri, 6 Jul 2012 18:07:52 -0400 Subject: [Freeswitch-users] mod_skypopen not generating inbound A-leg CDR Message-ID: Hi, Does mod_skypopen generate inbound A-leg CDRs? Simple scenario, inbound call from Skype to anywhere in Freeswitch (currently a Lua script, but also tried direct to extensions), does not generate any CDRs for that A-Leg. I tried transferring to another context, bridging, etc to no avail... only the b-leg, if present, generates a CDR. To eliminate my xml_cdr, I purposely set the post url to an invalid link, which throws a red crit error in console upon call hangup of regular mod_sofia calls. The mod_skypopen throws no error whatsoever, possibly indicating its not generating the CDR? Im running FreeSWITCH Version 1.0.head (git-5e4a514 2012-03-10 22-56-29 -0500). (Tried updating to the latest 1.2 branch last week but got random restarts and crashes of freeswitch, also kept getting "unclean" in the git version header after several make currents', had to quickly revert as this is a production box). Will clone the install again and try and test the latest git at a later date). Thanks, -Damian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120706/a48fdb20/attachment.html From all.eforums at gmail.com Sat Jul 7 02:23:35 2012 From: all.eforums at gmail.com (A E G) Date: Fri, 6 Jul 2012 18:23:35 -0400 Subject: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a In-Reply-To: References: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern> <4FED4DA1.10304@belrosbank.by> Message-ID: On Fri, Jul 6, 2012 at 5:43 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I would be insane to endorse using it in a virtual world. The support > expectations would consume the rest of my life. > I try to make the code work well on real boxes and if it works on > virtual ones then that's a bonus =D > > We have good luck with openVZ since its a single real kernel and > virtual runtimes but either way we do not recommend it so its a use at > your own risk kind of thing. > > Yup agree and acknowledge that you wouldn't endorse or offer to support it, and nor should you. Simply trying to pick your (and anyone else who wants to chime in) brain on the topic. Totally PoC'ing it at my own risk :) So, in that spirit, just one last message about it. Just like sharing the experience with OpenVZ, let's just say that I am insane and I still want to try it in a private cloud. Would you say the resource consumption of FS (assuming it's just receiving the call, having it routed by a custom routing engine via ESL, and switching it) be more than that in the "real" world for the same call volume barring any clock sync/skew/jitter issues? Would something like Cloudstack on hardware (like the one you recommended earlier) with a FS VM with 12 vcpu cores, 24GB RAM and DAS be able to handle something like 300-400 concurrent calls or am I out of my mind? Thx > On Fri, Jul 6, 2012 at 3:59 PM, A E G wrote: > > Yup got that, and are you saying the same is applicable and true even if > one > > was to try and run FreeSWITCH on a private cloud or in a virtual > > environment? assuming of course that those tips would then apply to the > > machine on which the Hypervisor will run. > > > > Just trying to get a handle on whether running FreeSWITCH to do something > > like wholesale or calling card traffic in a purely virtual environment > is / > > has proven to work. You probably know the most in terms of all different > > environments people are running FS in, and if you (or they) have pointers > > specific to it being run in a virtual environment. > > > > Thanks so much > > > > On Fri, Jul 6, 2012 at 12:54 PM, Anthony Minessale > > wrote: > >> > >> Like I said "Really Really Nice Motherboard", as many CPU as > >> possible/affordable, and a good chunk of RAM. > >> Motherboard is the most important, a cheap motherboard with a ton of > >> cores is a waste. > >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120706/94ab86bb/attachment.html From anthony.minessale at gmail.com Sat Jul 7 03:01:29 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 6 Jul 2012 18:01:29 -0500 Subject: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a In-Reply-To: References: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern> <4FED4DA1.10304@belrosbank.by> Message-ID: Hundreds of calls should be no problem. Thousands start to take some finesse.... Like Doc says, "roads, where we're going we don't need roads...." On Fri, Jul 6, 2012 at 5:23 PM, A E G wrote: > On Fri, Jul 6, 2012 at 5:43 PM, Anthony Minessale > wrote: >> >> I would be insane to endorse using it in a virtual world. The support >> expectations would consume the rest of my life. >> I try to make the code work well on real boxes and if it works on >> virtual ones then that's a bonus =D >> >> We have good luck with openVZ since its a single real kernel and >> virtual runtimes but either way we do not recommend it so its a use at >> your own risk kind of thing. >> > > Yup agree and acknowledge that you wouldn't endorse or offer to support it, > and nor should you. Simply trying to pick your (and anyone else who wants to > chime in) brain on the topic. Totally PoC'ing it at my own risk :) > > So, in that spirit, just one last message about it. Just like sharing the > experience with OpenVZ, let's just say that I am insane and I still want to > try it in a private cloud. Would you say the resource consumption of FS > (assuming it's just receiving the call, having it routed by a custom routing > engine via ESL, and switching it) be more than that in the "real" world for > the same call volume barring any clock sync/skew/jitter issues? Would > something like Cloudstack on hardware (like the one you recommended earlier) > with a FS VM with 12 vcpu cores, 24GB RAM and DAS be able to handle > something like 300-400 concurrent calls or am I out of my mind? > > Thx > > >> >> On Fri, Jul 6, 2012 at 3:59 PM, A E G wrote: >> > Yup got that, and are you saying the same is applicable and true even if >> > one >> > was to try and run FreeSWITCH on a private cloud or in a virtual >> > environment? assuming of course that those tips would then apply to the >> > machine on which the Hypervisor will run. >> > >> > Just trying to get a handle on whether running FreeSWITCH to do >> > something >> > like wholesale or calling card traffic in a purely virtual environment >> > is / >> > has proven to work. You probably know the most in terms of all different >> > environments people are running FS in, and if you (or they) have >> > pointers >> > specific to it being run in a virtual environment. >> > >> > Thanks so much >> > >> > On Fri, Jul 6, 2012 at 12:54 PM, Anthony Minessale >> > wrote: >> >> >> >> Like I said "Really Really Nice Motherboard", as many CPU as >> >> possible/affordable, and a good chunk of RAM. >> >> Motherboard is the most important, a cheap motherboard with a ton of >> >> cores is a waste. >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From moises.silva at gmail.com Sat Jul 7 03:19:59 2012 From: moises.silva at gmail.com (Moises Silva) Date: Fri, 6 Jul 2012 19:19:59 -0400 Subject: [Freeswitch-users] Unable to set effective_caller_id_number when bridging using Openzap In-Reply-To: <4FF57BA5.7090000@as-infodienste.de> References: <4FF57BA5.7090000@as-infodienste.de> Message-ID: On Thu, Jul 5, 2012 at 7:33 AM, Marcus M?lb?sch < muelbuesch at as-infodienste.de> wrote: > Hello all, > > yes, for hardware reasons (very old Sangoma card with very old > firmware) I have to use OpenZAP instead of freetTDM. > > So, when setting the effective_caller_id that value isn't used. See > that part of my dialplan here: > > Please use pastebin.com and provide relevant debug logs. How are you testing that it does not work? *Moises Silva **Manager, Software Engineering*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120706/08a8e946/attachment.html From ssinyagin at yahoo.com Sat Jul 7 03:27:32 2012 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Fri, 6 Jul 2012 16:27:32 -0700 (PDT) Subject: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a In-Reply-To: References: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern> <4FED4DA1.10304@belrosbank.by> Message-ID: <1341617252.56013.YahooMailNeo@web39301.mail.mud.yahoo.com> OpenVZ and XEN VPS'es are usually fine for quick prototyping and feature tests. Also they would be fine for private PBXes with low call volumes. For hundreds of simultaneous calls, it's probably cheaper to rent dedicated hardware servers. On the other side, if you have some smart load balancing, probably you can build a scalable design with VPS'es and eventually migrate to physical servers when volumes? are high enough. also for a calling card business, probably the most critical CPU resource would be needed for the IVR and real-time billing, but not for the calls themselves. >________________________________ > From: A E G >To: FreeSWITCH Users Help >Sent: Saturday, July 7, 2012 12:23 AM >Subject: Re: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a > > >On Fri, Jul 6, 2012 at 5:43 PM, Anthony Minessale wrote: > >I would be insane to endorse using it in a virtual world. ?The support >>expectations would consume the rest of my life. >>I try to make the code work well on real boxes and if it works on >>virtual ones then that's a bonus =D >> >>We have good luck with openVZ since its a single real kernel and >>virtual runtimes but either way we do not recommend it so its a use at >>your own risk kind of thing. >> >> >> > > >Yup agree and acknowledge that you wouldn't endorse or offer to support it, and nor should you.?Simply trying to pick your (and anyone else who wants to chime in) brain on the topic. Totally PoC'ing it at my own risk :) > > >So, in that spirit, just one last message about it. Just like sharing the experience with OpenVZ, let's just say that I am insane and I still want to try it in a private cloud. Would you say the resource consumption of FS (assuming it's just receiving the call, having it routed by a custom routing engine via ESL, and switching it) be more than that in the "real" world for the same call volume barring any clock sync/skew/jitter issues? Would something like Cloudstack on hardware (like the one you recommended earlier) with a FS VM with 12 vcpu cores, 24GB RAM and DAS be able to handle something like 300-400 concurrent calls or am I out of my mind? > > >Thx? > > >? >On Fri, Jul 6, 2012 at 3:59 PM, A E G wrote: >>> Yup got that, and are you saying the same is applicable and true even if one >>> was to try and run FreeSWITCH on a private cloud or in a virtual >>> environment? assuming of course that those tips would then apply to the >>> machine on which the Hypervisor will run. >>> >>> Just trying to get a handle on whether running FreeSWITCH to do something >>> like wholesale or calling card traffic in a purely virtual environment is / >>> has proven to work. You probably know the most in terms of all different >>> environments people are running FS in, and if you (or they) have pointers >>> specific to it being run in a virtual environment. >>> >>> Thanks so much >>> >>> On Fri, Jul 6, 2012 at 12:54 PM, Anthony Minessale >>> wrote: >>>> >>>> Like I said "Really Really Nice Motherboard", as many CPU as >>>> possible/affordable, and a good chunk of RAM. >>>> Motherboard is the most important, a cheap motherboard with a ton of >>>> cores is a waste. >>>> >> >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >Join Us At ClueCon - Aug 7-9, 2012 > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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/20120706/036dde39/attachment.html From all.eforums at gmail.com Sat Jul 7 05:57:53 2012 From: all.eforums at gmail.com (A E G) Date: Fri, 6 Jul 2012 21:57:53 -0400 Subject: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a In-Reply-To: <1341617252.56013.YahooMailNeo@web39301.mail.mud.yahoo.com> References: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern> <4FED4DA1.10304@belrosbank.by> <1341617252.56013.YahooMailNeo@web39301.mail.mud.yahoo.com> Message-ID: On Fri, Jul 6, 2012 at 7:27 PM, Stanislav Sinyagin wrote: > OpenVZ and XEN VPS'es are usually fine for quick prototyping and feature > tests. > Also they would be fine for private PBXes with low call volumes. > For hundreds of simultaneous calls, it's probably cheaper to rent > dedicated hardware servers. > On the other side, if you have some smart load balancing, probably you can > build a scalable design with VPS'es and eventually migrate to physical > servers when volumes are high enough. > > also for a calling card business, probably the most critical CPU resource > would be needed for the IVR and real-time billing, but not for the calls > themselves. > > Thanks Anthony and Stanislav. I think I have enough warning along with some encouraging words to build a strategy. Thanks again., Now the search for the right IaaS provider begins. > > > > ------------------------------ > *From:* A E G > *To:* FreeSWITCH Users Help > *Sent:* Saturday, July 7, 2012 12:23 AM > *Subject:* Re: [Freeswitch-users] RTP delay > 100ms by performance > testing with freeswitch as trancoding device SILK to G711a > > On Fri, Jul 6, 2012 at 5:43 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > > I would be insane to endorse using it in a virtual world. The support > expectations would consume the rest of my life. > I try to make the code work well on real boxes and if it works on > virtual ones then that's a bonus =D > > We have good luck with openVZ since its a single real kernel and > virtual runtimes but either way we do not recommend it so its a use at > your own risk kind of thing. > > > Yup agree and acknowledge that you wouldn't endorse or offer to support > it, and nor should you. Simply trying to pick your (and anyone else who > wants to chime in) brain on the topic. Totally PoC'ing it at my own risk :) > > So, in that spirit, just one last message about it. Just like sharing the > experience with OpenVZ, let's just say that I am insane and I still want to > try it in a private cloud. Would you say the resource consumption of FS > (assuming it's just receiving the call, having it routed by a custom > routing engine via ESL, and switching it) be more than that in the "real" > world for the same call volume barring any clock sync/skew/jitter issues? > Would something like Cloudstack on hardware (like the one you recommended > earlier) with a FS VM with 12 vcpu cores, 24GB RAM and DAS be able to > handle something like 300-400 concurrent calls or am I out of my mind? > > Thx > > > > On Fri, Jul 6, 2012 at 3:59 PM, A E G wrote: > > Yup got that, and are you saying the same is applicable and true even if > one > > was to try and run FreeSWITCH on a private cloud or in a virtual > > environment? assuming of course that those tips would then apply to the > > machine on which the Hypervisor will run. > > > > Just trying to get a handle on whether running FreeSWITCH to do something > > like wholesale or calling card traffic in a purely virtual environment > is / > > has proven to work. You probably know the most in terms of all different > > environments people are running FS in, and if you (or they) have pointers > > specific to it being run in a virtual environment. > > > > Thanks so much > > > > On Fri, Jul 6, 2012 at 12:54 PM, Anthony Minessale > > wrote: > >> > >> Like I said "Really Really Nice Motherboard", as many CPU as > >> possible/affordable, and a good chunk of RAM. > >> Motherboard is the most important, a cheap motherboard with a ton of > >> cores is a waste. > >> > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120706/8ad75a3e/attachment-0001.html From h.maghsoudy at gmail.com Sat Jul 7 11:46:16 2012 From: h.maghsoudy at gmail.com (Hanie Maghsoudy) Date: Sat, 7 Jul 2012 12:16:16 +0430 Subject: [Freeswitch-users] Performance Testing Message-ID: Hi all, I searched for FreeSwitch call capacity, but most of the results wasn't new. So, I wanna ask if anybody has either tested FreeSwitch's performance recently, or got a dramatic result in real environment? I tested call quality on this machine: Virtual FreeSwitch server OS: CentOS release 6.2 - x86_64 CPU: 8 processor - Intel(R) Xeon(R) CPU X5670 @ 2.93GHz Memory: 8 G After receiving incoming calls, FreeSwitch routed them to another sip server, without transcoding. The other server transmitted calls by playing an audio file. Meanwhile, I called an extension in FreeSwitch to test the call quality. The result was like this: 1000 Concurrent calls Call duration: 160s Call rate: 6 cps (just creating channels) Max used Memory: 1416M Max CPU load: 0.24 Max Network throughput (recv/send): 6711k/80k Quality: Good This test was taken before tearing down the channels. Then, I took another test to estimate calls per second, and it wasn't what I was expected! 150 Concurrent calls Call duration: 4s Call rate: 30 cps (creating and tearing down) Max used Memory: 1293M Max CPU load: 4.50 Max Network throughput (recv/send): 828k/60k Quality: Average And when I increase call rate to 50 cps: 1000 Concurrent calls Call duration: 4s Call rate: 50 cps (creating and tearing down) Max used Memory: 1730M Max CPU load: *29.9* Max Network throughput (recv/send): 1367k/202k Quality: Bad Why call per second is such a big problem? Did anyone get a better result on this? Thanks, Hanie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120707/ff24c2f5/attachment.html From peter.olsson at visionutveckling.se Sat Jul 7 12:05:04 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sat, 7 Jul 2012 08:05:04 +0000 Subject: [Freeswitch-users] Performance Testing In-Reply-To: References: Message-ID: <1FFF97C269757C458224B7C895F35F151318A5@cantor.std.visionutv.se> Before trying anything else, use real hardware instead of virtual. A virtual server with this kind of load will only cause you trouble. Actually - most virtual solutions will cause you problems :) Also, performance with average call duration of 4s is not really anything that would happen in real life. But first of all - get real hardware. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Hanie Maghsoudy [h.maghsoudy at gmail.com] Skickat: den 7 juli 2012 09:46 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Performance Testing Hi all, I searched for FreeSwitch call capacity, but most of the results wasn't new. So, I wanna ask if anybody has either tested FreeSwitch's performance recently, or got a dramatic result in real environment? I tested call quality on this machine: Virtual FreeSwitch server OS: CentOS release 6.2 - x86_64 CPU: 8 processor - Intel(R) Xeon(R) CPU X5670 @ 2.93GHz Memory: 8 G After receiving incoming calls, FreeSwitch routed them to another sip server, without transcoding. The other server transmitted calls by playing an audio file. Meanwhile, I called an extension in FreeSwitch to test the call quality. The result was like this: 1000 Concurrent calls Call duration: 160s Call rate: 6 cps (just creating channels) Max used Memory: 1416M Max CPU load: 0.24 Max Network throughput (recv/send): 6711k/80k Quality: Good This test was taken before tearing down the channels. Then, I took another test to estimate calls per second, and it wasn't what I was expected! 150 Concurrent calls Call duration: 4s Call rate: 30 cps (creating and tearing down) Max used Memory: 1293M Max CPU load: 4.50 Max Network throughput (recv/send): 828k/60k Quality: Average And when I increase call rate to 50 cps: 1000 Concurrent calls Call duration: 4s Call rate: 50 cps (creating and tearing down) Max used Memory: 1730M Max CPU load: 29.9 Max Network throughput (recv/send): 1367k/202k Quality: Bad Why call per second is such a big problem? Did anyone get a better result on this? Thanks, Hanie !DSPAM:4ff7e86232761844718139! From stefanshortner at yahoo.com Sat Jul 7 09:21:39 2012 From: stefanshortner at yahoo.com (Stefan Shortner) Date: Sat, 7 Jul 2012 06:21:39 +0100 (BST) Subject: [Freeswitch-users] how to prevent of changing the value of nonce ttl sip auth ? In-Reply-To: <1341638039.64151.YahooMailNeo@web133001.mail.ir2.yahoo.com> References: <1341638039.64151.YahooMailNeo@web133001.mail.ir2.yahoo.com> Message-ID: <1341638499.8030.YahooMailNeo@web133003.mail.ir2.yahoo.com> HI, i find out every time we want to register on sipclient the sip_nonce_auth? get the value,and every time we unregister the same sipclient the value of sip_nonce_auth was changed ! but is there any solutions to prevent of changing the value of sip_nonce_auth for same sipclient that registered and unregistered? because suppose we have 25 sip client that is registered on our server,now i want to check the sip client that is registered be the same that is unregistered so if the value of sip_nonce_auth for the sipcliet be the equal with the value of the same sipclient that is unregister ,the problem will solve, in the internal.xml we have it means every 60sec the value of? sip_nonce_auth for each sipclient was chnaged , and if the value of? nonce-ttl is less than the value of? parametere "register every ... " softphone, so the nonce-ttl was changed again. but i don't want change to check every sip client that is registered is the same that is unregistered i want when the status of sipclient was changed to 'unregistered' form 'registered' the value of 'sip_nonce_auth' be the same? to find out the sip client that is registered was the same that is unregistered ! is there any solutions? ( i know we can check using fs_cli with sofia status profile internal reg and sofia_count_reg ..) but i want to check with sip header ... for example when sipclient is registered 'Event-Name: REQUEST_PARAMS Core-UUID: 86fdb0e4-c6cd-11e1-a155-c382209fd02d FreeSWITCH-Hostname: PBX FreeSWITCH-Switchname: PBX FreeSWITCH-IPv4: 192.168.10.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-07-06%2009%3A42%3A44 Event-Date-GMT: Fri,%2006%20Jul%202012%2005%3A12%3A44%20GMT Event-Date-Timestamp: 1341551564411676 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_parse_auth Event-Calling-Line-Number: 2364 Event-Sequence: 5610 action: sip_auth sip_profile: internal sip_user_agent: eyeBeam%20release%201102q%20stamp%2051814 sip_auth_username: jack sip_auth_realm: 192.168.10.89 sip_auth_nonce: 37cb2234-c729-11e1-a170-c382209fd02d sip_auth_uri: sip%3A192.168.10.1 sip_contact_user: jack sip_contact_host: 192.168.12.1 sip_to_user: jack sip_to_host: 192.168.10.1 sip_from_user: jack sip_from_host: 192.168.10.1 sip_request_host: 192.168.10.1 sip_auth_qop: auth sip_auth_cnonce: c3f4cce1becf9f7fb6b20bafc27e2c68 sip_auth_nc: 00000001 sip_auth_response: 5489fc3afb45d7a826b865eef559baf5 sip_auth_method: REGISTER key: id user: jack domain: 192.168.10.1 ip: 192.168.12.1 when the same sip client is unregistered: 'Event-Name: REQUEST_PARAMS Core-UUID: 86fdb0e4-c6cd-11e1-a155-c382209fd02d FreeSWITCH-Hostname: PBX FreeSWITCH-Switchname: PBX FreeSWITCH-IPv4: 192.168.10.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-07-06%2009%3A49%3A56 Event-Date-GMT: Fri,%2006%20Jul%202012%2005%3A19%3A56%20GMT Event-Date-Timestamp: 1341551996651662 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_parse_auth Event-Calling-Line-Number: 2364 Event-Sequence: 5610 action: sip_auth sip_profile: internal sip_user_agent: eyeBeam%20release%201102q%20stamp%2051814 sip_auth_username:jack sip_auth_realm: 192.168.10.89 sip_auth_nonce:397d1b86-c72a-11e1-a171-c382209fd02d sip_auth_uri: sip%3A192.168.10.1 sip_contact_user:jack sip_contact_host: 192.168.12.1 sip_to_user: jack sip_to_host: 192.168.10.1 sip_from_user: jack sip_from_host: 192.168.10.1 sip_request_host: 192.168.10.1 sip_auth_qop: auth sip_auth_cnonce: c3f4cce1becf9f7fb6b20bafc27e2c68 sip_auth_nc: 00000001 sip_auth_response: 5489fc3afb45d7a826b865eef559baf5 sip_auth_method: REGISTER key: id user: jack domain: 192.168.10.1 ip: 192.168.12.1 as you see the sip_auth_method is the same when user is registered AND unregistered ... but how can i findout? that ? is there any method to change the 'sip_auth_method ' to another value when sipclient status was changed to 'unregistered'? thanks in advanced -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120707/4969d44d/attachment.html From motosota at gmail.com Sat Jul 7 12:25:45 2012 From: motosota at gmail.com (Mike) Date: Sat, 7 Jul 2012 09:25:45 +0100 Subject: [Freeswitch-users] Problem with Polycom behind firewall with freeswitch In-Reply-To: <005001cd5b18$4bc6b9b0$e3542d10$@com> References: <005001cd5b18$4bc6b9b0$e3542d10$@com> Message-ID: Akhil, You might want to have a look at this http://lists.freeswitch.org/pipermail/freeswitch-users/2011-November/077513.html Alternatively (what I do) - get the phone to send the keepalive, not the other way around. Set the following in the Polycom phone .xml configuration file: >From the manual: "If non-Null (or 0), the keepalive interval in seconds. This parameter is used to set the interval at which phones will send a keep-alive packet to the gateway/NAT device to keep the communication port open so that NAT can continue to function as setup initially." Default value is 0. Mike On Fri, Jul 6, 2012 at 2:40 AM, Akhil Chandra wrote: > Hi all, > > > > I come from a asterisk background and having trouble keeping the firewall > hole opened up for Polycom registering to freeswitch server. > > > > I am running freeswitch ( with bluebox UI) and trying to register a Polycom > behind a router. > > > > I have exhausted all the options in sip_profiles directory for the interface > I am using. The issue I am seeing is that the freeswitch sends the OPTIONS > message to the internal IP ie ( 192.168.X.X) rather than the public IP, > behind which the Polycom phones are. > > These phones register fine with asterisk in the stated scenario. > > > > I have tried changing these options in the xml file but still the OPTIONS > message ( as seen in tcpdump) is sent to 192.168.X.X > > > > > > > > > > > > > > > > Any help appreciated !! .. Thanks in advance. > > > > Regards, > > Akhil > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From alex at thewinelake.com Sat Jul 7 12:49:52 2012 From: alex at thewinelake.com (Alexander Lake) Date: Sat, 7 Jul 2012 09:49:52 +0100 Subject: [Freeswitch-users] Performance Testing In-Reply-To: <1FFF97C269757C458224B7C895F35F151318A5@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F151318A5@cantor.std.visionutv.se> Message-ID: <2D510CB3-B1D3-48E3-8AA6-6C7F47FCEE4A@thewinelake.com> This is an interesting area. I would be fascinated to compare real with virtual hardware. We're deploying some fs hosts on "the cloud" and some of these hosted systems are pretty virtualised (eg. Linode uses Xen XPS) and some aren't (Hetzner being an example). It would be nice to start an area of the Wiki for this. It may be fairly random/adhoc contributions to start with, but I'm sure a structure could emerge with time. Any scripts used (with instructions) that could quickly be deployed to a host somewhere would be a good start. Also sound recordings of sound quality (together, perhaps, with measurements of latency) would help improve the authority of results. We would most probably be prepared to donate a day or so of testing. On 7 Jul 2012, at 09:05, Peter Olsson wrote: > Before trying anything else, use real hardware instead of virtual. A virtual server with this kind of load will only cause you trouble. Actually - most virtual solutions will cause you problems :) > > Also, performance with average call duration of 4s is not really anything that would happen in real life. > > But first of all - get real hardware. > > /Peter > > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Hanie Maghsoudy [h.maghsoudy at gmail.com] > Skickat: den 7 juli 2012 09:46 > Till: FreeSWITCH Users Help > ?mne: [Freeswitch-users] Performance Testing > > Hi all, > > I searched for FreeSwitch call capacity, but most of the results wasn't new. So, I wanna ask if anybody has either tested FreeSwitch's performance recently, or got a dramatic result in real environment? > > I tested call quality on this machine: > > Virtual FreeSwitch server > OS: CentOS release 6.2 - x86_64 > CPU: 8 processor - Intel(R) Xeon(R) CPU X5670 @ 2.93GHz > Memory: 8 G > > After receiving incoming calls, FreeSwitch routed them to another sip server, without transcoding. The other server transmitted calls by playing an audio file. Meanwhile, I called an extension in FreeSwitch to test the call quality. > > The result was like this: > > 1000 Concurrent calls > Call duration: 160s > Call rate: 6 cps (just creating channels) > Max used Memory: 1416M > Max CPU load: 0.24 > Max Network throughput (recv/send): 6711k/80k > Quality: Good > > This test was taken before tearing down the channels. > > Then, I took another test to estimate calls per second, and it wasn't what I was expected! > > > 150 Concurrent calls > Call duration: 4s > Call rate: 30 cps (creating and tearing down) > Max used Memory: 1293M > Max CPU load: 4.50 > Max Network throughput (recv/send): 828k/60k > Quality: Average > > And when I increase call rate to 50 cps: > > 1000 Concurrent calls > Call duration: 4s > Call rate: 50 cps (creating and tearing down) > Max used Memory: 1730M > Max CPU load: 29.9 > Max Network throughput (recv/send): 1367k/202k > Quality: Bad > > Why call per second is such a big problem? Did anyone get a better result on this? > > Thanks, > Hanie > > !DSPAM:4ff7e86232761844718139! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From h.maghsoudy at gmail.com Sat Jul 7 13:18:44 2012 From: h.maghsoudy at gmail.com (Hanie Maghsoudy) Date: Sat, 7 Jul 2012 13:48:44 +0430 Subject: [Freeswitch-users] Performance Testing In-Reply-To: <1FFF97C269757C458224B7C895F35F151318A5@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F151318A5@cantor.std.visionutv.se> Message-ID: Thanks Peter, I will definitely test with a real server too. But, first of all, I'm really interested to know that what the result looks like for the others (perhaps with real servers in real environment). On Sat, Jul 7, 2012 at 12:35 PM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Before trying anything else, use real hardware instead of virtual. A > virtual server with this kind of load will only cause you trouble. Actually > - most virtual solutions will cause you problems :) > > Also, performance with average call duration of 4s is not really anything > that would happen in real life. > > But first of all - get real hardware. > > /Peter > > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] f?r Hanie Maghsoudy [ > h.maghsoudy at gmail.com] > Skickat: den 7 juli 2012 09:46 > Till: FreeSWITCH Users Help > ?mne: [Freeswitch-users] Performance Testing > > Hi all, > > I searched for FreeSwitch call capacity, but most of the results wasn't > new. So, I wanna ask if anybody has either tested FreeSwitch's performance > recently, or got a dramatic result in real environment? > > I tested call quality on this machine: > > Virtual FreeSwitch server > OS: CentOS release 6.2 - x86_64 > CPU: 8 processor - Intel(R) Xeon(R) CPU X5670 @ 2.93GHz > Memory: 8 G > > After receiving incoming calls, FreeSwitch routed them to another sip > server, without transcoding. The other server transmitted calls by playing > an audio file. Meanwhile, I called an extension in FreeSwitch to test the > call quality. > > The result was like this: > > 1000 Concurrent calls > Call duration: 160s > Call rate: 6 cps (just creating channels) > Max used Memory: 1416M > Max CPU load: 0.24 > Max Network throughput (recv/send): 6711k/80k > Quality: Good > > This test was taken before tearing down the channels. > > Then, I took another test to estimate calls per second, and it wasn't what > I was expected! > > > 150 Concurrent calls > Call duration: 4s > Call rate: 30 cps (creating and tearing down) > Max used Memory: 1293M > Max CPU load: 4.50 > Max Network throughput (recv/send): 828k/60k > Quality: Average > > And when I increase call rate to 50 cps: > > 1000 Concurrent calls > Call duration: 4s > Call rate: 50 cps (creating and tearing down) > Max used Memory: 1730M > Max CPU load: 29.9 > Max Network throughput (recv/send): 1367k/202k > Quality: Bad > > Why call per second is such a big problem? Did anyone get a better result > on this? > > Thanks, > Hanie > > !DSPAM:4ff7e86232761844718139! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120707/7992d5b3/attachment.html From peter.olsson at visionutveckling.se Sat Jul 7 14:16:18 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sat, 7 Jul 2012 10:16:18 +0000 Subject: [Freeswitch-users] Performance Testing In-Reply-To: References: <1FFF97C269757C458224B7C895F35F151318A5@cantor.std.visionutv.se>, Message-ID: <1FFF97C269757C458224B7C895F35F15131936@cantor.std.visionutv.se> The common answer is "it depends".. It depends on many different aspects, for instance: transcoding, script handling, media bypass etc. So there is no answer that is valid for everyone. There are also easy changes that can be made to improve performance, putting the sqlite databases on a RAM-drive is one common task. Also make sure to have as many CPU cores as possible, lots of RAM, and med sure to use 64-bit systems. I've never done any testing myself, so I don't know really. But I know there are people out there handling thousands of concurrent calls, at pretty high CPS. One more thing though, that will help you out right now. FS doesn't behave 100% correctly in CentOS 6.x, CentOS 6.x causes much higher CPU load, compared to CentOX 5.x. So right now CentOS 6.x is not officially supported by FS (until the reasons for this has been discovered). If you switch to real hardware and use CentOS 5.x I'm pretty sure that will give you much better results. These issues on Jira are related to CentOS 6.x problems. FS-4396 FS-4316 /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Hanie Maghsoudy [h.maghsoudy at gmail.com] Skickat: den 7 juli 2012 11:18 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Performance Testing Thanks Peter, I will definitely test with a real server too. But, first of all, I'm really interested to know that what the result looks like for the others (perhaps with real servers in real environment). On Sat, Jul 7, 2012 at 12:35 PM, Peter Olsson > wrote: Before trying anything else, use real hardware instead of virtual. A virtual server with this kind of load will only cause you trouble. Actually - most virtual solutions will cause you problems :) Also, performance with average call duration of 4s is not really anything that would happen in real life. But first of all - get real hardware. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Hanie Maghsoudy [h.maghsoudy at gmail.com] Skickat: den 7 juli 2012 09:46 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Performance Testing Hi all, I searched for FreeSwitch call capacity, but most of the results wasn't new. So, I wanna ask if anybody has either tested FreeSwitch's performance recently, or got a dramatic result in real environment? I tested call quality on this machine: Virtual FreeSwitch server OS: CentOS release 6.2 - x86_64 CPU: 8 processor - Intel(R) Xeon(R) CPU X5670 @ 2.93GHz Memory: 8 G After receiving incoming calls, FreeSwitch routed them to another sip server, without transcoding. The other server transmitted calls by playing an audio file. Meanwhile, I called an extension in FreeSwitch to test the call quality. The result was like this: 1000 Concurrent calls Call duration: 160s Call rate: 6 cps (just creating channels) Max used Memory: 1416M Max CPU load: 0.24 Max Network throughput (recv/send): 6711k/80k Quality: Good This test was taken before tearing down the channels. Then, I took another test to estimate calls per second, and it wasn't what I was expected! 150 Concurrent calls Call duration: 4s Call rate: 30 cps (creating and tearing down) Max used Memory: 1293M Max CPU load: 4.50 Max Network throughput (recv/send): 828k/60k Quality: Average And when I increase call rate to 50 cps: 1000 Concurrent calls Call duration: 4s Call rate: 50 cps (creating and tearing down) Max used Memory: 1730M Max CPU load: 29.9 Max Network throughput (recv/send): 1367k/202k Quality: Bad Why call per second is such a big problem? Did anyone get a better result on this? Thanks, Hanie _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:4ff7fdea32761519618044! From h.maghsoudy at gmail.com Sat Jul 7 14:27:53 2012 From: h.maghsoudy at gmail.com (Hanie Maghsoudy) Date: Sat, 7 Jul 2012 14:57:53 +0430 Subject: [Freeswitch-users] Performance Testing In-Reply-To: <1FFF97C269757C458224B7C895F35F15131936@cantor.std.visionutv.se> References: <1FFF97C269757C458224B7C895F35F151318A5@cantor.std.visionutv.se> <1FFF97C269757C458224B7C895F35F15131936@cantor.std.visionutv.se> Message-ID: That was really informative. Thanks alot. On Sat, Jul 7, 2012 at 2:46 PM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > The common answer is "it depends".. > > It depends on many different aspects, for instance: transcoding, script > handling, media bypass etc. So there is no answer that is valid for > everyone. There are also easy changes that can be made to improve > performance, putting the sqlite databases on a RAM-drive is one common task. > > Also make sure to have as many CPU cores as possible, lots of RAM, and med > sure to use 64-bit systems. > > I've never done any testing myself, so I don't know really. But I know > there are people out there handling thousands of concurrent calls, at > pretty high CPS. > > One more thing though, that will help you out right now. FS doesn't behave > 100% correctly in CentOS 6.x, CentOS 6.x causes much higher CPU load, > compared to CentOX 5.x. So right now CentOS 6.x is not officially supported > by FS (until the reasons for this has been discovered). If you switch to > real hardware and use CentOS 5.x I'm pretty sure that will give you much > better results. > > These issues on Jira are related to CentOS 6.x problems. > > FS-4396 > FS-4316 > > /Peter > > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] f?r Hanie Maghsoudy [ > h.maghsoudy at gmail.com] > Skickat: den 7 juli 2012 11:18 > Till: FreeSWITCH Users Help > ?mne: Re: [Freeswitch-users] Performance Testing > > Thanks Peter, I will definitely test with a real server too. But, first of > all, I'm really interested to know that what the result looks like for the > others (perhaps with real servers in real environment). > > On Sat, Jul 7, 2012 at 12:35 PM, Peter Olsson < > peter.olsson at visionutveckling.se> > wrote: > Before trying anything else, use real hardware instead of virtual. A > virtual server with this kind of load will only cause you trouble. Actually > - most virtual solutions will cause you problems :) > > Also, performance with average call duration of 4s is not really anything > that would happen in real life. > > But first of all - get real hardware. > > /Peter > > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org> [ > freeswitch-users-bounces at lists.freeswitch.org freeswitch-users-bounces at lists.freeswitch.org>] f?r Hanie Maghsoudy [ > h.maghsoudy at gmail.com] > Skickat: den 7 juli 2012 09:46 > Till: FreeSWITCH Users Help > ?mne: [Freeswitch-users] Performance Testing > > Hi all, > > I searched for FreeSwitch call capacity, but most of the results wasn't > new. So, I wanna ask if anybody has either tested FreeSwitch's performance > recently, or got a dramatic result in real environment? > > I tested call quality on this machine: > > Virtual FreeSwitch server > OS: CentOS release 6.2 - x86_64 > CPU: 8 processor - Intel(R) Xeon(R) CPU X5670 @ 2.93GHz > Memory: 8 G > > After receiving incoming calls, FreeSwitch routed them to another sip > server, without transcoding. The other server transmitted calls by playing > an audio file. Meanwhile, I called an extension in FreeSwitch to test the > call quality. > > The result was like this: > > 1000 Concurrent calls > Call duration: 160s > Call rate: 6 cps (just creating channels) > Max used Memory: 1416M > Max CPU load: 0.24 > Max Network throughput (recv/send): 6711k/80k > Quality: Good > > This test was taken before tearing down the channels. > > Then, I took another test to estimate calls per second, and it wasn't what > I was expected! > > > 150 Concurrent calls > Call duration: 4s > Call rate: 30 cps (creating and tearing down) > Max used Memory: 1293M > Max CPU load: 4.50 > Max Network throughput (recv/send): 828k/60k > Quality: Average > > And when I increase call rate to 50 cps: > > 1000 Concurrent calls > Call duration: 4s > Call rate: 50 cps (creating and tearing down) > Max used Memory: 1730M > Max CPU load: 29.9 > Max Network throughput (recv/send): 1367k/202k > Quality: Bad > > Why call per second is such a big problem? Did anyone get a better result > on this? > > Thanks, > Hanie > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4ff7fdea32761519618044! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120707/6f081bf2/attachment-0001.html From mgg at giagnocavo.net Sat Jul 7 14:56:47 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Sat, 7 Jul 2012 10:56:47 +0000 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: <537ACDFCE00243439E851C86C8C25E7A@freeswitch.org> References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <537ACDFCE00243439E851C86C8C25E7A@freeswitch.org> Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B604510494@BLUPRD0711MB413.namprd07.prod.outlook.com> But 0MQ is just the transport. You still need a broker to persist messages, participate in transactions, and so on. That?s what I was referring to, not just the transport layer acknowledgements. So if you?re just taking CDRs from FS memory and sending them out over 0MQ, what if the other side is down? In that case, aren?t you risking a lot of data, should something happen before the receiver gets up and saves all the CDRs? 0MQ isn?t really an alternative to a broker like RabbitMQ or SSSB. CDRs should be going to a local, durable, message queue. This doesn?t mean that you need to hit disk; the messages can stay in RAM for a few seconds, under the assumption that during normal ops, they?ll get delivered immediately and there?s no need for local persistence. But when something breaks, then the CDRs can just pile up on disk, and get pumped out whenever things stabilize. Am I missing something? -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita Sent: Thursday, July 05, 2012 3:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Pushing CDR Information into Zeromq Michael, You are partially correct because it depends on the socket type you use on ZeroMQ. Since the socket implemented on mod_zmq is a simple PUB/SUB, yes, messages would be discarded if they are not received by the other end. If we would use a XREQ/XREP then it would be queued until you are able to receive them. -- Jo?o Mesquita Sent with Sparrow On Thursday, July 5, 2012 at 3:08 PM, Michael Giagnocavo wrote: Wouldn?t the problem with ZeroMQ be no reliable delivery of CDRs? I was under the impression that 0MQ was more for high-performance, and things like guaranteed delivery are left as an exercise to the user. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita Sent: Thursday, July 05, 2012 9:56 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Pushing CDR Information into Zeromq I haven't actually looked at the code, but you have to look at the variables on CHANNEL_REPORTING for example. There you will see all the vars needed to produce CDR entries, but they won't be on XML format like xml_curl would create. Regards, -- Jo?o Mesquita Sent with Sparrow On Thursday, July 5, 2012 at 10:37 AM, Neo Cheema wrote: Hi all, I was hoping to find a way to push CDR information into a queue. Zeromq comes out as an obvious choice because mod_event_zmq already exits. However, I can't find a way to configure this module to push CDR info. Have any of you guys tried it? Regards _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120707/d92311ec/attachment-0001.html From mgg at giagnocavo.net Sat Jul 7 14:56:48 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Sat, 7 Jul 2012 10:56:48 +0000 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <9343F57A41054F8DB89BD4A0C17CFD7B@gmail.com> <0B4DC9FFE198449888339EA7BA4E3E24@freeswitch.org> Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B60451049B@BLUPRD0711MB413.namprd07.prod.outlook.com> Regarding "safe" variables like billsec: The issue is that most of those things are not calculated and set until later on in the lifecycle of a request, well after your Lua script has run. You can access them in C by adding a state handler, but I don't believe that's easily exposed to "API" apps/scripts. Anthony humoured me a quite a bit trying to get that kind of stuff working, but (at least a while ago), it was just not feasible. I would suggest that your Lua script add in its own data variables that it would use in billing calculations, then let the full CDR get written, then pick up processing again, restore the vars from the CDR, and continue calculations. I know, it's not the most elegant seeming approach, but it's the only reliable way AFAIK. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Friday, July 06, 2012 4:02 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Pushing CDR Information into Zeromq I'm pretty sure there's a variable to have the CDR in the ESL events, but I can't find it. -Avi On Fri, Jul 6, 2012 at 12:42 PM, Neo Cheema > wrote: Avi, Yea. That would work as well. I just thought it would be a 'nice-to-have' feature. Guess I'll go ahead with your model. Regards On Fri, Jul 6, 2012 at 2:07 PM, Avi Marcus > wrote: > I process CDRs on the same machine and then replicate the DB elsewhere. > Or you could have a daemon processing the CDRs on the main machine and doing > whatever with them. > The benefit is that then you'll have file persistence of the CDRs until they > actually get processed... > > -Avi > > > On Fri, Jul 6, 2012 at 11:10 AM, Neo Cheema > wrote: >> >> Avi, >> >> What I really want is to provide near-real time updates of CDRs on a >> webpage, the webserver not necessarily being on the same machine as >> Freeswitch. I would hate to continuously scan DB on freeswitch machine >> for latest CDR info. >> >> In a related issue, I read that the only reliable info on "call >> length", billsec etc should be read from CDRs. >> >> I mean, I can't do a getvariable on session in say a Lua script, to >> read call length, billsec session variables. Infact I tried it and got >> null responses, even after session hangup. Is there a way to get >> "correct" values of these variables in the Lua script itself? It would >> make my life simpler, because I can then just push the cdr info + >> custom values into any queue or webserver, without waiting for the >> Freeswitch to do it for me. >> >> Any Ideas? >> >> >> On Fri, Jul 6, 2012 at 1:18 PM, Avi Marcus > wrote: >> > xml_cdr by default writes to disk, no web server required. You want to >> > skip >> > the disk, I presume..? >> > >> > -Avi >> > >> > >> > On Fri, Jul 6, 2012 at 10:14 AM, Neo Cheema > >> > wrote: >> >> >> >> Jo?o, >> >> >> >> "Since the socket implemented on mod_zmq is a simple PUB/SUB, yes, >> >> messages would be discarded if they are not received by the other >> >> end." >> >> >> >> I thought it didn't matter if the 'worker' was up before the >> >> 'producer'/freeswitch started pushing in the messages. 0mq handles >> >> that part, if atleast one worker is connected. >> >> >> >> My main reason for looking towards 0mq was to bypass a webserver, >> >> which would be required for xml_cdr. >> >> >> >> On Fri, Jul 6, 2012 at 8:33 AM, jay binks > wrote: >> >> > while we are talking about funky CDR stuff. >> >> > >> >> > Ive been planning to look at riak ( http://wiki.basho.com/ ) for bulk >> >> > CDR >> >> > storage >> >> > ( storing data other than billing data ) but just not finding the >> >> > time. >> >> > >> >> > anyone done something similar ? >> >> > The idea is to store as much data in a distributed manner, so as to >> >> > aid >> >> > debugging. >> >> > ( not so much for billing ) >> >> > >> >> > Jay >> >> > >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ >> >> > Professional FreeSWITCH Consulting Services: >> >> > consulting at freeswitch.org >> >> > http://www.freeswitchsolutions.com >> >> > >> >> > >> >> > >> >> > >> >> > Official FreeSWITCH Sites >> >> > http://www.freeswitch.org >> >> > http://wiki.freeswitch.org >> >> > http://www.cluecon.com >> >> > >> >> > Join Us At ClueCon - Aug 7-9, 2012 >> >> > >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > >> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> > http://www.freeswitch.org >> >> > >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120707/d4c62470/attachment-0001.html From jack.nikolas at ymail.com Sat Jul 7 07:34:10 2012 From: jack.nikolas at ymail.com (Jack Nikolas) Date: Sat, 7 Jul 2012 04:34:10 +0100 (BST) Subject: [Freeswitch-users] how to prevent of changing the value of nonce ttl sip auth ? In-Reply-To: <1341557452.85542.YahooMailNeo@web171505.mail.ir2.yahoo.com> References: <1341553250.66528.YahooMailNeo@web171504.mail.ir2.yahoo.com> <1341553363.25866.YahooMailNeo@web171506.mail.ir2.yahoo.com> <1341557452.85542.YahooMailNeo@web171505.mail.ir2.yahoo.com> Message-ID: <1341632050.50592.YahooMailNeo@web171506.mail.ir2.yahoo.com> ----- Forwarded Message ----- From: Jack Nikolas To: "freeswitch-users at lists.freeswitch.org" Sent: Friday, 6 July 2012, 11:20:52 Subject: Fw: how to prevent the changes of nonce ttl sip auth ? ----- Forwarded Message ----- From: Jack Nikolas To: "freeswitch-users at lists.freeswitch.org" Sent: Friday, 6 July 2012, 10:12:43 Subject: how to prevent the changes of nonce ttl sip auth ? ----- Forwarded Message ----- From: Jack Nikolas To: "freeswitch-users at lists.freeswitch.org" Sent: Friday, 6 July 2012, 10:10:50 Subject: how to prevent the changes of nonce ttl sip auth ? HI, i find out every time we want to register on sipclient the sip_nonce_auth? get the value,and every time we unregister the same sipclient the value of sip_nonce_auth was changed ! but is there any solutions to prevent of changing the value of sip_nonce_auth for same sipclient that registered and unregistered? because suppose we have 25 sip client that is registered on our server,now i want to check the sip client that is registered be the same that is unregistered so if the value of sip_nonce_auth for the sipcliet be the equal with the value of the same sipclient that is unregister ,the problem will solve, in the internal.xml we have it means every 60sec the value of? sip_nonce_auth for each sipclient was chnaged , and if the value of? nonce-ttl is less than the value of? parametere "register every ... " softphone, so the nonce-ttl was changed again. but i don't want change to check every sip client that is registered is the same that is unregistered i want when the status of sipclient was changed to 'unregistered' form 'registered' the value of 'sip_nonce_auth' be the same? to find out the sip client that is registered was the same that is unregistered ! is there any solutions? ( i know we can check using fs_cli with sofia status profile internal reg and sofia_count_reg ..) but i want to check with sip header ... for example when sipclient is registered 'Event-Name: REQUEST_PARAMS Core-UUID: 86fdb0e4-c6cd-11e1-a155-c382209fd02d FreeSWITCH-Hostname: PBX FreeSWITCH-Switchname: PBX FreeSWITCH-IPv4: 192.168.10.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-07-06%2009%3A42%3A44 Event-Date-GMT: Fri,%2006%20Jul%202012%2005%3A12%3A44%20GMT Event-Date-Timestamp: 1341551564411676 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_parse_auth Event-Calling-Line-Number: 2364 Event-Sequence: 5610 action: sip_auth sip_profile: internal sip_user_agent: eyeBeam%20release%201102q%20stamp%2051814 sip_auth_username: jack sip_auth_realm: 192.168.10.89 sip_auth_nonce: 37cb2234-c729-11e1-a170-c382209fd02d sip_auth_uri: sip%3A192.168.10.1 sip_contact_user: jack sip_contact_host: 192.168.12.1 sip_to_user: jack sip_to_host: 192.168.10.1 sip_from_user: jack sip_from_host: 192.168.10.1 sip_request_host: 192.168.10.1 sip_auth_qop: auth sip_auth_cnonce: c3f4cce1becf9f7fb6b20bafc27e2c68 sip_auth_nc: 00000001 sip_auth_response: 5489fc3afb45d7a826b865eef559baf5 sip_auth_method: REGISTER key: id user: jack domain: 192.168.10.1 ip: 192.168.12.1 when the same sip client is unregistered: 'Event-Name: REQUEST_PARAMS Core-UUID: 86fdb0e4-c6cd-11e1-a155-c382209fd02d FreeSWITCH-Hostname: PBX FreeSWITCH-Switchname: PBX FreeSWITCH-IPv4: 192.168.10.1 FreeSWITCH-IPv6: %3A%3A1 Event-Date-Local: 2012-07-06%2009%3A49%3A56 Event-Date-GMT: Fri,%2006%20Jul%202012%2005%3A19%3A56%20GMT Event-Date-Timestamp: 1341551996651662 Event-Calling-File: sofia_reg.c Event-Calling-Function: sofia_reg_parse_auth Event-Calling-Line-Number: 2364 Event-Sequence: 5610 action: sip_auth sip_profile: internal sip_user_agent: eyeBeam%20release%201102q%20stamp%2051814 sip_auth_username:jack sip_auth_realm: 192.168.10.89 sip_auth_nonce:397d1b86-c72a-11e1-a171-c382209fd02d sip_auth_uri: sip%3A192.168.10.1 sip_contact_user:jack sip_contact_host: 192.168.12.1 sip_to_user: jack sip_to_host: 192.168.10.1 sip_from_user: jack sip_from_host: 192.168.10.1 sip_request_host: 192.168.10.1 sip_auth_qop: auth sip_auth_cnonce: c3f4cce1becf9f7fb6b20bafc27e2c68 sip_auth_nc: 00000001 sip_auth_response: 5489fc3afb45d7a826b865eef559baf5 sip_auth_method: REGISTER key: id user: jack domain: 192.168.10.1 ip: 192.168.12.1 as you see the sip_auth_method is the same when user is registered AND unregistered ... but how can i findout? that ? is there any method to change the 'sip_auth_method ' to another value when sipclient status was changed to 'unregistered'? thanks in advanced -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120707/b7a47d1a/attachment.html From neo.cheema at gmail.com Sat Jul 7 16:26:50 2012 From: neo.cheema at gmail.com (Neo Cheema) Date: Sat, 7 Jul 2012 17:56:50 +0530 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B60451049B@BLUPRD0711MB413.namprd07.prod.outlook.com> References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <9343F57A41054F8DB89BD4A0C17CFD7B@gmail.com> <0B4DC9FFE198449888339EA7BA4E3E24@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B60451049B@BLUPRD0711MB413.namprd07.prod.outlook.com> Message-ID: I looked around, and mod_event_zmq does infact send "all" the events. So I just need to filter the incoming messages like, hangup_complete etc to generate CDR-like info at the other end. Ofcourse, the listener should be up all the time at the other end. So enabling mod_cdr_csv can handle disk persistence in case worker goes down. But I also liked the idea of generating 'billsec' like variables inside the script independently. Thanks for the inputs. On Sat, Jul 7, 2012 at 4:26 PM, Michael Giagnocavo wrote: > Regarding ?safe? variables like billsec: The issue is that most of those > things are not calculated and set until later on in the lifecycle of a > request, well after your Lua script has run. You can access them in C by > adding a state handler, but I don?t believe that?s easily exposed to ?API? > apps/scripts. Anthony humoured me a quite a bit trying to get that kind of > stuff working, but (at least a while ago), it was just not feasible. I would > suggest that your Lua script add in its own data variables that it would use > in billing calculations, then let the full CDR get written, then pick up > processing again, restore the vars from the CDR, and continue calculations. > I know, it?s not the most elegant seeming approach, but it?s the only > reliable way AFAIK. > > -Michael > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi > Marcus > Sent: Friday, July 06, 2012 4:02 AM > > > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Pushing CDR Information into Zeromq > > > > I'm pretty sure there's a variable to have the CDR in the ESL events, but I > can't find it. > > > > -Avi > > > > On Fri, Jul 6, 2012 at 12:42 PM, Neo Cheema wrote: > > Avi, > Yea. That would work as well. I just thought it would be a > 'nice-to-have' feature. Guess I'll go ahead with your model. > > Regards > > > On Fri, Jul 6, 2012 at 2:07 PM, Avi Marcus wrote: >> I process CDRs on the same machine and then replicate the DB elsewhere. >> Or you could have a daemon processing the CDRs on the main machine and >> doing >> whatever with them. >> The benefit is that then you'll have file persistence of the CDRs until >> they >> actually get processed... >> >> -Avi >> >> >> On Fri, Jul 6, 2012 at 11:10 AM, Neo Cheema wrote: >>> >>> Avi, >>> >>> What I really want is to provide near-real time updates of CDRs on a >>> webpage, the webserver not necessarily being on the same machine as >>> Freeswitch. I would hate to continuously scan DB on freeswitch machine >>> for latest CDR info. >>> >>> In a related issue, I read that the only reliable info on "call >>> length", billsec etc should be read from CDRs. >>> >>> I mean, I can't do a getvariable on session in say a Lua script, to >>> read call length, billsec session variables. Infact I tried it and got >>> null responses, even after session hangup. Is there a way to get >>> "correct" values of these variables in the Lua script itself? It would >>> make my life simpler, because I can then just push the cdr info + >>> custom values into any queue or webserver, without waiting for the >>> Freeswitch to do it for me. >>> >>> Any Ideas? >>> >>> >>> On Fri, Jul 6, 2012 at 1:18 PM, Avi Marcus wrote: >>> > xml_cdr by default writes to disk, no web server required. You want to >>> > skip >>> > the disk, I presume..? >>> > >>> > -Avi >>> > >>> > >>> > On Fri, Jul 6, 2012 at 10:14 AM, Neo Cheema >>> > wrote: >>> >> >>> >> Jo?o, >>> >> >>> >> "Since the socket implemented on mod_zmq is a simple PUB/SUB, yes, >>> >> messages would be discarded if they are not received by the other >>> >> end." >>> >> >>> >> I thought it didn't matter if the 'worker' was up before the >>> >> 'producer'/freeswitch started pushing in the messages. 0mq handles >>> >> that part, if atleast one worker is connected. >>> >> >>> >> My main reason for looking towards 0mq was to bypass a webserver, >>> >> which would be required for xml_cdr. >>> >> >>> >> On Fri, Jul 6, 2012 at 8:33 AM, jay binks wrote: >>> >> > while we are talking about funky CDR stuff. >>> >> > >>> >> > Ive been planning to look at riak ( http://wiki.basho.com/ ) for >>> >> > bulk >>> >> > CDR >>> >> > storage >>> >> > ( storing data other than billing data ) but just not finding the >>> >> > time. >>> >> > >>> >> > anyone done something similar ? >>> >> > The idea is to store as much data in a distributed manner, so as to >>> >> > aid >>> >> > debugging. >>> >> > ( not so much for billing ) >>> >> > >>> >> > Jay >>> >> > >>> >> > >>> >> > >>> >> > >>> >> > >>> >> > _________________________________________________________________________ >>> >> > Professional FreeSWITCH Consulting Services: >>> >> > consulting at freeswitch.org >>> >> > http://www.freeswitchsolutions.com >>> >> > >>> >> > >>> >> > >>> >> > >>> >> > Official FreeSWITCH Sites >>> >> > http://www.freeswitch.org >>> >> > http://wiki.freeswitch.org >>> >> > http://www.cluecon.com >>> >> > >>> >> > Join Us At ClueCon - Aug 7-9, 2012 >>> >> > >>> >> > FreeSWITCH-users mailing list >>> >> > FreeSWITCH-users at lists.freeswitch.org >>> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> > >>> >> > >>> >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> > http://www.freeswitch.org >>> >> > >>> >> >>> >> >>> >> >>> >> _________________________________________________________________________ >>> >> Professional FreeSWITCH Consulting Services: >>> >> consulting at freeswitch.org >>> >> http://www.freeswitchsolutions.com >>> >> >>> >> >>> >> >>> >> >>> >> Official FreeSWITCH Sites >>> >> http://www.freeswitch.org >>> >> http://wiki.freeswitch.org >>> >> http://www.cluecon.com >>> >> >>> >> Join Us At ClueCon - Aug 7-9, 2012 >>> >> >>> >> FreeSWITCH-users mailing list >>> >> FreeSWITCH-users at lists.freeswitch.org >>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> >> >>> >> >>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> >> http://www.freeswitch.org >>> > >>> > >>> > >>> > >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > Join Us At ClueCon - Aug 7-9, 2012 >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >>> > >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From gavin.henry at gmail.com Sat Jul 7 18:06:39 2012 From: gavin.henry at gmail.com (Gavin Henry) Date: Sat, 7 Jul 2012 15:06:39 +0100 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <9343F57A41054F8DB89BD4A0C17CFD7B@gmail.com> <0B4DC9FFE198449888339EA7BA4E3E24@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B60451049B@BLUPRD0711MB413.namprd07.prod.outlook.com> Message-ID: <3A08C74A7601400885B90885A5AB2AEC@gmail.com> If you check the links I sent earlier this is what we do. -- Gavin Henry Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Saturday, 7 July 2012 at 13:26, Neo Cheema wrote: > I looked around, and mod_event_zmq does infact send "all" the events. > So I just need to filter the incoming messages like, hangup_complete > etc to generate CDR-like info at the other end. Ofcourse, the listener > should be up all the time at the other end. So enabling mod_cdr_csv > can handle disk persistence in case worker goes down. > > But I also liked the idea of generating 'billsec' like variables > inside the script independently. Thanks for the inputs. > > On Sat, Jul 7, 2012 at 4:26 PM, Michael Giagnocavo wrote: > > Regarding ?safe? variables like billsec: The issue is that most of those > > things are not calculated and set until later on in the lifecycle of a > > request, well after your Lua script has run. You can access them in C by > > adding a state handler, but I don?t believe that?s easily exposed to ?API? > > apps/scripts. Anthony humoured me a quite a bit trying to get that kind of > > stuff working, but (at least a while ago), it was just not feasible. I would > > suggest that your Lua script add in its own data variables that it would use > > in billing calculations, then let the full CDR get written, then pick up > > processing again, restore the vars from the CDR, and continue calculations. > > I know, it?s not the most elegant seeming approach, but it?s the only > > reliable way AFAIK. > > > > -Michael > > > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi > > Marcus > > Sent: Friday, July 06, 2012 4:02 AM > > > > > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Pushing CDR Information into Zeromq > > > > > > > > I'm pretty sure there's a variable to have the CDR in the ESL events, but I > > can't find it. > > > > > > > > -Avi > > > > > > > > On Fri, Jul 6, 2012 at 12:42 PM, Neo Cheema wrote: > > > > Avi, > > Yea. That would work as well. I just thought it would be a > > 'nice-to-have' feature. Guess I'll go ahead with your model. > > > > Regards > > > > > > On Fri, Jul 6, 2012 at 2:07 PM, Avi Marcus wrote: > > > I process CDRs on the same machine and then replicate the DB elsewhere. > > > Or you could have a daemon processing the CDRs on the main machine and > > > doing > > > whatever with them. > > > The benefit is that then you'll have file persistence of the CDRs until > > > they > > > actually get processed... > > > > > > -Avi > > > > > > > > > On Fri, Jul 6, 2012 at 11:10 AM, Neo Cheema wrote: > > > > > > > > Avi, > > > > > > > > What I really want is to provide near-real time updates of CDRs on a > > > > webpage, the webserver not necessarily being on the same machine as > > > > Freeswitch. I would hate to continuously scan DB on freeswitch machine > > > > for latest CDR info. > > > > > > > > In a related issue, I read that the only reliable info on "call > > > > length", billsec etc should be read from CDRs. > > > > > > > > I mean, I can't do a getvariable on session in say a Lua script, to > > > > read call length, billsec session variables. Infact I tried it and got > > > > null responses, even after session hangup. Is there a way to get > > > > "correct" values of these variables in the Lua script itself? It would > > > > make my life simpler, because I can then just push the cdr info + > > > > custom values into any queue or webserver, without waiting for the > > > > Freeswitch to do it for me. > > > > > > > > Any Ideas? > > > > > > > > > > > > On Fri, Jul 6, 2012 at 1:18 PM, Avi Marcus wrote: > > > > > xml_cdr by default writes to disk, no web server required. You want to > > > > > skip > > > > > the disk, I presume..? > > > > > > > > > > -Avi > > > > > > > > > > > > > > > On Fri, Jul 6, 2012 at 10:14 AM, Neo Cheema > > > > > wrote: > > > > > > > > > > > > Jo?o, > > > > > > > > > > > > "Since the socket implemented on mod_zmq is a simple PUB/SUB, yes, > > > > > > messages would be discarded if they are not received by the other > > > > > > end." > > > > > > > > > > > > I thought it didn't matter if the 'worker' was up before the > > > > > > 'producer'/freeswitch started pushing in the messages. 0mq handles > > > > > > that part, if atleast one worker is connected. > > > > > > > > > > > > My main reason for looking towards 0mq was to bypass a webserver, > > > > > > which would be required for xml_cdr. > > > > > > > > > > > > On Fri, Jul 6, 2012 at 8:33 AM, jay binks wrote: > > > > > > > while we are talking about funky CDR stuff. > > > > > > > > > > > > > > Ive been planning to look at riak ( http://wiki.basho.com/ ) for > > > > > > > bulk > > > > > > > CDR > > > > > > > storage > > > > > > > ( storing data other than billing data ) but just not finding the > > > > > > > time. > > > > > > > > > > > > > > anyone done something similar ? > > > > > > > The idea is to store as much data in a distributed manner, so as to > > > > > > > aid > > > > > > > debugging. > > > > > > > ( not so much for billing ) > > > > > > > > > > > > > > Jay > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > > > Professional FreeSWITCH Consulting Services: > > > > > > > consulting at freeswitch.org > > > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > > > http://www.freeswitch.org > > > > > > > http://wiki.freeswitch.org > > > > > > > http://www.cluecon.com > > > > > > > > > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > > > > > > > > > FreeSWITCH-users mailing list > > > > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > > > > > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > > Professional FreeSWITCH Consulting Services: > > > > > > consulting at freeswitch.org > > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > > http://www.freeswitch.org > > > > > > http://wiki.freeswitch.org > > > > > > http://www.cluecon.com > > > > > > > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > > > > > > > FreeSWITCH-users mailing list > > > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > Professional FreeSWITCH Consulting Services: > > > > > consulting at freeswitch.org > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > http://www.freeswitch.org > > > > > http://wiki.freeswitch.org > > > > > http://www.cluecon.com > > > > > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > > > > > FreeSWITCH-users mailing list > > > > > FreeSWITCH-users at lists.freeswitch.org > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120707/0cc745c8/attachment-0001.html From gavin.henry at gmail.com Sat Jul 7 18:07:13 2012 From: gavin.henry at gmail.com (Gavin Henry) Date: Sat, 7 Jul 2012 15:07:13 +0100 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B604510494@BLUPRD0711MB413.namprd07.prod.outlook.com> References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <537ACDFCE00243439E851C86C8C25E7A@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B604510494@BLUPRD0711MB413.namprd07.prod.outlook.com> Message-ID: <6D07CDCCEAC84C73841519D1FB19955E@gmail.com> 0mq does hold the messages. -- Gavin Henry Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Saturday, 7 July 2012 at 11:56, Michael Giagnocavo wrote: > > But 0MQ is just the transport. You still need a broker to persist messages, participate in transactions, and so on. That?s what I was referring to, not just the transport layer acknowledgements. So if you?re just taking CDRs from FS memory and sending them out over 0MQ, what if the other side is down? In that case, aren?t you risking a lot of data, should something happen before the receiver gets up and saves all the CDRs? 0MQ isn?t really an alternative to a broker like RabbitMQ or SSSB. > > > > > > CDRs should be going to a local, durable, message queue. This doesn?t mean that you need to hit disk; the messages can stay in RAM for a few seconds, under the assumption that during normal ops, they?ll get delivered immediately and there?s no need for local persistence. But when something breaks, then the CDRs can just pile up on disk, and get pumped out whenever things stabilize. > > > > > > Am I missing something? > > > > > > -Michael > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita > Sent: Thursday, July 05, 2012 3:09 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Pushing CDR Information into Zeromq > > > > > > Michael, > > > > > > > You are partially correct because it depends on the socket type you use on ZeroMQ. Since the socket implemented on mod_zmq is a simple PUB/SUB, yes, messages would be discarded if they are not received by the other end. If we would use a XREQ/XREP then it would be queued until you are able to receive them. > > > > > > > > > -- > > > > Jo?o Mesquita > > > > Sent with Sparrow (http://www.sparrowmailapp.com/?sig) > > > > > > > > On Thursday, July 5, 2012 at 3:08 PM, Michael Giagnocavo wrote: > > > > Wouldn?t the problem with ZeroMQ be no reliable delivery of CDRs? I was under the impression that 0MQ was more for high-performance, and things like guaranteed delivery are left as an exercise to the user. > > > > > > -Michael > > > > > > > > > > > > From: freeswitch-users-bounces at lists.freeswitch.org (mailto:freeswitch-users-bounces at lists.freeswitch.org) [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita > > Sent: Thursday, July 05, 2012 9:56 AM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Pushing CDR Information into Zeromq > > > > > > > > > > > > I haven't actually looked at the code, but you have to look at the variables on CHANNEL_REPORTING for example. There you will see all the vars needed to produce CDR entries, but they won't be on XML format like xml_curl would create. > > > > > > > > > > > > > > Regards, > > > > > > > > > > > > > > > > > > -- > > > > > > > > Jo?o Mesquita > > > > > > > > Sent with Sparrow (http://www.sparrowmailapp.com/?sig) > > > > > > > > > > > > > > > > On Thursday, July 5, 2012 at 10:37 AM, Neo Cheema wrote: > > > > > > Hi all, > > > > > > > > > > > > > > > > > > > > > > > > I was hoping to find a way to push CDR information into a queue. > > > > > > > > > > > > Zeromq comes out as an obvious choice because mod_event_zmq already > > > > > > > > > > > > exits. However, I can't find a way to configure this module to push > > > > > > > > > > > > CDR info. Have any of you guys tried it? > > > > > > > > > > > > > > > > > > > > > > > > Regards > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > > > > > > > > Professional FreeSWITCH Consulting Services: > > > > > > > > > > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > > > > > > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > > > > > > > > http://www.freeswitch.org > > > > > > > > > > > > http://wiki.freeswitch.org > > > > > > > > > > > > http://www.cluecon.com > > > > > > > > > > > > > > > > > > > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > > > > > > > > > > > > > > > > > > > FreeSWITCH-users mailing list > > > > > > > > > > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > > > > Professional FreeSWITCH Consulting Services: > > > > > > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > > > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > > > > > http://www.freeswitch.org > > > > > > > > http://wiki.freeswitch.org > > > > > > > > http://www.cluecon.com > > > > > > > > > > > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > > > > > > > > > > > FreeSWITCH-users mailing list > > > > > > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120707/3bdbc9e7/attachment.html From steveayre at gmail.com Sat Jul 7 18:24:47 2012 From: steveayre at gmail.com (Steven Ayre) Date: Sat, 7 Jul 2012 15:24:47 +0100 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <9343F57A41054F8DB89BD4A0C17CFD7B@gmail.com> <0B4DC9FFE198449888339EA7BA4E3E24@freeswitch.org> Message-ID: <6B689347-7E46-4BC6-907A-EF257F50226F@gmail.com> > What I really want is to provide near-real time updates of CDRs on a > webpage, the webserver not necessarily being on the same machine as > Freeswitch. I would hate to continuously scan DB on freeswitch machine > for latest CDR info. There's no need for this. mod_xml_cdr can submit CDRs to any URL accessible from the FS host, that means it can submit CDRs directly to a script on your webserver. The only need to do any scanning on your FS server is to watch the err log dir so you can resubmit failed CDRs. That doesn't need to be a constant polling - it can be a cronjob, manual (after seeing and correcting the problem) or... a daemon that uses inotify to only wake up when a failed cdr has been written to disk. Of course as well as retries you want it to notify you in case it's a problem on the webserver that needs you to correct it. Steve on iPhone On 6 Jul 2012, at 09:10, Neo Cheema wrote: > Avi, > > What I really want is to provide near-real time updates of CDRs on a > webpage, the webserver not necessarily being on the same machine as > Freeswitch. I would hate to continuously scan DB on freeswitch machine > for latest CDR info. > > In a related issue, I read that the only reliable info on "call > length", billsec etc should be read from CDRs. > > I mean, I can't do a getvariable on session in say a Lua script, to > read call length, billsec session variables. Infact I tried it and got > null responses, even after session hangup. Is there a way to get > "correct" values of these variables in the Lua script itself? It would > make my life simpler, because I can then just push the cdr info + > custom values into any queue or webserver, without waiting for the > Freeswitch to do it for me. > > Any Ideas? > > > On Fri, Jul 6, 2012 at 1:18 PM, Avi Marcus wrote: >> xml_cdr by default writes to disk, no web server required. You want to skip >> the disk, I presume..? >> >> -Avi >> >> >> On Fri, Jul 6, 2012 at 10:14 AM, Neo Cheema wrote: >>> >>> Jo?o, >>> >>> "Since the socket implemented on mod_zmq is a simple PUB/SUB, yes, >>> messages would be discarded if they are not received by the other >>> end." >>> >>> I thought it didn't matter if the 'worker' was up before the >>> 'producer'/freeswitch started pushing in the messages. 0mq handles >>> that part, if atleast one worker is connected. >>> >>> My main reason for looking towards 0mq was to bypass a webserver, >>> which would be required for xml_cdr. >>> >>> On Fri, Jul 6, 2012 at 8:33 AM, jay binks wrote: >>>> while we are talking about funky CDR stuff. >>>> >>>> Ive been planning to look at riak ( http://wiki.basho.com/ ) for bulk >>>> CDR >>>> storage >>>> ( storing data other than billing data ) but just not finding the time. >>>> >>>> anyone done something similar ? >>>> The idea is to store as much data in a distributed manner, so as to aid >>>> debugging. >>>> ( not so much for billing ) >>>> >>>> Jay >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From nasida at live.ru Sat Jul 7 19:19:15 2012 From: nasida at live.ru (Yuriy Nasida) Date: Sat, 7 Jul 2012 19:19:15 +0400 Subject: [Freeswitch-users] sensibility of voice level for Mod_pocketsphinx In-Reply-To: References: Message-ID: Solved. Ok. I have recieved good advise in IRC, thanks. I am able to not break playing of session:streamFile by returning 'true' from OnInput function.Thus there is not matter that Mod_pocketsphinx generate "bad" events when I don't say anything, the session:streamFile continue the playing of file.I still don't know 100% what settings in pocketsphinx.conf.xml means but it works fine with default. From: nasida at live.ru To: freeswitch-users at lists.freeswitch.org Date: Fri, 6 Jul 2012 21:18:58 +0400 Subject: [Freeswitch-users] sensibility of voice level for Mod_pocketsphinx Hi guys, Whether somebody know how to lower sensibility of voice level for Mod_pocketsphinx. I play with "threshold" in pocketsphinx.conf.xml but I have not any effect.the Mod_pocketsphinx begins to generate event == "begin-speaking" when I don't say anything... Also i would like do NOT break the current playing of session:streamFile when events of Mod_pocketsphinx is being generated. Is it possible ? Also can somebody explain that this parameters means ? I understand approximately but I would like to know for certain. I have found nothing in wiki. Thanks. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120707/4516b324/attachment.html From nasida at live.ru Sat Jul 7 19:41:07 2012 From: nasida at live.ru (Yuriy Nasida) Date: Sat, 7 Jul 2012 19:41:07 +0400 Subject: [Freeswitch-users] Disable reinvite In-Reply-To: <20120706133712335045.de3036fd@klingler.net> References: <20120702125026817891.9461489f@klingler.net>, , <20120706133712335045.de3036fd@klingler.net> Message-ID: Why do you set false ? Do you want to send RTP media without FS, right ?So you can simply set in dialplan. > Date: Fri, 6 Jul 2012 13:37:12 +0200 > From: richard at klingler.net > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Disable reinvite > > Hmm... > > Doesn't help setting: > > > > Seems to be som eother problem... > > Internal phones are on network 10.0/16 and connect to FS via internal network... > But as soon RTP traffic is flowing, RTP traffic goes out via internet gateway > and back in into FS via its WAN IP. (Internet FW and FS WAN are one same /29 subnet). > > So somehow FS thinks the internal phones have a public IP and not RFC1918. > > > On Mon, 2 Jul 2012 18:57:18 +0400, Yuriy Nasida wrote: > > You have to use bypass_media for it. > > > > http://wiki.freeswitch.org/wiki/Bypass_Media > > > > > > > > > >> Date: Mon, 2 Jul 2012 12:50:26 +0200 > >> From: richard at klingler.net > >> To: freeswitch-users at lists.freeswitch.org > >> Subject: [Freeswitch-users] Disable reinvite > >> > >> Hello > >> > >> Just switched this weekend from asterisk to freeswitch on freebsd 9.0. > >> And so far I was able to implement basic dialplan for > > inbound/outbound calls (o; > >> > >> What I'm trying to find is a similar configuration option from > > asterisk sip.conf > >> where individual user agents can have SIP reinvite disbaled with > > "canreinvite=no". > >> > >> Fomr the Freeswitch documentation it states that all RTP streams > > are passed through > >> by default, but I still see SIP clients trying to send UDP streams > > to my box directly. > >> > >> Also setting "" in > > sofia.conf.xml > >> doesn't solve this problem. I just want the UDP streams frmo > > outside are only allowed > >> from my SIP trunk provider. > >> > >> > >> thanx in advance > >> richard > >> > >> > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120707/d9694cf1/attachment.html From red.rain.seven at gmail.com Sat Jul 7 21:55:37 2012 From: red.rain.seven at gmail.com (Henry Huang) Date: Sat, 7 Jul 2012 10:55:37 -0700 Subject: [Freeswitch-users] Performance Testing In-Reply-To: References: Message-ID: >From my experience playing with different CPS values. The CPU spike is usually caused by the 'tearing down' process. If you increase the call duration, you will be able to see CPU spikes when channels times out and starting to tear down. Henry On Sat, Jul 7, 2012 at 12:46 AM, Hanie Maghsoudy wrote: > Hi all, > > I searched for FreeSwitch call capacity, but most of the results wasn't > new. So, I wanna ask if anybody has either tested FreeSwitch's performance > recently, or got a dramatic result in real environment? > > I tested call quality on this machine: > > Virtual FreeSwitch server > OS: CentOS release 6.2 - x86_64 > CPU: 8 processor - Intel(R) Xeon(R) CPU X5670 @ 2.93GHz > Memory: 8 G > > After receiving incoming calls, FreeSwitch routed them to another sip > server, without transcoding. The other server transmitted calls by playing > an audio file. Meanwhile, I called an extension in FreeSwitch to test the > call quality. > > The result was like this: > > 1000 Concurrent calls > Call duration: 160s > Call rate: 6 cps (just creating channels) > Max used Memory: 1416M > Max CPU load: 0.24 > Max Network throughput (recv/send): 6711k/80k > Quality: Good > > This test was taken before tearing down the channels. > > Then, I took another test to estimate calls per second, and it wasn't what > I was expected! > > > 150 Concurrent calls > Call duration: 4s > Call rate: 30 cps (creating and tearing down) > Max used Memory: 1293M > Max CPU load: 4.50 > Max Network throughput (recv/send): 828k/60k > Quality: Average > > And when I increase call rate to 50 cps: > > 1000 Concurrent calls > Call duration: 4s > Call rate: 50 cps (creating and tearing down) > Max used Memory: 1730M > Max CPU load: *29.9* > Max Network throughput (recv/send): 1367k/202k > Quality: Bad > > Why call per second is such a big problem? Did anyone get a better result > on this? > > Thanks, > Hanie > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120707/cd0294a8/attachment.html From toddb at toddbailey.net Sat Jul 7 22:19:24 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Sat, 07 Jul 2012 11:19:24 -0700 Subject: [Freeswitch-users] Routing inbound call to multiple extentions Message-ID: <1341685164.3454.51.camel@mythtv> Hi All, At witt's end on this, how do I get a inbound call to go to multiple extension? I've tried groups but so far no avail I using a spa 3102 as a analog adapter that is configured to rout calls to extension 1000, I want FS to automatically forward this call to extensions 1001-1009. Here is an extract from default.xml (/usr/local/freeswitch/conf/directory) From avi at avimarcus.net Sat Jul 7 22:54:25 2012 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 7 Jul 2012 21:54:25 +0300 Subject: [Freeswitch-users] Performance Testing In-Reply-To: References: Message-ID: What CDR mod are you using? Are you writing large CDRs, e.g. xml_cdr to disk? Perhaps that's messing with the load average. And regarding ACD... for example, my calling card "good" traffic has an ACD of about or over ~3 minutes. See if 50cps but 120-180 second ACD still has that issue... after you check the CDRs. -Avi On Sat, Jul 7, 2012 at 8:55 PM, Henry Huang wrote: > From my experience playing with different CPS values. The CPU spike is > usually caused by the 'tearing down' process. If you increase the call > duration, you will be able to see CPU spikes when channels times out and > starting to tear down. > > Henry > > On Sat, Jul 7, 2012 at 12:46 AM, Hanie Maghsoudy wrote: > >> Hi all, >> >> I searched for FreeSwitch call capacity, but most of the results wasn't >> new. So, I wanna ask if anybody has either tested FreeSwitch's performance >> recently, or got a dramatic result in real environment? >> >> I tested call quality on this machine: >> >> Virtual FreeSwitch server >> OS: CentOS release 6.2 - x86_64 >> CPU: 8 processor - Intel(R) Xeon(R) CPU X5670 @ 2.93GHz >> Memory: 8 G >> >> After receiving incoming calls, FreeSwitch routed them to another sip >> server, without transcoding. The other server transmitted calls by playing >> an audio file. Meanwhile, I called an extension in FreeSwitch to test the >> call quality. >> >> The result was like this: >> >> 1000 Concurrent calls >> Call duration: 160s >> Call rate: 6 cps (just creating channels) >> Max used Memory: 1416M >> Max CPU load: 0.24 >> Max Network throughput (recv/send): 6711k/80k >> Quality: Good >> >> This test was taken before tearing down the channels. >> >> Then, I took another test to estimate calls per second, and it wasn't >> what I was expected! >> >> >> 150 Concurrent calls >> Call duration: 4s >> Call rate: 30 cps (creating and tearing down) >> Max used Memory: 1293M >> Max CPU load: 4.50 >> Max Network throughput (recv/send): 828k/60k >> Quality: Average >> >> And when I increase call rate to 50 cps: >> >> 1000 Concurrent calls >> Call duration: 4s >> Call rate: 50 cps (creating and tearing down) >> Max used Memory: 1730M >> Max CPU load: *29.9* >> Max Network throughput (recv/send): 1367k/202k >> Quality: Bad >> >> Why call per second is such a big problem? Did anyone get a better result >> on this? >> >> Thanks, >> Hanie >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120707/91f05438/attachment.html From mgg at giagnocavo.net Sat Jul 7 22:54:58 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Sat, 7 Jul 2012 18:54:58 +0000 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: <6D07CDCCEAC84C73841519D1FB19955E@gmail.com> References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <537ACDFCE00243439E851C86C8C25E7A@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B604510494@BLUPRD0711MB413.namprd07.prod.outlook.com> <6D07CDCCEAC84C73841519D1FB19955E@gmail.com> Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B60451076D@BLUPRD0711MB413.namprd07.prod.outlook.com> What do you mean by hold? From the 0mq site: How do I use the ZMQ_SWAP socket feature to persist my data to disk? It is not the responsibility of the library to handle data persistence. If you need to make sure no messages are ever lost, you need to write application-level code to persist your data to non-volatile memory and handle the recovery modes of your application. Also, ZMQ_SWAP is deprecated and will be removed in the 3.x series of releases since it confuses many people over proper usage. So, what am I missing? Everything I read about 0MQ make it clear that persistence and durability are something the user needs to handle. Contrast to, say, RabbitMQ or other message brokers which exist to handle the persistence for you. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gavin Henry Sent: Saturday, July 07, 2012 8:07 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Pushing CDR Information into Zeromq 0mq does hold the messages. -- Gavin Henry Sent with Sparrow On Saturday, 7 July 2012 at 11:56, Michael Giagnocavo wrote: But 0MQ is just the transport. You still need a broker to persist messages, participate in transactions, and so on. That?s what I was referring to, not just the transport layer acknowledgements. So if you?re just taking CDRs from FS memory and sending them out over 0MQ, what if the other side is down? In that case, aren?t you risking a lot of data, should something happen before the receiver gets up and saves all the CDRs? 0MQ isn?t really an alternative to a broker like RabbitMQ or SSSB. CDRs should be going to a local, durable, message queue. This doesn?t mean that you need to hit disk; the messages can stay in RAM for a few seconds, under the assumption that during normal ops, they?ll get delivered immediately and there?s no need for local persistence. But when something breaks, then the CDRs can just pile up on disk, and get pumped out whenever things stabilize. Am I missing something? -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita Sent: Thursday, July 05, 2012 3:09 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Pushing CDR Information into Zeromq Michael, You are partially correct because it depends on the socket type you use on ZeroMQ. Since the socket implemented on mod_zmq is a simple PUB/SUB, yes, messages would be discarded if they are not received by the other end. If we would use a XREQ/XREP then it would be queued until you are able to receive them. -- Jo?o Mesquita Sent with Sparrow On Thursday, July 5, 2012 at 3:08 PM, Michael Giagnocavo wrote: Wouldn?t the problem with ZeroMQ be no reliable delivery of CDRs? I was under the impression that 0MQ was more for high-performance, and things like guaranteed delivery are left as an exercise to the user. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita Sent: Thursday, July 05, 2012 9:56 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Pushing CDR Information into Zeromq I haven't actually looked at the code, but you have to look at the variables on CHANNEL_REPORTING for example. There you will see all the vars needed to produce CDR entries, but they won't be on XML format like xml_curl would create. Regards, -- Jo?o Mesquita Sent with Sparrow On Thursday, July 5, 2012 at 10:37 AM, Neo Cheema wrote: Hi all, I was hoping to find a way to push CDR information into a queue. Zeromq comes out as an obvious choice because mod_event_zmq already exits. However, I can't find a way to configure this module to push CDR info. Have any of you guys tried it? Regards _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120707/8b12d090/attachment-0001.html From h.maghsoudy at gmail.com Sat Jul 7 23:03:05 2012 From: h.maghsoudy at gmail.com (Hanie Maghsoudy) Date: Sat, 7 Jul 2012 23:33:05 +0430 Subject: [Freeswitch-users] Performance Testing In-Reply-To: References: Message-ID: Yeah, I believe so, since when calls are establishing the cpu load is not a problem. On Sat, Jul 7, 2012 at 10:25 PM, Henry Huang wrote: > From my experience playing with different CPS values. The CPU spike is > usually caused by the 'tearing down' process. If you increase the call > duration, you will be able to see CPU spikes when channels times out and > starting to tear down. > > Henry > > On Sat, Jul 7, 2012 at 12:46 AM, Hanie Maghsoudy wrote: > >> Hi all, >> >> I searched for FreeSwitch call capacity, but most of the results wasn't >> new. So, I wanna ask if anybody has either tested FreeSwitch's performance >> recently, or got a dramatic result in real environment? >> >> I tested call quality on this machine: >> >> Virtual FreeSwitch server >> OS: CentOS release 6.2 - x86_64 >> CPU: 8 processor - Intel(R) Xeon(R) CPU X5670 @ 2.93GHz >> Memory: 8 G >> >> After receiving incoming calls, FreeSwitch routed them to another sip >> server, without transcoding. The other server transmitted calls by playing >> an audio file. Meanwhile, I called an extension in FreeSwitch to test the >> call quality. >> >> The result was like this: >> >> 1000 Concurrent calls >> Call duration: 160s >> Call rate: 6 cps (just creating channels) >> Max used Memory: 1416M >> Max CPU load: 0.24 >> Max Network throughput (recv/send): 6711k/80k >> Quality: Good >> >> This test was taken before tearing down the channels. >> >> Then, I took another test to estimate calls per second, and it wasn't >> what I was expected! >> >> >> 150 Concurrent calls >> Call duration: 4s >> Call rate: 30 cps (creating and tearing down) >> Max used Memory: 1293M >> Max CPU load: 4.50 >> Max Network throughput (recv/send): 828k/60k >> Quality: Average >> >> And when I increase call rate to 50 cps: >> >> 1000 Concurrent calls >> Call duration: 4s >> Call rate: 50 cps (creating and tearing down) >> Max used Memory: 1730M >> Max CPU load: *29.9* >> Max Network throughput (recv/send): 1367k/202k >> Quality: Bad >> >> Why call per second is such a big problem? Did anyone get a better result >> on this? >> >> Thanks, >> Hanie >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120707/64d59a88/attachment.html From avi at avimarcus.net Sat Jul 7 23:11:26 2012 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 7 Jul 2012 22:11:26 +0300 Subject: [Freeswitch-users] Performance Testing In-Reply-To: References: Message-ID: So have you tried writing with the smaller mod_json or even smaller mod_cdr_csv with your own small template? Or having mod_xml_cdr / mod_json_cdr POST the CDRs to another machine to process? -Avi On Sat, Jul 7, 2012 at 10:03 PM, Hanie Maghsoudy wrote: > Yeah, I believe so, since when calls are establishing the cpu load is not > a problem. > > > On Sat, Jul 7, 2012 at 10:25 PM, Henry Huang wrote: > >> From my experience playing with different CPS values. The CPU spike is >> usually caused by the 'tearing down' process. If you increase the call >> duration, you will be able to see CPU spikes when channels times out and >> starting to tear down. >> >> Henry >> >> On Sat, Jul 7, 2012 at 12:46 AM, Hanie Maghsoudy wrote: >> >>> Hi all, >>> >>> I searched for FreeSwitch call capacity, but most of the results wasn't >>> new. So, I wanna ask if anybody has either tested FreeSwitch's performance >>> recently, or got a dramatic result in real environment? >>> >>> I tested call quality on this machine: >>> >>> Virtual FreeSwitch server >>> OS: CentOS release 6.2 - x86_64 >>> CPU: 8 processor - Intel(R) Xeon(R) CPU X5670 @ 2.93GHz >>> Memory: 8 G >>> >>> After receiving incoming calls, FreeSwitch routed them to another sip >>> server, without transcoding. The other server transmitted calls by playing >>> an audio file. Meanwhile, I called an extension in FreeSwitch to test the >>> call quality. >>> >>> The result was like this: >>> >>> 1000 Concurrent calls >>> Call duration: 160s >>> Call rate: 6 cps (just creating channels) >>> Max used Memory: 1416M >>> Max CPU load: 0.24 >>> Max Network throughput (recv/send): 6711k/80k >>> Quality: Good >>> >>> This test was taken before tearing down the channels. >>> >>> Then, I took another test to estimate calls per second, and it wasn't >>> what I was expected! >>> >>> >>> 150 Concurrent calls >>> Call duration: 4s >>> Call rate: 30 cps (creating and tearing down) >>> Max used Memory: 1293M >>> Max CPU load: 4.50 >>> Max Network throughput (recv/send): 828k/60k >>> Quality: Average >>> >>> And when I increase call rate to 50 cps: >>> >>> 1000 Concurrent calls >>> Call duration: 4s >>> Call rate: 50 cps (creating and tearing down) >>> Max used Memory: 1730M >>> Max CPU load: *29.9* >>> Max Network throughput (recv/send): 1367k/202k >>> Quality: Bad >>> >>> Why call per second is such a big problem? Did anyone get a better >>> result on this? >>> >>> Thanks, >>> Hanie >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120707/ab466094/attachment.html From red.rain.seven at gmail.com Sun Jul 8 00:39:50 2012 From: red.rain.seven at gmail.com (Henry Huang) Date: Sat, 7 Jul 2012 13:39:50 -0700 Subject: [Freeswitch-users] Performance Testing In-Reply-To: References: Message-ID: Avi So are you saying that using mod_json for mod_cdr_csv can reduce the cpu load comparing to using xml_cdr? I knew it has to do with writing CDR when tearing down the channels and I thought it was inevitable. Thanks, Henry On Sat, Jul 7, 2012 at 12:11 PM, Avi Marcus wrote: > So have you tried writing with the smaller mod_json or even smaller > mod_cdr_csv with your own small template? > Or having mod_xml_cdr / mod_json_cdr POST the CDRs to another machine to > process? > > -Avi > > > On Sat, Jul 7, 2012 at 10:03 PM, Hanie Maghsoudy wrote: > >> Yeah, I believe so, since when calls are establishing the cpu load is not >> a problem. >> >> >> On Sat, Jul 7, 2012 at 10:25 PM, Henry Huang wrote: >> >>> From my experience playing with different CPS values. The CPU spike is >>> usually caused by the 'tearing down' process. If you increase the call >>> duration, you will be able to see CPU spikes when channels times out and >>> starting to tear down. >>> >>> Henry >>> >>> On Sat, Jul 7, 2012 at 12:46 AM, Hanie Maghsoudy wrote: >>> >>>> Hi all, >>>> >>>> I searched for FreeSwitch call capacity, but most of the results wasn't >>>> new. So, I wanna ask if anybody has either tested FreeSwitch's performance >>>> recently, or got a dramatic result in real environment? >>>> >>>> I tested call quality on this machine: >>>> >>>> Virtual FreeSwitch server >>>> OS: CentOS release 6.2 - x86_64 >>>> CPU: 8 processor - Intel(R) Xeon(R) CPU X5670 @ 2.93GHz >>>> Memory: 8 G >>>> >>>> After receiving incoming calls, FreeSwitch routed them to another sip >>>> server, without transcoding. The other server transmitted calls by playing >>>> an audio file. Meanwhile, I called an extension in FreeSwitch to test the >>>> call quality. >>>> >>>> The result was like this: >>>> >>>> 1000 Concurrent calls >>>> Call duration: 160s >>>> Call rate: 6 cps (just creating channels) >>>> Max used Memory: 1416M >>>> Max CPU load: 0.24 >>>> Max Network throughput (recv/send): 6711k/80k >>>> Quality: Good >>>> >>>> This test was taken before tearing down the channels. >>>> >>>> Then, I took another test to estimate calls per second, and it wasn't >>>> what I was expected! >>>> >>>> >>>> 150 Concurrent calls >>>> Call duration: 4s >>>> Call rate: 30 cps (creating and tearing down) >>>> Max used Memory: 1293M >>>> Max CPU load: 4.50 >>>> Max Network throughput (recv/send): 828k/60k >>>> Quality: Average >>>> >>>> And when I increase call rate to 50 cps: >>>> >>>> 1000 Concurrent calls >>>> Call duration: 4s >>>> Call rate: 50 cps (creating and tearing down) >>>> Max used Memory: 1730M >>>> Max CPU load: *29.9* >>>> Max Network throughput (recv/send): 1367k/202k >>>> Quality: Bad >>>> >>>> Why call per second is such a big problem? Did anyone get a better >>>> result on this? >>>> >>>> Thanks, >>>> Hanie >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120707/e9278e46/attachment-0001.html From avi at avimarcus.net Sun Jul 8 01:06:55 2012 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 8 Jul 2012 00:06:55 +0300 Subject: [Freeswitch-users] Performance Testing In-Reply-To: References: Message-ID: I just know the actual data size of mod_json_cdr is smaller than mod_xml_cdr... are you sure it's the CDR generation time and not the cpu usage of writing it to disk? Try a) mod_cdr_csv instead, b) mod_json_cdr instead, c) turning off all logging and/or d) writing the CDRs to a ramdisk to see the new usage. AND.. as said before.. calls aren't usually 4 seconds. "good" traffic is on average 180+ seconds. -Avi On Sat, Jul 7, 2012 at 11:39 PM, Henry Huang wrote: > Avi > > So are you saying that using mod_json for mod_cdr_csv can reduce the cpu > load comparing to using xml_cdr? I knew it has to do with writing CDR when > tearing down the channels and I thought it was inevitable. > > Thanks, > > Henry > > On Sat, Jul 7, 2012 at 12:11 PM, Avi Marcus wrote: > >> So have you tried writing with the smaller mod_json or even smaller >> mod_cdr_csv with your own small template? >> Or having mod_xml_cdr / mod_json_cdr POST the CDRs to another machine to >> process? >> >> -Avi >> >> >> On Sat, Jul 7, 2012 at 10:03 PM, Hanie Maghsoudy wrote: >> >>> Yeah, I believe so, since when calls are establishing the cpu load is >>> not a problem. >>> >>> >>> On Sat, Jul 7, 2012 at 10:25 PM, Henry Huang wrote: >>> >>>> From my experience playing with different CPS values. The CPU spike is >>>> usually caused by the 'tearing down' process. If you increase the call >>>> duration, you will be able to see CPU spikes when channels times out and >>>> starting to tear down. >>>> >>>> Henry >>>> >>>> On Sat, Jul 7, 2012 at 12:46 AM, Hanie Maghsoudy >>> > wrote: >>>> >>>>> Hi all, >>>>> >>>>> I searched for FreeSwitch call capacity, but most of the results >>>>> wasn't new. So, I wanna ask if anybody has either tested FreeSwitch's >>>>> performance recently, or got a dramatic result in real environment? >>>>> >>>>> I tested call quality on this machine: >>>>> >>>>> Virtual FreeSwitch server >>>>> OS: CentOS release 6.2 - x86_64 >>>>> CPU: 8 processor - Intel(R) Xeon(R) CPU X5670 @ 2.93GHz >>>>> Memory: 8 G >>>>> >>>>> After receiving incoming calls, FreeSwitch routed them to another sip >>>>> server, without transcoding. The other server transmitted calls by playing >>>>> an audio file. Meanwhile, I called an extension in FreeSwitch to test the >>>>> call quality. >>>>> >>>>> The result was like this: >>>>> >>>>> 1000 Concurrent calls >>>>> Call duration: 160s >>>>> Call rate: 6 cps (just creating channels) >>>>> Max used Memory: 1416M >>>>> Max CPU load: 0.24 >>>>> Max Network throughput (recv/send): 6711k/80k >>>>> Quality: Good >>>>> >>>>> This test was taken before tearing down the channels. >>>>> >>>>> Then, I took another test to estimate calls per second, and it wasn't >>>>> what I was expected! >>>>> >>>>> >>>>> 150 Concurrent calls >>>>> Call duration: 4s >>>>> Call rate: 30 cps (creating and tearing down) >>>>> Max used Memory: 1293M >>>>> Max CPU load: 4.50 >>>>> Max Network throughput (recv/send): 828k/60k >>>>> Quality: Average >>>>> >>>>> And when I increase call rate to 50 cps: >>>>> >>>>> 1000 Concurrent calls >>>>> Call duration: 4s >>>>> Call rate: 50 cps (creating and tearing down) >>>>> Max used Memory: 1730M >>>>> Max CPU load: *29.9* >>>>> Max Network throughput (recv/send): 1367k/202k >>>>> Quality: Bad >>>>> >>>>> Why call per second is such a big problem? Did anyone get a better >>>>> result on this? >>>>> >>>>> Thanks, >>>>> Hanie >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> Join Us At ClueCon - Aug 7-9, 2012 >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120708/f3f4215a/attachment.html From jmesquita at freeswitch.org Sun Jul 8 06:24:38 2012 From: jmesquita at freeswitch.org (Jmesquita@freeswitch.org) Date: Sat, 7 Jul 2012 23:24:38 -0300 Subject: [Freeswitch-users] Pushing CDR Information into Zeromq In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B60451076D@BLUPRD0711MB413.namprd07.prod.outlook.com> References: <3D0606A352164F4FAFE2CE9F14BD8512@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B603B6145B@CH1PRD0710MB393.namprd07.prod.outlook.com> <537ACDFCE00243439E851C86C8C25E7A@freeswitch.org> <63B00DD1DA6A364E9F64A3A0BD2FE7B604510494@BLUPRD0711MB413.namprd07.prod.outlook.com> <6D07CDCCEAC84C73841519D1FB19955E@gmail.com> <63B00DD1DA6A364E9F64A3A0BD2FE7B60451076D@BLUPRD0711MB413.namprd07.prod.outlook.com> Message-ID: <5F559242-D763-4C27-AC15-B089B2C2E1E7@freeswitch.org> Michael, As far as I know, you are correct. Zeromq is a broker less design in which you are the one who should write your own broker. It just enables you to put the pieces together. There are lot of protocols that teach you how to do data persistence properly tho on the zeromq guide. Written by Pieter, one of the project leaders. Once again, I really don't think that zeromq should be plugged onto the fs core because of its asserts that are still yet to be resolved but when they do, I will be the first one to propose some fancy stuff we can get done with it. Nowadays, I use zeromq worker that consumes ESL events. It works great and enables me to have redundancy when consuming those events by syncing state on the worker cloud. Jo?o Mesquita On 07/07/2012, at 03:54 p.m., Michael Giagnocavo wrote: > What do you mean by hold? From the 0mq site: > > How do I use the ZMQ_SWAP socket feature to persist my data to disk? > It is not the responsibility of the library to handle data persistence. If you need to make sure no messages are ever lost, you need to write application-level code to persist your data to non-volatile memory and handle the recovery modes of your application. Also, ZMQ_SWAP is deprecated and will be removed in the 3.x series of releases since it confuses many people over proper usage. > > So, what am I missing? Everything I read about 0MQ make it clear that persistence and durability are something the user needs to handle. Contrast to, say, RabbitMQ or other message brokers which exist to handle the persistence for you. > > -Michael > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Gavin Henry > Sent: Saturday, July 07, 2012 8:07 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Pushing CDR Information into Zeromq > > 0mq does hold the messages. > > -- > Gavin Henry > Sent with Sparrow > > On Saturday, 7 July 2012 at 11:56, Michael Giagnocavo wrote: > > But 0MQ is just the transport. You still need a broker to persist messages, participate in transactions, and so on. That?s what I was referring to, not just the transport layer acknowledgements. So if you?re just taking CDRs from FS memory and sending them out over 0MQ, what if the other side is down? In that case, aren?t you risking a lot of data, should something happen before the receiver gets up and saves all the CDRs? 0MQ isn?t really an alternative to a broker like RabbitMQ or SSSB. > > CDRs should be going to a local, durable, message queue. This doesn?t mean that you need to hit disk; the messages can stay in RAM for a few seconds, under the assumption that during normal ops, they?ll get delivered immediately and there?s no need for local persistence. But when something breaks, then the CDRs can just pile up on disk, and get pumped out whenever things stabilize. > > Am I missing something? > > -Michael > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita > Sent: Thursday, July 05, 2012 3:09 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Pushing CDR Information into Zeromq > > Michael, > > You are partially correct because it depends on the socket type you use on ZeroMQ. Since the socket implemented on mod_zmq is a simple PUB/SUB, yes, messages would be discarded if they are not received by the other end. If we would use a XREQ/XREP then it would be queued until you are able to receive them. > > -- > Jo?o Mesquita > Sent with Sparrow > > On Thursday, July 5, 2012 at 3:08 PM, Michael Giagnocavo wrote: > > Wouldn?t the problem with ZeroMQ be no reliable delivery of CDRs? I was under the impression that 0MQ was more for high-performance, and things like guaranteed delivery are left as an exercise to the user. > -Michael > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jo?o Mesquita > Sent: Thursday, July 05, 2012 9:56 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Pushing CDR Information into Zeromq > > I haven't actually looked at the code, but you have to look at the variables on CHANNEL_REPORTING for example. There you will see all the vars needed to produce CDR entries, but they won't be on XML format like xml_curl would create. > > Regards, > > -- > Jo?o Mesquita > Sent with Sparrow > > On Thursday, July 5, 2012 at 10:37 AM, Neo Cheema wrote: > > Hi all, > > I was hoping to find a way to push CDR information into a queue. > Zeromq comes out as an obvious choice because mod_event_zmq already > exits. However, I can't find a way to configure this module to push > CDR info. Have any of you guys tried it? > > Regards > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120707/bdaf7995/attachment-0001.html From h.maghsoudy at gmail.com Sun Jul 8 07:48:00 2012 From: h.maghsoudy at gmail.com (Hanie Maghsoudy) Date: Sun, 8 Jul 2012 08:18:00 +0430 Subject: [Freeswitch-users] Performance Testing In-Reply-To: References: Message-ID: Hi. Yes, I'm using mod_xml_cdr. So, I'll try real hardware server, older version of CentOS, and modify CDR modes to test again with more realistic ACD. Thank you all. On Sun, Jul 8, 2012 at 1:36 AM, Avi Marcus wrote: > I just know the actual data size of mod_json_cdr is smaller than > mod_xml_cdr... are you sure it's the CDR generation time and not the cpu > usage of writing it to disk? > Try a) mod_cdr_csv instead, b) mod_json_cdr instead, c) turning off all > logging and/or d) writing the CDRs to a ramdisk to see the new usage. > > AND.. as said before.. calls aren't usually 4 seconds. "good" traffic is > on average 180+ seconds. > > -Avi > > > On Sat, Jul 7, 2012 at 11:39 PM, Henry Huang wrote: > >> Avi >> >> So are you saying that using mod_json for mod_cdr_csv can reduce the cpu >> load comparing to using xml_cdr? I knew it has to do with writing CDR when >> tearing down the channels and I thought it was inevitable. >> >> Thanks, >> >> Henry >> >> On Sat, Jul 7, 2012 at 12:11 PM, Avi Marcus wrote: >> >>> So have you tried writing with the smaller mod_json or even smaller >>> mod_cdr_csv with your own small template? >>> Or having mod_xml_cdr / mod_json_cdr POST the CDRs to another machine to >>> process? >>> >>> -Avi >>> >>> >>> On Sat, Jul 7, 2012 at 10:03 PM, Hanie Maghsoudy wrote: >>> >>>> Yeah, I believe so, since when calls are establishing the cpu load is >>>> not a problem. >>>> >>>> >>>> On Sat, Jul 7, 2012 at 10:25 PM, Henry Huang wrote: >>>> >>>>> From my experience playing with different CPS values. The CPU spike is >>>>> usually caused by the 'tearing down' process. If you increase the call >>>>> duration, you will be able to see CPU spikes when channels times out and >>>>> starting to tear down. >>>>> >>>>> Henry >>>>> >>>>> On Sat, Jul 7, 2012 at 12:46 AM, Hanie Maghsoudy < >>>>> h.maghsoudy at gmail.com> wrote: >>>>> >>>>>> Hi all, >>>>>> >>>>>> I searched for FreeSwitch call capacity, but most of the results >>>>>> wasn't new. So, I wanna ask if anybody has either tested FreeSwitch's >>>>>> performance recently, or got a dramatic result in real environment? >>>>>> >>>>>> I tested call quality on this machine: >>>>>> >>>>>> Virtual FreeSwitch server >>>>>> OS: CentOS release 6.2 - x86_64 >>>>>> CPU: 8 processor - Intel(R) Xeon(R) CPU X5670 @ 2.93GHz >>>>>> Memory: 8 G >>>>>> >>>>>> After receiving incoming calls, FreeSwitch routed them to another sip >>>>>> server, without transcoding. The other server transmitted calls by playing >>>>>> an audio file. Meanwhile, I called an extension in FreeSwitch to test the >>>>>> call quality. >>>>>> >>>>>> The result was like this: >>>>>> >>>>>> 1000 Concurrent calls >>>>>> Call duration: 160s >>>>>> Call rate: 6 cps (just creating channels) >>>>>> Max used Memory: 1416M >>>>>> Max CPU load: 0.24 >>>>>> Max Network throughput (recv/send): 6711k/80k >>>>>> Quality: Good >>>>>> >>>>>> This test was taken before tearing down the channels. >>>>>> >>>>>> Then, I took another test to estimate calls per second, and it wasn't >>>>>> what I was expected! >>>>>> >>>>>> >>>>>> 150 Concurrent calls >>>>>> Call duration: 4s >>>>>> Call rate: 30 cps (creating and tearing down) >>>>>> Max used Memory: 1293M >>>>>> Max CPU load: 4.50 >>>>>> Max Network throughput (recv/send): 828k/60k >>>>>> Quality: Average >>>>>> >>>>>> And when I increase call rate to 50 cps: >>>>>> >>>>>> 1000 Concurrent calls >>>>>> Call duration: 4s >>>>>> Call rate: 50 cps (creating and tearing down) >>>>>> Max used Memory: 1730M >>>>>> Max CPU load: *29.9* >>>>>> Max Network throughput (recv/send): 1367k/202k >>>>>> Quality: Bad >>>>>> >>>>>> Why call per second is such a big problem? Did anyone get a better >>>>>> result on this? >>>>>> >>>>>> Thanks, >>>>>> Hanie >>>>>> >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> Join Us At ClueCon - Aug 7-9, 2012 >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE: >>>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>>> >>>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> Join Us At ClueCon - Aug 7-9, 2012 >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120708/c42d9893/attachment.html From steveu at coppice.org Sun Jul 8 08:16:51 2012 From: steveu at coppice.org (Steve Underwood) Date: Sun, 08 Jul 2012 12:16:51 +0800 Subject: [Freeswitch-users] Performance Testing In-Reply-To: References: Message-ID: <4FF909B3.1040107@coppice.org> On 07/08/2012 11:48 AM, Hanie Maghsoudy wrote: > Hi. Yes, I'm using mod_xml_cdr. So, I'll try real hardware server, > older version of CentOS, and modify CDR modes to test again with more > realistic ACD. Thank you all. > > On Sun, Jul 8, 2012 at 1:36 AM, Avi Marcus > wrote: > > I just know the actual data size of mod_json_cdr is smaller than > mod_xml_cdr... are you sure it's the CDR generation time and not > the cpu usage of writing it to disk? > Try a) mod_cdr_csv instead, b) mod_json_cdr instead, c) turning > off all logging and/or d) writing the CDRs to a ramdisk to see the > new usage. > > AND.. as said before.. calls aren't usually 4 seconds. "good" > traffic is on average 180+ seconds. > Good is highly variable. There used to be lots of good traffic that averaged 4s in the days when paging was big business. There is still plenty of call centre work where the average is 30s. Even proper conversations can average considerably less than 180s. You REALLY have to look at the application. Steve From alex at thewinelake.com Sun Jul 8 11:24:12 2012 From: alex at thewinelake.com (Alexander Lake) Date: Sun, 8 Jul 2012 08:24:12 +0100 Subject: [Freeswitch-users] Performance Testing In-Reply-To: <4FF909B3.1040107@coppice.org> References: <4FF909B3.1040107@coppice.org> Message-ID: <81255FC8-7F2D-409D-B3BA-F358E3EC5830@thewinelake.com> Our average call is 60s. Quite a lot of short ones due to people hitting voicemail and hanging up immediately. On 8 Jul 2012, at 05:16, Steve Underwood wrote: > On 07/08/2012 11:48 AM, Hanie Maghsoudy wrote: >> Hi. Yes, I'm using mod_xml_cdr. So, I'll try real hardware server, >> older version of CentOS, and modify CDR modes to test again with more >> realistic ACD. Thank you all. >> >> On Sun, Jul 8, 2012 at 1:36 AM, Avi Marcus > > wrote: >> >> I just know the actual data size of mod_json_cdr is smaller than >> mod_xml_cdr... are you sure it's the CDR generation time and not >> the cpu usage of writing it to disk? >> Try a) mod_cdr_csv instead, b) mod_json_cdr instead, c) turning >> off all logging and/or d) writing the CDRs to a ramdisk to see the >> new usage. >> >> AND.. as said before.. calls aren't usually 4 seconds. "good" >> traffic is on average 180+ seconds. >> > Good is highly variable. There used to be lots of good traffic that > averaged 4s in the days when paging was big business. There is still > plenty of call centre work where the average is 30s. Even proper > conversations can average considerably less than 180s. You REALLY have > to look at the application. > > Steve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From ssinyagin at yahoo.com Sun Jul 8 15:54:06 2012 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Sun, 8 Jul 2012 04:54:06 -0700 (PDT) Subject: [Freeswitch-users] Routing inbound call to multiple extentions In-Reply-To: <1341685164.3454.51.camel@mythtv> References: <1341685164.3454.51.camel@mythtv> Message-ID: <1341748446.53554.YahooMailNeo@web39304.mail.mud.yahoo.com> you forgot to specify how exactly you route the call to this group. >________________________________ > From: Todd Bailey >To: freeswitch >Sent: Saturday, July 7, 2012 8:19 PM >Subject: [Freeswitch-users] Routing inbound call to multiple extentions > >Hi All, > > >At witt's end on this,? how do I get a inbound call to go to multiple >extension? > >I've tried groups but so far no avail > >I using a spa 3102 as a analog adapter that is configured to rout calls >to extension 1000,? I want FS to automatically forward this call to >extensions 1001-1009. > >Here is an extract from default.xml >(/usr/local/freeswitch/conf/directory) > >? >??? >??? ? >??? ? >??? ? >??? ? >??? ? >??? ? >??? ? >??? ? >??? ? >??? ? >??? >? ? ? > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >Join Us At ClueCon - Aug 7-9, 2012 > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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/20120708/15d66b32/attachment.html From jack.nikolas at ymail.com Sun Jul 8 19:23:35 2012 From: jack.nikolas at ymail.com (Jack Nikolas) Date: Sun, 8 Jul 2012 16:23:35 +0100 (BST) Subject: [Freeswitch-users] how to make web call on mod_curl using proxy ? Message-ID: <1341761015.26568.YahooMailNeo@web171503.mail.ir2.yahoo.com> hi All, i am going to making web call using mod_curl in freeswitch, it is noticable that on my freeswitch for having? internet access ,user have to? set proxy, for example set proxy on the address like this: http_proxy=http://10.10.10.1:2343 now i want to running web call as the following way,but how can i set proxy on lua? web_url = "http://tycho.usno.navy.mil/cgi-bin/timer.pl" num_reads = 0 api = freeswitch.API() session:answer() while(session:ready() == true and num_reads < 10) do freeswitch.consoleLog("INFO","URL: " .. web_url .. "\n") raw_data = api:execute("curl", web_url) freeswitch.consoleLog("INFO","Raw data:\n" .. raw_data .. "\n\n") date_time = string.match(raw_data,"
.-UTC",1) if (date_time == nil) then freeswitch.consoleLog("INFO","UTC date and time not found\n") else freeswitch.consoleLog("INFO","UTC date and time is '" .. date_time .. "'\n") time = string.gsub(date_time,".-(%d+:%d+:%d+).+","%1") freeswitch.consoleLog("INFO","Time is '" .. time .. "'\n\n") session:streamFile("phrase:simple_time:" .. time) end num_reads = num_reads + 1 session:execute("sleep","1000") end session:hangup() any help is welcome, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120708/399030d5/attachment.html From sdevoy at bizfocused.com Sun Jul 8 20:06:57 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sun, 8 Jul 2012 12:06:57 -0400 Subject: [Freeswitch-users] Routing inbound call to multiple extentions In-Reply-To: <1341685164.3454.51.camel@mythtv> References: <1341685164.3454.51.camel@mythtv> Message-ID: <055b01cd5d23$b371a360$1a54ea20$@bizfocused.com> HI Todd, Based on this sample: http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#Groups and your directory data it looks like you need a dialplan something like this: Maybe IVR goes here??? What is your dial plan entry? Sean -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Todd Bailey Sent: Saturday, July 07, 2012 2:19 PM To: freeswitch Subject: [Freeswitch-users] Routing inbound call to multiple extentions Hi All, At witt's end on this, how do I get a inbound call to go to multiple extension? I've tried groups but so far no avail I using a spa 3102 as a analog adapter that is configured to rout calls to extension 1000, I want FS to automatically forward this call to extensions 1001-1009. Here is an extract from default.xml (/usr/local/freeswitch/conf/directory) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.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 Sun Jul 8 23:34:00 2012 From: brian at freeswitch.org (Brian West) Date: Sun, 8 Jul 2012 14:34:00 -0500 Subject: [Freeswitch-users] FreeSWITCH TLS with StartSSL Certificate In-Reply-To: <1338801843810-7579377.post@n2.nabble.com> References: <1338801843810-7579377.post@n2.nabble.com> Message-ID: You should post this on Jira so we can document and integrate this properly. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.918.420.9266 | F: +1.918.420.9267 | M: +1.918.424.WEST iNUM: +883 5100 1420 9266 UK: +44 20 3298 4900 On Jun 4, 2012, at 4:24 AM, sunsus wrote: > Hello > > To day I tried to add a Free StartSSL Class 1 Certificate to a FreeSWITCH > installation. Here I will share the script on how to generate the > Certificate Request: > > > #!/bin/sh > > CONFDIR=/usr/local/freeswitch/conf/ssl > DAYS=2190 > KEY_SIZE=2048 > > TMPFILE="/tmp/fs-ca-$$-$(date +%Y%m%d%H%M%S)" > > COMMON_NAME="FrwwSWICH VOIP" > ALT_NAME="DNS:sip.freeswitch.org" > ORG_NAME="FreeSWICHT" > OUTFILE="agent.pem" > > umask 037 > > generate_request() { > local val="" > > echo "Generating new request..." > > echo > echo "--------------------------------------------------------" > echo "CN: \"${COMMON_NAME}\"" > echo "ORG_NAME: \"${ORG_NAME}\"" > echo "ALT_NAME: \"${ALT_NAME}\"" > echo > echo "Certificate filename \"${OUTFILE}\"" > echo > echo "[Is this OK? (y/N)]" > read val > if [ "${val}" != "y" ] && [ "${val}" != "Y" ]; then > echo "Aborted" > return 2 > fi > > sed \ > -e "s|%CN%|$COMMON_NAME|" \ > -e "s|%ALTNAME%|$ALT_NAME|" \ > -e "s|%ORG%|$ORG_NAME|" \ > "${CONFDIR}/CA/config.tpl" \ >> "${TMPFILE}.cfg" || exit 1 > > > echo ${KEY_SIZE} > openssl req -new -out "${TMPFILE}.req" \ > -newkey rsa:${KEY_SIZE} -keyout "${TMPFILE}.key" \ > -config "${TMPFILE}.cfg" -nodes -sha1 >/dev/null || exit 1 > > echo > cat ${TMPFILE}.req > echo > echo "go to http://www.startssl.com/ and generate a certificate" > echo "past certificate:" > while read LINE > do > echo $LINE >> ${TMPFILE}.crt > if [ "$LINE" = "^A" ];then > break > fi > done > echo "other processing continues " > > # openssl x509 -req -CAkey "${CONFDIR}/CA/cakey.pem" -CA > "${CONFDIR}/CA/cacert.pem" -CAcreateserial \ > # -in "${TMPFILE}.req" -out "${TMPFILE}.crt" -extfile "${TMPFILE}.cfg" \ > # -extensions "${EXTENSIONS}" -days ${DAYS} -sha1 >/dev/null || exit 1 > cat "${TMPFILE}.crt" "${TMPFILE}.key" > "${CONFDIR}/${OUTFILE}" > > wget http://www.startssl.com/certs/sub.class1.server.ca.pem > wget http://www.startssl.com/certs/ca.pem > cat sub.class1.server.ca.pem ca.pem >> ${CONFDIR}/cafile.pem > rm -f sub.class1.server.ca.pem ca.pem > rm "${TMPFILE}.cfg" "${TMPFILE}.crt" "${TMPFILE}.key" "${TMPFILE}.req" > > echo "DONE" > } > > > remove_startssl() { > echo "Removing StartSSL" > > if [ -d "${CONFDIR}/agent.pem" ]; then > rm "${CONFDIR}/agent.pem" > fi > > echo "DONE" > } > OUTFILESET="0" > command="$1" > shift > > while [ $# -gt 0 ]; do > case $1 in > -cn) > shift > COMMON_NAME="$1" > ;; > -alt) > shift > ALT_NAME="$1" > ;; > -org) > shift > ORG_NAME="$1" > ;; > -out) > shift > OUTFILE="$1" > OUTFILESET="1" > ;; > -days) > shift > DAYS="$1" > ;; > esac > shift > done > > > case ${command} in > create_request) > EXTENSIONS="request" > generate_request > ;; > > remove) > echo "Are you sure you want to delete the StartSSL Certificate? [YES to > delete]" > read val > if [ "${val}" = "YES" ]; then > remove_startssl > else > echo "Not deleting CA" > fi > ;; > > *) > cat <<-EOF > $0 [options] > > * commands: > > remove - Remove StartSSL > > create_request - Create a new certificate request for startSSL > > * options: > > -cn Set common name > -alt Set alternative name (use prefix 'DNS:' or 'URI:') > -org Set organization name > -out Filename for new certificate (create only) > -days Certificate expires in X days (default: 365) > > EOF > exit 1 > ;; > esac > > > Everything seams to work, expect the validation of a SNOM phone. Does any > one know how to tell FreeSWITCH to publish the correct ca bundel and > certificate track. Because the CA Certificate of Start SSL is included in > the SNOM: > > regards > > Patrick > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-TLS-with-StartSSL-Certificate-tp7579377.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120708/b1a29a56/attachment-0001.html From toddb at toddbailey.net Mon Jul 9 09:44:00 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Sun, 08 Jul 2012 22:44:00 -0700 Subject: [Freeswitch-users] Routing inbound call to multiple extentions In-Reply-To: <1341748446.53554.YahooMailNeo@web39304.mail.mud.yahoo.com> References: <1341685164.3454.51.camel@mythtv> <1341748446.53554.YahooMailNeo@web39304.mail.mud.yahoo.com> Message-ID: <1341812640.3767.6.camel@mythtv> I'm using a cisco/linksys spa 3102 as a analog adapter that is configured to route calls to extension 1000. So an incoming call goes to voip extension 1000, & I want all calls on this ext to be forwarded to 1001-1009. Does this help answer your question? BTW: I'm being side tracked after a power failure took out a partition. So much for having a ups to cya. Anyway a 6tb jfs linux partition is probably lost, so I won't have a lot of time to spend with FreeSwitch issues. On Sun, 2012-07-08 at 04:54 -0700, Stanislav Sinyagin wrote: > you forgot to specify how exactly you route the call to this group. > > > > > ______________________________________________________________ > From: Todd Bailey > To: freeswitch > Sent: Saturday, July 7, 2012 8:19 PM > Subject: [Freeswitch-users] Routing inbound call to multiple > extentions > > > Hi All, > > > At witt's end on this, how do I get a inbound call to go to > multiple > extension? > > I've tried groups but so far no avail > > I using a spa 3102 as a analog adapter that is configured to > rout calls > to extension 1000, I want FS to automatically forward this > call to > extensions 1001-1009. > > Here is an extract from default.xml > (/usr/local/freeswitch/conf/directory) > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From toddb at toddbailey.net Mon Jul 9 09:57:02 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Sun, 08 Jul 2012 22:57:02 -0700 Subject: [Freeswitch-users] Polycom IP501 Message-ID: <1341813422.3767.16.camel@mythtv> Hey all, I just purchased this phone ( ebay $35 total) and should be here in a few days. I think I got a boot server (tftp) all set up so the phone will have a place to get the latest boot code. I've read through most of the FS user docs on this phone and it seems like it should be plug and play after I get the configuration files updated and in place. I not sure what to expect, for the initial install. Basic functionality and maybe a few higher features or does the fun begin after the phone is installed? This phone is intended as a beta test, if it works ok, meaning plug in and it works and advanced features don't take days or weeks to sort out, I'll be adding a few more. Any one have experience with this particular phone on FS and any issues or gotcha's to be aware of? thanks From yehavi.bourvine at gmail.com Mon Jul 9 10:20:07 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Mon, 9 Jul 2012 09:20:07 +0300 Subject: [Freeswitch-users] Polycom IP501 In-Reply-To: <1341813422.3767.16.camel@mythtv> References: <1341813422.3767.16.camel@mythtv> Message-ID: Hell Todd, If you've created the right config files then it should work immediately. However, if you haven't used Polycoms so far I doubt you have the perfect config files. It also depends on which additional functionality you want (like extended function keys, etc.). Polycom's guide includes everything, but it is sort of a refference manual and not a user guide. I suggest you start with the WIKI as it have examples. __Yehavi: 2012/7/9 Todd Bailey > Hey all, I just purchased this phone ( ebay $35 total) and should be > here in a few days. > I think I got a boot server (tftp) all set up so the phone will have a > place to get the latest boot code. > > I've read through most of the FS user docs on this phone and it seems > like it should be plug and play after I get the configuration files > updated and in place. I not sure what to expect, for the initial > install. Basic functionality and maybe a few higher features or does the > fun begin after the phone is installed? > > This phone is intended as a beta test, if it works ok, meaning plug in > and it works and advanced features don't take days or weeks to sort out, > I'll be adding a few more. > > Any one have experience with this particular phone on FS and any issues > or gotcha's to be aware of? > > > thanks > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120709/8518c229/attachment.html From ssinyagin at yahoo.com Mon Jul 9 12:49:17 2012 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Mon, 9 Jul 2012 01:49:17 -0700 (PDT) Subject: [Freeswitch-users] Routing inbound call to multiple extentions In-Reply-To: <1341812640.3767.6.camel@mythtv> References: <1341685164.3454.51.camel@mythtv> <1341748446.53554.YahooMailNeo@web39304.mail.mud.yahoo.com> <1341812640.3767.6.camel@mythtv> Message-ID: <1341823757.69136.YahooMailNeo@web39303.mail.mud.yahoo.com> the group_call function is designed exactly for that: http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide >________________________________ > From: Todd Bailey >To: FreeSWITCH Users Help >Sent: Monday, July 9, 2012 7:44 AM >Subject: Re: [Freeswitch-users] Routing inbound call to multiple extentions > >I'm using a cisco/linksys spa 3102 as a analog adapter that is >configured to route calls to extension 1000. > >So an incoming call goes to voip extension 1000, & I want all calls on >this ext to be forwarded to 1001-1009. > >Does this help answer your question? > >BTW: I'm being side tracked after a power failure took out a partition. >So much for having a ups to cya.? Anyway a 6tb jfs linux partition is >probably lost, so I won't have a lot of time to spend with FreeSwitch >issues.? > > >On Sun, 2012-07-08 at 04:54 -0700, Stanislav Sinyagin wrote: >> you forgot to specify how exactly you route the call to this group. >> >> >> >>? ? ? ? >>? ? ? ? ______________________________________________________________ >>? ? ? ? From: Todd Bailey >>? ? ? ? To: freeswitch >>? ? ? ? Sent: Saturday, July 7, 2012 8:19 PM >>? ? ? ? Subject: [Freeswitch-users] Routing inbound call to multiple >>? ? ? ? extentions >>? ? ? ? >>? ? ? ? >>? ? ? ? Hi All, >>? ? ? ? >>? ? ? ? >>? ? ? ? At witt's end on this,? how do I get a inbound call to go to >>? ? ? ? multiple >>? ? ? ? extension? >>? ? ? ? >>? ? ? ? I've tried groups but so far no avail >>? ? ? ? >>? ? ? ? I using a spa 3102 as a analog adapter that is configured to >>? ? ? ? rout calls >>? ? ? ? to extension 1000,? I want FS to automatically forward this >>? ? ? ? call to >>? ? ? ? extensions 1001-1009. >>? ? ? ? >>? ? ? ? Here is an extract from default.xml >>? ? ? ? (/usr/local/freeswitch/conf/directory) >>? ? ? ? >>? ? ? ? ? >>? ? ? ? ? ? >>? ? ? ? ? ? ? >>? ? ? ? ? ? ? >>? ? ? ? ? ? ? >>? ? ? ? ? ? ? >>? ? ? ? ? ? ? >>? ? ? ? ? ? ? >>? ? ? ? ? ? ? >>? ? ? ? ? ? ? >>? ? ? ? ? ? ? >>? ? ? ? ? ? ? >>? ? ? ? ? ? >>? ? ? ? ? ? ? >>? ? ? ? >>? ? ? ? >>? ? ? ? _________________________________________________________________________ >>? ? ? ? Professional FreeSWITCH Consulting Services: >>? ? ? ? consulting at freeswitch.org >>? ? ? ? http://www.freeswitchsolutions.com >>? ? ? ? >>? ? ? ? >>? ? ? ? >>? ? ? ? >>? ? ? ? Official FreeSWITCH Sites >>? ? ? ? http://www.freeswitch.org >>? ? ? ? http://wiki.freeswitch.org >>? ? ? ? http://www.cluecon.com >>? ? ? ? >>? ? ? ? Join Us At ClueCon - Aug 7-9, 2012 >>? ? ? ? >>? ? ? ? FreeSWITCH-users mailing list >>? ? ? ? FreeSWITCH-users at lists.freeswitch.org >>? ? ? ? http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>? ? ? ? UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>? ? ? ? http://www.freeswitch.org >>? ? ? ? >>? ? ? ? >>? ? ? ? >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >Join Us At ClueCon - Aug 7-9, 2012 > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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/20120709/65dbb08d/attachment-0001.html From =?ISO-8859-15?Q?=22Marcus_M=FClb=FCsch=2C_AS-Infodienste_GmbH?= Mon Jul 9 14:12:46 2012 From: =?ISO-8859-15?Q?=22Marcus_M=FClb=FCsch=2C_AS-Infodienste_GmbH?= (=?ISO-8859-15?Q?=22Marcus_M=FClb=FCsch=2C_AS-Infodienste_GmbH?=) Date: Mon, 09 Jul 2012 12:12:46 +0200 Subject: [Freeswitch-users] Unable to set effective_caller_id_number when bridging using Openzap In-Reply-To: References: <4FF57BA5.7090000@as-infodienste.de> Message-ID: <4FFAAE9E.3010705@as-infodienste.de> Am 07.07.2012 01:19, schrieb Moises Silva: > On Thu, Jul 5, 2012 at 7:33 AM, Marcus M?lb?sch < > muelbuesch at as-infodienste.de> wrote: > >> Hello all, >> >> yes, for hardware reasons (very old Sangoma card with very old >> firmware) I have to use OpenZAP instead of freetTDM. >> >> So, when setting the effective_caller_id that value isn't used. See >> that part of my dialplan here: >> >> > Please use pastebin.com and provide relevant debug logs. > > How are you testing that it does not work? Thank you, you'll find the freeswitch.log here: http://pastebin.com/mz5fSVwA openzap.conf and openzap.conf.cml are here: http://pastebin.com/7WYVitLN and http://pastebin.com/F1zB6hQq zt.conf and wanpipe.conf are here. http://pastebin.com/5M3ezCPv We're using freeswitch 1.0.4 (Sorry again) and WANPIPE Release: 3.3.14.11 The call comes in via an FXS Analog Line on a Digium TDM 2400P and gets bridged to the ISDN line to a local mobile phone. The number is not changed to the number set but remains the number allocated to the fixed line. Thanks for any pointers, Marcus From jayesh.voip at gmail.com Mon Jul 9 16:39:02 2012 From: jayesh.voip at gmail.com (Jayesh Nambiar) Date: Mon, 9 Jul 2012 18:09:02 +0530 Subject: [Freeswitch-users] Call recording quality problems when both legs have different codecs In-Reply-To: References: Message-ID: Hi just checking if anybody had this similar problem in recordings. If anyone can atleast have some pointers for me, I'll really appreciate it. Thanks, --- Jayesh On Wed, Jul 4, 2012 at 4:25 PM, Jayesh Nambiar wrote: > Hi, > I am using mod_callcenter and have enabled recording_template in my queues > for all my calls to be recorded. Now there is a problem in quality of > recorded file when freeswitch answers the call in one codec and connects > with the agent on a different codec. > For eg: The incoming call comes in and get connected with G711u codec. The > caller is listening to MoH while waiting in queue. The agents have G722, > G711u both enabled on their phones and the sofia has outbound-codec-prefs > as G722,PCMU. So when the agent answers, the call is connected with G722 > codec. > So in this case, the recorded file generated plays as if the call is > getting fast-forwarded. The audio in the call is very fast. Whereas if I > change the codec of Agent's phone to G711u only the recorded file quality > is perfect. I don't think it is related to callcenter module anywhere but > since I am using it in callcenter I have given this example. I think its > more related to the record function. > I am running latest GIT version and using mod_shout for recording calls in > mp3 format. I had the same problems when recording in wav format also. Any > help regarding this is really appreciated. I have attached two files; one > which has PCMu on first leg and 722 on other and the second one has PCMu on > both legs. > > Thanks in advance. > > --- Jayesh > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120709/417adcde/attachment.html From josh at foshee.info Mon Jul 9 17:33:08 2012 From: josh at foshee.info (Josh Foshee) Date: Mon, 9 Jul 2012 08:33:08 -0500 Subject: [Freeswitch-users] G_729 Virtual Server Audio Quality Problem Message-ID: Let me start off with the environment that I am using. I have a virtual server that is hosted on Linode.com. They are running Xen Server VPS. The Freeswitch version I am using is FreeSWITCH Version 1.2.0-rc2+git~20120629T194106Z~5f09b40381+unclean~20120630T202857Z I am using mod_com_g729 with 20+ license installed. I have two Sip Profiles setup (internal and external). The internal with authentication is on port 5060 and external with no authentication on port 5080. They have different codec values for each. The Sip provider that I use is Flowroute.com The Leg from Flowroute to Linode I will call Leg A. The Leg from Linode to phones I will call Leg B. The problem that I have is that I can setup the pathway with the codec PCMU. I can place a call and both parties sound find and no audio problems. I can change the Leg A to g729 and still have no audio problems. I then change Leg B to g729 and the remote party (the one on the PSTN network) will hear the site traffic as choppy. The local party will hear quality issues not as much. It is a constant problem on the remote party side though. I then change Leg B back to PCMU and the quality is great. I change it back to G729 and it goes back to bad audio quality. All testing was done between 1am to 2am but this was also reproducible during the daytime. I'm pretty sure it is not a bandwidth problem. I have noticed a problem in the console that when one call is connected and stable this keeps scrolling over and over. http://pastebin.freeswitch.org/19466 Any ideas as to the cause of the audio problems with G729? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120709/250ca652/attachment.html From milosz at kozakiewicz.pl Mon Jul 9 11:41:46 2012 From: milosz at kozakiewicz.pl (Milosz) Date: Mon, 9 Jul 2012 07:41:46 +0000 (UTC) Subject: [Freeswitch-users] spandsp error reading frame References: <4f53a4d6.a526340a.306b.ffff9ea1@mx.google.com> Message-ID: Anita Hall writes: > > > How does your spandsp.conf.xml look like?regards,Anita > On Sun, Mar 4, 2012 at 10:52 PM, Jean-Marc Hyppolite wrote: > > Hello, > ? > I am trying to use mod_spandsp to detect call progress. I haven t been successful so far. I am getting the following error messages. > ? > ================================================ > 2012-03-03 23:55:08.398205 [DEBUG] mod_spandsp_dsp.c:379 (sofia/outbound/? ...) Starting tone detection for '1' > 2012-03-03 23:55:08.398205 [INFO] mod_spandsp_dsp.c:411 (sofia/outbound/? ...) initializing tone detector > 2012-03-03 23:55:08.398205 [DEBUG] switch_core_media_bug.c:462 Attaching BUG to sofia/outbound/? ... > 2012-03-03 23:55:10.218175 [INFO] mod_spandsp_dsp.c:422 (sofia/outbound/? ...) error reading frame > 2012-03-03 23:55:10.218175 [INFO] mod_spandsp_dsp.c:447 (sofia/outbound/? ...) destroying tone detector > ================================================ > ? > Any help would be appreciated. > ? > Thanks > ? > Jean-Marc. > ? > N.B. Sorry if some people received this e-mail twice. I am not sure this email went through the first time. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting-YF8E +gPBBv73h3GqohbjpQ at public.gmane.orghttp://www.freeswitchsolutions.com > http:// www.cudatel.com > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp:// wiki.freeswitch.orghttp://www.cluecon.com > FreeSWITCH-users mailing listFreeSWITCH-users lists.freeswitch.orghttp:// lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- usershttp://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at ... > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at ... > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Hi, Did you followed and solved this issue so far ? Milosz From gabe at gundy.org Mon Jul 9 19:49:20 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 9 Jul 2012 09:49:20 -0600 Subject: [Freeswitch-users] Different Voices for FS In-Reply-To: <20120620132139.33e327b490679d2282e332758c73b55b.6e807b7d68.wbe@email14.secureserver.net> References: <20120620132139.33e327b490679d2282e332758c73b55b.6e807b7d68.wbe@email14.secureserver.net> Message-ID: On Wed, Jun 20, 2012 at 2:21 PM, wrote: > This is my 12th posting does any one ever respond to questions posted here? Yes, they do. Check the archives; there are 10 of thousands of emails. Not everyone will find your question interesting and choose to reply. Still, good luck. Gabe From gabe at gundy.org Mon Jul 9 19:59:32 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 9 Jul 2012 09:59:32 -0600 Subject: [Freeswitch-users] FaxxBochs hackers? In-Reply-To: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com> References: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com> Message-ID: On Tue, Jun 19, 2012 at 11:33 AM, Sean Devoy wrote: > Has anyone investigated the new FaxxBochs offering? It appears to handle the > analog fax machine locally, then send the fax as a data TCP connection to > the server (as a TIFF or JPG I would guess). Given how it works, they should have named it FauxBochs. Gabe From chris at gonumina.com Mon Jul 9 21:02:23 2012 From: chris at gonumina.com (Chris Ferreira) Date: Mon, 9 Jul 2012 13:02:23 -0400 Subject: [Freeswitch-users] FaxxBochs hackers? In-Reply-To: References: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com> Message-ID: <6432809444531730803@unknownmsgid> I think it would be wonderful if all VoIP admins and providers collectively told the people needing fax service that it has been completely phased out. I think it would be great if stores stopped selling fax machines and all remaining fax machines were recycled. The only reason faxing still exists is because of the few people that refuse to let it die. ___________________ Mobile Reply On Jul 9, 2012, at 12:58 PM, Gabriel Gunderson wrote: > On Tue, Jun 19, 2012 at 11:33 AM, Sean Devoy wrote: >> Has anyone investigated the new FaxxBochs offering? It appears to handle the >> analog fax machine locally, then send the fax as a data TCP connection to >> the server (as a TIFF or JPG I would guess). > > Given how it works, they should have named it FauxBochs. > > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From toddb at toddbailey.net Mon Jul 9 21:08:05 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Mon, 09 Jul 2012 10:08:05 -0700 Subject: [Freeswitch-users] Routing inbound call to multiple extentions In-Reply-To: <1341823757.69136.YahooMailNeo@web39303.mail.mud.yahoo.com> References: <1341685164.3454.51.camel@mythtv> <1341748446.53554.YahooMailNeo@web39304.mail.mud.yahoo.com> <1341812640.3767.6.camel@mythtv> <1341823757.69136.YahooMailNeo@web39303.mail.mud.yahoo.com> Message-ID: <1341853685.4860.3.camel@mythtv> I think what is missing in this wiki is it doesn't explicitly state what file(s) and in what folder(s) needs to be edited. here is an extract of the section that I suspect is what I need to follow Groups A group is a logical collection of users that FreeSWITCH can use to bridge calls in a serial or parallel fashion, depending on the arguments to the group_call application. Using groups is optional -- you can put your users straight into the domain section if you desire. This is specially useful if you use mod_xml_curl to provide the user directory to FreeSWITCH and want to group some users in a logical structure. The following group '200' groups four users. It's interesting to notice the "dial-string" param, which is used to bridge the calls to those users. Users 1000 and 1001 will use the default "dial-string" while user 2014 uses a loopback channel so FreeSWITCH can actually query the dialplan to figure out how to reach that user (this also works for external numbers through OpenZAP and gateways): type="pointer" is a pointer so you can have the same user in multiple groups. It basically means to keep searching for the user in the directory.
The dialplan for the above group can be defined like this: The extension 200 will ring all the users defined in the user directory for group 200 in a serial fashion (specified by the +F argument, if you want to ring all the users at once use the +A argument) for 15 seconds, then it will transfer the call to the same group again so the call will ring the group infinitely. To explain the variables set prior to the bridge: The hangup_after_bridge is set to true for this effect: if the bridge is successfully answered and the B-leg later hangs up, the A-leg will also be hung up. The continue_on_fail is set to true for this: if the bridge fails, dialplan execution will continue and the transfer will be executed. The originate_continue_on_timeout is set to true for this: if the bridge's dial string specifies several destinations separated by the "|" (this is for failover), the bridge will time out on an unanswered destination and will attempt the next specified destination. Without originate_continue_on_timeout set to true, the bridge will time out on the first destination it tries and the bridge itself will fail. (In the example above, the bridge string is generated by group_call with the +F option; this specifies a dial string with all the group's destinations separated by "|". So originate_continue_on_timeout needs to be set to true for serial calling behavior.) The call_timeout is set so that the bridge attempt to a destination that goes unanswered will time out. NOTE: If the destination sends early media, the bridge will be answered (pre-answered) and will NOT time out! To time out a bridge attempt to a destination sending early media, set ignore_early_media to true. The dialplan for user 2014, which in this example happens to be terminated through a gateway defined via mod_lcr, can be defined like this: On Mon, 2012-07-09 at 01:49 -0700, Stanislav Sinyagin wrote: > http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide From bdfoster at endigotech.com Mon Jul 9 21:20:01 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 9 Jul 2012 13:20:01 -0400 Subject: [Freeswitch-users] FaxxBochs hackers? In-Reply-To: References: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com> Message-ID: Gabe, ...they suck? Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 9, 2012 12:31 PM, "Gabriel Gunderson" wrote: > On Tue, Jun 19, 2012 at 11:33 AM, Sean Devoy > wrote: > > Has anyone investigated the new FaxxBochs offering? It appears to handle > the > > analog fax machine locally, then send the fax as a data TCP connection to > > the server (as a TIFF or JPG I would guess). > > Given how it works, they should have named it FauxBochs. > > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120709/ff0017f3/attachment.html From msc at freeswitch.org Mon Jul 9 21:47:05 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 9 Jul 2012 10:47:05 -0700 Subject: [Freeswitch-users] Routing inbound call to multiple extentions In-Reply-To: <1341853685.4860.3.camel@mythtv> References: <1341685164.3454.51.camel@mythtv> <1341748446.53554.YahooMailNeo@web39304.mail.mud.yahoo.com> <1341812640.3767.6.camel@mythtv> <1341823757.69136.YahooMailNeo@web39303.mail.mud.yahoo.com> <1341853685.4860.3.camel@mythtv> Message-ID: That may be true, because you don't always know what files/folders to edit. Technically, you may not even need files/folders because you are using mod_xml_curl or something like that. However, we could add a sentence like: "If you are using the default 'vanilla' configs, the groups are defined in conf/directory/default.xml and the dialplan entries are found in conf/dialplan/default.xml" Would that help? If so, please add that text as I'm positive that it's correct. If you are having trouble with wiki editing let me know and I'll be happy to help. At some point one of these days I'm gonna do screencast w/ training on how to do wiki editing... -MC On Mon, Jul 9, 2012 at 10:08 AM, Todd Bailey wrote: > I think what is missing in this wiki is it doesn't explicitly state what > file(s) and in what folder(s) needs to be edited. > > > > here is an extract of the section that I suspect is what I need to > follow > > Groups > > A group is a logical collection of users that FreeSWITCH can use to > bridge calls in a serial or parallel fashion, depending on the arguments > to the group_call application. Using groups is optional -- you can put > your users straight into the domain section if you desire. > > This is specially useful if you use mod_xml_curl to provide the user > directory to FreeSWITCH and want to group some users in a logical > structure. The following group '200' groups four users. It's interesting > to notice the "dial-string" param, which is used to bridge the calls to > those users. Users 1000 and 1001 will use the default "dial-string" > while user 2014 uses a loopback channel so FreeSWITCH can actually query > the dialplan to figure out how to reach that user (this also works for > external numbers through OpenZAP and gateways): > > type="pointer" is a pointer so you can have the same user in multiple > groups. It basically means to keep searching for the user in the > directory. > > >
> > > > > > > > > > > > > > > > > > > > > > > > > value="loopback/2014/default/XML"/> > > > > > > > > > > > >
> > > The dialplan for the above group can be defined like this: > > > > > > data="originate_continue_on_timeout=true"/> > > > > > > > > The extension 200 will ring all the users defined in the user directory > for group 200 in a serial fashion (specified by the +F argument, if you > want to ring all the users at once use the +A argument) for 15 seconds, > then it will transfer the call to the same group again so the call will > ring the group infinitely. > > To explain the variables set prior to the bridge: > > The hangup_after_bridge is set to true for this effect: if the bridge is > successfully answered and the B-leg later hangs up, the A-leg will also > be hung up. > > The continue_on_fail is set to true for this: if the bridge fails, > dialplan execution will continue and the transfer will be executed. > > The originate_continue_on_timeout is set to true for this: if the > bridge's dial string specifies several destinations separated by the > "|" (this is for failover), the bridge will time out on an unanswered > destination and will attempt the next specified destination. Without > originate_continue_on_timeout set to true, the bridge will time out on > the first destination it tries and the bridge itself will fail. (In the > example above, the bridge string is generated by group_call with the +F > option; this specifies a dial string with all the group's destinations > separated by "|". So originate_continue_on_timeout needs to be set to > true for serial calling behavior.) > > The call_timeout is set so that the bridge attempt to a destination that > goes unanswered will time out. NOTE: If the destination sends early > media, the bridge will be answered (pre-answered) and will NOT time out! > To time out a bridge attempt to a destination sending early media, set > ignore_early_media to true. > > The dialplan for user 2014, which in this example happens to be > terminated through a gateway defined via mod_lcr, can be defined like > this: > > > > > > > > > > > > > > > > > On Mon, 2012-07-09 at 01:49 -0700, Stanislav Sinyagin wrote: > > http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120709/307ecb0b/attachment.html From marketing at cluecon.com Mon Jul 9 22:26:54 2012 From: marketing at cluecon.com (Michael Collins) Date: Mon, 9 Jul 2012 11:26:54 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: Greetings all! We are back to a full week after taking last Wednesday off. This weekwe have Dave Kompel scheduled to talk to us on a subject that has been of much interest in recent weeks: firewall security on Windows deployments of FreeSWITCH. Many of you know that we have fail2banavailable for Linux/Unix based installs, but what about Windows ? There are indeed techniques for Windows systems to update the Windows firewall to prevent things like the friendly scanner . Recall: the friendly scanner is an abuse of a SIP testing tool called SIPVicious. People use it for nefarious purposes, so having the ability to lock down the source IP address on the fly is a powerful technique to employ. We look forward to this discussion . We are happy to report that we have a new GOLD sponsor at ClueCon: Flowroute! The folks at Flowroute have graciously offered to make sure that our attendees are well taken care of. First, they are supplying the gift bags that all of our attendees will receive. In those bags will be lots of goodies from Flowroute and our other sponsors. A special item will be a $25 gift card for Flowroute. Additionally, Flowroute is sponsoring a pizza party on Monday night, August 6 at Gino'sin downtown Chicago! We hope you all will be able to join us. Only a few more weeks until we're all together again! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120709/e5129182/attachment-0001.html From sip at inbox.com Mon Jul 9 22:41:26 2012 From: sip at inbox.com (Jimmy Godbout) Date: Mon, 9 Jul 2012 10:41:26 -0800 Subject: [Freeswitch-users] Why FS recreates indexes when it starts Message-ID: Hi, I'd like to know why does FS tries to recreate its indexes when it restarts. When using a clustered solution, FS will then fail because the DB is still running and the indexes a still there. ____________________________________________________________ Send any screenshot to your friends in seconds... Works in all emails, instant messengers, blogs, forums and social networks. TRY IM TOOLPACK at http://www.imtoolpack.com/default.aspx?rc=if2 for FREE From andrew at cassidywebservices.co.uk Mon Jul 9 22:42:59 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Mon, 9 Jul 2012 19:42:59 +0100 Subject: [Freeswitch-users] FaxxBochs hackers? In-Reply-To: <6432809444531730803@unknownmsgid> References: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com> <6432809444531730803@unknownmsgid> Message-ID: True, but in some jurisdictions a signed fax is admissible as court evidence where the same email transmission would not be. So from a legal standpoint there's still a need, unfortunately. However, building a 'virtual fax machine' on freeswitch isn't particularly difficult. On 9 July 2012 18:02, Chris Ferreira wrote: > I think it would be wonderful if all VoIP admins and providers > collectively told the people needing fax service that it has been > completely phased out. > > I think it would be great if stores stopped selling fax machines and > all remaining fax machines were recycled. > > The only reason faxing still exists is because of the few people that > refuse to let it die. > > > > > ___________________ > Mobile Reply > > On Jul 9, 2012, at 12:58 PM, Gabriel Gunderson wrote: > > > On Tue, Jun 19, 2012 at 11:33 AM, Sean Devoy > wrote: > >> Has anyone investigated the new FaxxBochs offering? It appears to > handle the > >> analog fax machine locally, then send the fax as a data TCP connection > to > >> the server (as a TIFF or JPG I would guess). > > > > Given how it works, they should have named it FauxBochs. > > > > > > Gabe > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120709/ab9e506c/attachment.html From gabe at gundy.org Mon Jul 9 22:56:31 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 9 Jul 2012 12:56:31 -0600 Subject: [Freeswitch-users] FaxxBochs hackers? In-Reply-To: References: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com> Message-ID: On Mon, Jul 9, 2012 at 9:59 AM, Gabriel Gunderson wrote: > Given how it works, they should have named it FauxBochs. No, it wasn't meant to be critical of the product; I've never used it. Rather, it was commentary on how they're faking faxs. " It appears to handle the analog fax machine locally, then send the fax as a data TCP connection to the server (as a TIFF or JPG I would guess)." Faux: Not genuine; fake or false... "her faux New York accent". Anything that kills real fax is okay by me. Nothing more :) Gabe From gabe at gundy.org Mon Jul 9 22:58:43 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Mon, 9 Jul 2012 12:58:43 -0600 Subject: [Freeswitch-users] [Freeswitch-dev] FreeSWITCH Weekly News and Notes In-Reply-To: References: Message-ID: On Mon, Jul 9, 2012 at 12:26 PM, Michael Collins wrote: > Additionally, Flowroute is sponsoring a pizza party on Monday night, August > 6 at Gino's in downtown Chicago! We hope you all will be able to join us. That's my kind of sponsor! Thanks Flowroute! Gabe From mgg at giagnocavo.net Mon Jul 9 23:03:39 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Mon, 9 Jul 2012 19:03:39 +0000 Subject: [Freeswitch-users] G_729 Virtual Server Audio Quality Problem In-Reply-To: References: Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B6045115E3@BLUPRD0711MB413.namprd07.prod.outlook.com> Grab a full RTP capture, and you can use Wireshark to see what is happening to your media. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Josh Foshee Sent: Monday, July 09, 2012 7:33 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] G_729 Virtual Server Audio Quality Problem Let me start off with the environment that I am using. I have a virtual server that is hosted on Linode.com. They are running Xen Server VPS. The Freeswitch version I am using is FreeSWITCH Version 1.2.0-rc2+git~20120629T194106Z~5f09b40381+unclean~20120630T202857Z I am using mod_com_g729 with 20+ license installed. I have two Sip Profiles setup (internal and external). The internal with authentication is on port 5060 and external with no authentication on port 5080. They have different codec values for each. The Sip provider that I use is Flowroute.com The Leg from Flowroute to Linode I will call Leg A. The Leg from Linode to phones I will call Leg B. The problem that I have is that I can setup the pathway with the codec PCMU. I can place a call and both parties sound find and no audio problems. I can change the Leg A to g729 and still have no audio problems. I then change Leg B to g729 and the remote party (the one on the PSTN network) will hear the site traffic as choppy. The local party will hear quality issues not as much. It is a constant problem on the remote party side though. I then change Leg B back to PCMU and the quality is great. I change it back to G729 and it goes back to bad audio quality. All testing was done between 1am to 2am but this was also reproducible during the daytime. I'm pretty sure it is not a bandwidth problem. I have noticed a problem in the console that when one call is connected and stable this keeps scrolling over and over. http://pastebin.freeswitch.org/19466 Any ideas as to the cause of the audio problems with G729? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120709/7885c57e/attachment-0001.html From chris at gonumina.com Tue Jul 10 00:10:48 2012 From: chris at gonumina.com (Chris Ferreira) Date: Mon, 9 Jul 2012 16:10:48 -0400 Subject: [Freeswitch-users] FaxxBochs hackers? In-Reply-To: References: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com> <6432809444531730803@unknownmsgid> Message-ID: <8849853856102862287@unknownmsgid> I was more speaking along the lines of analog fax machines. We continue to support them and it's stunting innovation. ___________________ Mobile Reply On Jul 9, 2012, at 4:05 PM, Andrew Cassidy wrote: True, but in some jurisdictions a signed fax is admissible as court evidence where the same email transmission would not be. So from a legal standpoint there's still a need, unfortunately. However, building a 'virtual fax machine' on freeswitch isn't particularly difficult. On 9 July 2012 18:02, Chris Ferreira wrote: > I think it would be wonderful if all VoIP admins and providers > collectively told the people needing fax service that it has been > completely phased out. > > I think it would be great if stores stopped selling fax machines and > all remaining fax machines were recycled. > > The only reason faxing still exists is because of the few people that > refuse to let it die. > > > > > ___________________ > Mobile Reply > > On Jul 9, 2012, at 12:58 PM, Gabriel Gunderson wrote: > > > On Tue, Jun 19, 2012 at 11:33 AM, Sean Devoy > wrote: > >> Has anyone investigated the new FaxxBochs offering? It appears to > handle the > >> analog fax machine locally, then send the fax as a data TCP connection > to > >> the server (as a TIFF or JPG I would guess). > > > > Given how it works, they should have named it FauxBochs. > > > > > > Gabe > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120709/b13b32a8/attachment.html From asaad2 at gmail.com Tue Jul 10 02:36:53 2012 From: asaad2 at gmail.com (BookBag) Date: Mon, 9 Jul 2012 17:36:53 -0500 Subject: [Freeswitch-users] FaxxBochs hackers? In-Reply-To: <6432809444531730803@unknownmsgid> References: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com> <6432809444531730803@unknownmsgid> Message-ID: You know if I told my boss that all fax machines were phased out . I wouldn't have a job. He's gonna say my job is to make our equipment work not force them to switch to different methods On Jul 9, 2012 2:18 PM, "Chris Ferreira" wrote: > I think it would be wonderful if all VoIP admins and providers > collectively told the people needing fax service that it has been > completely phased out. > > I think it would be great if stores stopped selling fax machines and > all remaining fax machines were recycled. > > The only reason faxing still exists is because of the few people that > refuse to let it die. > > > > > ___________________ > Mobile Reply > > On Jul 9, 2012, at 12:58 PM, Gabriel Gunderson wrote: > > > On Tue, Jun 19, 2012 at 11:33 AM, Sean Devoy > wrote: > >> Has anyone investigated the new FaxxBochs offering? It appears to > handle the > >> analog fax machine locally, then send the fax as a data TCP connection > to > >> the server (as a TIFF or JPG I would guess). > > > > Given how it works, they should have named it FauxBochs. > > > > > > Gabe > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120709/1b4b05f2/attachment.html From chris at gonumina.com Tue Jul 10 04:12:35 2012 From: chris at gonumina.com (Chris Ferreira) Date: Mon, 9 Jul 2012 20:12:35 -0400 Subject: [Freeswitch-users] FaxxBochs hackers? In-Reply-To: References: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com> <6432809444531730803@unknownmsgid> Message-ID: <8251983215579532490@unknownmsgid> I disagree. The job of any telecom or IT professional is to also educate and circulate out antiquated equipment. I don't know why we still accept the use of faxes when things like typewriters have pretty much been eliminated. ___________________ Mobile Reply On Jul 9, 2012, at 7:07 PM, BookBag wrote: You know if I told my boss that all fax machines were phased out . I wouldn't have a job. He's gonna say my job is to make our equipment work not force them to switch to different methods On Jul 9, 2012 2:18 PM, "Chris Ferreira" wrote: > I think it would be wonderful if all VoIP admins and providers > collectively told the people needing fax service that it has been > completely phased out. > > I think it would be great if stores stopped selling fax machines and > all remaining fax machines were recycled. > > The only reason faxing still exists is because of the few people that > refuse to let it die. > > > > > ___________________ > Mobile Reply > > On Jul 9, 2012, at 12:58 PM, Gabriel Gunderson wrote: > > > On Tue, Jun 19, 2012 at 11:33 AM, Sean Devoy > wrote: > >> Has anyone investigated the new FaxxBochs offering? It appears to > handle the > >> analog fax machine locally, then send the fax as a data TCP connection > to > >> the server (as a TIFF or JPG I would guess). > > > > Given how it works, they should have named it FauxBochs. > > > > > > Gabe > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120709/44690cd0/attachment-0001.html From mgg at giagnocavo.net Tue Jul 10 05:55:35 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Tue, 10 Jul 2012 01:55:35 +0000 Subject: [Freeswitch-users] FaxxBochs hackers? In-Reply-To: <8251983215579532490@unknownmsgid> References: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com> <6432809444531730803@unknownmsgid> <8251983215579532490@unknownmsgid> Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B604511892@BLUPRD0711MB413.namprd07.prod.outlook.com> Faxes carry legal standing that email does not. So, you can want a better solution, but the reality is that fax won't disappear soon. In fact, as I understand, part of T.38 was driven by legal requirements. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Ferreira Sent: Monday, July 09, 2012 6:13 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FaxxBochs hackers? I disagree. The job of any telecom or IT professional is to also educate and circulate out antiquated equipment. I don't know why we still accept the use of faxes when things like typewriters have pretty much been eliminated. ___________________ Mobile Reply On Jul 9, 2012, at 7:07 PM, BookBag > wrote: You know if I told my boss that all fax machines were phased out . I wouldn't have a job. He's gonna say my job is to make our equipment work not force them to switch to different methods On Jul 9, 2012 2:18 PM, "Chris Ferreira" > wrote: I think it would be wonderful if all VoIP admins and providers collectively told the people needing fax service that it has been completely phased out. I think it would be great if stores stopped selling fax machines and all remaining fax machines were recycled. The only reason faxing still exists is because of the few people that refuse to let it die. ___________________ Mobile Reply On Jul 9, 2012, at 12:58 PM, Gabriel Gunderson > wrote: > On Tue, Jun 19, 2012 at 11:33 AM, Sean Devoy > wrote: >> Has anyone investigated the new FaxxBochs offering? It appears to handle the >> analog fax machine locally, then send the fax as a data TCP connection to >> the server (as a TIFF or JPG I would guess). > > Given how it works, they should have named it FauxBochs. > > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120710/fcd69d2c/attachment.html From chris at gonumina.com Tue Jul 10 06:30:37 2012 From: chris at gonumina.com (Chris Ferreira) Date: Mon, 9 Jul 2012 22:30:37 -0400 Subject: [Freeswitch-users] FaxxBochs hackers? In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B604511892@BLUPRD0711MB413.namprd07.prod.outlook.com> References: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com> <6432809444531730803@unknownmsgid> <8251983215579532490@unknownmsgid> <63B00DD1DA6A364E9F64A3A0BD2FE7B604511892@BLUPRD0711MB413.namprd07.prod.outlook.com> Message-ID: <-9115573625816947171@unknownmsgid> There are more modern ways of legally transmitting documents. Signed and all. I have several clients that are practicing lawyers in multiple states and my own brother has a law firm. These are simply excuses. My point is that there ARE better ways. We just keep enabling a poor, old, and broken technology. ___________________ Mobile Reply On Jul 9, 2012, at 10:09 PM, Michael Giagnocavo wrote: Faxes carry legal standing that email does not. So, you can want a better solution, but the reality is that fax won?t disappear soon. In fact, as I understand, part of T.38 was driven by legal requirements. -Michael *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris Ferreira *Sent:* Monday, July 09, 2012 6:13 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] FaxxBochs hackers? I disagree. The job of any telecom or IT professional is to also educate and circulate out antiquated equipment. I don't know why we still accept the use of faxes when things like typewriters have pretty much been eliminated. ___________________ Mobile Reply On Jul 9, 2012, at 7:07 PM, BookBag wrote: You know if I told my boss that all fax machines were phased out . I wouldn't have a job. He's gonna say my job is to make our equipment work not force them to switch to different methods On Jul 9, 2012 2:18 PM, "Chris Ferreira" wrote: I think it would be wonderful if all VoIP admins and providers collectively told the people needing fax service that it has been completely phased out. I think it would be great if stores stopped selling fax machines and all remaining fax machines were recycled. The only reason faxing still exists is because of the few people that refuse to let it die. ___________________ Mobile Reply On Jul 9, 2012, at 12:58 PM, Gabriel Gunderson wrote: > On Tue, Jun 19, 2012 at 11:33 AM, Sean Devoy wrote: >> Has anyone investigated the new FaxxBochs offering? It appears to handle the >> analog fax machine locally, then send the fax as a data TCP connection to >> the server (as a TIFF or JPG I would guess). > > Given how it works, they should have named it FauxBochs. > > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120709/49a4d74f/attachment-0001.html From dujinfang at gmail.com Tue Jul 10 06:36:55 2012 From: dujinfang at gmail.com (Seven Du) Date: Tue, 10 Jul 2012 10:36:55 +0800 Subject: [Freeswitch-users] Attended transfer via ESL In-Reply-To: References: <1335539693783-7506468.post@n2.nabble.com> <1336657979551-7546785.post@n2.nabble.com> Message-ID: <229B56BE41DF403D81C0C19EAE96D47B@gmail.com> correction: uuid_broadcast att_xfer:user/1001 was wrong, should be uuid_broadcast att_xfer::user/1001 However my problem is different than Limit: 1) when I use att_xfer in erlang with sendmsg, it works, but no MOH on the other leg 2) when I use uuid_broadcast, everthing is ok, and DTMF works as expected, but uuid_recv_dtmf uuid *#0 doesn't work, I tried to use that to control a xfer On Friday, May 11, 2012 at 2:18 AM, Michael Collins wrote: > Cool, thanks for letting us know it worked. > -MC > > On Thu, May 10, 2012 at 6:52 AM, Limit wrote: > > *mercutioviz*, thank you! > > > > Looks like it works for me! Transfer works just as called from dialplan. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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/20120710/0d1bde1e/attachment.html From mgg at giagnocavo.net Tue Jul 10 08:14:28 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Tue, 10 Jul 2012 04:14:28 +0000 Subject: [Freeswitch-users] FaxxBochs hackers? In-Reply-To: <-9115573625816947171@unknownmsgid> References: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com> <6432809444531730803@unknownmsgid> <8251983215579532490@unknownmsgid> <63B00DD1DA6A364E9F64A3A0BD2FE7B604511892@BLUPRD0711MB413.namprd07.prod.outlook.com> <-9115573625816947171@unknownmsgid> Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B6045119B9@BLUPRD0711MB413.namprd07.prod.outlook.com> Sure, and I wholeheartedly agree. But none of these arguments are compelling to users, and making them just means you'll lose a sale. It's unlikely a large fax customer is going to say "Wow, you know, I now realize faxes are just ancient relics, and we're going to ditch them." By not enabling faxes, you only end up harming the customer. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Ferreira Sent: Monday, July 09, 2012 8:31 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FaxxBochs hackers? There are more modern ways of legally transmitting documents. Signed and all. I have several clients that are practicing lawyers in multiple states and my own brother has a law firm. These are simply excuses. My point is that there ARE better ways. We just keep enabling a poor, old, and broken technology. ___________________ Mobile Reply On Jul 9, 2012, at 10:09 PM, Michael Giagnocavo > wrote: Faxes carry legal standing that email does not. So, you can want a better solution, but the reality is that fax won't disappear soon. In fact, as I understand, part of T.38 was driven by legal requirements. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Chris Ferreira Sent: Monday, July 09, 2012 6:13 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FaxxBochs hackers? I disagree. The job of any telecom or IT professional is to also educate and circulate out antiquated equipment. I don't know why we still accept the use of faxes when things like typewriters have pretty much been eliminated. ___________________ Mobile Reply On Jul 9, 2012, at 7:07 PM, BookBag > wrote: You know if I told my boss that all fax machines were phased out . I wouldn't have a job. He's gonna say my job is to make our equipment work not force them to switch to different methods On Jul 9, 2012 2:18 PM, "Chris Ferreira" > wrote: I think it would be wonderful if all VoIP admins and providers collectively told the people needing fax service that it has been completely phased out. I think it would be great if stores stopped selling fax machines and all remaining fax machines were recycled. The only reason faxing still exists is because of the few people that refuse to let it die. ___________________ Mobile Reply On Jul 9, 2012, at 12:58 PM, Gabriel Gunderson > wrote: > On Tue, Jun 19, 2012 at 11:33 AM, Sean Devoy > wrote: >> Has anyone investigated the new FaxxBochs offering? It appears to handle the >> analog fax machine locally, then send the fax as a data TCP connection to >> the server (as a TIFF or JPG I would guess). > > Given how it works, they should have named it FauxBochs. > > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120710/50b58e18/attachment-0001.html From bdfoster at endigotech.com Tue Jul 10 15:12:30 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 10 Jul 2012 07:12:30 -0400 Subject: [Freeswitch-users] FaxxBochs hackers? In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B6045119B9@BLUPRD0711MB413.namprd07.prod.outlook.com> References: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com> <6432809444531730803@unknownmsgid> <8251983215579532490@unknownmsgid> <63B00DD1DA6A364E9F64A3A0BD2FE7B604511892@BLUPRD0711MB413.namprd07.prod.outlook.com> <-9115573625816947171@unknownmsgid> <63B00DD1DA6A364E9F64A3A0BD2FE7B6045119B9@BLUPRD0711MB413.namprd07.prod.outlook.com> Message-ID: Besides, fax is fun, right? Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 10, 2012 12:43 AM, "Michael Giagnocavo" wrote: > Sure, and I wholeheartedly agree. But none of these arguments are > compelling to users, and making them just means you?ll lose a sale. It?s > unlikely a large fax customer is going to say ?Wow, you know, I now realize > faxes are just ancient relics, and we?re going to ditch them.? By not > enabling faxes, you only end up harming the customer.**** > > ** ** > > -Michael**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris > Ferreira > *Sent:* Monday, July 09, 2012 8:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] FaxxBochs hackers?**** > > ** ** > > There are more modern ways of legally transmitting documents. Signed and > all. I have several clients that are practicing lawyers in multiple states > and my own brother has a law firm.**** > > ** ** > > These are simply excuses. My point is that there ARE better ways. We just > keep enabling a poor, old, and broken technology.**** > > ___________________**** > > Mobile Reply**** > > > On Jul 9, 2012, at 10:09 PM, Michael Giagnocavo > wrote:**** > > Faxes carry legal standing that email does not. So, you can want a > better solution, but the reality is that fax won?t disappear soon. In fact, > as I understand, part of T.38 was driven by legal requirements.**** > > **** > > -Michael**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris > Ferreira > *Sent:* Monday, July 09, 2012 6:13 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] FaxxBochs hackers?**** > > **** > > I disagree. The job of any telecom or IT professional is to also educate > and circulate out antiquated equipment.**** > > **** > > I don't know why we still accept the use of faxes when things like > typewriters have pretty much been eliminated.**** > > ___________________**** > > Mobile Reply**** > > > On Jul 9, 2012, at 7:07 PM, BookBag wrote:**** > > You know if I told my boss that all fax machines were phased out . I > wouldn't have a job. He's gonna say my job is to make our equipment work > not force them to switch to different methods**** > > On Jul 9, 2012 2:18 PM, "Chris Ferreira" wrote:**** > > I think it would be wonderful if all VoIP admins and providers > collectively told the people needing fax service that it has been > completely phased out. > > I think it would be great if stores stopped selling fax machines and > all remaining fax machines were recycled. > > The only reason faxing still exists is because of the few people that > refuse to let it die. > > > > > ___________________ > Mobile Reply > > On Jul 9, 2012, at 12:58 PM, Gabriel Gunderson wrote: > > > On Tue, Jun 19, 2012 at 11:33 AM, Sean Devoy > wrote: > >> Has anyone investigated the new FaxxBochs offering? It appears to > handle the > >> analog fax machine locally, then send the fax as a data TCP connection > to > >> the server (as a TIFF or JPG I would guess). > > > > Given how it works, they should have named it FauxBochs. > > > > > > Gabe > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120710/1a114a60/attachment.html From steveu at coppice.org Tue Jul 10 18:53:53 2012 From: steveu at coppice.org (Steve Underwood) Date: Tue, 10 Jul 2012 22:53:53 +0800 Subject: [Freeswitch-users] FaxxBochs hackers? In-Reply-To: <6432809444531730803@unknownmsgid> References: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com> <6432809444531730803@unknownmsgid> Message-ID: <4FFC4201.4000904@coppice.org> On 07/10/2012 01:02 AM, Chris Ferreira wrote: > I think it would be wonderful if all VoIP admins and providers > collectively told the people needing fax service that it has been > completely phased out. > > I think it would be great if stores stopped selling fax machines and > all remaining fax machines were recycled. > > The only reason faxing still exists is because of the few people that > refuse to let it die. Right, and burn them at the stake if they won't show allegiance to e-mail. Steve > > > > > ___________________ > Mobile Reply > > On Jul 9, 2012, at 12:58 PM, Gabriel Gunderson wrote: > >> On Tue, Jun 19, 2012 at 11:33 AM, Sean Devoy wrote: >>> Has anyone investigated the new FaxxBochs offering? It appears to handle the >>> analog fax machine locally, then send the fax as a data TCP connection to >>> the server (as a TIFF or JPG I would guess). >> Given how it works, they should have named it FauxBochs. >> >> >> Gabe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From wstephen80 at gmail.com Tue Jul 10 19:51:59 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Tue, 10 Jul 2012 17:51:59 +0200 Subject: [Freeswitch-users] Codec Negotiation Message-ID: Is this message flow valid? FS ---------------- MyProvider ---> INVITE (w/SDP G729, G711a, G711u) <--- 100 Trying <--- 183 Session Progress (w/SDP G729, G711a, G711u) <--- 200 OK (w/SDP G729, G711a, G711u) What is strange to me is that my provider send back a 183 Session Progress with all codec list of Freeswitch INVITE (I expect only the choosen codec). It's this message flow correct? Stephen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120710/c259c20e/attachment.html From shouldbeq931 at gmail.com Tue Jul 10 20:48:03 2012 From: shouldbeq931 at gmail.com (shouldbe q931) Date: Tue, 10 Jul 2012 17:48:03 +0100 Subject: [Freeswitch-users] FaxxBochs hackers? In-Reply-To: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com> References: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com> Message-ID: On Tue, Jun 19, 2012 at 6:33 PM, Sean Devoy wrote: > Has anyone investigated the new FaxxBochs offering? It appears to handle the > analog fax machine locally, then send the fax as a data TCP connection to > the server (as a TIFF or JPG I would guess). It totally removes the VOIP > problem with fax machines and is a clever solution. $30 a month unlimited > in and out is reasonable, but I would rather have that income myself. > > > > It would be great if I could resell it and make a cut, but they require 75 > device sales the first year. > > > > If someone knows anything about ?reconfiguring? one of these, I would love > to read about it. > > > > Sean > > > _________________________________________________________________________ looking at their other offerings, I would guess that its a four port asterisk box with IAX modem connecting to hylafax to "receive" the inbound fax and then send it to their "datacenter" hyalafax server(s) for onward transmission using the "proxy" feature in jobcontrol, inbound would work in the same way but the other way around. The AES 256 encryption could well just be SSL... not a bad solution if you must have fax and don't have PSTN (or ISDN) connectivity, but as its store and forward, I'm not sure how it plays out legally... From andrew at cassidywebservices.co.uk Tue Jul 10 20:50:44 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 10 Jul 2012 17:50:44 +0100 Subject: [Freeswitch-users] FaxxBochs hackers? In-Reply-To: References: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com> <6432809444531730803@unknownmsgid> <8251983215579532490@unknownmsgid> <63B00DD1DA6A364E9F64A3A0BD2FE7B604511892@BLUPRD0711MB413.namprd07.prod.outlook.com> <-9115573625816947171@unknownmsgid> <63B00DD1DA6A364E9F64A3A0BD2FE7B6045119B9@BLUPRD0711MB413.namprd07.prod.outlook.com> Message-ID: Yes, faxing is great fun! But it's true, if you're working for someone you can recommend better ways, but if a client says 'I want fax' and you say 'no', kiss that sale goodbye. I'd rather support faxing (which freeswitch makes pretty simple) than lose out on all the money! On 10 July 2012 12:12, Brian Foster wrote: > Besides, fax is fun, right? > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > On Jul 10, 2012 12:43 AM, "Michael Giagnocavo" wrote: > >> Sure, and I wholeheartedly agree. But none of these arguments are >> compelling to users, and making them just means you?ll lose a sale. It?s >> unlikely a large fax customer is going to say ?Wow, you know, I now realize >> faxes are just ancient relics, and we?re going to ditch them.? By not >> enabling faxes, you only end up harming the customer.**** >> >> ** ** >> >> -Michael**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris >> Ferreira >> *Sent:* Monday, July 09, 2012 8:31 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] FaxxBochs hackers?**** >> >> ** ** >> >> There are more modern ways of legally transmitting documents. Signed and >> all. I have several clients that are practicing lawyers in multiple states >> and my own brother has a law firm.**** >> >> ** ** >> >> These are simply excuses. My point is that there ARE better ways. We just >> keep enabling a poor, old, and broken technology.**** >> >> ___________________**** >> >> Mobile Reply**** >> >> >> On Jul 9, 2012, at 10:09 PM, Michael Giagnocavo >> wrote:**** >> >> Faxes carry legal standing that email does not. So, you can want a >> better solution, but the reality is that fax won?t disappear soon. In fact, >> as I understand, part of T.38 was driven by legal requirements.**** >> >> **** >> >> -Michael**** >> >> **** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Chris >> Ferreira >> *Sent:* Monday, July 09, 2012 6:13 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] FaxxBochs hackers?**** >> >> **** >> >> I disagree. The job of any telecom or IT professional is to also educate >> and circulate out antiquated equipment.**** >> >> **** >> >> I don't know why we still accept the use of faxes when things like >> typewriters have pretty much been eliminated.**** >> >> ___________________**** >> >> Mobile Reply**** >> >> >> On Jul 9, 2012, at 7:07 PM, BookBag wrote:**** >> >> You know if I told my boss that all fax machines were phased out . I >> wouldn't have a job. He's gonna say my job is to make our equipment work >> not force them to switch to different methods**** >> >> On Jul 9, 2012 2:18 PM, "Chris Ferreira" wrote:**** >> >> I think it would be wonderful if all VoIP admins and providers >> collectively told the people needing fax service that it has been >> completely phased out. >> >> I think it would be great if stores stopped selling fax machines and >> all remaining fax machines were recycled. >> >> The only reason faxing still exists is because of the few people that >> refuse to let it die. >> >> >> >> >> ___________________ >> Mobile Reply >> >> On Jul 9, 2012, at 12:58 PM, Gabriel Gunderson wrote: >> >> > On Tue, Jun 19, 2012 at 11:33 AM, Sean Devoy >> wrote: >> >> Has anyone investigated the new FaxxBochs offering? It appears to >> handle the >> >> analog fax machine locally, then send the fax as a data TCP connection >> to >> >> the server (as a TIFF or JPG I would guess). >> > >> > Given how it works, they should have named it FauxBochs. >> > >> > >> > Gabe >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org**** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120710/c62c5aae/attachment-0001.html From moises.silva at gmail.com Tue Jul 10 22:09:25 2012 From: moises.silva at gmail.com (Moises Silva) Date: Tue, 10 Jul 2012 14:09:25 -0400 Subject: [Freeswitch-users] Codec Negotiation In-Reply-To: References: Message-ID: On Tue, Jul 10, 2012 at 11:51 AM, Stephen Wilde wrote: > Is this message flow valid? > > FS ---------------- MyProvider > > ---> INVITE (w/SDP G729, G711a, G711u) > > <--- 100 Trying > > <--- 183 Session Progress (w/SDP G729, G711a, G711u) > > <--- 200 OK (w/SDP G729, G711a, G711u) > > > What is strange to me is that my provider send back a 183 Session Progress > with all codec list of Freeswitch INVITE (I expect only the choosen codec). > > It's this message flow correct? > > Although strange, I think per spec is correct. http://www.ietf.org/rfc/rfc3264.txt Section 7, mentions multiple answers are valid to the offer and it MAY send media in any of the accepted streams and MUST send media using one of the accepted formats and it SHOULD use the first one listed in the answer. Anyone feel free to correct me if my understanding is wrong, as I just started going thru the SIP related specs myself to troubleshoot and support this kind of scenarios. *Moises Silva **Manager, Software Engineering*** msilva at sangoma.com Sangoma Technologies 100 Renfrew Drive, Suite 100, Markham, ON L3R 9R6 Canada t. +1 800 388 2475 (N. America) t. +1 905 474 1990 x128 f. +1 905 474 9223 ** Products | Solutions | Events | Contact | Wiki | Facebook | Twitter`| | YouTube -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120710/aad28239/attachment.html From msc at freeswitch.org Tue Jul 10 23:15:49 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Jul 2012 12:15:49 -0700 Subject: [Freeswitch-users] It's That Time Again: Buy Dinner For The FreeSWITCH Developers Message-ID: Greetings all! It has been quite some time since we all got together and pitched in to buy dinner for the FreeSWITCH development team. A well-fed dev team means more features and fewer bugs, so let's all head over to the main FreeSWITCH pageand hit the donate button. Be sure to click the "special instructions" link in Paypal and mention that your donation is for feeding the developers. Feel free to be creative. :) If you prefer to use a payment method other than FreeSWITCH then please contact Brian West directly. Thanks again for being such a great community and support us over the years! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120710/f14a6288/attachment.html From aksrini at hotmail.com Tue Jul 10 23:19:23 2012 From: aksrini at hotmail.com (Srini K) Date: Tue, 10 Jul 2012 12:19:23 -0700 Subject: [Freeswitch-users] Sending INVITE without sdp Message-ID: Hi,FreeSWITCH can receive INVITE with no sdp by configuring in sip profile.Is there any way to send Invite out from FreeSWITCH without sdp? ThanksSrini -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120710/52712770/attachment.html From acrow at integrafin.co.uk Wed Jul 11 00:20:31 2012 From: acrow at integrafin.co.uk (Alex Crow) Date: Tue, 10 Jul 2012 21:20:31 +0100 Subject: [Freeswitch-users] FaxxBochs hackers? In-Reply-To: References: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com> <6432809444531730803@unknownmsgid> <8251983215579532490@unknownmsgid> <63B00DD1DA6A364E9F64A3A0BD2FE7B604511892@BLUPRD0711MB413.namprd07.prod.outlook.com> <-9115573625816947171@unknownmsgid> <63B00DD1DA6A364E9F64A3A0BD2FE7B6045119B9@BLUPRD0711MB413.namprd07.prod.outlook.com> Message-ID: <4FFC8E8F.6070004@integrafin.co.uk> All. Still a lot of fax requirements in the finance world here - some brokers still require it. T.38 makes things easier without the admin having to worry about legal issues due to the "realtime" nature of the transmission. Yes, it is silly, as anyone can fiddle with the time in their receiving device, anyone as a sender can change their T30 ID, but somehow it is quite fun seeing old fax machines talking to each other, over the internet, with no idea there isn't a real analog PSTN line involved anywhere! In an ideal world everyone would know how to verify, eg, a PGP signed document sent via email/sftp etc but I can't see the adoption reaching critical mass for an audience like ours in my lifetime. Too many proprietary "easy-to-use" "secure email" services that just fragment the market, ignore established standards and detract from everyone adopting a common mechanism. And the stuff with RFCs: S/MIME anyone? SSMTP? Feels like p**sing in the wind when I talk to non-techies about those, let alone PKI and trust networks. Only huge corps and govermnent ever use such things and even then it's a nightmare to enrol in their schemes due to the huge mass of forms you have to fill in! Cheers Alex On 10/07/12 17:50, Andrew Cassidy wrote: > Yes, faxing is great fun! But it's true, if you're working for someone > you can recommend better ways, but if a client says 'I want fax' and > you say 'no', kiss that sale goodbye. I'd rather support faxing (which > freeswitch makes pretty simple) than lose out on all the money! > > On 10 July 2012 12:12, Brian Foster > wrote: > > Besides, fax is fun, right? > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > > On Jul 10, 2012 12:43 AM, "Michael Giagnocavo" > wrote: > > Sure, and I wholeheartedly agree. But none of these arguments > are compelling to users, and making them just means you'll > lose a sale. It's unlikely a large fax customer is going to > say "Wow, you know, I now realize faxes are just ancient > relics, and we're going to ditch them." By not enabling faxes, > you only end up harming the customer. > > -Michael > > *From:*freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On > Behalf Of *Chris Ferreira > *Sent:* Monday, July 09, 2012 8:31 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] FaxxBochs hackers? > > There are more modern ways of legally transmitting documents. > Signed and all. I have several clients that are practicing > lawyers in multiple states and my own brother has a law firm. > > These are simply excuses. My point is that there ARE better > ways. We just keep enabling a poor, old, and broken technology. > > ___________________ > > Mobile Reply > > > On Jul 9, 2012, at 10:09 PM, Michael Giagnocavo > > wrote: > > Faxes carry legal standing that email does not. So, you > can want a better solution, but the reality is that fax > won't disappear soon. In fact, as I understand, part of > T.38 was driven by legal requirements. > > -Michael > > *From:*freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] > *On Behalf Of *Chris Ferreira > *Sent:* Monday, July 09, 2012 6:13 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] FaxxBochs hackers? > > I disagree. The job of any telecom or IT professional is > to also educate and circulate out antiquated equipment. > > I don't know why we still accept the use of faxes when > things like typewriters have pretty much been eliminated. > > ___________________ > > Mobile Reply > > > On Jul 9, 2012, at 7:07 PM, BookBag > wrote: > > You know if I told my boss that all fax machines were > phased out . I wouldn't have a job. He's gonna say my > job is to make our equipment work not force them to > switch to different methods > > On Jul 9, 2012 2:18 PM, "Chris Ferreira" > > wrote: > > I think it would be wonderful if all VoIP admins and > providers > collectively told the people needing fax service that > it has been > completely phased out. > > I think it would be great if stores stopped selling > fax machines and > all remaining fax machines were recycled. > > The only reason faxing still exists is because of the > few people that > refuse to let it die. > > > > > ___________________ > Mobile Reply > > On Jul 9, 2012, at 12:58 PM, Gabriel Gunderson > > wrote: > > > On Tue, Jun 19, 2012 at 11:33 AM, Sean Devoy > > > wrote: > >> Has anyone investigated the new FaxxBochs offering? > It appears to handle the > >> analog fax machine locally, then send the fax as a > data TCP connection to > >> the server (as a TIFF or JPG I would guess). > > > > Given how it works, they should have named it FauxBochs. > > > > > > Gabe > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E > *andrew at cassidywebservices.co.uk > *W > *www.cassidywebservices.co.uk > > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120710/28e951db/attachment-0001.html From andrew at cassidywebservices.co.uk Wed Jul 11 00:52:18 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Tue, 10 Jul 2012 21:52:18 +0100 Subject: [Freeswitch-users] It's That Time Again: Buy Dinner For The FreeSWITCH Developers In-Reply-To: References: Message-ID: Id' definitely prefer to use a payment method other than FreeSWITCH ;) On 10 July 2012 20:15, Michael Collins wrote: > Greetings all! > > It has been quite some time since we all got together and pitched in to > buy dinner for the FreeSWITCH development team. A well-fed dev team means > more features and fewer bugs, so let's all head over to the main FreeSWITCH > page and hit the donate button. Be > sure to click the "special instructions" link in Paypal and mention that > your donation is for feeding the developers. Feel free to be creative. :) > If you prefer to use a payment method other than FreeSWITCH then please > contact Brian West directly. > > Thanks again for being such a great community and support us over the > years! > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120710/8317cd4e/attachment.html From msc at freeswitch.org Wed Jul 11 01:12:05 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Jul 2012 14:12:05 -0700 Subject: [Freeswitch-users] It's That Time Again: Buy Dinner For The FreeSWITCH Developers In-Reply-To: References: Message-ID: Doh! On Tue, Jul 10, 2012 at 1:52 PM, Andrew Cassidy < andrew at cassidywebservices.co.uk> wrote: > Id' definitely prefer to use a payment method other than FreeSWITCH ;) > > On 10 July 2012 20:15, Michael Collins wrote: > >> Greetings all! >> >> It has been quite some time since we all got together and pitched in to >> buy dinner for the FreeSWITCH development team. A well-fed dev team means >> more features and fewer bugs, so let's all head over to the main FreeSWITCH >> page and hit the donate button. Be >> sure to click the "special instructions" link in Paypal and mention that >> your donation is for feeding the developers. Feel free to be creative. :) >> If you prefer to use a payment method other than FreeSWITCH then please >> contact Brian West directly. >> >> Thanks again for being such a great community and support us over the >> years! >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > *Andrew Cassidy BSc (Hons) MBCS SSCA* > Managing Director > > > *T *03300 100 960 *F > *03300 100 961 > *E *andrew at cassidywebservices.co.uk > *W *www.cassidywebservices.co.uk > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120710/bf6dde0d/attachment.html From rdmitry0911 at gmail.com Wed Jul 11 01:44:08 2012 From: rdmitry0911 at gmail.com (Dmitry R) Date: Wed, 11 Jul 2012 01:44:08 +0400 Subject: [Freeswitch-users] Problem with setting up a variable in section of a gateway Message-ID: Hi guys, I'm trying to set a channel variable effective_caller_id_number in gateway's section and it seems not working. However, if I set it up with application set command in dialplan it works fine. What is the difference between these two ways? My gateway section looks like this: and working dialplan strings look like this: Thank you, Dmitry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120711/29376d5f/attachment.html From msc at freeswitch.org Wed Jul 11 03:04:53 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Jul 2012 16:04:53 -0700 Subject: [Freeswitch-users] FreeSWITCH Conference Call Tomorrow Message-ID: Hello Community! Just a friendly reminder: our conference call is scheduled for 1PM Eastern/10AM Pacific time tomorrow. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_07_11 Dave Kompel (drk__) will be discussing Windows firewall security. Rumor has it that he's got an interesting example of how to do server push notification for those of you who want to see an example of how to do server updates to a browser. On the ClueCon front we just wanted to remind those of you who will be in Chicago for ClueCon that Anthony will be enjoying birthday number 40-something, so plan accordingly. :) Thanks! -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120710/a2e8ac80/attachment-0001.html From curriegrad2004 at gmail.com Wed Jul 11 03:12:14 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 10 Jul 2012 16:12:14 -0700 Subject: [Freeswitch-users] It's That Time Again: Buy Dinner For The FreeSWITCH Developers In-Reply-To: References: Message-ID: I thought I was on something but that definitely looks like a typo now On 7/10/12, Michael Collins wrote: > Doh! > > On Tue, Jul 10, 2012 at 1:52 PM, Andrew Cassidy < > andrew at cassidywebservices.co.uk> wrote: > >> Id' definitely prefer to use a payment method other than FreeSWITCH ;) >> >> On 10 July 2012 20:15, Michael Collins wrote: >> >>> Greetings all! >>> >>> It has been quite some time since we all got together and pitched in to >>> buy dinner for the FreeSWITCH development team. A well-fed dev team >>> means >>> more features and fewer bugs, so let's all head over to the main >>> FreeSWITCH >>> page and hit the donate button. Be >>> sure to click the "special instructions" link in Paypal and mention that >>> your donation is for feeding the developers. Feel free to be creative. >>> :) >>> If you prefer to use a payment method other than FreeSWITCH then please >>> contact Brian West directly. >>> >>> Thanks again for being such a great community and support us over the >>> years! >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> *Andrew Cassidy BSc (Hons) MBCS SSCA* >> Managing Director >> >> >> *T *03300 100 960 >> *F >> *03300 100 961 >> *E *andrew at cassidywebservices.co.uk >> *W *www.cassidywebservices.co.uk >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > From terry at digital-outpost.com Wed Jul 11 03:19:18 2012 From: terry at digital-outpost.com (Terry Barnum) Date: Tue, 10 Jul 2012 16:19:18 -0700 Subject: [Freeswitch-users] new freeswitch/fusionpbx user Message-ID: I'm starting to find my way around freeswitch via fusionpbx and have it running with a free DID, two X-Lite softphones and 3CXPhone on an iPhone. After creating a recording, an IVR menu and an inbound route to the IVR, a test call coming in on the DID complains it can't find the menu: 2012-07-08 11:28:36.451506 [NOTICE] switch_channel.c:926 New Channel sofia/external/@www.xxx.yyy.zzz [82327424-822d-44a4-9aa5-c20fdce2f356] 2012-07-08 11:28:36.451506 [INFO] mod_dialplan_xml.c:485 Processing <>-> in context public 2012-07-08 11:28:36.471501 [NOTICE] switch_ivr.c:1729 Transfer sofia/external/@www.xxx.yyy.zzz to XML[5002 at default] 2012-07-08 11:28:36.471501 [INFO] mod_dialplan_xml.c:485 Processing <>->5002 in context default 2012-07-08 11:28:36.551572 [NOTICE] mod_dptools.c:1148 Channel [sofia/external/@www.xxx.yyy.zzz] has been answered 2012-07-08 11:28:37.571514 [ERR] mod_dptools.c:1775 Unable to find menu 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:249 sofia/external/@www.xxx.yyy.zzz has executed the last dialplan instruction, hanging up. 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:251 Hangup sofia/external/@www.xxx.yyy.zzz [CS_EXECUTE] [NORMAL_CLEARING] 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1446 Session 1 (sofia/external/@www.xxx.yyy.zzz) Ended 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1448 Close Channel sofia/external/@www.xxx.yyy.zzz [CS_DESTROY] I can see that the newly created choosevm ivr menu exists: $ sudo ls -l *.xml /usr/local/freeswitch/conf/ivr_menus/ -rw-rw---- 1 _www _www 2878 Jun 20 00:08 demo_ivr.xml -rw-r--r-- 1 _www _www 632 Jul 8 11:08 v_choosevm.xml $ sudo cat /usr/local/freeswitch/conf/ivr_menus/v_choosevm.xml $ sudo cat /usr/local/freeswitch/conf/dialplan/default/333_v_choosevm.xml Is there something else necessary for the ivr menu to be found? I realize that I'm abstracted from freeswitch a bit by using fusionpbx as a front-end, but is this still the right place to ask? Thanks, -Terry From gcd at i.ph Wed Jul 11 03:54:42 2012 From: gcd at i.ph (Nandy Dagondon) Date: Wed, 11 Jul 2012 07:54:42 +0800 Subject: [Freeswitch-users] new freeswitch/fusionpbx user In-Reply-To: References: Message-ID: hi terry, FusionPBX has a forum http://www.fusionpbx.com/messageboard/rsslist.php. it's the best place to ask for help. /nandy On Wed, Jul 11, 2012 at 7:19 AM, Terry Barnum wrote: > I'm starting to find my way around freeswitch via fusionpbx and have it > running with a free DID, two X-Lite softphones and 3CXPhone on an iPhone. > After creating a recording, an IVR menu and an inbound route to the IVR, a > test call coming in on the DID complains it can't find the menu: > > 2012-07-08 11:28:36.451506 [NOTICE] switch_channel.c:926 New Channel > sofia/external/@www.xxx.yyy.zzz > [82327424-822d-44a4-9aa5-c20fdce2f356] > 2012-07-08 11:28:36.451506 [INFO] mod_dialplan_xml.c:485 Processing > <>-> in context public > 2012-07-08 11:28:36.471501 [NOTICE] switch_ivr.c:1729 Transfer > sofia/external/@www.xxx.yyy.zzz to XML[5002 at default] > 2012-07-08 11:28:36.471501 [INFO] mod_dialplan_xml.c:485 Processing > <>->5002 in context default > 2012-07-08 11:28:36.551572 [NOTICE] mod_dptools.c:1148 Channel > [sofia/external/@www.xxx.yyy.zzz] has been answered > 2012-07-08 11:28:37.571514 [ERR] mod_dptools.c:1775 Unable to find menu > 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:249 > sofia/external/@www.xxx.yyy.zzz has executed the last dialplan > instruction, hanging up. > 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:251 Hangup > sofia/external/@www.xxx.yyy.zzz [CS_EXECUTE] [NORMAL_CLEARING] > 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1446 Session 1 > (sofia/external/@www.xxx.yyy.zzz) Ended > 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1448 Close > Channel sofia/external/@www.xxx.yyy.zzz [CS_DESTROY] > > I can see that the newly created choosevm ivr menu exists: > > $ sudo ls -l *.xml /usr/local/freeswitch/conf/ivr_menus/ > -rw-rw---- 1 _www _www 2878 Jun 20 00:08 demo_ivr.xml > -rw-r--r-- 1 _www _www 632 Jul 8 11:08 v_choosevm.xml > > $ sudo cat /usr/local/freeswitch/conf/ivr_menus/v_choosevm.xml > > > greet-long="/usr/local/freeswitch/recordings/recording1.wav" > greet-short="/usr/local/freeswitch/recordings/" > invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" > exit-sound="" > confirm-macro="" > confirm-key="" > tts-engine="flite" > tts-voice="rms" > confirm-attempts="3" > timeout="3000" > inter-digit-timeout="2000" > max-failures="3" > max-timeouts="3" > digit-len="5"> > > > > > > $ sudo cat /usr/local/freeswitch/conf/dialplan/default/333_v_choosevm.xml > > > > > > > > > > Is there something else necessary for the ivr menu to be found? I realize > that I'm abstracted from freeswitch a bit by using fusionpbx as a > front-end, but is this still the right place to ask? > > Thanks, > -Terry > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120711/4d8b1400/attachment.html From bdfoster at endigotech.com Wed Jul 11 03:57:06 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 10 Jul 2012 19:57:06 -0400 Subject: [Freeswitch-users] new freeswitch/fusionpbx user In-Reply-To: References: Message-ID: Run: $ fs_cli -x 'reloadxml' Try the call again, see if that helps. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 10, 2012 7:27 PM, "Terry Barnum" wrote: > I'm starting to find my way around freeswitch via fusionpbx and have it > running with a free DID, two X-Lite softphones and 3CXPhone on an iPhone. > After creating a recording, an IVR menu and an inbound route to the IVR, a > test call coming in on the DID complains it can't find the menu: > > 2012-07-08 11:28:36.451506 [NOTICE] switch_channel.c:926 New Channel > sofia/external/@www.xxx.yyy.zzz > [82327424-822d-44a4-9aa5-c20fdce2f356] > 2012-07-08 11:28:36.451506 [INFO] mod_dialplan_xml.c:485 Processing > <>-> in context public > 2012-07-08 11:28:36.471501 [NOTICE] switch_ivr.c:1729 Transfer > sofia/external/@www.xxx.yyy.zzz to XML[5002 at default] > 2012-07-08 11:28:36.471501 [INFO] mod_dialplan_xml.c:485 Processing > <>->5002 in context default > 2012-07-08 11:28:36.551572 [NOTICE] mod_dptools.c:1148 Channel > [sofia/external/@www.xxx.yyy.zzz] has been answered > 2012-07-08 11:28:37.571514 [ERR] mod_dptools.c:1775 Unable to find menu > 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:249 > sofia/external/@www.xxx.yyy.zzz has executed the last dialplan > instruction, hanging up. > 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:251 Hangup > sofia/external/@www.xxx.yyy.zzz [CS_EXECUTE] [NORMAL_CLEARING] > 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1446 Session 1 > (sofia/external/@www.xxx.yyy.zzz) Ended > 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1448 Close > Channel sofia/external/@www.xxx.yyy.zzz [CS_DESTROY] > > I can see that the newly created choosevm ivr menu exists: > > $ sudo ls -l *.xml /usr/local/freeswitch/conf/ivr_menus/ > -rw-rw---- 1 _www _www 2878 Jun 20 00:08 demo_ivr.xml > -rw-r--r-- 1 _www _www 632 Jul 8 11:08 v_choosevm.xml > > $ sudo cat /usr/local/freeswitch/conf/ivr_menus/v_choosevm.xml > > > greet-long="/usr/local/freeswitch/recordings/recording1.wav" > greet-short="/usr/local/freeswitch/recordings/" > invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" > exit-sound="" > confirm-macro="" > confirm-key="" > tts-engine="flite" > tts-voice="rms" > confirm-attempts="3" > timeout="3000" > inter-digit-timeout="2000" > max-failures="3" > max-timeouts="3" > digit-len="5"> > > > > > > $ sudo cat /usr/local/freeswitch/conf/dialplan/default/333_v_choosevm.xml > > > > > > > > > > Is there something else necessary for the ivr menu to be found? I realize > that I'm abstracted from freeswitch a bit by using fusionpbx as a > front-end, but is this still the right place to ask? > > Thanks, > -Terry > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120710/d147cc1a/attachment-0001.html From dujinfang at gmail.com Wed Jul 11 04:03:54 2012 From: dujinfang at gmail.com (Seven Du) Date: Wed, 11 Jul 2012 08:03:54 +0800 Subject: [Freeswitch-users] It's That Time Again: Buy Dinner For The FreeSWITCH Developers In-Reply-To: References: Message-ID: <307BB5D5CEAD45A5893614BAC9A1C368@gmail.com> mod_dinner mod_donate mod_pay mod_wallet On Wednesday, July 11, 2012 at 7:12 AM, curriegrad2004 wrote: > I thought I was on something but that definitely looks like a typo now > > On 7/10/12, Michael Collins wrote: > > Doh! > > > > On Tue, Jul 10, 2012 at 1:52 PM, Andrew Cassidy < > > andrew at cassidywebservices.co.uk (mailto:andrew at cassidywebservices.co.uk)> wrote: > > > > > Id' definitely prefer to use a payment method other than FreeSWITCH ;) > > > > > > On 10 July 2012 20:15, Michael Collins wrote: > > > > > > > Greetings all! > > > > > > > > It has been quite some time since we all got together and pitched in to > > > > buy dinner for the FreeSWITCH development team. A well-fed dev team > > > > means > > > > more features and fewer bugs, so let's all head over to the main > > > > FreeSWITCH > > > > page and hit the donate button. Be > > > > sure to click the "special instructions" link in Paypal and mention that > > > > your donation is for feeding the developers. Feel free to be creative. > > > > :) > > > > If you prefer to use a payment method other than FreeSWITCH then please > > > > contact Brian West directly. > > > > > > > > Thanks again for being such a great community and support us over the > > > > years! > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > -- > > > *Andrew Cassidy BSc (Hons) MBCS SSCA* > > > Managing Director > > > > > > > > > *T *03300 100 960 > > > *F > > > *03300 100 961 > > > *E *andrew at cassidywebservices.co.uk (mailto:andrew at cassidywebservices.co.uk) > > > *W *www.cassidywebservices.co.uk (http://www.cassidywebservices.co.uk) > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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/20120711/1710fcd3/attachment.html From terry at digital-outpost.com Wed Jul 11 04:36:55 2012 From: terry at digital-outpost.com (Terry Barnum) Date: Tue, 10 Jul 2012 17:36:55 -0700 Subject: [Freeswitch-users] new freeswitch/fusionpbx user In-Reply-To: References: Message-ID: <8C819ECF-2E38-4139-9D27-300A7F087801@digital-outpost.com> Thank you for the suggestion Brian. I have been running freeswitch in a foreground terminal window. I shutdown and restarted it with no change in behavior. -Terry On Jul 10, 2012, at 4:57 PM, Brian Foster wrote: > Run: $ fs_cli -x 'reloadxml' > > Try the call again, see if that helps. > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > > On Jul 10, 2012 7:27 PM, "Terry Barnum" wrote: > I'm starting to find my way around freeswitch via fusionpbx and have it running with a free DID, two X-Lite softphones and 3CXPhone on an iPhone. After creating a recording, an IVR menu and an inbound route to the IVR, a test call coming in on the DID complains it can't find the menu: > > 2012-07-08 11:28:36.451506 [NOTICE] switch_channel.c:926 New Channel sofia/external/@www.xxx.yyy.zzz [82327424-822d-44a4-9aa5-c20fdce2f356] > 2012-07-08 11:28:36.451506 [INFO] mod_dialplan_xml.c:485 Processing <>-> in context public > 2012-07-08 11:28:36.471501 [NOTICE] switch_ivr.c:1729 Transfer sofia/external/@www.xxx.yyy.zzz to XML[5002 at default] > 2012-07-08 11:28:36.471501 [INFO] mod_dialplan_xml.c:485 Processing <>->5002 in context default > 2012-07-08 11:28:36.551572 [NOTICE] mod_dptools.c:1148 Channel [sofia/external/@www.xxx.yyy.zzz] has been answered > 2012-07-08 11:28:37.571514 [ERR] mod_dptools.c:1775 Unable to find menu > 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:249 sofia/external/@www.xxx.yyy.zzz has executed the last dialplan instruction, hanging up. > 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:251 Hangup sofia/external/@www.xxx.yyy.zzz [CS_EXECUTE] [NORMAL_CLEARING] > 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1446 Session 1 (sofia/external/@www.xxx.yyy.zzz) Ended > 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1448 Close Channel sofia/external/@www.xxx.yyy.zzz [CS_DESTROY] > > I can see that the newly created choosevm ivr menu exists: > > $ sudo ls -l *.xml /usr/local/freeswitch/conf/ivr_menus/ > -rw-rw---- 1 _www _www 2878 Jun 20 00:08 demo_ivr.xml > -rw-r--r-- 1 _www _www 632 Jul 8 11:08 v_choosevm.xml > > $ sudo cat /usr/local/freeswitch/conf/ivr_menus/v_choosevm.xml > > greet-long="/usr/local/freeswitch/recordings/recording1.wav" > greet-short="/usr/local/freeswitch/recordings/" > invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" > exit-sound="" > confirm-macro="" > confirm-key="" > tts-engine="flite" > tts-voice="rms" > confirm-attempts="3" > timeout="3000" > inter-digit-timeout="2000" > max-failures="3" > max-timeouts="3" > digit-len="5"> > > > > > > $ sudo cat /usr/local/freeswitch/conf/dialplan/default/333_v_choosevm.xml > > > > > > > > > > Is there something else necessary for the ivr menu to be found? I realize that I'm abstracted from freeswitch a bit by using fusionpbx as a front-end, but is this still the right place to ask? > > Thanks, > -Terry > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From terry at digital-outpost.com Wed Jul 11 04:39:14 2012 From: terry at digital-outpost.com (Terry Barnum) Date: Tue, 10 Jul 2012 17:39:14 -0700 Subject: [Freeswitch-users] new freeswitch/fusionpbx user In-Reply-To: References: Message-ID: <8361818B-6F13-4D15-BA98-C313DB72E546@digital-outpost.com> Thanks Nandy. I have posted there and the yahoo groups mailing list but it's very quiet. -Terry On Jul 10, 2012, at 4:54 PM, Nandy Dagondon wrote: > hi terry, > > FusionPBX has a forum http://www.fusionpbx.com/messageboard/rsslist.php. it's the best place to ask for help. > > /nandy > > On Wed, Jul 11, 2012 at 7:19 AM, Terry Barnum wrote: > I'm starting to find my way around freeswitch via fusionpbx and have it running with a free DID, two X-Lite softphones and 3CXPhone on an iPhone. After creating a recording, an IVR menu and an inbound route to the IVR, a test call coming in on the DID complains it can't find the menu: > > 2012-07-08 11:28:36.451506 [NOTICE] switch_channel.c:926 New Channel sofia/external/@www.xxx.yyy.zzz [82327424-822d-44a4-9aa5-c20fdce2f356] > 2012-07-08 11:28:36.451506 [INFO] mod_dialplan_xml.c:485 Processing <>-> in context public > 2012-07-08 11:28:36.471501 [NOTICE] switch_ivr.c:1729 Transfer sofia/external/@www.xxx.yyy.zzz to XML[5002 at default] > 2012-07-08 11:28:36.471501 [INFO] mod_dialplan_xml.c:485 Processing <>->5002 in context default > 2012-07-08 11:28:36.551572 [NOTICE] mod_dptools.c:1148 Channel [sofia/external/@www.xxx.yyy.zzz] has been answered > 2012-07-08 11:28:37.571514 [ERR] mod_dptools.c:1775 Unable to find menu > 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:249 sofia/external/@www.xxx.yyy.zzz has executed the last dialplan instruction, hanging up. > 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:251 Hangup sofia/external/@www.xxx.yyy.zzz [CS_EXECUTE] [NORMAL_CLEARING] > 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1446 Session 1 (sofia/external/@www.xxx.yyy.zzz) Ended > 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1448 Close Channel sofia/external/@www.xxx.yyy.zzz [CS_DESTROY] > > I can see that the newly created choosevm ivr menu exists: > > $ sudo ls -l *.xml /usr/local/freeswitch/conf/ivr_menus/ > -rw-rw---- 1 _www _www 2878 Jun 20 00:08 demo_ivr.xml > -rw-r--r-- 1 _www _www 632 Jul 8 11:08 v_choosevm.xml > > $ sudo cat /usr/local/freeswitch/conf/ivr_menus/v_choosevm.xml > > greet-long="/usr/local/freeswitch/recordings/recording1.wav" > greet-short="/usr/local/freeswitch/recordings/" > invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" > exit-sound="" > confirm-macro="" > confirm-key="" > tts-engine="flite" > tts-voice="rms" > confirm-attempts="3" > timeout="3000" > inter-digit-timeout="2000" > max-failures="3" > max-timeouts="3" > digit-len="5"> > > > > > > $ sudo cat /usr/local/freeswitch/conf/dialplan/default/333_v_choosevm.xml > > > > > > > > > > Is there something else necessary for the ivr menu to be found? I realize that I'm abstracted from freeswitch a bit by using fusionpbx as a front-end, but is this still the right place to ask? > > Thanks, > -Terry > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Jul 11 06:53:07 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 10 Jul 2012 19:53:07 -0700 Subject: [Freeswitch-users] new freeswitch/fusionpbx user In-Reply-To: <8361818B-6F13-4D15-BA98-C313DB72E546@digital-outpost.com> References: <8361818B-6F13-4D15-BA98-C313DB72E546@digital-outpost.com> Message-ID: Don't forget about the IRC channel: #fusionpbx on irc.freenode.net. On Tue, Jul 10, 2012 at 5:39 PM, Terry Barnum wrote: > Thanks Nandy. I have posted there and the yahoo groups mailing list but > it's very quiet. > > -Terry > > On Jul 10, 2012, at 4:54 PM, Nandy Dagondon wrote: > > > hi terry, > > > > FusionPBX has a forum http://www.fusionpbx.com/messageboard/rsslist.php. > it's the best place to ask for help. > > > > /nandy > > > > On Wed, Jul 11, 2012 at 7:19 AM, Terry Barnum > wrote: > > I'm starting to find my way around freeswitch via fusionpbx and have it > running with a free DID, two X-Lite softphones and 3CXPhone on an iPhone. > After creating a recording, an IVR menu and an inbound route to the IVR, a > test call coming in on the DID complains it can't find the menu: > > > > 2012-07-08 11:28:36.451506 [NOTICE] switch_channel.c:926 New Channel > sofia/external/@www.xxx.yyy.zzz > [82327424-822d-44a4-9aa5-c20fdce2f356] > > 2012-07-08 11:28:36.451506 [INFO] mod_dialplan_xml.c:485 Processing > <>-> in context public > > 2012-07-08 11:28:36.471501 [NOTICE] switch_ivr.c:1729 Transfer > sofia/external/@www.xxx.yyy.zzz to XML[5002 at default] > > 2012-07-08 11:28:36.471501 [INFO] mod_dialplan_xml.c:485 Processing > <>->5002 in context default > > 2012-07-08 11:28:36.551572 [NOTICE] mod_dptools.c:1148 Channel > [sofia/external/@www.xxx.yyy.zzz] has been answered > > 2012-07-08 11:28:37.571514 [ERR] mod_dptools.c:1775 Unable to find menu > > 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:249 > sofia/external/@www.xxx.yyy.zzz has executed the last dialplan > instruction, hanging up. > > 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:251 > Hangup sofia/external/@www.xxx.yyy.zzz [CS_EXECUTE] > [NORMAL_CLEARING] > > 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1446 Session 1 > (sofia/external/@www.xxx.yyy.zzz) Ended > > 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1448 Close > Channel sofia/external/@www.xxx.yyy.zzz [CS_DESTROY] > > > > I can see that the newly created choosevm ivr menu exists: > > > > $ sudo ls -l *.xml /usr/local/freeswitch/conf/ivr_menus/ > > -rw-rw---- 1 _www _www 2878 Jun 20 00:08 demo_ivr.xml > > -rw-r--r-- 1 _www _www 632 Jul 8 11:08 v_choosevm.xml > > > > $ sudo cat /usr/local/freeswitch/conf/ivr_menus/v_choosevm.xml > > > > > > greet-long="/usr/local/freeswitch/recordings/recording1.wav" > > greet-short="/usr/local/freeswitch/recordings/" > > invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" > > exit-sound="" > > confirm-macro="" > > confirm-key="" > > tts-engine="flite" > > tts-voice="rms" > > confirm-attempts="3" > > timeout="3000" > > inter-digit-timeout="2000" > > max-failures="3" > > max-timeouts="3" > > digit-len="5"> > > > > > > > > > > > > $ sudo cat /usr/local/freeswitch/conf/dialplan/default/333_v_choosevm.xml > > > > > > > > > > > > > > > > > > > > Is there something else necessary for the ivr menu to be found? I > realize that I'm abstracted from freeswitch a bit by using fusionpbx as a > front-end, but is this still the right place to ask? > > > > Thanks, > > -Terry > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120710/56b1beaa/attachment-0001.html From bdfoster at endigotech.com Wed Jul 11 07:14:24 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 10 Jul 2012 23:14:24 -0400 Subject: [Freeswitch-users] new freeswitch/fusionpbx user In-Reply-To: References: <8361818B-6F13-4D15-BA98-C313DB72E546@digital-outpost.com> Message-ID: On Jul 10, 2012 11:00 PM, "Michael Collins" wrote: > > Don't forget about the IRC channel: #fusionpbx on irc.freenode.net. > > > On Tue, Jul 10, 2012 at 5:39 PM, Terry Barnum wrote: >> >> Thanks Nandy. I have posted there and the yahoo groups mailing list but it's very quiet. >> Agreed, this is where the action happens in regards to fusionpbx. >> >> On Jul 10, 2012, at 4:54 PM, Nandy Dagondon wrote: >> >> > hi terry, >> > >> > FusionPBX has a forum http://www.fusionpbx.com/messageboard/rsslist.php. it's the best place to ask for help. >> > >> > /nandy >> > >> > On Wed, Jul 11, 2012 at 7:19 AM, Terry Barnum < terry at digital-outpost.com> wrote: >> > I'm starting to find my way around freeswitch via fusionpbx and have it running with a free DID, two X-Lite softphones and 3CXPhone on an iPhone. After creating a recording, an IVR menu and an inbound route to the IVR, a test call coming in on the DID complains it can't find the menu: >> > >> > 2012-07-08 11:28:36.451506 [NOTICE] switch_channel.c:926 New Channel sofia/external/@www.xxx.yyy.zzz [82327424-822d-44a4-9aa5-c20fdce2f356] >> > 2012-07-08 11:28:36.451506 [INFO] mod_dialplan_xml.c:485 Processing <>-> in context public >> > 2012-07-08 11:28:36.471501 [NOTICE] switch_ivr.c:1729 Transfer sofia/external/@www.xxx.yyy.zzz to XML[5002 at default] >> > 2012-07-08 11:28:36.471501 [INFO] mod_dialplan_xml.c:485 Processing <>->5002 in context default >> > 2012-07-08 11:28:36.551572 [NOTICE] mod_dptools.c:1148 Channel [sofia/external/@www.xxx.yyy.zzz] has been answered >> > 2012-07-08 11:28:37.571514 [ERR] mod_dptools.c:1775 Unable to find menu >> > 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:249 sofia/external/@www.xxx.yyy.zzz has executed the last dialplan instruction, hanging up. >> > 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:251 Hangup sofia/external/@www.xxx.yyy.zzz [CS_EXECUTE] [NORMAL_CLEARING] >> > 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1446 Session 1 (sofia/external/@www.xxx.yyy.zzz) Ended >> > 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1448 Close Channel sofia/external/@www.xxx.yyy.zzz [CS_DESTROY] >> > >> > I can see that the newly created choosevm ivr menu exists: >> > >> > $ sudo ls -l *.xml /usr/local/freeswitch/conf/ivr_menus/ >> > -rw-rw---- 1 _www _www 2878 Jun 20 00:08 demo_ivr.xml >> > -rw-r--r-- 1 _www _www 632 Jul 8 11:08 v_choosevm.xml >> > >> > $ sudo cat /usr/local/freeswitch/conf/ivr_menus/v_choosevm.xml >> > >> > > > greet-long="/usr/local/freeswitch/recordings/recording1.wav" >> > greet-short="/usr/local/freeswitch/recordings/" >> > invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" >> > exit-sound="" >> > confirm-macro="" >> > confirm-key="" >> > tts-engine="flite" >> > tts-voice="rms" >> > confirm-attempts="3" >> > timeout="3000" >> > inter-digit-timeout="2000" >> > max-failures="3" >> > max-timeouts="3" >> > digit-len="5"> >> > >> > >> > >> > >> > >> > $ sudo cat /usr/local/freeswitch/conf/dialplan/default/333_v_choosevm.xml >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > Is there something else necessary for the ivr menu to be found? I realize that I'm abstracted from freeswitch a bit by using fusionpbx as a front-end, but is this still the right place to ask? >> > >> > Thanks, >> > -Terry >> > >> > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120710/dd040ae4/attachment.html From jaasmailing at gmail.com Wed Jul 11 11:07:34 2012 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Wed, 11 Jul 2012 09:07:34 +0200 Subject: [Freeswitch-users] new freeswitch/fusionpbx user In-Reply-To: References: Message-ID: <4FFD2636.3040906@gmail.com> Hi Terry, what about calling directly ivr extension 5002? I'm using ivr in fusionpbx without problems... Il 11/07/12 01.19, Terry Barnum ha scritto: > I'm starting to find my way around freeswitch via fusionpbx and have it running with a free DID, two X-Lite softphones and 3CXPhone on an iPhone. After creating a recording, an IVR menu and an inbound route to the IVR, a test call coming in on the DID complains it can't find the menu: > > 2012-07-08 11:28:36.451506 [NOTICE] switch_channel.c:926 New Channel sofia/external/@www.xxx.yyy.zzz [82327424-822d-44a4-9aa5-c20fdce2f356] > 2012-07-08 11:28:36.451506 [INFO] mod_dialplan_xml.c:485 Processing <>-> in context public > 2012-07-08 11:28:36.471501 [NOTICE] switch_ivr.c:1729 Transfer sofia/extemernal/@www.xxx.yyy.zzz to XML[5002 at default] > 2012-07-08 11:28:36.471501 [INFO] mod_dialplan_xml.c:485 Processing <>->5002 in context default > 2012-07-08 11:28:36.551572 [NOTICE] mod_dptools.c:1148 Channel [sofia/external/@www.xxx.yyy.zzz] has been answered > 2012-07-08 11:28:37.571514 [ERR] mod_dptools.c:1775 Unable to find menu > 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:249 sofia/external/@www.xxx.yyy.zzz has executed the last dialplan instruction, hanging up. > 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:251 Hangup sofia/external/@www.xxx.yyy.zzz [CS_EXECUTE] [NORMAL_CLEARING] > 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1446 Session 1 (sofia/external/@www.xxx.yyy.zzz) Ended > 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1448 Close Channel sofia/external/@www.xxx.yyy.zzz [CS_DESTROY] > > I can see that the newly created choosevm ivr menu exists: > > $ sudo ls -l *.xml /usr/local/freeswitch/conf/ivr_menus/ > -rw-rw---- 1 _www _www 2878 Jun 20 00:08 demo_ivr.xml > -rw-r--r-- 1 _www _www 632 Jul 8 11:08 v_choosevm.xml > > $ sudo cat /usr/local/freeswitch/conf/ivr_menus/v_choosevm.xml > > greet-long="/usr/local/freeswitch/recordings/recording1.wav" > greet-short="/usr/local/freeswitch/recordings/" > invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" > exit-sound="" > confirm-macro="" > confirm-key="" > tts-engine="flite" > tts-voice="rms" > confirm-attempts="3" > timeout="3000" > inter-digit-timeout="2000" > max-failures="3" > max-timeouts="3" > digit-len="5"> > > > > > > $ sudo cat /usr/local/freeswitch/conf/dialplan/default/333_v_choosevm.xml > > > > > > > > > > Is there something else necessary for the ivr menu to be found? I realize that I'm abstracted from freeswitch a bit by using fusionpbx as a front-end, but is this still the right place to ask? > > Thanks, > -Terry > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From vipkilla at gmail.com Wed Jul 11 16:59:43 2012 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 11 Jul 2012 08:59:43 -0400 Subject: [Freeswitch-users] difference between bridging with 'user/XXXX@YYYY.com' and ${sofia_contact(XXXX@YYYY.com} Message-ID: what is the difference between bridging to a UA using 'user/XXX at xxx.com' and ${sofia_contact(XXX at xxx.com} ? i noticed things like presence (esp. with SLA) work better with user/XXX at xxx.com I also noticed the pickup dialplan app (,pickup/XXX at xxx.com) doesnt work with the 'user/XXX at xxx.com' it only seems to work with sofia_contact bridge Can anybody test or verify this? Thanks. RJ From vbvbrj at gmail.com Wed Jul 11 11:02:39 2012 From: vbvbrj at gmail.com (Vbvbrj) Date: Wed, 11 Jul 2012 10:02:39 +0300 Subject: [Freeswitch-users] Redirecting folders. Message-ID: <4FFD250F.2040001@gmail.com> Hello. I've compiled and build freeswitch with --prefix=/opt/freeswitch . After making and installing, I want to rename or move the whole installed folder with binaries and other stuff. I could use LD_LIBRARY_PATH and command line options to specify new folders, but there still remains other folders, like storage, recordings, grammar. Sounds dir I specify in vars.xml. How to redirect the rest of folders? On start freeswitch creates a folder in /opt/freeswitch with grammar, recordings, sounds and storage subfolders. This is not needed. It will be preferable if during compiling, to compile the freeswitch with prefix in such a way, that any needed file will be found relatively to bin file 'freeswitch' on starting it, ie, freeswitch will look for it default configs not using $prefix/conf, but using ../conf. So is for other directories. Thank you. A second problem. I use a dedicated server with just linux installed for freeswitch. And freeswitch is the only major service which is installed. Calling 5000 immediately after starting freeswitch, the IVR sound is good. But as time passes, sound get creeped, losed, sometimes it is not understandable what it is saying. With this default configurations if I register with 1000 and call 1001, I get into voicemail, as 1001 is not registered. When spelling the number 1001 the sounds seems to distort, like slowing down the speed of saying. What this can be a cause? From steveu at coppice.org Wed Jul 11 17:53:47 2012 From: steveu at coppice.org (Steve Underwood) Date: Wed, 11 Jul 2012 21:53:47 +0800 Subject: [Freeswitch-users] FaxxBochs hackers? In-Reply-To: References: <030f01cd4e41$9ffde5f0$dff9b1d0$@bizfocused.com> Message-ID: <4FFD856B.80404@coppice.org> As you move from the single port to the 4 port version of FaxxBochs, its only an extra $20 or so per port. The base unit is also pretty cheap compared to any of the little Asterisk boxes around. They just don't need the complexity of Asterisk. They just need a processor that can run 4 instances of V.17 (I don't think they do V.34) and a fairly simple TCP stack to interact with the server. Steve On 07/11/2012 12:48 AM, shouldbe q931 wrote: > On Tue, Jun 19, 2012 at 6:33 PM, Sean Devoy wrote: >> Has anyone investigated the new FaxxBochs offering? It appears to handle the >> analog fax machine locally, then send the fax as a data TCP connection to >> the server (as a TIFF or JPG I would guess). It totally removes the VOIP >> problem with fax machines and is a clever solution. $30 a month unlimited >> in and out is reasonable, but I would rather have that income myself. >> >> >> >> It would be great if I could resell it and make a cut, but they require 75 >> device sales the first year. >> >> >> >> If someone knows anything about ?reconfiguring? one of these, I would love >> to read about it. >> >> >> >> Sean >> >> >> _________________________________________________________________________ > > looking at their other offerings, I would guess that its a four port > asterisk box with IAX modem connecting to hylafax to "receive" the > inbound fax and then send it to their "datacenter" hyalafax server(s) > for onward transmission using the "proxy" feature in jobcontrol, > inbound would work in the same way but the other way around. The AES > 256 encryption could well just be SSL... > > not a bad solution if you must have fax and don't have PSTN (or ISDN) > connectivity, but as its store and forward, I'm not sure how it plays > out legally... > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Wed Jul 11 18:27:37 2012 From: kawarod at laposte.net (kawarod) Date: Wed, 11 Jul 2012 18:27:37 +0400 Subject: [Freeswitch-users] bind_digit_action and SIP INFO Message-ID: <4FFD8D59.6040506@laposte.net> Hi List, I'm using bypass_media on FS to avoid as much as possible RTP going on low bandwidth link. But I'm interested in using bind_digit_action for handling 3 way call/call hold on SIP UA with bad implementation of this feature. The only way I see for this to work is to have bind_digit_action listening to SIP INFO, so that I can put FS back in the media path to handle advanced features. If somebody has an idea for this. regards, Rod. From mario_fs at mgtech.com Wed Jul 11 18:58:49 2012 From: mario_fs at mgtech.com (Mario G) Date: Wed, 11 Jul 2012 07:58:49 -0700 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? Message-ID: I recently spent a lot of time learning how to keep an SSD fast and increase its life. It took a ton of research and I combined a lot of info from several sources to install the SSD. Turns out, just installing an SSD and using it as an HDD without doing anything else is not a good idea. Although the topic is not specific to FreeSwitch I am wondering how many out there would like me to put together a FS wiki page fr tuning an SSD. It's a lot of work so I won't do it unless there are enough requests or the developers want it. Mario G From sdevoy at bizfocused.com Wed Jul 11 19:34:42 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 11 Jul 2012 11:34:42 -0400 Subject: [Freeswitch-users] FreeSwitch Failover Question Message-ID: <025801cd5f7a$b19fa150$14dee3f0$@bizfocused.com> Hi All, I have a question concerning FS failover for high availability. My SIP provider offers the ability to route to a second server if the primary is down. That is simple enough, but what about the phones? The desktop devices are not registered with the back FS server. The only idea I have so far would be to change the DNS addresses to the backup upon failure, but that can take a long time to propagate to the phones. How do other people handle this issue? Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120711/375c9813/attachment.html From msc at freeswitch.org Wed Jul 11 19:45:29 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Jul 2012 08:45:29 -0700 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: References: Message-ID: This is an interesting topic, but I don't know how wide the audience is. Perhaps you could tell us what led you to use an SSD in your environment? -MC On Wed, Jul 11, 2012 at 7:58 AM, Mario G wrote: > I recently spent a lot of time learning how to keep an SSD fast and > increase its life. It took a ton of research and I combined a lot of info > from several sources to install the SSD. Turns out, just installing an SSD > and using it as an HDD without doing anything else is not a good idea. > Although the topic is not specific to FreeSwitch I am wondering how many > out there would like me to put together a FS wiki page fr tuning an SSD. > It's a lot of work so I won't do it unless there are enough requests or the > developers want it. > Mario G > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120711/407abb98/attachment.html From tnelson at rockbochs.com Wed Jul 11 19:55:47 2012 From: tnelson at rockbochs.com (Tim Nelson) Date: Wed, 11 Jul 2012 10:55:47 -0500 (CDT) Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: Message-ID: <1815260.1124954.1342022147321.JavaMail.root@rockbochs.com> ----- Original Message ----- > I recently spent a lot of time learning how to keep an SSD fast and > increase its life. It took a ton of research and I combined a lot of > info from several sources to install the SSD. Turns out, just > installing an SSD and using it as an HDD without doing anything else > is not a good idea. Although the topic is not specific to FreeSwitch > I am wondering how many out there would like me to put together a FS > wiki page fr tuning an SSD. It's a lot of work so I won't do it > unless there are enough requests or the developers want it. Yes, please do! I'm certain there are plenty of people running SSDs in general, and with Freeswitch, so that would seem to make it relevant. :) --Tim From chris at gonumina.com Wed Jul 11 20:15:56 2012 From: chris at gonumina.com (Chris Ferreira) Date: Wed, 11 Jul 2012 12:15:56 -0400 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: References: Message-ID: <-5316884151656024069@unknownmsgid> I would most definitely be interested in this. I have been using SSD's for a number of different applications and never had a problem. Am I missing something? And you talk about increasing its life? Most of the SSDs I have used are rated at about 4 Million Hours which is like 456 Years (which I don't believe). Very interested to hear what you have to say. Thanks, -Chris ___________________ Mobile Reply On Jul 11, 2012, at 11:39 AM, Mario G wrote: > I recently spent a lot of time learning how to keep an SSD fast and increase its life. It took a ton of research and I combined a lot of info from several sources to install the SSD. Turns out, just installing an SSD and using it as an HDD without doing anything else is not a good idea. Although the topic is not specific to FreeSwitch I am wondering how many out there would like me to put together a FS wiki page fr tuning an SSD. It's a lot of work so I won't do it unless there are enough requests or the developers want it. > Mario G > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mgg at giagnocavo.net Wed Jul 11 20:22:17 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Wed, 11 Jul 2012 16:22:17 +0000 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: References: Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B6045126AD@BLUPRD0711MB413.namprd07.prod.outlook.com> I'd be interested, assuming it was a business class SSD from a good vendor with decent firmware and not just a junk SSD burning out. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, July 11, 2012 9:45 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? This is an interesting topic, but I don't know how wide the audience is. Perhaps you could tell us what led you to use an SSD in your environment? -MC On Wed, Jul 11, 2012 at 7:58 AM, Mario G > wrote: I recently spent a lot of time learning how to keep an SSD fast and increase its life. It took a ton of research and I combined a lot of info from several sources to install the SSD. Turns out, just installing an SSD and using it as an HDD without doing anything else is not a good idea. Although the topic is not specific to FreeSwitch I am wondering how many out there would like me to put together a FS wiki page fr tuning an SSD. It's a lot of work so I won't do it unless there are enough requests or the developers want it. Mario G -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120711/ff188be3/attachment-0001.html From msc at freeswitch.org Wed Jul 11 20:27:28 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Jul 2012 09:27:28 -0700 Subject: [Freeswitch-users] FreeSWITCH Conf Call In 30 Minutes! Message-ID: Talk to you soon! http://wiki.freeswitch.org/wiki/FS_weekly_2012_07_11 -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120711/a9b2df59/attachment.html From lloyd.aloysius at gmail.com Wed Jul 11 20:27:46 2012 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Wed, 11 Jul 2012 12:27:46 -0400 Subject: [Freeswitch-users] FreeSwitch Failover Question In-Reply-To: <025801cd5f7a$b19fa150$14dee3f0$@bizfocused.com> References: <025801cd5f7a$b19fa150$14dee3f0$@bizfocused.com> Message-ID: Use the DNS SRV records for the Desk phones. Phones automatically register to the back server. Look into the DNS Srv records. You can specify the priority. Use the DNS SRV for the phone registration. I use Polycom and Cisco. Aastra phones... have bug in the firmware using DNS Srv records. http://wiki.freeswitch.org/wiki/Enterprise_deployment_DNS_SRV On Wed, Jul 11, 2012 at 11:34 AM, Sean Devoy wrote: > Hi All,**** > > ** ** > > I have a question concerning FS failover for high availability. My SIP > provider offers the ability to route to a second server if the primary is > down. That is simple enough, but what about the phones? The desktop > devices are not registered with the back FS server.**** > > ** ** > > The only idea I have so far would be to change the DNS addresses to the > backup upon failure, but that can take a long time to propagate to the > phones. **** > > ** ** > > How do other people handle this issue?**** > > ** ** > > Thanks,**** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120711/8404e195/attachment.html From ifoundthetao at gmail.com Wed Jul 11 20:37:39 2012 From: ifoundthetao at gmail.com (Timothy Bolton) Date: Wed, 11 Jul 2012 11:37:39 -0500 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: References: Message-ID: <4FFDABD3.5020705@gmail.com> I'm interested. And while it might not be specific to FreeSWITCH, I think it's worth it to have it on there when looking at performance and how to tweak configs. Also, I heard that SSDs and RAID don't get along so well, is this true? 'We who cut mere stones must always be envisioning cathedrals.' Quarry Worker's Creed On 7/11/2012 9:58 AM, Mario G wrote: > I recently spent a lot of time learning how to keep an SSD fast and increase its life. It took a ton of research and I combined a lot of info from several sources to install the SSD. Turns out, just installing an SSD and using it as an HDD without doing anything else is not a good idea. Although the topic is not specific to FreeSwitch I am wondering how many out there would like me to put together a FS wiki page fr tuning an SSD. It's a lot of work so I won't do it unless there are enough requests or the developers want it. > Mario G > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From avi at avimarcus.net Wed Jul 11 20:43:51 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 11 Jul 2012 19:43:51 +0300 Subject: [Freeswitch-users] FreeSwitch Failover Question In-Reply-To: References: <025801cd5f7a$b19fa150$14dee3f0$@bizfocused.com> Message-ID: Also, you can share the registration data between servers so that it know where to send incoming calls to. Might have NAT issues with that, though, if you send via a different IP. I'm not so fluent in NAT.. -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120711/67f923ac/attachment.html From sdevoy at bizfocused.com Wed Jul 11 21:06:13 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 11 Jul 2012 13:06:13 -0400 Subject: [Freeswitch-users] FreeSwitch Failover Question In-Reply-To: References: <025801cd5f7a$b19fa150$14dee3f0$@bizfocused.com> Message-ID: <030601cd5f87$7a644df0$6f2ce9d0$@bizfocused.com> I knew there was a way. Thanks Lloyd. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lloyd Aloysius Sent: Wednesday, July 11, 2012 12:28 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSwitch Failover Question Use the DNS SRV records for the Desk phones. Phones automatically register to the back server. Look into the DNS Srv records. You can specify the priority. Use the DNS SRV for the phone registration. I use Polycom and Cisco. Aastra phones... have bug in the firmware using DNS Srv records. http://wiki.freeswitch.org/wiki/Enterprise_deployment_DNS_SRV On Wed, Jul 11, 2012 at 11:34 AM, Sean Devoy wrote: Hi All, I have a question concerning FS failover for high availability. My SIP provider offers the ability to route to a second server if the primary is down. That is simple enough, but what about the phones? The desktop devices are not registered with the back FS server. The only idea I have so far would be to change the DNS addresses to the backup upon failure, but that can take a long time to propagate to the phones. How do other people handle this issue? Thanks, Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120711/4ac76c6e/attachment.html From fs-list at communicatefreely.net Wed Jul 11 21:36:18 2012 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Wed, 11 Jul 2012 13:36:18 -0400 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: References: Message-ID: <4FFDB992.1050505@communicatefreely.net> I'm interested in Mario's response, but in our environment, we had to go with SSDs to keep up with the database activity that happens when you run the core DB in sql and have 400 phones all with a list of subscriptions, and every call using the limit DB backend twice. I have only been using them for about a year, so if there is something I should do to extend their life, I'm quite interested! -Tim Michael Collins wrote: > This is an interesting topic, but I don't know how wide the audience > is. Perhaps you could tell us what led you to use an SSD in your > environment? > -MC > > On Wed, Jul 11, 2012 at 7:58 AM, Mario G > wrote: > > I recently spent a lot of time learning how to keep an SSD fast > and increase its life. It took a ton of research and I combined a > lot of info from several sources to install the SSD. Turns out, > just installing an SSD and using it as an HDD without doing > anything else is not a good idea. Although the topic is not > specific to FreeSwitch I am wondering how many out there would > like me to put together a FS wiki page fr tuning an SSD. It's a > lot of work so I won't do it unless there are enough requests or > the developers want it. > Mario G > > ------------------------------------------------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From cmrienzo at gmail.com Wed Jul 11 21:45:58 2012 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Wed, 11 Jul 2012 13:45:58 -0400 Subject: [Freeswitch-users] Redirecting folders. In-Reply-To: <4FFD250F.2040001@gmail.com> References: <4FFD250F.2040001@gmail.com> Message-ID: Latest code added command line options for temp, grammar, sounds, recordings, voicemail. Chris On Wed, Jul 11, 2012 at 3:02 AM, Vbvbrj wrote: > Hello. > > I've compiled and build freeswitch with --prefix=/opt/freeswitch . After > making and installing, I want to rename or move the whole installed > folder with binaries and other stuff. I could use LD_LIBRARY_PATH and > command line options to specify new folders, but there still remains > other folders, like storage, recordings, grammar. Sounds dir I specify > in vars.xml. > > How to redirect the rest of folders? On start freeswitch creates a > folder in /opt/freeswitch with grammar, recordings, sounds and storage > subfolders. This is not needed. It will be preferable if during > compiling, to compile the freeswitch with prefix in such a way, that any > needed file will be found relatively to bin file 'freeswitch' on > starting it, ie, freeswitch will look for it default configs not using > $prefix/conf, but using ../conf. So is for other directories. > > Thank you. > > A second problem. I use a dedicated server with just linux installed for > freeswitch. And freeswitch is the only major service which is installed. > Calling 5000 immediately after starting freeswitch, the IVR sound is > good. But as time passes, sound get creeped, losed, sometimes it is not > understandable what it is saying. With this default configurations if I > register with 1000 and call 1001, I get into voicemail, as 1001 is not > registered. When spelling the number 1001 the sounds seems to distort, > like slowing down the speed of saying. What this can be a cause? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120711/3393bd72/attachment.html From avi at avimarcus.net Wed Jul 11 22:09:47 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 11 Jul 2012 21:09:47 +0300 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: <4FFDB992.1050505@communicatefreely.net> References: <4FFDB992.1050505@communicatefreely.net> Message-ID: On Wed, Jul 11, 2012 at 8:36 PM, Tim St. Pierre < fs-list at communicatefreely.net> wrote: > I'm interested in Mario's response, but in our environment, we had to go > with SSDs to keep up with the database activity that happens when you > run the core DB in sql and have 400 phones all with a list > of subscriptions, Tim: What about ram-caching the sqlite DB? and every call using the limit DB backend twice. > Do you need this distribution ready / persistent? You could similarly use hash to store this in ram. -Avi > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120711/dd9a16d6/attachment.html From torstein.knutsen at gmail.com Wed Jul 11 22:38:45 2012 From: torstein.knutsen at gmail.com (Torstein Knutsen) Date: Wed, 11 Jul 2012 20:38:45 +0200 Subject: [Freeswitch-users] Help with cidlookup Message-ID: Hi I have cidlookup partly working. Im using a norwegian service, which returns a whole lot more than just the number. Anybody here have some Ideas on how I could proceed to map "Pizza & Kina Expressen" to the calling_id_name ? Thank you! Torstein snipplet *** : freeswitch at --hidden-ip--@internal> cidlookup 22222222 { "title" : "Gule Sider firma API", "query" : " http://api.eniro.com/cs/search/basic?country=no&search_word=22222222&to_list=1&version=1.1.3&from_list=1", "totalHits" : 1 , "totalCount" : 1 , "startIndex" : 1, "itemsPerPage" : 1, "adverts" : [ { "eniroId" : "P10000836357" , "companyInfo" : { "companyName" : "Pizza & Kina Expressen" , "orgNumber" : null , "companyText" : null }, "address" : { "streetName" : "Vitaminveien 11 B" , "postCode" : "0485" , "postArea" : "Oslo" , "postBox" : null }, "location" : { "coordinates" : [ { "longitude" : 10.7725744944096 , "latitude" : 59.9471107465998 }, { "use" : "route", "longitude" : 10.7725744944096 , "latitude" : 59.9471107465998 } ] }, "phoneNumbers" : [ { "type" : "std" , "phoneNumber" : "22 22 22 22" , "label" : null } ], "companyReviews" : " http://www.anbefalt.no/omtale/0003292695/22222222" , "homepage" : " http://api.eniro.com/proxy/homepage/uANwPf5aVK3QsMrfdwYjz8Olp1PSJ6L1-mCsL3_LC0d9Yem9mkC025y22P034JmT" , "infoPageLink" : " http://www.gulesider.no/firma/pizza-kina-expressen:p10000836357?search_word=22222222" } ] } freeswitch at --hidden-ip--@internal> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120711/5adcf179/attachment.html From terry at digital-outpost.com Wed Jul 11 23:08:02 2012 From: terry at digital-outpost.com (Terry Barnum) Date: Wed, 11 Jul 2012 12:08:02 -0700 Subject: [Freeswitch-users] new freeswitch/fusionpbx user In-Reply-To: <4FFD2636.3040906@gmail.com> References: <4FFD2636.3040906@gmail.com> Message-ID: Thank you Carlo. I'm sure it's something I've got wrong somewhere. I'll try 5002 directly when I'm at the machine tonight. -Terry On Jul 11, 2012, at 12:07 AM, Carlo Dimaggio wrote: > Hi Terry, > what about calling directly ivr extension 5002? > > I'm using ivr in fusionpbx without problems... > > > > Il 11/07/12 01.19, Terry Barnum ha scritto: >> I'm starting to find my way around freeswitch via fusionpbx and have it running with a free DID, two X-Lite softphones and 3CXPhone on an iPhone. After creating a recording, an IVR menu and an inbound route to the IVR, a test call coming in on the DID complains it can't find the menu: >> >> 2012-07-08 11:28:36.451506 [NOTICE] switch_channel.c:926 New Channel sofia/external/@www.xxx.yyy.zzz [82327424-822d-44a4-9aa5-c20fdce2f356] >> 2012-07-08 11:28:36.451506 [INFO] mod_dialplan_xml.c:485 Processing <>-> in context public >> 2012-07-08 11:28:36.471501 [NOTICE] switch_ivr.c:1729 Transfer sofia/extemernal/@www.xxx.yyy.zzz to XML[5002 at default] >> 2012-07-08 11:28:36.471501 [INFO] mod_dialplan_xml.c:485 Processing <>->5002 in context default >> 2012-07-08 11:28:36.551572 [NOTICE] mod_dptools.c:1148 Channel [sofia/external/@www.xxx.yyy.zzz] has been answered >> 2012-07-08 11:28:37.571514 [ERR] mod_dptools.c:1775 Unable to find menu >> 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:249 sofia/external/@www.xxx.yyy.zzz has executed the last dialplan instruction, hanging up. >> 2012-07-08 11:28:37.571514 [NOTICE] switch_core_state_machine.c:251 Hangup sofia/external/@www.xxx.yyy.zzz [CS_EXECUTE] [NORMAL_CLEARING] >> 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1446 Session 1 (sofia/external/@www.xxx.yyy.zzz) Ended >> 2012-07-08 11:28:37.591511 [NOTICE] switch_core_session.c:1448 Close Channel sofia/external/@www.xxx.yyy.zzz [CS_DESTROY] >> >> I can see that the newly created choosevm ivr menu exists: >> >> $ sudo ls -l *.xml /usr/local/freeswitch/conf/ivr_menus/ >> -rw-rw---- 1 _www _www 2878 Jun 20 00:08 demo_ivr.xml >> -rw-r--r-- 1 _www _www 632 Jul 8 11:08 v_choosevm.xml >> >> $ sudo cat /usr/local/freeswitch/conf/ivr_menus/v_choosevm.xml >> >> > greet-long="/usr/local/freeswitch/recordings/recording1.wav" >> greet-short="/usr/local/freeswitch/recordings/" >> invalid-sound="ivr/ivr-that_was_an_invalid_entry.wav" >> exit-sound="" >> confirm-macro="" >> confirm-key="" >> tts-engine="flite" >> tts-voice="rms" >> confirm-attempts="3" >> timeout="3000" >> inter-digit-timeout="2000" >> max-failures="3" >> max-timeouts="3" >> digit-len="5"> >> >> >> >> >> >> $ sudo cat /usr/local/freeswitch/conf/dialplan/default/333_v_choosevm.xml >> >> >> >> >> >> >> >> >> >> Is there something else necessary for the ivr menu to be found? I realize that I'm abstracted from freeswitch a bit by using fusionpbx as a front-end, but is this still the right place to ask? >> >> Thanks, >> -Terry >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From curriegrad2004 at gmail.com Wed Jul 11 23:18:29 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 11 Jul 2012 12:18:29 -0700 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: References: <4FFDB992.1050505@communicatefreely.net> Message-ID: This is sure an interesting topic to talk about. This is definitely going to be a great article on the wiki. On Wed, Jul 11, 2012 at 11:09 AM, Avi Marcus wrote: > > On Wed, Jul 11, 2012 at 8:36 PM, Tim St. Pierre > wrote: >> >> I'm interested in Mario's response, but in our environment, we had to go >> with SSDs to keep up with the database activity that happens when you >> run the core DB in sql and have 400 phones all with a list of >> subscriptions, > > Tim: What about ram-caching the sqlite DB? > >> and every call using the limit DB backend twice. > > Do you need this distribution ready / persistent? You could similarly use > hash to store this in ram. > > -Avi >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From sdevoy at bizfocused.com Wed Jul 11 23:28:02 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 11 Jul 2012 15:28:02 -0400 Subject: [Freeswitch-users] FreeSwitch Failover Question In-Reply-To: References: <025801cd5f7a$b19fa150$14dee3f0$@bizfocused.com> Message-ID: <040201cd5f9b$49f7d240$dde776c0$@bizfocused.com> If I am reading correctly, this does not lend itself to multi-tenant very nicely. For example, some of my users connect to fs_aa.mydomain.com and others to fs_bb.mydomain.com. Then I can reuse the extension numbers between the 2 clients. However, SRV records, like MX records, are by domain name (ie just mydomain.com). It seems to me that the SRV records for my 2 sample "domains" would evaluate down to the same name and I would lose my critical multi-tenant information. I am hoping that I am misinterpreting the usage of SRV records. If I use SRV and attempt to connect to fs_aa.mydomain.com and the SRV records sort out to sip1.mydomain.com, what does FS see as the "domain name"? If it is still fs_aa.mydomain.com, I LOVE THIS. If it is sip1.mydomain.com, it is going to be cumbersome at the least Thanks for your patients and help, Sean. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Lloyd Aloysius Sent: Wednesday, July 11, 2012 12:28 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSwitch Failover Question Use the DNS SRV records for the Desk phones. Phones automatically register to the back server. Look into the DNS Srv records. You can specify the priority. Use the DNS SRV for the phone registration. I use Polycom and Cisco. Aastra phones... have bug in the firmware using DNS Srv records. http://wiki.freeswitch.org/wiki/Enterprise_deployment_DNS_SRV On Wed, Jul 11, 2012 at 11:34 AM, Sean Devoy wrote: Hi All, I have a question concerning FS failover for high availability. My SIP provider offers the ability to route to a second server if the primary is down. That is simple enough, but what about the phones? The desktop devices are not registered with the back FS server. The only idea I have so far would be to change the DNS addresses to the backup upon failure, but that can take a long time to propagate to the phones. How do other people handle this issue? Thanks, Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120711/dfd05f2b/attachment-0001.html From gmaruzz at gmail.com Wed Jul 11 23:44:00 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 11 Jul 2012 21:44:00 +0200 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: References: <4FFDB992.1050505@communicatefreely.net> Message-ID: I'm interested, and btw SSD are very relevant both for embedded and for HA implementations. -giovanni On Wed, Jul 11, 2012 at 8:09 PM, Avi Marcus wrote: > > On Wed, Jul 11, 2012 at 8:36 PM, Tim St. Pierre < > fs-list at communicatefreely.net> wrote: > >> I'm interested in Mario's response, but in our environment, we had to go >> with SSDs to keep up with the database activity that happens when you >> run the core DB in sql and have 400 phones all with a list >> of subscriptions, > > Tim: What about ram-caching the sqlite DB? > > and every call using the limit DB backend twice. >> > Do you need this distribution ready / persistent? You could similarly use > hash to store this in ram. > > -Avi > >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120711/254936ee/attachment.html From spencer at 5ninesolutions.com Wed Jul 11 23:48:09 2012 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Wed, 11 Jul 2012 12:48:09 -0700 Subject: [Freeswitch-users] Help with cidlookup In-Reply-To: References: Message-ID: <04918207-4B96-4836-B2A6-ABDE7B2912D7@5ninesolutions.com> Hi Torstein, I ran into a similar problem with a US provider. I used a python script that wraps the api command and then parses the json response and sets effective_caller_id_name. in python e.g. import json from freeswitch import API def cidlookup(number): api = API() number = str(number) cmd = 'cidlookup %s' % (number) return api.executeString(cmd).strip() def handler(session, args): caller_num = session.getVariable("caller_id_number") cnam_data = json.loads(cidlookup(caller_num)) # get caller name from cnam_data here.. session.setVariable("effective_caller_id_name", caller_name) return (You could also do the api lookup in you script and bypass mod_cidlookup) I hope that helps, Spencer On Jul 11, 2012, at 11:38 AM, Torstein Knutsen wrote: > Hi > > I have cidlookup partly working. > Im using a norwegian service, which returns a whole lot more than just the number. > > Anybody here have some Ideas on how I could proceed to map "Pizza & Kina Expressen" to the calling_id_name ? > > Thank you! > Torstein > > snipplet *** > : > > freeswitch at --hidden-ip--@internal> cidlookup 22222222 > > > > > { "title" : "Gule Sider firma API", "query" : "http://api.eniro.com/cs/search/basic?country=no&search_word=22222222&to_list=1&version=1.1.3&from_list=1", "totalHits" : 1 , "totalCount" : 1 , "startIndex" : 1, "itemsPerPage" : 1, "adverts" : [ { "eniroId" : "P10000836357" , "companyInfo" : { "companyName" : "Pizza & Kina Expressen" , "orgNumber" : null , "companyText" : null }, "address" : { "streetName" : "Vitaminveien 11 B" , "postCode" : "0485" , "postArea" : "Oslo" , "postBox" : null }, "location" : { "coordinates" : [ { "longitude" : 10.7725744944096 , "latitude" : 59.9471107465998 }, { "use" : "route", "longitude" : 10.7725744944096 , "latitude" : 59.9471107465998 } ] }, "phoneNumbers" : [ { "type" : "std" , "phoneNumber" : "22 22 22 22" , "label" : null } ], "companyReviews" : "http://www.anbefalt.no/omtale/0003292695/22222222" , "homepage" : "http://api.eniro.com/proxy/homepage/uANwPf5aVK3QsMrfdwYjz8Olp1PSJ6L1-mCsL3_LC0d9Yem9mkC025y22P034JmT" , "infoPageLink" : "http://www.gulesider.no/firma/pizza-kina-expressen:p10000836357?search_word=22222222" } ] } > freeswitch at --hidden-ip--@internal> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120711/97bcc38f/attachment.html From toddb at toddbailey.net Wed Jul 11 23:51:46 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Wed, 11 Jul 2012 12:51:46 -0700 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: <-5316884151656024069@unknownmsgid> References: <-5316884151656024069@unknownmsgid> Message-ID: <1342036306.3587.7.camel@mythtv> Or 4 million micro seconds what ever comes first. I tried using a 90 gb ssd on linux and being curious, ran phoronix-test-suite against the drive and with 15 minutes the drive was non operational, rma'd the device and tried again, same results. returned for refund and never looked back... SSD devices may be fast but in a server environment reliability is key. currently running raid 5 across the enterprise. On Wed, 2012-07-11 at 12:15 -0400, Chris Ferreira wrote: > I would most definitely be interested in this. I have been using SSD's > for a number of different applications and never had a problem. Am I > missing something? And you talk about increasing its life? Most of the > SSDs I have used are rated at about 4 Million Hours which is like 456 > Years (which I don't believe). > > > > Very interested to hear what you have to say. > > > > Thanks, > > -Chris > > ___________________ > Mobile Reply > > On Jul 11, 2012, at 11:39 AM, Mario G wrote: > > > I recently spent a lot of time learning how to keep an SSD fast and increase its life. It took a ton of research and I combined a lot of info from several sources to install the SSD. Turns out, just installing an SSD and using it as an HDD without doing anything else is not a good idea. Although the topic is not specific to FreeSwitch I am wondering how many out there would like me to put together a FS wiki page fr tuning an SSD. It's a lot of work so I won't do it unless there are enough requests or the developers want it. > > Mario G > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From asaad2 at gmail.com Thu Jul 12 00:22:00 2012 From: asaad2 at gmail.com (BookBag) Date: Wed, 11 Jul 2012 16:22:00 -0400 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: References: Message-ID: I never refuse knowledge. Put it up fs wiki On Jul 11, 2012 12:32 PM, "Michael Collins" wrote: > This is an interesting topic, but I don't know how wide the audience is. > Perhaps you could tell us what led you to use an SSD in your environment? > -MC > > On Wed, Jul 11, 2012 at 7:58 AM, Mario G wrote: > >> I recently spent a lot of time learning how to keep an SSD fast and >> increase its life. It took a ton of research and I combined a lot of info >> from several sources to install the SSD. Turns out, just installing an SSD >> and using it as an HDD without doing anything else is not a good idea. >> Although the topic is not specific to FreeSwitch I am wondering how many >> out there would like me to put together a FS wiki page fr tuning an SSD. >> It's a lot of work so I won't do it unless there are enough requests or the >> developers want it. >> Mario G >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120711/a8a7a724/attachment.html From nbhatti at gmail.com Thu Jul 12 00:22:12 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 11 Jul 2012 23:22:12 +0300 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: <1342036306.3587.7.camel@mythtv> References: <-5316884151656024069@unknownmsgid> <1342036306.3587.7.camel@mythtv> Message-ID: Depends on what brand the drive is. We are using a lot of SSDs in a huge enterprise environment and never faced issues. On average each SSD is giving us around 2000 IOPS with a combine output of around 32000 IOPS and we will be upgrading to 32 soon. Currently have 16. For smaller scale, I am using one in my Macbook, and never had any issues. Reaches upto 5000 IOPS and data transfer rate of around 200+ MBps. Have been using a quite a few in the servers running for FS. They work very well and never deceived. Surely, the thread is a good one, if the OP posts the draft :) Thanks. On Wed, Jul 11, 2012 at 10:51 PM, Todd Bailey wrote: > > Or 4 million micro seconds what ever comes first. > > I tried using a 90 gb ssd on linux and being curious, ran > phoronix-test-suite against the drive and with 15 minutes the drive was > non operational, rma'd the device and tried again, same results. > > returned for refund and never looked back... > > SSD devices may be fast but in a server environment reliability is key. > currently running raid 5 across the enterprise. > > > On Wed, 2012-07-11 at 12:15 -0400, Chris Ferreira wrote: >> I would most definitely be interested in this. I have been using SSD's >> for a number of different applications and never had a problem. Am I >> missing something? And you talk about increasing its life? Most of the >> SSDs I have used are rated at about 4 Million Hours which is like 456 >> Years (which I don't believe). >> >> >> >> Very interested to hear what you have to say. >> >> >> >> Thanks, >> >> -Chris >> >> ___________________ >> Mobile Reply >> >> On Jul 11, 2012, at 11:39 AM, Mario G wrote: >> >> > I recently spent a lot of time learning how to keep an SSD fast and increase its life. It took a ton of research and I combined a lot of info from several sources to install the SSD. Turns out, just installing an SSD and using it as an HDD without doing anything else is not a good idea. Although the topic is not specific to FreeSwitch I am wondering how many out there would like me to put together a FS wiki page fr tuning an SSD. It's a lot of work so I won't do it unless there are enough requests or the developers want it. >> > Mario G >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bfmtl at hotmail.com Thu Jul 12 00:06:37 2012 From: bfmtl at hotmail.com (BF) Date: Wed, 11 Jul 2012 16:06:37 -0400 Subject: [Freeswitch-users] Record call in LUA Message-ID: Hello, I can't find how to record a call from a LUA script. DEBUG shows no error but file is not created. Snippets of my LUA script and FS console log are provide below. api = freeswitch.API(); api:execute("record_session",call_recording_path..call_id..".mp3"); 2012-07-11 15:59:59.553048 [DEBUG] switch_cpp.cpp:1227 record_session /mydir/fc157da2-cb92-11e1-b8ad-65e540468340.mp3 The "mydir" directory is in 777 privilege access mode. Any thought anyone? Thank you. Bernard From gabe at gundy.org Thu Jul 12 00:28:49 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 11 Jul 2012 14:28:49 -0600 Subject: [Freeswitch-users] FreeSwitch Failover Question In-Reply-To: <040201cd5f9b$49f7d240$dde776c0$@bizfocused.com> References: <025801cd5f7a$b19fa150$14dee3f0$@bizfocused.com> <040201cd5f9b$49f7d240$dde776c0$@bizfocused.com> Message-ID: On Wed, Jul 11, 2012 at 1:28 PM, Sean Devoy wrote: > I am hoping that I am misinterpreting the usage of SRV records. If I use > SRV and attempt to connect to fs_aa.mydomain.com and the SRV records sort > out to sip1.mydomain.com, what does FS see as the ?domain name?? If it is > still fs_aa.mydomain.com, I LOVE THIS. If it is sip1.mydomain.com, it is > going to be cumbersome at the least Turns out, you'll love it. But, you still have issues with tracking the registration on the back-end over several servers. That takes some work still. Great? Yes. Silver bullet? No :) Gabe From msc at freeswitch.org Thu Jul 12 00:54:14 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Jul 2012 13:54:14 -0700 Subject: [Freeswitch-users] Record call in LUA In-Reply-To: References: Message-ID: Is this Lua script called from the dialplan? If so you want to be using the session object and session:recordFile: http://wiki.freeswitch.org/wiki/Lua#session:recordFile If it's being called from outside the dialplan then you'll need to use the uuid_record API: http://wiki.freeswitch.org/wiki/Mod_commands#uuid_record -MC On Wed, Jul 11, 2012 at 1:06 PM, BF wrote: > Hello, > > I can't find how to record a call from a LUA script. DEBUG shows no error > but file is not created. Snippets of my LUA script and FS console log are > provide below. > > api = freeswitch.API(); > api:execute("record_session",call_recording_path..call_id..".mp3"); > > > 2012-07-11 15:59:59.553048 [DEBUG] switch_cpp.cpp:1227 record_session > /mydir/fc157da2-cb92-11e1-b8ad-65e540468340.mp3 > > The "mydir" directory is in 777 privilege access mode. > > Any thought anyone? > > Thank you. > > Bernard > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120711/b7616f00/attachment.html From fluixab at bellsouth.net Thu Jul 12 01:56:11 2012 From: fluixab at bellsouth.net (Bernard Fluixa) Date: Wed, 11 Jul 2012 17:56:11 -0400 Subject: [Freeswitch-users] Record call in LUA In-Reply-To: References: Message-ID: Michael, uuid_record works. Thanks. Bernard On Jul 11, 2012, at 4:54 PM, Michael Collins wrote: > Is this Lua script called from the dialplan? If so you want to be using the session object and session:recordFile: > http://wiki.freeswitch.org/wiki/Lua#session:recordFile > > If it's being called from outside the dialplan then you'll need to use the uuid_record API: > http://wiki.freeswitch.org/wiki/Mod_commands#uuid_record > > -MC > > On Wed, Jul 11, 2012 at 1:06 PM, BF wrote: > Hello, > > I can't find how to record a call from a LUA script. DEBUG shows no error but file is not created. Snippets of my LUA script and FS console log are provide below. > > api = freeswitch.API(); > api:execute("record_session",call_recording_path..call_id..".mp3"); > > > 2012-07-11 15:59:59.553048 [DEBUG] switch_cpp.cpp:1227 record_session /mydir/fc157da2-cb92-11e1-b8ad-65e540468340.mp3 > > The "mydir" directory is in 777 privilege access mode. > > Any thought anyone? > > Thank you. > > Bernard > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120711/969b108c/attachment.html From cjbujold at accra.ca Thu Jul 12 03:00:54 2012 From: cjbujold at accra.ca (Charles Bujold) Date: Wed, 11 Jul 2012 20:00:54 -0300 Subject: [Freeswitch-users] Error in updating Freeswitch Message-ID: <000001cd5fb9$06deddf0$149c99d0$@accra.ca> Ubuntu 10.4lts AMD Platform Tried to update Freeswitch using make current and I get the following error: How can I correct this. Do I need to update to Ubuntu 12.04lts? Compiling /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c... quiet_libtool: compile: gcc -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c -fPIC -DPIC -o .libs/mod_conference.o cc1: warnings being treated as errors /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c: In function 'send_rfc_event': /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c:125 1: error: format not a string literal and no format arguments /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c:125 2: error: format not a string literal and no format arguments /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c:125 5: error: format not a string literal and no format arguments /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c: In function 'call_setup_event_handler': /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c:805 6: error: format not a string literal and no format arguments /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c: In function 'conf_data_event_handler': /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c:809 1: error: format not a string literal and no format arguments make[6]: *** [mod_conference.lo] Error 1 make[6]: Leaving directory `/usr/src/freeswitch/src/mod/applications/mod_conference' make[5]: *** [all] Error 1 make[5]: Leaving directory `/usr/src/freeswitch/src/mod/applications/mod_conference' make[4]: *** [mod_conference-all] Error 1 make[4]: Leaving directory `/usr/src/freeswitch/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/src/freeswitch/src' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2 Thanks cjb -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120711/c5c2c0a9/attachment-0001.html From stkn at freeswitch.org Thu Jul 12 03:46:37 2012 From: stkn at freeswitch.org (Stefan Knoblich) Date: Thu, 12 Jul 2012 01:46:37 +0200 Subject: [Freeswitch-users] Error in updating Freeswitch In-Reply-To: <000001cd5fb9$06deddf0$149c99d0$@accra.ca> References: <000001cd5fb9$06deddf0$149c99d0$@accra.ca> Message-ID: <4FFE105D.6080307@freeswitch.org> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 07/12/12 01:00, Charles Bujold wrote: > /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c: In function ?send_rfc_event?: /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c:1251: error: > format not a string literal and no format arguments /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c:1252: error: format not a string literal and no format arguments > /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c:1255: error: format not a string literal and no format arguments > /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c: In function ?call_setup_event_handler?: /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c:8056: > error: format not a string literal and no format arguments /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c: In function ?conf_data_event_handler?: > /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c:8091: error: format not a string literal and no format arguments That's fixed now, please update. -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.19 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk/+EF0ACgkQjiIIAK4rYUrDggCgrmfq5vcthCyjohqTqW19B2Nz 1csAn0uywNmmluuarIOR8Ky81lUx8ttS =o4GJ -----END PGP SIGNATURE----- From basit.engg at gmail.com Thu Jul 12 04:28:47 2012 From: basit.engg at gmail.com (Abdul Basit) Date: Thu, 12 Jul 2012 05:28:47 +0500 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: References: <-5316884151656024069@unknownmsgid> <1342036306.3587.7.camel@mythtv> Message-ID: i m definitely interested and appreciate you willingness of knowledge sharing. Love hardware capacity, scalability and long life so waiting for the information you collected. -- Regards, Abdul Basit On Thu, Jul 12, 2012 at 1:22 AM, Muhammad Naseer Bhatti wrote: > Depends on what brand the drive is. We are using a lot of SSDs in a > huge enterprise environment and never faced issues. On average each > SSD is giving us around 2000 IOPS with a combine output of around > 32000 IOPS and we will be upgrading to 32 soon. Currently have 16. > > For smaller scale, I am using one in my Macbook, and never had any > issues. Reaches upto 5000 IOPS and data transfer rate of around 200+ > MBps. Have been using a quite a few in the servers running for FS. > They work very well and never deceived. Surely, the thread is a good > one, if the OP posts the draft :) > > Thanks. > > On Wed, Jul 11, 2012 at 10:51 PM, Todd Bailey > wrote: > > > > Or 4 million micro seconds what ever comes first. > > > > I tried using a 90 gb ssd on linux and being curious, ran > > phoronix-test-suite against the drive and with 15 minutes the drive was > > non operational, rma'd the device and tried again, same results. > > > > returned for refund and never looked back... > > > > SSD devices may be fast but in a server environment reliability is key. > > currently running raid 5 across the enterprise. > > > > > > On Wed, 2012-07-11 at 12:15 -0400, Chris Ferreira wrote: > >> I would most definitely be interested in this. I have been using SSD's > >> for a number of different applications and never had a problem. Am I > >> missing something? And you talk about increasing its life? Most of the > >> SSDs I have used are rated at about 4 Million Hours which is like 456 > >> Years (which I don't believe). > >> > >> > >> > >> Very interested to hear what you have to say. > >> > >> > >> > >> Thanks, > >> > >> -Chris > >> > >> ___________________ > >> Mobile Reply > >> > >> On Jul 11, 2012, at 11:39 AM, Mario G wrote: > >> > >> > I recently spent a lot of time learning how to keep an SSD fast and > increase its life. It took a ton of research and I combined a lot of info > from several sources to install the SSD. Turns out, just installing an SSD > and using it as an HDD without doing anything else is not a good idea. > Although the topic is not specific to FreeSwitch I am wondering how many > out there would like me to put together a FS wiki page fr tuning an SSD. > It's a lot of work so I won't do it unless there are enough requests or the > developers want it. > >> > Mario G > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > Join Us At ClueCon - Aug 7-9, 2012 > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120712/c345189a/attachment.html From mario_fs at mgtech.com Thu Jul 12 04:35:18 2012 From: mario_fs at mgtech.com (Mario G) Date: Wed, 11 Jul 2012 17:35:18 -0700 Subject: [Freeswitch-users] Error in updating Freeswitch In-Reply-To: <4FFE105D.6080307@freeswitch.org> References: <000001cd5fb9$06deddf0$149c99d0$@accra.ca> <4FFE105D.6080307@freeswitch.org> Message-ID: <9CABF627-B092-4AC0-A119-12C813838220@mgtech.com> I just tried and I get: cc1: warnings being treated as errors src/switch_core.c: In function 'send_heartbeat': src/switch_core.c:86: warning: format '%lu' expects type 'long unsigned int', but argument 5 has type 'switch_time_t' make[2]: *** [libfreeswitch_la-switch_core.lo] Error 1 make[1]: *** [all] Error 2 make: *** [current] Error 2 On Jul 11, 2012, at 4:46 PM, Stefan Knoblich wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On 07/12/12 01:00, Charles Bujold wrote: >> /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c: In function ?send_rfc_event?: /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c:1251: error: >> format not a string literal and no format arguments /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c:1252: error: format not a string literal and no format arguments >> /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c:1255: error: format not a string literal and no format arguments >> /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c: In function ?call_setup_event_handler?: /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c:8056: >> error: format not a string literal and no format arguments /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c: In function ?conf_data_event_handler?: >> /usr/src/freeswitch/src/mod/applications/mod_conference/mod_conference.c:8091: error: format not a string literal and no format arguments > > That's fixed now, please update. > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v2.0.19 (GNU/Linux) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ > > iEYEARECAAYFAk/+EF0ACgkQjiIIAK4rYUrDggCgrmfq5vcthCyjohqTqW19B2Nz > 1csAn0uywNmmluuarIOR8Ky81lUx8ttS > =o4GJ > -----END PGP SIGNATURE----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mario_fs at mgtech.com Thu Jul 12 04:39:50 2012 From: mario_fs at mgtech.com (Mario G) Date: Wed, 11 Jul 2012 17:39:50 -0700 Subject: [Freeswitch-users] make current fails today Message-ID: Was ok Tuesday...... on osX. c/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DMACOSX -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -pipe -no-cpp-precomp -MT libfreeswitch_la-switch_core.lo -MD -MP -MF .deps/libfreeswitch_la-switch_core.Tpo -c src/switch_core.c -fno-common -DPIC -o .libs/libfreeswitch_la-switch_core.o cc1: warnings being treated as errors src/switch_core.c: In function 'send_heartbeat': src/switch_core.c:86: warning: format '%lu' expects type 'long unsigned int', but argument 5 has type 'switch_time_t' make[2]: *** [libfreeswitch_la-switch_core.lo] Error 1 make[1]: *** [all] Error 2 make: *** [current] Error 2 From anthony.minessale at gmail.com Thu Jul 12 04:52:54 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 11 Jul 2012 19:52:54 -0500 Subject: [Freeswitch-users] make current fails today In-Reply-To: References: Message-ID: http://jira.freeswitch.org On Wed, Jul 11, 2012 at 7:39 PM, Mario G wrote: > Was ok Tuesday...... on osX. > > c/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DMACOSX -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -pipe -no-cpp-precomp -MT libfreeswitch_la-switch_core.lo -MD -MP -MF .deps/libfreeswitch_la-switch_core.Tpo -c src/switch_core.c -fno-common -DPIC -o .libs/libfreeswitch_la-switch_core.o > cc1: warnings being treated as errors > src/switch_core.c: In function 'send_heartbeat': > src/switch_core.c:86: warning: format '%lu' expects type 'long unsigned int', but argument 5 has type 'switch_time_t' > make[2]: *** [libfreeswitch_la-switch_core.lo] Error 1 > make[1]: *** [all] Error 2 > make: *** [current] Error 2 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From sdevoy at bizfocused.com Thu Jul 12 05:15:20 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 11 Jul 2012 21:15:20 -0400 Subject: [Freeswitch-users] make current fails today In-Reply-To: References: Message-ID: <054401cd5fcb$ce568970$6b039c50$@bizfocused.com> Mario, You do see you have the same error in your output, right? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mario G Sent: Wednesday, July 11, 2012 8:40 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] make current fails today Was ok Tuesday...... on osX. c/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DMACOSX -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -pipe -no-cpp-precomp -MT libfreeswitch_la-switch_core.lo -MD -MP -MF .deps/libfreeswitch_la-switch_core.Tpo -c src/switch_core.c -fno-common -DPIC -o .libs/libfreeswitch_la-switch_core.o cc1: warnings being treated as errors src/switch_core.c: In function 'send_heartbeat': src/switch_core.c:86: warning: format '%lu' expects type 'long unsigned int', but argument 5 has type 'switch_time_t' make[2]: *** [libfreeswitch_la-switch_core.lo] Error 1 make[1]: *** [all] Error 2 make: *** [current] Error 2 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120711/1c49a6b4/attachment.html From stkn at freeswitch.org Thu Jul 12 05:28:12 2012 From: stkn at freeswitch.org (Stefan Knoblich) Date: Thu, 12 Jul 2012 03:28:12 +0200 Subject: [Freeswitch-users] make current fails today In-Reply-To: References: Message-ID: <4FFE282C.7000802@freeswitch.org> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 07/12/12 02:39, Mario G wrote: > src/switch_core.c: In function 'send_heartbeat': src/switch_core.c:86: warning: format '%lu' expects type 'long unsigned int', but argument 5 has type 'switch_time_t' make[2]: *** > [libfreeswitch_la-switch_core.lo] Error 1 make[1]: *** [all] Error 2 make: *** [current] Error 2 Fixed in git. -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.19 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAk/+KCwACgkQjiIIAK4rYUqXsgCgnC23voZLdRtf2xVVmJARMrDR cXAAnRA68kbjONC82mFK0MWUphO+tcVO =v3Uk -----END PGP SIGNATURE----- From mario_fs at mgtech.com Thu Jul 12 06:00:13 2012 From: mario_fs at mgtech.com (Mario G) Date: Wed, 11 Jul 2012 19:00:13 -0700 Subject: [Freeswitch-users] make current fails today In-Reply-To: <054401cd5fcb$ce568970$6b039c50$@bizfocused.com> References: <054401cd5fcb$ce568970$6b039c50$@bizfocused.com> Message-ID: <292B2FA2-5B20-4F7D-8F8F-5BB14FDDAEF3@mgtech.com> Error the same now different: eeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DMACOSX -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -pipe -no-cpp-precomp -MT libfreeswitch_la-switch_rtp.lo -MD -MP -MF .deps/libfreeswitch_la-switch_rtp.Tpo -c src/switch_rtp.c -fno-common -DPIC -o .libs/libfreeswitch_la-switch_rtp.o cc1: warnings being treated as errors src/switch_rtp.c: In function 'check_srtp_and_ice': src/switch_rtp.c:1056: warning: format '%ld' expects type 'long int', but argument 8 has type 'switch_time_t' make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 make[1]: *** [all] Error 2 make: *** [current] Error 2 On Jul 11, 2012, at 6:15 PM, Sean Devoy wrote: > Mario, > > You do see you have the same error in your output, right? > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mario G > Sent: Wednesday, July 11, 2012 8:40 PM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] make current fails today > > Was ok Tuesday...... on osX. > > c/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DMACOSX -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 -pipe -no-cpp-precomp -MT libfreeswitch_la-switch_core.lo -MD -MP -MF .deps/libfreeswitch_la-switch_core.Tpo -c src/switch_core.c -fno-common -DPIC -o .libs/libfreeswitch_la-switch_core.o > > cc1: warnings being treated as errors > > src/switch_core.c: In function 'send_heartbeat': > > src/switch_core.c:86: warning: format '%lu' expects type 'long unsigned int', but argument 5 has type 'switch_time_t' > > make[2]: *** [libfreeswitch_la-switch_core.lo] Error 1 > > make[1]: *** [all] Error 2 > > make: *** [current] Error 2 > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120711/0125f276/attachment.html From vbvbrj at gmail.com Thu Jul 12 09:12:27 2012 From: vbvbrj at gmail.com (Vbvbrj) Date: Thu, 12 Jul 2012 08:12:27 +0300 Subject: [Freeswitch-users] Sound problems in voicemail. Message-ID: <4FFE5CBB.60406@gmail.com> Hello. I have a default installation with default config files. When calling voicemail, ie extension not registered, the sound of answering machine is poor. Not the quality, but rather with interrupts. And when spelling extenstion, ie 1001, the sound seems to be distorted like slowing down. What this cause a problem? Thank you. From msc at freeswitch.org Thu Jul 12 09:42:32 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 11 Jul 2012 22:42:32 -0700 Subject: [Freeswitch-users] make current fails today In-Reply-To: <292B2FA2-5B20-4F7D-8F8F-5BB14FDDAEF3@mgtech.com> References: <054401cd5fcb$ce568970$6b039c50$@bizfocused.com> <292B2FA2-5B20-4F7D-8F8F-5BB14FDDAEF3@mgtech.com> Message-ID: Please move this to jira.freeswitch.org so we can keep track of its progress. Besides, there are 4K+ people subscribed to this list and a very large percentage of them are not interested in this discussion. -MC On Wed, Jul 11, 2012 at 7:00 PM, Mario G wrote: > Error the same now different: > > eeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DMACOSX -DHAVE_OPENSSL -Wall -std=c99 -pedantic > -Wdeclaration-after-statement -g -O2 -pipe -no-cpp-precomp -MT > libfreeswitch_la-switch_rtp.lo -MD -MP -MF > .deps/libfreeswitch_la-switch_rtp.Tpo -c src/switch_rtp.c -fno-common > -DPIC -o .libs/libfreeswitch_la-switch_rtp.o > cc1: warnings being treated as errors > src/switch_rtp.c: In function 'check_srtp_and_ice': > src/switch_rtp.c:1056: warning: format '%ld' expects type 'long int', but > argument 8 has type 'switch_time_t' > make[2]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 > make[1]: *** [all] Error 2 > make: *** [current] Error 2 > > On Jul 11, 2012, at 6:15 PM, Sean Devoy wrote: > > Mario, > > You do see you have the same error in your output, right? > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org] > On Behalf Of Mario G > Sent: Wednesday, July 11, 2012 8:40 PM > To: FreeSWITCH Users Help > Subject: [Freeswitch-users] make current fails today > > Was ok Tuesday...... on osX. > > c/freeswitch/src/include -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DMACOSX -DHAVE_OPENSSL -Wall -std=c99 -pedantic > -Wdeclaration-after-statement -g -O2 -pipe -no-cpp-precomp -MT > libfreeswitch_la-switch_core.lo -MD -MP -MF > .deps/libfreeswitch_la-switch_core.Tpo -c src/switch_core.c -fno-common > -DPIC -o .libs/libfreeswitch_la-switch_core.o > > cc1: warnings being treated as errors > > src/switch_core.c: In function 'send_heartbeat': > > src/switch_core.c:86: warning: format '%lu' expects type 'long unsigned > int', but argument 5 has type 'switch_time_t' > > make[2]: *** [libfreeswitch_la-switch_core.lo] Error 1 > > make[1]: *** [all] Error 2 > > make: *** [current] Error 2 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120711/1a7e0b2d/attachment-0001.html From terry at digital-outpost.com Thu Jul 12 10:15:48 2012 From: terry at digital-outpost.com (Terry Barnum) Date: Wed, 11 Jul 2012 23:15:48 -0700 Subject: [Freeswitch-users] new freeswitch/fusionpbx user - SOLVED In-Reply-To: <4FFD2636.3040906@gmail.com> References: <4FFD2636.3040906@gmail.com> Message-ID: <3151B2D1-4243-4D46-9A11-33212D85D0EA@digital-outpost.com> Figured it out. After initially getting freeswitch compiled and running I briefly tried Bluebox but it felt a bit too basic. In that process of testing it seems to have altered freeswitch/conf/autoload_configs/ivr.conf.xml. So I copied src/freeswitch/conf/vanilla/autoload_configs/ivr.conf.xml over it and now it works like a charm. I apologize I didn't catch it sooner. Onward! -Terry On Jul 11, 2012, at 12:07 AM, Carlo Dimaggio wrote: > Hi Terry, > what about calling directly ivr extension 5002? > > I'm using ivr in fusionpbx without problems... > > > Il 11/07/12 01.19, Terry Barnum ha scritto: >> I'm starting to find my way around freeswitch via fusionpbx and have it running with a free DID, two X-Lite softphones and 3CXPhone on an iPhone. After creating a recording, an IVR menu and an inbound route to the IVR, a test call coming in on the DID complains it can't find the menu: >> [snip] From muelbuesch at as-infodienste.de Thu Jul 12 12:08:31 2012 From: muelbuesch at as-infodienste.de (=?ISO-8859-15?Q?Marcus_M=FClb=FCsch?=) Date: Thu, 12 Jul 2012 10:08:31 +0200 Subject: [Freeswitch-users] Unable to set effective_caller_id_number when bridging using Openzap In-Reply-To: <4FFAAE9E.3010705@as-infodienste.de> References: <4FF57BA5.7090000@as-infodienste.de> <4FFAAE9E.3010705@as-infodienste.de> Message-ID: <4FFE85FF.6020307@as-infodienste.de> Hello, More info I should have provided earlier:: we're using a Sangoma A104 with Firmware V14, thus no newer version of freeswitch is possible :( Marcus > Thank you, > > you'll find the freeswitch.log here: > > http://pastebin.com/mz5fSVwA > > openzap.conf and openzap.conf.cml are here: > > http://pastebin.com/7WYVitLN and > http://pastebin.com/F1zB6hQq > > zt.conf and wanpipe.conf are here. > > http://pastebin.com/5M3ezCPv > > We're using freeswitch 1.0.4 (Sorry again) and WANPIPE Release: > 3.3.14.11 > > The call comes in via an FXS Analog Line on a Digium TDM 2400P and > gets bridged to the ISDN line to a local mobile phone. The number is not > changed to the number set but remains the number allocated to the fixed > line. > > Thanks for any pointers, > > Marcus From anita.hall at simmortel.com Thu Jul 12 16:17:32 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Thu, 12 Jul 2012 17:47:32 +0530 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: References: <-5316884151656024069@unknownmsgid> <1342036306.3587.7.camel@mythtv> Message-ID: Count me in! We are using SSD in an embedded system. regards, Anita On Thu, Jul 12, 2012 at 5:58 AM, Abdul Basit wrote: > i m definitely interested and appreciate you willingness of knowledge > sharing. Love hardware capacity, scalability and long life so waiting for > the information you collected. > > -- > Regards, > > Abdul Basit > > > On Thu, Jul 12, 2012 at 1:22 AM, Muhammad Naseer Bhatti > wrote: > >> Depends on what brand the drive is. We are using a lot of SSDs in a >> huge enterprise environment and never faced issues. On average each >> SSD is giving us around 2000 IOPS with a combine output of around >> 32000 IOPS and we will be upgrading to 32 soon. Currently have 16. >> >> For smaller scale, I am using one in my Macbook, and never had any >> issues. Reaches upto 5000 IOPS and data transfer rate of around 200+ >> MBps. Have been using a quite a few in the servers running for FS. >> They work very well and never deceived. Surely, the thread is a good >> one, if the OP posts the draft :) >> >> Thanks. >> >> On Wed, Jul 11, 2012 at 10:51 PM, Todd Bailey >> wrote: >> > >> > Or 4 million micro seconds what ever comes first. >> > >> > I tried using a 90 gb ssd on linux and being curious, ran >> > phoronix-test-suite against the drive and with 15 minutes the drive was >> > non operational, rma'd the device and tried again, same results. >> > >> > returned for refund and never looked back... >> > >> > SSD devices may be fast but in a server environment reliability is key. >> > currently running raid 5 across the enterprise. >> > >> > >> > On Wed, 2012-07-11 at 12:15 -0400, Chris Ferreira wrote: >> >> I would most definitely be interested in this. I have been using SSD's >> >> for a number of different applications and never had a problem. Am I >> >> missing something? And you talk about increasing its life? Most of the >> >> SSDs I have used are rated at about 4 Million Hours which is like 456 >> >> Years (which I don't believe). >> >> >> >> >> >> >> >> Very interested to hear what you have to say. >> >> >> >> >> >> >> >> Thanks, >> >> >> >> -Chris >> >> >> >> ___________________ >> >> Mobile Reply >> >> >> >> On Jul 11, 2012, at 11:39 AM, Mario G wrote: >> >> >> >> > I recently spent a lot of time learning how to keep an SSD fast and >> increase its life. It took a ton of research and I combined a lot of info >> from several sources to install the SSD. Turns out, just installing an SSD >> and using it as an HDD without doing anything else is not a good idea. >> Although the topic is not specific to FreeSwitch I am wondering how many >> out there would like me to put together a FS wiki page fr tuning an SSD. >> It's a lot of work so I won't do it unless there are enough requests or the >> developers want it. >> >> > Mario G >> >> > >> _________________________________________________________________________ >> >> > Professional FreeSWITCH Consulting Services: >> >> > consulting at freeswitch.org >> >> > http://www.freeswitchsolutions.com >> >> > >> >> > >> >> > >> >> > >> >> > Official FreeSWITCH Sites >> >> > http://www.freeswitch.org >> >> > http://wiki.freeswitch.org >> >> > http://www.cluecon.com >> >> > >> >> > Join Us At ClueCon - Aug 7-9, 2012 >> >> > >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> > http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120712/d2a555b6/attachment.html From anita.hall at simmortel.com Thu Jul 12 16:28:35 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Thu, 12 Jul 2012 17:58:35 +0530 Subject: [Freeswitch-users] bridge_pre_execute_bleg_app playback Message-ID: Hi! I am trying to bridge the caller to the callee and play them different sound files foo.wav and bar.wav before they are "connected". Taking help from the wiki http://wiki.freeswitch.org/wiki/Variable_bridge_pre_execute_aleg_app, I put up this dialplan but it plays back the audio files to the a-leg and b-leg one after the other and not simultaneously. What will be the way to do simultaneous playback to both the legs and then bridge them? My current dialplan is this Thanks! regards, Anita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120712/d6145768/attachment-0001.html From mario_fs at mgtech.com Thu Jul 12 20:16:11 2012 From: mario_fs at mgtech.com (Mario G) Date: Thu, 12 Jul 2012 09:16:11 -0700 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: References: <-5316884151656024069@unknownmsgid> <1342036306.3587.7.camel@mythtv> Message-ID: <83D31466-D013-4078-A1A2-D95CC3FE1DEA@mgtech.com> Ok been working on it all day yesterday and should be done by Monday, will post a new subject when done. MC - I did not put the SSD into my FS Mac mini, it was another Linux machine. I spent months learning beforehand because I was thinking about it then I got a free Intel 320 160G. I just thought some of the FS folks might like to see what I did and learned. I replaced a WDC Velociraptor and this is WAAYY FASTER! On Jul 12, 2012, at 5:17 AM, Anita Hall wrote: > Count me in! We are using SSD in an embedded system. > > regards, > Anita > > > > On Thu, Jul 12, 2012 at 5:58 AM, Abdul Basit wrote: > i m definitely interested and appreciate you willingness of knowledge sharing. Love hardware capacity, scalability and long life so waiting for the information you collected. > > -- > Regards, > > Abdul Basit > > > On Thu, Jul 12, 2012 at 1:22 AM, Muhammad Naseer Bhatti wrote: > Depends on what brand the drive is. We are using a lot of SSDs in a > huge enterprise environment and never faced issues. On average each > SSD is giving us around 2000 IOPS with a combine output of around > 32000 IOPS and we will be upgrading to 32 soon. Currently have 16. > > For smaller scale, I am using one in my Macbook, and never had any > issues. Reaches upto 5000 IOPS and data transfer rate of around 200+ > MBps. Have been using a quite a few in the servers running for FS. > They work very well and never deceived. Surely, the thread is a good > one, if the OP posts the draft :) > > Thanks. > > On Wed, Jul 11, 2012 at 10:51 PM, Todd Bailey wrote: > > > > Or 4 million micro seconds what ever comes first. > > > > I tried using a 90 gb ssd on linux and being curious, ran > > phoronix-test-suite against the drive and with 15 minutes the drive was > > non operational, rma'd the device and tried again, same results. > > > > returned for refund and never looked back... > > > > SSD devices may be fast but in a server environment reliability is key. > > currently running raid 5 across the enterprise. > > > > > > On Wed, 2012-07-11 at 12:15 -0400, Chris Ferreira wrote: > >> I would most definitely be interested in this. I have been using SSD's > >> for a number of different applications and never had a problem. Am I > >> missing something? And you talk about increasing its life? Most of the > >> SSDs I have used are rated at about 4 Million Hours which is like 456 > >> Years (which I don't believe). > >> > >> > >> > >> Very interested to hear what you have to say. > >> > >> > >> > >> Thanks, > >> > >> -Chris > >> > >> ___________________ > >> Mobile Reply > >> > >> On Jul 11, 2012, at 11:39 AM, Mario G wrote: > >> > >> > I recently spent a lot of time learning how to keep an SSD fast and increase its life. It took a ton of research and I combined a lot of info from several sources to install the SSD. Turns out, just installing an SSD and using it as an HDD without doing anything else is not a good idea. Although the topic is not specific to FreeSwitch I am wondering how many out there would like me to put together a FS wiki page fr tuning an SSD. It's a lot of work so I won't do it unless there are enough requests or the developers want it. > >> > Mario G > >> > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > Join Us At ClueCon - Aug 7-9, 2012 > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120712/f93b27df/attachment.html From mgg at giagnocavo.net Thu Jul 12 20:29:05 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 12 Jul 2012 16:29:05 +0000 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: <83D31466-D013-4078-A1A2-D95CC3FE1DEA@mgtech.com> References: <-5316884151656024069@unknownmsgid> <1342036306.3587.7.camel@mythtv> <83D31466-D013-4078-A1A2-D95CC3FE1DEA@mgtech.com> Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B604512C92@BLUPRD0711MB413.namprd07.prod.outlook.com> Just as a side note for those that have not heard: Once you go SSD on your work machine, you can never go back. My Windows 7 laptop boots faster than my Android phone. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Mario G Sent: Thursday, July 12, 2012 10:16 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? Ok been working on it all day yesterday and should be done by Monday, will post a new subject when done. MC - I did not put the SSD into my FS Mac mini, it was another Linux machine. I spent months learning beforehand because I was thinking about it then I got a free Intel 320 160G. I just thought some of the FS folks might like to see what I did and learned. I replaced a WDC Velociraptor and this is WAAYY FASTER! On Jul 12, 2012, at 5:17 AM, Anita Hall wrote: Count me in! We are using SSD in an embedded system. regards, Anita On Thu, Jul 12, 2012 at 5:58 AM, Abdul Basit > wrote: i m definitely interested and appreciate you willingness of knowledge sharing. Love hardware capacity, scalability and long life so waiting for the information you collected. -- Regards, Abdul Basit On Thu, Jul 12, 2012 at 1:22 AM, Muhammad Naseer Bhatti > wrote: Depends on what brand the drive is. We are using a lot of SSDs in a huge enterprise environment and never faced issues. On average each SSD is giving us around 2000 IOPS with a combine output of around 32000 IOPS and we will be upgrading to 32 soon. Currently have 16. For smaller scale, I am using one in my Macbook, and never had any issues. Reaches upto 5000 IOPS and data transfer rate of around 200+ MBps. Have been using a quite a few in the servers running for FS. They work very well and never deceived. Surely, the thread is a good one, if the OP posts the draft :) Thanks. On Wed, Jul 11, 2012 at 10:51 PM, Todd Bailey > wrote: > > Or 4 million micro seconds what ever comes first. > > I tried using a 90 gb ssd on linux and being curious, ran > phoronix-test-suite against the drive and with 15 minutes the drive was > non operational, rma'd the device and tried again, same results. > > returned for refund and never looked back... > > SSD devices may be fast but in a server environment reliability is key. > currently running raid 5 across the enterprise. > > > On Wed, 2012-07-11 at 12:15 -0400, Chris Ferreira wrote: >> I would most definitely be interested in this. I have been using SSD's >> for a number of different applications and never had a problem. Am I >> missing something? And you talk about increasing its life? Most of the >> SSDs I have used are rated at about 4 Million Hours which is like 456 >> Years (which I don't believe). >> >> >> >> Very interested to hear what you have to say. >> >> >> >> Thanks, >> >> -Chris >> >> ___________________ >> Mobile Reply >> >> On Jul 11, 2012, at 11:39 AM, Mario G > wrote: >> >> > I recently spent a lot of time learning how to keep an SSD fast and increase its life. It took a ton of research and I combined a lot of info from several sources to install the SSD. Turns out, just installing an SSD and using it as an HDD without doing anything else is not a good idea. Although the topic is not specific to FreeSwitch I am wondering how many out there would like me to put together a FS wiki page fr tuning an SSD. It's a lot of work so I won't do it unless there are enough requests or the developers want it. >> > Mario G >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120712/50a42781/attachment-0001.html From bdfoster at endigotech.com Thu Jul 12 20:36:05 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 12 Jul 2012 12:36:05 -0400 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B604512C92@BLUPRD0711MB413.namprd07.prod.outlook.com> References: <-5316884151656024069@unknownmsgid> <1342036306.3587.7.camel@mythtv> <83D31466-D013-4078-A1A2-D95CC3FE1DEA@mgtech.com> <63B00DD1DA6A364E9F64A3A0BD2FE7B604512C92@BLUPRD0711MB413.namprd07.prod.outlook.com> Message-ID: My linux laptop always boots faster than my android phone, maybe there are problems with Windows? ;-) Laptop: 1.8ghz, 1GB 5400RPM Western Digital running Ubuntu 12.04 Phone: Motorola ATRIX Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 12, 2012 12:30 PM, "Michael Giagnocavo" wrote: > Just as a side note for those that have not heard: Once you go SSD on > your work machine, you can never go back. My Windows 7 laptop boots faster > than my Android phone.**** > > -Michael**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Mario G > *Sent:* Thursday, July 12, 2012 10:16 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki?* > *** > > ** ** > > Ok been working on it all day yesterday and should be done by Monday, will > post a new subject when done.**** > > ** ** > > MC - I did not put the SSD into my FS Mac mini, it was another Linux > machine. I spent months learning beforehand because I was thinking about it > then I got a free Intel 320 160G. I just thought some of the FS folks might > like to see what I did and learned. I replaced a WDC Velociraptor and this > is WAAYY FASTER!**** > > ** ** > > On Jul 12, 2012, at 5:17 AM, Anita Hall wrote:**** > > > > **** > > Count me in! We are using SSD in an embedded system. > > regards, > Anita > > > **** > > On Thu, Jul 12, 2012 at 5:58 AM, Abdul Basit wrote: > **** > > i m definitely interested and appreciate you willingness of knowledge > sharing. Love hardware capacity, scalability and long life so waiting for > the information you collected.**** > > ** ** > > -- **** > > Regards,**** > > ** ** > > Abdul Basit**** > > **** > > On Thu, Jul 12, 2012 at 1:22 AM, Muhammad Naseer Bhatti > wrote:**** > > Depends on what brand the drive is. We are using a lot of SSDs in a > huge enterprise environment and never faced issues. On average each > SSD is giving us around 2000 IOPS with a combine output of around > 32000 IOPS and we will be upgrading to 32 soon. Currently have 16. > > For smaller scale, I am using one in my Macbook, and never had any > issues. Reaches upto 5000 IOPS and data transfer rate of around 200+ > MBps. Have been using a quite a few in the servers running for FS. > They work very well and never deceived. Surely, the thread is a good > one, if the OP posts the draft :) > > Thanks.**** > > > On Wed, Jul 11, 2012 at 10:51 PM, Todd Bailey > wrote: > > > > Or 4 million micro seconds what ever comes first. > > > > I tried using a 90 gb ssd on linux and being curious, ran > > phoronix-test-suite against the drive and with 15 minutes the drive was > > non operational, rma'd the device and tried again, same results. > > > > returned for refund and never looked back... > > > > SSD devices may be fast but in a server environment reliability is key. > > currently running raid 5 across the enterprise. > > > > > > On Wed, 2012-07-11 at 12:15 -0400, Chris Ferreira wrote: > >> I would most definitely be interested in this. I have been using SSD's > >> for a number of different applications and never had a problem. Am I > >> missing something? And you talk about increasing its life? Most of the > >> SSDs I have used are rated at about 4 Million Hours which is like 456 > >> Years (which I don't believe). > >> > >> > >> > >> Very interested to hear what you have to say. > >> > >> > >> > >> Thanks, > >> > >> -Chris > >> > >> ___________________ > >> Mobile Reply > >> > >> On Jul 11, 2012, at 11:39 AM, Mario G wrote: > >> > >> > I recently spent a lot of time learning how to keep an SSD fast and > increase its life. It took a ton of research and I combined a lot of info > from several sources to install the SSD. Turns out, just installing an SSD > and using it as an HDD without doing anything else is not a good idea. > Although the topic is not specific to FreeSwitch I am wondering how many > out there would like me to put together a FS wiki page fr tuning an SSD. > It's a lot of work so I won't do it unless there are enough requests or the > developers want it. > >> > Mario G > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > Join Us At ClueCon - Aug 7-9, 2012 > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120712/e1289b0f/attachment-0001.html From kunalkerkar at gmail.com Thu Jul 12 17:44:57 2012 From: kunalkerkar at gmail.com (tsudot) Date: Thu, 12 Jul 2012 06:44:57 -0700 (PDT) Subject: [Freeswitch-users] Playing of mms stream by means of FS In-Reply-To: References: Message-ID: <1342100697222-7580783.post@n2.nabble.com> You need to install the latest version of VLC as mentioned on the mod_vlc wiki. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Playing-of-mms-stream-by-means-of-FS-tp7579341p7580783.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ocset at the800group.com Thu Jul 12 21:00:59 2012 From: ocset at the800group.com (ocset) Date: Fri, 13 Jul 2012 01:00:59 +0800 Subject: [Freeswitch-users] Security settings for FXO gateway Message-ID: <4FFF02CB.3030603@the800group.com> Hi There is a wiki on the web which describes how to use a Grandstream FXO device with freesiwtch and recommends the following settings. How big a security hole is this creating and is there a better way to do this? Also, what would the syntax be for the "Add ip=" recommendation? *add file conf/directory/default/pstn.xml containing* *change following in conf/sip_profiles/internal.xml* Thanks advance O -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120713/df96412d/attachment.html From aaron.berndsen at gmail.com Thu Jul 12 20:56:06 2012 From: aaron.berndsen at gmail.com (Aaron Berndsen) Date: Thu, 12 Jul 2012 09:56:06 -0700 Subject: [Freeswitch-users] trunk_1 registration failed (timeout) In-Reply-To: References: Message-ID: My freeswitch configuration was working fine until I upgraded to ubuntu 12.04. Now sofia status indicates that registering trunk_1 to my sip provider times out: 2012-07-12 09:02:51.252018 [WARNING] sofia_reg.c:449 Timeout Registering trunk_1 2012-07-12 09:02:53.252075 [WARNING] sofia_reg.c:468 trunk_1 Failed Registration [908], setting retry to 45 seconds. A major clue is that sofia_dig on the ipv4 of my sip provider returns instantly, while if I use the dns hostname it takes about 3 minutes to return the information. freeswitch at internal> sofia_dig 64.254.145.209 Preference Weight Transport Port Address ================================================================================ 1 1.000 udp 5060 64.254.145.209 2 1.000 tcp 5060 64.254.145.209 Ubuntu 12.04 introduced dns-cacheing, but I have tried turning this off (comment out dns=dnsmasq in /etc/NetworkManager/NetworkManager.conf) and the problem persists. Any tips or suggestions would be helpful. I should point out that my system is behind a router, though I've had the FS configuration working with NAT before. Cheers, -Aaron From qasimakhan at gmail.com Thu Jul 12 21:26:37 2012 From: qasimakhan at gmail.com (qasimakhan at gmail.com) Date: Thu, 12 Jul 2012 22:26:37 +0500 Subject: [Freeswitch-users] Media Bypass in Mod_rtmp. Message-ID: Hi, A quick question if someone could help me out. Can we use mediabypass mode when using mod_rtmp? Regards, Qasim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120712/195e6b1a/attachment.html From krice at freeswitch.org Thu Jul 12 22:50:53 2012 From: krice at freeswitch.org (Ken Rice) Date: Thu, 12 Jul 2012 13:50:53 -0500 Subject: [Freeswitch-users] Media Bypass in Mod_rtmp. In-Reply-To: Message-ID: No you can not... Bypass media only works on sip to sip calls On 7/12/12 12:26 PM, "qasimakhan at gmail.com" wrote: > Hi, > > A quick question if someone could help me out. Can we use mediabypass mode > when using mod_rtmp? > > Regards, > Qasim > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120712/76edbfa3/attachment.html From qasimakhan at gmail.com Thu Jul 12 22:56:36 2012 From: qasimakhan at gmail.com (qasimakhan at gmail.com) Date: Thu, 12 Jul 2012 23:56:36 +0500 Subject: [Freeswitch-users] Media Bypass in Mod_rtmp. In-Reply-To: References: Message-ID: Thanks man. On Jul 12, 2012 11:55 PM, "Ken Rice" wrote: > No you can not... Bypass media only works on sip to sip calls > > > On 7/12/12 12:26 PM, "qasimakhan at gmail.com" wrote: > > Hi, > > A quick question if someone could help me out. Can we use mediabypass mode > when using mod_rtmp? > > Regards, > Qasim > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120712/0913c23a/attachment.html From toddb at toddbailey.net Thu Jul 12 23:57:00 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Thu, 12 Jul 2012 12:57:00 -0700 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: References: <-5316884151656024069@unknownmsgid> <1342036306.3587.7.camel@mythtv> Message-ID: <1342123020.3613.2.camel@mythtv> SSD and Long Life should never be used in the same sentence. On Thu, 2012-07-12 at 05:28 +0500, Abdul Basit wrote: > i m definitely interested and appreciate you willingness of knowledge > sharing. Love hardware capacity, scalability and long life so waiting > for the information you collected. > > > -- > Regards, > > > Abdul Basit > > > On Thu, Jul 12, 2012 at 1:22 AM, Muhammad Naseer Bhatti > wrote: > Depends on what brand the drive is. We are using a lot of SSDs > in a > huge enterprise environment and never faced issues. On average > each > SSD is giving us around 2000 IOPS with a combine output of > around > 32000 IOPS and we will be upgrading to 32 soon. Currently have > 16. > > For smaller scale, I am using one in my Macbook, and never had > any > issues. Reaches upto 5000 IOPS and data transfer rate of > around 200+ > MBps. Have been using a quite a few in the servers running for > FS. > They work very well and never deceived. Surely, the thread is > a good > one, if the OP posts the draft :) > > Thanks. > > On Wed, Jul 11, 2012 at 10:51 PM, Todd Bailey > wrote: > > > > Or 4 million micro seconds what ever comes first. > > > > I tried using a 90 gb ssd on linux and being curious, ran > > phoronix-test-suite against the drive and with 15 minutes > the drive was > > non operational, rma'd the device and tried again, same > results. > > > > returned for refund and never looked back... > > > > SSD devices may be fast but in a server environment > reliability is key. > > currently running raid 5 across the enterprise. > > > > > > On Wed, 2012-07-11 at 12:15 -0400, Chris Ferreira wrote: > >> I would most definitely be interested in this. I have been > using SSD's > >> for a number of different applications and never had a > problem. Am I > >> missing something? And you talk about increasing its life? > Most of the > >> SSDs I have used are rated at about 4 Million Hours which > is like 456 > >> Years (which I don't believe). > >> > >> > >> > >> Very interested to hear what you have to say. > >> > >> > >> > >> Thanks, > >> > >> -Chris > >> > >> ___________________ > >> Mobile Reply > >> > >> On Jul 11, 2012, at 11:39 AM, Mario G > wrote: > >> > >> > I recently spent a lot of time learning how to keep an > SSD fast and increase its life. It took a ton of research and > I combined a lot of info from several sources to install the > SSD. Turns out, just installing an SSD and using it as an HDD > without doing anything else is not a good idea. Although the > topic is not specific to FreeSwitch I am wondering how many > out there would like me to put together a FS wiki page fr > tuning an SSD. It's a lot of work so I won't do it unless > there are enough requests or the developers want it. > >> > Mario G > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > FreeSWITCH-powered IP PBX: The CudaTel Communication > Server > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > Join Us At ClueCon - Aug 7-9, 2012 > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From toddb at toddbailey.net Fri Jul 13 00:02:33 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Thu, 12 Jul 2012 13:02:33 -0700 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B604512C92@BLUPRD0711MB413.namprd07.prod.outlook.com> References: <-5316884151656024069@unknownmsgid> <1342036306.3587.7.camel@mythtv> <83D31466-D013-4078-A1A2-D95CC3FE1DEA@mgtech.com> <63B00DD1DA6A364E9F64A3A0BD2FE7B604512C92@BLUPRD0711MB413.namprd07.prod.outlook.com> Message-ID: <1342123353.3613.7.camel@mythtv> I'm using 4 SAS 2.5 73 gb 10K rpm drives raid 5 (205 gb formatted total) on a HP P600 controller The machine boots faster that a single 64gb ssd device used briefly for testing. Just as fast, quiet, cheap, reliable and long life. Best of both worlds. On Thu, 2012-07-12 at 16:29 +0000, Michael Giagnocavo wrote: > Just as a side note for those that have not heard: Once you go SSD on > your work machine, you can never go back. My Windows 7 laptop boots > faster than my Android phone. > > -Michael > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Mario G > Sent: Thursday, July 12, 2012 10:16 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Anyone want my SSD tips on the FS > Wiki? > > > > > Ok been working on it all day yesterday and should be done by Monday, > will post a new subject when done. > > > > > MC - I did not put the SSD into my FS Mac mini, it was another Linux > machine. I spent months learning beforehand because I was thinking > about it then I got a free Intel 320 160G. I just thought some of the > FS folks might like to see what I did and learned. I replaced a WDC > Velociraptor and this is WAAYY FASTER! > > > > > On Jul 12, 2012, at 5:17 AM, Anita Hall wrote: > > > > > Count me in! We are using SSD in an embedded system. > > regards, > Anita > > > > > On Thu, Jul 12, 2012 at 5:58 AM, Abdul Basit > wrote: > > i m definitely interested and appreciate you willingness of knowledge > sharing. Love hardware capacity, scalability and long life so waiting > for the information you collected. > > > > > -- > > > Regards, > > > > > > Abdul Basit > > > > > On Thu, Jul 12, 2012 at 1:22 AM, Muhammad Naseer Bhatti > wrote: > > Depends on what brand the drive is. We are using a lot of SSDs in a > huge enterprise environment and never faced issues. On average each > SSD is giving us around 2000 IOPS with a combine output of around > 32000 IOPS and we will be upgrading to 32 soon. Currently have 16. > > For smaller scale, I am using one in my Macbook, and never had any > issues. Reaches upto 5000 IOPS and data transfer rate of around 200+ > MBps. Have been using a quite a few in the servers running for FS. > They work very well and never deceived. Surely, the thread is a good > one, if the OP posts the draft :) > > Thanks. > > > On Wed, Jul 11, 2012 at 10:51 PM, Todd Bailey > wrote: > > > > Or 4 million micro seconds what ever comes first. > > > > I tried using a 90 gb ssd on linux and being curious, ran > > phoronix-test-suite against the drive and with 15 minutes the drive > was > > non operational, rma'd the device and tried again, same results. > > > > returned for refund and never looked back... > > > > SSD devices may be fast but in a server environment reliability is > key. > > currently running raid 5 across the enterprise. > > > > > > On Wed, 2012-07-11 at 12:15 -0400, Chris Ferreira wrote: > >> I would most definitely be interested in this. I have been using > SSD's > >> for a number of different applications and never had a problem. Am > I > >> missing something? And you talk about increasing its life? Most of > the > >> SSDs I have used are rated at about 4 Million Hours which is like > 456 > >> Years (which I don't believe). > >> > >> > >> > >> Very interested to hear what you have to say. > >> > >> > >> > >> Thanks, > >> > >> -Chris > >> > >> ___________________ > >> Mobile Reply > >> > >> On Jul 11, 2012, at 11:39 AM, Mario G wrote: > >> > >> > I recently spent a lot of time learning how to keep an SSD fast > and increase its life. It took a ton of research and I combined a lot > of info from several sources to install the SSD. Turns out, just > installing an SSD and using it as an HDD without doing anything else > is not a good idea. Although the topic is not specific to FreeSwitch I > am wondering how many out there would like me to put together a FS > wiki page fr tuning an SSD. It's a lot of work so I won't do it unless > there are enough requests or the developers want it. > >> > Mario G > >> > > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > Join Us At ClueCon - Aug 7-9, 2012 > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> > http://www.freeswitch.org > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From hexade at hotmail.com Fri Jul 13 00:39:03 2012 From: hexade at hotmail.com (Adelia C.) Date: Thu, 12 Jul 2012 16:39:03 -0400 Subject: [Freeswitch-users] One Way RTP (no NAT) - Freeswitch sends RTP to the remote source port instead of SDP negotiated destination port (with disable_rtp_auto_adjust=true) Message-ID: We're testing a feature that requires an extra Application Server (B2BUA) on our call path, on the egress side of the call from FreeSwitch. With the extra AS, we have only one-way RTP (Caller -> Callee). Without, RTP is fine. We narrowed it down to FS enforcing symmetric RTP while AS only works with asymmetric RTP. We cannot change the behavior of the AS at this time. Call: Caller - >NTW -> SBC -> FS -> AS -> SBC -> NTW -> Callee. RTP streams, as negotiated (SDP): Caller - > SBCOut:50460 ->FSIn:29526 -> FSOut:17818 -> ASIn:8080 -> ASOutB:19040 -> SBCOut:49162 -> Callee (can't hear a thing) Callee -> SBCOut:49162 ->ASIn:8082 -> ASOut:19000 -> FSIn:17818 -> FSOut:29526 -> SBCIn:50460 -> Caller (good RTP) RTP streams, as sent/receiced: FS negotiates 8080 as the RTP destination to AS in the "caller - > callee" direction. For about .5s, that's how the RTP flows : FSOut:17818 -> ASIn:8080 FS negotiates 19000 as the RTP source from AS in the "callee -> caller" direction. This is how the RTP flows for the duration of the call.: ASOut:19000 -> FSIn:17818 FS switches its RTP stream in the "caller - > callee" direction ~.5s into the call : FSOut:17818 -> ASIn:19000. [ASIn listens on 8080, the caller - > called RTP is dead.] All this from Wireshark trace. I tried the following, with no success: 1. Added to MyFSApp.xml and dialplan\public.xml: Restarted, FS, same result. 2. Commented out Restarted, FS, same result. 3. Started FS with -nonat, then nonatmap Same result. Other info : - Our App on FS runs in B2BUA mode, doesn't proxy the call. We answers the incoming leg before initiating the outgoing leg. I can attach the Wireshark is I didn't provide enough info - I have both FS and AS pcaps. - Our FS code about 2 weeks old. What else can I try? Any other settings I can enable/disable? Can I enable extra RTP logs on FreeSwitch for debug? Thanks, A.C. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120712/23897043/attachment-0001.html From mgg at giagnocavo.net Fri Jul 13 01:43:58 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 12 Jul 2012 21:43:58 +0000 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: <1342123353.3613.7.camel@mythtv> References: <-5316884151656024069@unknownmsgid> <1342036306.3587.7.camel@mythtv> <83D31466-D013-4078-A1A2-D95CC3FE1DEA@mgtech.com> <63B00DD1DA6A364E9F64A3A0BD2FE7B604512C92@BLUPRD0711MB413.namprd07.prod.outlook.com> <1342123353.3613.7.camel@mythtv> Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B604514098@BLUPRD0711MB413.namprd07.prod.outlook.com> 4 10K drives cannot do the IOPS of any remotely decent SSD. We're talking what, 600-800 IOPS max on 4 spindles versus 20,000 IOPS at the low end for a modern SSD? (Like a $200 one.) As far as lifetime, a decent server-class SSD should be fine. Where you draw the line between junky consumer and server-grade is somewhat open to discussion, but at least at the FusionIO end, I don't think the lifetime should be a concern. Don't let cheap SSD vendors with crap firmware colour an entire class of technology. -Michael -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Todd Bailey Sent: Thursday, July 12, 2012 2:03 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? I'm using 4 SAS 2.5 73 gb 10K rpm drives raid 5 (205 gb formatted total) on a HP P600 controller The machine boots faster that a single 64gb ssd device used briefly for testing. Just as fast, quiet, cheap, reliable and long life. Best of both worlds. On Thu, 2012-07-12 at 16:29 +0000, Michael Giagnocavo wrote: > Just as a side note for those that have not heard: Once you go SSD on > your work machine, you can never go back. My Windows 7 laptop boots > faster than my Android phone. > > -Michael > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Mario G > Sent: Thursday, July 12, 2012 10:16 AM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Anyone want my SSD tips on the FS > Wiki? > > > > > Ok been working on it all day yesterday and should be done by Monday, > will post a new subject when done. > > > > > MC - I did not put the SSD into my FS Mac mini, it was another Linux > machine. I spent months learning beforehand because I was thinking > about it then I got a free Intel 320 160G. I just thought some of the > FS folks might like to see what I did and learned. I replaced a WDC > Velociraptor and this is WAAYY FASTER! > > > > > On Jul 12, 2012, at 5:17 AM, Anita Hall wrote: > > > > > Count me in! We are using SSD in an embedded system. > > regards, > Anita > > > > > On Thu, Jul 12, 2012 at 5:58 AM, Abdul Basit > wrote: > > i m definitely interested and appreciate you willingness of knowledge > sharing. Love hardware capacity, scalability and long life so waiting > for the information you collected. > > > > > -- > > > Regards, > > > > > > Abdul Basit > > > > > On Thu, Jul 12, 2012 at 1:22 AM, Muhammad Naseer Bhatti > wrote: > > Depends on what brand the drive is. We are using a lot of SSDs in a > huge enterprise environment and never faced issues. On average each > SSD is giving us around 2000 IOPS with a combine output of around > 32000 IOPS and we will be upgrading to 32 soon. Currently have 16. > > For smaller scale, I am using one in my Macbook, and never had any > issues. Reaches upto 5000 IOPS and data transfer rate of around 200+ > MBps. Have been using a quite a few in the servers running for FS. > They work very well and never deceived. Surely, the thread is a good > one, if the OP posts the draft :) > > Thanks. > > > On Wed, Jul 11, 2012 at 10:51 PM, Todd Bailey > wrote: > > > > Or 4 million micro seconds what ever comes first. > > > > I tried using a 90 gb ssd on linux and being curious, ran > > phoronix-test-suite against the drive and with 15 minutes the drive > was > > non operational, rma'd the device and tried again, same results. > > > > returned for refund and never looked back... > > > > SSD devices may be fast but in a server environment reliability is > key. > > currently running raid 5 across the enterprise. > > > > > > On Wed, 2012-07-11 at 12:15 -0400, Chris Ferreira wrote: > >> I would most definitely be interested in this. I have been using > SSD's > >> for a number of different applications and never had a problem. Am > I > >> missing something? And you talk about increasing its life? Most of > the > >> SSDs I have used are rated at about 4 Million Hours which is like > 456 > >> Years (which I don't believe). > >> > >> > >> > >> Very interested to hear what you have to say. > >> > >> > >> > >> Thanks, > >> > >> -Chris > >> > >> ___________________ > >> Mobile Reply > >> > >> On Jul 11, 2012, at 11:39 AM, Mario G wrote: > >> > >> > I recently spent a lot of time learning how to keep an SSD fast > and increase its life. It took a ton of research and I combined a lot > of info from several sources to install the SSD. Turns out, just > installing an SSD and using it as an HDD without doing anything else > is not a good idea. Although the topic is not specific to FreeSwitch I > am wondering how many out there would like me to put together a FS > wiki page fr tuning an SSD. It's a lot of work so I won't do it unless > there are enough requests or the developers want it. > >> > Mario G > >> > > ______________________________________________________________________ > ___ > >> > Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > Join Us At ClueCon - Aug 7-9, 2012 > >> > > >> > FreeSWITCH-users 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 > >> > >> > ______________________________________________________________________ > ___ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users 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 > >> > > > > > > > > > ______________________________________________________________________ > ___ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users 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 > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users 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 > > > > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users 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 > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users 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 > > > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From toddb at toddbailey.net Fri Jul 13 04:14:56 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Thu, 12 Jul 2012 17:14:56 -0700 Subject: [Freeswitch-users] Polycom IP 501 DOA ? Message-ID: <1342138496.3613.11.camel@mythtv> Hey all, I know this isn't the best place for phone troubleshooting, but can someone tell me how long it takes the phone to behave like it's working? when I plugged every thing in, no display, no network connection indicated at the ip switch, no dial tones, lights etc, the only indication is a popping or clicking sound in the handset. Any thoughts ? From bdfoster at endigotech.com Fri Jul 13 04:17:20 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 12 Jul 2012 20:17:20 -0400 Subject: [Freeswitch-users] Polycom IP 501 DOA ? In-Reply-To: <1342138496.3613.11.camel@mythtv> References: <1342138496.3613.11.camel@mythtv> Message-ID: DOA. Sorry :( Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 12, 2012 8:16 PM, "Todd Bailey" wrote: > Hey all, > > I know this isn't the best place for phone troubleshooting, but can > someone tell me how long it takes the phone to behave like it's working? > > when I plugged every thing in, no display, no network connection > indicated at the ip switch, no dial tones, lights etc, the only > indication is a popping or clicking sound in the handset. > > Any thoughts ? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120712/2cc9b73b/attachment.html From toddb at toddbailey.net Fri Jul 13 04:28:26 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Thu, 12 Jul 2012 17:28:26 -0700 Subject: [Freeswitch-users] Polycom IP 501 DOA ? In-Reply-To: References: <1342138496.3613.11.camel@mythtv> Message-ID: <1342139306.3613.12.camel@mythtv> Are you implying that I should have a display or some functionality the moment power is applied? On Thu, 2012-07-12 at 20:17 -0400, Brian Foster wrote: > DOA. Sorry :( > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > > On Jul 12, 2012 8:16 PM, "Todd Bailey" wrote: > Hey all, > > I know this isn't the best place for phone troubleshooting, > but can > someone tell me how long it takes the phone to behave like > it's working? > > when I plugged every thing in, no display, no network > connection > indicated at the ip switch, no dial tones, lights etc, the > only > indication is a popping or clicking sound in the handset. > > Any thoughts ? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jrichey at itltd.net Fri Jul 13 04:39:54 2012 From: jrichey at itltd.net (JRichey) Date: Thu, 12 Jul 2012 17:39:54 -0700 Subject: [Freeswitch-users] Polycom IP 501 DOA ? Message-ID: <6ECAF1527329364583AB525CF34ABF950B31A69F@ms.kallback.com> You should see the LEDs blink and the Polycom logo on the screen followed by a boot countdown message. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of Todd Bailey Sent: Thursday, July 12, 2012 5:28 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Polycom IP 501 DOA ? Are you implying that I should have a display or some functionality the moment power is applied? On Thu, 2012-07-12 at 20:17 -0400, Brian Foster wrote: > DOA. Sorry :( > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > > On Jul 12, 2012 8:16 PM, "Todd Bailey" wrote: > Hey all, > > I know this isn't the best place for phone troubleshooting, > but can > someone tell me how long it takes the phone to behave like > it's working? > > when I plugged every thing in, no display, no network > connection > indicated at the ip switch, no dial tones, lights etc, the > only > indication is a popping or clicking sound in the handset. > > Any thoughts ? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From toddb at toddbailey.net Fri Jul 13 04:51:25 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Thu, 12 Jul 2012 17:51:25 -0700 Subject: [Freeswitch-users] Polycom IP 501 DOA ? In-Reply-To: <6ECAF1527329364583AB525CF34ABF950B31A69F@ms.kallback.com> References: <6ECAF1527329364583AB525CF34ABF950B31A69F@ms.kallback.com> Message-ID: <1342140685.3613.14.camel@mythtv> OK, DOA then, I'll pack this up send it back and get a refund then try again On Thu, 2012-07-12 at 17:39 -0700, JRichey wrote: > You should see the LEDs blink and the Polycom logo on the screen followed by > a boot countdown message. > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of Todd > Bailey > Sent: Thursday, July 12, 2012 5:28 PM > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] Polycom IP 501 DOA ? > > > Are you implying that I should have a display or some functionality the > moment power is applied? > > > On Thu, 2012-07-12 at 20:17 -0400, Brian Foster wrote: > > DOA. Sorry :( > > > > Brian Foster > > Endigo Computer LLC > > > > Sent from a mobile device. > > > > On Jul 12, 2012 8:16 PM, "Todd Bailey" wrote: > > Hey all, > > > > I know this isn't the best place for phone troubleshooting, > > but can > > someone tell me how long it takes the phone to behave like > > it's working? > > > > when I plugged every thing in, no display, no network > > connection > > indicated at the ip switch, no dial tones, lights etc, the > > only > > indication is a popping or clicking sound in the handset. > > > > Any thoughts ? > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jmsucasas at me.com Fri Jul 13 05:39:03 2012 From: jmsucasas at me.com (Jose Miguel Sucasas Mejuto) Date: Fri, 13 Jul 2012 03:39:03 +0200 Subject: [Freeswitch-users] Receive response of an event in ESL Message-ID: Hello, I am trying to send an SMS using mod_sms via ESL using: require ESL; my $con = new ESL::ESLconnection("localhost", "8021", "ClueCon"); my $e = new ESL::ESLevent("custom", "SMS::SEND_MESSAGE"); $e->addHeader("to", "1004\@www.freeswitch.org"); $e->addHeader("from", "testing\@foobar.com"); $e->addHeader("sip_profile", "internal"); $e->addHeader("dest_proto", "sip"); $e->addBody(shift); $con->sendEvent($e); as it appears in mod_sms wiki. But, how can I read the result of the event?. sendRcv waits for the response, but only works with commands not with events. Thanks in advance. Regards From bdfoster at endigotech.com Fri Jul 13 05:59:24 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 12 Jul 2012 21:59:24 -0400 Subject: [Freeswitch-users] Polycom IP 501 DOA ? In-Reply-To: <1342139306.3613.12.camel@mythtv> References: <1342138496.3613.11.camel@mythtv> <1342139306.3613.12.camel@mythtv> Message-ID: Yes, there is no on switch Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 12, 2012 8:29 PM, "Todd Bailey" wrote: > Are you implying that I should have a display or some functionality the > moment power is applied? > > > On Thu, 2012-07-12 at 20:17 -0400, Brian Foster wrote: > > DOA. Sorry :( > > > > Brian Foster > > Endigo Computer LLC > > > > Sent from a mobile device. > > > > On Jul 12, 2012 8:16 PM, "Todd Bailey" wrote: > > Hey all, > > > > I know this isn't the best place for phone troubleshooting, > > but can > > someone tell me how long it takes the phone to behave like > > it's working? > > > > when I plugged every thing in, no display, no network > > connection > > indicated at the ip switch, no dial tones, lights etc, the > > only > > indication is a popping or clicking sound in the handset. > > > > Any thoughts ? > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120712/d2950411/attachment.html From claytondus at gmail.com Fri Jul 13 05:33:04 2012 From: claytondus at gmail.com (Clayton Davis) Date: Thu, 12 Jul 2012 20:33:04 -0500 Subject: [Freeswitch-users] Polycom IP 501 DOA ? In-Reply-To: <1342140685.3613.14.camel@mythtv> References: <6ECAF1527329364583AB525CF34ABF950B31A69F@ms.kallback.com> <1342140685.3613.14.camel@mythtv> Message-ID: Are you using the Polycom 501 series power cable? The PoE on the 501 requires a particular nonstandard cable. PoE or AC adaptor? See here as well: http://lists.digium.com/pipermail/asterisk-users/2006-March/142485.html HTH -cd 2012/7/12 Todd Bailey > OK, DOA then, > > > I'll pack this up send it back and get a refund then try again > > > On Thu, 2012-07-12 at 17:39 -0700, JRichey wrote: > > You should see the LEDs blink and the Polycom logo on the screen > followed by > > a boot countdown message. > > > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of Todd > > Bailey > > Sent: Thursday, July 12, 2012 5:28 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] Polycom IP 501 DOA ? > > > > > > Are you implying that I should have a display or some functionality the > > moment power is applied? > > > > > > On Thu, 2012-07-12 at 20:17 -0400, Brian Foster wrote: > > > DOA. Sorry :( > > > > > > Brian Foster > > > Endigo Computer LLC > > > > > > Sent from a mobile device. > > > > > > On Jul 12, 2012 8:16 PM, "Todd Bailey" wrote: > > > Hey all, > > > > > > I know this isn't the best place for phone troubleshooting, > > > but can > > > someone tell me how long it takes the phone to behave like > > > it's working? > > > > > > when I plugged every thing in, no display, no network > > > connection > > > indicated at the ip switch, no dial tones, lights etc, the > > > only > > > indication is a popping or clicking sound in the handset. > > > > > > Any thoughts ? > > > > > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- __________________ Clayton Davis ADTRAN claytondus at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120712/b8656c95/attachment-0001.html From toddb at toddbailey.net Fri Jul 13 11:18:40 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Fri, 13 Jul 2012 00:18:40 -0700 Subject: [Freeswitch-users] Polycom IP 501 DOA ? In-Reply-To: <1342138496.3613.11.camel@mythtv> References: <1342138496.3613.11.camel@mythtv> Message-ID: <1342163920.3613.33.camel@mythtv> Dodged another bullet, As it turns out the vendor sent me the wrong psu. the ip 301 and 501 use a 12 vdc psu they sent a 24volt psu, atter scrounging around for a 12 volt ad adapter, the phone has a life, unlike the new owner. But all might not be well, it seems caught in a updating boot rom and formatting file system loop On Thu, 2012-07-12 at 17:14 -0700, Todd Bailey wrote: > Hey all, > > I know this isn't the best place for phone troubleshooting, but can > someone tell me how long it takes the phone to behave like it's working? > > when I plugged every thing in, no display, no network connection > indicated at the ip switch, no dial tones, lights etc, the only > indication is a popping or clicking sound in the handset. > > Any thoughts ? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From torstein.knutsen at gmail.com Fri Jul 13 11:36:26 2012 From: torstein.knutsen at gmail.com (Torstein Knutsen) Date: Fri, 13 Jul 2012 09:36:26 +0200 Subject: [Freeswitch-users] Help with cidlookup In-Reply-To: <04918207-4B96-4836-B2A6-ABDE7B2912D7@5ninesolutions.com> References: <04918207-4B96-4836-B2A6-ABDE7B2912D7@5ninesolutions.com> Message-ID: Thank you. I will try this route ! Best regards Torstein On 11 July 2012 21:48, Spencer Thomason wrote: > Hi Torstein, > I ran into a similar problem with a US provider. > > I used a python script that wraps the api command and then parses the json > response and sets effective_caller_id_name. > > in python e.g. > > import json > from freeswitch import API > > > def cidlookup(number): > > api = API() > number = str(number) > cmd = 'cidlookup %s' % (number) > return api.executeString(cmd).strip() > > def handler(session, args): > > caller_num = session.getVariable("caller_id_number") > cnam_data = json.loads(cidlookup(caller_num)) > # get caller name from cnam_data here.. > > > > session.setVariable("effective_caller_id_name", caller_name) > return > > > (You could also do the api lookup in you script and bypass mod_cidlookup) > > > I hope that helps, > Spencer > > > > On Jul 11, 2012, at 11:38 AM, Torstein Knutsen wrote: > > Hi > > I have cidlookup partly working. > Im using a norwegian service, which returns a whole lot more than just the > number. > > Anybody here have some Ideas on how I could proceed to map "Pizza & Kina > Expressen" to the calling_id_name ? > > Thank you! > Torstein > > snipplet *** > : > > freeswitch at --hidden-ip--@internal> cidlookup 22222222 > > > > > { "title" : "Gule Sider firma API", "query" : " > http://api.eniro.com/cs/search/basic?country=no&search_word=22222222&to_list=1&version=1.1.3&from_list=1", > "totalHits" : 1 , "totalCount" : 1 , "startIndex" : 1, "itemsPerPage" : 1, > "adverts" : [ { "eniroId" : "P10000836357" , "companyInfo" : { > "companyName" : "Pizza & Kina Expressen" , "orgNumber" : null , > "companyText" : null }, "address" : { "streetName" : "Vitaminveien 11 B" , > "postCode" : "0485" , "postArea" : "Oslo" , "postBox" : null }, "location" > : { "coordinates" : [ { "longitude" : 10.7725744944096 , "latitude" : > 59.9471107465998 }, { "use" : "route", "longitude" : 10.7725744944096 , > "latitude" : 59.9471107465998 } ] }, "phoneNumbers" : [ { "type" : "std" , > "phoneNumber" : "22 22 22 22" , "label" : null } ], "companyReviews" : " > http://www.anbefalt.no/omtale/0003292695/22222222" , "homepage" : " > http://api.eniro.com/proxy/homepage/uANwPf5aVK3QsMrfdwYjz8Olp1PSJ6L1-mCsL3_LC0d9Yem9mkC025y22P034JmT" > , "infoPageLink" : " > http://www.gulesider.no/firma/pizza-kina-expressen:p10000836357?search_word=22222222" > } ] } > freeswitch at --hidden-ip--@internal> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120713/1852eac4/attachment.html From hiryu23 at gmail.com Fri Jul 13 12:24:09 2012 From: hiryu23 at gmail.com (hiryu23) Date: Fri, 13 Jul 2012 01:24:09 -0700 (PDT) Subject: [Freeswitch-users] Subscribe for MWI In-Reply-To: References: <9D1F6AAA-D64F-45BC-A70D-C6E469D38C30@5ninesolutions.com> <8b33f052-3895-4252-8f47-6ce672e810c6@blur> <850DE19F-0F8B-4E4F-9B22-534311287478@5ninesolutions.com> <1338282624250-7579201.post@n2.nabble.com> <1338282786982-7579202.post@n2.nabble.com> Message-ID: <1342167849415-7580806.post@n2.nabble.com> Thanks Spencer. May i know how do you write the contact header before passing it to FreeSwitch? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Subscribe-for-MWI-tp7557448p7580806.html Sent from the freeswitch-users mailing list archive at Nabble.com. From getallad at gmail.com Fri Jul 13 14:04:15 2012 From: getallad at gmail.com (Sergei) Date: Fri, 13 Jul 2012 13:04:15 +0300 Subject: [Freeswitch-users] FreeSwitch Failover Question In-Reply-To: <040201cd5f9b$49f7d240$dde776c0$@bizfocused.com> References: <025801cd5f7a$b19fa150$14dee3f0$@bizfocused.com> <040201cd5f9b$49f7d240$dde776c0$@bizfocused.com> Message-ID: <00e701cd60de$dd2fb570$978f2050$@gmail.com> I could also recommend LinuxHA product - http://www.linux-ha.org . It allows to share virtual IP between servers, and failover this IP in case of primary server failure. Still, there could be issues with registrations sync J -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120713/79e2b6cc/attachment.html From peter.olsson at visionutveckling.se Fri Jul 13 15:48:14 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 13 Jul 2012 11:48:14 +0000 Subject: [Freeswitch-users] trunk_1 registration failed (timeout) In-Reply-To: References: , Message-ID: <52D2F757-D97B-495D-AC6F-E170DD34FD28@visionutveckling.se> Maybe a local firewall (iptables) that is blocking? /Peter 12 jul 2012 kl. 19:10 skrev "Aaron Berndsen" : > My freeswitch configuration was working fine until I upgraded to > ubuntu 12.04. Now sofia status indicates that registering trunk_1 to > my sip provider times out: > > 2012-07-12 09:02:51.252018 [WARNING] sofia_reg.c:449 Timeout Registering trunk_1 > 2012-07-12 09:02:53.252075 [WARNING] sofia_reg.c:468 trunk_1 Failed > Registration [908], setting retry to 45 seconds. > > A major clue is that sofia_dig on the ipv4 of my sip provider returns > instantly, while if I use the dns hostname it takes about 3 minutes to > return the information. > freeswitch at internal> sofia_dig 64.254.145.209 > Preference Weight Transport Port Address > ================================================================================ > 1 1.000 udp 5060 64.254.145.209 > 2 1.000 tcp 5060 64.254.145.209 > > Ubuntu 12.04 introduced dns-cacheing, but I have tried turning this > off (comment out dns=dnsmasq in > /etc/NetworkManager/NetworkManager.conf) and the problem persists. > > Any tips or suggestions would be helpful. I should point out that my > system is behind a router, though I've had the FS configuration > working with NAT before. > > Cheers, > -Aaron > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:4fff019832761881313408! > From steveayre at gmail.com Fri Jul 13 16:30:38 2012 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 13 Jul 2012 13:30:38 +0100 Subject: [Freeswitch-users] One Way RTP (no NAT) - Freeswitch sends RTP to the remote source port instead of SDP negotiated destination port (with disable_rtp_auto_adjust=true) In-Reply-To: References: Message-ID: <53970340-F914-470A-8BC8-446489102D5B@gmail.com> Can you supply a pastebin of adebug log of the call? Steve on iPhone On 12 Jul 2012, at 21:39, Adelia C. wrote: > We're testing a feature that requires an extra Application Server (B2BUA) on our call path, on the egress side of the call from FreeSwitch. > With the extra AS, we have only one-way RTP (Caller -> Callee). Without, RTP is fine. > We narrowed it down to FS enforcing symmetric RTP while AS only works with asymmetric RTP. We cannot change the behavior of the AS at this time. > > Call: > > Caller - >NTW -> SBC -> FS -> AS -> SBC -> NTW -> Callee. > > > RTP streams, as negotiated (SDP): > > > Caller - > SBCOut:50460 ->FSIn:29526 -> FSOut:17818 -> ASIn:8080 -> ASOutB:19040 -> SBCOut:49162 -> Callee (can't hear a thing) > > Callee -> SBCOut:49162 ->ASIn:8082 -> ASOut:19000 -> FSIn:17818 -> FSOut:29526 -> SBCIn:50460 -> Caller (good RTP) > > > > RTP streams, as sent/receiced: > > FS negotiates 8080 as the RTP destination to AS in the "caller - > callee" direction. For about .5s, that's how the RTP flows : FSOut:17818 -> ASIn:8080 > FS negotiates 19000 as the RTP source from AS in the "callee -> caller" direction. This is how the RTP flows for the duration of the call.: ASOut:19000 -> FSIn:17818 > FS switches its RTP stream in the "caller - > callee" direction ~.5s into the call : FSOut:17818 -> ASIn:19000. [ASIn listens on 8080, the caller - > called RTP is dead.] > > All this from Wireshark trace. > > I tried the following, with no success: > > 1. Added to MyFSApp.xml and dialplan\public.xml: > > > Restarted, FS, same result. > > 2. Commented out > > > > Restarted, FS, same result. > > 3. Started FS with -nonat, then nonatmap > Same result. > > Other info : > - Our App on FS runs in B2BUA mode, doesn't proxy the call. We answers the incoming leg before initiating the outgoing leg. I can attach the Wireshark is I didn't provide enough info - I have both FS and AS pcaps. > - Our FS code about 2 weeks old. > > What else can I try? Any other settings I can enable/disable? Can I enable extra RTP logs on FreeSwitch for debug? > > Thanks, > A.C. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120713/6baf5120/attachment-0001.html From steveayre at gmail.com Fri Jul 13 16:39:20 2012 From: steveayre at gmail.com (Steven Ayre) Date: Fri, 13 Jul 2012 13:39:20 +0100 Subject: [Freeswitch-users] FreeSwitch Failover Question In-Reply-To: <00e701cd60de$dd2fb570$978f2050$@gmail.com> References: <025801cd5f7a$b19fa150$14dee3f0$@bizfocused.com> <040201cd5f9b$49f7d240$dde776c0$@bizfocused.com> <00e701cd60de$dd2fb570$978f2050$@gmail.com> Message-ID: <44E1EE29-ABEC-429A-A499-251AA96083D4@gmail.com> I'm currently in the process of setting up a corosync/pacemaker install to provide a virtual ip and IPVS routed mode to send the sip packets to a FS server. FS hosts have the virtual ip bound to lo:0, and a sip profile listening on the virtual ip (allows FS to receive packets from the load balancer on ingress, on egress the packets from FS are sent directly to the endpoint with the correct ip, reducing load on the load balancer. Being on lo:0 prevents FS answering ARP requests. rtp-ip is the real FS address, so rtp bypasses the load balancer reducing its load and ensures RTP goes to the correct FS host. mod_sofia uses ODBC to use a shared database, MySQL Cluster (NDB) for HA. That means inbound registrations are visible from all FS servers regardless of which host handled the REGISTER. Outbound registrations (gateways) will be trickier as each FS host will currently need to register separately. Steve on iPhone On 13 Jul 2012, at 11:04, "Sergei" wrote: > I could also recommend LinuxHA product - http://www.linux-ha.org. It allows to share virtual IP between servers, and failover this IP in case of primary server failure. Still, there could be issues with registrations sync J > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120713/1ff505e1/attachment.html From wstephen80 at gmail.com Fri Jul 13 18:26:50 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Fri, 13 Jul 2012 16:26:50 +0200 Subject: [Freeswitch-users] FreeSwitch Failover Question In-Reply-To: <44E1EE29-ABEC-429A-A499-251AA96083D4@gmail.com> References: <025801cd5f7a$b19fa150$14dee3f0$@bizfocused.com> <040201cd5f9b$49f7d240$dde776c0$@bizfocused.com> <00e701cd60de$dd2fb570$978f2050$@gmail.com> <44E1EE29-ABEC-429A-A499-251AA96083D4@gmail.com> Message-ID: I'm using a SIP proxy acting as load balancer installed in a Cloud (high availability) in front of a pool of FS Gateways that handle media. I don't have phone registered so the SIP proxy is a FS in "bypass_media=true" and the balancer is a FS module that handles Gateway load and fail. Stephen On Fri, Jul 13, 2012 at 2:39 PM, Steven Ayre wrote: > I'm currently in the process of setting up a corosync/pacemaker install to > provide a virtual ip and IPVS routed mode to send the sip packets to a FS > server. > > FS hosts have the virtual ip bound to lo:0, and a sip profile listening on > the virtual ip (allows FS to receive packets from the load balancer on > ingress, on egress the packets from FS are sent directly to the endpoint > with the correct ip, reducing load on the load balancer. Being on lo:0 > prevents FS answering ARP requests. > > rtp-ip is the real FS address, so rtp bypasses the load balancer reducing > its load and ensures RTP goes to the correct FS host. > > mod_sofia uses ODBC to use a shared database, MySQL Cluster (NDB) for HA. > That means inbound registrations are visible from all FS servers regardless > of which host handled the REGISTER. > > Outbound registrations (gateways) will be trickier as each FS host will > currently need to register separately. > > Steve on iPhone > > > > On 13 Jul 2012, at 11:04, "Sergei" wrote: > > I could also recommend LinuxHA product - http://www.linux-ha.org. It > allows to share virtual IP between servers, and failover this IP in case of > primary server failure. Still, there could be issues with registrations > sync J **** > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120713/7ac8c322/attachment.html From mstockton at harqen.com Fri Jul 13 19:19:40 2012 From: mstockton at harqen.com (Matt Stockton) Date: Fri, 13 Jul 2012 10:19:40 -0500 Subject: [Freeswitch-users] FreeSwitch Failover Question In-Reply-To: References: <025801cd5f7a$b19fa150$14dee3f0$@bizfocused.com> <040201cd5f9b$49f7d240$dde776c0$@bizfocused.com> <00e701cd60de$dd2fb570$978f2050$@gmail.com> <44E1EE29-ABEC-429A-A499-251AA96083D4@gmail.com> Message-ID: Hey Stephen - I have a question about your setup for using FS as a SIP proxy with bypass_media. I have a similar need where I dont need registration, only inbound and bound calls from/to the PSTN --- Right now I have OpenSips sitting in front of a pool of FS, but if I could have the same setup and just use another FS, this would be preferable. My FS Pool have public IPs so I have no NAT issues with RTP, its just that the SIP signaling for incoming and outgoing calls have to come from a single IP. Are you using the proxy as I described above? If so, I'm interested to hear more details on how you have things setup. Thanks! On Fri, Jul 13, 2012 at 9:26 AM, Stephen Wilde wrote: > I'm using a SIP proxy acting as load balancer installed in a Cloud (high > availability) in front of a pool of FS Gateways that handle media. > I don't have phone registered so the SIP proxy is a FS in > "bypass_media=true" and the balancer is a FS module that handles Gateway > load and fail. > > Stephen > > On Fri, Jul 13, 2012 at 2:39 PM, Steven Ayre wrote: > >> I'm currently in the process of setting up a corosync/pacemaker install >> to provide a virtual ip and IPVS routed mode to send the sip packets to a >> FS server. >> >> FS hosts have the virtual ip bound to lo:0, and a sip profile listening >> on the virtual ip (allows FS to receive packets from the load balancer on >> ingress, on egress the packets from FS are sent directly to the endpoint >> with the correct ip, reducing load on the load balancer. Being on lo:0 >> prevents FS answering ARP requests. >> >> rtp-ip is the real FS address, so rtp bypasses the load balancer >> reducing its load and ensures RTP goes to the correct FS host. >> >> mod_sofia uses ODBC to use a shared database, MySQL Cluster (NDB) for HA. >> That means inbound registrations are visible from all FS servers regardless >> of which host handled the REGISTER. >> >> Outbound registrations (gateways) will be trickier as each FS host will >> currently need to register separately. >> >> Steve on iPhone >> >> >> >> On 13 Jul 2012, at 11:04, "Sergei" wrote: >> >> I could also recommend LinuxHA product - http://www.linux-ha.org. It >> allows to share virtual IP between servers, and failover this IP in case of >> primary server failure. Still, there could be issues with registrations >> sync J **** >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120713/4733d234/attachment-0001.html From spencer at 5ninesolutions.com Fri Jul 13 19:29:02 2012 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Fri, 13 Jul 2012 08:29:02 -0700 Subject: [Freeswitch-users] Subscribe for MWI In-Reply-To: <1342167849415-7580806.post@n2.nabble.com> References: <9D1F6AAA-D64F-45BC-A70D-C6E469D38C30@5ninesolutions.com> <8b33f052-3895-4252-8f47-6ce672e810c6@blur> <850DE19F-0F8B-4E4F-9B22-534311287478@5ninesolutions.com> <1338282624250-7579201.post@n2.nabble.com> <1338282786982-7579202.post@n2.nabble.com> <1342167849415-7580806.post@n2.nabble.com> Message-ID: <11AAB3D9-F17F-4AC1-B66C-591CF93155D1@5ninesolutions.com> Here's how I did it: in the main route block of kamailio.cfg: if (is_method("SUBSCRIBE")) { subst('/^Contact:(.*)sip:(.*)@[a-zA-Z0-9.:]+(.*)$/Contact:\1sip:$au@$Ri: $Rp\3/ig'); $du = "sip:" + "" + ":" + ""; if (!t_relay()) { sl_reply_error(); } exit; } That wil rewrite the contact on SUBSCRIBEs to the public IP and port of the kamailio instance that received the message. You can then handle the NOTIFYs from FreeSWITCH with lookup("") and route them to the registered endpoint. On Jul 13, 2012, at 1:24 AM, hiryu23 wrote: > Thanks Spencer. > > May i know how do you write the contact header before passing it to > FreeSwitch? > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Subscribe-for-MWI-tp7557448p7580806.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mario_fs at mgtech.com Fri Jul 13 19:46:15 2012 From: mario_fs at mgtech.com (Mario G) Date: Fri, 13 Jul 2012 08:46:15 -0700 Subject: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B604514098@BLUPRD0711MB413.namprd07.prod.outlook.com> References: <-5316884151656024069@unknownmsgid> <1342036306.3587.7.camel@mythtv> <83D31466-D013-4078-A1A2-D95CC3FE1DEA@mgtech.com> <63B00DD1DA6A364E9F64A3A0BD2FE7B604512C92@BLUPRD0711MB413.namprd07.prod.outlook.com> <1342123353.3613.7.camel@mythtv> <63B00DD1DA6A364E9F64A3A0BD2FE7B604514098@BLUPRD0711MB413.namprd07.prod.outlook.com> Message-ID: I agree, I have friends who work at companies that provide embedded systems to the military and they only use SSDs that use low power to reduce heat and increase reliability. They say no HDD can match the overall reliability. As for speed, every article I read says even the cheapest SSDs beat the best HDDs. On Jul 12, 2012, at 2:43 PM, Michael Giagnocavo wrote: > 4 10K drives cannot do the IOPS of any remotely decent SSD. We're talking what, 600-800 IOPS max on 4 spindles versus 20,000 IOPS at the low end for a modern SSD? (Like a $200 one.) > > As far as lifetime, a decent server-class SSD should be fine. Where you draw the line between junky consumer and server-grade is somewhat open to discussion, but at least at the FusionIO end, I don't think the lifetime should be a concern. > > Don't let cheap SSD vendors with crap firmware colour an entire class of technology. > > -Michael > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Todd Bailey > Sent: Thursday, July 12, 2012 2:03 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Anyone want my SSD tips on the FS Wiki? > > I'm using 4 SAS 2.5 73 gb 10K rpm drives raid 5 (205 gb formatted total) on a HP P600 controller The machine boots faster that a single 64gb ssd > device used briefly for testing. Just as fast, quiet, cheap, reliable > and long life. Best of both worlds. > > > > On Thu, 2012-07-12 at 16:29 +0000, Michael Giagnocavo wrote: >> Just as a side note for those that have not heard: Once you go SSD on >> your work machine, you can never go back. My Windows 7 laptop boots >> faster than my Android phone. >> >> -Michael >> >> >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Mario G >> Sent: Thursday, July 12, 2012 10:16 AM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Anyone want my SSD tips on the FS >> Wiki? >> >> >> >> >> Ok been working on it all day yesterday and should be done by Monday, >> will post a new subject when done. >> >> >> >> >> MC - I did not put the SSD into my FS Mac mini, it was another Linux >> machine. I spent months learning beforehand because I was thinking >> about it then I got a free Intel 320 160G. I just thought some of the >> FS folks might like to see what I did and learned. I replaced a WDC >> Velociraptor and this is WAAYY FASTER! >> >> >> >> >> On Jul 12, 2012, at 5:17 AM, Anita Hall wrote: >> >> >> >> >> Count me in! We are using SSD in an embedded system. >> >> regards, >> Anita >> >> >> >> >> On Thu, Jul 12, 2012 at 5:58 AM, Abdul Basit >> wrote: >> >> i m definitely interested and appreciate you willingness of knowledge >> sharing. Love hardware capacity, scalability and long life so waiting >> for the information you collected. >> >> >> >> >> -- >> >> >> Regards, >> >> >> >> >> >> Abdul Basit >> >> >> >> >> On Thu, Jul 12, 2012 at 1:22 AM, Muhammad Naseer Bhatti >> wrote: >> >> Depends on what brand the drive is. We are using a lot of SSDs in a >> huge enterprise environment and never faced issues. On average each >> SSD is giving us around 2000 IOPS with a combine output of around >> 32000 IOPS and we will be upgrading to 32 soon. Currently have 16. >> >> For smaller scale, I am using one in my Macbook, and never had any >> issues. Reaches upto 5000 IOPS and data transfer rate of around 200+ >> MBps. Have been using a quite a few in the servers running for FS. >> They work very well and never deceived. Surely, the thread is a good >> one, if the OP posts the draft :) >> >> Thanks. >> >> >> On Wed, Jul 11, 2012 at 10:51 PM, Todd Bailey >> wrote: >>> >>> Or 4 million micro seconds what ever comes first. >>> >>> I tried using a 90 gb ssd on linux and being curious, ran >>> phoronix-test-suite against the drive and with 15 minutes the drive >> was >>> non operational, rma'd the device and tried again, same results. >>> >>> returned for refund and never looked back... >>> >>> SSD devices may be fast but in a server environment reliability is >> key. >>> currently running raid 5 across the enterprise. >>> >>> >>> On Wed, 2012-07-11 at 12:15 -0400, Chris Ferreira wrote: >>>> I would most definitely be interested in this. I have been using >> SSD's >>>> for a number of different applications and never had a problem. Am >> I >>>> missing something? And you talk about increasing its life? Most of >> the >>>> SSDs I have used are rated at about 4 Million Hours which is like >> 456 >>>> Years (which I don't believe). >>>> >>>> >>>> >>>> Very interested to hear what you have to say. >>>> >>>> >>>> >>>> Thanks, >>>> >>>> -Chris >>>> >>>> ___________________ >>>> Mobile Reply >>>> >>>> On Jul 11, 2012, at 11:39 AM, Mario G wrote: >>>> >>>>> I recently spent a lot of time learning how to keep an SSD fast >> and increase its life. It took a ton of research and I combined a lot >> of info from several sources to install the SSD. Turns out, just >> installing an SSD and using it as an HDD without doing anything else >> is not a good idea. Although the topic is not specific to FreeSwitch I >> am wondering how many out there would like me to put together a FS >> wiki page fr tuning an SSD. It's a lot of work so I won't do it unless >> there are enough requests or the developers want it. >>>>> Mario G >>>>> >> ______________________________________________________________________ >> ___ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> Join Us At ClueCon - Aug 7-9, 2012 >>>>> >>>>> FreeSWITCH-users 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 >>>> >>>> >> ______________________________________________________________________ >> ___ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users 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 >>>> >>> >>> >>> >>> >> ______________________________________________________________________ >> ___ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users 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 >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users 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 >> >> >> >> >> >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users 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 >> >> >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users 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 >> >> >> >> >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From wstephen80 at gmail.com Fri Jul 13 20:04:56 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Fri, 13 Jul 2012 18:04:56 +0200 Subject: [Freeswitch-users] FreeSwitch Failover Question In-Reply-To: References: <025801cd5f7a$b19fa150$14dee3f0$@bizfocused.com> <040201cd5f9b$49f7d240$dde776c0$@bizfocused.com> <00e701cd60de$dd2fb570$978f2050$@gmail.com> <44E1EE29-ABEC-429A-A499-251AA96083D4@gmail.com> Message-ID: In my setup I don't have the necessity to have one IP for inbound/outbound calls so I have 1 IP for inbound calls (the ip of FS in bypass_media) and 'n' IP for outbound calls. But I don't see any difficult to implement your configuration (the FS that handle media can send the invite to the FS in bypass_media instead of to send them directly to providers). I have all FS with public IP and also, to increment the cps, I have all FS running with '-nosql'. My frontend FS forward the calls with a dialplan like: where "mymodule" write channel variable "server" based on load/fail. For the load I have a script in each FS backend that uses "dstat" to get cpu load. For the fail I have a script that uses "sipp" to ping each FS with a sip options. On Fri, Jul 13, 2012 at 5:19 PM, Matt Stockton wrote: > Hey Stephen - > > I have a question about your setup for using FS as a SIP proxy with > bypass_media. I have a similar need where I dont need registration, only > inbound and bound calls from/to the PSTN --- Right now I have OpenSips > sitting in front of a pool of FS, but if I could have the same setup and > just use another FS, this would be preferable. > > My FS Pool have public IPs so I have no NAT issues with RTP, its just that > the SIP signaling for incoming and outgoing calls have to come from a > single IP. > > Are you using the proxy as I described above? If so, I'm interested to > hear more details on how you have things setup. > > Thanks! > > > On Fri, Jul 13, 2012 at 9:26 AM, Stephen Wilde wrote: > >> I'm using a SIP proxy acting as load balancer installed in a Cloud (high >> availability) in front of a pool of FS Gateways that handle media. >> I don't have phone registered so the SIP proxy is a FS in >> "bypass_media=true" and the balancer is a FS module that handles Gateway >> load and fail. >> >> Stephen >> >> On Fri, Jul 13, 2012 at 2:39 PM, Steven Ayre wrote: >> >>> I'm currently in the process of setting up a corosync/pacemaker install >>> to provide a virtual ip and IPVS routed mode to send the sip packets to a >>> FS server. >>> >>> FS hosts have the virtual ip bound to lo:0, and a sip profile listening >>> on the virtual ip (allows FS to receive packets from the load balancer on >>> ingress, on egress the packets from FS are sent directly to the endpoint >>> with the correct ip, reducing load on the load balancer. Being on lo:0 >>> prevents FS answering ARP requests. >>> >>> rtp-ip is the real FS address, so rtp bypasses the load balancer >>> reducing its load and ensures RTP goes to the correct FS host. >>> >>> mod_sofia uses ODBC to use a shared database, MySQL Cluster (NDB) for >>> HA. That means inbound registrations are visible from all FS servers >>> regardless of which host handled the REGISTER. >>> >>> Outbound registrations (gateways) will be trickier as each FS host will >>> currently need to register separately. >>> >>> Steve on iPhone >>> >>> >>> >>> On 13 Jul 2012, at 11:04, "Sergei" wrote: >>> >>> I could also recommend LinuxHA product - http://www.linux-ha.org. It >>> allows to share virtual IP between servers, and failover this IP in case of >>> primary server failure. Still, there could be issues with registrations >>> sync J **** >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120713/0ca3b16e/attachment-0001.html From toddb at toddbailey.net Fri Jul 13 20:38:12 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Fri, 13 Jul 2012 09:38:12 -0700 Subject: [Freeswitch-users] Polycom IP501 In-Reply-To: References: <1341813422.3767.16.camel@mythtv> Message-ID: <1342197492.3512.18.camel@mythtv> Thanks, I'm working through the wiki and it's fairly cryptic. The tftp boot server work right off the bat and once I got past getting the phone to work, once I connected and powered up it went out and got the latest code. So now the phone is running 3.1.8 Initially, I built all the config files as outlined in the wiki, but the phone never auto provisioned, eventually I just went into the web set on the phone and got minimal functionality working. One of my to-do items is to get full functionality for phone services via tftp services, so I can add additional lines and phones, and a nice to have feature is perhaps a few side apps when I can have misc. apps available like weather, traffic, streaming music, etc. I'm not sure what the capabilities of this phone are so it all a research and work in progress project at this point. perhaps I'll find some prebuilt apps so I don't have to spend a lot of time reinventing things. For now I can send and receive calls and nothing else, but it's a start. Buyers remorse? yes I would have liked a high res color screen, and newer & more capabilities but considering the total cost of $25 bucks, it's workable starting point. On Mon, 2012-07-09 at 09:20 +0300, Yehavi Bourvine wrote: > Hell Todd, > > If you've created the right config files then it should work > immediately. However, if you haven't used Polycoms so far I doubt you > have the perfect config files. It also depends on which additional > functionality you want (like extended function keys, etc.). > > Polycom's guide includes everything, but it is sort of a refference > manual and not a user guide. > > I suggest you start with the WIKI as it have examples. > > __Yehavi: > > > 2012/7/9 Todd Bailey > Hey all, I just purchased this phone ( ebay $35 total) and > should be > here in a few days. > I think I got a boot server (tftp) all set up so the phone > will have a > place to get the latest boot code. > > I've read through most of the FS user docs on this phone and > it seems > like it should be plug and play after I get the configuration > files > updated and in place. I not sure what to expect, for the > initial > install. Basic functionality and maybe a few higher features > or does the > fun begin after the phone is installed? > > This phone is intended as a beta test, if it works ok, > meaning plug in > and it works and advanced features don't take days or weeks to > sort out, > I'll be adding a few more. > > Any one have experience with this particular phone on FS and > any issues > or gotcha's to be aware of? > > > thanks > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From adrottenberg at gmail.com Fri Jul 13 21:34:52 2012 From: adrottenberg at gmail.com (Duvid Rottenberg) Date: Fri, 13 Jul 2012 13:34:52 -0400 Subject: [Freeswitch-users] Username in SUBSCRIBE Request URI Message-ID: I am using embedded freeswitch as a softphone client and I am trying to subscribe to call-info on the server, (see config below) but the server is responding with a 481 Call/Transaction Does not exist. I compared the freeswitch SIP messages with SIP messages sent by a polycom phone for this feature and I noticed that freeswitch doesn't send the username in the request line. I think that this is causing the 481 response. Polycom Version: SUBSCRIBE sip:user at server:5060;transport=udp SIP/2.0 Freeswitch: SUBSCRIBE sip:server:5060;transport=udp SIP/2.0 Below is my gateway configuration Is there any way to tell freeswitch to include the username in the request line? Thank You, Duvid Rottenberg -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120713/720c0b0a/attachment.html From hexade at hotmail.com Fri Jul 13 21:37:25 2012 From: hexade at hotmail.com (Adelia C.) Date: Fri, 13 Jul 2012 13:37:25 -0400 Subject: [Freeswitch-users] One Way RTP (no NAT) - Freeswitch sends RTP to the remote source port instead of SDP negotiated destination port (with disable_rtp_auto_adjust=true) In-Reply-To: <53970340-F914-470A-8BC8-446489102D5B@gmail.com> References: , <53970340-F914-470A-8BC8-446489102D5B@gmail.com> Message-ID: Thank you Steve - here it is - http://pastebin.freeswitch.org/19510. Fresh call, new ports. Notice this line, this is what I want to stop: 2012-07-13 10:18:42.935692 [INFO] switch_rtp.c:3133 Auto Changing port from 10.3.212.105:8024 to 10.3.212.105:12016 Thank you, A.C. CC: freeswitch-users at lists.freeswitch.org From: steveayre at gmail.com Date: Fri, 13 Jul 2012 13:30:38 +0100 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] One Way RTP (no NAT) - Freeswitch sends RTP to the remote source port instead of SDP negotiated destination port (with disable_rtp_auto_adjust=true) Can you supply a pastebin of adebug log of the call? Steve on iPhone On 12 Jul 2012, at 21:39, Adelia C. wrote: We're testing a feature that requires an extra Application Server (B2BUA) on our call path, on the egress side of the call from FreeSwitch. With the extra AS, we have only one-way RTP (Caller -> Callee). Without, RTP is fine. We narrowed it down to FS enforcing symmetric RTP while AS only works with asymmetric RTP. We cannot change the behavior of the AS at this time. Call: Caller - >NTW -> SBC -> FS -> AS -> SBC -> NTW -> Callee. RTP streams, as negotiated (SDP): Caller - > SBCOut:50460 ->FSIn:29526 -> FSOut:17818 -> ASIn:8080 -> ASOutB:19040 -> SBCOut:49162 -> Callee (can't hear a thing) Callee -> SBCOut:49162 ->ASIn:8082 -> ASOut:19000 -> FSIn:17818 -> FSOut:29526 -> SBCIn:50460 -> Caller (good RTP) RTP streams, as sent/receiced: FS negotiates 8080 as the RTP destination to AS in the "caller - > callee" direction. For about .5s, that's how the RTP flows : FSOut:17818 -> ASIn:8080 FS negotiates 19000 as the RTP source from AS in the "callee -> caller" direction. This is how the RTP flows for the duration of the call.: ASOut:19000 -> FSIn:17818 FS switches its RTP stream in the "caller - > callee" direction ~.5s into the call : FSOut:17818 -> ASIn:19000. [ASIn listens on 8080, the caller - > called RTP is dead.] All this from Wireshark trace. I tried the following, with no success: 1. Added to MyFSApp.xml and dialplan\public.xml: Restarted, FS, same result. 2. Commented out Restarted, FS, same result. 3. Started FS with -nonat, then nonatmap Same result. Other info : - Our App on FS runs in B2BUA mode, doesn't proxy the call. We answers the incoming leg before initiating the outgoing leg. I can attach the Wireshark is I didn't provide enough info - I have both FS and AS pcaps. - Our FS code about 2 weeks old. What else can I try? Any other settings I can enable/disable? Can I enable extra RTP logs on FreeSwitch for debug? Thanks, A.C. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120713/f7366ae3/attachment-0001.html From sdevoy at bizfocused.com Sat Jul 14 02:35:24 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Fri, 13 Jul 2012 18:35:24 -0400 Subject: [Freeswitch-users] _sip._udp SRV DNS records on Windows 2008 R2 server Message-ID: <103e01cd6147$cb652c30$622f8490$@bizfocused.com> Hi all, This is a Windows DNS Server question. I have added SRV records for _sip._udp.bizfocused.com, even though _sip is not one of the stock choices in the dropdown list from the add SRV record dialog. I can see them in DNS console listed under my domain: bizfocused.com +_udp (2 _sip records show up here) However, I cannot find any tool that can successfully request/display those values (like nslookup). And of course when I turn on "Use DNS SRV" on my phone it just hangs on "CHECKING DNS". Any ideas for hosting _sip._udp SRV records would be greatly appreciated. Do I have to move my DNS to a UNIX box or DNS Service provider? Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120713/fbd7788c/attachment.html From ash at url.net.au Sat Jul 14 03:41:47 2012 From: ash at url.net.au (Ashley Breeden) Date: Sat, 14 Jul 2012 09:41:47 +1000 Subject: [Freeswitch-users] _sip._udp SRV DNS records on Windows 2008 R2 server In-Reply-To: <103e01cd6147$cb652c30$622f8490$@bizfocused.com> References: <103e01cd6147$cb652c30$622f8490$@bizfocused.com> Message-ID: <0DE5FB25-9E43-43FB-81C1-B045E7F89668@url.net.au> Hi Sean, I don't know a lot about the windows DNS server but you should be able to display the values using: nslookup -q=SRV _sip._udp.bizfocussed.com You should see something like this: Non-authoritative answer: _sip._udp.bizfocussed.com service = 50 0 5060 YOUR SIP SERVER. _sip._udp.bizfocussed.com service = 0 0 5060 YOUR SIP SERVER. Cheers, Ash. On 14/07/2012, at 8:35 AM, Sean Devoy wrote: > Hi all, > > This is a Windows DNS Server question. I have added SRV records for _sip._udp.bizfocused.com, even though _sip is not one of the stock choices in the dropdown list from the add SRV record dialog. I can see them in DNS console listed under my domain: > bizfocused.com > +_udp (2 _sip records show up here) > > However, I cannot find any tool that can successfully request/display those values (like nslookup). > > And of course when I turn on ?Use DNS SRV? on my phone it just hangs on ?CHECKING DNS?. > > Any ideas for hosting _sip._udp SRV records would be greatly appreciated. Do I have to move my DNS to a UNIX box or DNS Service provider? > > Thanks, > Sean > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120714/c6f946cf/attachment.html From andrew at cassidywebservices.co.uk Sat Jul 14 03:46:27 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Sat, 14 Jul 2012 00:46:27 +0100 Subject: [Freeswitch-users] _sip._udp SRV DNS records on Windows 2008 R2 server In-Reply-To: <103e01cd6147$cb652c30$622f8490$@bizfocused.com> References: <103e01cd6147$cb652c30$622f8490$@bizfocused.com> Message-ID: That's strange, and you shouldn't have to as MS Lync uses _siptls._tcp SRV records. Try a Wireshark capture, make sure the request is being responded to. Can you query other records ok? On 13 July 2012 23:35, Sean Devoy wrote: > Hi all,**** > > ** ** > > This is a Windows DNS Server question. I have added SRV records for _sip* > .*_udp.bizfocused.com, even though _sip is not one of the stock choices > in the dropdown list from the add SRV record dialog. I can see them in DNS > console listed under my domain:**** > > bizfocused.com**** > > +_udp (2 _sip records show up here)**** > > ** ** > > However, I cannot find any tool that can successfully request/display > those values (like nslookup).**** > > ** ** > > And of course when I turn on ?Use DNS SRV? on my phone it just hangs on > ?CHECKING DNS?.**** > > ** ** > > Any ideas for hosting _sip*.*_udp SRV records would be greatly > appreciated. Do I have to move my DNS to a UNIX box or DNS Service > provider?**** > > ** ** > > Thanks,**** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120714/894fac8d/attachment.html From sdevoy at bizfocused.com Sat Jul 14 09:29:20 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sat, 14 Jul 2012 01:29:20 -0400 Subject: [Freeswitch-users] _sip._udp SRV DNS records on Windows 2008 R2 server In-Reply-To: References: <103e01cd6147$cb652c30$622f8490$@bizfocused.com> Message-ID: <10a701cd6181$9ed9c600$dc8d5200$@bizfocused.com> Thanks. My problem seems to really be with my lack of knowledge of nslookup In fact: Nslookup Set querytype=srv _sip._udp.bizfocused.com worked fine The VERY LONG boot time oin the phone is sucky though. However, since the client phones are all cisco spa504g's they have a proxy field and an alternate proxy to login to if the primary fails! That is what I was looking for anyway. It appears worse case the phone will have to be rebooted to come back online. Thanks for the feedback though. Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Andrew Cassidy Sent: Friday, July 13, 2012 7:46 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] _sip._udp SRV DNS records on Windows 2008 R2 server That's strange, and you shouldn't have to as MS Lync uses _siptls._tcp SRV records. Try a Wireshark capture, make sure the request is being responded to. Can you query other records ok? On 13 July 2012 23:35, Sean Devoy wrote: Hi all, This is a Windows DNS Server question. I have added SRV records for _sip._udp.bizfocused.com, even though _sip is not one of the stock choices in the dropdown list from the add SRV record dialog. I can see them in DNS console listed under my domain: bizfocused.com +_udp (2 _sip records show up here) However, I cannot find any tool that can successfully request/display those values (like nslookup). And of course when I turn on "Use DNS SRV" on my phone it just hangs on "CHECKING DNS". Any ideas for hosting _sip._udp SRV records would be greatly appreciated. Do I have to move my DNS to a UNIX box or DNS Service provider? Thanks, Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Andrew Cassidy BSc (Hons) MBCS SSCA Managing Director T 03300 100 960 F 03300 100 961 E andrew at cassidywebservices.co.uk W www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120714/a607066e/attachment-0001.html From yehavi.bourvine at gmail.com Sat Jul 14 10:45:50 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sat, 14 Jul 2012 09:45:50 +0300 Subject: [Freeswitch-users] Polycom IP501 In-Reply-To: <1342197492.3512.18.camel@mythtv> References: <1341813422.3767.16.camel@mythtv> <1342197492.3512.18.camel@mythtv> Message-ID: Hi, A few things to follow: - Make sure the phone is set to use DHCP. - Make sure in the same configuration screen that it honors option 66. - Here is a fraction from my dhcpd.conf file: option domain-name "phones.huji.ac.il"; option domain-name-servers 10.64.0.3, 10.64.0.4; ddns-update-style ad-hoc; default-lease-time 3600; max-lease-time 7200; authoritative; option option-66 code 66 = string; option option-67 code 67 = string; option option-128 code 128 = string; option option-160 code 160 = string; subnet 10.64.1.0 netmask 255.255.255.0 { default-lease-time 3600; max-lease-time 7200; option routers 10.64.1.1; option ntp-servers 10.64.0.3; range 10.64.1.250 10.64.1.252; range dynamic-bootp 10.64.1.253 10.64.1.254; } host phone-cc-80601-401 { default-lease-time 600000; hardware ethernet 00:04:f2:14:7a:9d; fixed-address 10.64.1.20; option option-66 "tftp://10.64.0.3/POLYCOM_OLD_501/"; } Next, try to boot and run TCPDUMP in parallel. Send me its output (preferably the CAP file). Regards, __Yehavi: 2012/7/13 Todd Bailey > Thanks, > > I'm working through the wiki and it's fairly cryptic. > > The tftp boot server work right off the bat and once I got past getting > the phone to work, once I connected and powered up it went out and got > the latest code. So now the phone is running 3.1.8 > > Initially, I built all the config files as outlined in the wiki, but the > phone never auto provisioned, eventually I just went into the web set on > the phone and got minimal functionality working. > > One of my to-do items is to get full functionality for phone services > via tftp services, so I can add additional lines and phones, and a nice > to have feature is perhaps a few side apps when I can have misc. apps > available like weather, traffic, streaming music, etc. > > I'm not sure what the capabilities of this phone are so it all a > research and work in progress project at this point. > > perhaps I'll find some prebuilt apps so I don't have to spend a lot of > time reinventing things. > > For now I can send and receive calls and nothing else, but it's a > start. > > > Buyers remorse? yes I would have liked a high res color screen, and > newer & more capabilities but considering the total cost of $25 bucks, > it's workable starting point. > > > > On Mon, 2012-07-09 at 09:20 +0300, Yehavi Bourvine wrote: > > Hell Todd, > > > > If you've created the right config files then it should work > > immediately. However, if you haven't used Polycoms so far I doubt you > > have the perfect config files. It also depends on which additional > > functionality you want (like extended function keys, etc.). > > > > Polycom's guide includes everything, but it is sort of a refference > > manual and not a user guide. > > > > I suggest you start with the WIKI as it have examples. > > > > __Yehavi: > > > > > > 2012/7/9 Todd Bailey > > Hey all, I just purchased this phone ( ebay $35 total) and > > should be > > here in a few days. > > I think I got a boot server (tftp) all set up so the phone > > will have a > > place to get the latest boot code. > > > > I've read through most of the FS user docs on this phone and > > it seems > > like it should be plug and play after I get the configuration > > files > > updated and in place. I not sure what to expect, for the > > initial > > install. Basic functionality and maybe a few higher features > > or does the > > fun begin after the phone is installed? > > > > This phone is intended as a beta test, if it works ok, > > meaning plug in > > and it works and advanced features don't take days or weeks to > > sort out, > > I'll be adding a few more. > > > > Any one have experience with this particular phone on FS and > > any issues > > or gotcha's to be aware of? > > > > > > thanks > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120714/5d5ff459/attachment.html From peterwood.sd at gmail.com Sat Jul 14 04:12:33 2012 From: peterwood.sd at gmail.com (Wood Peter) Date: Fri, 13 Jul 2012 17:12:33 -0700 Subject: [Freeswitch-users] Using the freeswitch.spec to build RPM Message-ID: Hi, I'm using CentOS5 and I can successfully compile and install freeswitch using the latest Git clone. But, I'm not very successful building an RPM with the freeswitch.spec file. I couldn't find any documentation or instructions how to build the RPM. This will be a great start for me instead of sending long error messages to the list. Any pointers are appreciated. Thank you Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120713/04344f04/attachment.html From peterwood.sd at gmail.com Sat Jul 14 04:23:18 2012 From: peterwood.sd at gmail.com (Wood Peter) Date: Fri, 13 Jul 2012 17:23:18 -0700 Subject: [Freeswitch-users] Failed to compile on CentOS5 Message-ID: Latest Git clone fails to compile on CentOS5. The compile error out with this: ... Compiling src/switch_regex.c ... Compiling src/switch_rtp.c ... src/switch_rtp.c: In function 'read_rtp_packet': src/switch_rtp.c:2950: error: 'switch_rtp_t' has no member named 'rtp_recv_msg' src/switch_rtp.c: In function 'read_rtcp_packet': src/switch_rtp.c:3069: error: 'switch_rtp_t' has no member named 'rtp_recv_msg' make[1]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 make: *** [all] Error 2 Using the last commit from yesterday compiles successfully: git checkout ba6c404 Any idea what could be wrong? Thank you, Peter From peter.olsson at visionutveckling.se Sat Jul 14 12:08:24 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sat, 14 Jul 2012 08:08:24 +0000 Subject: [Freeswitch-users] Failed to compile on CentOS5 In-Reply-To: References: Message-ID: It should be fixed in latest git head already. Please try latest again. /Peter 14 jul 2012 kl. 09:54 skrev "Wood Peter" : > Latest Git clone fails to compile on CentOS5. > > The compile error out with this: > > ... > > Compiling src/switch_regex.c ... > > Compiling src/switch_rtp.c ... > > src/switch_rtp.c: In function 'read_rtp_packet': > > src/switch_rtp.c:2950: error: 'switch_rtp_t' has no member named 'rtp_recv_msg' > > src/switch_rtp.c: In function 'read_rtcp_packet': > > src/switch_rtp.c:3069: error: 'switch_rtp_t' has no member named 'rtp_recv_msg' > > make[1]: *** [libfreeswitch_la-switch_rtp.lo] Error 1 > > make: *** [all] Error 2 > > > Using the last commit from yesterday compiles successfully: git checkout ba6c404 > > Any idea what could be wrong? > > Thank you, > > Peter > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:5001227132761618476333! > From krice at freeswitch.org Sat Jul 14 19:38:26 2012 From: krice at freeswitch.org (Ken Rice) Date: Sat, 14 Jul 2012 10:38:26 -0500 Subject: [Freeswitch-users] Using the freeswitch.spec to build RPM In-Reply-To: Message-ID: You use it like any other spec file ... Rpmbuild it... you need to have all the support sources and such in place in the correct places on centos5... Theres some scripts in the scripts/ci directory that handle this on the FS Automatted build system On 7/13/12 7:12 PM, "Wood Peter" wrote: > Hi, > > I'm using CentOS5 and I can successfully compile and install freeswitch using > the latest Git clone. > > But, I'm not very?successful?building an RPM with the freeswitch.spec file. > > I couldn't find any documentation or instructions how to build the RPM. This > will be a great start for me instead of sending long error messages to the > list. > > Any pointers are appreciated. > > Thank you > Peter > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120714/f133b1aa/attachment-0001.html From andrew at cassidywebservices.co.uk Sat Jul 14 21:44:38 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Sat, 14 Jul 2012 18:44:38 +0100 Subject: [Freeswitch-users] Outgoing Registration Contact Message-ID: Hi all, is it possible to force FreeSWITCH to use an FQDN in the contact field of an outgoing registration? Say for example you wanted to haev a trunk that you have to register terminate at multiple public IP addresses. Is that something that can be done with FreeSWITCH at the moment? It's not a massive deal, I'll probably end up putting in SBC(s) at some point. -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120714/80c0b30d/attachment.html From toddb at toddbailey.net Sun Jul 15 00:15:35 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Sat, 14 Jul 2012 13:15:35 -0700 Subject: [Freeswitch-users] Music on Hold overly loud and distorted. Message-ID: <1342296935.3512.47.camel@mythtv> Hi All, I'm still setting up FS and got a complaint from callers that music on hold is overly loud and distorted. But from an internal line, no issues. Could this be a setting issue with FS or perhaps the spa-31002 ata adapter? thanks From curriegrad2004 at gmail.com Sun Jul 15 03:23:23 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sat, 14 Jul 2012 16:23:23 -0700 Subject: [Freeswitch-users] Music on Hold overly loud and distorted. In-Reply-To: <1342296935.3512.47.camel@mythtv> References: <1342296935.3512.47.camel@mythtv> Message-ID: Did you supply your own MOH sounds? If you did, put them through audacity and attenuate the volume by at least 20dB down from full levels. On Sat, Jul 14, 2012 at 1:15 PM, Todd Bailey wrote: > Hi All, > > I'm still setting up FS and got a complaint from callers that music on > hold is overly loud and distorted. > > But from an internal line, no issues. Could this be a setting issue > with FS or perhaps the spa-31002 ata adapter? > > thanks > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From admin at blindi.net Sun Jul 15 12:53:22 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Sun, 15 Jul 2012 10:53:22 +0200 (CEST) Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR after about 62 transfers of a sesseion In-Reply-To: References: <1342296935.3512.47.camel@mythtv> Message-ID: Hi guys, i have setup a voicechat. In this chat, i have many extensions to transfer the uuser from confernce to conference and so on. or transfer back to a menu to enter selections and transfer next. For example: transfer mailbox logon, transfer in sendvoicemessage menu, leave a message, transfer back. and so on. These works fine: My problem: after about 62 transfers in the same session, the line break. How can i set the transferlimit higher please? Thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From peter.olsson at visionutveckling.se Sun Jul 15 14:55:58 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 15 Jul 2012 10:55:58 +0000 Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR after about 62 transfers of a sesseion In-Reply-To: References: <1342296935.3512.47.camel@mythtv> , Message-ID: <1FFF97C269757C458224B7C895F35F15134E74@cantor.std.visionutv.se> Set the channel variable "max_forwards" to a higher value, it should do the trick. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Thomas Hoellriegel [admin at blindi.net] Skickat: den 15 juli 2012 10:53 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] EXCHANGE_ROUTING_ERROR after about 62 transfers of a sesseion Hi guys, i have setup a voicechat. In this chat, i have many extensions to transfer the uuser from confernce to conference and so on. or transfer back to a menu to enter selections and transfer next. For example: transfer mailbox logon, transfer in sendvoicemessage menu, leave a message, transfer back. and so on. These works fine: My problem: after about 62 transfers in the same session, the line break. How can i set the transferlimit higher please? Thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc !DSPAM:5002839d32761280617049! From admin at blindi.net Sun Jul 15 16:38:16 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Sun, 15 Jul 2012 14:38:16 +0200 (CEST) Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR after about 62 transfers of a sesseion In-Reply-To: <1FFF97C269757C458224B7C895F35F15134E74@cantor.std.visionutv.se> References: <1342296935.3512.47.camel@mythtv> , <1FFF97C269757C458224B7C895F35F15134E74@cantor.std.visionutv.se> Message-ID: Hi peter, > Set the channel variable "max_forwards" to a higher value, it should do the trick. The variable don.t working. I have only inboundcalls. my extension is: Dorf is the german name of the voicechat. fs ignore the channel variable why? thanks. > --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From peter.olsson at visionutveckling.se Sun Jul 15 17:27:24 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 15 Jul 2012 13:27:24 +0000 Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR after about 62 transfers of a sesseion In-Reply-To: References: <1342296935.3512.47.camel@mythtv> , <1FFF97C269757C458224B7C895F35F15134E74@cantor.std.visionutv.se>, Message-ID: <1FFF97C269757C458224B7C895F35F15134EDF@cantor.std.visionutv.se> Try dumping the channel variables in a few different places in the dialplan (execute application "info"). And see what's going on. The only thing that should be able to override max_forwards is if there's lots of reinvites going on (it's set from the SIP-attribute "Max-Forwards"). It's definately the correct variable to change (the variable "Max-Forwards" won't make a difference though, but I see you set them both). /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Thomas Hoellriegel [admin at blindi.net] Skickat: den 15 juli 2012 14:38 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] EXCHANGE_ROUTING_ERROR after about 62 transfers of a sesseion Hi peter, > Set the channel variable "max_forwards" to a higher value, it should do the trick. The variable don.t working. I have only inboundcalls. my extension is: Dorf is the german name of the voicechat. fs ignore the channel variable why? thanks. > --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc !DSPAM:5002b88532761569116435! From toddb at toddbailey.net Mon Jul 16 05:00:50 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Sun, 15 Jul 2012 18:00:50 -0700 Subject: [Freeswitch-users] programming/config Polycom ip501 Message-ID: <1342400450.3892.23.camel@mythtv> Hi All, I'm running a vanilla install of Fs and current have a hard phone attached - the ip501. tftp config files. The address of the phone is 192.168.1.23 & 00:04:F2:04:79:68 If I read the setup wiki correctly, I need to have created 3 different files 0004F2047968.cfg 0004F2047968-settings.cfg 0004F2047968-directory.cfg these don't appear to have any effect on phone settings or functionality, did I name these properly? and what if anything should I do with the files from the firmware zip file I downloaded: 000000000000.cfg 000000000000-directory~.xml phone1_318.cfg sip_318.cfg Speed dial (stored contacts) ? I was able to config the phone using the web interface but I'm running into several issues in getting various functions to work. how do I setup and store a contact to get to an outside line I have to dial 9 then send then dial the rest of the number. Needless to say , speed dial doesn't work, What sections do I need to study to get 9 + ac + phone number to work either manually or via a saved contact? Messages? how do I program the phone access voice mail? when I press "messages", I can select 1 of the 3 extensions after doing so it dials that extension instead of going to the voice mail. sorry for being verbose, it couldn't be helped. From amilkhanzada at gmail.com Mon Jul 16 07:57:26 2012 From: amilkhanzada at gmail.com (Amil) Date: Sun, 15 Jul 2012 20:57:26 -0700 Subject: [Freeswitch-users] Reloading mod_python dist-packages? Message-ID: So, I am writing some python scripts that are copied into the dist-packages directory of python. However, I am also using python scripts in the FreeSWITCH scripts directory that refer to these python scripts in the dist-packages. To re-copy the files, I need to run "sudo python setup_fs.py" each time I make a change. However, I also need to restart FreeSWITCH because I don't know of a way to reload the dist-packages used by mod_python. Any ideas? PS: This is the project I am referring to: < https://github.com/kheimerl/libvbts> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120715/6058f92a/attachment-0001.html From yehavi.bourvine at gmail.com Mon Jul 16 08:04:13 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Mon, 16 Jul 2012 07:04:13 +0300 Subject: [Freeswitch-users] programming/config Polycom ip501 In-Reply-To: <1342400450.3892.23.camel@mythtv> References: <1342400450.3892.23.camel@mythtv> Message-ID: Hello Todd, 1. You need a DHCP server to tell the phone where to get the configuration files from. I sent you an example in a private mail a fwe days ago. 2. Note that the file *mac-address.*cfg should point to the other files. 3. As I told you: send me a TCPDUMP of the boot process and then we might be able to help you. __Yehavi: 2012/7/16 Todd Bailey > Hi All, > > > I'm running a vanilla install of Fs and current have a hard phone > attached - the ip501. > > tftp config files. > > The address of the phone is 192.168.1.23 & 00:04:F2:04:79:68 > > If I read the setup wiki correctly, I need to have created 3 different > files > > 0004F2047968.cfg > 0004F2047968-settings.cfg > 0004F2047968-directory.cfg > > these don't appear to have any effect on phone settings or > functionality, did I name these properly? and what if anything should > I do with the files from the firmware zip file I downloaded: > 000000000000.cfg > 000000000000-directory~.xml > phone1_318.cfg > sip_318.cfg > > > Speed dial (stored contacts) ? > > I was able to config the phone using the web interface but I'm running > into several issues in getting various functions to work. > how do I setup and store a contact to get to an outside line I have to > dial 9 then send then dial the rest of the number. Needless to say , > speed dial doesn't work, > > What sections do I need to study to get 9 + ac + phone number to work > either manually or via a saved contact? > > > Messages? > > how do I program the phone access voice mail? > when I press "messages", I can select 1 of the 3 extensions after doing > so it dials that extension instead of going to the voice mail. > > sorry for being verbose, it couldn't be helped. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120716/d440df6d/attachment.html From vbvbrj at gmail.com Mon Jul 16 10:27:24 2012 From: vbvbrj at gmail.com (Vbvbrj) Date: Mon, 16 Jul 2012 09:27:24 +0300 Subject: [Freeswitch-users] Sound problems in voicemail. Message-ID: <5003B44C.50409@gmail.com> Hello. Searching for a resolve. I've found in default dialplan from distro in file conf/dialplan/default.xml, line 270: also on line 284 is: Changing line 270 to: and leaving line 284 as is, solves the problem with the sound when calling some extension which is not registered from 1000 to 1019. But problem with sounds persists when calling 1100 to 1119. Messages "Goodbay" is very distorted. Is this a bug or something else that "application="bridge" data="loopback/app=voicemail:default" distort sound and "application="voicemail" data="default" does not distort? From peter.olsson at visionutveckling.se Mon Jul 16 11:43:33 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 16 Jul 2012 07:43:33 +0000 Subject: [Freeswitch-users] Sound problems in voicemail. In-Reply-To: <5003B44C.50409@gmail.com> References: <5003B44C.50409@gmail.com> Message-ID: Are you on latest git head and are you using real hardware (no VMware etc)? /Peter 16 jul 2012 kl. 08:37 skrev "Vbvbrj" : > Hello. > > Searching for a resolve. I've found in default dialplan from distro in > file conf/dialplan/default.xml, line 270: > > > > also on line 284 is: > > > > Changing line 270 to: > > > > and leaving line 284 as is, solves the problem with the sound when > calling some extension which is not registered from 1000 to 1019. But > problem with sounds persists when calling 1100 to 1119. Messages > "Goodbay" is very distorted. > > Is this a bug or something else that "application="bridge" > data="loopback/app=voicemail:default" distort sound and > "application="voicemail" data="default" does not distort? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:5003b32d32761191015625! > From vbvbrj at gmail.com Mon Jul 16 11:47:22 2012 From: vbvbrj at gmail.com (Vbvbrj) Date: Mon, 16 Jul 2012 10:47:22 +0300 Subject: [Freeswitch-users] Sound problems in voicemail. In-Reply-To: References: <5003B44C.50409@gmail.com> Message-ID: <5003C70A.5070301@gmail.com> Yes, latest git and real hardware, dedicated debian server with only FS installed. But same problem is in VM too. On 16.07.2012 10:43, Peter Olsson wrote: > Are you on latest git head and are you using real hardware (no VMware etc)? > > /Peter > > 16 jul 2012 kl. 08:37 skrev "Vbvbrj" : > >> Hello. >> >> Searching for a resolve. I've found in default dialplan from distro in >> file conf/dialplan/default.xml, line 270: >> >> >> >> also on line 284 is: >> >> >> >> Changing line 270 to: >> >> >> >> and leaving line 284 as is, solves the problem with the sound when >> calling some extension which is not registered from 1000 to 1019. But >> problem with sounds persists when calling 1100 to 1119. Messages >> "Goodbay" is very distorted. >> >> Is this a bug or something else that "application="bridge" >> data="loopback/app=voicemail:default" distort sound and >> "application="voicemail" data="default" does not distort? From peter.olsson at visionutveckling.se Mon Jul 16 11:58:57 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 16 Jul 2012 07:58:57 +0000 Subject: [Freeswitch-users] Sound problems in voicemail. In-Reply-To: <5003C70A.5070301@gmail.com> References: <5003B44C.50409@gmail.com> , <5003C70A.5070301@gmail.com> Message-ID: <5CFE5DF4-8C62-4A4F-8914-8F8947FBD56A@visionutveckling.se> In that case, please submit a Jira report. Make sure to attach all the needed information (debug logs etc), and in this case, a pcap for the distorted audio might be good. /Peter 16 jul 2012 kl. 09:51 skrev "Vbvbrj" : > Yes, latest git and real hardware, dedicated debian server with only FS > installed. But same problem is in VM too. > > On 16.07.2012 10:43, Peter Olsson wrote: >> Are you on latest git head and are you using real hardware (no VMware etc)? >> >> /Peter >> >> 16 jul 2012 kl. 08:37 skrev "Vbvbrj" : >> >>> Hello. >>> >>> Searching for a resolve. I've found in default dialplan from distro in >>> file conf/dialplan/default.xml, line 270: >>> >>> >>> >>> also on line 284 is: >>> >>> >>> >>> Changing line 270 to: >>> >>> >>> >>> and leaving line 284 as is, solves the problem with the sound when >>> calling some extension which is not registered from 1000 to 1019. But >>> problem with sounds persists when calling 1100 to 1119. Messages >>> "Goodbay" is very distorted. >>> >>> Is this a bug or something else that "application="bridge" >>> data="loopback/app=voicemail:default" distort sound and >>> "application="voicemail" data="default" does not distort? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:5003c4b632761996510574! > From hexade at hotmail.com Mon Jul 16 18:34:31 2012 From: hexade at hotmail.com (Adelia C.) Date: Mon, 16 Jul 2012 10:34:31 -0400 Subject: [Freeswitch-users] One Way RTP (no NAT) - Freeswitch sends RTP to the remote source port instead of SDP negotiated destination port (with disable_rtp_auto_adjust=true) In-Reply-To: References: , , <53970340-F914-470A-8BC8-446489102D5B@gmail.com>, Message-ID: Guys, any idea what this is not working? - http://pastebin.freeswitch.org/19510 (logs from call). FS is enforcing symmetrical RTP in a config with an asymmetrical-only B2BUA. It starts by sending RTP to the negotiated port (remote destination) but switches to the actual remote source in ~.3-.5s.I tried setting disable_rtp_auto_adjust=true. Thanks,A.C. From: hexade at hotmail.com To: freeswitch-users at lists.freeswitch.org Date: Fri, 13 Jul 2012 13:37:25 -0400 Subject: Re: [Freeswitch-users] One Way RTP (no NAT) - Freeswitch sends RTP to the remote source port instead of SDP negotiated destination port (with disable_rtp_auto_adjust=true) Thank you Steve - here it is - http://pastebin.freeswitch.org/19510. Fresh call, new ports. Notice this line, this is what I want to stop: 2012-07-13 10:18:42.935692 [INFO] switch_rtp.c:3133 Auto Changing port from 10.3.212.105:8024 to 10.3.212.105:12016 Thank you, A.C. CC: freeswitch-users at lists.freeswitch.org From: steveayre at gmail.com Date: Fri, 13 Jul 2012 13:30:38 +0100 To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] One Way RTP (no NAT) - Freeswitch sends RTP to the remote source port instead of SDP negotiated destination port (with disable_rtp_auto_adjust=true) Can you supply a pastebin of adebug log of the call? Steve on iPhone -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120716/8376ed97/attachment.html From ian.bonham at gmail.com Mon Jul 16 12:44:44 2012 From: ian.bonham at gmail.com (Ian Bonham) Date: Mon, 16 Jul 2012 10:44:44 +0200 Subject: [Freeswitch-users] AVMD Message-ID: Hi Everyone, New member, so nice to 'meet' you all! I'm experimenting with FreeSwitch to see how it could replace Asterisk in a mass dialler system that I run in Europe, and I have been playing with the AVMD module. It seems to work perfectly, and does detect voicemail very well. What I am trying to do through is have the dialler generate calls and run as much as possible through FreeSwitch's core, rather than external scripts (I'm using PHP for most of my external functionality as thats what I use to write the agent interfaces too). So I generate individual calls and launch them in FreeSwitch, and it then does most of the call logic from XML dialplan commands, jumping to PHP shell scripts for quick MySQL functions. I'd like very much to be able to run AVMD from the second the call is answered, which is simple enough, but at the same time have my outgoing message playing. So we are assuming the call is a human, but if during the OGM playout AVMD hears a beep, it just terminates the call. Can anyone tell me if this is possible please, either using XML dialplan (I'm guessing not as the XML is sequential), or using a programming script like LUA. Many thanks for any advise anyone can offer, Ian -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120716/76acfbe3/attachment-0001.html From mbodbg at gmx.net Mon Jul 16 15:45:20 2012 From: mbodbg at gmx.net (mbo) Date: Mon, 16 Jul 2012 13:45:20 +0200 Subject: [Freeswitch-users] mod_spandsp dosn't compile Message-ID: <54696C79-10AD-4B32-9E94-644C98839647@gmx.net> Hello, I just updated my freeswitch test installation (make sure). mod_spandsp doesn't compile anymore, I get the following error: making all mod_spandsp make[5]: Entering directory `/usr/src/freeswitch/src/mod/applications/mod_spandsp' Creating mod_spandsp_la-mod_spandsp.lo quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../../../../src/include -I../../../../libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -I/usr/src/freeswitch/libs/spandsp/src -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -I/usr/src/freeswitch/libs/spandsp/src -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT mod_spandsp_la-mod_spandsp.lo -MD -MP -MF .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o .libs/mod_spandsp_la-mod_spandsp.o In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:82, from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:109: error: expected specifier-qualifier-list before ?t81_t82_arith_encode_state_t? /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:202: error: expected specifier-qualifier-list before ?t81_t82_arith_decode_state_t? In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:85, from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_rx.h:118: error: expected specifier-qualifier-list before ?t85_decode_state_t? In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:86, from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:132: error: expected specifier-qualifier-list before ?t85_encode_state_t? In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:96, from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:33: error: expected specifier-qualifier-list before ?ademco_contactid_report_func_t? cc1: warnings being treated as errors /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:48: error: struct has no members Thanks Markus From bdfoster at endigotech.com Mon Jul 16 19:16:52 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 16 Jul 2012 11:16:52 -0400 Subject: [Freeswitch-users] mod_spandsp dosn't compile In-Reply-To: <54696C79-10AD-4B32-9E94-644C98839647@gmx.net> References: <54696C79-10AD-4B32-9E94-644C98839647@gmx.net> Message-ID: Correct usage is make current, try that and let us know what happens. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 16, 2012 11:09 AM, "mbo" wrote: > Hello, > > I just updated my freeswitch test installation (make sure). mod_spandsp > doesn't compile anymore, I get the following error: > > making all mod_spandsp > make[5]: Entering directory > `/usr/src/freeswitch/src/mod/applications/mod_spandsp' > Creating mod_spandsp_la-mod_spandsp.lo > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../../../../src/include > -I../../../../libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include > -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 > -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/src/include > -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement > -I/usr/src/freeswitch/libs/spandsp/src > -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff > -I/usr/src/freeswitch/libs/spandsp/src > -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT > mod_spandsp_la-mod_spandsp.lo -MD -MP -MF > .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o > .libs/mod_spandsp_la-mod_spandsp.o > In file included from > /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:82, > from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, > from mod_spandsp.h:50, > from mod_spandsp.c:39: > /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:109: error: > expected specifier-qualifier-list before ?t81_t82_arith_encode_state_t? > /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:202: error: > expected specifier-qualifier-list before ?t81_t82_arith_decode_state_t? > In file included from > /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:85, > from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, > from mod_spandsp.h:50, > from mod_spandsp.c:39: > /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_rx.h:118: error: > expected specifier-qualifier-list before ?t85_decode_state_t? > In file included from > /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:86, > from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, > from mod_spandsp.h:50, > from mod_spandsp.c:39: > /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:132: error: > expected specifier-qualifier-list before ?t85_encode_state_t? > In file included from > /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:96, > from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, > from mod_spandsp.h:50, > from mod_spandsp.c:39: > /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:33: > error: expected specifier-qualifier-list before > ?ademco_contactid_report_func_t? > cc1: warnings being treated as errors > /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:48: > error: struct has no members > > > Thanks > > Markus > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120716/e0b137f8/attachment.html From peter.olsson at visionutveckling.se Mon Jul 16 19:19:04 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 16 Jul 2012 15:19:04 +0000 Subject: [Freeswitch-users] mod_spandsp dosn't compile In-Reply-To: <54696C79-10AD-4B32-9E94-644C98839647@gmx.net> References: <54696C79-10AD-4B32-9E94-644C98839647@gmx.net> Message-ID: <1FFF97C269757C458224B7C895F35F151351DB@cantor.std.visionutv.se> make spandsp-reconf ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för mbo [mbodbg at gmx.net] Skickat: den 16 juli 2012 13:45 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] mod_spandsp dosn't compile Hello, I just updated my freeswitch test installation (make sure). mod_spandsp doesn't compile anymore, I get the following error: making all mod_spandsp make[5]: Entering directory `/usr/src/freeswitch/src/mod/applications/mod_spandsp' Creating mod_spandsp_la-mod_spandsp.lo quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../../../../src/include -I../../../../libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -I/usr/src/freeswitch/libs/spandsp/src -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -I/usr/src/freeswitch/libs/spandsp/src -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT mod_spandsp_la-mod_spandsp.lo -MD -MP -MF .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o .libs/mod_spandsp_la-mod_spandsp.o In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:82, from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:109: error: expected specifier-qualifier-list before ?t81_t82_arith_encode_state_t? /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:202: error: expected specifier-qualifier-list before ?t81_t82_arith_decode_state_t? In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:85, from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_rx.h:118: error: expected specifier-qualifier-list before ?t85_decode_state_t? In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:86, from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:132: error: expected specifier-qualifier-list before ?t85_encode_state_t? In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:96, from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:33: error: expected specifier-qualifier-list before ?ademco_contactid_report_func_t? cc1: warnings being treated as errors /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:48: error: struct has no members Thanks Markus _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:50042c8032761028316903! From toddb at toddbailey.net Mon Jul 16 19:47:07 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Mon, 16 Jul 2012 08:47:07 -0700 Subject: [Freeswitch-users] programming/config Polycom ip501 In-Reply-To: References: <1342400450.3892.23.camel@mythtv> Message-ID: <1342453627.31445.14.camel@mythtv> Hello, DHCP is running on the network router. The mac address is 00:04:F2:04:79:68 and I've created these files 0004F2047968.cfg 0004F2047968-settings.cfg 004F2047968-directory.cfg did I name these properly? or do I want these files to use a naming convention like 00:04:F2:04:79:68.cfg? I've never used tcpdump before so I'll have to read up on it's usage. When I first installed the phone it connected to the boot server and went through a update process, ie loading newer boot rom, formatting file system etc. The phone does work, and I can dial or receive calls so connectivity isn't a problem, But I would like to add services, program soft keys and be able to retrieve messages. On Mon, 2012-07-16 at 07:04 +0300, Yehavi Bourvine wrote: > Hello Todd, > > 1. You need a DHCP server to tell the phone where to get the > configuration files from. I sent you an example in a private > mail a fwe days ago. > 2. Note that the file mac-address.cfg should point to the other > files. > 3. As I told you: send me a TCPDUMP of the boot process and then > we might be able to help you. > > __Yehavi: > > 2012/7/16 Todd Bailey > Hi All, > > > I'm running a vanilla install of Fs and current have a hard > phone > attached - the ip501. > > tftp config files. > > The address of the phone is 192.168.1.23 & 00:04:F2:04:79:68 > > If I read the setup wiki correctly, I need to have created 3 > different > files > > 0004F2047968.cfg > 0004F2047968-settings.cfg > 0004F2047968-directory.cfg > > these don't appear to have any effect on phone settings or > functionality, did I name these properly? and what if > anything should > I do with the files from the firmware zip file I downloaded: > 000000000000.cfg > 000000000000-directory~.xml > phone1_318.cfg > sip_318.cfg > > > Speed dial (stored contacts) ? > > I was able to config the phone using the web interface but I'm > running > into several issues in getting various functions to work. > how do I setup and store a contact to get to an outside line I > have to > dial 9 then send then dial the rest of the number. Needless to > say , > speed dial doesn't work, > > What sections do I need to study to get 9 + ac + phone number > to work > either manually or via a saved contact? > > > Messages? > > how do I program the phone access voice mail? > when I press "messages", I can select 1 of the 3 extensions > after doing > so it dials that extension instead of going to the voice mail. > > sorry for being verbose, it couldn't be helped. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mbodbg at gmx.net Mon Jul 16 19:29:50 2012 From: mbodbg at gmx.net (mbo) Date: Mon, 16 Jul 2012 17:29:50 +0200 Subject: [Freeswitch-users] mod_spandsp dosn't compile In-Reply-To: References: <54696C79-10AD-4B32-9E94-644C98839647@gmx.net> Message-ID: Ok, make current worked. Thanks Markus Am 16.07.2012 um 17:16 schrieb Brian Foster: > Correct usage is make current, try that and let us know what happens. > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > > On Jul 16, 2012 11:09 AM, "mbo" wrote: > Hello, > > I just updated my freeswitch test installation (make sure). mod_spandsp doesn't compile anymore, I get the following error: > > making all mod_spandsp > make[5]: Entering directory `/usr/src/freeswitch/src/mod/applications/mod_spandsp' > Creating mod_spandsp_la-mod_spandsp.lo > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../../../../src/include -I../../../../libs/xmlrpc-c -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -I/usr/src/freeswitch/libs/spandsp/src -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -I/usr/src/freeswitch/libs/spandsp/src -I/usr/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT mod_spandsp_la-mod_spandsp.lo -MD -MP -MF .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o .libs/mod_spandsp_la-mod_spandsp.o > In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:82, > from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, > from mod_spandsp.h:50, > from mod_spandsp.c:39: > /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:109: error: expected specifier-qualifier-list before ?t81_t82_arith_encode_state_t? > /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t85.h:202: error: expected specifier-qualifier-list before ?t81_t82_arith_decode_state_t? > In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:85, > from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, > from mod_spandsp.h:50, > from mod_spandsp.c:39: > /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_rx.h:118: error: expected specifier-qualifier-list before ?t85_decode_state_t? > In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:86, > from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, > from mod_spandsp.h:50, > from mod_spandsp.c:39: > /usr/src/freeswitch/libs/spandsp/src/spandsp/private/t4_tx.h:132: error: expected specifier-qualifier-list before ?t85_encode_state_t? > In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp/expose.h:96, > from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:135, > from mod_spandsp.h:50, > from mod_spandsp.c:39: > /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:33: error: expected specifier-qualifier-list before ?ademco_contactid_report_func_t? > cc1: warnings being treated as errors > /usr/src/freeswitch/libs/spandsp/src/spandsp/private/ademco_contactid.h:48: error: struct has no members > > > Thanks > > Markus > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120716/3c363186/attachment.html From yehavi.bourvine at gmail.com Mon Jul 16 20:31:42 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Mon, 16 Jul 2012 19:31:42 +0300 Subject: [Freeswitch-users] programming/config Polycom ip501 In-Reply-To: <1342453627.31445.14.camel@mythtv> References: <1342400450.3892.23.camel@mythtv> <1342453627.31445.14.camel@mythtv> Message-ID: The file names are ok, except the directory which should have .xml extension. Please send me the contents of 0004F2047968.cfg, as it tells which files to load. Thanks, __Yehavi: 2012/7/16 Todd Bailey > Hello, > > DHCP is running on the network router. > > The mac address is 00:04:F2:04:79:68 and > I've created these files > 0004F2047968.cfg > 0004F2047968-settings.cfg > 004F2047968-directory.cfg > > did I name these properly? > or do I want these files to use a naming convention like > 00:04:F2:04:79:68.cfg? > > I've never used tcpdump before so I'll have to read up on it's usage. > > When I first installed the phone it connected to the boot server and > went through a update process, ie loading newer boot rom, formatting > file system etc. > > The phone does work, and I can dial or receive calls so connectivity > isn't a problem, But I would like to add services, program soft keys > and be able to retrieve messages. > > > On Mon, 2012-07-16 at 07:04 +0300, Yehavi Bourvine wrote: > > Hello Todd, > > > > 1. You need a DHCP server to tell the phone where to get the > > configuration files from. I sent you an example in a private > > mail a fwe days ago. > > 2. Note that the file mac-address.cfg should point to the other > > files. > > 3. As I told you: send me a TCPDUMP of the boot process and then > > we might be able to help you. > > > > __Yehavi: > > > > 2012/7/16 Todd Bailey > > Hi All, > > > > > > I'm running a vanilla install of Fs and current have a hard > > phone > > attached - the ip501. > > > > tftp config files. > > > > The address of the phone is 192.168.1.23 & 00:04:F2:04:79:68 > > > > If I read the setup wiki correctly, I need to have created 3 > > different > > files > > > > 0004F2047968.cfg > > 0004F2047968-settings.cfg > > 0004F2047968-directory.cfg > > > > these don't appear to have any effect on phone settings or > > functionality, did I name these properly? and what if > > anything should > > I do with the files from the firmware zip file I downloaded: > > 000000000000.cfg > > 000000000000-directory~.xml > > phone1_318.cfg > > sip_318.cfg > > > > > > Speed dial (stored contacts) ? > > > > I was able to config the phone using the web interface but I'm > > running > > into several issues in getting various functions to work. > > how do I setup and store a contact to get to an outside line I > > have to > > dial 9 then send then dial the rest of the number. Needless to > > say , > > speed dial doesn't work, > > > > What sections do I need to study to get 9 + ac + phone number > > to work > > either manually or via a saved contact? > > > > > > Messages? > > > > how do I program the phone access voice mail? > > when I press "messages", I can select 1 of the 3 extensions > > after doing > > so it dials that extension instead of going to the voice mail. > > > > sorry for being verbose, it couldn't be helped. > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120716/a519f194/attachment-0001.html From admin at blindi.net Mon Jul 16 20:50:55 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Mon, 16 Jul 2012 18:50:55 +0200 (CEST) Subject: [Freeswitch-users] EXCHANGE_ROUTING_ERROR after about 62 transfers of a sesseion In-Reply-To: <1FFF97C269757C458224B7C895F35F15134EDF@cantor.std.visionutv.se> References: <1342296935.3512.47.camel@mythtv> , <1FFF97C269757C458224B7C895F35F15134E74@cantor.std.visionutv.se>, <1FFF97C269757C458224B7C895F35F15134EDF@cantor.std.visionutv.se> Message-ID: Hi peter, Thanks for you nice help. it works fine. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From admin at blindi.net Mon Jul 16 20:57:46 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Mon, 16 Jul 2012 18:57:46 +0200 (CEST) Subject: [Freeswitch-users] say date time for other countrys? In-Reply-To: <5003B44C.50409@gmail.com> References: <5003B44C.50409@gmail.com> Message-ID: Hi guys, i.m in germany the date time format is: day, month, year, hour, minute and secound. how can i say the correct format please? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From marketing at cluecon.com Mon Jul 16 21:24:07 2012 From: marketing at cluecon.com (Michael Collins) Date: Mon, 16 Jul 2012 10:24:07 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: One week closer to ClueCon! Things have been very busy for the FreeSWITCH team and community. In addition to gearing up for ClueCon we've all been working on our day jobs and mixing in the occasional summer vacation. We've also had inclement weather play a role in scheduling one of our weekly conferencecall presentations. Dave Kompel, one of our resident Windows gurus, had been scheduled to discuss a technique for handling the Windows firewall in a manner similar to how we use Fail2Banin a Linux environment, however thunderstorms in his area required his full attention on protecting and keeping his servers up and running. We will reschedule as soon as possible. This week we have scheduled Darren Schreiber from 2600hz. He will be doing a follow up to his SIP 101 discussionfrom several weeks ago. We will be diving a bit deeper into the SIP protocol. In the initial discussion we focused most of our attention on getting SIP endpoints registered and the setup/teardown process. This week we will look at some of the other things SIP can do, like presence. We invite you to join us this Wednesday at 1PM Eastern, 10AM Pacific. In ClueCon news we are very happy to report that 2600hz is back as a Silver sponsor! In fact, the aforementioned Darren Schreiber will be giving a presentationat ClueCon, discussing some of the hard-earned lessons that 2600hz has learned over the past few years. If you are interested in cloud telephony then definitely be there for Darren's talk. For those who've not registered yet please keep in mind that the "bits will shift again" this Wednesday and registrants will receive four entries instead of the current eight. Register now to maximize your opportunity to win lots of great prizes. The ClueCon countdown is already down to 21 days! We are looking forward to seeing everyone together again in Chicago. -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0716 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120716/95448617/attachment.html From mbodbg at gmx.net Mon Jul 16 20:53:22 2012 From: mbodbg at gmx.net (mbo) Date: Mon, 16 Jul 2012 18:53:22 +0200 Subject: [Freeswitch-users] Setting caller_ton doesn't work Message-ID: <18CE9710-7978-4CF1-88A6-4CEEEFBB0DC1@gmx.net> In the sangoma wiki I can find a sample how to configure the caller_ton per call (A200 E1 card): http://wiki.sangoma.com/wanpipe-freeswitch-config-appendix#ton-npi-per-call Based on that, I've configured the following test extension: I'm dialing out from sip extension 1001 to the pstn. If I look in the ISDN Trace, i can see ... Calling Party Number:49111111111111111(l:17) plan:isdn(1) type:national(2)scr:user, passed(1) pres:allowed(0) ... that caller_ton is still set to 2 (national). You can find the full freeswitch trace here: http://pastebin.freeswitch.org/19525 It seems it's not overwriting the default parameter outbound-calling-ton from freetdm.conf.xml. Here my freetdm.conf.xml: I'm pretty sure that was working in the past. Has something changed here? From mario_fs at mgtech.com Mon Jul 16 21:36:35 2012 From: mario_fs at mgtech.com (Mario G) Date: Mon, 16 Jul 2012 10:36:35 -0700 Subject: [Freeswitch-users] SSD Tuning for Linux now on wiki Message-ID: <545AC920-E5A3-4570-A980-234C80E4A15B@mgtech.com> As promised, small but lot's of work when into this, everything was actually tested and in use. During the past few months I found many misconceptions and errors on the web about SSDs so this should be a time saver. Enjoy! Mario G http://wiki.freeswitch.org/wiki/SSD_Tuning_for_Linux Now referenced in: http://wiki.freeswitch.org/wiki/Performance_Tuning <- New page created my MC http://wiki.freeswitch.org/wiki/Performance_testing_and_configurations From msc at freeswitch.org Mon Jul 16 22:32:41 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Jul 2012 11:32:41 -0700 Subject: [Freeswitch-users] AVMD In-Reply-To: References: Message-ID: Hi Ian, Just curious - at what point during the call are you currently launching AVMD? In any case, this sounds like a perfect job for "execute_on_answer": http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer Just be nice and don't abuse your new found powers! :) -MC P.S. - Welcome to the FreeSWITCH community. On Mon, Jul 16, 2012 at 1:44 AM, Ian Bonham wrote: > Hi Everyone, > > New member, so nice to 'meet' you all! > > I'm experimenting with FreeSwitch to see how it could replace Asterisk in > a mass dialler system that I run in Europe, and I have been playing with > the AVMD module. It seems to work perfectly, and does detect voicemail very > well. What I am trying to do through is have the dialler generate calls and > run as much as possible through FreeSwitch's core, rather than external > scripts (I'm using PHP for most of my external functionality as thats what > I use to write the agent interfaces too). So I generate individual calls > and launch them in FreeSwitch, and it then does most of the call logic from > XML dialplan commands, jumping to PHP shell scripts for quick MySQL > functions. > > I'd like very much to be able to run AVMD from the second the call is > answered, which is simple enough, but at the same time have my outgoing > message playing. So we are assuming the call is a human, but if during the > OGM playout AVMD hears a beep, it just terminates the call. Can anyone tell > me if this is possible please, either using XML dialplan (I'm guessing not > as the XML is sequential), or using a programming script like LUA. > > Many thanks for any advise anyone can offer, > > Ian > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120716/219e81a3/attachment.html From anthony.minessale at gmail.com Mon Jul 16 23:46:23 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 16 Jul 2012 14:46:23 -0500 Subject: [Freeswitch-users] Sound problems in voicemail. In-Reply-To: <5CFE5DF4-8C62-4A4F-8914-8F8947FBD56A@visionutveckling.se> References: <5003B44C.50409@gmail.com> <5003C70A.5070301@gmail.com> <5CFE5DF4-8C62-4A4F-8914-8F8947FBD56A@visionutveckling.se> Message-ID: check for linksys/sipura ptime bug On Mon, Jul 16, 2012 at 2:58 AM, Peter Olsson wrote: > In that case, please submit a Jira report. Make sure to attach all the needed information (debug logs etc), and in this case, a pcap for the distorted audio might be good. > > /Peter > > 16 jul 2012 kl. 09:51 skrev "Vbvbrj" : > >> Yes, latest git and real hardware, dedicated debian server with only FS >> installed. But same problem is in VM too. >> >> On 16.07.2012 10:43, Peter Olsson wrote: >>> Are you on latest git head and are you using real hardware (no VMware etc)? >>> >>> /Peter >>> >>> 16 jul 2012 kl. 08:37 skrev "Vbvbrj" : >>> >>>> Hello. >>>> >>>> Searching for a resolve. I've found in default dialplan from distro in >>>> file conf/dialplan/default.xml, line 270: >>>> >>>> >>>> >>>> also on line 284 is: >>>> >>>> >>>> >>>> Changing line 270 to: >>>> >>>> >>>> >>>> and leaving line 284 as is, solves the problem with the sound when >>>> calling some extension which is not registered from 1000 to 1019. But >>>> problem with sounds persists when calling 1100 to 1119. Messages >>>> "Goodbay" is very distorted. >>>> >>>> Is this a bug or something else that "application="bridge" >>>> data="loopback/app=voicemail:default" distort sound and >>>> "application="voicemail" data="default" does not distort? >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> !DSPAM:5003c4b632761996510574! >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From red.rain.seven at gmail.com Tue Jul 17 00:26:33 2012 From: red.rain.seven at gmail.com (Henry Huang) Date: Mon, 16 Jul 2012 13:26:33 -0700 Subject: [Freeswitch-users] AVMD In-Reply-To: References: Message-ID: Michael, I have also been looking into AVMD. The point being that I want to play some pre-recorded message once the phone is answered and start AVMD at the same time, and if the 'beep' is detected later during the call(say 10 sec. later). I want to be able to stop the original Playback and start a new Playback. Is there a way to do that? Henry On Mon, Jul 16, 2012 at 11:32 AM, Michael Collins wrote: > Hi Ian, > > Just curious - at what point during the call are you currently launching > AVMD? In any case, this sounds like a perfect job for "execute_on_answer": > http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer > > Just be nice and don't abuse your new found powers! :) > > -MC > > P.S. - Welcome to the FreeSWITCH community. > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120716/e64d891d/attachment.html From ian.bonham at gmail.com Tue Jul 17 00:29:10 2012 From: ian.bonham at gmail.com (Ian Bonham) Date: Mon, 16 Jul 2012 22:29:10 +0200 Subject: [Freeswitch-users] AVMD In-Reply-To: References: Message-ID: Hi Michael, Thanks for getting back to me, this is why I on the whole love the Linux/FOSS community (most of the time!) I've actually had a fun afternoon studying more and more about OpenSwitch and ended up changing my dial logic totally. I now have a PHP script staying resident, that queries fs_cli every 5 seconds for the number of active calls. If that number is less than the target number of calls per agent (in total) I know I need to add some more calls. So the script grabs a load out the database (based on how many I need to get back to the target # of calls) and chucks them into FreeSwitch using fs_cli -x and just an originate. I originate the call and lob it to an extension for that campaign. That 'extension' then launches an LUA script for each call which starts a playback and runs AVMD at the same time. So we start playing the OGM, but if I hear a beep tone I just hang up as we're wasting money. If a DTMF has been detected I process that and handle the call accordingly (If they press 9, I go back to the database and add the number to the DNC list, if they press 5 I look for an agent that has been waiting the longest and bridge the call to them). If neither of those conditions is reached I mark the call in the database for a retry and count how many times we've tried it (unless it's a duff number, then I mark it as such and the call centre get a report at the end of the day on dead lines so they can get a refund on the dead numbers in any particular data set). It's almost running now, few bugs to stamp on tomorrow, and then I'll run it with 20 or so 'test' agents tomorrow or Wednesday and see how hard I can push it. I'm on an 8 core Xeon dedicated server at the moment, so I'll see how hard I can thrash that and then see about migrating it to AWS maybe. Thanks to all the FreeSwitch developers, my job is suddenly fun again! Bon On 16 July 2012 20:32, Michael Collins wrote: > Hi Ian, > > Just curious - at what point during the call are you currently launching > AVMD? In any case, this sounds like a perfect job for "execute_on_answer": > http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer > > Just be nice and don't abuse your new found powers! :) > > -MC > > P.S. - Welcome to the FreeSWITCH community. > > > On Mon, Jul 16, 2012 at 1:44 AM, Ian Bonham wrote: > >> Hi Everyone, >> >> New member, so nice to 'meet' you all! >> >> I'm experimenting with FreeSwitch to see how it could replace Asterisk in >> a mass dialler system that I run in Europe, and I have been playing with >> the AVMD module. It seems to work perfectly, and does detect voicemail very >> well. What I am trying to do through is have the dialler generate calls and >> run as much as possible through FreeSwitch's core, rather than external >> scripts (I'm using PHP for most of my external functionality as thats what >> I use to write the agent interfaces too). So I generate individual calls >> and launch them in FreeSwitch, and it then does most of the call logic from >> XML dialplan commands, jumping to PHP shell scripts for quick MySQL >> functions. >> >> I'd like very much to be able to run AVMD from the second the call is >> answered, which is simple enough, but at the same time have my outgoing >> message playing. So we are assuming the call is a human, but if during the >> OGM playout AVMD hears a beep, it just terminates the call. Can anyone tell >> me if this is possible please, either using XML dialplan (I'm guessing not >> as the XML is sequential), or using a programming script like LUA. >> >> Many thanks for any advise anyone can offer, >> >> Ian >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120716/1cce16c6/attachment.html From msc at freeswitch.org Tue Jul 17 00:38:33 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Jul 2012 13:38:33 -0700 Subject: [Freeswitch-users] AVMD In-Reply-To: References: Message-ID: On Mon, Jul 16, 2012 at 1:26 PM, Henry Huang wrote: > Michael, > > I have also been looking into AVMD. The point being that I want to play > some pre-recorded message once the phone is answered and start AVMD at the > same time, and if the 'beep' is detected later during the call(say 10 sec. > later). I want to be able to stop the original Playback and start a new > Playback. Is there a way to do that? > Yes, but "some assembly required" as the saying goes. AVMD only throws events, so whatever solution you are using must be able to consume and react to those events. The OP is using Lua which definitely allows you to consume the events. There's even a really simple example on the wiki about using the setInputCallback function to look for "events" and then do something once the event comes in. It's a really sparse example but it demonstrates how to catch an event. From there you'd need to put in your commands. I'm assuming you'll need a break app or uuid_break API to stop the current playback and then you'll need to queue up another playback. It's also good to turn off avmd at this point as well. I'm just throwing this out there off the top of my head, so I highly recommend some hands on tinkering. Please let us know what you end up doing. :) -MC > > Henry > > On Mon, Jul 16, 2012 at 11:32 AM, Michael Collins wrote: > >> Hi Ian, >> >> Just curious - at what point during the call are you currently launching >> AVMD? In any case, this sounds like a perfect job for "execute_on_answer": >> http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer >> >> Just be nice and don't abuse your new found powers! :) >> >> -MC >> >> P.S. - Welcome to the FreeSWITCH community. >> >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120716/12dc898b/attachment.html From ian.bonham at gmail.com Tue Jul 17 00:58:30 2012 From: ian.bonham at gmail.com (Ian Bonham) Date: Mon, 16 Jul 2012 22:58:30 +0200 Subject: [Freeswitch-users] AVMD In-Reply-To: References: Message-ID: Michael is totally right there! Hands on Tinkering is what I've been doing all afternoon and picked up so much. I also hit Amazon (Other book shops are available!) and got myself the FreeSwitch 1.0.6 book by our Mr Collins himself, Mr Minessale and Mr Schriber. A fine read guys, and a good start when you're really wanting to get stuck in! I was thinking, oh great LUA, something else new to learn but the book has really got me going in the right direction, thanks! Bon On 16 July 2012 22:38, Michael Collins wrote: > > > On Mon, Jul 16, 2012 at 1:26 PM, Henry Huang wrote: > >> Michael, >> >> I have also been looking into AVMD. The point being that I want to play >> some pre-recorded message once the phone is answered and start AVMD at the >> same time, and if the 'beep' is detected later during the call(say 10 sec. >> later). I want to be able to stop the original Playback and start a new >> Playback. Is there a way to do that? >> > Yes, but "some assembly required" as the saying goes. AVMD only throws > events, so whatever solution you are using must be able to consume and > react to those events. The OP is using Lua which definitely allows you to > consume the events. There's even a really simple example on the wiki about > using the setInputCallback function to look for "events" and then do > something once the event comes in. It's a really sparse example but it > demonstrates how to catch an event. From there you'd need to put in your > commands. I'm assuming you'll need a break app or uuid_break API to stop > the current playback and then you'll need to queue up another playback. > It's also good to turn off avmd at this point as well. I'm just throwing > this out there off the top of my head, so I highly recommend some hands on > tinkering. > > Please let us know what you end up doing. :) > > -MC > >> >> Henry >> >> On Mon, Jul 16, 2012 at 11:32 AM, Michael Collins wrote: >> >>> Hi Ian, >>> >>> Just curious - at what point during the call are you currently launching >>> AVMD? In any case, this sounds like a perfect job for "execute_on_answer": >>> http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer >>> >>> Just be nice and don't abuse your new found powers! :) >>> >>> -MC >>> >>> P.S. - Welcome to the FreeSWITCH community. >>> >>> >>> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120716/4ad10724/attachment-0001.html From ifoundthetao at gmail.com Tue Jul 17 01:01:46 2012 From: ifoundthetao at gmail.com (Timothy Bolton) Date: Mon, 16 Jul 2012 16:01:46 -0500 Subject: [Freeswitch-users] AVMD In-Reply-To: References: Message-ID: <5004813A.5000804@gmail.com> I've been back and forth on getting the book (time to read and blah blah), but you just sold me on it. (: 'We who cut mere stones must always be envisioning cathedrals.' Quarry Worker's Creed On 7/16/2012 3:58 PM, Ian Bonham wrote: > Michael is totally right there! Hands on Tinkering is what I've been > doing all afternoon and picked up so much. I also hit Amazon (Other > book shops are available!) and got myself the FreeSwitch 1.0.6 book by > our Mr Collins himself, Mr Minessale and Mr Schriber. A fine read > guys, and a good start when you're really wanting to get stuck in! > I was thinking, oh great LUA, something else new to learn but the book > has really got me going in the right direction, thanks! > > Bon > > > On 16 July 2012 22:38, Michael Collins > wrote: > > > > On Mon, Jul 16, 2012 at 1:26 PM, Henry Huang > > wrote: > > Michael, > > I have also been looking into AVMD. The point being that I > want to play some pre-recorded message once the phone is > answered and start AVMD at the same time, and if the 'beep' is > detected later during the call(say 10 sec. later). I want to > be able to stop the original Playback and start a new > Playback. Is there a way to do that? > > Yes, but "some assembly required" as the saying goes. AVMD only > throws events, so whatever solution you are using must be able to > consume and react to those events. The OP is using Lua which > definitely allows you to consume the events. There's even a really > simple example on the wiki about using the setInputCallback > function to look for "events" and then do something once the event > comes in. It's a really sparse example but it demonstrates how to > catch an event. From there you'd need to put in your commands. I'm > assuming you'll need a break app or uuid_break API to stop the > current playback and then you'll need to queue up another > playback. It's also good to turn off avmd at this point as well. > I'm just throwing this out there off the top of my head, so I > highly recommend some hands on tinkering. > > Please let us know what you end up doing. :) > > -MC > > > Henry > > On Mon, Jul 16, 2012 at 11:32 AM, Michael Collins > > wrote: > > Hi Ian, > > Just curious - at what point during the call are you > currently launching AVMD? In any case, this sounds like a > perfect job for "execute_on_answer": > http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer > > Just be nice and don't abuse your new found powers! :) > > -MC > > P.S. - Welcome to the FreeSWITCH community. > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120716/bb0aa9c4/attachment.html From ian.bonham at gmail.com Tue Jul 17 01:14:03 2012 From: ian.bonham at gmail.com (Ian Bonham) Date: Mon, 16 Jul 2012 23:14:03 +0200 Subject: [Freeswitch-users] AVMD In-Reply-To: <5004813A.5000804@gmail.com> References: <5004813A.5000804@gmail.com> Message-ID: It's really good actually, some dry humour in there, some great ideas, and I got it on Kindle so I can reference it anywhere. I'm not on commission btw! ;) On 16 July 2012 23:01, Timothy Bolton wrote: > I've been back and forth on getting the book (time to read and blah > blah), but you just sold me on it. (: > > 'We who cut mere stones must always be envisioning cathedrals.' > Quarry Worker's Creed > > On 7/16/2012 3:58 PM, Ian Bonham wrote: > > Michael is totally right there! Hands on Tinkering is what I've been doing > all afternoon and picked up so much. I also hit Amazon (Other book shops > are available!) and got myself the FreeSwitch 1.0.6 book by our Mr Collins > himself, Mr Minessale and Mr Schriber. A fine read guys, and a good start > when you're really wanting to get stuck in! > I was thinking, oh great LUA, something else new to learn but the book has > really got me going in the right direction, thanks! > > Bon > > > On 16 July 2012 22:38, Michael Collins wrote: > >> >> >> On Mon, Jul 16, 2012 at 1:26 PM, Henry Huang wrote: >> >>> Michael, >>> >>> I have also been looking into AVMD. The point being that I want to play >>> some pre-recorded message once the phone is answered and start AVMD at the >>> same time, and if the 'beep' is detected later during the call(say 10 sec. >>> later). I want to be able to stop the original Playback and start a new >>> Playback. Is there a way to do that? >>> >> Yes, but "some assembly required" as the saying goes. AVMD only throws >> events, so whatever solution you are using must be able to consume and >> react to those events. The OP is using Lua which definitely allows you to >> consume the events. There's even a really simple example on the wiki about >> using the setInputCallback function to look for "events" and then do >> something once the event comes in. It's a really sparse example but it >> demonstrates how to catch an event. From there you'd need to put in your >> commands. I'm assuming you'll need a break app or uuid_break API to stop >> the current playback and then you'll need to queue up another playback. >> It's also good to turn off avmd at this point as well. I'm just throwing >> this out there off the top of my head, so I highly recommend some hands on >> tinkering. >> >> Please let us know what you end up doing. :) >> >> -MC >> >>> >>> Henry >>> >>> On Mon, Jul 16, 2012 at 11:32 AM, Michael Collins wrote: >>> >>>> Hi Ian, >>>> >>>> Just curious - at what point during the call are you currently >>>> launching AVMD? In any case, this sounds like a perfect job for >>>> "execute_on_answer": >>>> http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer >>>> >>>> Just be nice and don't abuse your new found powers! :) >>>> >>>> -MC >>>> >>>> P.S. - Welcome to the FreeSWITCH community. >>>> >>>> >>>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120716/b1567bb2/attachment-0001.html From msc at freeswitch.org Tue Jul 17 02:38:16 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 16 Jul 2012 15:38:16 -0700 Subject: [Freeswitch-users] AVMD In-Reply-To: References: <5004813A.5000804@gmail.com> Message-ID: On Mon, Jul 16, 2012 at 2:14 PM, Ian Bonham wrote: > It's really good actually, some dry humour in there, some great ideas, and > I got it on Kindle so I can reference it anywhere. Dry humo[u]r? Us? I think you may have purchased the *our* FS book?! :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120716/8f23e69d/attachment.html From Joann.DeJesus at convergys.com Tue Jul 17 02:46:01 2012 From: Joann.DeJesus at convergys.com (Joann R DeJesus) Date: Mon, 16 Jul 2012 18:46:01 -0400 Subject: [Freeswitch-users] Duplicate DTMF Events - Please Help! Message-ID: <978316856E3C7F4184F9BE1D0AD2B4A0017D43DFA9ED@JAXMW10E.na.convergys.com> Hello All, I hope someone has experience this issue and can quickly assist me! We are supposed to be going live with a customer in a couple of days and need to find a resolution quickly in order to stay on track... We are using Freeswitch with Sangoma T1 cards to bridge TDM with our internal VoIP system. Call flow is as follows: PSTN --> T1/Sangoma cards --> Freeswitch --> IVR I have configured Freeswitch for RFC2833. When I make a call to the IVR from a landline or cell phone and press "5" at a prompt, FS will sometimes send two separate DTMF events to the IVR. This causes the IVR to receive "55" which then sends the call to the incorrect destination. I can verify with Wireshark that FS is sending two separate RFC2833 events. For some reason, this is exacerbated with heavy call volume (easy to reproduce during load testing). Has anyone experienced this symptom or have any ideas what I can check? Thanks in advance for any assistance!! - Joann ________________________________ NOTICE: The information contained in this electronic mail transmission is intended by Convergys Corporation for the use of the named individual or entity to which it is directed and may contain information that is privileged or otherwise confidential. If you have received this electronic mail transmission in error, please delete it from your system without copying or forwarding it, and notify the sender of the error by reply email or by telephone (collect), so that the sender's address records can be corrected. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120716/a1809de0/attachment.html From ian.bonham at gmail.com Tue Jul 17 03:07:19 2012 From: ian.bonham at gmail.com (Ian Bonham) Date: Tue, 17 Jul 2012 01:07:19 +0200 Subject: [Freeswitch-users] AVMD In-Reply-To: References: <5004813A.5000804@gmail.com> Message-ID: I'll get a 'proper' book tomorrow :p On 17 July 2012 00:38, Michael Collins wrote: > > > On Mon, Jul 16, 2012 at 2:14 PM, Ian Bonham wrote: > >> It's really good actually, some dry humour in there, some great ideas, >> and I got it on Kindle so I can reference it anywhere. > > > Dry humo[u]r? Us? I think you may have purchased the *our* FS book?! :) > > -MC > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120717/26fd39ee/attachment.html From mel0torme at gmail.com Tue Jul 17 03:29:50 2012 From: mel0torme at gmail.com (Tom C) Date: Mon, 16 Jul 2012 16:29:50 -0700 Subject: [Freeswitch-users] Profiling FreeSwitch? Message-ID: Are there any tools already in use for profiling FreeSwitch? That is, getting statistics on how often various procedures are called, and how much processor time they take, etc. Being a linux noob, I spent hours looking at "gprof", and when I finally got it working, I learned that it doesn't handle multi-threaded apps like FS. Is there a recommended tool that people are already using? Or should I just add my own logging code and recompile? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120716/794ecad2/attachment.html From Joshua.Foshee at LogixCom.com Tue Jul 17 03:49:06 2012 From: Joshua.Foshee at LogixCom.com (Joshua Foshee) Date: Mon, 16 Jul 2012 18:49:06 -0500 Subject: [Freeswitch-users] Callcenter and FIFO Mod Question Message-ID: <06502C073AD9394AADB3CA7FD94931BC0971F6B9@okc1x1.Logixcom.com> Does the Callcenter have any chime feature to play announces while a user is on hold? I know the FIFO does have this option with chime but adding on that does it have ability to setup a IVR while playing the chime? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120716/325b4f26/attachment-0001.html From bdfoster at endigotech.com Tue Jul 17 07:32:33 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 16 Jul 2012 23:32:33 -0400 Subject: [Freeswitch-users] Duplicate DTMF Events - Please Help! In-Reply-To: <978316856E3C7F4184F9BE1D0AD2B4A0017D43DFA9ED@JAXMW10E.na.convergys.com> References: <978316856E3C7F4184F9BE1D0AD2B4A0017D43DFA9ED@JAXMW10E.na.convergys.com> Message-ID: Rfc2833 is for sip, not for analog. Other than that I don't have much to offer. Try using inband only. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 16, 2012 6:50 PM, "Joann R DeJesus" wrote: > Hello All,**** > > ** ** > > I hope someone has experience this issue and can quickly assist me! We > are supposed to be going live with a customer in a couple of days and need > to find a resolution quickly in order to stay on track?**** > > ** ** > > We are using Freeswitch with Sangoma T1 cards to bridge TDM with our > internal VoIP system. Call flow is as follows: **** > > ** ** > > PSTN --> T1/Sangoma cards --> Freeswitch --> IVR **** > > ** ** > > I have configured Freeswitch for RFC2833. When I make a call to the IVR > from a landline or cell phone and press "5" at a prompt, FS will sometimes > send two separate DTMF events to the IVR. This causes the IVR to receive > "55" which then sends the call to the incorrect destination. **** > > ** ** > > I can verify with Wireshark that FS is sending two separate RFC2833events. For some reason, this is exacerbated with heavy call volume (easy > to reproduce during load testing). Has anyone experienced this symptom or > have any ideas what I can check? **** > > ** ** > > Thanks in advance for any assistance!! **** > > ** ** > > - Joann**** > > ** ** > > ------------------------------ > NOTICE: The information contained in this electronic mail transmission is > intended by Convergys Corporation for the use of the named individual or > entity to which it is directed and may contain information that is > privileged or otherwise confidential. If you have received this electronic > mail transmission in error, please delete it from your system without > copying or forwarding it, and notify the sender of the error by reply email > or by telephone (collect), so that the sender's address records can be > corrected. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120716/27077d11/attachment.html From curriegrad2004 at gmail.com Tue Jul 17 07:58:19 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Mon, 16 Jul 2012 20:58:19 -0700 Subject: [Freeswitch-users] Duplicate DTMF Events - Please Help! In-Reply-To: References: <978316856E3C7F4184F9BE1D0AD2B4A0017D43DFA9ED@JAXMW10E.na.convergys.com> Message-ID: All TDM links send DTMF tones as in-band only. Your only option for detecting such tones is to use in-band DTMF tone detection. If Sangoma offered a RFC2833 option for their cards, the most likely the card is doing the DTMF detection, which also could lead to a buggy detector on the card. On Mon, Jul 16, 2012 at 8:32 PM, Brian Foster wrote: > Rfc2833 is for sip, not for analog. Other than that I don't have much to > offer. Try using inband only. > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > > On Jul 16, 2012 6:50 PM, "Joann R DeJesus" > wrote: >> >> Hello All, >> >> >> >> I hope someone has experience this issue and can quickly assist me! We >> are supposed to be going live with a customer in a couple of days and need >> to find a resolution quickly in order to stay on track? >> >> >> >> We are using Freeswitch with Sangoma T1 cards to bridge TDM with our >> internal VoIP system. Call flow is as follows: >> >> >> >> PSTN --> T1/Sangoma cards --> Freeswitch --> IVR >> >> >> >> I have configured Freeswitch for RFC2833. When I make a call to the IVR >> from a landline or cell phone and press "5" at a prompt, FS will sometimes >> send two separate DTMF events to the IVR. This causes the IVR to receive >> "55" which then sends the call to the incorrect destination. >> >> >> >> I can verify with Wireshark that FS is sending two separate RFC2833 >> events. For some reason, this is exacerbated with heavy call volume (easy >> to reproduce during load testing). Has anyone experienced this symptom or >> have any ideas what I can check? >> >> >> >> Thanks in advance for any assistance!! >> >> >> >> - Joann >> >> >> >> >> ________________________________ >> NOTICE: The information contained in this electronic mail transmission is >> intended by Convergys Corporation for the use of the named individual or >> entity to which it is directed and may contain information that is >> privileged or otherwise confidential. If you have received this electronic >> mail transmission in error, please delete it from your system without >> copying or forwarding it, and notify the sender of the error by reply email >> or by telephone (collect), so that the sender's address records can be >> corrected. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From piyush.sharma at coraltele.com Tue Jul 17 09:39:27 2012 From: piyush.sharma at coraltele.com (Piyush Sharma) Date: Tue, 17 Jul 2012 11:09:27 +0530 (IST) Subject: [Freeswitch-users] Need help in IVR Message-ID: Hello Everyone, I am trying to create an IVR with menu and sub menus, but getting following error when I make the call, 2012-07-17 11:05:37.755034 [ERR] switch_ivr_menu.c:870 Unable to build xml menu 2012-07-17 11:05:37.755034 [ERR] switch_ivr_menu.c:870 Unable to build xml menu 2012-07-17 11:05:37.755034 [ERR] mod_dptools.c:1774 Unable to create menu I am new to this, so there might be little or a big error. My XML contents in ... conf/ivr_menus is and in .../en/demo is Thanks everyone, thanks for this support forum !!! -- Regards, Piyush Sharma. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120717/84377220/attachment.html From vbvbrj at gmail.com Tue Jul 17 10:54:15 2012 From: vbvbrj at gmail.com (Vbvbrj) Date: Tue, 17 Jul 2012 09:54:15 +0300 Subject: [Freeswitch-users] Sound problems in voicemail. In-Reply-To: References: <5003B44C.50409@gmail.com> <5003C70A.5070301@gmail.com> <5CFE5DF4-8C62-4A4F-8914-8F8947FBD56A@visionutveckling.se> Message-ID: <50050C17.9010406@gmail.com> I use only soft-phones like MicroSIP and PhonerLite. On 16.07.2012 22:46, Anthony Minessale wrote: > check for linksys/sipura ptime bug From miha at softnet.si Tue Jul 17 12:15:45 2012 From: miha at softnet.si (Miha) Date: Tue, 17 Jul 2012 10:15:45 +0200 Subject: [Freeswitch-users] mod_dptools.c:3831 Can't find user, different sip profiles Message-ID: <50051F31.9040807@softnet.si> HI, I have created two different sip_profiles due to NAT issue (5060 and 5070 port). FS also has multi tentat configuration. If I make call between two phones which are registered on same profile calls goes throught, If I make phone call on with phones, which are registered on two different profiles I get mod_dptools.c:3831 Can't find user. I have resolve this issue with presence but few members on IRC told me that this is not the bast way due to load. How can I deal with this issue? Thanks! Miha From anita.hall at simmortel.com Tue Jul 17 13:14:29 2012 From: anita.hall at simmortel.com (Anita Hall) Date: Tue, 17 Jul 2012 14:44:29 +0530 Subject: [Freeswitch-users] a=crypto in RTP/AVP, refer to rfc3711 Message-ID: Hi I am getting an error a=crypto in RTP/AVP, refer to rfc3711 when using sipml5 client with FS. There is some info on this thread, which I am currently experimenting with. https://groups.google.com/forum/#!msg/doubango/m7FCqYxRcQc/-ThdAwWJxP8J Is there any change in setting on FS side that can deal with this ? Complete SIP Log v=0 o=- 2834273280 1 IN IP4 127.0.0.1 s=webrtc (chrome 20.0.1127.0) - Doubango Telecom (sipML5 r000) t=0 0 a=group:BUNDLE audio video m=audio 51704 RTP/SAVPF 103 104 0 8 106 105 13 126 c=IN IP4 182.68.141.91 a=rtcp:51704 IN IP4 182.68.141.91 a=candidate:829852397 1 udp 2130714367 192.168.1.31 51703 typ host generation 0 a=candidate:829852397 2 udp 2130714367 192.168.1.31 51703 typ host generation 0 a=candidate:3345412921 1 udp 1912610559 182.68.141.91 51704 typ srflx generation 0 a=candidate:3345412921 2 udp 1912610559 182.68.141.91 51704 typ srflx generation 0 a=candidate:2146792989 1 tcp 1694506751 192.168.1.31 50295 typ host generation 0 a=candidate:2146792989 2 tcp 1694506751 192.168.1.31 50295 typ host generation 0 a=ice-ufrag:Jz1C79lLNJodMTz2 a=ice-pwd:UsowydSx3Uy9ehq8lKyR+JmJ a=mid:audio a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ccVX0xSFpsuRG8lnNQgnSiMdckCgtHs5SET8r9Jt a=rtpmap:103 ISAC/16000 a=rtpmap:104 ISAC/32000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:106 CN/32000 a=rtpmap:105 CN/16000 a=rtpmap:13 CN/8000 a=rtpmap:126 telephone-event/8000 a=ssrc:862982555 cname:QP1jAi3c1C/sIjMM a=ssrc:862982555 mslabel:JTCmb2lovY15B6rH1xoGTpSOwige5Py22GgE a=ssrc:862982555 label:JTCmb2lovY15B6rH1xoGTpSOwige5Py22GgE00 m=video 51704 RTP/SAVPF 100 101 102 c=IN IP4 182.68.141.91 a=rtcp:51704 IN IP4 182.68.141.91 a=candidate:829852397 1 udp 2130714367 192.168.1.31 51703 typ host generation 0 a=candidate:829852397 2 udp 2130714367 192.168.1.31 51703 typ host generation 0 a=candidate:3345412921 1 udp 1912610559 182.68.141.91 51704 typ srflx generation 0 a=candidate:3345412921 2 udp 1912610559 182.68.141.91 51704 typ srflx generation 0 a=candidate:2146792989 1 tcp 1694506751 192.168.1.31 50295 typ host generation 0 a=candidate:2146792989 2 tcp 1694506751 192.168.1.31 50295 typ host generation 0 a=ice-ufrag:Jz1C79lLNJodMTz2 a=ice-pwd:UsowydSx3Uy9ehq8lKyR+JmJ a=mid:video a=rtcp-mux a=crypto:1 AES_CM_128_HMAC_SHA1_80 inline:ccVX0xSFpsuRG8lnNQgnSiMdckCgtHs5SET8r9Jt a=rtpmap:100 VP8/90000 a=rtpmap:101 red/90000 a=rtpmap:102 ulpfec/90000 a=ssrc:1143670773 cname:QP1jAi3c1C/sIjMM a=ssrc:1143670773 mslabel:JTCmb2lovY15B6rH1xoGTpSOwige5Py22GgE a=ssrc:1143670773 label:JTCmb2lovY15B6rH1xoGTpSOwige5Py22GgE10 2012-07-17 14:48:02.285368 [ERR] sofia_glue.c:4673 a=crypto in RTP/AVP, refer to rfc3711 2012-07-17 14:48:02.285368 [DEBUG] switch_channel.c:2848 (sofia/internal/ 1015 at 122.180.97.198) Callstate Change DOWN -> HANGUP 2012-07-17 14:48:02.285368 [NOTICE] sofia.c:5813 Hangup sofia/internal/ 1015 at 122.180.97.198 [CS_NEW] [INCOMPATIBLE_DESTINATION] regards, Anita -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120717/5d0168dd/attachment.html From jalsot at gmail.com Tue Jul 17 14:17:02 2012 From: jalsot at gmail.com (Tamas Jalsovszky) Date: Tue, 17 Jul 2012 12:17:02 +0200 Subject: [Freeswitch-users] Webrtc Message-ID: Hello, Is there any plan to support webrtc in FS? Right I've found this post about asterisk: http://blogs.digium.com/2012/05/23/asterisk-11-webrtc/ T. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120717/dc03a3f2/attachment.html From vallimamod.abdullah at imtelecom.fr Tue Jul 17 14:54:42 2012 From: vallimamod.abdullah at imtelecom.fr (Vallimamod ABDULLAH) Date: Tue, 17 Jul 2012 12:54:42 +0200 Subject: [Freeswitch-users] Need help in IVR In-Reply-To: References: Message-ID: <96C4D4B3-7C57-458B-9E5D-146B84F2E8AF@imtelecom.fr> hi, On Jul 17, 2012, at 7:39 AM, Piyush Sharma wrote: > Hello Everyone, > > I am trying to create an IVR with menu and sub menus, but getting following error when I make the call, > 2012-07-17 11:05:37.755034 [ERR] switch_ivr_menu.c:870 Unable to build xml menu > 2012-07-17 11:05:37.755034 [ERR] switch_ivr_menu.c:870 Unable to build xml menu > 2012-07-17 11:05:37.755034 [ERR] mod_dptools.c:1774 Unable to create menu > > I am new to this, so there might be little or a big error. > > My XML contents in ...conf/ivr_menus is > > > greet-long="phrase:ps_main" > greet-short="phrase:ps_short" > exit-sound="ivr/good_bye.wav" > invalid-sound="ivr/ivr-not_available.wav" > tts-engine="flite" > tts-voice="rms" > timeout="3000" > max-failures="2" > ; phrase_lang="en" -------^^ This is strange: ";" is not a valid sign in xml for comments. All comments must be between "". Also, you cannot put a comment inside a tag. Try to remove it and test again. Best Regards, -vma . From piyush.sharma at coraltele.com Tue Jul 17 16:03:38 2012 From: piyush.sharma at coraltele.com (Piyush Sharma) Date: Tue, 17 Jul 2012 17:33:38 +0530 (IST) Subject: [Freeswitch-users] Need help in IVR In-Reply-To: Message-ID: There were no semicolon in my XML, but somehow they are visible here it might be my mistake, so please ignore them. ----- Original Message ----- From: "Piyush Sharma" To: "FreeSWITCH Users Help" Sent: Tuesday, July 17, 2012 11:09:27 AM Subject: [Freeswitch-users] Need help in IVR Hello Everyone, I am trying to create an IVR with menu and sub menus, but getting following error when I make the call, 2012-07-17 11:05:37.755034 [ERR] switch_ivr_menu.c:870 Unable to build xml menu 2012-07-17 11:05:37.755034 [ERR] switch_ivr_menu.c:870 Unable to build xml menu 2012-07-17 11:05:37.755034 [ERR] mod_dptools.c:1774 Unable to create menu I am new to this, so there might be little or a big error. My XML contents in ... conf/ivr_menus is and in .../en/demo is &n bsp; ; Thanks everyone, thanks for this support forum !!! -- Regards, Piyush Sharma. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120717/bb96d20a/attachment.html From shaik.bawajan at gmail.com Tue Jul 17 16:54:33 2012 From: shaik.bawajan at gmail.com (bawajan) Date: Tue, 17 Jul 2012 05:54:33 -0700 (PDT) Subject: [Freeswitch-users] A fatal error has been detected by the Java Runtime Environment: Message-ID: <1342529673372-7580878.post@n2.nabble.com> Hi, Am using ESL inbound connection to make calls and have a below flow (written in java) a) originating 15 calls simultaneously with park function b) play audio file c) creating new thread and passing eslconnection and playing a IVR in originate call. Here while handling events getting the below error : # A fatal error has been detected by the Java Runtime Environment: INFO | jvm 3 | 2012/07/17 17:54:36 | # INFO | jvm 3 | 2012/07/17 17:54:36 | # SIGSEGV (0xb) at pc=0x00007fc01f1d3943, pid=14759, tid=140463084390144 INFO | jvm 3 | 2012/07/17 17:54:36 | # INFO | jvm 3 | 2012/07/17 17:54:36 | # JRE version: 6.0_25-b06 INFO | jvm 3 | 2012/07/17 17:54:36 | # Java VM: Java HotSpot(TM) 64-Bit Server VM (20.0-b11 mixed mode linux-amd64 compressed oops) INFO | jvm 3 | 2012/07/17 17:54:36 | # Problematic frame: INFO | jvm 3 | 2012/07/17 17:54:36 | # C [libesljni.so+0xd943] long double+0x183 INFO | jvm 3 | 2012/07/17 17:54:36 | # INFO | jvm 3 | 2012/07/17 17:54:36 | # An error report file with more information is saved as: INFO | jvm 3 | 2012/07/17 17:54:36 | # /usr/local/freeswitch/hs_err_pid14759.log INFO | jvm 3 | 2012/07/17 17:54:36 | # INFO | jvm 3 | 2012/07/17 17:54:36 | # If you would like to submit a bug report, please visit: INFO | jvm 3 | 2012/07/17 17:54:36 | # http://java.sun.com/webapps/bugreport/crash.jsp INFO | jvm 3 | 2012/07/17 17:54:36 | # The crash happened outside the Java Virtual Machine in native code. INFO | jvm 3 | 2012/07/17 17:54:36 | # See problematic frame for where to report the bug. plz let me know, where am doing mistake and how to resolve it. Thanks in advance. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/A-fatal-error-has-been-detected-by-the-Java-Runtime-Environment-tp7580878.html Sent from the freeswitch-users mailing list archive at Nabble.com. From piyush.sharma at coraltele.com Tue Jul 17 16:51:12 2012 From: piyush.sharma at coraltele.com (Piyush Sharma) Date: Tue, 17 Jul 2012 18:21:12 +0530 (IST) Subject: [Freeswitch-users] Need help in IVR In-Reply-To: <96C4D4B3-7C57-458B-9E5D-146B84F2E8AF@imtelecom.fr> Message-ID: <376e7ac7-41ff-4add-84fc-57c6f4dbfb7b@mail.coraltele.com> Thanks for the response, I don't know how semicolon came in the email, but I have no semicolon in my XML, and please avoid visible smiles as well. ----- Original Message ----- From: "Vallimamod ABDULLAH" To: "FreeSWITCH Users Help" Sent: Tuesday, July 17, 2012 4:24:42 PM Subject: Re: [Freeswitch-users] Need help in IVR hi, On Jul 17, 2012, at 7:39 AM, Piyush Sharma wrote: > Hello Everyone, > > I am trying to create an IVR with menu and sub menus, but getting following error when I make the call, > 2012-07-17 11:05:37.755034 [ERR] switch_ivr_menu.c:870 Unable to build xml menu > 2012-07-17 11:05:37.755034 [ERR] switch_ivr_menu.c:870 Unable to build xml menu > 2012-07-17 11:05:37.755034 [ERR] mod_dptools.c:1774 Unable to create menu > > I am new to this, so there might be little or a big error. > > My XML contents in ...conf/ivr_menus is > > > greet-long="phrase:ps_main" > greet-short="phrase:ps_short" > exit-sound="ivr/good_bye.wav" > invalid-sound="ivr/ivr-not_available.wav" > tts-engine="flite" > tts-voice="rms" > timeout="3000" > max-failures="2" > ; phrase_lang="en" -------^^ This is strange: ";" is not a valid sign in xml for comments. All comments must be between "". Also, you cannot put a comment inside a tag. Try to remove it and test again. Best Regards, -vma . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120717/34f9f95f/attachment.html From gcd at i.ph Tue Jul 17 17:18:28 2012 From: gcd at i.ph (Nandy Dagondon) Date: Tue, 17 Jul 2012 21:18:28 +0800 Subject: [Freeswitch-users] Need help in IVR In-Reply-To: <376e7ac7-41ff-4add-84fc-57c6f4dbfb7b@mail.coraltele.com> References: <96C4D4B3-7C57-458B-9E5D-146B84F2E8AF@imtelecom.fr> <376e7ac7-41ff-4add-84fc-57c6f4dbfb7b@mail.coraltele.com> Message-ID: how about this line? wrote: > > Thanks for the response, I don't know how semicolon came in the email, but > I have no semicolon in my XML, and please avoid visible smiles as well. > > > greet-long="phrases_main" > greet-short="phrases_short" > exit-sound="ivr/good_bye.wav" > invalid-sound="ivr/ivr-not_available.wav" > t ts-engine="flite" > > tts-voice="rms" > timeout="3000" > max-failures="2" > phrase_lang="en" > digit-len="1"> > > > > > greet-long="phrases1_main" > greet-short="phrases1_short" > exit-sound="ivr/good_bye.wav" > invalid-sound="ivr/ivr-not_available.wav" > timeout ="3000" > > max-failure="3"> > > > > > > > ------------------------------ > *From: *"Vallimamod ABDULLAH" > > *To: *"FreeSWITCH Users Help" > *Sent: *Tuesday, July 17, 2012 4:24:42 PM > *Subject: *Re: [Freeswitch-users] Need help in IVR > > > hi, > > On Jul 17, 2012, at 7:39 AM, Piyush Sharma wrote: > > > Hello Everyone, > > > > I am trying to create an IVR with menu and sub menus, but getting > following error when I make the call, > > 2012-07-17 11:05:37.755034 [ERR] switch_ivr_menu.c:870 Unable to build > xml menu > > 2012-07-17 11:05:37.755034 [ERR ] switch_ivr_menu.c:870 Unable to build > xml menu > > 2012-07-17 11:05:37.755034 [ERR] mod_dptools.c:1774 Unable to create menu > > > > I am new to this, so there might be little or a big error. > > > > My XML contents in ...conf/ivr_menus is > > > > > > > greet-long="phrase:ps_main" > > greet-short="phrase:ps_short" > > exit-sound="ivr/good_bye.wav" > > invalid-sound="ivr/ivr-not_available.wav" > > tts-engine="flite" > > tts-voice="rms" > > timeout="3000" > > max-failures="2" > > ; phrase_lang="en" > > -------^^ > This is strange: ";" is not a valid sign in xml for comments. All comments > must be between "". Also, you cannot put a comment inside a > tag. > Try to remove it and test again. > > Best Regards, > -vma > . > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120717/11f768da/attachment.html From mstockton at harqen.com Tue Jul 17 17:27:38 2012 From: mstockton at harqen.com (Matt Stockton) Date: Tue, 17 Jul 2012 08:27:38 -0500 Subject: [Freeswitch-users] How can I use fs_path when originating an external call through a gateway? Message-ID: Hi all, I am trying to do the following: Originate a call from Freeswitch to an external gateway. I want the SIP signaling to go through a different Freeswitch box that is acting as an outbound proxy. It seems that I should be able to use fs_path to do this, however I am having trouble getting this to work. I have tried the following originate commands, and for each one, the SIP signaling is not going through the other_fs box. Note that the originate is working just fine, its just going directly to the gateway: originate sofia/gateway/my_gateway/+14145551212;fs_path=sip:other_fs &park() originate sofia/gateway/my_gateway/+14145551212;fs_path=sip:other_fs:5060 &park() originate 'sofia/gateway/my_gateway/+14145551212;fs_path=sip:other_fs' &park() originate 'sofia/gateway/my_gateway/+14145551212;fs_path=sip:other_fs:5060' &park() Am I doing something wrong with fs_path? My goal is to have a FS node (the other_fs) act as an outbound proxy to the gateway so that the gateway sees all SIP signaling traffic come from other_fs. Currently, I am using OpenSIPS to act as the outbound proxy but ideally I would just have Freeswitch nodes, so I am trying to remove OpenSIPS from the SIP path. Any help on this is appreciated! - Matt -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120717/65a292c4/attachment.html From Joann.DeJesus at convergys.com Tue Jul 17 07:54:44 2012 From: Joann.DeJesus at convergys.com (Joann R DeJesus) Date: Mon, 16 Jul 2012 23:54:44 -0400 Subject: [Freeswitch-users] Duplicate DTMF Events - Please Help! In-Reply-To: References: Message-ID: Hi Brian, Thank you for taking the time to respond. We are using Sangoma cards with Freeswitch as a gateway between the PSTN and our internal network which uses SIP. We prefer not to use inband with SIP due to numerous other potential issues, and are trying to determine why FS sometimes creates double DTMF events. Thanks for the tip though! Joann -----Original message----- From: Brian Foster To: FreeSWITCH Users Help Sent: Mon, Jul 16, 2012 23:34:17 EDT Subject: Re: [Freeswitch-users] Duplicate DTMF Events - Please Help! Rfc2833 is for sip, not for analog. Other than that I don't have much to offer. Try using inband only. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 16, 2012 6:50 PM, "Joann R DeJesus" > wrote: Hello All, I hope someone has experience this issue and can quickly assist me! We are supposed to be going live with a customer in a couple of days and need to find a resolution quickly in order to stay on track? We are using Freeswitch with Sangoma T1 cards to bridge TDM with our internal VoIP system. Call flow is as follows: PSTN --> T1/Sangoma cards --> Freeswitch --> IVR I have configured Freeswitch for RFC2833. When I make a call to the IVR from a landline or cell phone and press "5" at a prompt, FS will sometimes send two separate DTMF events to the IVR. This causes the IVR to receive "55" which then sends the call to the incorrect destination. I can verify with Wireshark that FS is sending two separate RFC2833 events. For some reason, this is exacerbated with heavy call volume (easy to reproduce during load testing). Has anyone experienced this symptom or have any ideas what I can check? Thanks in advance for any assistance!! - Joann ________________________________ NOTICE: The information contained in this electronic mail transmission is intended by Convergys Corporation for the use of the named individual or entity to which it is directed and may contain information that is privileged or otherwise confidential. If you have received this electronic mail transmission in error, please delete it from your system without copying or forwarding it, and notify the sender of the error by reply email or by telephone (collect), so that the sender's address records can be corrected. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ NOTICE: The information contained in this electronic mail transmission is intended by Convergys Corporation for the use of the named individual or entity to which it is directed and may contain information that is privileged or otherwise confidential. If you have received this electronic mail transmission in error, please delete it from your system without copying or forwarding it, and notify the sender of the error by reply email or by telephone (collect), so that the sender's address records can be corrected. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120716/52297f25/attachment-0001.html From saulius.kibilda at lantel.lt Tue Jul 17 15:24:42 2012 From: saulius.kibilda at lantel.lt (Saulius Kibilda) Date: Tue, 17 Jul 2012 11:24:42 +0000 Subject: [Freeswitch-users] Not droping calls after BUSY signal Message-ID: Hi All I have Dinstar GSM gateway and configured like in this how to -> http://wiki.freeswitch.org/wiki/Dinstar_GSM_gateway_FreeSwitch_HowTo Everything works well except when calling from an IP phone to a mobile phone and mobile phone dropps the call. Freeswitch does not understand the BUSY signal and calls again sending INVITE and making new call to mobile phone. Hare is SIP debug http://pastebin.freeswitch.org/19539 BTW: in Freeswitch 1.1-beta I not observed this problem Dinstar with Asterisk works fine, so problem i think not in Dinstar gateway. Saulius -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120717/0109a3b9/attachment.html From brett at launch3.net Tue Jul 17 19:39:01 2012 From: brett at launch3.net (Brett Wilson) Date: Tue, 17 Jul 2012 11:39:01 -0400 Subject: [Freeswitch-users] setting max-forwards? Message-ID: <01ba01cd6432$4a6c3400$df449c00$@launch3.net> Hello, Can anyone tell me how and where to set max-forwards? I can't find much of anything on the net. I get exchange routing errors and sometimes calls will not forward to an extension. Also, On another install, emailing the voicemail does not work. The settings seem all ok, the log says emailing, but I did some snooping with filesystem monitoring and the script to send the email is never touched by freeswitch in any way. Where should I start diagnosing this? ******************************************* Brett Wilson IT Department Launch 3 Ventures, LLC 134 Myer Street Hackensack, NJ 07601 Phone: 877.878.9134 Fax: 646.536.3866 Email: Brett.Wilson at launch3.net AOL IM: Brett.Wilson at launch3.net www.Launch3.net www.Launch3telecom.com ******************************************* Description: Description: Description: Blogger-logo Description: Description: Description: FaceBook-Logo Description: Description: Description: Twitter-Logo Description: Description: Description: GPlus-Logo -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/png Size: 3063 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120717/c019b92d/attachment-0007.png From bdfoster at endigotech.com Tue Jul 17 19:53:07 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 17 Jul 2012 11:53:07 -0400 Subject: [Freeswitch-users] setting max-forwards? In-Reply-To: <01ba01cd6432$4a6c3400$df449c00$@launch3.net> References: <01ba01cd6432$4a6c3400$df449c00$@launch3.net> Message-ID: Your email issue is probably a permissions issue. Check to see.if you have the proper permissions on the script and make sure its executable. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 17, 2012 11:40 AM, "Brett Wilson" wrote: > Hello,**** > > Can anyone tell me how and where to set max-forwards?**** > > I can?t find much of anything on the net.**** > > I get exchange routing errors and sometimes calls will not forward to an > extension.**** > > ** ** > > Also,**** > > On another install, emailing the voicemail does not work. The settings > seem all ok, the log says emailing, but I did some snooping with filesystem > monitoring and the script to send the email is never touched by freeswitch > in any way. Where should I start diagnosing this?**** > > ** ** > > ******************************************* > *Brett Wilson* > *IT Department* > *Launch 3 Ventures, LLC* > 134 Myer Street > Hackensack, NJ 07601 > *Phone:* 877.878.9134 > *Fax:* 646.536.3866 > *Email:* Brett.Wilson at launch3.net > *AOL IM:* Brett.Wilson at launch3.net > www.Launch3.net > *www.Launch3telecom.com * > ******************************************* > [image: Description: Description: Description: Blogger-logo][image: > Description: Description: Description: FaceBook-Logo][image: > Description: Description: Description: Twitter-Logo][image: > Description: Description: Description: GPlus-Logo] > **** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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... 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Name: not available Type: image/png Size: 3063 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120717/b90c3c07/attachment-0003.png From msc at freeswitch.org Tue Jul 17 20:08:57 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Jul 2012 09:08:57 -0700 Subject: [Freeswitch-users] Need French-Candian translator Message-ID: Bonjour! We could use some assistance with a few sounds that need to be translated from English to French. These will be prompts that are recorded and made available to the community at large. Please email me off list if you are able to assist with translating about 20 or so phrases. Thanks, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120717/112a433a/attachment-0001.html From mario_fs at mgtech.com Tue Jul 17 20:54:51 2012 From: mario_fs at mgtech.com (Mario G) Date: Tue, 17 Jul 2012 09:54:51 -0700 Subject: [Freeswitch-users] how to get leg_delay_start to work for bridge enterprise Message-ID: <8645413F-9FD0-4E9D-BE4A-A2FC75591A97@mgtech.com> I am try to delay the second target by 20 seconds. I used [..] and {..} but no dice. The wiki has them both for enterprise, can someone shed light on what's wrong, thanks. Main is supposed to ring, 20 secs later the second target is added while main keeps going, all timeout after a total of 43 secs. Mario G -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120717/2a7764c0/attachment.html From adrottenberg at gmail.com Tue Jul 17 22:01:20 2012 From: adrottenberg at gmail.com (Duvid Rottenberg) Date: Tue, 17 Jul 2012 14:01:20 -0400 Subject: [Freeswitch-users] Username in SUBSCRIBE Request URI In-Reply-To: References: Message-ID: Apparently this is not currently possible. I have submitted a patch on JIRA to add a new parameter to the subscriptions section named user-in-register, when set to true the username will be included in the request-uri. If/when this patch is approved I will update the wiki. On Fri, Jul 13, 2012 at 1:34 PM, Duvid Rottenberg wrote: > I am using embedded freeswitch as a softphone client and I am trying to > subscribe to call-info on the server, (see config below) but the server is > responding with a 481 Call/Transaction Does not exist. > I compared the freeswitch SIP messages with SIP messages sent by a polycom > phone for this feature and I noticed that freeswitch doesn't send the > username in the request line. I think that this is causing the 481 response. > > Polycom Version: > SUBSCRIBE sip:user at server:5060;transport=udp SIP/2.0 > Freeswitch: > SUBSCRIBE sip:server:5060;transport=udp SIP/2.0 > > Below is my gateway configuration > > > > > > > > > > > > > > > > > Is there any way to tell freeswitch to include the username in the request > line? > > Thank You, > Duvid Rottenberg > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120717/d6dc6a74/attachment.html From eugeneazuka at gmail.com Tue Jul 17 20:33:56 2012 From: eugeneazuka at gmail.com (Eugene Azuka) Date: Tue, 17 Jul 2012 17:33:56 +0100 Subject: [Freeswitch-users] Central FreeSWITCH nodes management with Mod_XML_Curl vs Mod_Lua Message-ID: Hi FreeSWITCH experts, My question is as regards performance and which is the best option. I am trying to scale and manage multiple nodes of FreeSWITCH centrally from my external web server/site using Mod_XML_Curl, But someone more experience than me says otherwise that Mod_XML_Curl may not be the best option: Here is his statement below: *" The thing I don't like about mod_xml_curl is that to scale it you have to scale the web server with more listeners. If you run out of listeners your ability to answer new calls fails. If the web server crashes your ability to handle calls is gone.* * * *Instead, I use Lua which is embedded into FreeSWITCH as the XML handler. The Lua reads the info from the database and hands it off to FreeSWITCH. This approach is not dependent on a service that can fail, or that can run out of listeners. "* What do you experts think about this comment above? Wouldn't Lua reading directly from database still face some of the issues i may be running from using mod_xml_curl, like database crashing, slow reading from database? Unlike using mod_xml_curl whereby i can optimise my code to make use of memcached or radis to reduce some calls to database, can i still do such with mod_lua? So what do experts think, should i go with Mod_Lua approach just as he said above or should i continue with using Mod_XML_Curl? You opinion will be appreciated. Regards Eugene -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120717/b3af42c3/attachment.html From vipkilla at gmail.com Tue Jul 17 23:30:53 2012 From: vipkilla at gmail.com (Vik Killa) Date: Tue, 17 Jul 2012 15:30:53 -0400 Subject: [Freeswitch-users] Central FreeSWITCH nodes management with Mod_XML_Curl vs Mod_Lua In-Reply-To: References: Message-ID: I dont understand what they meant by 'listeners' AFAIK if the web server process is running it will always return the XML From msc at freeswitch.org Wed Jul 18 00:56:11 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 17 Jul 2012 13:56:11 -0700 Subject: [Freeswitch-users] Central FreeSWITCH nodes management with Mod_XML_Curl vs Mod_Lua In-Reply-To: References: Message-ID: On Tue, Jul 17, 2012 at 12:30 PM, Vik Killa wrote: > I dont understand what they meant by 'listeners' > AFAIK if the web server process is running it will always return the XML > Last time I checked, there were one or two websites out there that could handle thousands of requests per second. The web has given birth to tools that allow for many, many concurrent requests. The LAMP stack on a beefy machine can do quite a lot of traffic. Furthermore, MySQL/Postgres/et al all have backup/redundancy/HA options built in, as does Apache/HTTP. It seems to me that you could scale farther and have more redundancy using these time-tested tools. Just my $0.02. -MC P.S. - Don't get me wrong - I really like Lua. I just don't know if it's really a "better" solution to this problem. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120717/44aa7e95/attachment.html From toddb at toddbailey.net Wed Jul 18 02:37:51 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Tue, 17 Jul 2012 15:37:51 -0700 Subject: [Freeswitch-users] Looking for dial plan examples for FS and SPA3102 router Message-ID: <1342564671.3414.63.camel@mythtv> Hi All, I'm having issues getting a dial plan to work on FS and a Cisco SPA 3102 router. when I dial 0, 1 or 9 plus a 10 number, I get to the router's dial tone but I have to reenter the number I want to connect to. The expected action is to only need to enter the number to dial one Can some one provide dial plan and/or other config file example on how to resolve this issue? here is what I have so far: /usr/local/freeswitch/conf/dialplan/default.xml /usr/local/freeswitch/conf/dialplan/default/00_spa3102.xml From gabe at gundy.org Wed Jul 18 02:40:41 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Tue, 17 Jul 2012 16:40:41 -0600 Subject: [Freeswitch-users] Central FreeSWITCH nodes management with Mod_XML_Curl vs Mod_Lua In-Reply-To: References: Message-ID: Comments inline: On Tue, Jul 17, 2012 at 10:33 AM, Eugene Azuka wrote: > " The thing I don't like about mod_xml_curl is that to scale it you have to > scale the web server with more listeners. If you run out of listeners your > ability to answer new calls fails. If the web server crashes your ability to > handle calls is gone. Scaling web servers is an easy, well known and a common problem to solve. Yes, you have to make sure you have the resources, but this it true of any technology out there. > Instead, I use Lua which is embedded into FreeSWITCH as the XML handler. The > Lua reads the info from the database and hands it off to FreeSWITCH. This > approach is not dependent on a service that can fail, or that can run out of > listeners. " This is a great approach, but I don't think the logic is sound. Lua (if it's doing anything fancy) will run out of resources that it depends on -- memory, database connections, sockets etc. If one accepts that they'll have to build out to *really* scale something, they'll also come to appreciate the fact that they can move the HTTP stack to another box when needed (or 50 other boxen if required). The Lua (or any other embedded language) is pretty well tied to that same box. In the end, it probably doesn't matter. If you end up scaling to the ends of the Earth, you'll have to rewrite it anyway. Just give thanks that you've found FreeSWITCH and it was flexible enough to give you the amazing configurability needed to build it in anyway you like :) > What do you experts think about this comment above? There are experts on this list now?! Awesome ;) > Wouldn't Lua reading directly from database still face some of the issues i > may be running from using mod_xml_curl, like database crashing, slow reading > from database? Yep, see above. > Unlike using mod_xml_curl whereby i can optimise my code to make use of > memcached or radis to reduce some calls to database, can i still do such > with mod_lua? If you want and Lua supports it. > So what do experts think, should i go with Mod_Lua approach just as he said > above or should i continue with using Mod_XML_Curl? Your choice! Happy hacking! Best, Gabe From kris at kriskinc.com Wed Jul 18 04:07:34 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 17 Jul 2012 20:07:34 -0400 Subject: [Freeswitch-users] Central FreeSWITCH nodes management with Mod_XML_Curl vs Mod_Lua In-Reply-To: References: Message-ID: In my testing from a few months ago mod_xml_curl provided the best performance with FreeSWITCH (in terms of CPS). Additionally, as others have noted, scaling HTTP is a well known problem. There are very few cases where mod_xml_curl and your preferred web technology would not be the ideal solution. On Tue, Jul 17, 2012 at 12:33 PM, Eugene Azuka wrote: > Hi FreeSWITCH experts, > > My question is as regards performance and which is the best option. > > I am trying to scale and manage multiple nodes of FreeSWITCH centrally from > my external web server/site using Mod_XML_Curl, But someone more experience > than me says otherwise that Mod_XML_Curl may not be the best option: Here is > his statement below: > > " The thing I don't like about mod_xml_curl is that to scale it you have to > scale the web server with more listeners. If you run out of listeners your > ability to answer new calls fails. If the web server crashes your ability to > handle calls is gone. > > Instead, I use Lua which is embedded into FreeSWITCH as the XML handler. The > Lua reads the info from the database and hands it off to FreeSWITCH. This > approach is not dependent on a service that can fail, or that can run out of > listeners. " > > > What do you experts think about this comment above? > > Wouldn't Lua reading directly from database still face some of the issues i > may be running from using mod_xml_curl, like database crashing, slow reading > from database? > > Unlike using mod_xml_curl whereby i can optimise my code to make use of > memcached or radis to reduce some calls to database, can i still do such > with mod_lua? > > > So what do experts think, should i go with Mod_Lua approach just as he said > above or should i continue with using Mod_XML_Curl? > > You opinion will be appreciated. > > Regards > > Eugene > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 From luis.daniel.lucio at gmail.com Wed Jul 18 06:59:30 2012 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Tue, 17 Jul 2012 21:59:30 -0500 Subject: [Freeswitch-users] Strange 400 with some clients Message-ID: Using FS 1.0, I'm having problems registering some devices, Android 4.0.4, Linphone 3.4.0 under Wifi works Android 2.3.4, Linphone 3.4.0 under Wifi works it fails here iPhone IOS 5.1.1, Linphone 3.5.2 under Wifi fails iPod IOS 5.1.1, Linphone 3.5.2 under Wifi fails As far as i seen im getting error 400 when IOS is trying to register here is failling dump REGISTER sip:corporate.xxxx.com SIP/2.0 Via: SIP/2.0/UDP 189.146.85.112:10649;rport;branch=z9hG4bK842453785 From: ;tag=1058009119 To: Call-ID: 508811206 CSeq: 19 REGISTER Contact: Max-Forwards: 70 User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) Expires: 600 Content-Length: 0 SIP/2.0 400 Bad Contact Header Via: SIP/2.0/UDP 189.146.85.112:10649;rport=10649;branch=z9hG4bK842453785 From: ;tag=1058009119 To: ;tag=1FvU4KjDH0ggg Call-ID: 508811206 CSeq: 19 REGISTER Content-Length: 0 here is workig request REGISTER sip:corporate.xxxx.com SIP/2.0 Via: SIP/2.0/UDP 189.146.85.112:5060;rport;branch=z9hG4bK86735004 From: ;tag=134885940 To: Call-ID: 591954255 CSeq: 2 REGISTER Contact: Authorization: Digest username="104", realm="corporate.xxxx.com", nonce="f418e392-b7d1-43e9-98d5-b841134ad19c", uri="sip:corporate.xxxx.com", response="f599fa83cb2660595d61bb3cadba6c5b", algorithm=MD5, cnonce="0a4f113b", qop=auth, nc=00000001 Max-Forwards: 70 User-Agent: Linphone/3.4.0 (eXosip2/unknown) Expires: 3600 Content-Length: 0 SIP/2.0 200 OK Via: SIP/2.0/UDP 189.146.85.112:5060;rport=5060;branch=z9hG4bK86735004 From: ;tag=134885940 To: ;tag=Xc0Hp99F0S0Um Call-ID: 591954255 CSeq: 2 REGISTER Contact: ;expires=3600 Date: Wed, 18 Jul 2012 02:54:14 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-261505a 2012-02-13 13-50-21 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 Just wondering how to workarround this regards LD From jbaclor at ezuce.com Wed Jul 18 07:20:54 2012 From: jbaclor at ezuce.com (Joegen Baclor) Date: Wed, 18 Jul 2012 11:20:54 +0800 Subject: [Freeswitch-users] Strange 400 with some clients In-Reply-To: References: Message-ID: <50062B96.9070706@ezuce.com> Contact: There are two ports (10649) in the contact uri. On 07/18/2012 10:59 AM, Luis Daniel Lucio Quiroz wrote: > Using FS 1.0, I'm having problems registering some devices, > > Android 4.0.4, Linphone 3.4.0 under Wifi works > Android 2.3.4, Linphone 3.4.0 under Wifi works > > it fails here > > iPhone IOS 5.1.1, Linphone 3.5.2 under Wifi fails > iPod IOS 5.1.1, Linphone 3.5.2 under Wifi fails > > As far as i seen im getting error 400 when IOS is trying to register > > > here is failling dump > > REGISTER sip:corporate.xxxx.com SIP/2.0 > Via: SIP/2.0/UDP 189.146.85.112:10649;rport;branch=z9hG4bK842453785 > From: ;tag=1058009119 > To: > Call-ID: 508811206 > CSeq: 19 REGISTER > Contact: > Max-Forwards: 70 > User-Agent: Linphone/3.5.2 (eXosip2/3.6.0) > Expires: 600 > Content-Length: 0 > > > > SIP/2.0 400 Bad Contact Header > Via: SIP/2.0/UDP > 189.146.85.112:10649;rport=10649;branch=z9hG4bK842453785 > From: ;tag=1058009119 > To: ;tag=1FvU4KjDH0ggg > Call-ID: 508811206 > CSeq: 19 REGISTER > Content-Length: 0 > > here is workig request > > > REGISTER sip:corporate.xxxx.com SIP/2.0 > Via: SIP/2.0/UDP 189.146.85.112:5060;rport;branch=z9hG4bK86735004 > From: ;tag=134885940 > To: > Call-ID: 591954255 > CSeq: 2 REGISTER > Contact: > Authorization: Digest username="104", > realm="corporate.xxxx.com", > nonce="f418e392-b7d1-43e9-98d5-b841134ad19c", > uri="sip:corporate.xxxx.com", > response="f599fa83cb2660595d61bb3cadba6c5b", algorithm=MD5, > cnonce="0a4f113b", qop=auth, nc=00000001 > Max-Forwards: 70 > User-Agent: Linphone/3.4.0 (eXosip2/unknown) > Expires: 3600 > Content-Length: 0 > > > > SIP/2.0 200 OK > Via: SIP/2.0/UDP 189.146.85.112:5060;rport=5060;branch=z9hG4bK86735004 > From: ;tag=134885940 > To: ;tag=Xc0Hp99F0S0Um > Call-ID: 591954255 > CSeq: 2 REGISTER > Contact: ;expires=3600 > Date: Wed, 18 Jul 2012 02:54:14 GMT > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-261505a > 2012-02-13 13-50-21 -0600 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, > INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > > Just wondering how to workarround this > > regards > > > LD > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From luis.daniel.lucio at gmail.com Wed Jul 18 08:02:29 2012 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Tue, 17 Jul 2012 23:02:29 -0500 Subject: [Freeswitch-users] Strange 400 with some clients In-Reply-To: <50062B96.9070706@ezuce.com> References: <50062B96.9070706@ezuce.com> Message-ID: and how can i fix this? I know it is a linphone issue, but i wonder if there is a workarround in server side. 2012/7/17 Joegen Baclor : > Contact: > > > There are two ports (10649) in the contact uri. From piyush.sharma at coraltele.com Wed Jul 18 09:18:46 2012 From: piyush.sharma at coraltele.com (Piyush Sharma) Date: Wed, 18 Jul 2012 10:48:46 +0530 (IST) Subject: [Freeswitch-users] Need help in IVR In-Reply-To: Message-ID: Hello Nandy, I didn't get you completely, I named my main menu "ps" it is just a test case, I want to prepare a IVR, that may have sub menus upto some level, so I will name menus & sub menus with some logic so that it is easy to maintain, If it is possible, I want to generate it automatically via a script on the basis of some input. I am new to this so your little support can help me a lot. If you are not clear, tell me I will explain more about this. (I am sick of visibility, it is not clear again.) ----- Original Message ----- From: "Nandy Dagondon" To: "FreeSWITCH Users Help" Sent: Tuesday, July 17, 2012 6:48:28 PM Subject: Re: [Freeswitch-users] Need help in IVR how about this line? wrote: Thanks for the response, I don't know how semicolon came in the email, but I have no semicolon in my XML, and please avoid visible smiles as well. greet-long="phrase s1_main" greet-short="phrase s1_short" exit-sound="ivr/good_bye.wav" invalid-sound="ivr/ivr-not_available.wav" timeout ="3000" max-failure="3"> From: "Vallimamod ABDULLAH" < vallimamod.abdullah at imtelecom.fr > To: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > Sent: Tuesday, July 17, 2012 4:24:42 PM Subject: Re: [Freeswitch-users] Need help in IVR hi, On Jul 17, 2012, at 7:39 AM, Piyush Sharma wrote: > Hello Everyone, > > I am trying to create an IVR with menu and sub menus, but getting following error when I make the call, > 2012-07-17 11:05:37.755034 [ERR] switch_ivr_menu.c:870 Unable to build xml menu > 2012-07-17 11:05:37.755034 [ERR ] switch_ivr_menu.c:870 Unable to build xml menu > 2012-07-17 11:05:37.755034 [ERR] mod_dptools.c:1774 Unable to create menu > > I am new to this, so there might be little or a big error. > > My XML contents in ...conf/ivr_menus is > > > greet-long="phrase:ps_main" > greet-short="phrase:ps_short" > exit-sound="ivr/good_bye.wav" > invalid-sound="ivr/ivr-not_available.wav" > tts-engine="flite" > tts-voice="rms" > timeout="3000" > max-failures="2" > ; phrase_lang="en" -------^^ This is strange: ";" is not a valid sign in xml for comments. All comments must be between "". Also, you cannot put a comment inside a tag. Try to remove it and test again. Best Regards, -vma . _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120718/e9b5657c/attachment-0001.html From nandy1925 at gmail.com Wed Jul 18 09:56:34 2012 From: nandy1925 at gmail.com (Nandy Dagondon) Date: Wed, 18 Jul 2012 13:56:34 +0800 Subject: [Freeswitch-users] Need help in IVR In-Reply-To: References: Message-ID: hi, i thought there was a typo error in the parameter "na me". /nandy On Wed, Jul 18, 2012 at 1:18 PM, Piyush Sharma wrote: > > Hello Nandy, > > I didn't get you completely, I named my main menu "ps" it is just a test > case, I want to prepare a IVR, that may have sub menus upto some level, so > I will name menus & sub menus with some logic so that it is easy to > maintain, If it is possible, I want to generate it automatically via a > script on the basis of some input. I am new to this so your little support > can help me a lot. If you are not clear, tell me I will explain more about > this. (I am sick of visibility, it is not clear again.) > > > greet-long="phrase: ps_main" > greet-short="phrase: ps_short" > ............ > digit_len = 1> > > ------------------------------ > *From: *"Nandy Dagondon" > > *To: *"FreeSWITCH Users Help" > *Sent: *Tuesday, July 17, 2012 6:48:28 PM > > *Subject: *Re: [Freeswitch-users] Need help in IVR > > how about this line? > > On Tue, Jul 17, 2012 at 8:51 PM, Piyush Sharma < > piyush.sharma at coraltele.com> wrote: > >> >> Thanks for the response, I don't know how semicolon came in the email, >> but I have no semicolon in my XML, and please avoid visible smiles as well. >> >> >> > greet-long="phrases_main" >> greet-short="phrases_short" >> exit-sound="ivr/good_bye.wav" >> invalid-sound="ivr/ivr-not_available.wav" >> t ts-engine="flite" >> >> tts-voice="rms" >> timeout="3000" >> max-failures="2" >> phrase_lang="en" >> digit-len="1"> >> >> >> >> >> greet-long="phrases1_main" >> greet-short="phrases1_short" >> exit-sound="ivr/good_bye.wav" >> invalid-sound="ivr/ivr-not_available.wav" >> timeout ="3000" >> >> max-failure="3"> >> >> >> >> >> >> >> ------------------------------ >> *From: *"Vallimamod ABDULLAH" >> >> *To: *"FreeSWITCH Users Help" >> *Sent: *Tuesday, July 17, 2012 4:24:42 PM >> *Subject: *Re: [Freeswitch-users] Need help in IVR >> >> >> hi, >> >> On Jul 17, 2012, at 7:39 AM, Piyush Sharma wrote: >> >> > Hello Everyone, >> > >> > I am trying to create an IVR with menu and sub menus, but getting >> following error when I make the call, >> > 2012-07-17 11:05:37.755034 [ERR] switch_ivr_menu.c:870 Unable to build >> xml menu >> > 2012-07-17 11:05:37.755034 [ERR ] switch_ivr_menu.c:870 Unable to build >> xml menu >> > 2012-07-17 11:05:37.755034 [ERR] mod_dptools.c:1774 Unable to create >> menu >> > >> > I am new to this, so there might be little or a big error. >> > >> > My XML contents in ...conf/ivr_menus is >> > >> > >> > > > greet-long="phrase:ps_main" >> > greet-short="phrase:ps_short" >> > exit-sound="ivr/good_bye.wav" >> > invalid-sound="ivr/ivr-not_available.wav" >> > tts-engine="flite" >> > tts-voice="rms" >> > timeout="3000" >> > max-failures="2" >> > ; phrase_lang="en" >> >> -------^^ >> This is strange: ";" is not a valid sign in xml for comments. All >> comments must be between "". Also, you cannot put a comment >> inside a tag. >> Try to remove it and test again. >> >> Best Regards, >> -vma >> . >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120718/c0301bdf/attachment.html From william.king at quentustech.com Wed Jul 18 10:59:08 2012 From: william.king at quentustech.com (William King) Date: Tue, 17 Jul 2012 23:59:08 -0700 Subject: [Freeswitch-users] Strange 400 with some clients In-Reply-To: References: <50062B96.9070706@ezuce.com> Message-ID: <50065EBC.1090705@quentustech.com> You have probably configured the ip field in the client with the port, as well as configured the port field. Confirm the ip field doesn't have the address with the port on the end. This would be wrong: 189.146.85.112:10649 William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 07/17/2012 09:02 PM, Luis Daniel Lucio Quiroz wrote: > and how can i fix this? > > I know it is a linphone issue, but i wonder if there is a workarround > in server side. > > 2012/7/17 Joegen Baclor: >> Contact: >> >> >> There are two ports (10649) in the contact uri. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From miha at softnet.si Wed Jul 18 11:32:33 2012 From: miha at softnet.si (Miha) Date: Wed, 18 Jul 2012 09:32:33 +0200 Subject: [Freeswitch-users] ENUM do not work Message-ID: <50066691.9060700@softnet.si> Hi, is there any problem with enum on new git? When I do nslookup in linux, I get result but if I do it from fs_cli (enum 1231231231) I can see with wireshark that FS do not send lookup. p.s.: I have same configuration on different older FS and works. Thanks! miha From eugeneazuka at gmail.com Wed Jul 18 12:53:54 2012 From: eugeneazuka at gmail.com (Eugene Azuka) Date: Wed, 18 Jul 2012 09:53:54 +0100 Subject: [Freeswitch-users] Central FreeSWITCH nodes management with Mod_XML_Curl vs Mod_Lua Message-ID: Thank you FreeSWITCH experts: Michael Collins, Kristian Kielhofner, Vik Killa, Gabriel Gunderson. Thanks you all, i appreciate your comments. Gabriel Gunderson, Anyone who has practical experience of FreeSWITCH and able to reply and give meaningful comment/solution to help request, to me that person an experts. So you are an expert. :) Regards Eugene On Tue, Jul 17, 2012 at 11:41 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > 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. how to get leg_delay_start to work for bridge enterprise (Mario > G) > 2. Re: Username in SUBSCRIBE Request URI (Duvid Rottenberg) > 3. Central FreeSWITCH nodes management with Mod_XML_Curl vs > Mod_Lua (Eugene Azuka) > 4. Re: Central FreeSWITCH nodes management with Mod_XML_Curl vs > Mod_Lua (Vik Killa) > 5. Re: Central FreeSWITCH nodes management with Mod_XML_Curl vs > Mod_Lua (Michael Collins) > 6. Looking for dial plan examples for FS and SPA3102 router > (Todd Bailey) > 7. Re: Central FreeSWITCH nodes management with Mod_XML_Curl vs > Mod_Lua (Gabriel Gunderson) > > > ---------- Forwarded message ---------- > From: Mario G > To: FreeSWITCH Users Help > Cc: > Date: Tue, 17 Jul 2012 09:54:51 -0700 > Subject: [Freeswitch-users] how to get leg_delay_start to work for bridge > enterprise > I am try to delay the second target by 20 seconds. I used [..] and {..} > but no dice. The wiki has them both for enterprise, can someone shed light > on what's wrong, thanks. Main is supposed to ring, 20 secs later the second > target is added while main keeps going, all timeout after a total of 43 > secs. > Mario G > > "${group_call(main@ > ${domain_name}+E)}:_:{leg_delay_start=20}sofia/gateway/${dial_gateway}/19161234567" > /> > > > ---------- Forwarded message ---------- > From: Duvid Rottenberg > To: FreeSWITCH Users Help > Cc: > Date: Tue, 17 Jul 2012 14:01:20 -0400 > Subject: Re: [Freeswitch-users] Username in SUBSCRIBE Request URI > Apparently this is not currently possible. I have submitted a patch on > JIRA to add a new parameter to the subscriptions section named > user-in-register, when set to true the username will be included in the > request-uri. If/when this patch is approved I will update the wiki. > > On Fri, Jul 13, 2012 at 1:34 PM, Duvid Rottenberg wrote: > >> I am using embedded freeswitch as a softphone client and I am trying to >> subscribe to call-info on the server, (see config below) but the server is >> responding with a 481 Call/Transaction Does not exist. >> I compared the freeswitch SIP messages with SIP messages sent by a >> polycom phone for this feature and I noticed that freeswitch doesn't send >> the username in the request line. I think that this is causing the 481 >> response. >> >> Polycom Version: >> SUBSCRIBE sip:user at server:5060;transport=udp SIP/2.0 >> Freeswitch: >> SUBSCRIBE sip:server:5060;transport=udp SIP/2.0 >> >> Below is my gateway configuration >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Is there any way to tell freeswitch to include the username in the >> request line? >> >> Thank You, >> Duvid Rottenberg >> > > > > ---------- Forwarded message ---------- > From: Eugene Azuka > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Tue, 17 Jul 2012 17:33:56 +0100 > Subject: [Freeswitch-users] Central FreeSWITCH nodes management with > Mod_XML_Curl vs Mod_Lua > Hi FreeSWITCH experts, > > My question is as regards performance and which is the best option. > > I am trying to scale and manage multiple nodes of FreeSWITCH centrally > from my external web server/site using Mod_XML_Curl, But someone more > experience than me says otherwise that Mod_XML_Curl may not be the best > option: Here is his statement below: > > *" The thing I don't like about mod_xml_curl is that to scale it you have > to scale the web server with more listeners. If you run out of listeners > your ability to answer new calls fails. If the web server crashes your > ability to handle calls is gone.* > * > * > *Instead, I use Lua which is embedded into FreeSWITCH as the XML handler. > The Lua reads the info from the database and hands it off to FreeSWITCH. > This approach is not dependent on a service that can fail, or that can run > out of listeners. "* > > > What do you experts think about this comment above? > > Wouldn't Lua reading directly from database still face some of the issues > i may be running from using mod_xml_curl, like database crashing, slow > reading from database? > > Unlike using mod_xml_curl whereby i can optimise my code to make use of > memcached or radis to reduce some calls to database, can i still do such > with mod_lua? > > > So what do experts think, should i go with Mod_Lua approach just as he > said above or should i continue with using Mod_XML_Curl? > > You opinion will be appreciated. > > Regards > > Eugene > > > > > > > ---------- Forwarded message ---------- > From: Vik Killa > To: FreeSWITCH Users Help > Cc: > Date: Tue, 17 Jul 2012 15:30:53 -0400 > Subject: Re: [Freeswitch-users] Central FreeSWITCH nodes management with > Mod_XML_Curl vs Mod_Lua > I dont understand what they meant by 'listeners' > AFAIK if the web server process is running it will always return the XML > > > > > ---------- Forwarded message ---------- > From: Michael Collins > To: FreeSWITCH Users Help > Cc: > Date: Tue, 17 Jul 2012 13:56:11 -0700 > Subject: Re: [Freeswitch-users] Central FreeSWITCH nodes management with > Mod_XML_Curl vs Mod_Lua > > > On Tue, Jul 17, 2012 at 12:30 PM, Vik Killa wrote: > >> I dont understand what they meant by 'listeners' >> AFAIK if the web server process is running it will always return the XML >> > > Last time I checked, there were one or two websites out there that could > handle thousands of requests per second. The web has given birth to tools > that allow for many, many concurrent requests. The LAMP stack on a beefy > machine can do quite a lot of traffic. Furthermore, MySQL/Postgres/et al > all have backup/redundancy/HA options built in, as does Apache/HTTP. It > seems to me that you could scale farther and have more redundancy using > these time-tested tools. Just my $0.02. > > -MC > > P.S. - Don't get me wrong - I really like Lua. I just don't know if it's > really a "better" solution to this problem. > > > ---------- Forwarded message ---------- > From: Todd Bailey > To: freeswitch > Cc: > Date: Tue, 17 Jul 2012 15:37:51 -0700 > Subject: [Freeswitch-users] Looking for dial plan examples for FS and > SPA3102 router > Hi All, > > > I'm having issues getting a dial plan to work on FS and a Cisco SPA 3102 > router. > > when I dial 0, 1 or 9 plus a 10 number, I get to the router's dial tone > but I have to reenter the number I want to connect to. > > The expected action is to only need to enter the number to dial one > > Can some one provide dial plan and/or other config file example on how > to resolve this issue? > > here is what I have so far: > > /usr/local/freeswitch/conf/dialplan/default.xml > > > > > > expression="^(1{0,1,9}\d{10})$"> > > data="effective_caller_id_number=12223334444"/> > > > > > data="sofia/internal/$1 at 192.168.1.5:5061" /> > > > > > /usr/local/freeswitch/conf/dialplan/default/00_spa3102.xml > > > > > data="sofia/internal/${destination_number}@192.168.1.5:5061" /> > > > > > > > > > > > ---------- Forwarded message ---------- > From: Gabriel Gunderson > To: FreeSWITCH Users Help > Cc: > Date: Tue, 17 Jul 2012 16:40:41 -0600 > Subject: Re: [Freeswitch-users] Central FreeSWITCH nodes management with > Mod_XML_Curl vs Mod_Lua > Comments inline: > > On Tue, Jul 17, 2012 at 10:33 AM, Eugene Azuka > wrote: > > " The thing I don't like about mod_xml_curl is that to scale it you have > to > > scale the web server with more listeners. If you run out of listeners > your > > ability to answer new calls fails. If the web server crashes your > ability to > > handle calls is gone. > > Scaling web servers is an easy, well known and a common problem to > solve. Yes, you have to make sure you have the resources, but this it > true of any technology out there. > > > > Instead, I use Lua which is embedded into FreeSWITCH as the XML handler. > The > > Lua reads the info from the database and hands it off to FreeSWITCH. This > > approach is not dependent on a service that can fail, or that can run > out of > > listeners. " > > This is a great approach, but I don't think the logic is sound. Lua > (if it's doing anything fancy) will run out of resources that it > depends on -- memory, database connections, sockets etc. > > If one accepts that they'll have to build out to *really* scale > something, they'll also come to appreciate the fact that they can move > the HTTP stack to another box when needed (or 50 other boxen if > required). The Lua (or any other embedded language) is pretty well > tied to that same box. > > In the end, it probably doesn't matter. If you end up scaling to the > ends of the Earth, you'll have to rewrite it anyway. Just give thanks > that you've found FreeSWITCH and it was flexible enough to give you > the amazing configurability needed to build it in anyway you like :) > > > > What do you experts think about this comment above? > > There are experts on this list now?! Awesome ;) > > > > Wouldn't Lua reading directly from database still face some of the > issues i > > may be running from using mod_xml_curl, like database crashing, slow > reading > > from database? > > Yep, see above. > > > > Unlike using mod_xml_curl whereby i can optimise my code to make use of > > memcached or radis to reduce some calls to database, can i still do such > > with mod_lua? > > If you want and Lua supports it. > > > > So what do experts think, should i go with Mod_Lua approach just as he > said > > above or should i continue with using Mod_XML_Curl? > > Your choice! Happy hacking! > > > Best, > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120718/356b70ad/attachment-0001.html From jaybinks at gmail.com Wed Jul 18 12:57:31 2012 From: jaybinks at gmail.com (Jay Binks) Date: Wed, 18 Jul 2012 18:57:31 +1000 Subject: [Freeswitch-users] ENUM do not work In-Reply-To: <50066691.9060700@softnet.si> References: <50066691.9060700@softnet.si> Message-ID: <2007FC18-287D-4EC4-8258-2022D63A4370@gmail.com> Divide and conquer , figure out which revision broke it and I'll help you. What do you see on the console and what enum config do you have . On 18/07/2012, at 5:32 PM, Miha wrote: > Hi, > > is there any problem with enum on new git? When I do nslookup in linux, > I get result but if I do it from fs_cli (enum 1231231231) I can see with > wireshark that FS do not send lookup. > > p.s.: I have same configuration on different older FS and works. > > Thanks! > miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From piyush.sharma at coraltele.com Wed Jul 18 12:52:18 2012 From: piyush.sharma at coraltele.com (Piyush Sharma) Date: Wed, 18 Jul 2012 14:22:18 +0530 (IST) Subject: [Freeswitch-users] Need help in IVR In-Reply-To: Message-ID: Ya it seems so, some contents were not properly displayed that seems like typo error. I don't know why I am getting error when I make call but menu-sub & menu-top is not working for me. ----- Original Message ----- From: "Nandy Dagondon" To: "FreeSWITCH Users Help" Sent: Wednesday, July 18, 2012 11:26:34 AM Subject: Re: [Freeswitch-users] Need help in IVR hi, i thought there was a typo error in the parameter "na me". /nandy On Wed, Jul 18, 2012 at 1:18 PM, Piyush Sharma < piyush.sharma at coraltele.com > wrote: Hello Nandy, I didn't get you completely, I named my main menu "ps" it is just a test case, I want to prepare a IVR, that may have sub menus upto some level, so I will name menus & sub menus with some logic so that it is easy to maintain, If it is possible, I want to generate it automatically via a script on the basis of some input. I am new to this so your little support can help me a lot. If you are not clear, tell me I will explain more about this. (I am sick of visibility, it is not clear again.) From: "Nandy Dagondon" < gcd at i.ph > To: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > Sent: Tuesday, July 17, 2012 6:48:28 PM Subject: Re: [Freeswitch-users] Need help in IVR how about this line? wrote:
Thanks for the response, I don't know how semicolon came in the email, but I have no semicolon in my XML, and please avoid visible smiles as well. greet-long="phrase s1_main" greet-short="phrase s1_short" exit-sound="ivr/good_bye.wav" invalid-sound="ivr/ivr-not_available.wav" timeout ="3000" max-failure="3"> From: "Vallimamod ABDULLAH" < vallimamod.abdullah at imtelecom.fr > To: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > Sent: Tuesday, July 17, 2012 4:24:42 PM Subject: Re: [Freeswitch-users] Need help in IVR hi, On Jul 17, 2012, at 7:39 AM, Piyush Sharma wrote: > Hello Everyone, > > I am trying to create an IVR with menu and sub menus, but getting following error when I make the call, > 2012-07-17 11:05:37.755034 [ERR] switch_ivr_menu.c:870 Unable to build xml menu > 2012-07-17 11:05:37.755034 [ERR ] switch_ivr_menu.c:870 Unable to build xml menu > 2012-07-17 11:05:37.755034 [ERR] mod_dptools.c:1774 Unable to create menu > > I am new to this, so there might be little or a big error. > > My XML contents in ...conf/ivr_menus is > > > greet-long="phrase:ps_main" > greet-short="phrase:ps_short" > exit-sound="ivr/good_bye.wav" > invalid-sound="ivr/ivr-not_available.wav" > tts-engine="flite" > tts-voice="rms" > timeout="3000" > max-failures="2" > ; phrase_lang="en" -------^^ This is strange: ";" is not a valid sign in xml for comments. All comments must be between "". Also, you cannot put a comment inside a tag. Try to remove it and test again. Best Regards, -vma . _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120718/2ae1e7a2/attachment.html From miha at softnet.si Wed Jul 18 13:18:48 2012 From: miha at softnet.si (Miha) Date: Wed, 18 Jul 2012 11:18:48 +0200 Subject: [Freeswitch-users] ENUM do not work In-Reply-To: <2007FC18-287D-4EC4-8258-2022D63A4370@gmail.com> References: <50066691.9060700@softnet.si> <2007FC18-287D-4EC4-8258-2022D63A4370@gmail.com> Message-ID: <50067F78.3060803@softnet.si> Hi, it is revision 1.2 rc2 (installed from git two days ago). First I have change server to which dns lookup is made, than I add default one 164.... First I tought that is something wrong with configuration but than I noticed with wireshark that FS does not send nslookup. In 1.06 work and also on 1.2 rc1. Regards, Miha On 7/18/2012 10:57 AM, Jay Binks wrote: > Divide and conquer , figure out which revision broke it and I'll help you. > > What do you see on the console and what enum config do you have . > > > > On 18/07/2012, at 5:32 PM, Miha wrote: > >> Hi, >> >> is there any problem with enum on new git? When I do nslookup in linux, >> I get result but if I do it from fs_cli (enum 1231231231) I can see with >> wireshark that FS do not send lookup. >> >> p.s.: I have same configuration on different older FS and works. >> >> Thanks! >> miha >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From a.afzali2003 at gmail.com Wed Jul 18 14:54:49 2012 From: a.afzali2003 at gmail.com (afshin afzali) Date: Wed, 18 Jul 2012 15:24:49 +0430 Subject: [Freeswitch-users] Fax processing not successful - result (48) Disconnected after permitted retries Message-ID: Hi, There are some inbound fax sessions which I got error code 48 (and in some cases 49). FreeSWITCH Version 1.0.head (git-54ddef0 2011-12-06 21-53-45 -0600) AS5350XM Voice / Fax ( T.38 ) Gateway I've noticed that this error appears in cases which there are complex documents (graphics or gray areas). appreciate all comments ! -- afshin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120718/bb21694c/attachment.html From piyush.sharma at coraltele.com Wed Jul 18 15:08:47 2012 From: piyush.sharma at coraltele.com (Piyush Sharma) Date: Wed, 18 Jul 2012 16:38:47 +0530 (IST) Subject: [Freeswitch-users] Need help in IVR In-Reply-To: Message-ID: It seems that issue is regarding menu-sub. If I comment that line() from main menu, IVR starts but when I want to work with sub menu, I get these error messages 2012-07-18 16:25:07.623017 [ERR] switch_ivr_menu.c:870 Unable to build xml menu 2012-07-18 16:25:07.623017 [ERR] switch_ivr_menu.c:870 Unable to build xml menu 2012-07-18 16:25:07.623017 [ERR] mod_dptools.c:1774 Unable to create menu this is the code of sub menu in XML. is there anything wrong in this, well I have made the macros as same as in main menu ? ----- Original Message ----- From: "Nandy Dagondon" To: "FreeSWITCH Users Help" Sent: Wednesday, July 18, 2012 11:26:34 AM Subject: Re: [Freeswitch-users] Need help in IVR hi, i thought there was a typo error in the parameter "na me". /nandy On Wed, Jul 18, 2012 at 1:18 PM, Piyush Sharma < piyush.sharma at coraltele.com > wrote: Hello Nandy, I didn't get you completely, I named my main menu "ps" it is just a test case, I want to prepare a IVR, that may have sub menus upto some level, so I will name menus & sub menus with some logic so that it is easy to maintain, If it is possible, I want to generate it automatically via a script on the basis of some input. I am new to this so your little support can help me a lot. If you are not clear, tell me I will explain more about this. (I am sick of visibility, it is not clear again.) From: "Nandy Dagondon" < gcd at i.ph > To: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > Sent: Tuesday, July 17, 2012 6:48:28 PM Subject: Re: [Freeswitch-users] Need help in IVR how about this line? wrote:
Thanks for the response, I don't know how semicolon came in the email, but I have no semicolon in my XML, and please avoid visible smiles as well. greet-long="phrase s1_main" greet-short="phrase s1_short" exit-sound="ivr/good_bye.wav" invalid-sound="ivr/ivr-not_available.wav" timeout ="3000" max-failure="3"> From: "Vallimamod ABDULLAH" < vallimamod.abdullah at imtelecom.fr > To: "FreeSWITCH Users Help" < freeswitch-users at lists.freeswitch.org > Sent: Tuesday, July 17, 2012 4:24:42 PM Subject: Re: [Freeswitch-users] Need help in IVR hi, On Jul 17, 2012, at 7:39 AM, Piyush Sharma wrote: > Hello Everyone, > > I am trying to create an IVR with menu and sub menus, but getting following error when I make the call, > 2012-07-17 11:05:37.755034 [ERR] switch_ivr_menu.c:870 Unable to build xml menu > 2012-07-17 11:05:37.755034 [ERR ] switch_ivr_menu.c:870 Unable to build xml menu > 2012-07-17 11:05:37.755034 [ERR] mod_dptools.c:1774 Unable to create menu > > I am new to this, so there might be little or a big error. > > My XML contents in ...conf/ivr_menus is > > > greet-long="phrase:ps_main" > greet-short="phrase:ps_short" > exit-sound="ivr/good_bye.wav" > invalid-sound="ivr/ivr-not_available.wav" > tts-engine="flite" > tts-voice="rms" > timeout="3000" > max-failures="2" > ; phrase_lang="en" -------^^ This is strange: ";" is not a valid sign in xml for comments. All comments must be between "". Also, you cannot put a comment inside a tag. Try to remove it and test again. Best Regards, -vma . _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120718/7afb0f33/attachment-0001.html From ben at langfeld.co.uk Wed Jul 18 15:27:53 2012 From: ben at langfeld.co.uk (Ben Langfeld) Date: Wed, 18 Jul 2012 12:27:53 +0100 Subject: [Freeswitch-users] Central FreeSWITCH nodes management with Mod_XML_Curl vs Mod_Lua In-Reply-To: References: Message-ID: Does mod_xml_curl not support caching? If so, you could reduce load on your web servers by setting the correct response headers and avoid having to return a payload, or even FS making a request in the first place. In the case of failure, FS could fall back to its cache. Regards, Ben Langfeld On 18 July 2012 09:53, Eugene Azuka wrote: > Thank you FreeSWITCH experts: > > Michael Collins, Kristian Kielhofner, Vik Killa, Gabriel Gunderson. > > Thanks you all, i appreciate your comments. > > Gabriel Gunderson, Anyone who has practical experience of FreeSWITCH and > able to reply and give meaningful comment/solution to help request, to me > that person an experts. So you are an expert. :) > > > Regards > > Eugene > > > > > > On Tue, Jul 17, 2012 at 11:41 PM, < > freeswitch-users-request at lists.freeswitch.org> wrote: > >> 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. how to get leg_delay_start to work for bridge enterprise (Mario >> G) >> 2. Re: Username in SUBSCRIBE Request URI (Duvid Rottenberg) >> 3. Central FreeSWITCH nodes management with Mod_XML_Curl vs >> Mod_Lua (Eugene Azuka) >> 4. Re: Central FreeSWITCH nodes management with Mod_XML_Curl vs >> Mod_Lua (Vik Killa) >> 5. Re: Central FreeSWITCH nodes management with Mod_XML_Curl vs >> Mod_Lua (Michael Collins) >> 6. Looking for dial plan examples for FS and SPA3102 router >> (Todd Bailey) >> 7. Re: Central FreeSWITCH nodes management with Mod_XML_Curl vs >> Mod_Lua (Gabriel Gunderson) >> >> >> ---------- Forwarded message ---------- >> From: Mario G >> To: FreeSWITCH Users Help >> Cc: >> Date: Tue, 17 Jul 2012 09:54:51 -0700 >> Subject: [Freeswitch-users] how to get leg_delay_start to work for bridge >> enterprise >> I am try to delay the second target by 20 seconds. I used [..] and {..} >> but no dice. The wiki has them both for enterprise, can someone shed light >> on what's wrong, thanks. Main is supposed to ring, 20 secs later the second >> target is added while main keeps going, all timeout after a total of 43 >> secs. >> Mario G >> >> > "${group_call(main@ >> ${domain_name}+E)}:_:{leg_delay_start=20}sofia/gateway/${dial_gateway}/19161234567" >> /> >> >> >> ---------- Forwarded message ---------- >> From: Duvid Rottenberg >> To: FreeSWITCH Users Help >> Cc: >> Date: Tue, 17 Jul 2012 14:01:20 -0400 >> Subject: Re: [Freeswitch-users] Username in SUBSCRIBE Request URI >> Apparently this is not currently possible. I have submitted a patch on >> JIRA to add a new parameter to the subscriptions section named >> user-in-register, when set to true the username will be included in the >> request-uri. If/when this patch is approved I will update the wiki. >> >> On Fri, Jul 13, 2012 at 1:34 PM, Duvid Rottenberg > > wrote: >> >>> I am using embedded freeswitch as a softphone client and I am trying to >>> subscribe to call-info on the server, (see config below) but the server is >>> responding with a 481 Call/Transaction Does not exist. >>> I compared the freeswitch SIP messages with SIP messages sent by a >>> polycom phone for this feature and I noticed that freeswitch doesn't send >>> the username in the request line. I think that this is causing the 481 >>> response. >>> >>> Polycom Version: >>> SUBSCRIBE sip:user at server:5060;transport=udp SIP/2.0 >>> Freeswitch: >>> SUBSCRIBE sip:server:5060;transport=udp SIP/2.0 >>> >>> Below is my gateway configuration >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> Is there any way to tell freeswitch to include the username in the >>> request line? >>> >>> Thank You, >>> Duvid Rottenberg >>> >> >> >> >> ---------- Forwarded message ---------- >> From: Eugene Azuka >> To: freeswitch-users at lists.freeswitch.org >> Cc: >> Date: Tue, 17 Jul 2012 17:33:56 +0100 >> Subject: [Freeswitch-users] Central FreeSWITCH nodes management with >> Mod_XML_Curl vs Mod_Lua >> Hi FreeSWITCH experts, >> >> My question is as regards performance and which is the best option. >> >> I am trying to scale and manage multiple nodes of FreeSWITCH centrally >> from my external web server/site using Mod_XML_Curl, But someone more >> experience than me says otherwise that Mod_XML_Curl may not be the best >> option: Here is his statement below: >> >> *" The thing I don't like about mod_xml_curl is that to scale it you >> have to scale the web server with more listeners. If you run out of >> listeners your ability to answer new calls fails. If the web server crashes >> your ability to handle calls is gone.* >> * >> * >> *Instead, I use Lua which is embedded into FreeSWITCH as the XML >> handler. The Lua reads the info from the database and hands it off to >> FreeSWITCH. This approach is not dependent on a service that can fail, or >> that can run out of listeners. "* >> >> >> What do you experts think about this comment above? >> >> Wouldn't Lua reading directly from database still face some of the issues >> i may be running from using mod_xml_curl, like database crashing, slow >> reading from database? >> >> Unlike using mod_xml_curl whereby i can optimise my code to make use of >> memcached or radis to reduce some calls to database, can i still do such >> with mod_lua? >> >> >> So what do experts think, should i go with Mod_Lua approach just as he >> said above or should i continue with using Mod_XML_Curl? >> >> You opinion will be appreciated. >> >> Regards >> >> Eugene >> >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: Vik Killa >> To: FreeSWITCH Users Help >> Cc: >> Date: Tue, 17 Jul 2012 15:30:53 -0400 >> Subject: Re: [Freeswitch-users] Central FreeSWITCH nodes management with >> Mod_XML_Curl vs Mod_Lua >> I dont understand what they meant by 'listeners' >> AFAIK if the web server process is running it will always return the XML >> >> >> >> >> ---------- Forwarded message ---------- >> From: Michael Collins >> To: FreeSWITCH Users Help >> Cc: >> Date: Tue, 17 Jul 2012 13:56:11 -0700 >> Subject: Re: [Freeswitch-users] Central FreeSWITCH nodes management with >> Mod_XML_Curl vs Mod_Lua >> >> >> On Tue, Jul 17, 2012 at 12:30 PM, Vik Killa wrote: >> >>> I dont understand what they meant by 'listeners' >>> AFAIK if the web server process is running it will always return the XML >>> >> >> Last time I checked, there were one or two websites out there that could >> handle thousands of requests per second. The web has given birth to tools >> that allow for many, many concurrent requests. The LAMP stack on a beefy >> machine can do quite a lot of traffic. Furthermore, MySQL/Postgres/et al >> all have backup/redundancy/HA options built in, as does Apache/HTTP. It >> seems to me that you could scale farther and have more redundancy using >> these time-tested tools. Just my $0.02. >> >> -MC >> >> P.S. - Don't get me wrong - I really like Lua. I just don't know if it's >> really a "better" solution to this problem. >> >> >> ---------- Forwarded message ---------- >> From: Todd Bailey >> To: freeswitch >> Cc: >> Date: Tue, 17 Jul 2012 15:37:51 -0700 >> Subject: [Freeswitch-users] Looking for dial plan examples for FS and >> SPA3102 router >> Hi All, >> >> >> I'm having issues getting a dial plan to work on FS and a Cisco SPA 3102 >> router. >> >> when I dial 0, 1 or 9 plus a 10 number, I get to the router's dial tone >> but I have to reenter the number I want to connect to. >> >> The expected action is to only need to enter the number to dial one >> >> Can some one provide dial plan and/or other config file example on how >> to resolve this issue? >> >> here is what I have so far: >> >> /usr/local/freeswitch/conf/dialplan/default.xml >> >> >> >> >> >> > expression="^(1{0,1,9}\d{10})$"> >> >> > data="effective_caller_id_number=12223334444"/> >> >> >> >> >> > data="sofia/internal/$1 at 192.168.1.5:5061" /> >> >> >> >> >> /usr/local/freeswitch/conf/dialplan/default/00_spa3102.xml >> >> >> >> >> > data="sofia/internal/${destination_number}@192.168.1.5:5061" /> >> >> >> >> >> >> >> >> >> >> >> >> ---------- Forwarded message ---------- >> From: Gabriel Gunderson >> To: FreeSWITCH Users Help >> Cc: >> Date: Tue, 17 Jul 2012 16:40:41 -0600 >> Subject: Re: [Freeswitch-users] Central FreeSWITCH nodes management with >> Mod_XML_Curl vs Mod_Lua >> Comments inline: >> >> On Tue, Jul 17, 2012 at 10:33 AM, Eugene Azuka >> wrote: >> > " The thing I don't like about mod_xml_curl is that to scale it you >> have to >> > scale the web server with more listeners. If you run out of listeners >> your >> > ability to answer new calls fails. If the web server crashes your >> ability to >> > handle calls is gone. >> >> Scaling web servers is an easy, well known and a common problem to >> solve. Yes, you have to make sure you have the resources, but this it >> true of any technology out there. >> >> >> > Instead, I use Lua which is embedded into FreeSWITCH as the XML >> handler. The >> > Lua reads the info from the database and hands it off to FreeSWITCH. >> This >> > approach is not dependent on a service that can fail, or that can run >> out of >> > listeners. " >> >> This is a great approach, but I don't think the logic is sound. Lua >> (if it's doing anything fancy) will run out of resources that it >> depends on -- memory, database connections, sockets etc. >> >> If one accepts that they'll have to build out to *really* scale >> something, they'll also come to appreciate the fact that they can move >> the HTTP stack to another box when needed (or 50 other boxen if >> required). The Lua (or any other embedded language) is pretty well >> tied to that same box. >> >> In the end, it probably doesn't matter. If you end up scaling to the >> ends of the Earth, you'll have to rewrite it anyway. Just give thanks >> that you've found FreeSWITCH and it was flexible enough to give you >> the amazing configurability needed to build it in anyway you like :) >> >> >> > What do you experts think about this comment above? >> >> There are experts on this list now?! Awesome ;) >> >> >> > Wouldn't Lua reading directly from database still face some of the >> issues i >> > may be running from using mod_xml_curl, like database crashing, slow >> reading >> > from database? >> >> Yep, see above. >> >> >> >> > Unlike using mod_xml_curl whereby i can optimise my code to make use of >> > memcached or radis to reduce some calls to database, can i still do such >> > with mod_lua? >> >> If you want and Lua supports it. >> >> >> > So what do experts think, should i go with Mod_Lua approach just as he >> said >> > above or should i continue with using Mod_XML_Curl? >> >> Your choice! Happy hacking! >> >> >> Best, >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120718/147b3239/attachment-0001.html From steveu at coppice.org Wed Jul 18 15:30:14 2012 From: steveu at coppice.org (Steve Underwood) Date: Wed, 18 Jul 2012 19:30:14 +0800 Subject: [Freeswitch-users] Fax processing not successful - result (48) Disconnected after permitted retries In-Reply-To: References: Message-ID: <50069E46.3030003@coppice.org> On 07/18/2012 06:54 PM, afshin afzali wrote: > Hi, > > There are some inbound fax sessions which I got error code 48 (and in > some cases 49). > > FreeSWITCH Version 1.0.head (git-54ddef0 2011-12-06 21-53-45 -0600) > AS5350XM Voice / Fax ( T.38 ) Gateway > > I've noticed that this error appears in cases which there are complex > documents (graphics or gray areas). > appreciate all comments ! If you have graphics, the pages generally take a lot longer to send. It is not uncommon for T.38 gateways to have stupidly short timeouts, which will cause a call to drop after a page has been in progress for just one minute, or one and a half minutes. Could that be your problem? Steve From avi at avimarcus.net Wed Jul 18 15:32:52 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 18 Jul 2012 14:32:52 +0300 Subject: [Freeswitch-users] Central FreeSWITCH nodes management with Mod_XML_Curl vs Mod_Lua In-Reply-To: References: Message-ID: Caching the entire response and xml curl don't usually go together... If it's something unchanging, then do it in static XML. To serve that over the network from a central location, see: http://wiki.freeswitch.org/wiki/Mod_xml_curl#Storing_Static_Dialplans Although you'd still have to trigger a reloadxml when that changes. -Avi On Wed, Jul 18, 2012 at 2:27 PM, Ben Langfeld wrote: > Does mod_xml_curl not support caching? If so, you could reduce load on > your web servers by setting the correct response headers and avoid having > to return a payload, or even FS making a request in the first place. In the > case of failure, FS could fall back to its cache. > > Regards, > Ben Langfeld > > > On 18 July 2012 09:53, Eugene Azuka wrote: > >> Thank you FreeSWITCH experts: >> >> Michael Collins, Kristian Kielhofner, Vik Killa, Gabriel Gunderson. >> >> Thanks you all, i appreciate your comments. >> >> Gabriel Gunderson, Anyone who has practical experience of FreeSWITCH and >> able to reply and give meaningful comment/solution to help request, to me >> that person an experts. So you are an expert. :) >> >> >> Regards >> >> Eugene >> >> >> >> >> >> On Tue, Jul 17, 2012 at 11:41 PM, < >> freeswitch-users-request at lists.freeswitch.org> wrote: >> >>> 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. how to get leg_delay_start to work for bridge enterprise >>> (Mario G) >>> 2. Re: Username in SUBSCRIBE Request URI (Duvid Rottenberg) >>> 3. Central FreeSWITCH nodes management with Mod_XML_Curl vs >>> Mod_Lua (Eugene Azuka) >>> 4. Re: Central FreeSWITCH nodes management with Mod_XML_Curl vs >>> Mod_Lua (Vik Killa) >>> 5. Re: Central FreeSWITCH nodes management with Mod_XML_Curl vs >>> Mod_Lua (Michael Collins) >>> 6. Looking for dial plan examples for FS and SPA3102 router >>> (Todd Bailey) >>> 7. Re: Central FreeSWITCH nodes management with Mod_XML_Curl vs >>> Mod_Lua (Gabriel Gunderson) >>> >>> >>> ---------- Forwarded message ---------- >>> From: Mario G >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Tue, 17 Jul 2012 09:54:51 -0700 >>> Subject: [Freeswitch-users] how to get leg_delay_start to work for >>> bridge enterprise >>> I am try to delay the second target by 20 seconds. I used [..] and {..} >>> but no dice. The wiki has them both for enterprise, can someone shed light >>> on what's wrong, thanks. Main is supposed to ring, 20 secs later the second >>> target is added while main keeps going, all timeout after a total of 43 >>> secs. >>> Mario G >>> >>> >> "${group_call(main@ >>> ${domain_name}+E)}:_:{leg_delay_start=20}sofia/gateway/${dial_gateway}/19161234567" >>> /> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Duvid Rottenberg >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Tue, 17 Jul 2012 14:01:20 -0400 >>> Subject: Re: [Freeswitch-users] Username in SUBSCRIBE Request URI >>> Apparently this is not currently possible. I have submitted a patch on >>> JIRA to add a new parameter to the subscriptions section named >>> user-in-register, when set to true the username will be included in the >>> request-uri. If/when this patch is approved I will update the wiki. >>> >>> On Fri, Jul 13, 2012 at 1:34 PM, Duvid Rottenberg < >>> adrottenberg at gmail.com> wrote: >>> >>>> I am using embedded freeswitch as a softphone client and I am trying to >>>> subscribe to call-info on the server, (see config below) but the server is >>>> responding with a 481 Call/Transaction Does not exist. >>>> I compared the freeswitch SIP messages with SIP messages sent by a >>>> polycom phone for this feature and I noticed that freeswitch doesn't send >>>> the username in the request line. I think that this is causing the 481 >>>> response. >>>> >>>> Polycom Version: >>>> SUBSCRIBE sip:user at server:5060;transport=udp SIP/2.0 >>>> Freeswitch: >>>> SUBSCRIBE sip:server:5060;transport=udp SIP/2.0 >>>> >>>> Below is my gateway configuration >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Is there any way to tell freeswitch to include the username in the >>>> request line? >>>> >>>> Thank You, >>>> Duvid Rottenberg >>>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Eugene Azuka >>> To: freeswitch-users at lists.freeswitch.org >>> Cc: >>> Date: Tue, 17 Jul 2012 17:33:56 +0100 >>> Subject: [Freeswitch-users] Central FreeSWITCH nodes management with >>> Mod_XML_Curl vs Mod_Lua >>> Hi FreeSWITCH experts, >>> >>> My question is as regards performance and which is the best option. >>> >>> I am trying to scale and manage multiple nodes of FreeSWITCH centrally >>> from my external web server/site using Mod_XML_Curl, But someone more >>> experience than me says otherwise that Mod_XML_Curl may not be the best >>> option: Here is his statement below: >>> >>> *" The thing I don't like about mod_xml_curl is that to scale it you >>> have to scale the web server with more listeners. If you run out of >>> listeners your ability to answer new calls fails. If the web server crashes >>> your ability to handle calls is gone.* >>> * >>> * >>> *Instead, I use Lua which is embedded into FreeSWITCH as the XML >>> handler. The Lua reads the info from the database and hands it off to >>> FreeSWITCH. This approach is not dependent on a service that can fail, or >>> that can run out of listeners. "* >>> >>> >>> What do you experts think about this comment above? >>> >>> Wouldn't Lua reading directly from database still face some of the >>> issues i may be running from using mod_xml_curl, like database crashing, >>> slow reading from database? >>> >>> Unlike using mod_xml_curl whereby i can optimise my code to make use of >>> memcached or radis to reduce some calls to database, can i still do such >>> with mod_lua? >>> >>> >>> So what do experts think, should i go with Mod_Lua approach just as he >>> said above or should i continue with using Mod_XML_Curl? >>> >>> You opinion will be appreciated. >>> >>> Regards >>> >>> Eugene >>> >>> >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Vik Killa >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Tue, 17 Jul 2012 15:30:53 -0400 >>> Subject: Re: [Freeswitch-users] Central FreeSWITCH nodes management with >>> Mod_XML_Curl vs Mod_Lua >>> I dont understand what they meant by 'listeners' >>> AFAIK if the web server process is running it will always return the XML >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Michael Collins >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Tue, 17 Jul 2012 13:56:11 -0700 >>> Subject: Re: [Freeswitch-users] Central FreeSWITCH nodes management with >>> Mod_XML_Curl vs Mod_Lua >>> >>> >>> On Tue, Jul 17, 2012 at 12:30 PM, Vik Killa wrote: >>> >>>> I dont understand what they meant by 'listeners' >>>> AFAIK if the web server process is running it will always return the XML >>>> >>> >>> Last time I checked, there were one or two websites out there that could >>> handle thousands of requests per second. The web has given birth to tools >>> that allow for many, many concurrent requests. The LAMP stack on a beefy >>> machine can do quite a lot of traffic. Furthermore, MySQL/Postgres/et al >>> all have backup/redundancy/HA options built in, as does Apache/HTTP. It >>> seems to me that you could scale farther and have more redundancy using >>> these time-tested tools. Just my $0.02. >>> >>> -MC >>> >>> P.S. - Don't get me wrong - I really like Lua. I just don't know if it's >>> really a "better" solution to this problem. >>> >>> >>> ---------- Forwarded message ---------- >>> From: Todd Bailey >>> To: freeswitch >>> Cc: >>> Date: Tue, 17 Jul 2012 15:37:51 -0700 >>> Subject: [Freeswitch-users] Looking for dial plan examples for FS and >>> SPA3102 router >>> Hi All, >>> >>> >>> I'm having issues getting a dial plan to work on FS and a Cisco SPA 3102 >>> router. >>> >>> when I dial 0, 1 or 9 plus a 10 number, I get to the router's dial tone >>> but I have to reenter the number I want to connect to. >>> >>> The expected action is to only need to enter the number to dial one >>> >>> Can some one provide dial plan and/or other config file example on how >>> to resolve this issue? >>> >>> here is what I have so far: >>> >>> /usr/local/freeswitch/conf/dialplan/default.xml >>> >>> >>> >>> >>> >>> >> expression="^(1{0,1,9}\d{10})$"> >>> >>> >> data="effective_caller_id_number=12223334444"/> >>> >>> >>> >>> >>> >> data="sofia/internal/$1 at 192.168.1.5:5061" /> >>> >>> >>> >>> >>> /usr/local/freeswitch/conf/dialplan/default/00_spa3102.xml >>> >>> >>> >>> >>> >> data="sofia/internal/${destination_number}@192.168.1.5:5061" /> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> ---------- Forwarded message ---------- >>> From: Gabriel Gunderson >>> To: FreeSWITCH Users Help >>> Cc: >>> Date: Tue, 17 Jul 2012 16:40:41 -0600 >>> Subject: Re: [Freeswitch-users] Central FreeSWITCH nodes management with >>> Mod_XML_Curl vs Mod_Lua >>> Comments inline: >>> >>> On Tue, Jul 17, 2012 at 10:33 AM, Eugene Azuka >>> wrote: >>> > " The thing I don't like about mod_xml_curl is that to scale it you >>> have to >>> > scale the web server with more listeners. If you run out of listeners >>> your >>> > ability to answer new calls fails. If the web server crashes your >>> ability to >>> > handle calls is gone. >>> >>> Scaling web servers is an easy, well known and a common problem to >>> solve. Yes, you have to make sure you have the resources, but this it >>> true of any technology out there. >>> >>> >>> > Instead, I use Lua which is embedded into FreeSWITCH as the XML >>> handler. The >>> > Lua reads the info from the database and hands it off to FreeSWITCH. >>> This >>> > approach is not dependent on a service that can fail, or that can run >>> out of >>> > listeners. " >>> >>> This is a great approach, but I don't think the logic is sound. Lua >>> (if it's doing anything fancy) will run out of resources that it >>> depends on -- memory, database connections, sockets etc. >>> >>> If one accepts that they'll have to build out to *really* scale >>> something, they'll also come to appreciate the fact that they can move >>> the HTTP stack to another box when needed (or 50 other boxen if >>> required). The Lua (or any other embedded language) is pretty well >>> tied to that same box. >>> >>> In the end, it probably doesn't matter. If you end up scaling to the >>> ends of the Earth, you'll have to rewrite it anyway. Just give thanks >>> that you've found FreeSWITCH and it was flexible enough to give you >>> the amazing configurability needed to build it in anyway you like :) >>> >>> >>> > What do you experts think about this comment above? >>> >>> There are experts on this list now?! Awesome ;) >>> >>> >>> > Wouldn't Lua reading directly from database still face some of the >>> issues i >>> > may be running from using mod_xml_curl, like database crashing, slow >>> reading >>> > from database? >>> >>> Yep, see above. >>> >>> >>> >>> > Unlike using mod_xml_curl whereby i can optimise my code to make use of >>> > memcached or radis to reduce some calls to database, can i still do >>> such >>> > with mod_lua? >>> >>> If you want and Lua supports it. >>> >>> >>> > So what do experts think, should i go with Mod_Lua approach just as he >>> said >>> > above or should i continue with using Mod_XML_Curl? >>> >>> Your choice! Happy hacking! >>> >>> >>> Best, >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120718/4532648e/attachment-0001.html From piyush.sharma at coraltele.com Wed Jul 18 15:38:02 2012 From: piyush.sharma at coraltele.com (Piyush Sharma) Date: Wed, 18 Jul 2012 17:08:02 +0530 (IST) Subject: [Freeswitch-users] Fax processing not successful - result (48) Disconnected after permitted retries In-Reply-To: <50069E46.3030003@coppice.org> Message-ID: I also got this code, but in my scenario the called party didn't generating fax tone. I have tested FreeSWITCH with Images as well but never got such kind of code in that case. ----- Original Message ----- From: "Steve Underwood" To: "FreeSWITCH Users Help" Sent: Wednesday, July 18, 2012 5:00:14 PM Subject: Re: [Freeswitch-users] Fax processing not successful - result (48) Disconnected after permitted retries On 07/18/2012 06:54 PM, afshin afzali wrote: > Hi, > > There are some inbound fax sessions which I got error code 48 (and in > some cases 49). > > FreeSWITCH Version 1.0.head (git-54ddef0 2011-12-06 21-53-45 -0600) > AS5350XM Voice / Fax ( T.38 ) Gateway > > I've noticed that this error appears in cases which there are complex > documents (graphics or gray areas). > appreciate all comments ! If you have graphics, the pages generally take a lot longer to send. It is not uncommon for T.38 gateways to have stupidly short timeouts, which will cause a call to drop after a page has been in progress for just one minute, or one and a half minutes. Could that be your problem? Steve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120718/50e078c6/attachment.html From a.afzali2003 at gmail.com Wed Jul 18 15:48:38 2012 From: a.afzali2003 at gmail.com (afshin afzali) Date: Wed, 18 Jul 2012 16:18:38 +0430 Subject: [Freeswitch-users] Fax processing not successful - result (48) Disconnected after permitted retries In-Reply-To: <50069E46.3030003@coppice.org> References: <50069E46.3030003@coppice.org> Message-ID: Hi Steve, I don't think that my problem relates straightly to the time that a fax takes to be received. In some cases I could get better result to decrease fax speed (normally causes longer time). Let me tell you my today case. There is a tabular document that I could receive from a Panasonic fax machine and the same time unable to receive that from a Canon machine. Although I'm receiving other (simpler than) documents from the Canon one ! I've tested various configurations on my 5350 device of modifying fax rate, ls_redundancy and hs_redundancy but could not managed to fix this. Afshin On Wed, Jul 18, 2012 at 4:00 PM, Steve Underwood wrote: > On 07/18/2012 06:54 PM, afshin afzali wrote: > > Hi, > > > > There are some inbound fax sessions which I got error code 48 (and in > > some cases 49). > > > > FreeSWITCH Version 1.0.head (git-54ddef0 2011-12-06 21-53-45 -0600) > > AS5350XM Voice / Fax ( T.38 ) Gateway > > > > I've noticed that this error appears in cases which there are complex > > documents (graphics or gray areas). > > appreciate all comments ! > If you have graphics, the pages generally take a lot longer to send. It > is not uncommon for T.38 gateways to have stupidly short timeouts, which > will cause a call to drop after a page has been in progress for just one > minute, or one and a half minutes. Could that be your problem? > > Steve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120718/c3e4db3b/attachment.html From ben at langfeld.co.uk Wed Jul 18 15:50:21 2012 From: ben at langfeld.co.uk (Ben Langfeld) Date: Wed, 18 Jul 2012 12:50:21 +0100 Subject: [Freeswitch-users] Central FreeSWITCH nodes management with Mod_XML_Curl vs Mod_Lua In-Reply-To: References: Message-ID: I was thinking more short-term caching. It would be a valid way of optimising the entity providing the XML. Regards, Ben Langfeld On 18 July 2012 12:32, Avi Marcus wrote: > Caching the entire response and xml curl don't usually go together... > > If it's something unchanging, then do it in static XML. > To serve that over the network from a central location, see: > http://wiki.freeswitch.org/wiki/Mod_xml_curl#Storing_Static_Dialplans > > Although you'd still have to trigger a reloadxml when that changes. > -Avi > > > On Wed, Jul 18, 2012 at 2:27 PM, Ben Langfeld wrote: > >> Does mod_xml_curl not support caching? If so, you could reduce load on >> your web servers by setting the correct response headers and avoid having >> to return a payload, or even FS making a request in the first place. In the >> case of failure, FS could fall back to its cache. >> >> Regards, >> Ben Langfeld >> >> >> On 18 July 2012 09:53, Eugene Azuka wrote: >> >>> Thank you FreeSWITCH experts: >>> >>> Michael Collins, Kristian Kielhofner, Vik Killa, Gabriel Gunderson. >>> >>> Thanks you all, i appreciate your comments. >>> >>> Gabriel Gunderson, Anyone who has practical experience of FreeSWITCH and >>> able to reply and give meaningful comment/solution to help request, to me >>> that person an experts. So you are an expert. :) >>> >>> >>> Regards >>> >>> Eugene >>> >>> >>> >>> >>> >>> On Tue, Jul 17, 2012 at 11:41 PM, < >>> freeswitch-users-request at lists.freeswitch.org> wrote: >>> >>>> 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. how to get leg_delay_start to work for bridge enterprise >>>> (Mario G) >>>> 2. Re: Username in SUBSCRIBE Request URI (Duvid Rottenberg) >>>> 3. Central FreeSWITCH nodes management with Mod_XML_Curl vs >>>> Mod_Lua (Eugene Azuka) >>>> 4. Re: Central FreeSWITCH nodes management with Mod_XML_Curl vs >>>> Mod_Lua (Vik Killa) >>>> 5. Re: Central FreeSWITCH nodes management with Mod_XML_Curl vs >>>> Mod_Lua (Michael Collins) >>>> 6. Looking for dial plan examples for FS and SPA3102 router >>>> (Todd Bailey) >>>> 7. Re: Central FreeSWITCH nodes management with Mod_XML_Curl vs >>>> Mod_Lua (Gabriel Gunderson) >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Mario G >>>> To: FreeSWITCH Users Help >>>> Cc: >>>> Date: Tue, 17 Jul 2012 09:54:51 -0700 >>>> Subject: [Freeswitch-users] how to get leg_delay_start to work for >>>> bridge enterprise >>>> I am try to delay the second target by 20 seconds. I used [..] and {..} >>>> but no dice. The wiki has them both for enterprise, can someone shed light >>>> on what's wrong, thanks. Main is supposed to ring, 20 secs later the second >>>> target is added while main keeps going, all timeout after a total of 43 >>>> secs. >>>> Mario G >>>> >>>> >>> "${group_call(main@ >>>> ${domain_name}+E)}:_:{leg_delay_start=20}sofia/gateway/${dial_gateway}/19161234567" >>>> /> >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Duvid Rottenberg >>>> To: FreeSWITCH Users Help >>>> Cc: >>>> Date: Tue, 17 Jul 2012 14:01:20 -0400 >>>> Subject: Re: [Freeswitch-users] Username in SUBSCRIBE Request URI >>>> Apparently this is not currently possible. I have submitted a patch on >>>> JIRA to add a new parameter to the subscriptions section named >>>> user-in-register, when set to true the username will be included in the >>>> request-uri. If/when this patch is approved I will update the wiki. >>>> >>>> On Fri, Jul 13, 2012 at 1:34 PM, Duvid Rottenberg < >>>> adrottenberg at gmail.com> wrote: >>>> >>>>> I am using embedded freeswitch as a softphone client and I am trying >>>>> to subscribe to call-info on the server, (see config below) but the server >>>>> is responding with a 481 Call/Transaction Does not exist. >>>>> I compared the freeswitch SIP messages with SIP messages sent by a >>>>> polycom phone for this feature and I noticed that freeswitch doesn't send >>>>> the username in the request line. I think that this is causing the 481 >>>>> response. >>>>> >>>>> Polycom Version: >>>>> SUBSCRIBE sip:user at server:5060;transport=udp SIP/2.0 >>>>> Freeswitch: >>>>> SUBSCRIBE sip:server:5060;transport=udp SIP/2.0 >>>>> >>>>> Below is my gateway configuration >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Is there any way to tell freeswitch to include the username in the >>>>> request line? >>>>> >>>>> Thank You, >>>>> Duvid Rottenberg >>>>> >>>> >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Eugene Azuka >>>> To: freeswitch-users at lists.freeswitch.org >>>> Cc: >>>> Date: Tue, 17 Jul 2012 17:33:56 +0100 >>>> Subject: [Freeswitch-users] Central FreeSWITCH nodes management with >>>> Mod_XML_Curl vs Mod_Lua >>>> Hi FreeSWITCH experts, >>>> >>>> My question is as regards performance and which is the best option. >>>> >>>> I am trying to scale and manage multiple nodes of FreeSWITCH centrally >>>> from my external web server/site using Mod_XML_Curl, But someone more >>>> experience than me says otherwise that Mod_XML_Curl may not be the best >>>> option: Here is his statement below: >>>> >>>> *" The thing I don't like about mod_xml_curl is that to scale it you >>>> have to scale the web server with more listeners. If you run out of >>>> listeners your ability to answer new calls fails. If the web server crashes >>>> your ability to handle calls is gone.* >>>> * >>>> * >>>> *Instead, I use Lua which is embedded into FreeSWITCH as the XML >>>> handler. The Lua reads the info from the database and hands it off to >>>> FreeSWITCH. This approach is not dependent on a service that can fail, or >>>> that can run out of listeners. "* >>>> >>>> >>>> What do you experts think about this comment above? >>>> >>>> Wouldn't Lua reading directly from database still face some of the >>>> issues i may be running from using mod_xml_curl, like database crashing, >>>> slow reading from database? >>>> >>>> Unlike using mod_xml_curl whereby i can optimise my code to make use of >>>> memcached or radis to reduce some calls to database, can i still do such >>>> with mod_lua? >>>> >>>> >>>> So what do experts think, should i go with Mod_Lua approach just as he >>>> said above or should i continue with using Mod_XML_Curl? >>>> >>>> You opinion will be appreciated. >>>> >>>> Regards >>>> >>>> Eugene >>>> >>>> >>>> >>>> >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Vik Killa >>>> To: FreeSWITCH Users Help >>>> Cc: >>>> Date: Tue, 17 Jul 2012 15:30:53 -0400 >>>> Subject: Re: [Freeswitch-users] Central FreeSWITCH nodes management >>>> with Mod_XML_Curl vs Mod_Lua >>>> I dont understand what they meant by 'listeners' >>>> AFAIK if the web server process is running it will always return the XML >>>> >>>> >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Michael Collins >>>> To: FreeSWITCH Users Help >>>> Cc: >>>> Date: Tue, 17 Jul 2012 13:56:11 -0700 >>>> Subject: Re: [Freeswitch-users] Central FreeSWITCH nodes management >>>> with Mod_XML_Curl vs Mod_Lua >>>> >>>> >>>> On Tue, Jul 17, 2012 at 12:30 PM, Vik Killa wrote: >>>> >>>>> I dont understand what they meant by 'listeners' >>>>> AFAIK if the web server process is running it will always return the >>>>> XML >>>>> >>>> >>>> Last time I checked, there were one or two websites out there that >>>> could handle thousands of requests per second. The web has given birth to >>>> tools that allow for many, many concurrent requests. The LAMP stack on a >>>> beefy machine can do quite a lot of traffic. Furthermore, MySQL/Postgres/et >>>> al all have backup/redundancy/HA options built in, as does Apache/HTTP. It >>>> seems to me that you could scale farther and have more redundancy using >>>> these time-tested tools. Just my $0.02. >>>> >>>> -MC >>>> >>>> P.S. - Don't get me wrong - I really like Lua. I just don't know if >>>> it's really a "better" solution to this problem. >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Todd Bailey >>>> To: freeswitch >>>> Cc: >>>> Date: Tue, 17 Jul 2012 15:37:51 -0700 >>>> Subject: [Freeswitch-users] Looking for dial plan examples for FS and >>>> SPA3102 router >>>> Hi All, >>>> >>>> >>>> I'm having issues getting a dial plan to work on FS and a Cisco SPA 3102 >>>> router. >>>> >>>> when I dial 0, 1 or 9 plus a 10 number, I get to the router's dial tone >>>> but I have to reenter the number I want to connect to. >>>> >>>> The expected action is to only need to enter the number to dial one >>>> >>>> Can some one provide dial plan and/or other config file example on how >>>> to resolve this issue? >>>> >>>> here is what I have so far: >>>> >>>> /usr/local/freeswitch/conf/dialplan/default.xml >>>> >>>> >>>> >>>> >>>> >>>> >>> expression="^(1{0,1,9}\d{10})$"> >>>> >>>> >>> data="effective_caller_id_number=12223334444"/> >>>> >>>> >>>> >>>> >>>> >>> data="sofia/internal/$1 at 192.168.1.5:5061" /> >>>> >>>> >>>> >>>> >>>> /usr/local/freeswitch/conf/dialplan/default/00_spa3102.xml >>>> >>>> >>>> >>>> >>>> >>> data="sofia/internal/${destination_number}@192.168.1.5:5061" /> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> ---------- Forwarded message ---------- >>>> From: Gabriel Gunderson >>>> To: FreeSWITCH Users Help >>>> Cc: >>>> Date: Tue, 17 Jul 2012 16:40:41 -0600 >>>> Subject: Re: [Freeswitch-users] Central FreeSWITCH nodes management >>>> with Mod_XML_Curl vs Mod_Lua >>>> Comments inline: >>>> >>>> On Tue, Jul 17, 2012 at 10:33 AM, Eugene Azuka >>>> wrote: >>>> > " The thing I don't like about mod_xml_curl is that to scale it you >>>> have to >>>> > scale the web server with more listeners. If you run out of listeners >>>> your >>>> > ability to answer new calls fails. If the web server crashes your >>>> ability to >>>> > handle calls is gone. >>>> >>>> Scaling web servers is an easy, well known and a common problem to >>>> solve. Yes, you have to make sure you have the resources, but this it >>>> true of any technology out there. >>>> >>>> >>>> > Instead, I use Lua which is embedded into FreeSWITCH as the XML >>>> handler. The >>>> > Lua reads the info from the database and hands it off to FreeSWITCH. >>>> This >>>> > approach is not dependent on a service that can fail, or that can run >>>> out of >>>> > listeners. " >>>> >>>> This is a great approach, but I don't think the logic is sound. Lua >>>> (if it's doing anything fancy) will run out of resources that it >>>> depends on -- memory, database connections, sockets etc. >>>> >>>> If one accepts that they'll have to build out to *really* scale >>>> something, they'll also come to appreciate the fact that they can move >>>> the HTTP stack to another box when needed (or 50 other boxen if >>>> required). The Lua (or any other embedded language) is pretty well >>>> tied to that same box. >>>> >>>> In the end, it probably doesn't matter. If you end up scaling to the >>>> ends of the Earth, you'll have to rewrite it anyway. Just give thanks >>>> that you've found FreeSWITCH and it was flexible enough to give you >>>> the amazing configurability needed to build it in anyway you like :) >>>> >>>> >>>> > What do you experts think about this comment above? >>>> >>>> There are experts on this list now?! Awesome ;) >>>> >>>> >>>> > Wouldn't Lua reading directly from database still face some of the >>>> issues i >>>> > may be running from using mod_xml_curl, like database crashing, slow >>>> reading >>>> > from database? >>>> >>>> Yep, see above. >>>> >>>> >>>> >>>> > Unlike using mod_xml_curl whereby i can optimise my code to make use >>>> of >>>> > memcached or radis to reduce some calls to database, can i still do >>>> such >>>> > with mod_lua? >>>> >>>> If you want and Lua supports it. >>>> >>>> >>>> > So what do experts think, should i go with Mod_Lua approach just as >>>> he said >>>> > above or should i continue with using Mod_XML_Curl? >>>> >>>> Your choice! Happy hacking! >>>> >>>> >>>> Best, >>>> 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120718/659dc4d7/attachment-0001.html From steveu at coppice.org Wed Jul 18 17:01:54 2012 From: steveu at coppice.org (Steve Underwood) Date: Wed, 18 Jul 2012 21:01:54 +0800 Subject: [Freeswitch-users] Fax processing not successful - result (48) Disconnected after permitted retries In-Reply-To: References: <50069E46.3030003@coppice.org> Message-ID: <5006B3C2.3060203@coppice.org> On 07/18/2012 07:48 PM, afshin afzali wrote: > Hi Steve, > > I don't think that my problem relates straightly to the time that a > fax takes to be received. In some cases I could get better result to > decrease fax speed (normally causes longer time). Let me tell you my > today case. There is a tabular document that I could receive from a > Panasonic fax machine and the same time unable to receive that from a > Canon machine. Although I'm receiving other (simpler than) documents > from the Canon one ! A huge number of Canon machines have a broken V.29 modem, which produces such a poor signal you are lucky if other machines can decode it. The only real difference a complex image makes is the amount of data. This has only two effects: - Obviously it takes longer. - In ECM mode, if the image exceeds 64k it will be sent in 64k blocks, with error correction between each block. If your Cisco is running software more than a couple of years old, its T.38 engine may be horribly broken. It seems that had so much trouble with it, they tossed it out and put a complete replacement in there. The new one usually works well. > > I've tested various configurations on my 5350 device of modifying fax > rate, ls_redundancy and hs_redundancy but could not managed to fix this. > > Afshin > > On Wed, Jul 18, 2012 at 4:00 PM, Steve Underwood > wrote: > > On 07/18/2012 06:54 PM, afshin afzali wrote: > > Hi, > > > > There are some inbound fax sessions which I got error code 48 > (and in > > some cases 49). > > > > FreeSWITCH Version 1.0.head (git-54ddef0 2011-12-06 21-53-45 -0600) > > AS5350XM Voice / Fax ( T.38 ) Gateway > > > > I've noticed that this error appears in cases which there are > complex > > documents (graphics or gray areas). > > appreciate all comments ! > If you have graphics, the pages generally take a lot longer to > send. It > is not uncommon for T.38 gateways to have stupidly short timeouts, > which > will cause a call to drop after a page has been in progress for > just one > minute, or one and a half minutes. Could that be your problem? > > Steve > Steve From a.afzali2003 at gmail.com Wed Jul 18 17:10:49 2012 From: a.afzali2003 at gmail.com (afshin afzali) Date: Wed, 18 Jul 2012 17:40:49 +0430 Subject: [Freeswitch-users] Fax processing not successful - result (48) Disconnected after permitted retries In-Reply-To: <5006B3C2.3060203@coppice.org> References: <50069E46.3030003@coppice.org> <5006B3C2.3060203@coppice.org> Message-ID: Steve, Thank you very very much :) I'll inform you of the result. Best, Afshin On Wed, Jul 18, 2012 at 5:31 PM, Steve Underwood wrote: > On 07/18/2012 07:48 PM, afshin afzali wrote: > > Hi Steve, > > > > I don't think that my problem relates straightly to the time that a > > fax takes to be received. In some cases I could get better result to > > decrease fax speed (normally causes longer time). Let me tell you my > > today case. There is a tabular document that I could receive from a > > Panasonic fax machine and the same time unable to receive that from a > > Canon machine. Although I'm receiving other (simpler than) documents > > from the Canon one ! > A huge number of Canon machines have a broken V.29 modem, which produces > such a poor signal you are lucky if other machines can decode it. The > only real difference a complex image makes is the amount of data. This > has only two effects: > - Obviously it takes longer. > - In ECM mode, if the image exceeds 64k it will be sent in 64k > blocks, with error correction between each block. > If your Cisco is running software more than a couple of years old, its > T.38 engine may be horribly broken. It seems that had so much trouble > with it, they tossed it out and put a complete replacement in there. The > new one usually works well. > > > > I've tested various configurations on my 5350 device of modifying fax > > rate, ls_redundancy and hs_redundancy but could not managed to fix this. > > > > Afshin > > > > On Wed, Jul 18, 2012 at 4:00 PM, Steve Underwood > > wrote: > > > > On 07/18/2012 06:54 PM, afshin afzali wrote: > > > Hi, > > > > > > There are some inbound fax sessions which I got error code 48 > > (and in > > > some cases 49). > > > > > > FreeSWITCH Version 1.0.head (git-54ddef0 2011-12-06 21-53-45 -0600) > > > AS5350XM Voice / Fax ( T.38 ) Gateway > > > > > > I've noticed that this error appears in cases which there are > > complex > > > documents (graphics or gray areas). > > > appreciate all comments ! > > If you have graphics, the pages generally take a lot longer to > > send. It > > is not uncommon for T.38 gateways to have stupidly short timeouts, > > which > > will cause a call to drop after a page has been in progress for > > just one > > minute, or one and a half minutes. Could that be your problem? > > > > Steve > > > > Steve > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120718/95a72f5f/attachment.html From Hector.Geraldino at ipsoft.com Wed Jul 18 18:00:13 2012 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Wed, 18 Jul 2012 10:00:13 -0400 Subject: [Freeswitch-users] A fatal error has been detected by the Java Runtime Environment: In-Reply-To: <1342529673372-7580878.post@n2.nabble.com> References: <1342529673372-7580878.post@n2.nabble.com> Message-ID: <6A6B4C284AD15042B429EB9D904544AD022FAE2819@NY1-EXMB-01.ip-soft.net> Hello, Trying to help you to solve a crash in the jvm for a multithreaded application is damn hard. Doing it using a mailing list is even harder, and without looking at your source code is almost impossible. However I want to recommend you to drop the use of this library (which is a java wrapper of the FS core lib written in C) and use the pure Java ESL Client (http://wiki.freeswitch.org/wiki/Java_ESL_Client). You will have full access to the source code for debug, no dependencies on native libraries, and a good set of examples. Good luck! -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of bawajan Sent: Tuesday, July 17, 2012 8:55 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] A fatal error has been detected by the Java Runtime Environment: Hi, Am using ESL inbound connection to make calls and have a below flow (written in java) a) originating 15 calls simultaneously with park function b) play audio file c) creating new thread and passing eslconnection and playing a IVR in originate call. Here while handling events getting the below error : # A fatal error has been detected by the Java Runtime Environment: INFO | jvm 3 | 2012/07/17 17:54:36 | # INFO | jvm 3 | 2012/07/17 17:54:36 | # SIGSEGV (0xb) at pc=0x00007fc01f1d3943, pid=14759, tid=140463084390144 INFO | jvm 3 | 2012/07/17 17:54:36 | # INFO | jvm 3 | 2012/07/17 17:54:36 | # JRE version: 6.0_25-b06 INFO | jvm 3 | 2012/07/17 17:54:36 | # Java VM: Java HotSpot(TM) 64-Bit Server VM (20.0-b11 mixed mode linux-amd64 compressed oops) INFO | jvm 3 | 2012/07/17 17:54:36 | # Problematic frame: INFO | jvm 3 | 2012/07/17 17:54:36 | # C [libesljni.so+0xd943] long double+0x183 INFO | jvm 3 | 2012/07/17 17:54:36 | # INFO | jvm 3 | 2012/07/17 17:54:36 | # An error report file with more information is saved as: INFO | jvm 3 | 2012/07/17 17:54:36 | # /usr/local/freeswitch/hs_err_pid14759.log INFO | jvm 3 | 2012/07/17 17:54:36 | # INFO | jvm 3 | 2012/07/17 17:54:36 | # If you would like to submit a bug report, please visit: INFO | jvm 3 | 2012/07/17 17:54:36 | # http://java.sun.com/webapps/bugreport/crash.jsp INFO | jvm 3 | 2012/07/17 17:54:36 | # The crash happened outside the Java Virtual Machine in native code. INFO | jvm 3 | 2012/07/17 17:54:36 | # See problematic frame for where to report the bug. plz let me know, where am doing mistake and how to resolve it. Thanks in advance. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/A-fatal-error-has-been-detected-by-the-Java-Runtime-Environment-tp7580878.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From gomtesh at gmail.com Wed Jul 18 17:52:36 2012 From: gomtesh at gmail.com (Gomtesh Jain) Date: Wed, 18 Jul 2012 19:22:36 +0530 Subject: [Freeswitch-users] Freeswitch as Media proxy Message-ID: Hi All, Can we use freeswitch as media proxy? So that I can relay media to sip end points sitting behind NAT . Thanx, Gomtesh -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120718/0198f484/attachment.html From bdfoster at endigotech.com Wed Jul 18 19:13:08 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 18 Jul 2012 11:13:08 -0400 Subject: [Freeswitch-users] Freeswitch as Media proxy In-Reply-To: References: Message-ID: Freeswitch is a B2BUA. It doesn't work in the same sense as a proxy but you can configure to get the desired effect. I don't really understand what exactly you are trying to do though. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 18, 2012 11:04 AM, "Gomtesh Jain" wrote: > Hi All, > Can we use freeswitch as media proxy? So that I can relay media to sip > end points sitting behind NAT . > > Thanx, > Gomtesh > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120718/0f5a563e/attachment.html From freeswitch-list at puzzled.xs4all.nl Wed Jul 18 19:15:28 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 18 Jul 2012 17:15:28 +0200 Subject: [Freeswitch-users] Freeswitch as Media proxy In-Reply-To: References: Message-ID: <5006D310.7060504@puzzled.xs4all.nl> On 18-07-12 15:52, Gomtesh Jain wrote: > Hi All, > Can we use freeswitch as media proxy? So that I can relay media to > sip end points sitting behind NAT . FreeSWITCH is a B2BUA. For proxy functionality look at OpenSIPS or Kamailio. Having said that, FreeSWITCH is quite capable of handling clients behind NAT. Maybe you could try to setup a proof of concept. Regards, Patrick From msc at freeswitch.org Wed Jul 18 20:04:46 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Jul 2012 09:04:46 -0700 Subject: [Freeswitch-users] FreeSWITCH Community Conference Call Reminder Message-ID: Hello all, Just a reminder that the conference call will start in less than an hour. Today's agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_07_18 Talk to you soon, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120718/bea809fe/attachment.html From mi.ke at null.net Wed Jul 18 23:06:10 2012 From: mi.ke at null.net (Mi Ke) Date: Wed, 18 Jul 2012 15:06:10 -0400 Subject: [Freeswitch-users] DTMF detection in the core & spandsp Message-ID: <20120718190610.154580@gmx.com> Hi All, Do I need to disable DTMF detection in the core when I use spandsp for the same? Calling stop_dtmf before spandsp_start_dtmf didn't work for me and when I dial e.g. 0 I get: 2012-07-18 18:47:25.155435 [DEBUG] mod_spandsp_dsp.c:61 DTMF BEGIN DETECTED: [0] 2012-07-18 18:47:25.155435 [DEBUG] mod_spandsp_dsp.c:73 DTMF END DETECTED: [0], duration = 89 ms 2012-07-18 18:47:25.295432 [DEBUG] switch_rtp.c:3457 RTP RECV DTMF 0:2400 which comes to 00 after digit collection completes. Setting dtmf-type to none in sofia profile didn't help either. What's the right way of switching to spandsp's DTMF ? Thanks in advance for all yours hints. Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120718/e12c776e/attachment.html From peter.olsson at visionutveckling.se Wed Jul 18 23:29:30 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 18 Jul 2012 19:29:30 +0000 Subject: [Freeswitch-users] DTMF detection in the core & spandsp In-Reply-To: <20120718190610.154580@gmx.com> References: <20120718190610.154580@gmx.com> Message-ID: You probably get both inband and RFC2833 DTMF. So don't enable DTMF tone detection at all, and let FS parse out the RFC2833 DTMF events. /Peter 18 jul 2012 kl. 21:23 skrev "Mi Ke" >: Hi All, Do I need to disable DTMF detection in the core when I use spandsp for the same? Calling stop_dtmf before spandsp_start_dtmf didn't work for me and when I dial e.g. 0 I get: 2012-07-18 18:47:25.155435 [DEBUG] mod_spandsp_dsp.c:61 DTMF BEGIN DETECTED: [0] 2012-07-18 18:47:25.155435 [DEBUG] mod_spandsp_dsp.c:73 DTMF END DETECTED: [0], duration = 89 ms 2012-07-18 18:47:25.295432 [DEBUG] switch_rtp.c:3457 RTP RECV DTMF 0:2400 which comes to 00 after digit collection completes. Setting dtmf-type to none in sofia profile didn't help either. What's the right way of switching to spandsp's DTMF ? Thanks in advance for all yours hints. Mike !DSPAM:500709e632761477125784! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:500709e632761477125784! From mi.ke at null.net Wed Jul 18 23:59:46 2012 From: mi.ke at null.net (Mi Ke) Date: Wed, 18 Jul 2012 15:59:46 -0400 Subject: [Freeswitch-users] DTMF detection in the core & spandsp Message-ID: <20120718195946.154620@gmx.com> Hi Peter, My goal is to disable core DTMF detection completely and try spandsp instead since duplicate digit detection in the core works bad for me. DTMF in inbound call are inband. Thanks / Mike ----- Original Message ----- From: Peter Olsson Sent: 07/18/12 10:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] DTMF detection in the core & spandsp You probably get both inband and RFC2833 DTMF. So don't enable DTMF tone detection at all, and let FS parse out the RFC2833 DTMF events. /Peter 18 jul 2012 kl. 21:23 skrev "Mi Ke" >: Hi All, Do I need to disable DTMF detection in the core when I use spandsp for the same? Calling stop_dtmf before spandsp_start_dtmf didn't work for me and when I dial e.g. 0 I get: 2012-07-18 18:47:25.155435 [DEBUG] mod_spandsp_dsp.c:61 DTMF BEGIN DETECTED: [0] 2012-07-18 18:47:25.155435 [DEBUG] mod_spandsp_dsp.c:73 DTMF END DETECTED: [0], duration = 89 ms 2012-07-18 18:47:25.295432 [DEBUG] switch_rtp.c:3457 RTP RECV DTMF 0:2400 which comes to 00 after digit collection completes. Setting dtmf-type to none in sofia profile didn't help either. What's the right way of switching to spandsp's DTMF ? Thanks in advance for all yours hints. Mike !DSPAM:500709e632761477125784! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:500709e632761477125784! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120718/dd76a929/attachment.html From peter.olsson at visionutveckling.se Thu Jul 19 00:38:39 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 18 Jul 2012 20:38:39 +0000 Subject: [Freeswitch-users] DTMF detection in the core & spandsp In-Reply-To: <20120718195946.154620@gmx.com> References: <20120718195946.154620@gmx.com> Message-ID: <1FFF97C269757C458224B7C895F35F15135BAF@cantor.std.visionutv.se> Core DTMF tone detection is disabled by default, and started by using start_dtmf. However DTMF over RFC2833 is always enabled (maybe it's possible to disable, I don't know how though). RFC2833 is not DTMF tone detection, but rather a specially formatted RTP packet. In your sample you get both inband and RFC2833 events, so that's why I think it's better not to detect inband tones at all, but only use RFC2833. However, if you only get inband you will need to enable DTMF tone detection. Are you sure they don't support RFC2833 events, according to your log sample they seem to do... /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Mi Ke [mi.ke at null.net] Skickat: den 18 juli 2012 21:59 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] DTMF detection in the core & spandsp Hi Peter, My goal is to disable core DTMF detection completely and try spandsp instead since duplicate digit detection in the core works bad for me. DTMF in inbound call are inband. Thanks / Mike ----- Original Message ----- From: Peter Olsson Sent: 07/18/12 10:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] DTMF detection in the core & spandsp You probably get both inband and RFC2833 DTMF. So don't enable DTMF tone detection at all, and let FS parse out the RFC2833 DTMF events. /Peter 18 jul 2012 kl. 21:23 skrev "Mi Ke" >: Hi All, Do I need to disable DTMF detection in the core when I use spandsp for the same? Calling stop_dtmf before spandsp_start_dtmf didn't work for me and when I dial e.g. 0 I get: 2012-07-18 18:47:25.155435 [DEBUG] mod_spandsp_dsp.c:61 DTMF BEGIN DETECTED: [0] 2012-07-18 18:47:25.155435 [DEBUG] mod_spandsp_dsp.c:73 DTMF END DETECTED: [0], duration = 89 ms 2012-07-18 18:47:25.295432 [DEBUG] switch_rtp.c:3457 RTP RECV DTMF 0:2400 which comes to 00 after digit collection completes. Setting dtmf-type to none in sofia profile didn't help either. What's the right way of switching to spandsp's DTMF ? Thanks in advance for all yours hints. Mike _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:500709e632761477125784! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:5007142232761731370519! From mi.ke at null.net Thu Jul 19 02:27:33 2012 From: mi.ke at null.net (Mi Ke) Date: Wed, 18 Jul 2012 18:27:33 -0400 Subject: [Freeswitch-users] DTMF detection in the core & spandsp Message-ID: <20120718222733.154600@gmx.com> So in order to use just spandsp I need either to disable RFC2833 detection at my side or to ask originating party to disable sending us RFC2833 and just send us audio intact - have I got your point ? Thanks / Mike ----- Original Message ----- From: Peter Olsson Sent: 07/18/12 11:38 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] DTMF detection in the core & spandsp Core DTMF tone detection is disabled by default, and started by using start_dtmf. However DTMF over RFC2833 is always enabled (maybe it's possible to disable, I don't know how though). RFC2833 is not DTMF tone detection, but rather a specially formatted RTP packet. In your sample you get both inband and RFC2833 events, so that's why I think it's better not to detect inband tones at all, but only use RFC2833. However, if you only get inband you will need to enable DTMF tone detection. Are you sure they don't support RFC2833 events, according to your log sample they seem to do... /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Mi Ke [mi.ke at null.net] Skickat: den 18 juli 2012 21:59 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] DTMF detection in the core & spandsp Hi Peter, My goal is to disable core DTMF detection completely and try spandsp instead since duplicate digit detection in the core works bad for me. DTMF in inbound call are inband. Thanks / Mike ----- Original Message ----- From: Peter Olsson Sent: 07/18/12 10:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] DTMF detection in the core & spandsp You probably get both inband and RFC2833 DTMF. So don't enable DTMF tone detection at all, and let FS parse out the RFC2833 DTMF events. /Peter 18 jul 2012 kl. 21:23 skrev "Mi Ke" >: Hi All, Do I need to disable DTMF detection in the core when I use spandsp for the same? Calling stop_dtmf before spandsp_start_dtmf didn't work for me and when I dial e.g. 0 I get: 2012-07-18 18:47:25.155435 [DEBUG] mod_spandsp_dsp.c:61 DTMF BEGIN DETECTED: [0] 2012-07-18 18:47:25.155435 [DEBUG] mod_spandsp_dsp.c:73 DTMF END DETECTED: [0], duration = 89 ms 2012-07-18 18:47:25.295432 [DEBUG] switch_rtp.c:3457 RTP RECV DTMF 0:2400 which comes to 00 after digit collection completes. Setting dtmf-type to none in sofia profile didn't help either. What's the right way of switching to spandsp's DTMF ? Thanks in advance for all yours hints. Mike _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:500709e632761477125784! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:5007142232761731370519! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120718/0be81c4a/attachment-0001.html From msc at freeswitch.org Thu Jul 19 02:32:35 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 18 Jul 2012 15:32:35 -0700 Subject: [Freeswitch-users] DTMF detection in the core & spandsp In-Reply-To: <20120718222733.154600@gmx.com> References: <20120718222733.154600@gmx.com> Message-ID: On Wed, Jul 18, 2012 at 3:27 PM, Mi Ke wrote: > So in order to use just spandsp I need either to disable RFC2833 detection > at my side or to ask originating party to disable sending us RFC2833 and > just send us audio intact - have I got your point ? > That's kinda sorta what he's saying. A better way of saying it is this: if your carrier sends you RFC2833 DTMFs then you don't even need to turn on dtmf detect stuff. Unless there is a compelling reason to go in-band I'd say just use RFC2833 and be done with it. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120718/038ac330/attachment.html From mi.ke at null.net Thu Jul 19 02:56:46 2012 From: mi.ke at null.net (Mi Ke) Date: Wed, 18 Jul 2012 18:56:46 -0400 Subject: [Freeswitch-users] DTMF detection in the core & spandsp Message-ID: <20120718225646.154580@gmx.com> It appears that we cannot use carrier's RFC2833 sequence as it's missing up to 50% of digits sent to them by the original caller. And this is not happening on the same system with 2 other carriers. Peter and Michael, thank you very much for your help and explanations. WBR / Mike ----- Original Message ----- From: Michael Collins Sent: 07/19/12 01:32 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] DTMF detection in the core & spandsp On Wed, Jul 18, 2012 at 3:27 PM, Mi Ke < mi.ke at null.net > wrote: So in order to use just spandsp I need either to disable RFC2833 detection at my side or to ask originating party to disable sending us RFC2833 and just send us audio intact - have I got your point ? That's kinda sorta what he's saying. A better way of saying it is this: if your carrier sends you RFC2833 DTMFs then you don't even need to turn on dtmf detect stuff. Unless there is a compelling reason to go in-band I'd say just use RFC2833 and be done with it. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120718/2b9e2e05/attachment.html From mel0torme at gmail.com Thu Jul 19 03:13:04 2012 From: mel0torme at gmail.com (Tom C) Date: Wed, 18 Jul 2012 16:13:04 -0700 Subject: [Freeswitch-users] Profiling FreeSwitch Message-ID: Are there any tools already in use for profiling FreeSwitch? That is, getting statistics on how often various procedures are called, and how much processor time they take, etc. Being a linux noob, I spent hours looking at "gprof", and when I finally got it working, I learned that it doesn't handle multi-threaded apps like FS. Is there a recommended tool that people are already using? Or should I just add my own logging code and recompile? (PS, I originally sent this question a couple days ago, but it didn't seem to go out to the list. This has happened to my last three emails. Strange.) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120718/54462388/attachment.html From curriegrad2004 at gmail.com Thu Jul 19 03:17:57 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 18 Jul 2012 16:17:57 -0700 Subject: [Freeswitch-users] Profiling FreeSwitch In-Reply-To: References: Message-ID: Try the FreeSWITCH-dev list On Wed, Jul 18, 2012 at 4:13 PM, Tom C wrote: > Are there any tools already in use for profiling FreeSwitch? That is, > getting statistics on how often various procedures are called, and how much > processor time they take, etc. > > Being a linux noob, I spent hours looking at "gprof", and when I finally got > it working, I learned that it doesn't handle multi-threaded apps like FS. > > Is there a recommended tool that people are already using? Or should I just > add my own logging code and recompile? > > (PS, I originally sent this question a couple days ago, but it didn't seem > to go out to the list. This has happened to my last three emails. Strange.) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Jul 19 04:50:14 2012 From: brian at freeswitch.org (Brian West) Date: Wed, 18 Jul 2012 19:50:14 -0500 Subject: [Freeswitch-users] Hotel Rooms for ClueCon Message-ID: <1812A19D-C80C-4824-888B-AF59565680DC@freeswitch.org> Seems we have filled our block on some days, I'll call tomorrow and get this fixed. If anyone has problems booking please call me. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120718/14088345/attachment.html From curriegrad2004 at gmail.com Thu Jul 19 04:53:15 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 18 Jul 2012 17:53:15 -0700 Subject: [Freeswitch-users] Hotel Rooms for ClueCon In-Reply-To: <1812A19D-C80C-4824-888B-AF59565680DC@freeswitch.org> References: <1812A19D-C80C-4824-888B-AF59565680DC@freeswitch.org> Message-ID: Hehe, if you can't get it fixed, it seems as if the attendees will have to bunk up with bkw_ On 7/18/12, Brian West wrote: > Seems we have filled our block on some days, I'll call tomorrow and get this > fixed. If anyone has problems booking please call me. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST > iNUM: +883 5100 1286 0410 > > > > > From bdfoster at endigotech.com Thu Jul 19 05:09:44 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 18 Jul 2012 21:09:44 -0400 Subject: [Freeswitch-users] Hotel Rooms for ClueCon In-Reply-To: References: <1812A19D-C80C-4824-888B-AF59565680DC@freeswitch.org> Message-ID: Shotgun! Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 18, 2012 8:54 PM, "curriegrad2004" wrote: > Hehe, if you can't get it fixed, it seems as if the attendees will > have to bunk up with bkw_ > > On 7/18/12, Brian West wrote: > > Seems we have filled our block on some days, I'll call tomorrow and get > this > > fixed. If anyone has problems booking please call me. > > > > -- > > Brian West > > brian at freeswitch.org > > FreeSWITCH Solutions, LLC > > PO BOX PO BOX 2531 > > Brookfield, WI 53008-2531 > > Twitter: @FreeSWITCH_Wire > > T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST > > iNUM: +883 5100 1286 0410 > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120718/e5570628/attachment-0001.html From o.belykh at kustogroup.com Thu Jul 19 08:55:18 2012 From: o.belykh at kustogroup.com (Oleg Belykh) Date: Thu, 19 Jul 2012 10:55:18 +0600 Subject: [Freeswitch-users] Freebsd mod_mp4v Message-ID: <598B35E4-0299-4FC5-BED0-FF14BFC52491@kustogroup.com> Hello. There is no flags for mod_mp4v module building (from ports). How i can build and install in manually? Oleg Belykh From aerocomputer at gmail.com Thu Jul 19 09:55:19 2012 From: aerocomputer at gmail.com (A.S. Shaja) Date: Thu, 19 Jul 2012 11:25:19 +0530 Subject: [Freeswitch-users] Re. trigger custom freeswitch application to notify users in a conference Message-ID: Hi, Is it possible to create an application in FreeSWITCH that when a user presses a series of keys, it could trigger a custom FreeSWITCH application to notify all moderators via an audio message? Can you please help me on how to achieve this? Regards, Shaja. -- "TajMahal would not have been so beautiful if shajahan had asked for three quotations and had chosen the least one!!!" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120719/e242968d/attachment.html From gabe at gundy.org Thu Jul 19 11:06:21 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Thu, 19 Jul 2012 01:06:21 -0600 Subject: [Freeswitch-users] Hotel Rooms for ClueCon In-Reply-To: <1812A19D-C80C-4824-888B-AF59565680DC@freeswitch.org> References: <1812A19D-C80C-4824-888B-AF59565680DC@freeswitch.org> Message-ID: On Wed, Jul 18, 2012 at 6:50 PM, Brian West wrote: > Seems we have filled our block on some days, I'll call tomorrow and get this > fixed. Can you update this thread so we can know when to call in? Thanks! Gabe From gomtesh at gmail.com Thu Jul 19 11:14:52 2012 From: gomtesh at gmail.com (Gomtesh Jain) Date: Thu, 19 Jul 2012 12:44:52 +0530 Subject: [Freeswitch-users] Freeswitch as Media proxy In-Reply-To: <5006D310.7060504@puzzled.xs4all.nl> References: <5006D310.7060504@puzzled.xs4all.nl> Message-ID: Here I explain scenario... UA1-------------->Opensips/Registrar----------->UA2 UA1 and UA2 are behind Symm NAT. For signaling I we can fix contact from opensips. But for media relay I want to use freeswitch. So the N/w I am looking for is ... Signaling:- UA1-------------->Opensips/Registrar-------->Freeswitch------>UA2 Media: UA1------->Freeswitch------>UA2 For signaling freeswitch should not make any change except contact/SDP . Please let me know if it is possible with freeswitch Thanx, Gomtesh On Wed, Jul 18, 2012 at 8:45 PM, Patrick Lists < freeswitch-list at puzzled.xs4all.nl> wrote: > On 18-07-12 15:52, Gomtesh Jain wrote: > > Hi All, > > Can we use freeswitch as media proxy? So that I can relay media to > > sip end points sitting behind NAT . > > FreeSWITCH is a B2BUA. For proxy functionality look at OpenSIPS or > Kamailio. Having said that, FreeSWITCH is quite capable of handling > clients behind NAT. Maybe you could try to setup a proof of concept. > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120719/80658782/attachment.html From miha at softnet.si Thu Jul 19 13:35:10 2012 From: miha at softnet.si (Miha) Date: Thu, 19 Jul 2012 11:35:10 +0200 Subject: [Freeswitch-users] ENUM do not work In-Reply-To: <50067F78.3060803@softnet.si> References: <50066691.9060700@softnet.si> <2007FC18-287D-4EC4-8258-2022D63A4370@gmail.com> <50067F78.3060803@softnet.si> Message-ID: <5007D4CE.8030302@softnet.si> On 7/18/2012 11:18 AM, Miha wrote: > Hi, > > it is revision 1.2 rc2 (installed from git two days ago). First I have > change server to which dns lookup is made, than I add default one 164.... > First I tought that is something wrong with configuration but than I > noticed with wireshark that FS does not send nslookup. > > In 1.06 work and also on 1.2 rc1. > > Regards, > Miha > > On 7/18/2012 10:57 AM, Jay Binks wrote: >> Divide and conquer , figure out which revision broke it and I'll help you. >> >> What do you see on the console and what enum config do you have . >> >> >> >> On 18/07/2012, at 5:32 PM, Miha wrote: >> >>> Hi, >>> >>> is there any problem with enum on new git? When I do nslookup in linux, >>> I get result but if I do it from fs_cli (enum 1231231231) I can see with >>> wireshark that FS do not send lookup. >>> >>> p.s.: I have same configuration on different older FS and works. >>> >>> Thanks! >>> miha >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > Revision of FS is: FreeSWITCH Version 1.2.0-rc2+git~20120719T040232Z~ec412c07d2+unclean~20120719T084055Z Regards,Miha From miha at softnet.si Thu Jul 19 14:43:36 2012 From: miha at softnet.si (Miha) Date: Thu, 19 Jul 2012 12:43:36 +0200 Subject: [Freeswitch-users] Which FS version to use? Message-ID: <5007E4D8.7050108@softnet.si> Hi, on my production server I have version: freeswitch at default> version FreeSWITCH Version 1.0.head (git-00de8e6 2011-11-01 17-27-13 -0600) which is in my case very stable, I can say perfect. As I have prepare other, more powerfull server for FS I have installed FS from latest git. With version 1.2 rc1 I was having problems with load and all users did not regisered (loosing connection). I do no know if this was FS problem but I do not have this problems on production server. Few day ago I did a new git pull. Now I have version 1.2 rc2 which enum do not work. I traced with wireshark but afer I put in fs_cli enum 123123123, Fs sends nothing. So on git.freeswitch.com are versions 1.0.6, 1.2 rc2 and 1.2 rc1. Which version to use? Is 1.0.6 most stable for now as 1.2 is rc? Where can I find version of FS which I am running on production server? Thanks for all your help! miha From bdfoster at endigotech.com Thu Jul 19 15:16:10 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 19 Jul 2012 07:16:10 -0400 Subject: [Freeswitch-users] Freeswitch as Media proxy In-Reply-To: References: <5006D310.7060504@puzzled.xs4all.nl> Message-ID: Yes. You could also do it with Opensips. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 19, 2012 3:15 AM, "Gomtesh Jain" wrote: > Here I explain scenario... > > UA1-------------->Opensips/Registrar----------->UA2 > > UA1 and UA2 are behind Symm NAT. For signaling I we can fix contact from > opensips. But for media relay I want to use freeswitch. > > So the N/w I am looking for is ... > > Signaling:- > > UA1-------------->Opensips/Registrar-------->Freeswitch------>UA2 > > Media: > > UA1------->Freeswitch------>UA2 > > For signaling freeswitch should not make any change except contact/SDP . > > > Please let me know if it is possible with freeswitch > > Thanx, > Gomtesh > > On Wed, Jul 18, 2012 at 8:45 PM, Patrick Lists < > freeswitch-list at puzzled.xs4all.nl> wrote: > >> On 18-07-12 15:52, Gomtesh Jain wrote: >> > Hi All, >> > Can we use freeswitch as media proxy? So that I can relay media to >> > sip end points sitting behind NAT . >> >> FreeSWITCH is a B2BUA. For proxy functionality look at OpenSIPS or >> Kamailio. Having said that, FreeSWITCH is quite capable of handling >> clients behind NAT. Maybe you could try to setup a proof of concept. >> >> Regards, >> Patrick >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120719/1ef138f9/attachment-0001.html From peter.olsson at visionutveckling.se Thu Jul 19 16:47:18 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 19 Jul 2012 12:47:18 +0000 Subject: [Freeswitch-users] Which FS version to use? In-Reply-To: <5007E4D8.7050108@softnet.si> References: <5007E4D8.7050108@softnet.si> Message-ID: <5B0A12B0-7D51-4A8C-A9EE-9C5A2200967B@visionutveckling.se> Current git head is always the recommended version. If you find any problems, please report them to Jira. Git head is currently at 1.2 RC2 state (waiting for a stable 1.2 release), so the more things we get fixed, the better! Please retry the ENUM problems on latest git again, and if it still fails, report this to Jira. /Peter 19 jul 2012 kl. 12:51 skrev "Miha" : > Hi, > > on my production server I have version: > > freeswitch at default> version > FreeSWITCH Version 1.0.head (git-00de8e6 2011-11-01 17-27-13 -0600) > > which is in my case very stable, I can say perfect. > > As I have prepare other, more powerfull server for FS I have installed > FS from latest git. With version 1.2 rc1 I was having problems with load > and all users did not regisered (loosing connection). I do no know if > this was FS problem but I do not have this problems on production > server. Few day ago I did a new git pull. Now I have version 1.2 rc2 > which enum do not work. I traced with wireshark but afer I put in fs_cli > enum 123123123, Fs sends nothing. > > > So on git.freeswitch.com are versions 1.0.6, 1.2 rc2 and 1.2 rc1. Which > version to use? Is 1.0.6 most stable for now as 1.2 is rc? > > Where can I find version of FS which I am running on production server? > > Thanks for all your help! > > miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:5007e36132765726119943! > From peter.olsson at visionutveckling.se Thu Jul 19 17:18:53 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 19 Jul 2012 13:18:53 +0000 Subject: [Freeswitch-users] Which FS version to use? In-Reply-To: <5007E4D8.7050108@softnet.si> References: <5007E4D8.7050108@softnet.si> Message-ID: <1FFF97C269757C458224B7C895F35F15135F07@cantor.std.visionutv.se> Just did a quick check on ENUM as well. It seems that the current version requires you to configure nameservers for enum. The default stuff that used to exist seems to be overwritten by these settings. The default enum settings file also reflect these changes. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Miha [miha at softnet.si] Skickat: den 19 juli 2012 12:43 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] Which FS version to use? Hi, on my production server I have version: freeswitch at default> version FreeSWITCH Version 1.0.head (git-00de8e6 2011-11-01 17-27-13 -0600) which is in my case very stable, I can say perfect. As I have prepare other, more powerfull server for FS I have installed FS from latest git. With version 1.2 rc1 I was having problems with load and all users did not regisered (loosing connection). I do no know if this was FS problem but I do not have this problems on production server. Few day ago I did a new git pull. Now I have version 1.2 rc2 which enum do not work. I traced with wireshark but afer I put in fs_cli enum 123123123, Fs sends nothing. So on git.freeswitch.com are versions 1.0.6, 1.2 rc2 and 1.2 rc1. Which version to use? Is 1.0.6 most stable for now as 1.2 is rc? Where can I find version of FS which I am running on production server? Thanks for all your help! miha _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:5007e36132765726119943! From alphaniner at me.com Thu Jul 19 17:18:10 2012 From: alphaniner at me.com (Alpha Niner) Date: Thu, 19 Jul 2012 13:18:10 +0000 (GMT) Subject: [Freeswitch-users] Lua outbound executing another lua script Message-ID: <3cd5711b-f322-6acd-80d0-dc2d41aadc5f@me.com> Hi, Firstly apologies for the newbie question. I've tried searching the various documentation and trying a few different things but to no avail, so I've come here to ask for help. I think my problem relates to sessions, but I'm not totally sure. What I'm trying to do is: 1. A lua script is run by someone calling into FS (the initiator) 2. They select an option, which originates and makes an outbound call to another phone number 3. The 3rd party (hopefully) answers their phone and receives a prompt that they're going to be connected (the receiver) 4. Both parties join a conference and talk to each other (a conference as there may be other parties in future) To do this I'm creating a session and using execute_on_answer to run the second script and provide the prompt to the receiving party. However the receiving party never hears a prompt, comes into the conference, immediately drops out and hangs up. Here are the scripts, they're pretty simple; The initiator: session:answer() session:streamFile("WaitForOutdial.wav") new_session = freeswitch.Session("[execute_on_answer=lua /usr/local/freeswitch/scripts/receivewithwhisper.lua]sofia/external/12345 at mydest"); session:execute("conference", "CONFTEST at defau?lt+flags{endconf, moderator}") session:hangup(); The above works fine and the outbound call is placed. The script below is then executed but the receiving phone rings but nothing is heard, the conference is joined/exited and it terminates again; The receiver (receivewithwhisper.lua): session:answer() session:streamFile("ConnectingToCaller.wav") session:execute("conference", "CONFTEST at default+flags{endconf, moderator}") session:hangup(); The receiving scripts runs when the receiver answers their phone, and executes (quickly) and terminates. I was expecting it to drop into the conference and I'd be able to talk to the other person. It does join the conference, but then exits right away and falls through to the hangup section. As I say, I'm not exactly sure if this is the best way (maybe there's another - suggestions are welcome) and not exactly sure what the problem is although I suspect it's something to do with the sessions? Any help, advice, corrections or examples would be appreciated. It's all done in lua and I've tried to avoid session.originate as I understand this is deprecated. Thanks Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120719/51a90377/attachment.html From ssinyagin at yahoo.com Thu Jul 19 17:37:19 2012 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Thu, 19 Jul 2012 06:37:19 -0700 (PDT) Subject: [Freeswitch-users] Which FS version to use? In-Reply-To: <5B0A12B0-7D51-4A8C-A9EE-9C5A2200967B@visionutveckling.se> References: <5007E4D8.7050108@softnet.si> <5B0A12B0-7D51-4A8C-A9EE-9C5A2200967B@visionutveckling.se> Message-ID: <1342705039.82548.YahooMailNeo@web39304.mail.mud.yahoo.com> by the way, any plans to fork a "stable" Git branch and use it only for bugfixes, whereas "master" would absorb also all new features? What I see now in the commit log, is the "master" branch also used for normal development, and not only for bugfixes. This way it's difficult to expect the final 1.2 release any time soon. >________________________________ > From: Peter Olsson >To: FreeSWITCH Users Help >Sent: Thursday, July 19, 2012 2:47 PM >Subject: Re: [Freeswitch-users] Which FS version to use? > >Current git head is always the recommended version. If you find any problems, please report them to Jira. Git head is currently at 1.2 RC2 state (waiting for a stable 1.2 release), so the more things we get fixed, the better! > >Please retry the ENUM problems on latest git again, and if it still fails, report this to Jira. > >/Peter > >19 jul 2012 kl. 12:51 skrev "Miha" : > >> Hi, >> >> on my production server I have version: >> >> freeswitch at default> version >> FreeSWITCH Version 1.0.head (git-00de8e6 2011-11-01 17-27-13 -0600) >> >> which is in my case very stable, I can say perfect. >> >> As I have prepare other, more powerfull server for FS I have installed >> FS from latest git. With version 1.2 rc1 I was having problems with load >> and all users did not regisered (loosing connection). I do no know if >> this was FS problem but I do not have this problems on production >> server. Few day ago I did a new git pull. Now I have version 1.2 rc2 >> which enum do not work. I traced with wireshark but afer I put in fs_cli >> enum 123123123, Fs sends nothing. >> >> >> So on git.freeswitch.com are versions 1.0.6, 1.2 rc2 and 1.2 rc1. Which >> version to use? Is 1.0.6 most stable for now as 1.2 is rc? >> >> Where can I find version of FS which I am running on production server? >> >> Thanks for all your help! >> >> miha >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> !DSPAM:5007e36132765726119943! >> > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >Join Us At ClueCon - Aug 7-9, 2012 > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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/20120719/99e6291f/attachment.html From ben at langfeld.co.uk Thu Jul 19 17:45:23 2012 From: ben at langfeld.co.uk (Ben Langfeld) Date: Thu, 19 Jul 2012 15:45:23 +0200 Subject: [Freeswitch-users] Which FS version to use? In-Reply-To: <1342705039.82548.YahooMailNeo@web39304.mail.mud.yahoo.com> References: <5007E4D8.7050108@softnet.si> <5B0A12B0-7D51-4A8C-A9EE-9C5A2200967B@visionutveckling.se> <1342705039.82548.YahooMailNeo@web39304.mail.mud.yahoo.com> Message-ID: git-flow is a very nice branching strategy (master branch for releases, develop branch for active development, feature branches for work in progress). Unfortunately, more or less all freeswitch development happens on a single branch thanks to svn-think. Additionally, the "the latest release is HEAD" strategy is a huge problem for those people writing software which depends on freeswitch. FreeSWITCH does not follow any defined versioning pattern, and thus determining wether updates will break a system is next to impossible. Thus, many people run out-of-date freeswitch systems out of fear. If the FS team were to adopt semver (http://semver.org) then this issue would be all but resolved, but they appear to be dead set against version numbers which actually have meaning. Unfortunately the core team don't seem to realise how much of a problem this is for downstream developers, and seem unwilling to do anything to help in that regard. A sad state of affairs, and my only real criticism of FreeSWITCH. I hope 1.2 represents the beginning of a sensible and mature approach to versioning for FreeSWITCH, and it can't be released soon enough for that very reason. Regards, Ben Langfeld On 19 July 2012 15:37, Stanislav Sinyagin wrote: > > by the way, any plans to fork a "stable" Git branch and use it only for > bugfixes, whereas "master" would absorb also all new features? > > What I see now in the commit log, is the "master" branch also used for > normal development, and not only for bugfixes. This way it's difficult to > expect the final 1.2 release any time soon. > > > > > > ------------------------------ > *From:* Peter Olsson > *To:* FreeSWITCH Users Help > *Sent:* Thursday, July 19, 2012 2:47 PM > *Subject:* Re: [Freeswitch-users] Which FS version to use? > > Current git head is always the recommended version. If you find any > problems, please report them to Jira. Git head is currently at 1.2 RC2 > state (waiting for a stable 1.2 release), so the more things we get fixed, > the better! > > Please retry the ENUM problems on latest git again, and if it still fails, > report this to Jira. > > /Peter > > 19 jul 2012 kl. 12:51 skrev "Miha" : > > > Hi, > > > > on my production server I have version: > > > > freeswitch at default> version > > FreeSWITCH Version 1.0.head (git-00de8e6 2011-11-01 17-27-13 -0600) > > > > which is in my case very stable, I can say perfect. > > > > As I have prepare other, more powerfull server for FS I have installed > > FS from latest git. With version 1.2 rc1 I was having problems with load > > and all users did not regisered (loosing connection). I do no know if > > this was FS problem but I do not have this problems on production > > server. Few day ago I did a new git pull. Now I have version 1.2 rc2 > > which enum do not work. I traced with wireshark but afer I put in fs_cli > > enum 123123123, Fs sends nothing. > > > > > > So on git.freeswitch.com are versions 1.0.6, 1.2 rc2 and 1.2 rc1. Which > > version to use? Is 1.0.6 most stable for now as 1.2 is rc? > > > > Where can I find version of FS which I am running on production server? > > > > Thanks for all your help! > > > > miha > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > !DSPAM:5007e36132765726119943! > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120719/7ca5811b/attachment-0001.html From bdfoster at endigotech.com Thu Jul 19 18:02:07 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 19 Jul 2012 10:02:07 -0400 Subject: [Freeswitch-users] Which FS version to use? In-Reply-To: References: <5007E4D8.7050108@softnet.si> <5B0A12B0-7D51-4A8C-A9EE-9C5A2200967B@visionutveckling.se> <1342705039.82548.YahooMailNeo@web39304.mail.mud.yahoo.com> Message-ID: I think the point of releasing 1.2 is exactly because of the reasons you are stating. FreeSWITCH gets many commits a day, and is continually improved feature-wise and stability wise. That presents precisely the problems you are pointing out. It's like art. Artwork is never really finished, you just have to make a decision to stop. +1 for a stable branch of FreeSWITCH. But keep those juices flowing! Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 19, 2012 9:47 AM, "Ben Langfeld" wrote: > git-flow is a very nice branching strategy (master branch for releases, > develop branch for active development, feature branches for work in > progress). Unfortunately, more or less all freeswitch development happens > on a single branch thanks to svn-think. > > Additionally, the "the latest release is HEAD" strategy is a huge problem > for those people writing software which depends on freeswitch. FreeSWITCH > does not follow any defined versioning pattern, and thus determining wether > updates will break a system is next to impossible. Thus, many people run > out-of-date freeswitch systems out of fear. > > If the FS team were to adopt semver (http://semver.org) then this issue > would be all but resolved, but they appear to be dead set against version > numbers which actually have meaning. Unfortunately the core team don't seem > to realise how much of a problem this is for downstream developers, and > seem unwilling to do anything to help in that regard. A sad state of > affairs, and my only real criticism of FreeSWITCH. > > I hope 1.2 represents the beginning of a sensible and mature approach to > versioning for FreeSWITCH, and it can't be released soon enough for that > very reason. > > Regards, > Ben Langfeld > > > On 19 July 2012 15:37, Stanislav Sinyagin wrote: > >> >> by the way, any plans to fork a "stable" Git branch and use it only for >> bugfixes, whereas "master" would absorb also all new features? >> >> What I see now in the commit log, is the "master" branch also used for >> normal development, and not only for bugfixes. This way it's difficult to >> expect the final 1.2 release any time soon. >> >> >> >> >> >> ------------------------------ >> *From:* Peter Olsson >> *To:* FreeSWITCH Users Help >> *Sent:* Thursday, July 19, 2012 2:47 PM >> *Subject:* Re: [Freeswitch-users] Which FS version to use? >> >> Current git head is always the recommended version. If you find any >> problems, please report them to Jira. Git head is currently at 1.2 RC2 >> state (waiting for a stable 1.2 release), so the more things we get fixed, >> the better! >> >> Please retry the ENUM problems on latest git again, and if it still >> fails, report this to Jira. >> >> /Peter >> >> 19 jul 2012 kl. 12:51 skrev "Miha" : >> >> > Hi, >> > >> > on my production server I have version: >> > >> > freeswitch at default> version >> > FreeSWITCH Version 1.0.head (git-00de8e6 2011-11-01 17-27-13 -0600) >> > >> > which is in my case very stable, I can say perfect. >> > >> > As I have prepare other, more powerfull server for FS I have installed >> > FS from latest git. With version 1.2 rc1 I was having problems with >> load >> > and all users did not regisered (loosing connection). I do no know if >> > this was FS problem but I do not have this problems on production >> > server. Few day ago I did a new git pull. Now I have version 1.2 rc2 >> > which enum do not work. I traced with wireshark but afer I put in >> fs_cli >> > enum 123123123, Fs sends nothing. >> > >> > >> > So on git.freeswitch.com are versions 1.0.6, 1.2 rc2 and 1.2 rc1. >> Which >> > version to use? Is 1.0.6 most stable for now as 1.2 is rc? >> > >> > Where can I find version of FS which I am running on production server? >> > >> > Thanks for all your help! >> > >> > miha >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> > !DSPAM:5007e36132765726119943! >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120719/c1932945/attachment.html From aerocomputer at gmail.com Thu Jul 19 17:37:23 2012 From: aerocomputer at gmail.com (A.S. Shaja) Date: Thu, 19 Jul 2012 19:07:23 +0530 Subject: [Freeswitch-users] Re. trigger custom freeswitch application to notify users in a conference In-Reply-To: References: Message-ID: Any comments/suggestions on the below request? On Thu, Jul 19, 2012 at 11:25 AM, A.S. Shaja wrote: > Hi, > > Is it possible to create an application in FreeSWITCH that when a user > presses a series of keys, it could trigger a custom FreeSWITCH application > to notify all moderators via an audio message? Can you please help me on > how to achieve this? > > Regards, > Shaja. > -- > "TajMahal would not have been so beautiful if shajahan had asked for three > quotations and > had chosen the least one!!!" > -- "TajMahal would not have been so beautiful if shajahan had asked for three quotations and had chosen the least one!!!" -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120719/fd25c7f0/attachment.html From Finn.Hoeck at gmx.de Thu Jul 19 18:01:22 2012 From: Finn.Hoeck at gmx.de (=?iso-8859-1?Q?=22Finn_H=F6ck=22?=) Date: Thu, 19 Jul 2012 16:01:22 +0200 Subject: [Freeswitch-users] ZRTP - SAS-strings not the same Message-ID: <20120719140122.17800@gmx.net> Hi, I got freeswitch working perfectly on CentOS 6.3 by following the guide on http://wiki.freeswitch.org/wiki/Installation_Guide. Then I followed the steps listet at http://wiki.freeswitch.org/wiki/ZRTP to get ZRTP working. Now, when I do calls from a zrtp-enabled client to another, communication still works BUT the sas-strings showing up on the clients aren't the same. Now my question is: why? Does anyone have some troubleshooting-tips for me? What possible reasons are there? Thanks in advance for any help. Greetings, Finn From brian at freeswitch.org Thu Jul 19 19:18:00 2012 From: brian at freeswitch.org (Brian West) Date: Thu, 19 Jul 2012 10:18:00 -0500 Subject: [Freeswitch-users] Rooms Added Message-ID: <688791BA-B908-4ACF-93E5-3F5897855960@freeswitch.org> I have just sent the paperwork in to get more rooms on the dates everyone was having problems with. Please give them an hour or so to get them into the system. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120719/cef1d450/attachment-0001.html From ben at langfeld.co.uk Thu Jul 19 19:18:33 2012 From: ben at langfeld.co.uk (Ben Langfeld) Date: Thu, 19 Jul 2012 17:18:33 +0200 Subject: [Freeswitch-users] Re. trigger custom freeswitch application to notify users in a conference In-Reply-To: References: Message-ID: Your first post only hit the list 8 hours previously. All in good time. On 19 July 2012 15:37, A.S. Shaja wrote: > Any comments/suggestions on the below request? > > > On Thu, Jul 19, 2012 at 11:25 AM, A.S. Shaja wrote: > >> Hi, >> >> Is it possible to create an application in FreeSWITCH that when a user >> presses a series of keys, it could trigger a custom FreeSWITCH application >> to notify all moderators via an audio message? Can you please help me on >> how to achieve this? >> >> Regards, >> Shaja. >> -- >> "TajMahal would not have been so beautiful if shajahan had asked for >> three quotations and >> had chosen the least one!!!" >> > > > > -- > "TajMahal would not have been so beautiful if shajahan had asked for three > quotations and > had chosen the least one!!!" > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120719/cb6b7757/attachment.html From msc at freeswitch.org Thu Jul 19 19:46:07 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Jul 2012 08:46:07 -0700 Subject: [Freeswitch-users] ENUM do not work In-Reply-To: <5007D4CE.8030302@softnet.si> References: <50066691.9060700@softnet.si> <2007FC18-287D-4EC4-8258-2022D63A4370@gmail.com> <50067F78.3060803@softnet.si> <5007D4CE.8030302@softnet.si> Message-ID: I believe Ken Rice just applied a fix in tree. Please git pull and try again. -MC On Thu, Jul 19, 2012 at 2:35 AM, Miha wrote: > On 7/18/2012 11:18 AM, Miha wrote: > > Hi, > > > > it is revision 1.2 rc2 (installed from git two days ago). First I have > > change server to which dns lookup is made, than I add default one 164.... > > First I tought that is something wrong with configuration but than I > > noticed with wireshark that FS does not send nslookup. > > > > In 1.06 work and also on 1.2 rc1. > > > > Regards, > > Miha > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120719/c3c015a3/attachment.html From Hector.Geraldino at ipsoft.com Thu Jul 19 19:51:51 2012 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Thu, 19 Jul 2012 11:51:51 -0400 Subject: [Freeswitch-users] Re. trigger custom freeswitch application to notify users in a conference In-Reply-To: References: Message-ID: <6A6B4C284AD15042B429EB9D904544AD022FAE294F@NY1-EXMB-01.ip-soft.net> What is a moderator? Are you talking about conference? Can you elaborate? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ben Langfeld Sent: Thursday, July 19, 2012 11:19 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Re. trigger custom freeswitch application to notify users in a conference Your first post only hit the list 8 hours previously. All in good time. On 19 July 2012 15:37, A.S. Shaja > wrote: Any comments/suggestions on the below request? On Thu, Jul 19, 2012 at 11:25 AM, A.S. Shaja > wrote: Hi, Is it possible to create an application in FreeSWITCH that when a user presses a series of keys, it could trigger a custom FreeSWITCH application to notify all moderators via an audio message? Can you please help me on how to achieve this? Regards, Shaja. -- "TajMahal would not have been so beautiful if shajahan had asked for three quotations and had chosen the least one!!!" -- "TajMahal would not have been so beautiful if shajahan had asked for three quotations and had chosen the least one!!!" _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120719/3e55ed7b/attachment.html From bdfoster at endigotech.com Thu Jul 19 19:52:35 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 19 Jul 2012 11:52:35 -0400 Subject: [Freeswitch-users] Re. trigger custom freeswitch application to notify users in a conference In-Reply-To: References: Message-ID: Check out bind_digit, bind_app, and I think uuid_break Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 19, 2012 3:02 AM, "A.S. Shaja" wrote: > Hi, > > Is it possible to create an application in FreeSWITCH that when a user > presses a series of keys, it could trigger a custom FreeSWITCH application > to notify all moderators via an audio message? Can you please help me on > how to achieve this? > > Regards, > Shaja. > -- > "TajMahal would not have been so beautiful if shajahan had asked for three > quotations and > had chosen the least one!!!" > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120719/33f95c2c/attachment.html From toddb at toddbailey.net Thu Jul 19 19:55:10 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Thu, 19 Jul 2012 08:55:10 -0700 Subject: [Freeswitch-users] Looking for dial plan examples for FS and SPA3102 router In-Reply-To: <1342564671.3414.63.camel@mythtv> References: <1342564671.3414.63.camel@mythtv> Message-ID: <1342713310.18217.16.camel@mythtv> Any one ? On Tue, 2012-07-17 at 15:37 -0700, Todd Bailey wrote: > Hi All, > > > I'm having issues getting a dial plan to work on FS and a Cisco SPA 3102 > router. > > when I dial 0, 1 or 9 plus a 10 number, I get to the router's dial tone > but I have to reenter the number I want to connect to. > > The expected action is to only need to enter the number to dial one > > Can some one provide dial plan and/or other config file example on how > to resolve this issue? > > here is what I have so far: > > /usr/local/freeswitch/conf/dialplan/default.xml > > > > > > expression="^(1{0,1,9}\d{10})$"> > > data="effective_caller_id_number=12223334444"/> > > > > > data="sofia/internal/$1 at 192.168.1.5:5061" /> > > > > > /usr/local/freeswitch/conf/dialplan/default/00_spa3102.xml > > > > > data="sofia/internal/${destination_number}@192.168.1.5:5061" /> > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 freeswitch.org Thu Jul 19 20:44:39 2012 From: krice at freeswitch.org (Ken Rice) Date: Thu, 19 Jul 2012 11:44:39 -0500 Subject: [Freeswitch-users] ENUM do not work In-Reply-To: <5007D4CE.8030302@softnet.si> Message-ID: This has been fixed in tree as of this morning... Please update and try again... There was a slight regression recently with a patch that allows you to specify specific name servers enum to query against... Note: this didn't appear to affect the dialplan only the event socket API call (the API call is what you get from the console) K On 7/19/12 4:35 AM, "Miha" wrote: > On 7/18/2012 11:18 AM, Miha wrote: >> Hi, >> >> it is revision 1.2 rc2 (installed from git two days ago). First I have >> change server to which dns lookup is made, than I add default one 164.... >> First I tought that is something wrong with configuration but than I >> noticed with wireshark that FS does not send nslookup. >> >> In 1.06 work and also on 1.2 rc1. >> >> Regards, >> Miha >> >> On 7/18/2012 10:57 AM, Jay Binks wrote: >>> Divide and conquer , figure out which revision broke it and I'll help you. >>> >>> What do you see on the console and what enum config do you have . >>> >>> >>> >>> On 18/07/2012, at 5:32 PM, Miha wrote: >>> >>>> Hi, >>>> >>>> is there any problem with enum on new git? When I do nslookup in linux, >>>> I get result but if I do it from fs_cli (enum 1231231231) I can see with >>>> wireshark that FS do not send lookup. >>>> >>>> p.s.: I have same configuration on different older FS and works. >>>> >>>> Thanks! >>>> miha >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > Revision of FS is: FreeSWITCH Version > 1.2.0-rc2+git~20120719T040232Z~ec412c07d2+unclean~20120719T084055Z > > Regards,Miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From paul at cupis.co.uk Thu Jul 19 20:48:18 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Thu, 19 Jul 2012 17:48:18 +0100 Subject: [Freeswitch-users] ENUM do not work In-Reply-To: References: <50066691.9060700@softnet.si> <2007FC18-287D-4EC4-8258-2022D63A4370@gmail.com> <50067F78.3060803@softnet.si> <5007D4CE.8030302@softnet.si> Message-ID: <20120719164818.GA3588@eagle.cupis.co.uk> On Thu, Jul 19, 2012 at 08:46:07AM -0700, Michael Collins wrote: > I believe Ken Rice just applied a fix in tree. Please git pull and try > again. I can confirm that this update does fix the issue. Specifically, mod_enum was changed to required the nameservers to be specified in the config file, rather than using the host systems DNS resolution. This caused the failure we are talking about, as existing configuration did not contain DNS servers in the mod_enum settings. With the latest version, if there are no nameservers in the mod_enum configuration file, it will default to using the system DNS lookup instead of not doing any DNS lookup. Regards, From msc at freeswitch.org Thu Jul 19 20:52:04 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Jul 2012 09:52:04 -0700 Subject: [Freeswitch-users] ENUM do not work In-Reply-To: <20120719164818.GA3588@eagle.cupis.co.uk> References: <50066691.9060700@softnet.si> <2007FC18-287D-4EC4-8258-2022D63A4370@gmail.com> <50067F78.3060803@softnet.si> <5007D4CE.8030302@softnet.si> <20120719164818.GA3588@eagle.cupis.co.uk> Message-ID: Thank you for not only testing but reporting back what you actually found and with a nice explanation! We appreciate you taking the extra 15 minutes to keep us all updated. If you come to ClueCon we'll all chip in and buy you pizza and beer. :) -MC On Thu, Jul 19, 2012 at 9:48 AM, Paul Cupis wrote: > On Thu, Jul 19, 2012 at 08:46:07AM -0700, Michael Collins wrote: > > I believe Ken Rice just applied a fix in tree. Please git pull and try > > again. > > I can confirm that this update does fix the issue. > > Specifically, mod_enum was changed to required the nameservers to be > specified in the config file, rather than using the host systems DNS > resolution. This caused the failure we are talking about, as existing > configuration did not contain DNS servers in the mod_enum settings. > > With the latest version, if there are no nameservers in the mod_enum > configuration file, it will default to using the system DNS lookup > instead of not doing any DNS lookup. > > Regards, > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120719/2c7c2d19/attachment.html From mario_fs at mgtech.com Thu Jul 19 21:01:35 2012 From: mario_fs at mgtech.com (Mario G) Date: Thu, 19 Jul 2012 10:01:35 -0700 Subject: [Freeswitch-users] leg_delay_start does not work in bridge enterprise Message-ID: <9D0E55F4-768E-4C9D-B1DF-9D074E44ADD7@mgtech.com> leg_delay_start is ignored when using the enterprise syntax below. I need the second target delayed and there appears no other way to do this. Both targets are called at the same time. Mario G -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120719/f4b4d51c/attachment.html From krice at freeswitch.org Thu Jul 19 21:17:14 2012 From: krice at freeswitch.org (Ken Rice) Date: Thu, 19 Jul 2012 12:17:14 -0500 Subject: [Freeswitch-users] Which FS version to use? In-Reply-To: Message-ID: A little history here... We used to do releases... See 1.0... And the 1.0.x releases Why did we stop? Because no one would test (as developers we can only test so far... Users fine new and exciting ways to break software all the time... There is probably 5 times the number of what I would call edge use cases then there are what I would consider normal use cases... We cant possibly test every possible permutation... So there ended up frustrations and pains cause people were using old revs then complaing about bugs that were fixed 2 months back or we had just released 1.0.2 and within 3 hours of the release a mjor bug pops up... We are moving back to a release cycle with 1.2... No amount of testing software that we have to run can address all the automatted testing... We have to have users testing and reporting bugs and when we fix bugs we need those fixes tested and reported back good/bad... Now more recently I came on board with the dev team to help facilitate that... So guys help us help you... Test and report the bugs... Test head... And report bugs... If you have a possible solution to a bug let us know... We want 1.2 to get here and theres been a lot of changes since a previous release.... K On 7/19/12 9:02 AM, "Brian Foster" wrote: > I think the point of releasing 1.2 is exactly because of the reasons you are > stating. > > FreeSWITCH gets many commits a day, and is continually improved feature-wise > and stability wise. That presents precisely the problems you are pointing out. > It's like art. Artwork is never really finished, you just have to make a > decision to stop. > > +1 for a stable branch of FreeSWITCH. But keep those juices flowing! > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > > On Jul 19, 2012 9:47 AM, "Ben Langfeld" wrote: >> git-flow is a very nice branching strategy (master branch for releases, >> develop branch for active development, feature branches for work in >> progress). Unfortunately, more or less all freeswitch development happens on >> a single branch thanks to svn-think. >> >> Additionally, the "the latest release is HEAD" strategy is a huge problem for >> those people writing software which depends on freeswitch. FreeSWITCH does >> not follow any defined versioning pattern, and thus determining wether >> updates will break a system is next to impossible. Thus, many people run >> out-of-date freeswitch systems out of fear. >> >> If the FS team were to adopt semver (http://semver.org) then this issue would >> be all but resolved, but they appear to be dead set against version numbers >> which actually have meaning. Unfortunately the core team don't seem to >> realise how much of a problem this is for downstream developers, and seem >> unwilling to do anything to help in that regard. A sad state of affairs, and >> my only real criticism of FreeSWITCH. >> >> I hope 1.2 represents the beginning of a sensible and mature approach to >> versioning for FreeSWITCH, and it can't be released soon enough for that very >> reason. >> >> Regards, >> Ben Langfeld >> >> >> On 19 July 2012 15:37, Stanislav Sinyagin wrote: >>> >>> by the way, any plans to fork a "stable" Git branch and use it only for >>> bugfixes, whereas "master" would absorb also all new features? >>> >>> What I see now in the commit log, is the "master" branch also used for >>> normal development, and not only for bugfixes. This way it's difficult to >>> expect the final 1.2 release any time soon. >>> >>> >>> >>> >>> >>>> >>>> >>>> >>>> >>>> >>>> From: Peter Olsson >>>> To: FreeSWITCH Users Help >>>> Sent: Thursday, July 19, 2012 2:47 PM >>>> Subject: Re: [Freeswitch-users] Which FS version to use? >>>> >>>> >>>> Current git head is always the recommended version. If you find any >>>> problems, please report them to Jira. Git head is currently at 1.2 RC2 >>>> state (waiting for a stable 1.2 release), so the more things we get fixed, >>>> the better! >>>> >>>> Please retry the ENUM problems on latest git again, and if it still fails, >>>> report this to Jira. >>>> >>>> /Peter >>>> >>>> 19 jul 2012 kl. 12:51 skrev "Miha" : >>>> >>>>> > Hi, >>>>> > >>>>> > on my production server I have version: >>>>> > >>>>> > freeswitch at default> version >>>>> > FreeSWITCH Version 1.0.head (git-00de8e6 2011-11-01 17-27-13 -0600) >>>>> > >>>>> > which is in my case very stable, I can say perfect. >>>>> > >>>>> > As I have prepare other, more powerfull server for FS I have installed >>>>> > FS from latest git. With version 1.2 rc1 I was having problems with load >>>>> > and all users did not regisered (loosing connection). I do no know if >>>>> > this was FS problem but I do not have this problems on production >>>>> > server. Few day ago I did a new git pull. Now I have version 1.2 rc2 >>>>> > which enum do not work. I traced with wireshark but afer I put in fs_cli >>>>> > enum 123123123, Fs sends nothing. >>>>> > >>>>> > >>>>> > So on git.freeswitch.com are versions >>>>> 1.0.6, 1.2 rc2 and 1.2 rc1. Which >>>>> > version to use? Is 1.0.6 most stable for now as 1.2 is rc? >>>>> > >>>>> > Where can I find version of FS which I am running on production server? >>>>> > >>>>> > Thanks for all your help! >>>>> > >>>>> > miha >>>>> > >>>>> > >>>>> _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://wiki.freeswitch.org >>>>> > http://www.cluecon.com >>>>> > >>>>> > Join Us At ClueCon - Aug 7-9, 2012 >>>>> > >>>>> > FreeSWITCH-users mailing list >>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>>> > >>>>> > !DSPAM:5007e36132765726119943! >>>>> > >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120719/4acccaac/attachment-0001.html From curriegrad2004 at gmail.com Thu Jul 19 21:23:24 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Thu, 19 Jul 2012 10:23:24 -0700 Subject: [Freeswitch-users] Which FS version to use? In-Reply-To: References: Message-ID: In other words, we need people testing and reporting bugs on JIRA and providing feedback on the fixes. We could always "assume" that the issue has been fixed from people who never leave feedback on the fixes anyways. On Thu, Jul 19, 2012 at 10:17 AM, Ken Rice wrote: > A little history here... We used to do releases... See 1.0... And the 1.0.x > releases > > Why did we stop? > > Because no one would test (as developers we can only test so far... Users > fine new and exciting ways to break software all the time... > > There is probably 5 times the number of what I would call edge use cases > then there are what I would consider normal use cases... We cant possibly > test every possible permutation... So there ended up frustrations and pains > cause people were using old revs then complaing about bugs that were fixed 2 > months back or we had just released 1.0.2 and within 3 hours of the release > a mjor bug pops up... > > We are moving back to a release cycle with 1.2... No amount of testing > software that we have to run can address all the automatted testing... We > have to have users testing and reporting bugs and when we fix bugs we need > those fixes tested and reported back good/bad... > > Now more recently I came on board with the dev team to help facilitate > that... So guys help us help you... Test and report the bugs... Test head... > And report bugs... If you have a possible solution to a bug let us know... > We want 1.2 to get here and theres been a lot of changes since a previous > release.... > > K > > > > On 7/19/12 9:02 AM, "Brian Foster" wrote: > > I think the point of releasing 1.2 is exactly because of the reasons you are > stating. > > FreeSWITCH gets many commits a day, and is continually improved feature-wise > and stability wise. That presents precisely the problems you are pointing > out. It's like art. Artwork is never really finished, you just have to make > a decision to stop. > > +1 for a stable branch of FreeSWITCH. But keep those juices flowing! > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > > On Jul 19, 2012 9:47 AM, "Ben Langfeld" wrote: > > git-flow is a very nice branching strategy (master branch for releases, > develop branch for active development, feature branches for work in > progress). Unfortunately, more or less all freeswitch development happens on > a single branch thanks to svn-think. > > Additionally, the "the latest release is HEAD" strategy is a huge problem > for those people writing software which depends on freeswitch. FreeSWITCH > does not follow any defined versioning pattern, and thus determining wether > updates will break a system is next to impossible. Thus, many people run > out-of-date freeswitch systems out of fear. > > If the FS team were to adopt semver (http://semver.org) then this issue > would be all but resolved, but they appear to be dead set against version > numbers which actually have meaning. Unfortunately the core team don't seem > to realise how much of a problem this is for downstream developers, and seem > unwilling to do anything to help in that regard. A sad state of affairs, and > my only real criticism of FreeSWITCH. > > I hope 1.2 represents the beginning of a sensible and mature approach to > versioning for FreeSWITCH, and it can't be released soon enough for that > very reason. > > Regards, > Ben Langfeld > > > On 19 July 2012 15:37, Stanislav Sinyagin wrote: > > > by the way, any plans to fork a "stable" Git branch and use it only for > bugfixes, whereas "master" would absorb also all new features? > > What I see now in the commit log, is the "master" branch also used for > normal development, and not only for bugfixes. This way it's difficult to > expect the final 1.2 release any time soon. > > > > > > > > > > ________________________________ > From: Peter Olsson > To: FreeSWITCH Users Help > Sent: Thursday, July 19, 2012 2:47 PM > Subject: Re: [Freeswitch-users] Which FS version to use? > > > > Current git head is always the recommended version. If you find any > problems, please report them to Jira. Git head is currently at 1.2 RC2 state > (waiting for a stable 1.2 release), so the more things we get fixed, the > better! > > Please retry the ENUM problems on latest git again, and if it still fails, > report this to Jira. > > /Peter > > 19 jul 2012 kl. 12:51 skrev "Miha" : > >> Hi, >> >> on my production server I have version: >> >> freeswitch at default> version >> FreeSWITCH Version 1.0.head (git-00de8e6 2011-11-01 17-27-13 -0600) >> >> which is in my case very stable, I can say perfect. >> >> As I have prepare other, more powerfull server for FS I have installed >> FS from latest git. With version 1.2 rc1 I was having problems with load >> and all users did not regisered (loosing connection). I do no know if >> this was FS problem but I do not have this problems on production >> server. Few day ago I did a new git pull. Now I have version 1.2 rc2 >> which enum do not work. I traced with wireshark but afer I put in fs_cli >> enum 123123123, Fs sends nothing. >> >> >> So on git.freeswitch.com are versions 1.0.6, >> 1.2 rc2 and 1.2 rc1. Which >> version to use? Is 1.0.6 most stable for now as 1.2 is rc? >> >> Where can I find version of FS which I am running on production server? >> >> Thanks for all your help! >> >> miha >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> !DSPAM:5007e36132765726119943! >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ________________________________ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From toddb at toddbailey.net Thu Jul 19 21:51:29 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Thu, 19 Jul 2012 10:51:29 -0700 Subject: [Freeswitch-users] Skype (+connect) Freeswitch/SIPfoundry Message-ID: <1342720289.28405.9.camel@mythtv> I'll be installing Fedora F17 over the next few days and hope to take my successes and er well, failures on FreeSwitch (FS) and move them into the new install. In looking at Skype it states to use SIP approved pbx's and SIPFoundry (SF) is one of them. In looking at SF I see a lot of FS rpms etc. Any one care to help minimize my confusion on what the differences are and what might be the better product to install? I'd like to use skype for outbound long distance calls, an added plus would be to have bi-directional calls, but outbound calling is key. I'm still using a analog line for local bidirectional calling but it has it's own set of issues. thanks From marketing at cluecon.com Thu Jul 19 22:06:11 2012 From: marketing at cluecon.com (Michael Collins) Date: Thu, 19 Jul 2012 11:06:11 -0700 Subject: [Freeswitch-users] How You Can Help With H.323 In FreeSWITCH Message-ID: Greetings! We would like to let you know about an opportunity we all have to assist with H.323 support in FreeSWITCH. We are working on getting Robert Jongbloed, author of mod_opal and all around cool guy, to ClueCon 2012. Robert does a lot of work with H.323 and his presence with us at ClueCon would go a long way in helping us get H.323 fully implemented in FreeSWITCH. The challenge is that Robert lives in Australia! To that end we are taking up a collection to help defray the substantial travel costs. Here's what you can do: visit FreeSWITCH.org and click the Donate button. In your Paypal donation, click "Add special instructions to the seller" and make a note that this donation is to assist with getting Robert to ClueCon. We'll take care of the rest. Thanks again for all of your wonderful support! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120719/061d26ff/attachment.html From ssinyagin at yahoo.com Thu Jul 19 22:23:48 2012 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Thu, 19 Jul 2012 11:23:48 -0700 (PDT) Subject: [Freeswitch-users] Which FS version to use? In-Reply-To: References: Message-ID: <1342722228.4238.YahooMailNeo@web39305.mail.mud.yahoo.com> Ken, a clean branching strategy and committing policy should actually reduce the number of faults and reduce the maintenance effort. looking at the current commit log, you can easily see that about 20% of commits are actually introducing new features or changing old ones. So at least we should stop calling it a release candidate. >________________________________ > From: curriegrad2004 >To: FreeSWITCH Users Help >Sent: Thursday, July 19, 2012 7:23 PM >Subject: Re: [Freeswitch-users] Which FS version to use? > >In other words, we need people testing and reporting bugs on JIRA and >providing feedback on the fixes. > >We could always "assume" that the issue has been fixed from people who >never leave feedback on the fixes anyways. > >On Thu, Jul 19, 2012 at 10:17 AM, Ken Rice wrote: >> A little history here... We used to do releases... See 1.0... And the 1.0.x >> releases >> >> Why did we stop? >> >> Because no one would test (as developers we can only test so far... Users >> fine new and exciting ways to break software all the time... >> >> There is probably 5 times the number of what I would call edge use cases >> then there are what I would consider normal use cases... We cant possibly >> test every possible permutation... So there ended up frustrations and pains >> cause people were using old revs then complaing about bugs that were fixed 2 >> months back or we had just released 1.0.2 and within 3 hours of the release >> a mjor bug pops up... >> >> We are moving back to a release cycle with 1.2... No amount of testing >> software that we have to run can address all the automatted testing... We >> have to have users testing and reporting bugs and when we fix bugs we need >> those fixes tested and reported back good/bad... >> >> Now more recently I came on board with the dev team to help facilitate >> that... So guys help us help you... Test and report the bugs... Test head... >> And report bugs... If you have a possible solution to a bug let us know... >> We want 1.2 to get here and theres been a lot of changes since a previous >> release.... >> >> K >> >> >> >> On 7/19/12 9:02 AM, "Brian Foster" wrote: >> >> I think the point of releasing 1.2 is exactly because of the reasons you are >> stating. >> >> FreeSWITCH gets many commits a day, and is continually improved feature-wise >> and stability wise. That presents precisely the problems you are pointing >> out. It's like art. Artwork is never really finished, you just have to make >> a decision to stop. >> >> +1 for a stable branch of FreeSWITCH. But keep those juices flowing! >> >> Brian Foster >> Endigo Computer LLC >> >> Sent from a mobile device. >> >> On Jul 19, 2012 9:47 AM, "Ben Langfeld" wrote: >> >> git-flow is a very nice branching strategy (master branch for releases, >> develop branch for active development, feature branches for work in >> progress). Unfortunately, more or less all freeswitch development happens on >> a single branch thanks to svn-think. >> >> Additionally, the "the latest release is HEAD" strategy is a huge problem >> for those people writing software which depends on freeswitch. FreeSWITCH >> does not follow any defined versioning pattern, and thus determining wether >> updates will break a system is next to impossible. Thus, many people run >> out-of-date freeswitch systems out of fear. >> >> If the FS team were to adopt semver (http://semver.org) then this issue >> would be all but resolved, but they appear to be dead set against version >> numbers which actually have meaning. Unfortunately the core team don't seem >> to realise how much of a problem this is for downstream developers, and seem >> unwilling to do anything to help in that regard. A sad state of affairs, and >> my only real criticism of FreeSWITCH. >> >> I hope 1.2 represents the beginning of a sensible and mature approach to >> versioning for FreeSWITCH, and it can't be released soon enough for that >> very reason. >> >> Regards, >> Ben Langfeld >> >> >> On 19 July 2012 15:37, Stanislav Sinyagin wrote: >> >> >> by the way, any plans to fork a "stable" Git branch and use it only for >> bugfixes, whereas "master" would absorb also all new features? >> >> What I see now in the commit log, is the "master" branch also used for >> normal development, and not only for bugfixes. This way it's difficult to >> expect the final 1.2 release any time soon. >> >> >> >> >> >> >> >> >> >> ________________________________ >>? From: Peter Olsson >>? To: FreeSWITCH Users Help >>? Sent: Thursday, July 19, 2012 2:47 PM >>? Subject: Re: [Freeswitch-users] Which FS version to use? >> >> >> >> Current git head is always the recommended version. If you find any >> problems, please report them to Jira. Git head is currently at 1.2 RC2 state >> (waiting for a stable 1.2 release), so the more things we get fixed, the >> better! >> >> Please retry the ENUM problems on latest git again, and if it still fails, >> report this to Jira. >> >> /Peter >> >> 19 jul 2012 kl. 12:51 skrev "Miha" : >> >>> Hi, >>> >>> on my production server I have version: >>> >>> freeswitch at default> version >>> FreeSWITCH Version 1.0.head (git-00de8e6 2011-11-01 17-27-13 -0600) >>> >>> which is in my case very stable, I can say perfect. >>> >>> As I have prepare other, more powerfull server for FS I have installed >>> FS from latest git. With version 1.2 rc1 I was having problems with load >>> and all users did not regisered (loosing connection). I do no know if >>> this was FS problem but I do not have this problems on production >>> server. Few day ago I did a new git pull. Now I have version 1.2 rc2 >>> which enum do not work. I traced with wireshark but afer I put in fs_cli >>> enum 123123123, Fs sends nothing. >>> >>> >>> So on git.freeswitch.com ? are versions 1.0.6, >>> 1.2 rc2 and 1.2 rc1. Which >>> version to use? Is 1.0.6 most stable for now as 1.2 is rc? >>> >>> Where can I find version of FS which I am running on production server? >>> >>> Thanks for all your help! >>> >>> miha >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> !DSPAM:5007e36132765726119943! >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ________________________________ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >Join Us At ClueCon - Aug 7-9, 2012 > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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/20120719/c47a9f7e/attachment-0001.html From msc at freeswitch.org Thu Jul 19 22:36:56 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Jul 2012 11:36:56 -0700 Subject: [Freeswitch-users] Re. trigger custom freeswitch application to notify users in a conference In-Reply-To: References: Message-ID: On Wed, Jul 18, 2012 at 10:55 PM, A.S. Shaja wrote: > Hi, > > Is it possible to create an application in FreeSWITCH that when a user > presses a series of keys, it could trigger a custom FreeSWITCH application > to notify all moderators via an audio message? Can you please help me on > how to achieve this? > This will most likely require some scripting. I recommend reading up on bind_digit_action and also Lua. It's not a simple thing to show in just a few lines of code and I'm positive that there's nothing already on the wiki to show it. If you figure it out we'd love to hear what you did. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120719/89b6d686/attachment.html From bdfoster at endigotech.com Thu Jul 19 22:45:53 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 19 Jul 2012 14:45:53 -0400 Subject: [Freeswitch-users] Skype (+connect) Freeswitch/SIPfoundry In-Reply-To: <1342720289.28405.9.camel@mythtv> References: <1342720289.28405.9.camel@mythtv> Message-ID: Don't see a reason to use sipfoundry when mod_skypopen works fine. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 19, 2012 1:53 PM, "Todd Bailey" wrote: > I'll be installing Fedora F17 over the next few days and hope to take my > successes and er well, failures on FreeSwitch (FS) and move them into > the new install. > > In looking at Skype it states to use SIP approved pbx's and SIPFoundry > (SF) is one of them. In looking at SF I see a lot of FS rpms etc. > > Any one care to help minimize my confusion on what the differences are > and what might be the better product to install? > > I'd like to use skype for outbound long distance calls, an added plus > would be to have bi-directional calls, but outbound calling is key. > > I'm still using a analog line for local bidirectional calling but it > has it's own set of issues. > > thanks > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120719/81bf77ac/attachment.html From msc at freeswitch.org Thu Jul 19 22:50:07 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Jul 2012 11:50:07 -0700 Subject: [Freeswitch-users] Lua outbound executing another lua script In-Reply-To: <3cd5711b-f322-6acd-80d0-dc2d41aadc5f@me.com> References: <3cd5711b-f322-6acd-80d0-dc2d41aadc5f@me.com> Message-ID: First off, welcome to FreeSWITCH! I would recommend that you do as much as possible in the dialplan and as little as possible in Lua. Lua makes things like if/then/else blocks trivially easy. The XML dialplan makes connecting calls trivially easy. The sweet spot is to find a nice balance. From what I see in your samples below, you don't even need the receivewithwhisper.lua on the b-leg. You could create a very simple dialplan extension that has each of the dp apps that you are executing in that Lua script. Then your execute_on_answer will be a transfer to that extension instead of calling Lua. Maybe something like this... Modify initiator script: new_session = freeswitch.Session("[execute_on_answer=transfer:whisper_answer]sofia/external/12345 at mydest"); Then add an extension: Tinker with that and see what you can get accomplished. -MC On Thu, Jul 19, 2012 at 6:18 AM, Alpha Niner wrote: > Hi, > > Firstly apologies for the newbie question. I've tried searching the > various documentation and trying a few different things but to no avail, so > I've come here to ask for help. I think my problem relates to sessions, > but I'm not totally sure. > > What I'm trying to do is: > > 1. A lua script is run by someone calling into FS (the initiator) > 2. They select an option, which originates and makes an outbound call to > another phone number > 3. The 3rd party (hopefully) answers their phone and receives a prompt > that they're going to be connected (the receiver) > 4. Both parties join a conference and talk to each other (a conference as > there may be other parties in future) > > To do this I'm creating a session and using execute_on_answer to run the > second script and provide the prompt to the receiving party. However the > receiving party never hears a prompt, comes into the conference, > immediately drops out and hangs up. > > Here are the scripts, they're pretty simple; > > The initiator: > > session:answer() > session:streamFile("WaitForOutdial.wav") > new_session = freeswitch.Session("[execute_on_answer=lua > /usr/local/freeswitch/scripts/receivewithwhisper.lua]sofia/external/12345 at mydest"); > > session:execute("conference", "CONFTEST at default+flags{endconf, > moderator}") > session:hangup(); > > The above works fine and the outbound call is placed. The script below is > then executed but the receiving phone rings but nothing is heard, the > conference is joined/exited and it terminates again; > > > The receiver (receivewithwhisper.lua): > > session:answer() > session:streamFile("ConnectingToCaller.wav") > session:execute("conference", "CONFTEST at default+flags{endconf, > moderator}") > session:hangup(); > > The receiving scripts runs when the receiver answers their phone, and > executes (quickly) and terminates. I was expecting it to drop into the > conference and I'd be able to talk to the other person. It does join the > conference, but then exits right away and falls through to the hangup > section. > > As I say, I'm not exactly sure if this is the best way (maybe there's > another - suggestions are welcome) and not exactly sure what the problem is > although I suspect it's something to do with the sessions? > > Any help, advice, corrections or examples would be appreciated. It's all > done in lua and I've tried to avoid session.originate as I understand this > is deprecated. > > Thanks > > Adam > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120719/503c7d3a/attachment.html From msc at freeswitch.org Thu Jul 19 22:52:11 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Jul 2012 11:52:11 -0700 Subject: [Freeswitch-users] Looking for dial plan examples for FS and SPA3102 router In-Reply-To: <1342713310.18217.16.camel@mythtv> References: <1342564671.3414.63.camel@mythtv> <1342713310.18217.16.camel@mythtv> Message-ID: Do you have console log and sip trace between fs and spa3102? If so, drop into pastebin.freeswitch.org and link back here. -MC On Thu, Jul 19, 2012 at 8:55 AM, Todd Bailey wrote: > Any one ? > > > On Tue, 2012-07-17 at 15:37 -0700, Todd Bailey wrote: > > Hi All, > > > > > > I'm having issues getting a dial plan to work on FS and a Cisco SPA 3102 > > router. > > > > when I dial 0, 1 or 9 plus a 10 number, I get to the router's dial tone > > but I have to reenter the number I want to connect to. > > > > The expected action is to only need to enter the number to dial one > > > > Can some one provide dial plan and/or other config file example on how > > to resolve this issue? > > > > here is what I have so far: > > > > /usr/local/freeswitch/conf/dialplan/default.xml > > > > > > > > > > > > > expression="^(1{0,1,9}\d{10})$"> > > > > > data="effective_caller_id_number=12223334444"/> > > > > > > > > > > > data="sofia/internal/$1 at 192.168.1.5:5061" /> > > > > > > > > > > /usr/local/freeswitch/conf/dialplan/default/00_spa3102.xml > > > > > > > > > > > data="sofia/internal/${destination_number}@192.168.1.5:5061" > /> > > > > > > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120719/32464da3/attachment-0001.html From mitch.capper at gmail.com Thu Jul 19 23:02:07 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Thu, 19 Jul 2012 15:02:07 -0400 Subject: [Freeswitch-users] leg_delay_start does not work in bridge enterprise In-Reply-To: <9D0E55F4-768E-4C9D-B1DF-9D074E44ADD7@mgtech.com> References: <9D0E55F4-768E-4C9D-B1DF-9D074E44ADD7@mgtech.com> Message-ID: leg_delay_start is meant to not wait if there are no other legs it is waiting on, this was designed to try a local user for example before trying their cell phone, but if they are not registered it goes right to the secondary leg. ~Mitch On Thu, Jul 19, 2012 at 1:01 PM, Mario G wrote: > leg_delay_start is ignored when using the enterprise syntax below. I need > the second target delayed and there appears no other way to do this. Both > targets are called at the same time. > > Mario G > > data="{originate_timeout=60}${group_call(main@${domain_name}+A)}:_:{leg_delay_start=20,leg_timeout=23}sofia/gateway/${dial_gateway}/19161234567"/> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From amilkhanzada at gmail.com Thu Jul 19 23:17:30 2012 From: amilkhanzada at gmail.com (Amil) Date: Thu, 19 Jul 2012 12:17:30 -0700 Subject: [Freeswitch-users] Reloading mod_python dist-packages? In-Reply-To: References: Message-ID: Any ideas? If it is not possible, please let me know. Thanks, Amil http://www.amilkhanzada.com/2012/06/gmail-keeping-mailing-list-topics-you.html On Sun, Jul 15, 2012 at 8:57 PM, Amil wrote: > So, I am writing some python scripts that are copied into the > dist-packages directory of python. > However, I am also using python scripts in the FreeSWITCH scripts > directory that refer to these python scripts in the dist-packages. > To re-copy the files, I need to run "sudo python setup_fs.py" each time I > make a change. > > However, I also need to restart FreeSWITCH because I don't know of a way > to reload the dist-packages used by mod_python. > > Any ideas? > > PS: This is the project I am referring to: < > https://github.com/kheimerl/libvbts> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120719/8fc0c99e/attachment.html From sshah at crexendo.com Thu Jul 19 23:20:05 2012 From: sshah at crexendo.com (Shaunak Shah) Date: Thu, 19 Jul 2012 19:20:05 +0000 Subject: [Freeswitch-users] mod_opus difficulties Message-ID: <46B74B1B3585614A83A1EE696DD942A24CAEAC@SOMAIL1.Storesonline.local> I am trying to get mod_opus to work with freeswitch. I am able to load mod_opus alright but to be able to use OPUS there needs to be something related to OPUS in vars.xml. What do I add is vars.xml. Currently my vars.xml has following: I want to test audio quality with OPUS so I need to put some string in vars.xml ? I tried "Opus-0.9.0" and "OPUS (BETA 0.9.0)" but none of them seem to work and I get a 488 - Not Accepted here while negotiating codecs. Thanks in advance. -- Shaunak Shah -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120719/8458837f/attachment.html From mario_fs at mgtech.com Fri Jul 20 00:02:17 2012 From: mario_fs at mgtech.com (Mario G) Date: Thu, 19 Jul 2012 13:02:17 -0700 Subject: [Freeswitch-users] leg_delay_start does not work in bridge enterprise In-Reply-To: References: <9D0E55F4-768E-4C9D-B1DF-9D074E44ADD7@mgtech.com> Message-ID: <458A0010-EC8C-4ACC-B08D-A02E0AD26077@mgtech.com> I thought that might be the case. OK then we need an enhancement request to FreeSwitch I guess. Thanks! On Jul 19, 2012, at 12:02 PM, Mitch Capper wrote: > leg_delay_start is meant to not wait if there are no other legs it is > waiting on, this was designed to try a local user for example before > trying their cell phone, but if they are not registered it goes right > to the secondary leg. > > ~Mitch > > On Thu, Jul 19, 2012 at 1:01 PM, Mario G wrote: >> leg_delay_start is ignored when using the enterprise syntax below. I need >> the second target delayed and there appears no other way to do this. Both >> targets are called at the same time. >> >> Mario G >> >> > data="{originate_timeout=60}${group_call(main@${domain_name}+A)}:_:{leg_delay_start=20,leg_timeout=23}sofia/gateway/${dial_gateway}/19161234567"/> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From prasd.d.b at gmail.com Fri Jul 20 01:35:27 2012 From: prasd.d.b at gmail.com (Prasd D) Date: Thu, 19 Jul 2012 14:35:27 -0700 Subject: [Freeswitch-users] Webrtc In-Reply-To: References: Message-ID: I think support for this should be added soon too ! On 7/17/12, Tamas Jalsovszky wrote: > Hello, > > Is there any plan to support webrtc in FS? Right I've found this post about > asterisk: > http://blogs.digium.com/2012/05/23/asterisk-11-webrtc/ > > T. > -- Thanks, Prasd From anthony.minessale at gmail.com Fri Jul 20 02:15:50 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Jul 2012 17:15:50 -0500 Subject: [Freeswitch-users] Webrtc In-Reply-To: References: Message-ID: Now accepting sponsors for this effort! On Thu, Jul 19, 2012 at 4:35 PM, Prasd D wrote: > I think support for this should be added soon too ! > > On 7/17/12, Tamas Jalsovszky wrote: >> Hello, >> >> Is there any plan to support webrtc in FS? Right I've found this post about >> asterisk: >> http://blogs.digium.com/2012/05/23/asterisk-11-webrtc/ >> >> T. >> > > > -- > Thanks, > Prasd > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From kris at kriskinc.com Fri Jul 20 02:19:59 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Thu, 19 Jul 2012 18:19:59 -0400 Subject: [Freeswitch-users] Webrtc In-Reply-To: References: Message-ID: In all seriousness for WebRTC support you need to start with full STUN/TURN/ICE support. FreeSWITCH already has mod_opus and mod_isac so audio is supported. On Thu, Jul 19, 2012 at 6:15 PM, Anthony Minessale wrote: > Now accepting sponsors for this effort! > > On Thu, Jul 19, 2012 at 4:35 PM, Prasd D wrote: >> I think support for this should be added soon too ! >> >> On 7/17/12, Tamas Jalsovszky wrote: >>> Hello, >>> >>> Is there any plan to support webrtc in FS? Right I've found this post about >>> asterisk: >>> http://blogs.digium.com/2012/05/23/asterisk-11-webrtc/ >>> >>> T. >>> >> >> >> -- >> Thanks, >> Prasd >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 From anthony.minessale at gmail.com Fri Jul 20 02:47:13 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 19 Jul 2012 17:47:13 -0500 Subject: [Freeswitch-users] Webrtc In-Reply-To: References: Message-ID: Oops, I was actually trying to sound serious and garner some sponsorship for the effort ;) Also anyone who wants to see OPAL H.323 working, we're accepting pledges to get Robert from OPAL to the upcoming ClueCon so we can provide moral support. He's down under so it takes some hefty coin 3k.... Maybe we should start a how you can help section on our homepage! On Thu, Jul 19, 2012 at 5:19 PM, Kristian Kielhofner wrote: > In all seriousness for WebRTC support you need to start with full > STUN/TURN/ICE support. FreeSWITCH already has mod_opus and mod_isac > so audio is supported. > > On Thu, Jul 19, 2012 at 6:15 PM, Anthony Minessale > wrote: >> Now accepting sponsors for this effort! >> >> On Thu, Jul 19, 2012 at 4:35 PM, Prasd D wrote: >>> I think support for this should be added soon too ! >>> >>> On 7/17/12, Tamas Jalsovszky wrote: >>>> Hello, >>>> >>>> Is there any plan to support webrtc in FS? Right I've found this post about >>>> asterisk: >>>> http://blogs.digium.com/2012/05/23/asterisk-11-webrtc/ >>>> >>>> T. >>>> >>> >>> >>> -- >>> Thanks, >>> Prasd >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From msc at freeswitch.org Fri Jul 20 02:58:30 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 19 Jul 2012 15:58:30 -0700 Subject: [Freeswitch-users] Webrtc In-Reply-To: References: Message-ID: On Thu, Jul 19, 2012 at 3:47 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Oops, I was actually trying to sound serious and garner some > sponsorship for the effort ;) > > Also anyone who wants to see OPAL H.323 working, we're accepting > pledges to get Robert from OPAL to the upcoming ClueCon so we can > provide moral support. He's down under so it takes some hefty coin > 3k.... > > Maybe we should start a how you can help section on our homepage! > How about doing a kickstarter for each of these items? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120719/5691d030/attachment.html From marketing at cluecon.com Fri Jul 20 03:19:59 2012 From: marketing at cluecon.com (Michael Collins) Date: Thu, 19 Jul 2012 16:19:59 -0700 Subject: [Freeswitch-users] Update On Donations For H.323 Message-ID: We have received several donations already - thank you very much! Just a reminder: please be sure to leave a note with your donation so that we know what your donation is for. Our target for getting Robert Jongbloed to Chicago from Australia is US $3000. We will let you know when we've met our goal and we'll give you status updates along the way. Thanks again for your generosity! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120719/b8a12726/attachment.html From miha at softnet.si Fri Jul 20 10:36:10 2012 From: miha at softnet.si (Miha) Date: Fri, 20 Jul 2012 08:36:10 +0200 Subject: [Freeswitch-users] ENUM do not work In-Reply-To: References: Message-ID: <5008FC5A.7040505@softnet.si> On 7/19/2012 6:44 PM, Ken Rice wrote: > This has been fixed in tree as of this morning... Please update and try > again... > > There was a slight regression recently with a patch that allows you to > specify specific name servers enum to query against... > > Note: this didn't appear to affect the dialplan only the event socket API > call (the API call is what you get from the console) > > K > > > On 7/19/12 4:35 AM, "Miha" wrote: > >> On 7/18/2012 11:18 AM, Miha wrote: >>> Hi, >>> >>> it is revision 1.2 rc2 (installed from git two days ago). First I have >>> change server to which dns lookup is made, than I add default one 164.... >>> First I tought that is something wrong with configuration but than I >>> noticed with wireshark that FS does not send nslookup. >>> >>> In 1.06 work and also on 1.2 rc1. >>> >>> Regards, >>> Miha >>> >>> On 7/18/2012 10:57 AM, Jay Binks wrote: >>>> Divide and conquer , figure out which revision broke it and I'll help you. >>>> >>>> What do you see on the console and what enum config do you have . >>>> >>>> >>>> >>>> On 18/07/2012, at 5:32 PM, Miha wrote: >>>> >>>>> Hi, >>>>> >>>>> is there any problem with enum on new git? When I do nslookup in linux, >>>>> I get result but if I do it from fs_cli (enum 1231231231) I can see with >>>>> wireshark that FS do not send lookup. >>>>> >>>>> p.s.: I have same configuration on different older FS and works. >>>>> >>>>> Thanks! >>>>> miha >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> Join Us At ClueCon - Aug 7-9, 2012 >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> Revision of FS is: FreeSWITCH Version >> 1.2.0-rc2+git~20120719T040232Z~ec412c07d2+unclean~20120719T084055Z >> >> Regards,Miha >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > @HI Ken, thanks for all your great work! it works now. Guys you rock! MIha From miha at softnet.si Fri Jul 20 10:38:54 2012 From: miha at softnet.si (Miha) Date: Fri, 20 Jul 2012 08:38:54 +0200 Subject: [Freeswitch-users] Which FS version to use? In-Reply-To: <1342722228.4238.YahooMailNeo@web39305.mail.mud.yahoo.com> References: <1342722228.4238.YahooMailNeo@web39305.mail.mud.yahoo.com> Message-ID: <5008FCFE.5060607@softnet.si> On 7/19/2012 8:23 PM, Stanislav Sinyagin wrote: > Ken, > > a clean branching strategy and committing policy should actually > reduce the number of faults and reduce the maintenance effort. > > looking at the current commit log, you can easily see that about 20% > of commits are actually introducing new features or changing old ones. > So at least we should stop calling it a release candidate. > > > > ------------------------------------------------------------------------ > *From:* curriegrad2004 > *To:* FreeSWITCH Users Help > *Sent:* Thursday, July 19, 2012 7:23 PM > *Subject:* Re: [Freeswitch-users] Which FS version to use? > > In other words, we need people testing and reporting bugs on JIRA and > providing feedback on the fixes. > > We could always "assume" that the issue has been fixed from people who > never leave feedback on the fixes anyways. > > On Thu, Jul 19, 2012 at 10:17 AM, Ken Rice > wrote: > > A little history here... We used to do releases... See 1.0... > And the 1.0.x > > releases > > > > Why did we stop? > > > > Because no one would test (as developers we can only test so > far... Users > > fine new and exciting ways to break software all the time... > > > > There is probably 5 times the number of what I would call edge > use cases > > then there are what I would consider normal use cases... We cant > possibly > > test every possible permutation... So there ended up > frustrations and pains > > cause people were using old revs then complaing about bugs that > were fixed 2 > > months back or we had just released 1.0.2 and within 3 hours of > the release > > a mjor bug pops up... > > > > We are moving back to a release cycle with 1.2... No amount of > testing > > software that we have to run can address all the automatted > testing... We > > have to have users testing and reporting bugs and when we fix > bugs we need > > those fixes tested and reported back good/bad... > > > > Now more recently I came on board with the dev team to help > facilitate > > that... So guys help us help you... Test and report the bugs... > Test head... > > And report bugs... If you have a possible solution to a bug let > us know... > > We want 1.2 to get here and theres been a lot of changes since a > previous > > release.... > > > > K > > > > > > > > On 7/19/12 9:02 AM, "Brian Foster" > wrote: > > > > I think the point of releasing 1.2 is exactly because of the > reasons you are > > stating. > > > > FreeSWITCH gets many commits a day, and is continually improved > feature-wise > > and stability wise. That presents precisely the problems you are > pointing > > out. It's like art. Artwork is never really finished, you just > have to make > > a decision to stop. > > > > +1 for a stable branch of FreeSWITCH. But keep those juices flowing! > > > > Brian Foster > > Endigo Computer LLC > > > > Sent from a mobile device. > > > > On Jul 19, 2012 9:47 AM, "Ben Langfeld" > wrote: > > > > git-flow is a very nice branching strategy (master branch for > releases, > > develop branch for active development, feature branches for work in > > progress). Unfortunately, more or less all freeswitch > development happens on > > a single branch thanks to svn-think. > > > > Additionally, the "the latest release is HEAD" strategy is a > huge problem > > for those people writing software which depends on freeswitch. > FreeSWITCH > > does not follow any defined versioning pattern, and thus > determining wether > > updates will break a system is next to impossible. Thus, many > people run > > out-of-date freeswitch systems out of fear. > > > > If the FS team were to adopt semver (http://semver.org > ) then this issue > > would be all but resolved, but they appear to be dead set > against version > > numbers which actually have meaning. Unfortunately the core team > don't seem > > to realise how much of a problem this is for downstream > developers, and seem > > unwilling to do anything to help in that regard. A sad state of > affairs, and > > my only real criticism of FreeSWITCH. > > > > I hope 1.2 represents the beginning of a sensible and mature > approach to > > versioning for FreeSWITCH, and it can't be released soon enough > for that > > very reason. > > > > Regards, > > Ben Langfeld > > > > > > On 19 July 2012 15:37, Stanislav Sinyagin > wrote: > > > > > > by the way, any plans to fork a "stable" Git branch and use it > only for > > bugfixes, whereas "master" would absorb also all new features? > > > > What I see now in the commit log, is the "master" branch also > used for > > normal development, and not only for bugfixes. This way it's > difficult to > > expect the final 1.2 release any time soon. > > > > > > > > > > > > > > > > > > > > ________________________________ > > From: Peter Olsson > > > To: FreeSWITCH Users Help > > > > Sent: Thursday, July 19, 2012 2:47 PM > > Subject: Re: [Freeswitch-users] Which FS version to use? > > > > > > > > Current git head is always the recommended version. If you find any > > problems, please report them to Jira. Git head is currently at > 1.2 RC2 state > > (waiting for a stable 1.2 release), so the more things we get > fixed, the > > better! > > > > Please retry the ENUM problems on latest git again, and if it > still fails, > > report this to Jira. > > > > /Peter > > > > 19 jul 2012 kl. 12:51 skrev "Miha" >: > > > >> Hi, > >> > >> on my production server I have version: > >> > >> freeswitch at default> version > >> FreeSWITCH Version 1.0.head (git-00de8e6 2011-11-01 17-27-13 -0600) > >> > >> which is in my case very stable, I can say perfect. > >> > >> As I have prepare other, more powerfull server for FS I have > installed > >> FS from latest git. With version 1.2 rc1 I was having problems > with load > >> and all users did not regisered (loosing connection). I do no > know if > >> this was FS problem but I do not have this problems on production > >> server. Few day ago I did a new git pull. Now I have version > 1.2 rc2 > >> which enum do not work. I traced with wireshark but afer I put > in fs_cli > >> enum 123123123, Fs sends nothing. > >> > >> > >> So on git.freeswitch.com > are versions 1.0.6, > >> 1.2 rc2 and 1.2 rc1. Which > >> version to use? Is 1.0.6 most stable for now as 1.2 is rc? > >> > >> Where can I find version of FS which I am running on production > server? > >> > >> Thanks for all your help! > >> > >> miha > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > >> > >> !DSPAM:5007e36132765726119943! > >> > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > ________________________________ > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org HI, thank you for all you answers. Ken did I fix for enum and now it works:) I did not set nameserver in enum_conf as if was not option in previous version. I will try to use lates git and we will see what will happen when server will be in production. Now everything works in testing environment. Regards, Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120720/39459402/attachment-0001.html From rajkumar.kanniappan at sasken.com Fri Jul 20 09:34:34 2012 From: rajkumar.kanniappan at sasken.com (Rajkumar Kanniappan) Date: Fri, 20 Jul 2012 11:04:34 +0530 Subject: [Freeswitch-users] Making a call from PSTN to SoftPhone Message-ID: <6F91E0FFDA542149961F7BDED2D2B94B568E46D0C5@EXGMBX01.sasken.com> Hi, if I make a call from PSTN extension to a IP phone, call has to reach from Extension -> PRI card(VoIP gateway) -> Drivers -> FreeTDM, and from here what is going to be the job of FreeTDM. Is FreeTDM going to contact Sofia module, if so what is going to be the configuration changes for freetdm and sofia. Will FreeTDM module select any sofia profile(internal or external) to make calls. And how sofia profile is getting selected for this particular call. Please clarify this to me... Thanks, Rajkumar ________________________________ SASKEN BUSINESS DISCLAIMER: This message may contain confidential, proprietary or legally privileged information. In case you are not the original intended Recipient of the message, you must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message and you are requested to delete it and inform the sender. Any views expressed in this message are those of the individual sender unless otherwise stated. Nothing contained in this message shall be construed as an offer or acceptance of any offer by Sasken Communication Technologies Limited ("Sasken") unless sent with that express intent and with due authority of Sasken. Sasken has taken enough precautions to prevent the spread of viruses. However the company accepts no liability for any damage caused by any virus transmitted by this email. Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120720/ce18c001/attachment.html From curriegrad2004 at gmail.com Fri Jul 20 11:31:06 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 20 Jul 2012 00:31:06 -0700 Subject: [Freeswitch-users] Making a call from PSTN to SoftPhone In-Reply-To: <6F91E0FFDA542149961F7BDED2D2B94B568E46D0C5@EXGMBX01.sasken.com> References: <6F91E0FFDA542149961F7BDED2D2B94B568E46D0C5@EXGMBX01.sasken.com> Message-ID: FreeTDM in FreeSWITCH will not even touch sofia at all. All it is going to do is bridge your particular timeslot from your PRI into an endpoint of your choosing. FreeSWITCH will then do the rest of the dirty job of selecting which sofia profile your IP set is registered at. On Thu, Jul 19, 2012 at 10:34 PM, Rajkumar Kanniappan wrote: > Hi, > > > > if I make a call from PSTN extension to a IP phone, > > call has to reach from Extension -> PRI card(VoIP gateway) -> Drivers -> > FreeTDM, and from here what is going to be the job of FreeTDM. Is FreeTDM > going to contact Sofia module, if so what is going to be the configuration > changes for freetdm and sofia. > Will FreeTDM module select any sofia profile(internal or external) to make > calls. > > And how sofia profile is getting selected for this particular call. > > > Please clarify this to me... > > > > Thanks, > > Rajkumar > > > > > ________________________________ > SASKEN BUSINESS DISCLAIMER: This message may contain confidential, > proprietary or legally privileged information. In case you are not the > original intended Recipient of the message, you must not, directly or > indirectly, use, disclose, distribute, print, or copy any part of this > message and you are requested to delete it and inform the sender. Any views > expressed in this message are those of the individual sender unless > otherwise stated. Nothing contained in this message shall be construed as an > offer or acceptance of any offer by Sasken Communication Technologies > Limited ("Sasken") unless sent with that express intent and with due > authority of Sasken. Sasken has taken enough precautions to prevent the > spread of viruses. However the company accepts no liability for any damage > caused by any virus transmitted by this email. > Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From alex at thewinelake.com Fri Jul 20 12:40:55 2012 From: alex at thewinelake.com (Alex) Date: Fri, 20 Jul 2012 09:40:55 +0100 Subject: [Freeswitch-users] Getting started with db from php In-Reply-To: References: <5007E4D8.7050108@softnet.si> <5B0A12B0-7D51-4A8C-A9EE-9C5A2200967B@visionutveckling.se> <1342705039.82548.YahooMailNeo@web39304.mail.mud.yahoo.com> Message-ID: <50091997.8080902@thewinelake.com> Trying to do some stuff with the voicemail database from a php script. Couldn't quickly find the connection runes to the sqlite DB. Tried the following, but got the error "file is encrypted or is not a database" "; echo phpversion(); $dbhandle = sqlite_open('/usr/local/freeswitch/db/voicemail_default.db', 0666, $error); if (!$dbhandle) die ($error); $sql = "SELECT * from voicemail_prefs"; $result = sqlite_exec($dbhandle, $stm, $error); while ($row = sqlite_fetch_array($result, SQLITE_ASSOC)) { print_r($row); } ?> From avi at avimarcus.net Fri Jul 20 12:44:37 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 20 Jul 2012 11:44:37 +0300 Subject: [Freeswitch-users] Getting started with db from php In-Reply-To: <50091997.8080902@thewinelake.com> References: <5007E4D8.7050108@softnet.si> <5B0A12B0-7D51-4A8C-A9EE-9C5A2200967B@visionutveckling.se> <1342705039.82548.YahooMailNeo@web39304.mail.mud.yahoo.com> <50091997.8080902@thewinelake.com> Message-ID: Try with SQLite3 -Avi On Fri, Jul 20, 2012 at 11:40 AM, Alex wrote: > Trying to do some stuff with the voicemail database from a php script. > Couldn't quickly find the connection runes to the sqlite DB. > > Tried the following, but got the error "file is encrypted or is not a > database" > > echo sqlite_libversion(); > echo "
"; > > echo phpversion(); > $dbhandle = sqlite_open('/usr/local/freeswitch/db/voicemail_default.db', > 0666, $error); > if (!$dbhandle) die ($error); > > $sql = "SELECT * from voicemail_prefs"; > > $result = sqlite_exec($dbhandle, $stm, $error); > while ($row = sqlite_fetch_array($result, SQLITE_ASSOC)) { > print_r($row); > } > > ?> > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120720/fed556d7/attachment.html From wstephen80 at gmail.com Fri Jul 20 16:48:28 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Fri, 20 Jul 2012 14:48:28 +0200 Subject: [Freeswitch-users] ACK to 200 OK in a Bridge Message-ID: I have the following issue: if I do a bridge of an inbound call to an outbound call, when the called number answer the call with a 200 OK, FreeSwitch sends the ACK instantely, without wait the ACK of the calling. The message flow (as shown in the attached .png) is: CALLING FS CALLED INVITE ---------> 100 Trying <--------- -------------> INVITE <--------- 100 Trying <--------- 200 OK -------------> ACK <--------- 200 OK -------------> ACK When I expected that the ACK to called is related to ACK from the calling: CALLING FS CALLED INVITE ---------> 100 Trying <--------- -------------> INVITE <--------- 100 Trying <--------- 200 OK <--------- 200 OK -------------> ACK -------------> ACK For this reason, if sometimes the Calling doesn't send the ACK (because it reject the 200 OK), the call is considered connected (and billed) by Freeswitch (30s, probably due to a timeout waiting ack) and not by Calling (0s). There is a way (config?) to send ACK to called only when is received the ACK from calling? Stephen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120720/499ebb4a/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: trace.png Type: image/png Size: 17619 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120720/499ebb4a/attachment-0001.png From bdfoster at endigotech.com Fri Jul 20 16:57:09 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 20 Jul 2012 08:57:09 -0400 Subject: [Freeswitch-users] ACK to 200 OK in a Bridge In-Reply-To: References: Message-ID: What is the billsec of each leg of the call? Is it what you expect to see? Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 20, 2012 8:49 AM, "Stephen Wilde" wrote: > I have the following issue: if I do a bridge of an inbound call to an > outbound call, when the called number answer the call with a 200 OK, > FreeSwitch sends the ACK instantely, without wait the ACK of the calling. > > The message flow (as shown in the attached .png) is: > > CALLING FS CALLED > > INVITE ---------> > 100 Trying <--------- > -------------> INVITE > <--------- 100 Trying > <--------- 200 OK > -------------> ACK > <--------- 200 OK > -------------> ACK > > > > When I expected that the ACK to called is related to ACK from the calling: > > CALLING FS CALLED > > INVITE ---------> > 100 Trying <--------- > -------------> INVITE > <--------- 100 Trying > <--------- 200 OK > <--------- 200 OK > -------------> ACK > -------------> ACK > > > > For this reason, if sometimes the Calling doesn't send the ACK (because it > reject the 200 OK), the call is considered connected (and billed) by > Freeswitch (30s, probably due to a timeout waiting ack) and not by Calling > (0s). > There is a way (config?) to send ACK to called only when is received the > ACK from calling? > > Stephen > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120720/7102e4a6/attachment.html From bdfoster at endigotech.com Fri Jul 20 16:59:55 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 20 Jul 2012 08:59:55 -0400 Subject: [Freeswitch-users] ACK to 200 OK in a Bridge In-Reply-To: References: Message-ID: Sorry, I think I understand a little better now. Please post a siptrace of both legs of the entire call on pastebin.freeswitch.org and link back here. Sanitize of you like, but dont make it too ambiguous. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 20, 2012 8:49 AM, "Stephen Wilde" wrote: > I have the following issue: if I do a bridge of an inbound call to an > outbound call, when the called number answer the call with a 200 OK, > FreeSwitch sends the ACK instantely, without wait the ACK of the calling. > > The message flow (as shown in the attached .png) is: > > CALLING FS CALLED > > INVITE ---------> > 100 Trying <--------- > -------------> INVITE > <--------- 100 Trying > <--------- 200 OK > -------------> ACK > <--------- 200 OK > -------------> ACK > > > > When I expected that the ACK to called is related to ACK from the calling: > > CALLING FS CALLED > > INVITE ---------> > 100 Trying <--------- > -------------> INVITE > <--------- 100 Trying > <--------- 200 OK > <--------- 200 OK > -------------> ACK > -------------> ACK > > > > For this reason, if sometimes the Calling doesn't send the ACK (because it > reject the 200 OK), the call is considered connected (and billed) by > Freeswitch (30s, probably due to a timeout waiting ack) and not by Calling > (0s). > There is a way (config?) to send ACK to called only when is received the > ACK from calling? > > Stephen > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120720/eaee31b9/attachment.html From rajkumar.kanniappan at sasken.com Fri Jul 20 12:43:10 2012 From: rajkumar.kanniappan at sasken.com (Rajkumar Kanniappan) Date: Fri, 20 Jul 2012 14:13:10 +0530 Subject: [Freeswitch-users] Making a call from PSTN to SoftPhone In-Reply-To: References: <6F91E0FFDA542149961F7BDED2D2B94B568E46D0C5@EXGMBX01.sasken.com> Message-ID: <6F91E0FFDA542149961F7BDED2D2B94B568E46D12A@EXGMBX01.sasken.com> Isnt it necessary that bridging of a call has to take place only after the selection of sofia profile. Otherwise when is it going to select the sofia profile... Can you please explain in detail about the sofia profile selection in this scenario Thanks Rajkumar -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004 Sent: Friday, July 20, 2012 1:01 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Making a call from PSTN to SoftPhone FreeTDM in FreeSWITCH will not even touch sofia at all. All it is going to do is bridge your particular timeslot from your PRI into an endpoint of your choosing. FreeSWITCH will then do the rest of the dirty job of selecting which sofia profile your IP set is registered at. On Thu, Jul 19, 2012 at 10:34 PM, Rajkumar Kanniappan wrote: > Hi, > > > > if I make a call from PSTN extension to a IP phone, > > call has to reach from Extension -> PRI card(VoIP gateway) -> Drivers > -> FreeTDM, and from here what is going to be the job of FreeTDM. Is > FreeTDM going to contact Sofia module, if so what is going to be the > configuration changes for freetdm and sofia. > Will FreeTDM module select any sofia profile(internal or external) to > make calls. > > And how sofia profile is getting selected for this particular call. > > > Please clarify this to me... > > > > Thanks, > > Rajkumar > > > > > ________________________________ > SASKEN BUSINESS DISCLAIMER: This message may contain confidential, > proprietary or legally privileged information. In case you are not the > original intended Recipient of the message, you must not, directly or > indirectly, use, disclose, distribute, print, or copy any part of this > message and you are requested to delete it and inform the sender. Any > views expressed in this message are those of the individual sender > unless otherwise stated. Nothing contained in this message shall be > construed as an offer or acceptance of any offer by Sasken > Communication Technologies Limited ("Sasken") unless sent with that > express intent and with due authority of Sasken. Sasken has taken > enough precautions to prevent the spread of viruses. However the > company accepts no liability for any damage caused by any virus transmitted by this email. > Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users 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 > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From wstephen80 at gmail.com Fri Jul 20 20:15:32 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Fri, 20 Jul 2012 18:15:32 +0200 Subject: [Freeswitch-users] ACK to 200 OK in a Bridge In-Reply-To: References: Message-ID: This is not a particulare case: in all calls bridged by FS the ACK to the 200OK to the outbound leg is sent not related to the ACK received from inbound leg. I expect a flow as: http://www.packetizer.com/ipmc/sip/papers/understanding_sip_voip/sip_call_flow.png These are my traces: LegA: http://pastebin.freeswitch.org/19565 LegB: http://pastebin.freeswitch.org/19566 On Fri, Jul 20, 2012 at 2:59 PM, Brian Foster wrote: > Sorry, I think I understand a little better now. > > Please post a siptrace of both legs of the entire call on > pastebin.freeswitch.org and link back here. Sanitize of you like, but > dont make it too ambiguous. > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > On Jul 20, 2012 8:49 AM, "Stephen Wilde" wrote: > >> I have the following issue: if I do a bridge of an inbound call to an >> outbound call, when the called number answer the call with a 200 OK, >> FreeSwitch sends the ACK instantely, without wait the ACK of the calling. >> >> The message flow (as shown in the attached .png) is: >> >> CALLING FS CALLED >> >> INVITE ---------> >> 100 Trying <--------- >> -------------> INVITE >> <--------- 100 Trying >> <--------- 200 OK >> -------------> ACK >> <--------- 200 OK >> -------------> ACK >> >> >> >> When I expected that the ACK to called is related to ACK from the calling: >> >> CALLING FS CALLED >> >> INVITE ---------> >> 100 Trying <--------- >> -------------> INVITE >> <--------- 100 Trying >> <--------- 200 OK >> <--------- 200 OK >> -------------> ACK >> -------------> ACK >> >> >> >> For this reason, if sometimes the Calling doesn't send the ACK (because >> it reject the 200 OK), the call is considered connected (and billed) by >> Freeswitch (30s, probably due to a timeout waiting ack) and not by Calling >> (0s). >> There is a way (config?) to send ACK to called only when is received the >> ACK from calling? >> >> Stephen >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120720/ab0b5209/attachment-0001.html From mgg at giagnocavo.net Fri Jul 20 20:45:21 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Fri, 20 Jul 2012 16:45:21 +0000 Subject: [Freeswitch-users] Which FS version to use? In-Reply-To: <5008FCFE.5060607@softnet.si> References: <1342722228.4238.YahooMailNeo@web39305.mail.mud.yahoo.com> <5008FCFE.5060607@softnet.si> Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B612F7FEC0@BLUPRD0711MB413.namprd07.prod.outlook.com> I'd second that. If the project really wants to call a stable release, why on earth would any new functionality be allowed in? If there's not even a branch, then we're not even really pretending to go for a stable release... That said, I've not really had much issues, so the process of "grab head, test until my use case works, never upgrade again" works for a lot of people. But, it'd work a whole lot better if there was actually a release branch, because then I would replace the "never upgrade again" part with "keep doing minor updates" like we do with Asterisk, OpenSIPS, and pretty much any other piece of software in the world. What would be nice is to know about security patches... a million lines of C and no security bugs? If that's true, well, the Queen Elizabeth Prize for Engineering closes nominations in two months. But hey, it works well enough for me, and, I guess, a lot of other folks, too, so I'm not really complaining. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Miha Sent: Friday, July 20, 2012 12:39 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Which FS version to use? Ken, a clean branching strategy and committing policy should actually reduce the number of faults and reduce the maintenance effort. looking at the current commit log, you can easily see that about 20% of commits are actually introducing new features or changing old ones. So at least we should stop calling it a release candidate. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120720/7aba0f46/attachment.html From anthony.minessale at gmail.com Fri Jul 20 21:11:27 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 20 Jul 2012 12:11:27 -0500 Subject: [Freeswitch-users] a=crypto in RTP/AVP, refer to rfc3711 In-Reply-To: References: Message-ID: This is because its a violation to have crypto in an AVP vs SAVP If you get latest HEAD I added a profile param / channel var to allow this. profile param: NDLB-allow-crypto-in-avp var: sip_allow_crypto_in_avp On Tue, Jul 17, 2012 at 4:14 AM, Anita Hall wrote: > Hi > > I am getting an error a=crypto in RTP/AVP, refer to rfc3711 when using > sipml5 client with FS. > > There is some info on this thread, which I am currently experimenting with. > https://groups.google.com/forum/#!msg/doubango/m7FCqYxRcQc/-ThdAwWJxP8J > > Is there any change in setting on FS side that can deal with this ? > > Complete SIP Log > > v=0 > o=- 2834273280 1 IN IP4 127.0.0.1 > s=webrtc (chrome 20.0.1127.0) - Doubango Telecom (sipML5 r000) > t=0 0 > a=group:BUNDLE audio video > m=audio 51704 RTP/SAVPF 103 104 0 8 106 105 13 126 > c=IN IP4 182.68.141.91 > a=rtcp:51704 IN IP4 182.68.141.91 > a=candidate:829852397 1 udp 2130714367 192.168.1.31 51703 typ host > generation 0 > a=candidate:829852397 2 udp 2130714367 192.168.1.31 51703 typ host > generation 0 > a=candidate:3345412921 1 udp 1912610559 182.68.141.91 51704 typ srflx > generation 0 > a=candidate:3345412921 2 udp 1912610559 182.68.141.91 51704 typ srflx > generation 0 > a=candidate:2146792989 1 tcp 1694506751 192.168.1.31 50295 typ host > generation 0 > a=candidate:2146792989 2 tcp 1694506751 192.168.1.31 50295 typ host > generation 0 > a=ice-ufrag:Jz1C79lLNJodMTz2 > a=ice-pwd:UsowydSx3Uy9ehq8lKyR+JmJ > a=mid:audio > a=rtcp-mux > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:ccVX0xSFpsuRG8lnNQgnSiMdckCgtHs5SET8r9Jt > a=rtpmap:103 ISAC/16000 > a=rtpmap:104 ISAC/32000 > a=rtpmap:0 PCMU/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:106 CN/32000 > a=rtpmap:105 CN/16000 > a=rtpmap:13 CN/8000 > a=rtpmap:126 telephone-event/8000 > a=ssrc:862982555 cname:QP1jAi3c1C/sIjMM > a=ssrc:862982555 mslabel:JTCmb2lovY15B6rH1xoGTpSOwige5Py22GgE > a=ssrc:862982555 label:JTCmb2lovY15B6rH1xoGTpSOwige5Py22GgE00 > m=video 51704 RTP/SAVPF 100 101 102 > c=IN IP4 182.68.141.91 > a=rtcp:51704 IN IP4 182.68.141.91 > a=candidate:829852397 1 udp 2130714367 192.168.1.31 51703 typ host > generation 0 > a=candidate:829852397 2 udp 2130714367 192.168.1.31 51703 typ host > generation 0 > a=candidate:3345412921 1 udp 1912610559 182.68.141.91 51704 typ srflx > generation 0 > a=candidate:3345412921 2 udp 1912610559 182.68.141.91 51704 typ srflx > generation 0 > a=candidate:2146792989 1 tcp 1694506751 192.168.1.31 50295 typ host > generation 0 > a=candidate:2146792989 2 tcp 1694506751 192.168.1.31 50295 typ host > generation 0 > a=ice-ufrag:Jz1C79lLNJodMTz2 > a=ice-pwd:UsowydSx3Uy9ehq8lKyR+JmJ > a=mid:video > a=rtcp-mux > a=crypto:1 AES_CM_128_HMAC_SHA1_80 > inline:ccVX0xSFpsuRG8lnNQgnSiMdckCgtHs5SET8r9Jt > a=rtpmap:100 VP8/90000 > a=rtpmap:101 red/90000 > a=rtpmap:102 ulpfec/90000 > a=ssrc:1143670773 cname:QP1jAi3c1C/sIjMM > a=ssrc:1143670773 mslabel:JTCmb2lovY15B6rH1xoGTpSOwige5Py22GgE > a=ssrc:1143670773 label:JTCmb2lovY15B6rH1xoGTpSOwige5Py22GgE10 > > 2012-07-17 14:48:02.285368 [ERR] sofia_glue.c:4673 a=crypto in RTP/AVP, > refer to rfc3711 > 2012-07-17 14:48:02.285368 [DEBUG] switch_channel.c:2848 > (sofia/internal/1015 at 122.180.97.198) Callstate Change DOWN -> HANGUP > 2012-07-17 14:48:02.285368 [NOTICE] sofia.c:5813 Hangup > sofia/internal/1015 at 122.180.97.198 [CS_NEW] [INCOMPATIBLE_DESTINATION] > > > regards, > Anita > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From curriegrad2004 at gmail.com Fri Jul 20 21:11:27 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 20 Jul 2012 10:11:27 -0700 Subject: [Freeswitch-users] Making a call from PSTN to SoftPhone In-Reply-To: <6F91E0FFDA542149961F7BDED2D2B94B568E46D12A@EXGMBX01.sasken.com> References: <6F91E0FFDA542149961F7BDED2D2B94B568E46D0C5@EXGMBX01.sasken.com> <6F91E0FFDA542149961F7BDED2D2B94B568E46D12A@EXGMBX01.sasken.com> Message-ID: Not exactly. Sofia is just an endpoint, selection of the profile is done automatically when you bridge the call under the user/XXX at XXX endpoint. On Fri, Jul 20, 2012 at 1:43 AM, Rajkumar Kanniappan wrote: > Isnt it necessary that bridging of a call has to take place only after the selection of sofia profile. > > Otherwise when is it going to select the sofia profile... Can you please explain in detail about the sofia profile selection in this scenario > > Thanks > Rajkumar > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004 > Sent: Friday, July 20, 2012 1:01 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Making a call from PSTN to SoftPhone > > FreeTDM in FreeSWITCH will not even touch sofia at all. All it is going to do is bridge your particular timeslot from your PRI into an endpoint of your choosing. FreeSWITCH will then do the rest of the dirty job of selecting which sofia profile your IP set is registered at. > > On Thu, Jul 19, 2012 at 10:34 PM, Rajkumar Kanniappan wrote: >> Hi, >> >> >> >> if I make a call from PSTN extension to a IP phone, >> >> call has to reach from Extension -> PRI card(VoIP gateway) -> Drivers >> -> FreeTDM, and from here what is going to be the job of FreeTDM. Is >> FreeTDM going to contact Sofia module, if so what is going to be the >> configuration changes for freetdm and sofia. >> Will FreeTDM module select any sofia profile(internal or external) to >> make calls. >> >> And how sofia profile is getting selected for this particular call. >> >> >> Please clarify this to me... >> >> >> >> Thanks, >> >> Rajkumar >> >> >> >> >> ________________________________ >> SASKEN BUSINESS DISCLAIMER: This message may contain confidential, >> proprietary or legally privileged information. In case you are not the >> original intended Recipient of the message, you must not, directly or >> indirectly, use, disclose, distribute, print, or copy any part of this >> message and you are requested to delete it and inform the sender. Any >> views expressed in this message are those of the individual sender >> unless otherwise stated. Nothing contained in this message shall be >> construed as an offer or acceptance of any offer by Sasken >> Communication Technologies Limited ("Sasken") unless sent with that >> express intent and with due authority of Sasken. Sasken has taken >> enough precautions to prevent the spread of viruses. However the >> company accepts no liability for any damage caused by any virus transmitted by this email. >> Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users 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 >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From miha at softnet.si Fri Jul 20 21:22:30 2012 From: miha at softnet.si (Miha) Date: Fri, 20 Jul 2012 19:22:30 +0200 Subject: [Freeswitch-users] xml cdr with rc2 Message-ID: HI, just one questione before I post to Jira. Is there any cdr problem with rc2. With the same dialplan that I have on FS verison 1.0.xxx from git (befor 1.2 appeared) I get on RC2 different values when 302 happens. Olso noticed that cdr for b leg are different (like on production FS with same dialplan). Is that normal or has anyone also noticed? THanks! Miha From msc at freeswitch.org Fri Jul 20 21:25:30 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 20 Jul 2012 10:25:30 -0700 Subject: [Freeswitch-users] xml cdr with rc2 In-Reply-To: References: Message-ID: Do you have any samples to share? Drop them on PB. -MC On Fri, Jul 20, 2012 at 10:22 AM, Miha wrote: > > HI, > > just one questione before I post to Jira. > > Is there any cdr problem with rc2. With the same dialplan > that I have on FS verison 1.0.xxx from git (befor 1.2 > appeared) I get on RC2 different values when 302 happens. > Olso noticed that cdr for b leg are different (like on > production FS with same dialplan). > > Is that normal or has anyone also noticed? > > THanks! > > Miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120720/cb35e73c/attachment-0001.html From mi.ke at null.net Fri Jul 20 21:30:20 2012 From: mi.ke at null.net (Mi Ke) Date: Fri, 20 Jul 2012 13:30:20 -0400 Subject: [Freeswitch-users] hunting in failover mode Message-ID: <20120720173020.154600@gmx.com> Hi All, When I do a failover using bridge app and ep1|ep2|ep3 construstion - how do I stop further hunting if any of already hunted endpoints returned e.g. USER_BUSY ? Thanks / Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120720/3ec14110/attachment.html From msc at freeswitch.org Fri Jul 20 21:39:04 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 20 Jul 2012 10:39:04 -0700 Subject: [Freeswitch-users] hunting in failover mode In-Reply-To: <20120720173020.154600@gmx.com> References: <20120720173020.154600@gmx.com> Message-ID: Perhaps you could use this? http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail Specify the conditions on which the bridge should keep going. Note: you may need to separate these out into individual bridge apps with a single endpoint. I haven't tried it lately, so be sure to report back what you find out. -MC On Fri, Jul 20, 2012 at 10:30 AM, Mi Ke wrote: > Hi All, > > When I do a failover using bridge app and ep1|ep2|ep3 construstion - how > do I stop further hunting if any of already hunted endpoints returned e.g. > USER_BUSY ? > > Thanks / Mike > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120720/7180971f/attachment.html From toddb at toddbailey.net Fri Jul 20 22:00:36 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Fri, 20 Jul 2012 11:00:36 -0700 Subject: [Freeswitch-users] Passing Caller id from ata to phone Message-ID: <1342807236.5783.65.camel@mythtv> Hi all, Using a spa3102 ata, Frees switch and a Polycom ip501, I'm not getting caller id to display on the phone. Is there some 'magical' setting that I need to address, somewhere ? Note: when an internal softphone extension calls the ip501, I see the caller id. This implies the phone and FS are working. From mi.ke at null.net Fri Jul 20 22:03:01 2012 From: mi.ke at null.net (Mi Ke) Date: Fri, 20 Jul 2012 14:03:01 -0400 Subject: [Freeswitch-users] hunting in failover mode Message-ID: <20120720180301.154600@gmx.com> Michael, thanks for answering - I already tried this but I need to set continue_on_fail to TRUE because that seems to be the only way to pass original hangup code from leg B to A when bridge completes. And when COF is set to TRUE it behaves like it's related to the "whole" bridge i.e. until all 3 endpoints are tried entire bridge is not considered as failed of succeded. Re separate approach - I'm not sure how can I do that as the number of EPs in my case is dynamic, bridge app receives them as a 1|2|3 dialstring from rouing lua app. Thanks / Mike ----- Original Message ----- From: Michael Collins Sent: 07/20/12 08:39 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] hunting in failover mode Perhaps you could use this? http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail Specify the conditions on which the bridge should keep going. Note: you may need to separate these out into individual bridge apps with a single endpoint. I haven't tried it lately, so be sure to report back what you find out. -MC On Fri, Jul 20, 2012 at 10:30 AM, Mi Ke < mi.ke at null.net > wrote: Hi All, When I do a failover using bridge app and ep1|ep2|ep3 construstion - how do I stop further hunting if any of already hunted endpoints returned e.g. USER_BUSY ? Thanks / Mike _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120720/3fc84d6d/attachment.html From bdfoster at endigotech.com Fri Jul 20 22:12:58 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 20 Jul 2012 14:12:58 -0400 Subject: [Freeswitch-users] Passing Caller id from ata to phone In-Reply-To: <1342807236.5783.65.camel@mythtv> References: <1342807236.5783.65.camel@mythtv> Message-ID: On FXS or FXO? Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 20, 2012 2:02 PM, "Todd Bailey" wrote: > Hi all, > > Using a spa3102 ata, Frees switch and a Polycom ip501, I'm not getting > caller id to display on the phone. > > Is there some 'magical' setting that I need to address, somewhere ? > > Note: when an internal softphone extension calls the ip501, I see the > caller id. This implies the phone and FS are working. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120720/b3664d3e/attachment.html From krice at freeswitch.org Fri Jul 20 22:21:57 2012 From: krice at freeswitch.org (Ken Rice) Date: Fri, 20 Jul 2012 13:21:57 -0500 Subject: [Freeswitch-users] ACK to 200 OK in a Bridge In-Reply-To: Message-ID: This shouldn?t be a problem... The call is answered as soon as FS gets the 200 OK from the termination leg... Also keep in mind that FreeSWITCH has sofia do a few things automattically like the 100 trying is done automagic by the sofia lib... Also FreeSwitch is a B2BUA not a Proxy so the the A Leg can operate independently of the B leg... On 7/20/12 7:48 AM, "Stephen Wilde" wrote: > I have the following issue: if I do a bridge of an inbound call to an outbound > call, when the called number answer the call with a 200 OK, FreeSwitch sends > the ACK instantely, without wait the ACK of the calling. > > The message flow (as shown in the attached .png) is: > > CALLING ? ? ? ? ? ? ?FS ? ? ? CALLED > > INVITE ? ? ? ---------> > 100 Trying <--------- > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?-------------> INVITE > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<---------?100 Trying? > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<---------?200 OK > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?-------------> ACK > <---------?200 OK > -------------> ACK > > > > When I expected that the ACK to called is related to ACK from the calling: > > CALLING ? ? ? ? ? ? ?FS ? ? ? CALLED > > INVITE ? ? ? ---------> > 100 Trying <--------- > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?-------------> INVITE > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<---------?100 Trying? > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?<---------?200 OK > <---------?200 OK > -------------> ACK > ? ? ? ? ? ? ? ? ? ? ? ? ? ? ?-------------> ACK > > > > For this reason, if sometimes the Calling doesn't send the ACK (because it > reject the 200 OK), the call is considered connected (and billed) by > Freeswitch (30s, probably due to a timeout waiting ack) and not by Calling > (0s). > There is a way (config?) to send ACK to called only when is received the ACK > from calling? > > Stephen > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120720/8f94c5c1/attachment-0001.html From toddb at toddbailey.net Fri Jul 20 22:22:03 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Fri, 20 Jul 2012 11:22:03 -0700 Subject: [Freeswitch-users] Passing Caller id from ata to phone In-Reply-To: <1342807236.5783.65.camel@mythtv> References: <1342807236.5783.65.camel@mythtv> Message-ID: <1342808523.5783.68.camel@mythtv> Never mind I fixed it.. On Fri, 2012-07-20 at 11:00 -0700, Todd Bailey wrote: > Hi all, > > Using a spa3102 ata, Frees switch and a Polycom ip501, I'm not getting > caller id to display on the phone. > > Is there some 'magical' setting that I need to address, somewhere ? > > Note: when an internal softphone extension calls the ip501, I see the > caller id. This implies the phone and FS are working. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From toddb at toddbailey.net Fri Jul 20 22:39:36 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Fri, 20 Jul 2012 11:39:36 -0700 Subject: [Freeswitch-users] Skype use with mod_skypopen Message-ID: <1342809576.5783.77.camel@mythtv> 2 Easy question for Skype users, 1: Do I need a special or business account to use with mod_skypopen or will any account type work? Reason: I'd like to create a dial plan that when I dial 9+ number etc it will connect to skype account, for uses for long distance calls. I want to be able to dial 1+ number for ld through pstn carrier. 2: Do I need to install any special skype applications or does mod_skypopen handle all aspects of Fs to Skype connections? reason: Skype applications and support for linix is basically nill. From bdfoster at endigotech.com Fri Jul 20 22:46:22 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 20 Jul 2012 14:46:22 -0400 Subject: [Freeswitch-users] Skype use with mod_skypopen In-Reply-To: <1342809576.5783.77.camel@mythtv> References: <1342809576.5783.77.camel@mythtv> Message-ID: 1. No, any acct will do. 2. Follow the instructions for mod_skypopen on the silicon and you'll end up installing everything needed. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 20, 2012 2:40 PM, "Todd Bailey" wrote: > 2 Easy question for Skype users, > > 1: Do I need a special or business account to use with mod_skypopen or > will any account type work? > > Reason: I'd like to create a dial plan that when I dial 9+ number etc > it will connect to skype account, for uses for long distance calls. > I want to be able to dial 1+ number for ld through pstn carrier. > > > 2: Do I need to install any special skype applications or does > mod_skypopen handle all aspects of Fs to Skype connections? > > reason: Skype applications and support for linix is basically nill. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120720/83cf448f/attachment.html From bdfoster at endigotech.com Fri Jul 20 22:47:24 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Fri, 20 Jul 2012 14:47:24 -0400 Subject: [Freeswitch-users] Passing Caller id from ata to phone In-Reply-To: <1342808523.5783.68.camel@mythtv> References: <1342807236.5783.65.camel@mythtv> <1342808523.5783.68.camel@mythtv> Message-ID: Care to share the problem/solution for google's sake? Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 20, 2012 2:22 PM, "Todd Bailey" wrote: > > Never mind I fixed it.. > > On Fri, 2012-07-20 at 11:00 -0700, Todd Bailey wrote: > > Hi all, > > > > Using a spa3102 ata, Frees switch and a Polycom ip501, I'm not getting > > caller id to display on the phone. > > > > Is there some 'magical' setting that I need to address, somewhere ? > > > > Note: when an internal softphone extension calls the ip501, I see the > > caller id. This implies the phone and FS are working. > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120720/607ca73c/attachment.html From gmaruzz at gmail.com Fri Jul 20 22:54:33 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Fri, 20 Jul 2012 20:54:33 +0200 Subject: [Freeswitch-users] Skype use with mod_skypopen In-Reply-To: References: <1342809576.5783.77.camel@mythtv> Message-ID: For all details: http://wiki.freeswitch.org/wiki/Skypopen On Fri, Jul 20, 2012 at 8:46 PM, Brian Foster wrote: > 1. No, any acct will do. > > 2. Follow the instructions for mod_skypopen on the silicon and you'll end > up installing everything needed. > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > On Jul 20, 2012 2:40 PM, "Todd Bailey" wrote: > >> 2 Easy question for Skype users, >> >> 1: Do I need a special or business account to use with mod_skypopen or >> will any account type work? >> >> Reason: I'd like to create a dial plan that when I dial 9+ number etc >> it will connect to skype account, for uses for long distance calls. >> I want to be able to dial 1+ number for ld through pstn carrier. >> >> >> 2: Do I need to install any special skype applications or does >> mod_skypopen handle all aspects of Fs to Skype connections? >> >> reason: Skype applications and support for linix is basically nill. >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120720/481aeb63/attachment.html From toddb at toddbailey.net Fri Jul 20 23:47:10 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Fri, 20 Jul 2012 12:47:10 -0700 Subject: [Freeswitch-users] Passing Caller id from ata to phone In-Reply-To: References: <1342807236.5783.65.camel@mythtv> <1342808523.5783.68.camel@mythtv> Message-ID: <1342813630.5783.82.camel@mythtv> I'd be happy to: I was thinking I could take screen shots and post the images. It's kind of a package solution, where I was having issues in getting dialing out to work and also getting Caller ID to appear. I'm no expert on any of this so I'm sure someone would spot errors in my experiments in getting the Cisco spa3102 to work with FS and FS to work with the Polycom IP501 phone. On Fri, 2012-07-20 at 14:47 -0400, Brian Foster wrote: > Care to share the problem/solution for google's sake? > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > > On Jul 20, 2012 2:22 PM, "Todd Bailey" wrote: > > Never mind I fixed it.. > > On Fri, 2012-07-20 at 11:00 -0700, Todd Bailey wrote: > > Hi all, > > > > Using a spa3102 ata, Frees switch and a Polycom ip501, I'm > not getting > > caller id to display on the phone. > > > > Is there some 'magical' setting that I need to address, > somewhere ? > > > > Note: when an internal softphone extension calls the ip501, > I see the > > caller id. This implies the phone and FS are working. > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From toddb at toddbailey.net Fri Jul 20 23:52:42 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Fri, 20 Jul 2012 12:52:42 -0700 Subject: [Freeswitch-users] Saving phone information from a Polycom IP501 Message-ID: <1342813962.5783.88.camel@mythtv> Just wondering if it's possible to download data from a phone and save it into separate files or into a database? I just save a number into the phone and it occurred to me the minute I have to reboot or power cycle the phone, all contact info will be lost. If possible, I'd like to store the contact information and call history into a database for future reference, and perhaps being able to store that information into my email and contact management application (currently Evolution) From aksrini at hotmail.com Sat Jul 21 00:30:28 2012 From: aksrini at hotmail.com (Srini K) Date: Fri, 20 Jul 2012 13:30:28 -0700 Subject: [Freeswitch-users] No reINVITE when bridging two sessions from mod_managed with bypass_media_after_bridge=true Message-ID: Hi, Iam trying to bypass media from FS after two call legs are bridged using mod_managed. The code looks like public void Run(AppContext context) { var fsApi = new FreeSWITCH.Native.Api(); var aLegSession = context.Session; // Answer the incoming call aLegSession.Answer(); // Play the prompt aLegSession.StreamFile("ivr/ThankYou.wav", 0); // Create outBound session var bLegSession = new ManagedSession("sofia/gateway/95/4151230000"); // Bypass Media aLegSession.SetVariable("bypass_media_after_bridge", "true"); bLegSession.SetVariable("bypass_media_after_bridge", "true"); fsApi.ExecuteString(string.Format("uuid_bridge {0} {1}", aLegSession.GetUuid(), bLegSession.GetUuid())); } I don't see FreeSWITCH sending re-Invite after the call is bridged. What I've already tried and did not succeed: 1) set bypass_media=true, on A leg only, on B leg only, on both legs 2) set bypass_media_after_bridge=true, on A leg only, on B leg only, on both legs When I tried without using mod_managed using only dialplan, FS sends re-Invite. Whether Iam doing anything stupid in mod_managed? Regards Srini -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120720/d61b05b1/attachment.html From jason.holden at teksavvy.com Sat Jul 21 00:41:26 2012 From: jason.holden at teksavvy.com (Jason Holden) Date: Fri, 20 Jul 2012 16:41:26 -0400 Subject: [Freeswitch-users] dtmf problems Message-ID: Hi, I have noticed a problem with some IP phones and am unsure how to correct this issue in fs. There are no settings on the IP phone that would correct the problem experienced. If anyone has any suggestions please let me know. Note this problem is experienced with git versions from late last year and from January this year. Vender notes The sequence of DTMF event messages we send is actually pretty logical when you look at it closely. We send three packets with 0 duration to announce the beginning of the DTMF event. This mirrors the 3 packets sent at the end with the final duration. The Polycom does the latter, but not the former. In between we send a packet with a new duration every 10ms. The "duration" in each packet increases by 80. Duration is in "samples", 80 make up 10 ms. Once the codec knows that the DTMF event has ended, it begins the sequence of 3 redundant end packets showing the final duration. So, looking at the sequence of durations, the normal last three are supplanted by the 3 redundant end packets. The Polycom does the same thing. The difference between us and the Polycom is that the PC sends exactly one packet every 20 ms in the stream, in place of the audio packet that would have been sent. We, OTOH, 1) send the DTMF event packets as soon as they are ready, 2) do not send the same number of packets as the audio codec would have. As a result, we send a different number of packets during a DTMF event than during normal audio. The Freeswitch appears to assume that all packets in an RTP stream represent the same amount of time, 20ms in this case. During our DTMF event, this is not the case. If you look at the rtp coming into the FS and going out, our first 5 DTMF packets arrive very soon after the last audio packet. The FS delays them and sent each one in a 20ms slot. But DTMF packets do not map linearly to the timeline. Therefore distorts the stream of packets. Our DTMF packets arrive spread over a much longer time than the actual duration of the DTMF event. That is almost certainly the cause of the problem. Since the Polycom sends exactly one packet in each 20 ms time slot, even during a DTMF event, that works with the FS. When you think about it, the whole reason for RTP is to handle changes in latency, missing packets and so on, it does not make sense for the FS to force every packet into a 20 ms timeslot. The answer is simply for the FS to forward DTMF packets immediately instead of delaying them for a 20 ms timeslot. IMHO, all RTP packets should be forwarded this way. I am pretty sure about what is happening in our phone that causes the timing oddities. When a DTMF event physically starts, the codec starts sending DTMF packets. However, the DTMF packets have no encoding latency, so the first packets seem to arrive immediately after the last audio packet. Then when the latency has fallen to zero, packets go out every 10 ms with corresponding increases in duration. AT the end, the encoding latency must be re-established so there is a gap in the time line after the last DTMF event is received and the first audio packet. Note the issue was noticed when the information was leaving our freeswitch and heading out to the carrier. We use bypass media and passive dtmf is set to true. Any assistance would be great. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120720/3cc6586d/attachment.html From fiorix at gmail.com Sat Jul 21 08:58:23 2012 From: fiorix at gmail.com (Alexandre Fiori) Date: Sat, 21 Jul 2012 00:58:23 -0400 Subject: [Freeswitch-users] dialplan commands on playback Message-ID: <5117A1BD-A549-4091-8F0F-577DE29C747C@gmail.com> is there any way to tell fs to execute a command at, say, 75% of a playback? like, call or schedule a detect_speech (or read) halfway through - af -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120721/83d02433/attachment.html From yehavi.bourvine at gmail.com Sat Jul 21 09:37:38 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sat, 21 Jul 2012 08:37:38 +0300 Subject: [Freeswitch-users] Saving phone information from a Polycom IP501 In-Reply-To: <1342813962.5783.88.camel@mythtv> References: <1342813962.5783.88.camel@mythtv> Message-ID: Hi Todd, Create a file named *mac-address*-directory.xml and make it world writeable (or at least something that allows your tftp server to write). When the user changes his/her directory this file should be re-written by the phone (I don't remember whether it is immediately or after a while). You cannot save the calls lists. __Yehavi: 2012/7/20 Todd Bailey > Just wondering if it's possible to download data from a phone and save > it into separate files or into a database? > > I just save a number into the phone and it occurred to me the minute I > have to reboot or power cycle the phone, all contact info will be lost. > > If possible, I'd like to store the contact information and call history > into a database for future reference, and perhaps being able to store > that information into my email and contact management application > (currently Evolution) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120721/16576461/attachment-0001.html From toddb at toddbailey.net Sat Jul 21 10:17:25 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Fri, 20 Jul 2012 23:17:25 -0700 Subject: [Freeswitch-users] Saving phone information from a Polycom IP501 In-Reply-To: References: <1342813962.5783.88.camel@mythtv> Message-ID: <1342851445.5783.116.camel@mythtv> OK I changed the permissions to 777 on files in the tftpboot folder and made and saved a few changes to the "contact directory" entries on the phone. Nothing has happened immediately to the 0004F2047968-directory.xml file, but will wait and check again tomorrow. Since the phone is able to boot from this folder, I'm assuming it can write changes as well. On Sat, 2012-07-21 at 08:37 +0300, Yehavi Bourvine wrote: > Hi Todd, > > Create a file named mac-address-directory.xml and make it world > writeable (or at least something that allows your tftp server to > write). When the user changes his/her directory this file should be > re-written by the phone (I don't remember whether it is immediately or > after a while). > > You cannot save the calls lists. > > __Yehavi: > > > 2012/7/20 Todd Bailey > Just wondering if it's possible to download data from a phone > and save > it into separate files or into a database? > > I just save a number into the phone and it occurred to me the > minute I > have to reboot or power cycle the phone, all contact info will > be lost. > > If possible, I'd like to store the contact information and > call history > into a database for future reference, and perhaps being able > to store > that information into my email and contact management > application > (currently Evolution) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From rajkumar.kanniappan at sasken.com Sat Jul 21 10:21:13 2012 From: rajkumar.kanniappan at sasken.com (Rajkumar Kanniappan) Date: Sat, 21 Jul 2012 11:51:13 +0530 Subject: [Freeswitch-users] Making a call from PSTN to SoftPhone In-Reply-To: References: <6F91E0FFDA542149961F7BDED2D2B94B568E46D0C5@EXGMBX01.sasken.com> <6F91E0FFDA542149961F7BDED2D2B94B568E46D12A@EXGMBX01.sasken.com>, Message-ID: <6F91E0FFDA542149961F7BDED2D2B94B568E108EE4@EXGMBX01.sasken.com> I have configured 2 sip profile in my FS, sofia1: under sip_ip is my ip, sip_port is 5060 sofia2: under sip_ip is my ip, sip_port is 5080 now which one will get selected when i bridge the call... ________________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004 [curriegrad2004 at gmail.com] Sent: Friday, July 20, 2012 22:41 To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Making a call from PSTN to SoftPhone Not exactly. Sofia is just an endpoint, selection of the profile is done automatically when you bridge the call under the user/XXX at XXX endpoint. On Fri, Jul 20, 2012 at 1:43 AM, Rajkumar Kanniappan wrote: > Isnt it necessary that bridging of a call has to take place only after the selection of sofia profile. > > Otherwise when is it going to select the sofia profile... Can you please explain in detail about the sofia profile selection in this scenario > > Thanks > Rajkumar > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004 > Sent: Friday, July 20, 2012 1:01 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Making a call from PSTN to SoftPhone > > FreeTDM in FreeSWITCH will not even touch sofia at all. All it is going to do is bridge your particular timeslot from your PRI into an endpoint of your choosing. FreeSWITCH will then do the rest of the dirty job of selecting which sofia profile your IP set is registered at. > > On Thu, Jul 19, 2012 at 10:34 PM, Rajkumar Kanniappan wrote: >> Hi, >> >> >> >> if I make a call from PSTN extension to a IP phone, >> >> call has to reach from Extension -> PRI card(VoIP gateway) -> Drivers >> -> FreeTDM, and from here what is going to be the job of FreeTDM. Is >> FreeTDM going to contact Sofia module, if so what is going to be the >> configuration changes for freetdm and sofia. >> Will FreeTDM module select any sofia profile(internal or external) to >> make calls. >> >> And how sofia profile is getting selected for this particular call. >> >> >> Please clarify this to me... >> >> >> >> Thanks, >> >> Rajkumar >> >> >> >> >> ________________________________ >> SASKEN BUSINESS DISCLAIMER: This message may contain confidential, >> proprietary or legally privileged information. In case you are not the >> original intended Recipient of the message, you must not, directly or >> indirectly, use, disclose, distribute, print, or copy any part of this >> message and you are requested to delete it and inform the sender. Any >> views expressed in this message are those of the individual sender >> unless otherwise stated. Nothing contained in this message shall be >> construed as an offer or acceptance of any offer by Sasken >> Communication Technologies Limited ("Sasken") unless sent with that >> express intent and with due authority of Sasken. Sasken has taken >> enough precautions to prevent the spread of viruses. However the >> company accepts no liability for any damage caused by any virus transmitted by this email. >> Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users 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 >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From curriegrad2004 at gmail.com Sat Jul 21 10:29:35 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 20 Jul 2012 23:29:35 -0700 Subject: [Freeswitch-users] Making a call from PSTN to SoftPhone In-Reply-To: <6F91E0FFDA542149961F7BDED2D2B94B568E108EE4@EXGMBX01.sasken.com> References: <6F91E0FFDA542149961F7BDED2D2B94B568E46D0C5@EXGMBX01.sasken.com> <6F91E0FFDA542149961F7BDED2D2B94B568E46D12A@EXGMBX01.sasken.com> <6F91E0FFDA542149961F7BDED2D2B94B568E108EE4@EXGMBX01.sasken.com> Message-ID: Again this will depend on where you are sending your calls to which SIP profile. If you have a phone registered on sofia1, then sofia1 receives the request and transfers it to the dialplan and FreeSWITCH will do the rest depending on how you configured it to act. On Fri, Jul 20, 2012 at 11:21 PM, Rajkumar Kanniappan wrote: > I have configured 2 sip profile in my FS, > > sofia1: under sip_ip is my ip, sip_port is 5060 > sofia2: under sip_ip is my ip, sip_port is 5080 > > now which one will get selected when i bridge the call... > > > > > ________________________________________ > From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004 [curriegrad2004 at gmail.com] > Sent: Friday, July 20, 2012 22:41 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Making a call from PSTN to SoftPhone > > Not exactly. Sofia is just an endpoint, selection of the profile is > done automatically when you bridge the call under the user/XXX at XXX > endpoint. > > On Fri, Jul 20, 2012 at 1:43 AM, Rajkumar Kanniappan > wrote: >> Isnt it necessary that bridging of a call has to take place only after the selection of sofia profile. >> >> Otherwise when is it going to select the sofia profile... Can you please explain in detail about the sofia profile selection in this scenario >> >> Thanks >> Rajkumar >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004 >> Sent: Friday, July 20, 2012 1:01 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Making a call from PSTN to SoftPhone >> >> FreeTDM in FreeSWITCH will not even touch sofia at all. All it is going to do is bridge your particular timeslot from your PRI into an endpoint of your choosing. FreeSWITCH will then do the rest of the dirty job of selecting which sofia profile your IP set is registered at. >> >> On Thu, Jul 19, 2012 at 10:34 PM, Rajkumar Kanniappan wrote: >>> Hi, >>> >>> >>> >>> if I make a call from PSTN extension to a IP phone, >>> >>> call has to reach from Extension -> PRI card(VoIP gateway) -> Drivers >>> -> FreeTDM, and from here what is going to be the job of FreeTDM. Is >>> FreeTDM going to contact Sofia module, if so what is going to be the >>> configuration changes for freetdm and sofia. >>> Will FreeTDM module select any sofia profile(internal or external) to >>> make calls. >>> >>> And how sofia profile is getting selected for this particular call. >>> >>> >>> Please clarify this to me... >>> >>> >>> >>> Thanks, >>> >>> Rajkumar >>> >>> >>> >>> >>> ________________________________ >>> SASKEN BUSINESS DISCLAIMER: This message may contain confidential, >>> proprietary or legally privileged information. In case you are not the >>> original intended Recipient of the message, you must not, directly or >>> indirectly, use, disclose, distribute, print, or copy any part of this >>> message and you are requested to delete it and inform the sender. Any >>> views expressed in this message are those of the individual sender >>> unless otherwise stated. Nothing contained in this message shall be >>> construed as an offer or acceptance of any offer by Sasken >>> Communication Technologies Limited ("Sasken") unless sent with that >>> express intent and with due authority of Sasken. Sasken has taken >>> enough precautions to prevent the spread of viruses. However the >>> company accepts no liability for any damage caused by any virus transmitted by this email. >>> Read Disclaimer at http://www.sasken.com/extras/mail_disclaimer.html >>> >>> ______________________________________________________________________ >>> ___ Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users 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 >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From shaik.bawajan at gmail.com Sat Jul 21 10:44:57 2012 From: shaik.bawajan at gmail.com (shaik bawajan) Date: Sat, 21 Jul 2012 12:14:57 +0530 Subject: [Freeswitch-users] A fatal error has been detected by the Java Runtime Environment: In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD022FAE2819@NY1-EXMB-01.ip-soft.net> References: <1342529673372-7580878.post@n2.nabble.com> <6A6B4C284AD15042B429EB9D904544AD022FAE2819@NY1-EXMB-01.ip-soft.net> Message-ID: Thanks a lot, It is very helpful to me. But, the each event are receiving multiple times ( each event coming 3 or 4 times ). It will happen like this or is there any way to stop this. here am attaching my java class, where am creating a single thread and making outbound, playing a file. The class itself is a Event Listener and am adding below event listeners from other initiate Class. client.setEventSubscriptions("plain", "all"); client.addEventFilter("Event-Name", "CHANNEL_CREATE"); client.addEventFilter("Event-Name","BACKGROUND_JOB"); client.addEventFilter("Event-Name","CHANNEL_STATE"); client.addEventFilter("Event-Name","CHANNEL_EXECUTE_COMPLETE"); client.addEventFilter("Event-Name","CHANNEL_HANGUP"); client.addEventFilter("Event-Name","CHANNEL_HANGUP_COMPLETE"); client.addEventFilter("Event-Name","DTMF"); client.addEventFilter("Event-Name","HEARTBEAT"); Thanks in advance, On Wed, Jul 18, 2012 at 7:30 PM, Hector Geraldino < Hector.Geraldino at ipsoft.com> wrote: > Hello, > > Trying to help you to solve a crash in the jvm for a multithreaded > application is damn hard. Doing it using a mailing list is even harder, and > without looking at your source code is almost impossible. > > However I want to recommend you to drop the use of this library (which is > a java wrapper of the FS core lib written in C) and use the pure Java ESL > Client (http://wiki.freeswitch.org/wiki/Java_ESL_Client). You will have > full access to the source code for debug, no dependencies on native > libraries, and a good set of examples. > > Good luck! > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of bawajan > Sent: Tuesday, July 17, 2012 8:55 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] A fatal error has been detected by the Java > Runtime Environment: > > Hi, > > Am using ESL inbound connection to make calls and have a below flow > (written in java) > > a) originating 15 calls simultaneously with park function > b) play audio file > c) creating new thread and passing eslconnection and playing a IVR in > originate call. > > Here while handling events getting the below error : > > > # A fatal error has been > detected by the Java Runtime Environment: > > INFO | jvm 3 | 2012/07/17 17:54:36 | # > INFO | jvm 3 | 2012/07/17 17:54:36 | # SIGSEGV (0xb) at > pc=0x00007fc01f1d3943, pid=14759, tid=140463084390144 > INFO | jvm 3 | 2012/07/17 17:54:36 | # > INFO | jvm 3 | 2012/07/17 17:54:36 | # JRE version: 6.0_25-b06 > INFO | jvm 3 | 2012/07/17 17:54:36 | # Java VM: Java HotSpot(TM) > 64-Bit > Server VM (20.0-b11 mixed mode linux-amd64 compressed oops) > INFO | jvm 3 | 2012/07/17 17:54:36 | # Problematic frame: > INFO | jvm 3 | 2012/07/17 17:54:36 | # C [libesljni.so+0xd943] long > double+0x183 > INFO | jvm 3 | 2012/07/17 17:54:36 | # > INFO | jvm 3 | 2012/07/17 17:54:36 | # An error report file with more > information is saved as: > INFO | jvm 3 | 2012/07/17 17:54:36 | # > /usr/local/freeswitch/hs_err_pid14759.log > INFO | jvm 3 | 2012/07/17 17:54:36 | # > INFO | jvm 3 | 2012/07/17 17:54:36 | # If you would like to submit a > bug report, please visit: > INFO | jvm 3 | 2012/07/17 17:54:36 | # > http://java.sun.com/webapps/bugreport/crash.jsp > INFO | jvm 3 | 2012/07/17 17:54:36 | # The crash happened outside the > Java Virtual Machine in native code. > INFO | jvm 3 | 2012/07/17 17:54:36 | # See problematic frame for where > to report the bug. > > > > plz let me know, where am doing mistake and how to resolve it. > > Thanks in advance. > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/A-fatal-error-has-been-detected-by-the-Java-Runtime-Environment-tp7580878.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120721/1acf9735/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: ExecutorProcess.java Type: application/octet-stream Size: 9780 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120721/1acf9735/attachment-0001.obj From yehavi.bourvine at gmail.com Sat Jul 21 11:45:03 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Sat, 21 Jul 2012 10:45:03 +0300 Subject: [Freeswitch-users] Saving phone information from a Polycom IP501 In-Reply-To: <1342851445.5783.116.camel@mythtv> References: <1342813962.5783.88.camel@mythtv> <1342851445.5783.116.camel@mythtv> Message-ID: Verify that your tftp server allows writing. The best way is to use TCPDUMP as I wrote you in order to see what the phone is trying to do. Look also in your syslog to see whether htere is some error message there. __Yehavi: 2012/7/21 Todd Bailey > > > > OK I changed the permissions to 777 on files in the tftpboot folder and > made and saved a few changes to the "contact directory" entries on the > phone. > > Nothing has happened immediately to the 0004F2047968-directory.xml > file, but will wait and check again tomorrow. > > Since the phone is able to boot from this folder, I'm assuming it can > write changes as well. > > > > > On Sat, 2012-07-21 at 08:37 +0300, Yehavi Bourvine wrote: > > Hi Todd, > > > > Create a file named mac-address-directory.xml and make it world > > writeable (or at least something that allows your tftp server to > > write). When the user changes his/her directory this file should be > > re-written by the phone (I don't remember whether it is immediately or > > after a while). > > > > You cannot save the calls lists. > > > > __Yehavi: > > > > > > 2012/7/20 Todd Bailey > > Just wondering if it's possible to download data from a phone > > and save > > it into separate files or into a database? > > > > I just save a number into the phone and it occurred to me the > > minute I > > have to reboot or power cycle the phone, all contact info will > > be lost. > > > > If possible, I'd like to store the contact information and > > call history > > into a database for future reference, and perhaps being able > > to store > > that information into my email and contact management > > application > > (currently Evolution) > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120721/86ee7277/attachment.html From Nabble_01394 at slickdeals.endjunk.com Sat Jul 21 20:51:42 2012 From: Nabble_01394 at slickdeals.endjunk.com (mazilo) Date: Sat, 21 Jul 2012 09:51:42 -0700 (PDT) Subject: [Freeswitch-users] libs/spandsp woes in compilation Message-ID: <1342889501930-7581017.post@n2.nabble.com> Just updated to git HEAD be6739e19856c0e84b7d973dc46c4f40cef3ecac and cross compilation woes as shown below: Upon inspecting the newly updated spandsp source files, i.e. libs/spandsp/src/v17tx.c, libs/spandsp/src/v17_v32bis_tx_constellation_maps.h, libs/spandsp/src/v22bis_rx.c, etc., I noticed the following statement: instead of: Searching through the libs/spandsp/configure.ac file, I noticed there is a reference to SPANDSP_USE_FIXED_POINT and not SPANDSP_USE_FIXED_POINTx. So, I suspect this is a typo. ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/libs-spandsp-woes-in-compilation-tp7581017.html Sent from the freeswitch-users mailing list archive at Nabble.com. From andrew at cassidywebservices.co.uk Sat Jul 21 21:12:06 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Sat, 21 Jul 2012 18:12:06 +0100 Subject: [Freeswitch-users] No reINVITE when bridging two sessions from mod_managed with bypass_media_after_bridge=true In-Reply-To: References: Message-ID: Try using uuid_media after the bridge instaed, see if that helps? http://wiki.freeswitch.org/wiki/Mod_commands#uuid_media On 20 July 2012 21:30, Srini K wrote: > Hi, > Iam trying to bypass media from FS after two call legs are bridged using > mod_managed. > The code looks like > public void Run(AppContext context) > { > var fsApi = new FreeSWITCH.Native.Api(); > var aLegSession = context.Session; > // Answer the incoming call > aLegSession.Answer(); > // Play the prompt > aLegSession.StreamFile("ivr/ThankYou.wav", 0); > // Create outBound session > var bLegSession = new > ManagedSession("sofia/gateway/95/4151230000"); > > // Bypass Media > aLegSession.SetVariable("bypass_media_after_bridge", "true"); > bLegSession.SetVariable("bypass_media_after_bridge", "true"); > fsApi.ExecuteString(string.Format("uuid_bridge {0} {1}", > aLegSession.GetUuid(), bLegSession.GetUuid())); > } > > I don't see FreeSWITCH sending re-Invite after the call is bridged. > What I've already tried and did not succeed: > 1) set bypass_media=true, on A leg only, on B leg only, on both legs > 2) set bypass_media_after_bridge=true, on A leg only, on B leg only, on > both legs > > When I tried without using mod_managed using only dialplan, FS sends > re-Invite. > > > > > > > Whether Iam doing anything stupid in mod_managed? > > Regards > Srini > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120721/76d78907/attachment.html From drk at drkngs.net Sat Jul 21 22:15:14 2012 From: drk at drkngs.net (Dave R. Kompel) Date: Sat, 21 Jul 2012 11:15:14 -0700 Subject: [Freeswitch-users] No reINVITE when bridging two sessions from mod_managed with bypass_media_after_bridge=true In-Reply-To: Message-ID: <20120721181514.c5444fbc@mail.tritonwest.net> Andrew, There are a bunch of things wrong with your example for what you're trying to do. The first has nothing to do w/ mod_managed, which is that you are trying to set variables on the a_leg, that are used by the bridge application, after the b_leg is up. The media variables that you set on the a_leg, are arguments to the dial plan app "bridge". Since you are never using it to place the outbound call for the b_leg, they're not going to work. Another problem is that mod_managed doesen't really work right to place an outbound call with the "new ManagedSession". That is left over from old days. What's wrong with doing a "Session.Execute("bridge","sofia/gateway...")"? If you do this then any flags that are used as parameters by the bridge applicaiton will work, since you are just calling it as you would from the dialplan. --Dave _____ From: Andrew Cassidy [mailto:andrew at cassidywebservices.co.uk] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Sat, 21 Jul 2012 10:12:06 -0700 Subject: Re: [Freeswitch-users] No reINVITE when bridging two sessions from mod_managed with bypass_media_after_bridge=true Try using uuid_media after the bridge instaed, see if that helps? http://wiki.freeswitch.org/wiki/Mod_commands#uuid_media On 20 July 2012 21:30, Srini K wrote: Hi, Iam trying to bypass media from FS after two call legs are bridged using mod_managed. The code looks like public void Run(AppContext context) { var fsApi = new FreeSWITCH.Native.Api(); var aLegSession = context.Session; // Answer the incoming call aLegSession.Answer(); // Play the prompt aLegSession.StreamFile("ivr/ThankYou.wav", 0); // Create outBound session var bLegSession = new ManagedSession("sofia/gateway/95/4151230000"); // Bypass Media aLegSession.SetVariable("bypass_media_after_bridge", "true"); bLegSession.SetVariable("bypass_media_after_bridge", "true"); fsApi.ExecuteString(string.Format("uuid_bridge {0} {1}", aLegSession.GetUuid(), bLegSession.GetUuid())); } I don't see FreeSWITCH sending re-Invite after the call is bridged. What I've already tried and did not succeed: 1) set bypass_media=true, on A leg only, on B leg only, on both legs 2) set bypass_media_after_bridge=true, on A leg only, on B leg only, on both legs When I tried without using mod_managed using only dialplan, FS sends re-Invite. Whether Iam doing anything stupid in mod_managed? Regards Srini _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Andrew Cassidy BSc (Hons) MBCS SSCA Managing Director T 03300 100 960 F 03300 100 961 E andrew at cassidywebservices.co.uk W www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120721/f40f2a46/attachment-0001.html From bdfoster at endigotech.com Sun Jul 22 00:03:46 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 21 Jul 2012 16:03:46 -0400 Subject: [Freeswitch-users] libs/spandsp woes in compilation In-Reply-To: <1342889501930-7581017.post@n2.nabble.com> References: <1342889501930-7581017.post@n2.nabble.com> Message-ID: http://jira.freeswitch.org Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 21, 2012 12:52 PM, "mazilo" wrote: > Just updated to git HEAD be6739e19856c0e84b7d973dc46c4f40cef3ecac and cross > compilation woes as shown below: > > Upon inspecting the newly updated spandsp source files, i.e. > libs/spandsp/src/v17tx.c, > libs/spandsp/src/v17_v32bis_tx_constellation_maps.h, > libs/spandsp/src/v22bis_rx.c, etc., I noticed the following statement: > > instead of: > > Searching through the libs/spandsp/configure.ac file, I noticed there is a > reference to SPANDSP_USE_FIXED_POINT and not SPANDSP_USE_FIXED_POINTx. So, > I > suspect this is a typo. > > > > ----- > FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 > Watts of electricity. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/libs-spandsp-woes-in-compilation-tp7581017.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120721/fdf7904d/attachment.html From peter.olsson at visionutveckling.se Sun Jul 22 01:02:55 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sat, 21 Jul 2012 21:02:55 +0000 Subject: [Freeswitch-users] libs/spandsp woes in compilation In-Reply-To: <1342889501930-7581017.post@n2.nabble.com> References: <1342889501930-7581017.post@n2.nabble.com> Message-ID: For the fifth time last couple of weeks: make spandsp-reconf, then make current.... /Peter 21 jul 2012 kl. 18:58 skrev "mazilo" : > Just updated to git HEAD be6739e19856c0e84b7d973dc46c4f40cef3ecac and cross > compilation woes as shown below: > > Upon inspecting the newly updated spandsp source files, i.e. > libs/spandsp/src/v17tx.c, > libs/spandsp/src/v17_v32bis_tx_constellation_maps.h, > libs/spandsp/src/v22bis_rx.c, etc., I noticed the following statement: > > instead of: > > Searching through the libs/spandsp/configure.ac file, I noticed there is a > reference to SPANDSP_USE_FIXED_POINT and not SPANDSP_USE_FIXED_POINTx. So, I > suspect this is a typo. > > > > ----- > FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/libs-spandsp-woes-in-compilation-tp7581017.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:500adc6232761438866156! > From peter.olsson at visionutveckling.se Sun Jul 22 01:22:32 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sat, 21 Jul 2012 21:22:32 +0000 Subject: [Freeswitch-users] libs/spandsp woes in compilation In-Reply-To: References: <1342889501930-7581017.post@n2.nabble.com>, Message-ID: And of course, if it doesn't help, submit to Jira. /Peter 21 jul 2012 kl. 23:10 skrev "Peter Olsson" : > For the fifth time last couple of weeks: make spandsp-reconf, then make current.... > > /Peter > > 21 jul 2012 kl. 18:58 skrev "mazilo" : > >> Just updated to git HEAD be6739e19856c0e84b7d973dc46c4f40cef3ecac and cross >> compilation woes as shown below: >> >> Upon inspecting the newly updated spandsp source files, i.e. >> libs/spandsp/src/v17tx.c, >> libs/spandsp/src/v17_v32bis_tx_constellation_maps.h, >> libs/spandsp/src/v22bis_rx.c, etc., I noticed the following statement: >> >> instead of: >> >> Searching through the libs/spandsp/configure.ac file, I noticed there is a >> reference to SPANDSP_USE_FIXED_POINT and not SPANDSP_USE_FIXED_POINTx. So, I >> suspect this is a typo. >> >> >> >> ----- >> FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. >> -- >> View this message in context: http://freeswitch-users.2379917.n2.nabble.com/libs-spandsp-woes-in-compilation-tp7581017.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:500b175832767053524390! > From Nabble_01394 at slickdeals.endjunk.com Sun Jul 22 18:01:32 2012 From: Nabble_01394 at slickdeals.endjunk.com (mazilo) Date: Sun, 22 Jul 2012 07:01:32 -0700 (PDT) Subject: [Freeswitch-users] libs/spandsp woes in compilation In-Reply-To: <1342889501930-7581017.post@n2.nabble.com> References: <1342889501930-7581017.post@n2.nabble.com> Message-ID: <1342965692888-7581023.post@n2.nabble.com> Finally, I reverted to git HEAD 9fe08675a1d3f0a8ba9c777befed7d6cc8a921f9 and the cross compilation is OK. ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/libs-spandsp-woes-in-compilation-tp7581017p7581023.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jeff at jefflenk.com Sun Jul 22 18:38:33 2012 From: jeff at jefflenk.com (Jeff Lenk) Date: Sun, 22 Jul 2012 07:38:33 -0700 (PDT) Subject: [Freeswitch-users] libs/spandsp woes in compilation In-Reply-To: <1342965692888-7581023.post@n2.nabble.com> References: <1342889501930-7581017.post@n2.nabble.com> <1342965692888-7581023.post@n2.nabble.com> Message-ID: <1342967913848-7581024.post@n2.nabble.com> Please report this to Jira or it will not get fixed. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/libs-spandsp-woes-in-compilation-tp7581017p7581024.html Sent from the freeswitch-users mailing list archive at Nabble.com. From anthony.minessale at gmail.com Sun Jul 22 20:38:43 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 22 Jul 2012 11:38:43 -0500 Subject: [Freeswitch-users] libs/spandsp woes in compilation In-Reply-To: <1342889501930-7581017.post@n2.nabble.com> References: <1342889501930-7581017.post@n2.nabble.com> Message-ID: Report issues or bugs to Jira.freeswitch.org On Jul 21, 2012 11:52 AM, "mazilo" wrote: > Just updated to git HEAD be6739e19856c0e84b7d973dc46c4f40cef3ecac and cross > compilation woes as shown below: > > Upon inspecting the newly updated spandsp source files, i.e. > libs/spandsp/src/v17tx.c, > libs/spandsp/src/v17_v32bis_tx_constellation_maps.h, > libs/spandsp/src/v22bis_rx.c, etc., I noticed the following statement: > > instead of: > > Searching through the libs/spandsp/configure.ac file, I noticed there is a > reference to SPANDSP_USE_FIXED_POINT and not SPANDSP_USE_FIXED_POINTx. So, > I > suspect this is a typo. > > > > ----- > FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 > Watts of electricity. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/libs-spandsp-woes-in-compilation-tp7581017.html > Sent from the freeswitch-users mailing list archive at Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120722/00411b90/attachment.html From philq at qsystemsengineering.com Sun Jul 22 21:16:11 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Sun, 22 Jul 2012 13:16:11 -0400 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! Message-ID: <001901cd682d$b3e4dd70$1bae9850$@com> Recently through several of the latest git pulls, FS stops routing calls with the following error, usually within a few minutes of making a call through Google Voice, not sure if receiving a call through GV will cause the problem yet: sofia_glue.c:1083 No RTP ports available! Restarting FS fixes the problem. Before I file the requisite Jira, is anyone aware of this problem or a potential configuration issue that might not be compatible with the newer builds of FS? This wasn't happening until the last week or two. Current version we're running is: FreeSWITCH Version 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z Here's a quick snip from the console when attempting to dial out to show the context of the error: 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to pass zrtp-hash between legs 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/102 at 192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 sofia/internal/102 at 192.168.1.6:5060 Original read codec set to PCMU:0 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports available! 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/102 at 192.168.1.6:5060 [BREAK] 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 (sofia/internal/102 at 192.168.1.6:5060) Callstate Change RINGING -> HANGUP 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup sofia/internal/102 at 192.168.1.6:5060 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Regards, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120722/b89912ae/attachment-0001.html From curriegrad2004 at gmail.com Sun Jul 22 22:35:24 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sun, 22 Jul 2012 11:35:24 -0700 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! In-Reply-To: <001901cd682d$b3e4dd70$1bae9850$@com> References: <001901cd682d$b3e4dd70$1bae9850$@com> Message-ID: Repeat the call again and dump a SIP trace here while you're at it On Sun, Jul 22, 2012 at 10:16 AM, Phil Quesinberry wrote: > Recently through several of the latest git pulls, FS stops routing calls > with the following error, usually within a few minutes of making a call > through Google Voice, not sure if receiving a call through GV will cause the > problem yet: > > sofia_glue.c:1083 No RTP ports available! > > Restarting FS fixes the problem. > > Before I file the requisite Jira, is anyone aware of this problem or a > potential configuration issue that might not be compatible with the newer > builds of FS? This wasn?t happening until the last week or two. > > Current version we?re running is: FreeSWITCH Version > 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z > > Here?s a quick snip from the console when attempting to dial out to show the > context of the error: > > 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash > > 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to > pass zrtp-hash between legs > > 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ > not set, so not propagating zrtp-hash > > 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare > [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > > 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec > sofia/internal/102 at 192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits > > 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 > sofia/internal/102 at 192.168.1.6:5060 Original read codec set to PCMU:0 > > 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. > Disable 2833 dtmf and switch to INFO > > 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports available! > > 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal > sofia/internal/102 at 192.168.1.6:5060 [BREAK] > > 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 > (sofia/internal/102 at 192.168.1.6:5060) Callstate Change RINGING -> HANGUP > > 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup > sofia/internal/102 at 192.168.1.6:5060 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] > > Regards, > > Phil Quesinberry > > Q Systems Engineering, Inc. > > Electronic Controls and Embedded Systems Development > > (410) 969-8002 > > http://www.qsystemsengineering.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Sun Jul 22 22:58:52 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sun, 22 Jul 2012 13:58:52 -0500 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! In-Reply-To: <001901cd682d$b3e4dd70$1bae9850$@com> References: <001901cd682d$b3e4dd70$1bae9850$@com> Message-ID: Jira Jira jira On Jul 22, 2012 12:41 PM, "Phil Quesinberry" wrote: > ** > > Recently through several of the latest git pulls, FS stops routing calls > with the following error, usually within a few minutes of making a call > through Google Voice, not sure if receiving a call through GV will cause > the problem yet: > > sofia_glue.c:1083 No RTP ports available! > > Restarting FS fixes the problem. > > Before I file the requisite Jira, is anyone aware of this problem or a > potential configuration issue that might not be compatible with the newer > builds of FS? This wasn?t happening until the last week or two. > > Current version we?re running is: FreeSWITCH Version > 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z > > Here?s a quick snip from the console when attempting to dial out to show > the context of the error: > > 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash > > 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to > pass zrtp-hash between legs > > 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ > not set, so not propagating zrtp-hash > > 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare > [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > > 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec > sofia/internal/102 at 192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits > > 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 sofia/internal/ > 102 at 192.168.1.6:5060 Original read codec set to PCMU:0 > > 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. > Disable 2833 dtmf and switch to INFO > > 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports available! > > 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal > sofia/internal/102 at 192.168.1.6:5060 [BREAK] > > 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 (sofia/internal/ > 102 at 192.168.1.6:5060) Callstate Change RINGING -> HANGUP > > 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup > sofia/internal/102 at 192.168.1.6:5060 [CS_EXECUTE] > [INCOMPATIBLE_DESTINATION] > > Regards, > > *******Phil Quesinberry* > > Q Systems Engineering, Inc. > > Electronic Controls and Embedded Systems Development > > (410) 969-8002 > > *****http://www.qsystemsengineering.com* > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120722/cc715856/attachment.html From peter.olsson at visionutveckling.se Sun Jul 22 23:16:07 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 22 Jul 2012 19:16:07 +0000 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! In-Reply-To: <001901cd682d$b3e4dd70$1bae9850$@com> References: <001901cd682d$b3e4dd70$1bae9850$@com> Message-ID: <1FFF97C269757C458224B7C895F35F15136AA6@cantor.std.visionutv.se> This means that the internal port allocator in FS doesn't have any RTP ports available. Either because you have not enough ports configured (less then you use), or if there is a bug somewhere not returning ports back to the system. You should report to Jira, and attach the needed logs in there. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Phil Quesinberry [philq at qsystemsengineering.com] Skickat: den 22 juli 2012 19:16 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! Recently through several of the latest git pulls, FS stops routing calls with the following error, usually within a few minutes of making a call through Google Voice, not sure if receiving a call through GV will cause the problem yet: sofia_glue.c:1083 No RTP ports available! Restarting FS fixes the problem. Before I file the requisite Jira, is anyone aware of this problem or a potential configuration issue that might not be compatible with the newer builds of FS? This wasn?t happening until the last week or two. Current version we?re running is: FreeSWITCH Version 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z Here?s a quick snip from the console when attempting to dial out to show the context of the error: 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to pass zrtp-hash between legs 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/102 at 192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 sofia/internal/102 at 192.168.1.6:5060 Original read codec set to PCMU:0 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports available! 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/102 at 192.168.1.6:5060 [BREAK] 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 (sofia/internal/102 at 192.168.1.6:5060) Callstate Change RINGING -> HANGUP 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup sofia/internal/102 at 192.168.1.6:5060 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Regards, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com !DSPAM:500c37ce32761523320785! From tculjaga at gmail.com Sun Jul 22 23:31:14 2012 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Sun, 22 Jul 2012 21:31:14 +0200 Subject: [Freeswitch-users] TLS and connection reuse Message-ID: hello, im just wondering if it is possible to make FS re-use the existing TLS connection established on UA registratioin for incoming calls to UA? as an example... i have extensions 1002 (uses TLS) and 1009 (uses SIP/UDP). Both extensions are registered to the same FS. I can place calls from 1002 everywhere but 1002 cannot get any calls. http://pastebin.freeswitch.org/19575 recv 394 bytes from tls/[109.227.38.121]:60591 at 18:16:03.019253: ------------------------------------------------------------------------ REGISTER sip:85.114.35.241 SIP/2.0 Via: SIP/2.0/TLS 109.227.38.121:5061;rport;branch=z9hG4bK419550279 From: ;tag=376877386 To: Call-ID: 665211288 CSeq: 1 REGISTER Contact: Max-Forwards: 70 User-Agent: Linphone/3.4.0 (eXosip2/unknown) Expires: 2000 Content-Length: 0 ------------------------------------------------------------------------ I know this register message is broken... good for TCP/UDP but bad for TLS :=) anyhow, extension 1002 establishes a TLS flow 109.227.38.121:60591 <> 85.114.35.241:5061 of course you know what happens when 1009 calls 1002, since 1002 advertized address:port in contact header different than the source port of the existing flow, FS tries to establish a new transport towards the UA and it fails. :=) So is there any way we can force FS to re-use the existing flow and send subsequent request messages via existing connection rather than trying to establish a new one that is going to fail miserably? The more i write this e-mail, the more i realize its a UA problem... don't advertize something you cannot get a call on... but anyhow... needed to ask that question :=) so, how do we fix this ? is the implementation of RFC5626 on client side the answer for that... does FS support this rfc at all ? the other solution should be to tie contact port to transport port but that's a hack.. any good advice in how to fix the UA behavior so it can send and receive calls via TLS connections? regards, Tihomir. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120722/8fb52390/attachment-0001.html From jlyons at quikvoip.net Mon Jul 23 01:10:41 2012 From: jlyons at quikvoip.net (James) Date: Sun, 22 Jul 2012 17:10:41 -0400 Subject: [Freeswitch-users] Originate with python ESL Message-ID: <20120722171041.276b0157@calico> I'm trying to do an originate command via python script and ESL. FS attempts to make this call but is rejected by my gateway with a 407. FS does not follow-up with an INVITE with the credentials. Is there a variable I need to set in order to do this? Here's the originate I'm using: e = con.api("create_uuid").getBody() res = con.bgapi("originate", "{origination_uuid=" + e + ",fax_verbose=true,ignore_early_media=true}sofia/gateway/myprovider/XXXXXXXXXX &txfax(/tmp/fax.5661.tiff)") -- James Lyons QuikVoIP, LLC. p: +1 (786) 369-5308 e: jlyons at quikvoip.net From peter.olsson at visionutveckling.se Mon Jul 23 01:49:43 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 22 Jul 2012 21:49:43 +0000 Subject: [Freeswitch-users] Originate with python ESL In-Reply-To: <20120722171041.276b0157@calico> References: <20120722171041.276b0157@calico> Message-ID: <3A3C9159-A6C5-4C41-94B9-5E028400C94B@visionutveckling.se> That should be enough. Are you sure the gateway is properly configured? /Peter 22 jul 2012 kl. 23:37 skrev "James" : > > I'm trying to do an originate command via python script and ESL. FS > attempts to make this call but is rejected by my gateway with a 407. > FS does not follow-up with an INVITE with the credentials. Is there a > variable I need to set in order to do this? > > Here's the originate I'm using: > > e = con.api("create_uuid").getBody() > res = con.bgapi("originate", "{origination_uuid=" + e + > ",fax_verbose=true,ignore_early_media=true}sofia/gateway/myprovider/XXXXXXXXXX > &txfax(/tmp/fax.5661.tiff)") > > > -- > James Lyons > QuikVoIP, LLC. > p: +1 (786) 369-5308 > e: jlyons at quikvoip.net > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:500c6f3f32761203936802! > From jlyons at quikvoip.net Mon Jul 23 02:10:55 2012 From: jlyons at quikvoip.net (James) Date: Sun, 22 Jul 2012 18:10:55 -0400 Subject: [Freeswitch-users] Originate with python ESL In-Reply-To: <3A3C9159-A6C5-4C41-94B9-5E028400C94B@visionutveckling.se> References: <20120722171041.276b0157@calico> <3A3C9159-A6C5-4C41-94B9-5E028400C94B@visionutveckling.se> Message-ID: <20120722181055.4c7fe23a@calico> Yes, this same gw file has been functioning properly with a bash call to fs_cli -x "originate sofia/gateway/mygateway/XXXXXXXXXX &txfax(/tmp/fax.5661.tiff)" I'm now trying to do this with python and ESL to be able to monitor the result of the fax. -James On Sun, 22 Jul 2012 21:49:43 +0000 Peter Olsson wrote: > That should be enough. Are you sure the gateway is properly > configured? > > /Peter > > 22 jul 2012 kl. 23:37 skrev "James" : > > > > > I'm trying to do an originate command via python script and ESL. FS > > attempts to make this call but is rejected by my gateway with a 407. > > FS does not follow-up with an INVITE with the credentials. Is there > > a variable I need to set in order to do this? > > > > Here's the originate I'm using: > > > > e = con.api("create_uuid").getBody() > > res = con.bgapi("originate", "{origination_uuid=" + e + > > ",fax_verbose=true,ignore_early_media=true}sofia/gateway/myprovider/XXXXXXXXXX > > &txfax(/tmp/fax.5661.tiff)") > > > > > > -- > > James Lyons > > QuikVoIP, LLC. > > p: +1 (786) 369-5308 > > e: jlyons at quikvoip.net > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > !DSPAM:500c6f3f32761203936802! > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From peter.olsson at visionutveckling.se Mon Jul 23 02:46:28 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Sun, 22 Jul 2012 22:46:28 +0000 Subject: [Freeswitch-users] Originate with python ESL In-Reply-To: <20120722181055.4c7fe23a@calico> References: <20120722171041.276b0157@calico> <3A3C9159-A6C5-4C41-94B9-5E028400C94B@visionutveckling.se>, <20120722181055.4c7fe23a@calico> Message-ID: <1FFF97C269757C458224B7C895F35F15136B2E@cantor.std.visionutv.se> Seems strange, that would execute exactly the same code within FS. I can't se anything abvious that you're doing wrong. Try doing it exactly the same way, skip the uuid creation, and use api instead of bgapi, and see if anything is doing a difference. fs_cli will send using ESL as well, so there must be something else to it. /Peter ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för James [jlyons at quikvoip.net] Skickat: den 23 juli 2012 00:10 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Originate with python ESL Yes, this same gw file has been functioning properly with a bash call to fs_cli -x "originate sofia/gateway/mygateway/XXXXXXXXXX &txfax(/tmp/fax.5661.tiff)" I'm now trying to do this with python and ESL to be able to monitor the result of the fax. -James On Sun, 22 Jul 2012 21:49:43 +0000 Peter Olsson wrote: > That should be enough. Are you sure the gateway is properly > configured? > > /Peter > > 22 jul 2012 kl. 23:37 skrev "James" : > > > > > I'm trying to do an originate command via python script and ESL. FS > > attempts to make this call but is rejected by my gateway with a 407. > > FS does not follow-up with an INVITE with the credentials. Is there > > a variable I need to set in order to do this? > > > > Here's the originate I'm using: > > > > e = con.api("create_uuid").getBody() > > res = con.bgapi("originate", "{origination_uuid=" + e + > > ",fax_verbose=true,ignore_early_media=true}sofia/gateway/myprovider/XXXXXXXXXX > > &txfax(/tmp/fax.5661.tiff)") > > > > > > -- > > James Lyons > > QuikVoIP, LLC. > > p: +1 (786) 369-5308 > > e: jlyons at quikvoip.net > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:500c7a9832768724918108! From curriegrad2004 at gmail.com Mon Jul 23 10:12:07 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sun, 22 Jul 2012 23:12:07 -0700 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! In-Reply-To: References: <001901cd682d$b3e4dd70$1bae9850$@com> Message-ID: Tony, What can cause this error to happen really? Does FS allow re-use of RTP ports at all given if the media isn't going to the same destination, or doesn't the SDP header define some kind of session tracking mechnaism so ports can be reused? On Sun, Jul 22, 2012 at 11:58 AM, Anthony Minessale wrote: > Jira Jira jira > > On Jul 22, 2012 12:41 PM, "Phil Quesinberry" > wrote: >> >> Recently through several of the latest git pulls, FS stops routing calls >> with the following error, usually within a few minutes of making a call >> through Google Voice, not sure if receiving a call through GV will cause the >> problem yet: >> >> sofia_glue.c:1083 No RTP ports available! >> >> Restarting FS fixes the problem. >> >> Before I file the requisite Jira, is anyone aware of this problem or a >> potential configuration issue that might not be compatible with the newer >> builds of FS? This wasn?t happening until the last week or two. >> >> Current version we?re running is: FreeSWITCH Version >> 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z >> >> Here?s a quick snip from the console when attempting to dial out to show >> the context of the error: >> >> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash >> >> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to >> pass zrtp-hash between legs >> >> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ >> not set, so not propagating zrtp-hash >> >> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare >> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >> >> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec >> sofia/internal/102 at 192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits >> >> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 >> sofia/internal/102 at 192.168.1.6:5060 Original read codec set to PCMU:0 >> >> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. >> Disable 2833 dtmf and switch to INFO >> >> 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports >> available! >> >> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal >> sofia/internal/102 at 192.168.1.6:5060 [BREAK] >> >> 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 >> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change RINGING -> HANGUP >> >> 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup >> sofia/internal/102 at 192.168.1.6:5060 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] >> >> Regards, >> >> Phil Quesinberry >> >> Q Systems Engineering, Inc. >> >> Electronic Controls and Embedded Systems Development >> >> (410) 969-8002 >> >> http://www.qsystemsengineering.com >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 opencsta.org Mon Jul 23 10:25:15 2012 From: chris at opencsta.org (Chris Mylonas) Date: Mon, 23 Jul 2012 16:25:15 +1000 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! In-Reply-To: References: <001901cd682d$b3e4dd70$1bae9850$@com> Message-ID: <52739C91-D2E8-43EC-BE5F-D5F5FACD30A1@opencsta.org> Sorry to but in, How many RTP ports have you got configured? If you only have a range of 5 and you have 5 concurrent calls, then no more RTP ports will be available. Doing a `netstat -anp | grep -i udp` should show what ports are open (on linux) On 23/07/2012, at 4:12 PM, curriegrad2004 wrote: > Tony, > > What can cause this error to happen really? Does FS allow re-use of > RTP ports at all given if the media isn't going to the same > destination, or doesn't the SDP header define some kind of session > tracking mechnaism so ports can be reused? > > On Sun, Jul 22, 2012 at 11:58 AM, Anthony Minessale > wrote: >> Jira Jira jira >> >> On Jul 22, 2012 12:41 PM, "Phil Quesinberry" >> wrote: >>> >>> Recently through several of the latest git pulls, FS stops routing calls >>> with the following error, usually within a few minutes of making a call >>> through Google Voice, not sure if receiving a call through GV will cause the >>> problem yet: >>> >>> sofia_glue.c:1083 No RTP ports available! >>> >>> Restarting FS fixes the problem. >>> >>> Before I file the requisite Jira, is anyone aware of this problem or a >>> potential configuration issue that might not be compatible with the newer >>> builds of FS? This wasn?t happening until the last week or two. >>> >>> Current version we?re running is: FreeSWITCH Version >>> 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z >>> >>> Here?s a quick snip from the console when attempting to dial out to show >>> the context of the error: >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to >>> pass zrtp-hash between legs >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ >>> not set, so not propagating zrtp-hash >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare >>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec >>> sofia/internal/102 at 192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 >>> sofia/internal/102 at 192.168.1.6:5060 Original read codec set to PCMU:0 >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. >>> Disable 2833 dtmf and switch to INFO >>> >>> 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports >>> available! >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal >>> sofia/internal/102 at 192.168.1.6:5060 [BREAK] >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 >>> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change RINGING -> HANGUP >>> >>> 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup >>> sofia/internal/102 at 192.168.1.6:5060 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] >>> >>> Regards, >>> >>> Phil Quesinberry >>> >>> Q Systems Engineering, Inc. >>> >>> Electronic Controls and Embedded Systems Development >>> >>> (410) 969-8002 >>> >>> http://www.qsystemsengineering.com >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From miha at softnet.si Mon Jul 23 11:37:49 2012 From: miha at softnet.si (Miha) Date: Mon, 23 Jul 2012 09:37:49 +0200 Subject: [Freeswitch-users] xml cdr with rc2 In-Reply-To: References: Message-ID: <500CFF4D.1090204@softnet.si> On 7/20/2012 7:25 PM, Michael Collins wrote: > Do you have any samples to share? Drop them on PB. > -MC > > On Fri, Jul 20, 2012 at 10:22 AM, Miha > wrote: > > > HI, > > just one questione before I post to Jira. > > Is there any cdr problem with rc2. With the same dialplan > that I have on FS verison 1.0.xxx from git (befor 1.2 > appeared) I get on RC2 different values when 302 happens. > Olso noticed that cdr for b leg are different (like on > production FS with same dialplan). > > Is that normal or has anyone also noticed? > > THanks! > > Miha > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org Hi @Michael, here is my pastebin. http://pastebin.freeswitch.org/19576 I posted log from cdr and xml. You can see that FS save different values for same variables. Regards, Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/43304332/attachment.html From miha at softnet.si Mon Jul 23 11:54:21 2012 From: miha at softnet.si (Miha) Date: Mon, 23 Jul 2012 09:54:21 +0200 Subject: [Freeswitch-users] xml cdr with rc2 In-Reply-To: <500CFF4D.1090204@softnet.si> References: <500CFF4D.1090204@softnet.si> Message-ID: <500D032D.4040602@softnet.si> I also noticed that I just get a and b leg. I should get for one call a and b leg and for other part of the call when 302 happens also a an b leg. I just get a and b leg after 302 happens. First part of the call is not logged. With the same dialplan on verison 1.0.6 I get also first part of call in my xml cdr. Is this a bug? Regards, Miha On 7/23/2012 9:37 AM, Miha wrote: > On 7/20/2012 7:25 PM, Michael Collins wrote: >> Do you have any samples to share? Drop them on PB. >> -MC >> >> On Fri, Jul 20, 2012 at 10:22 AM, Miha > > wrote: >> >> >> HI, >> >> just one questione before I post to Jira. >> >> Is there any cdr problem with rc2. With the same dialplan >> that I have on FS verison 1.0.xxx from git (befor 1.2 >> appeared) I get on RC2 different values when 302 happens. >> Olso noticed that cdr for b leg are different (like on >> production FS with same dialplan). >> >> Is that normal or has anyone also noticed? >> >> THanks! >> >> Miha >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > Hi @Michael, > > here is my pastebin. > http://pastebin.freeswitch.org/19576 > > I posted log from cdr and xml. You can see that FS save different > values for same variables. > > Regards, > Miha > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120723/b584f874/attachment.html From peter.olsson at visionutveckling.se Mon Jul 23 11:58:46 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 23 Jul 2012 07:58:46 +0000 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! In-Reply-To: References: <001901cd682d$b3e4dd70$1bae9850$@com> , Message-ID: <432B5B7B-DA11-4B91-B6F6-6C007E64D65C@visionutveckling.se> It's just an internal allocation within FS, so ports can be reused as soon as it's not allocated anymore. Lets say you have 5 ports, first call will get port 1, then 2 etc. When the first call hangs up, port 1 will be returned, and may be reused again. /Peter 23 jul 2012 kl. 08:20 skrev "curriegrad2004" : > Tony, > > What can cause this error to happen really? Does FS allow re-use of > RTP ports at all given if the media isn't going to the same > destination, or doesn't the SDP header define some kind of session > tracking mechnaism so ports can be reused? > > On Sun, Jul 22, 2012 at 11:58 AM, Anthony Minessale > wrote: >> Jira Jira jira >> >> On Jul 22, 2012 12:41 PM, "Phil Quesinberry" >> wrote: >>> >>> Recently through several of the latest git pulls, FS stops routing calls >>> with the following error, usually within a few minutes of making a call >>> through Google Voice, not sure if receiving a call through GV will cause the >>> problem yet: >>> >>> sofia_glue.c:1083 No RTP ports available! >>> >>> Restarting FS fixes the problem. >>> >>> Before I file the requisite Jira, is anyone aware of this problem or a >>> potential configuration issue that might not be compatible with the newer >>> builds of FS? This wasn?t happening until the last week or two. >>> >>> Current version we?re running is: FreeSWITCH Version >>> 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z >>> >>> Here?s a quick snip from the console when attempting to dial out to show >>> the context of the error: >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to >>> pass zrtp-hash between legs >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ >>> not set, so not propagating zrtp-hash >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare >>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec >>> sofia/internal/102 at 192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 >>> sofia/internal/102 at 192.168.1.6:5060 Original read codec set to PCMU:0 >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. >>> Disable 2833 dtmf and switch to INFO >>> >>> 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports >>> available! >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal >>> sofia/internal/102 at 192.168.1.6:5060 [BREAK] >>> >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 >>> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change RINGING -> HANGUP >>> >>> 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup >>> sofia/internal/102 at 192.168.1.6:5060 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] >>> >>> Regards, >>> >>> Phil Quesinberry >>> >>> Q Systems Engineering, Inc. >>> >>> Electronic Controls and Embedded Systems Development >>> >>> (410) 969-8002 >>> >>> http://www.qsystemsengineering.com >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:500ce9de32761973692718! > From support at sping.nl Mon Jul 23 11:29:44 2012 From: support at sping.nl (Systeembeheer) Date: Mon, 23 Jul 2012 09:29:44 +0200 Subject: [Freeswitch-users] 'Dialed number' field does not contain true dialed number? Message-ID: <500CFD68.8060603@sping.nl> Hi, Trying to get freeswitch to answer to a trunk with 10 different phone numbers. With all inbound calls however, the 'destination number' in conditions in the dialplan contains the login name of the gateway (where the trunk comes from), not the dialed number. The variable 'sip_req_user' does contain the true dialed number, but why doesn't the 'dialed number' field? Thanks! From avi at avimarcus.net Mon Jul 23 15:31:27 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 23 Jul 2012 14:31:27 +0300 Subject: [Freeswitch-users] 'Dialed number' field does not contain true dialed number? In-Reply-To: <500CFD68.8060603@sping.nl> References: <500CFD68.8060603@sping.nl> Message-ID: You want: in your gateway config. Via: http://wiki.freeswitch.org/wiki/Clarification:gateways -Avi BestFone On Mon, Jul 23, 2012 at 10:29 AM, Systeembeheer wrote: > Hi, > > Trying to get freeswitch to answer to a trunk with 10 different phone > numbers. With all inbound calls however, the 'destination number' in > conditions in the dialplan contains the login name of the gateway (where > the trunk comes from), not the dialed number. The variable > 'sip_req_user' does contain the true dialed number, but why doesn't the > 'dialed number' field? > > Thanks! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120723/7ea0d243/attachment.html From bdfoster at endigotech.com Mon Jul 23 16:17:53 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 23 Jul 2012 08:17:53 -0400 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! In-Reply-To: <52739C91-D2E8-43EC-BE5F-D5F5FACD30A1@opencsta.org> References: <001901cd682d$b3e4dd70$1bae9850$@com> <52739C91-D2E8-43EC-BE5F-D5F5FACD30A1@opencsta.org> Message-ID: Remember each call is 2 rtp ports. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 23, 2012 2:28 AM, "Chris Mylonas" wrote: > Sorry to but in, > > How many RTP ports have you got configured? > > If you only have a range of 5 and you have 5 concurrent calls, then no > more RTP ports will be available. > Doing a `netstat -anp | grep -i udp` should show what ports are open (on > linux) > > > > > On 23/07/2012, at 4:12 PM, curriegrad2004 wrote: > > > Tony, > > > > What can cause this error to happen really? Does FS allow re-use of > > RTP ports at all given if the media isn't going to the same > > destination, or doesn't the SDP header define some kind of session > > tracking mechnaism so ports can be reused? > > > > On Sun, Jul 22, 2012 at 11:58 AM, Anthony Minessale > > wrote: > >> Jira Jira jira > >> > >> On Jul 22, 2012 12:41 PM, "Phil Quesinberry" < > philq at qsystemsengineering.com> > >> wrote: > >>> > >>> Recently through several of the latest git pulls, FS stops routing > calls > >>> with the following error, usually within a few minutes of making a call > >>> through Google Voice, not sure if receiving a call through GV will > cause the > >>> problem yet: > >>> > >>> sofia_glue.c:1083 No RTP ports available! > >>> > >>> Restarting FS fixes the problem. > >>> > >>> Before I file the requisite Jira, is anyone aware of this problem or a > >>> potential configuration issue that might not be compatible with the > newer > >>> builds of FS? This wasn?t happening until the last week or two. > >>> > >>> Current version we?re running is: FreeSWITCH Version > >>> 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z > >>> > >>> Here?s a quick snip from the console when attempting to dial out to > show > >>> the context of the error: > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for > zrtp-hash > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether > to > >>> pass zrtp-hash between legs > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 > CF_ZRTP_PASSTHRU_REQ > >>> not set, so not propagating zrtp-hash > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec > Compare > >>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec > >>> sofia/internal/102 at 192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 > bits > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 > >>> sofia/internal/102 at 192.168.1.6:5060 Original read codec set to PCMU:0 > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. > >>> Disable 2833 dtmf and switch to INFO > >>> > >>> 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports > >>> available! > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send > signal > >>> sofia/internal/102 at 192.168.1.6:5060 [BREAK] > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 > >>> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change RINGING -> > HANGUP > >>> > >>> 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup > >>> sofia/internal/102 at 192.168.1.6:5060 [CS_EXECUTE] > [INCOMPATIBLE_DESTINATION] > >>> > >>> Regards, > >>> > >>> Phil Quesinberry > >>> > >>> Q Systems Engineering, Inc. > >>> > >>> Electronic Controls and Embedded Systems Development > >>> > >>> (410) 969-8002 > >>> > >>> http://www.qsystemsengineering.com > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> Join Us At ClueCon - Aug 7-9, 2012 > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120723/0e895fed/attachment-0001.html From adam.kelloway at newpace.ca Mon Jul 23 17:17:36 2012 From: adam.kelloway at newpace.ca (Adam Kelloway) Date: Mon, 23 Jul 2012 10:17:36 -0300 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! In-Reply-To: References: <001901cd682d$b3e4dd70$1bae9850$@com> <52739C91-D2E8-43EC-BE5F-D5F5FACD30A1@opencsta.org> Message-ID: <500D4EF0.8000809@newpace.ca> I have recently experienced this exact behavior. There doesn't need to be any active calls. It does seem related to the fact that an outbound call was previously made through google voice. A restart also resolves the issue for me as well. If there is already a JIRA ticket for this, can someone post the link? I can try and help by providing more testing details. On 23/07/2012 9:17 AM, Brian Foster wrote: > > Remember each call is 2 rtp ports. > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > > On Jul 23, 2012 2:28 AM, "Chris Mylonas" > wrote: > > Sorry to but in, > > How many RTP ports have you got configured? > > If you only have a range of 5 and you have 5 concurrent calls, > then no more RTP ports will be available. > Doing a `netstat -anp | grep -i udp` should show what ports are > open (on linux) > > > > > On 23/07/2012, at 4:12 PM, curriegrad2004 wrote: > > > Tony, > > > > What can cause this error to happen really? Does FS allow re-use of > > RTP ports at all given if the media isn't going to the same > > destination, or doesn't the SDP header define some kind of session > > tracking mechnaism so ports can be reused? > > > > On Sun, Jul 22, 2012 at 11:58 AM, Anthony Minessale > > > wrote: > >> Jira Jira jira > >> > >> On Jul 22, 2012 12:41 PM, "Phil Quesinberry" > > > >> wrote: > >>> > >>> Recently through several of the latest git pulls, FS stops > routing calls > >>> with the following error, usually within a few minutes of > making a call > >>> through Google Voice, not sure if receiving a call through GV > will cause the > >>> problem yet: > >>> > >>> sofia_glue.c:1083 No RTP ports available! > >>> > >>> Restarting FS fixes the problem. > >>> > >>> Before I file the requisite Jira, is anyone aware of this > problem or a > >>> potential configuration issue that might not be compatible > with the newer > >>> builds of FS? This wasn't happening until the last week or two. > >>> > >>> Current version we're running is: FreeSWITCH Version > >>> 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z > >>> > >>> Here's a quick snip from the console when attempting to dial > out to show > >>> the context of the error: > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking > for zrtp-hash > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding > whether to > >>> pass zrtp-hash between legs > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 > CF_ZRTP_PASSTHRU_REQ > >>> not set, so not propagating zrtp-hash > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio > Codec Compare > >>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec > >>> sofia/internal/102 at 192.168.1.6:5060 > PCMU/8000 20 ms 160 samples 64000 bits > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 > >>> sofia/internal/102 at 192.168.1.6:5060 > Original read codec set to PCMU:0 > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 > in SDP. > >>> Disable 2833 dtmf and switch to INFO > >>> > >>> 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports > >>> available! > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 > Send signal > >>> sofia/internal/102 at 192.168.1.6:5060 > [BREAK] > >>> > >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 > >>> (sofia/internal/102 at 192.168.1.6:5060 > ) Callstate Change RINGING -> HANGUP > >>> > >>> 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup > >>> sofia/internal/102 at 192.168.1.6:5060 > [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] > >>> > >>> Regards, > >>> > >>> Phil Quesinberry > >>> > >>> Q Systems Engineering, Inc. > >>> > >>> Electronic Controls and Embedded Systems Development > >>> > >>> (410) 969-8002 > >>> > >>> http://www.qsystemsengineering.com > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> Join Us At ClueCon - Aug 7-9, 2012 > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >>> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Adam -- NewPace Logo Adam Kelloway Software Engineer, NewPace phone +1 (902) 406--8375 x1031 email Adam.Kelloway at NewPace.com aim /msn Adam.Kelloway @NewPace.ca -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/1fd18192/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: Newpace_50x50.png Type: image/png Size: 4454 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/1fd18192/attachment-0001.png From krice at freeswitch.org Mon Jul 23 17:18:34 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 23 Jul 2012 08:18:34 -0500 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! In-Reply-To: Message-ID: Atleast 2 RTP ports... Possibly more... Also keep in mind that each RTP port requires a handle... So if you have things set to tight you?ll run into issues... Check the ulimits On 7/23/12 7:17 AM, "Brian Foster" wrote: > Remember each call is 2 rtp ports. > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > > On Jul 23, 2012 2:28 AM, "Chris Mylonas" wrote: >> Sorry to but in, >> >> How many RTP ports have you got configured? >> >> If you only have a range of 5 and you have 5 concurrent calls, then no more >> RTP ports will be available. >> Doing a `netstat -anp | grep -i udp` should show what ports are open (on >> linux) >> >> >> >> >> On 23/07/2012, at 4:12 PM, curriegrad2004 wrote: >> >>> > Tony, >>> > >>> > What can cause this error to happen really? Does FS allow re-use of >>> > RTP ports at all given if the media isn't going to the same >>> > destination, or doesn't the SDP header define some kind of session >>> > tracking mechnaism so ports can be reused? >>> > >>> > On Sun, Jul 22, 2012 at 11:58 AM, Anthony Minessale >>> > wrote: >>>> >> Jira Jira jira >>>> >> >>>> >> On Jul 22, 2012 12:41 PM, "Phil Quesinberry" >>>> >>>> >> wrote: >>>>> >>> >>>>> >>> Recently through several of the latest git pulls, FS stops routing >>>>> calls >>>>> >>> with the following error, usually within a few minutes of making a >>>>> call >>>>> >>> through Google Voice, not sure if receiving a call through GV will >>>>> cause the >>>>> >>> problem yet: >>>>> >>> >>>>> >>> sofia_glue.c:1083 No RTP ports available! >>>>> >>> >>>>> >>> Restarting FS fixes the problem. >>>>> >>> >>>>> >>> Before I file the requisite Jira, is anyone aware of this problem or a >>>>> >>> potential configuration issue that might not be compatible with the >>>>> newer >>>>> >>> builds of FS? ?This wasn?t happening until the last week or two. >>>>> >>> >>>>> >>> Current version we?re running is: ?FreeSWITCH Version >>>>> >>> 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z >>>>> >>> >>>>> >>> Here?s a quick snip from the console when attempting to dial out to >>>>> show >>>>> >>> the context of the error: >>>>> >>> >>>>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for >>>>> zrtp-hash >>>>> >>> >>>>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to >>>>> >>> pass zrtp-hash between legs >>>>> >>> >>>>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 >>>>> CF_ZRTP_PASSTHRU_REQ >>>>> >>> not set, so not propagating zrtp-hash >>>>> >>> >>>>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec >>>>> Compare >>>>> >>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >>>>> >>> >>>>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec >>>>> >>> sofia/internal/102 at 192.168.1.6:5060 >>>>> PCMU/8000 20 ms 160 samples 64000 bits >>>>> >>> >>>>> >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 >>>>> >>> sofia/internal/102 at 192.168.1.6:5060 >>>>> Original read codec set to PCMU:0 >>>>> >>> >>>>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. >>>>> >>> Disable 2833 dtmf and switch to INFO >>>>> >>> >>>>> >>> 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports >>>>> >>> available! >>>>> >>> >>>>> >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send >>>>> signal >>>>> >>> sofia/internal/102 at 192.168.1.6:5060 >>>>> [BREAK] >>>>> >>> >>>>> >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 >>>>> >>> (sofia/internal/102 at 192.168.1.6:5060 ) >>>>> Callstate Change RINGING -> HANGUP >>>>> >>> >>>>> >>> 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup >>>>> >>> sofia/internal/102 at 192.168.1.6:5060 >>>>> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] >>>>> >>> >>>>> >>> Regards, >>>>> >>> >>>>> >>> Phil Quesinberry >>>>> >>> >>>>> >>> Q Systems Engineering, Inc. >>>>> >>> >>>>> >>> Electronic Controls and Embedded Systems Development >>>>> >>> >>>>> >>> (410) 969-8002 >>>>> >>> >>>>> >>> http://www.qsystemsengineering.com >>>>> >>> >>>>> >>> >>>>> >>> >>>>> _________________________________________________________________________ >>>>> >>> Professional FreeSWITCH Consulting Services: >>>>> >>> consulting at freeswitch.org >>>>> >>> http://www.freeswitchsolutions.com >>>>> >>> >>>>> >>> >>>>> >>> >>>>> >>> >>>>> >>> Official FreeSWITCH Sites >>>>> >>> http://www.freeswitch.org >>>>> >>> http://wiki.freeswitch.org >>>>> >>> http://www.cluecon.com >>>>> >>> >>>>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>>>> >>> >>>>> >>> FreeSWITCH-users mailing list >>>>> >>> FreeSWITCH-users at lists.freeswitch.org >>>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> >>> >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> >>> http://www.freeswitch.org >>>>> >>> >>>> >> >>>> >> >>>> _________________________________________________________________________ >>>> >> Professional FreeSWITCH Consulting Services: >>>> >> consulting at freeswitch.org >>>> >> http://www.freeswitchsolutions.com >>>> >> >>>> >> >>>> >> >>>> >> >>>> >> Official FreeSWITCH Sites >>>> >> http://www.freeswitch.org >>>> >> http://wiki.freeswitch.org >>>> >> http://www.cluecon.com >>>> >> >>>> >> Join Us At ClueCon - Aug 7-9, 2012 >>>> >> >>>> >> FreeSWITCH-users mailing list >>>> >> FreeSWITCH-users at lists.freeswitch.org >>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> >> http://www.freeswitch.org >>>> >> >>> > >>> > _________________________________________________________________________ >>> > Professional FreeSWITCH Consulting Services: >>> > consulting at freeswitch.org >>> > http://www.freeswitchsolutions.com >>> > >>> > >>> > >>> > >>> > Official FreeSWITCH Sites >>> > http://www.freeswitch.org >>> > http://wiki.freeswitch.org >>> > http://www.cluecon.com >>> > >>> > Join Us At ClueCon - Aug 7-9, 2012 >>> > >>> > FreeSWITCH-users mailing list >>> > FreeSWITCH-users at lists.freeswitch.org >>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> > http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120723/7b0b7f61/attachment.html From bfmtl at hotmail.com Mon Jul 23 17:47:53 2012 From: bfmtl at hotmail.com (BF) Date: Mon, 23 Jul 2012 09:47:53 -0400 Subject: [Freeswitch-users] mod_avmd Message-ID: Hello, I'm trying to use it from a Lua script. My understanding is that mod_avmd detects beep from voicemail systems and is CPU intensive, please correct me if I'm wrong. The Lua example at FreeSWITCH Wiki is local human_detected = false; local voicemail_detected = false; function onInput(session, type, obj) if type == "dtmf" and obj['digit'] == '1' and human_detected == false then human_detected = true; return "break"; end if type == "event" and voicemail_detected == false then voicemail_detected = true; return "break"; end end session:setInputCallback("onInput"); session:execute("avmd","start"); In order to implement this example, the script must wait for the beep to be detected or not to process the case accordingly. What is no beep is detected? How can I prevent called party to hear only silence while potential beep detection is being executed? Thank you Bernard From miha at softnet.si Mon Jul 23 17:50:41 2012 From: miha at softnet.si (Miha) Date: Mon, 23 Jul 2012 15:50:41 +0200 Subject: [Freeswitch-users] xml cdr with rc2 In-Reply-To: <500D032D.4040602@softnet.si> References: <500CFF4D.1090204@softnet.si> <500D032D.4040602@softnet.si> Message-ID: <500D56B1.1020304@softnet.si> Forget aboout this email, it was my mistake, as something was missing in my configuration file:( Sorry for bothering you! On 7/23/2012 9:54 AM, Miha wrote: > I also noticed that I just get a and b leg. I should get for one call > a and b leg and for other part of the call when 302 happens also a an > b leg. I just get a and b leg after 302 happens. > First part of the call is not logged. > > With the same dialplan on verison 1.0.6 I get also first part of call > in my xml cdr. Is this a bug? > > Regards, > Miha > > On 7/23/2012 9:37 AM, Miha wrote: >> On 7/20/2012 7:25 PM, Michael Collins wrote: >>> Do you have any samples to share? Drop them on PB. >>> -MC >>> >>> On Fri, Jul 20, 2012 at 10:22 AM, Miha >> > wrote: >>> >>> >>> HI, >>> >>> just one questione before I post to Jira. >>> >>> Is there any cdr problem with rc2. With the same dialplan >>> that I have on FS verison 1.0.xxx from git (befor 1.2 >>> appeared) I get on RC2 different values when 302 happens. >>> Olso noticed that cdr for b leg are different (like on >>> production FS with same dialplan). >>> >>> Is that normal or has anyone also noticed? >>> >>> THanks! >>> >>> Miha >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> Hi @Michael, >> >> here is my pastebin. >> http://pastebin.freeswitch.org/19576 >> >> I posted log from cdr and xml. You can see that FS save different >> values for same variables. >> >> Regards, >> Miha >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120723/9507aa2f/attachment-0001.html From Hector.Geraldino at ipsoft.com Mon Jul 23 18:18:48 2012 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Mon, 23 Jul 2012 10:18:48 -0400 Subject: [Freeswitch-users] A fatal error has been detected by the Java Runtime Environment: In-Reply-To: References: <1342529673372-7580878.post@n2.nabble.com> <6A6B4C284AD15042B429EB9D904544AD022FAE2819@NY1-EXMB-01.ip-soft.net> Message-ID: <6A6B4C284AD15042B429EB9D904544AD022FAE2AB7@NY1-EXMB-01.ip-soft.net> Sounds to me like multiple calls to client.addEventListener. Can you try to make it work by modifying the ClientTest.java class (org.freeswitch.esl.client.inbound.ClientTest)? Please note also that you can pass all events you want to filter while setting the subscriptions, in the form: client.setEventSubscriptions("plain", "CHANNEL_ANSWER DTMF CHANNEL_HANGUP"); It's recommended to set the filter to exactly the number of events you want to be notified for, no more, no less, so probably you should skip adding events like HEARTBEAT, BACKGROUND_JOB, to your filters. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of shaik bawajan Sent: Saturday, July 21, 2012 2:45 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] A fatal error has been detected by the Java Runtime Environment: Thanks a lot, It is very helpful to me. But, the each event are receiving multiple times ( each event coming 3 or 4 times ). It will happen like this or is there any way to stop this. here am attaching my java class, where am creating a single thread and making outbound, playing a file. The class itself is a Event Listener and am adding below event listeners from other initiate Class. client.setEventSubscriptions("plain", "all"); client.addEventFilter("Event-Name", "CHANNEL_CREATE"); client.addEventFilter("Event-Name","BACKGROUND_JOB"); client.addEventFilter("Event-Name","CHANNEL_STATE"); client.addEventFilter("Event-Name","CHANNEL_EXECUTE_COMPLETE"); client.addEventFilter("Event-Name","CHANNEL_HANGUP"); client.addEventFilter("Event-Name","CHANNEL_HANGUP_COMPLETE"); client.addEventFilter("Event-Name","DTMF"); client.addEventFilter("Event-Name","HEARTBEAT"); Thanks in advance, On Wed, Jul 18, 2012 at 7:30 PM, Hector Geraldino > wrote: Hello, Trying to help you to solve a crash in the jvm for a multithreaded application is damn hard. Doing it using a mailing list is even harder, and without looking at your source code is almost impossible. However I want to recommend you to drop the use of this library (which is a java wrapper of the FS core lib written in C) and use the pure Java ESL Client (http://wiki.freeswitch.org/wiki/Java_ESL_Client). You will have full access to the source code for debug, no dependencies on native libraries, and a good set of examples. Good luck! -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of bawajan Sent: Tuesday, July 17, 2012 8:55 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] A fatal error has been detected by the Java Runtime Environment: Hi, Am using ESL inbound connection to make calls and have a below flow (written in java) a) originating 15 calls simultaneously with park function b) play audio file c) creating new thread and passing eslconnection and playing a IVR in originate call. Here while handling events getting the below error : # A fatal error has been detected by the Java Runtime Environment: INFO | jvm 3 | 2012/07/17 17:54:36 | # INFO | jvm 3 | 2012/07/17 17:54:36 | # SIGSEGV (0xb) at pc=0x00007fc01f1d3943, pid=14759, tid=140463084390144 INFO | jvm 3 | 2012/07/17 17:54:36 | # INFO | jvm 3 | 2012/07/17 17:54:36 | # JRE version: 6.0_25-b06 INFO | jvm 3 | 2012/07/17 17:54:36 | # Java VM: Java HotSpot(TM) 64-Bit Server VM (20.0-b11 mixed mode linux-amd64 compressed oops) INFO | jvm 3 | 2012/07/17 17:54:36 | # Problematic frame: INFO | jvm 3 | 2012/07/17 17:54:36 | # C [libesljni.so+0xd943] long double+0x183 INFO | jvm 3 | 2012/07/17 17:54:36 | # INFO | jvm 3 | 2012/07/17 17:54:36 | # An error report file with more information is saved as: INFO | jvm 3 | 2012/07/17 17:54:36 | # /usr/local/freeswitch/hs_err_pid14759.log INFO | jvm 3 | 2012/07/17 17:54:36 | # INFO | jvm 3 | 2012/07/17 17:54:36 | # If you would like to submit a bug report, please visit: INFO | jvm 3 | 2012/07/17 17:54:36 | # http://java.sun.com/webapps/bugreport/crash.jsp INFO | jvm 3 | 2012/07/17 17:54:36 | # The crash happened outside the Java Virtual Machine in native code. INFO | jvm 3 | 2012/07/17 17:54:36 | # See problematic frame for where to report the bug. plz let me know, where am doing mistake and how to resolve it. Thanks in advance. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/A-fatal-error-has-been-detected-by-the-Java-Runtime-Environment-tp7580878.html Sent from the freeswitch-users mailing list archive at Nabble.com. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120723/0ad7d72c/attachment.html From chmp99 at gmail.com Mon Jul 23 19:39:44 2012 From: chmp99 at gmail.com (=?ISO-8859-1?Q?Germ=E1n_Ruiz?=) Date: Mon, 23 Jul 2012 12:39:44 -0300 Subject: [Freeswitch-users] RTMP user not registered error Message-ID: <500D7040.5090100@gmail.com> Hi. I 'm new with FreeSwitch and I 'm trying to make work the RTMP module in both directions. Now, I can do a call from an user registered in the flash application (using the example client) to an user connected with a SIP client (desktop or android application). But when I call from a SIP client to another user connected with RTMP, get error: mod_dptools.c:2960 Originate Failed. Cause: USER_NOT_REGISTERED I used the configuration example for RTMP (autoload_configs/rtmp.conf.xml). In the dialplan configuration, the bridge option applied is: ?What is missing to register the users connected with RTMP? Thanks and sorry for my bad english. Germ?n From msc at freeswitch.org Mon Jul 23 19:53:08 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Jul 2012 08:53:08 -0700 Subject: [Freeswitch-users] dialplan commands on playback In-Reply-To: <5117A1BD-A549-4091-8F0F-577DE29C747C@gmail.com> References: <5117A1BD-A549-4091-8F0F-577DE29C747C@gmail.com> Message-ID: If you know the length of the file then you could use sched_api to schedule something. For example, if you know that your file is 12 seconds long you could do something like this: sched_api +9 none uuid_broadcast :: bleg Check out both of these on the wiki - there's a lot that these can do. -MC http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast http://wiki.freeswitch.org/wiki/Mod_commands#sched_api -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/c20d5247/attachment.html From msc at freeswitch.org Mon Jul 23 19:59:42 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Jul 2012 08:59:42 -0700 Subject: [Freeswitch-users] mod_avmd In-Reply-To: References: Message-ID: Bernard, Keep in mind that avmd is non-blocking, that is, it won't cause your dialplan or script to pause while it is attempting to detect the beep. Your dialplan will keep doing what it normally does, and if avmd detects a beep then it will throw an event which you catch and handle in your onInput function. -MC On Mon, Jul 23, 2012 at 6:47 AM, BF wrote: > Hello, > > I'm trying to use it from a Lua script. My understanding is that mod_avmd > detects beep from voicemail systems and is CPU intensive, please correct me > if I'm wrong. > > The Lua example at FreeSWITCH Wiki is > local human_detected = false; > local voicemail_detected = false; > > function onInput(session, type, obj) > if type == "dtmf" and obj['digit'] == '1' and human_detected == false > then > human_detected = true; > return "break"; > end > > if type == "event" and voicemail_detected == false then > voicemail_detected = true; > return "break"; > end > end > > session:setInputCallback("onInput"); > session:execute("avmd","start"); > In order to implement this example, the script must wait for the beep to > be detected or not to process the case accordingly. What is no beep is > detected? How can I prevent called party to hear only silence while > potential beep detection is being executed? > > Thank you > > Bernard > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/fd90f783/attachment-0001.html From fiorix at gmail.com Mon Jul 23 20:09:26 2012 From: fiorix at gmail.com (Alexandre Fiori) Date: Mon, 23 Jul 2012 12:09:26 -0400 Subject: [Freeswitch-users] dialplan commands on playback In-Reply-To: References: <5117A1BD-A549-4091-8F0F-577DE29C747C@gmail.com> Message-ID: <7AB73CC2-06A0-46E2-BA18-8603C208FADD@gmail.com> that's the problem, I don't know the length, but playback does :) thanks anyway On 2012-07-23, at 11:53 AM, Michael Collins wrote: > If you know the length of the file then you could use sched_api to schedule something. For example, if you know that your file is 12 seconds long you could do something like this: > > sched_api +9 none uuid_broadcast :: bleg > > Check out both of these on the wiki - there's a lot that these can do. > -MC > > http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast > http://wiki.freeswitch.org/wiki/Mod_commands#sched_api > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org - af -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/72f6e7b5/attachment.html From anthony.minessale at gmail.com Mon Jul 23 20:37:09 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Jul 2012 11:37:09 -0500 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! In-Reply-To: <500D4EF0.8000809@newpace.ca> References: <001901cd682d$b3e4dd70$1bae9850$@com> <52739C91-D2E8-43EC-BE5F-D5F5FACD30A1@opencsta.org> <500D4EF0.8000809@newpace.ca> Message-ID: commit 524468be7ba5ee0887d2e975e4f131171c601a54 Author: Anthony Minessale Date: Mon Jul 23 11:36:19 2012 -0500 FS-4373 --resolve this is probably the same issue nobody would file a bug about on the mailing list even after I begged them to from last weekend please please stop filing bugs on the mailing list. I will still work on them and try to fix them but you are making me do way more work. Unless everyone wants to pay for support I cannot keep this up. On Mon, Jul 23, 2012 at 8:17 AM, Adam Kelloway wrote: > I have recently experienced this exact behavior. There doesn't need to be > any active calls. It does seem related to the fact that an outbound call > was previously made through google voice. A restart also resolves the issue > for me as well. > > If there is already a JIRA ticket for this, can someone post the link? I > can try and help by providing more testing details. > > > On 23/07/2012 9:17 AM, Brian Foster wrote: > > Remember each call is 2 rtp ports. > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > On Jul 23, 2012 2:28 AM, "Chris Mylonas" wrote: > >> Sorry to but in, >> >> How many RTP ports have you got configured? >> >> If you only have a range of 5 and you have 5 concurrent calls, then no >> more RTP ports will be available. >> Doing a `netstat -anp | grep -i udp` should show what ports are open (on >> linux) >> >> >> >> >> On 23/07/2012, at 4:12 PM, curriegrad2004 wrote: >> >> > Tony, >> > >> > What can cause this error to happen really? Does FS allow re-use of >> > RTP ports at all given if the media isn't going to the same >> > destination, or doesn't the SDP header define some kind of session >> > tracking mechnaism so ports can be reused? >> > >> > On Sun, Jul 22, 2012 at 11:58 AM, Anthony Minessale >> > wrote: >> >> Jira Jira jira >> >> >> >> On Jul 22, 2012 12:41 PM, "Phil Quesinberry" < >> philq at qsystemsengineering.com> >> >> wrote: >> >>> >> >>> Recently through several of the latest git pulls, FS stops routing >> calls >> >>> with the following error, usually within a few minutes of making a >> call >> >>> through Google Voice, not sure if receiving a call through GV will >> cause the >> >>> problem yet: >> >>> >> >>> sofia_glue.c:1083 No RTP ports available! >> >>> >> >>> Restarting FS fixes the problem. >> >>> >> >>> Before I file the requisite Jira, is anyone aware of this problem or a >> >>> potential configuration issue that might not be compatible with the >> newer >> >>> builds of FS? This wasn?t happening until the last week or two. >> >>> >> >>> Current version we?re running is: FreeSWITCH Version >> >>> 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z >> >>> >> >>> Here?s a quick snip from the console when attempting to dial out to >> show >> >>> the context of the error: >> >>> >> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for >> zrtp-hash >> >>> >> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether >> to >> >>> pass zrtp-hash between legs >> >>> >> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 >> CF_ZRTP_PASSTHRU_REQ >> >>> not set, so not propagating zrtp-hash >> >>> >> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec >> Compare >> >>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >> >>> >> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec >> >>> sofia/internal/102 at 192.168.1.6:5060 PCMU/8000 20 ms 160 samples >> 64000 bits >> >>> >> >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 >> >>> sofia/internal/102 at 192.168.1.6:5060 Original read codec set to PCMU:0 >> >>> >> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. >> >>> Disable 2833 dtmf and switch to INFO >> >>> >> >>> 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports >> >>> available! >> >>> >> >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send >> signal >> >>> sofia/internal/102 at 192.168.1.6:5060 [BREAK] >> >>> >> >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 >> >>> (sofia/internal/102 at 192.168.1.6:5060) Callstate Change RINGING -> >> HANGUP >> >>> >> >>> 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup >> >>> sofia/internal/102 at 192.168.1.6:5060 [CS_EXECUTE] >> [INCOMPATIBLE_DESTINATION] >> >>> >> >>> Regards, >> >>> >> >>> Phil Quesinberry >> >>> >> >>> Q Systems Engineering, Inc. >> >>> >> >>> Electronic Controls and Embedded Systems Development >> >>> >> >>> (410) 969-8002 <%28410%29%20969-8002> >> >>> >> >>> http://www.qsystemsengineering.com >> >>> >> >>> >> >>> >> _________________________________________________________________________ >> >>> Professional FreeSWITCH Consulting Services: >> >>> consulting at freeswitch.org >> >>> http://www.freeswitchsolutions.com >> >>> >> >>> >> >>> >> >>> >> >>> Official FreeSWITCH Sites >> >>> http://www.freeswitch.org >> >>> http://wiki.freeswitch.org >> >>> http://www.cluecon.com >> >>> >> >>> Join Us At ClueCon - Aug 7-9, 2012 >> >>> >> >>> FreeSWITCH-users mailing list >> >>> FreeSWITCH-users at lists.freeswitch.org >> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >>> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >>> http://www.freeswitch.org >> >>> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> consulting at freeswitch.org >> >> http://www.freeswitchsolutions.com >> >> >> >> >> >> >> >> >> >> Official FreeSWITCH Sites >> >> http://www.freeswitch.org >> >> http://wiki.freeswitch.org >> >> http://www.cluecon.com >> >> >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> >> >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -- > Adam > -- > [image: NewPace Logo] > > > Adam Kelloway > Software Engineer, NewPace phone +1 (902) 406?8375 x1031 email > Adam.Kelloway at NewPace.com aim/msn > Adam.Kelloway at NewPace.ca > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/05566885/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/png Size: 4454 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/05566885/attachment-0001.png From fluixab at bellsouth.net Mon Jul 23 20:59:44 2012 From: fluixab at bellsouth.net (Bernard Fluixa) Date: Mon, 23 Jul 2012 12:59:44 -0400 Subject: [Freeswitch-users] mod_avmd In-Reply-To: References: Message-ID: <4211B6CA-C067-4B6D-B271-CC1BB82C0206@bellsouth.net> Michael, Thank you or your response. So it is my responsibility to do whatever needs to be done while mod_avmd attempts to detect a beep and to manually stop it after a beep as been detected or after a certain timeout. Correct? Bernard On Jul 23, 2012, at 11:59 AM, Michael Collins wrote: > Bernard, > > Keep in mind that avmd is non-blocking, that is, it won't cause your dialplan or script to pause while it is attempting to detect the beep. Your dialplan will keep doing what it normally does, and if avmd detects a beep then it will throw an event which you catch and handle in your onInput function. > > -MC > > On Mon, Jul 23, 2012 at 6:47 AM, BF wrote: > Hello, > > I'm trying to use it from a Lua script. My understanding is that mod_avmd detects beep from voicemail systems and is CPU intensive, please correct me if I'm wrong. > > The Lua example at FreeSWITCH Wiki is > local human_detected = false; > local voicemail_detected = false; > > function onInput(session, type, obj) > if type == "dtmf" and obj['digit'] == '1' and human_detected == false then > human_detected = true; > return "break"; > end > > if type == "event" and voicemail_detected == false then > voicemail_detected = true; > return "break"; > end > end > > session:setInputCallback("onInput"); > session:execute("avmd","start"); > In order to implement this example, the script must wait for the beep to be detected or not to process the case accordingly. What is no beep is detected? How can I prevent called party to hear only silence while potential beep detection is being executed? > > Thank you > > Bernard > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120723/9775ec25/attachment.html From getj2k at yahoo.com Mon Jul 23 21:02:22 2012 From: getj2k at yahoo.com (mtle) Date: Mon, 23 Jul 2012 10:02:22 -0700 (PDT) Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! In-Reply-To: <500D4EF0.8000809@newpace.ca> Message-ID: <1343062942.42529.YahooMailClassic@web161505.mail.bf1.yahoo.com> So do I, since the last rebuild from git a few days ago.?In autoload_configs/switch.conf.xml, I have:? ? ? ? ? ? and iptables:-A INPUT -i eth0 -p udp -m udp --dport 10000:20000 -j ACCEPT The 'netstat' command returns about 15 open UDP ports. --- On Mon, 7/23/12, Adam Kelloway wrote: From: Adam Kelloway Subject: Re: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! To: "FreeSWITCH Users Help" Date: Monday, July 23, 2012, 6:17 AM I have recently experienced this exact behavior. There doesn't need to be any active calls. It does seem related to the fact that an outbound call was previously made through google voice. A restart also resolves the issue for me as well. If there is already a JIRA ticket for this, can someone post the link? I can try and help by providing more testing details. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/71ec8ec0/attachment.html From krice at freeswitch.org Mon Jul 23 21:12:45 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 23 Jul 2012 12:12:45 -0500 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! In-Reply-To: Message-ID: Hey Tony, Thanks for fixing this... However, Everyone else, Open the freaking Jira, it doesn?t take that long... Or take the 2 minutes to search jira to see if theres an existing ticket for your problem, and if you have more information add it... Why? It saves Developers like Tony time in tracking down the issue and getting you a fix... As much as I hate to say it, there could come a time where No Jira ticket == Its not really a problem... You think tony hasn?t spent a crazy amount of time on FreeSWITCH? Lets do a little math... 50 weeks a year (he?s supposed to get 2 weeks of holiday a year right?) and I know he works at least 40 hours a week on FreeSWITCH so that?s 2000 hours a Year, then factor in that he has been working on it since atleast 2006... So that?s 12000 he has spent working on FreeSWITCH. And that?s at the low end of the estimate... Ohloh.net estimates something like $40million dollars worth of developer time has gone into FreeSWITCH ( http://www.ohloh.net/p/freeswitch/estimated_cost ) most of which is the developers doing their part for the love of the project (ie: not getting paid a dime)... So come on people... Just a little help... That?s all we are asking for here.... On 7/23/12 11:37 AM, "Anthony Minessale" wrote: > commit 524468be7ba5ee0887d2e975e4f131171c601a54 > Author: Anthony Minessale > Date: ? Mon Jul 23 11:36:19 2012 -0500 > > ? ? FS-4373 --resolve this is probably the same issue nobody would file a bug > about on the mailing list even after I begged them to from last weekend > > > please please stop filing bugs on the mailing list. ?I will still work on them > and try to fix them but you are making me do way more work. > Unless everyone wants to pay for support I cannot keep this up. > > > On Mon, Jul 23, 2012 at 8:17 AM, Adam Kelloway > wrote: >> >> I have recently experienced this exact behavior. There doesn't need to be >> any active calls. It does seem related to the fact that an outbound call was >> previously made through google voice. A restart also resolves the issue for >> me as well. >> >> If there is already a JIRA ticket for this, can someone post the link? I can >> try and help by providing more testing details. >> >> >> >> On 23/07/2012 9:17 AM, Brian Foster wrote: >> >> >>> >>> >>> Remember each call is 2 rtp ports. >>> >>> >>> Brian Foster >>> Endigo Computer LLC >>> >>> >>> Sent from a mobile device. >>> >>> On Jul 23, 2012 2:28 AM, "Chris Mylonas" wrote: >>> >>>> Sorry to but in, >>>> >>>> How many RTP ports have you got configured? >>>> >>>> If you only have a range of 5 and you have 5 concurrent calls, then no >>>> more RTP ports will be available. >>>> Doing a `netstat -anp | grep -i udp` should show what ports are open (on >>>> linux) >>>> >>>> >>>> >>>> >>>> On 23/07/2012, at 4:12 PM, curriegrad2004 wrote: >>>> >>>>> > Tony, >>>>> > >>>>> > What can cause this error to happen really? Does FS allow re-use of >>>>> > RTP ports at all given if the media isn't going to the same >>>>> > destination, or doesn't the SDP header define some kind of session >>>>> > tracking mechnaism so ports can be reused? >>>>> > >>>>> > On Sun, Jul 22, 2012 at 11:58 AM, Anthony Minessale >>>>> > wrote: >>>>>> >> Jira Jira jira >>>>>> >> >>>>>> >> On Jul 22, 2012 12:41 PM, "Phil Quesinberry" >>>>>> >>>>>> >> wrote: >>>>>>> >>> >>>>>>> >>> Recently through several of the latest git pulls, FS stops routing >>>>>>> calls >>>>>>> >>> with the following error, usually within a few minutes of making a call >>>>>>> >>> through Google Voice, not sure if receiving a call through GV will >>>>>>> cause the >>>>>>> >>> problem yet: >>>>>>> >>> >>>>>>> >>> sofia_glue.c:1083 No RTP ports available! >>>>>>> >>> >>>>>>> >>> Restarting FS fixes the problem. >>>>>>> >>> >>>>>>> >>> Before I file the requisite Jira, is anyone aware of this problem or a >>>>>>> >>> potential configuration issue that might not be compatible with the >>>>>>> newer >>>>>>> >>> builds of FS? ?This wasn?t happening until the last week or two. >>>>>>> >>> >>>>>>> >>> Current version we?re running is: ?FreeSWITCH Version >>>>>>> >>> 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z >>>>>>> >>> >>>>>>> >>> Here?s a quick snip from the console when attempting to dial out to show >>>>>>> >>> the context of the error: >>>>>>> >>> >>>>>>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for >>>>>>> zrtp-hash >>>>>>> >>> >>>>>>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding >>>>>>> whether to >>>>>>> >>> pass zrtp-hash between legs >>>>>>> >>> >>>>>>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 >>>>>>> CF_ZRTP_PASSTHRU_REQ >>>>>>> >>> not set, so not propagating zrtp-hash >>>>>>> >>> >>>>>>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec >>>>>>> Compare >>>>>>> >>> [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] >>>>>>> >>> >>>>>>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec >>>>>>> >>> sofia/internal/102 at 192.168.1.6:5060 >>>>>>> PCMU/8000 20 ms 160 samples 64000 bits >>>>>>> >>> >>>>>>> >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 >>>>>>> >>> sofia/internal/102 at 192.168.1.6:5060 >>>>>>> Original read codec set to PCMU:0 >>>>>>> >>> >>>>>>> >>> 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. >>>>>>> >>> Disable 2833 dtmf and switch to INFO >>>>>>> >>> >>>>>>> >>> 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports >>>>>>> >>> available! >>>>>>> >>> >>>>>>> >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send >>>>>>> signal >>>>>>> >>> sofia/internal/102 at 192.168.1.6:5060 >>>>>>> [BREAK] >>>>>>> >>> >>>>>>> >>> 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 >>>>>>> >>> (sofia/internal/102 at 192.168.1.6:5060 >>>>>>> ) Callstate Change RINGING -> HANGUP >>>>>>> >>> >>>>>>> >>> 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup >>>>>>> >>> sofia/internal/102 at 192.168.1.6:5060 >>>>>>> [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] >>>>>>> >>> >>>>>>> >>> Regards, >>>>>>> >>> >>>>>>> >>> Phil Quesinberry >>>>>>> >>> >>>>>>> >>> Q Systems Engineering, Inc. >>>>>>> >>> >>>>>>> >>> Electronic Controls and Embedded Systems Development >>>>>>> >>> >>>>>>> >>> (410) 969-8002 >>>>>>> >>> >>>>>>> >>> http://www.qsystemsengineering.com >>>>>>> >>> >>>>>>> >>> >>>>>>> >>> >>>>>>> _________________________________________________________________________ >>>>>>> >>> Professional FreeSWITCH Consulting Services: >>>>>>> >>> consulting at freeswitch.org >>>>>>> >>> http://www.freeswitchsolutions.com >>>>>>> >>> >>>>>>> >>> >>>>>>> >>> >>>>>>> >>> >>>>>>> >>> Official FreeSWITCH Sites >>>>>>> >>> http://www.freeswitch.org >>>>>>> >>> http://wiki.freeswitch.org >>>>>>> >>> http://www.cluecon.com >>>>>>> >>> >>>>>>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>>>>>> >>> >>>>>>> >>> FreeSWITCH-users mailing list >>>>>>> >>> FreeSWITCH-users at lists.freeswitch.org >>>>>>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>>> >>> >>>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>>> >>> http://www.freeswitch.org >>>>>>> >>> >>>>>> >> >>>>>> >> >>>>>> _________________________________________________________________________ >>>>>> >> Professional FreeSWITCH Consulting Services: >>>>>> >> consulting at freeswitch.org >>>>>> >> http://www.freeswitchsolutions.com >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> >>>>>> >> Official FreeSWITCH Sites >>>>>> >> http://www.freeswitch.org >>>>>> >> http://wiki.freeswitch.org >>>>>> >> http://www.cluecon.com >>>>>> >> >>>>>> >> Join Us At ClueCon - Aug 7-9, 2012 >>>>>> >> >>>>>> >> FreeSWITCH-users mailing list >>>>>> >> FreeSWITCH-users at lists.freeswitch.org >>>>>> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> >> >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> >> http://www.freeswitch.org >>>>>> >> >>>>> > >>>>> > >>>>> _________________________________________________________________________ >>>>> > Professional FreeSWITCH Consulting Services: >>>>> > consulting at freeswitch.org >>>>> > http://www.freeswitchsolutions.com >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > Official FreeSWITCH Sites >>>>> > http://www.freeswitch.org >>>>> > http://wiki.freeswitch.org >>>>> > http://www.cluecon.com >>>>> > >>>>> > Join Us At ClueCon - Aug 7-9, 2012 >>>>> > >>>>> > FreeSWITCH-users mailing list >>>>> > FreeSWITCH-users at lists.freeswitch.org >>>>> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> > >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> > http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at 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/20120723/e35182b8/attachment-0001.html From msc at freeswitch.org Mon Jul 23 21:17:13 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Jul 2012 10:17:13 -0700 Subject: [Freeswitch-users] mod_avmd In-Reply-To: <4211B6CA-C067-4B6D-B271-CC1BB82C0206@bellsouth.net> References: <4211B6CA-C067-4B6D-B271-CC1BB82C0206@bellsouth.net> Message-ID: Precisely. -MC On Mon, Jul 23, 2012 at 9:59 AM, Bernard Fluixa wrote: > Michael, > > Thank you or your response. So it is my responsibility to do whatever > needs to be done while mod_avmd attempts to detect a beep and to manually > stop it after a beep as been detected or after a certain timeout. Correct? > > Bernard > > > > > On Jul 23, 2012, at 11:59 AM, Michael Collins wrote: > > Bernard, > > Keep in mind that avmd is non-blocking, that is, it won't cause your > dialplan or script to pause while it is attempting to detect the beep. Your > dialplan will keep doing what it normally does, and if avmd detects a beep > then it will throw an event which you catch and handle in your onInput > function. > > -MC > > On Mon, Jul 23, 2012 at 6:47 AM, BF wrote: > >> Hello, >> >> I'm trying to use it from a Lua script. My understanding is that mod_avmd >> detects beep from voicemail systems and is CPU intensive, please correct me >> if I'm wrong. >> >> The Lua example at FreeSWITCH Wiki is >> local human_detected = false; >> local voicemail_detected = false; >> >> function onInput(session, type, obj) >> if type == "dtmf" and obj['digit'] == '1' and human_detected == false >> then >> human_detected = true; >> return "break"; >> end >> >> if type == "event" and voicemail_detected == false then >> voicemail_detected = true; >> return "break"; >> end >> end >> >> session:setInputCallback("onInput"); >> session:execute("avmd","start"); >> In order to implement this example, the script must wait for the beep to >> be detected or not to process the case accordingly. What is no beep is >> detected? How can I prevent called party to hear only silence while >> potential beep detection is being executed? >> >> Thank you >> >> Bernard >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/2b4e7989/attachment.html From brian at freeswitch.org Mon Jul 23 21:37:32 2012 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Jul 2012 12:37:32 -0500 Subject: [Freeswitch-users] Bria iOS does SRTP wrong. Message-ID: <28D6D8F1-8758-4EEC-AFCA-B971BAD1CCAA@freeswitch.org> Dear FreeSWITCHologists, Anyone that uses Bria on iOS, Please open a support case with CounterPath. I've tried, and they need to not offer a=crypto in an RTP/AVP and offer two m= lines one for RTP/AVP and RTP/SAVP in order of preference. They seem to think its not a critical thing to do SRTP the right way. (Guessing its not just SRTP LULZ!) Anyone see the refer to rfc3711 CRIT message? Why do I care so much... lolz -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/53e0011c/attachment.html From marketing at cluecon.com Mon Jul 23 21:41:55 2012 From: marketing at cluecon.com (Michael Collins) Date: Mon, 23 Jul 2012 10:41:55 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: Hello all! Last week we enjoyed a nice conference call with Darren Schreiberfrom 2600hz . Darren gave us a follow uppresentation to the original SIP 101 discussionthat was held in June. This was a great presentation that went beyond the basics of SIP. We discussed important concepts like Via and Record Route headers and NOTIFY messages. SIP can do a lot, and with these two presentations we really have only scratched the surface. If we all bug Darren enough then perhaps we'll get a third presentation! The audio of the presentation is in the usual placeand the slides will be available shortly. This week we will be hearing from the community. We will have a brief report from Dave Kompel who will get us up to speed on using Windows Advanced Firewall to implement Fail2Ban-like functionality. After Dave speaks we will be opening things up for community discussion. Please bring your questions and topics! ClueCon 2012 is shaping up nicely! We have recently added Ditech Networks , Bandwidth.com , and Yealinkas sponsors. We are also happy to announce that Darrell Hensley will be speaking on the subject of Transitioning to Professional Voice. Darrell is the CEO of GMVoices , a ClueCon media sponsor for many years and the supplier of Callie, the voice of FreeSWITCH. We look forward to seeing Darrell in person again. As a quick reminder, if you registerby the end of the day on July 25th you will still receive four chances to win in the great ClueCon giveaway. See you in two weeks! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0723 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/7a710848/attachment-0001.html From brian at freeswitch.org Mon Jul 23 22:01:23 2012 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Jul 2012 13:01:23 -0500 Subject: [Freeswitch-users] TLS and connection reuse In-Reply-To: References: Message-ID: <313B0898-293C-4A4F-8262-9DF896CBEF91@freeswitch.org> It should already be doing this. I suspect your contact port != real network port the request came from. Sofia can't find and reuse the connection if I recall correctly. On Jul 22, 2012, at 2:31 PM, Tihomir Culjaga wrote: > hello, > > im just wondering if it is possible to make FS re-use the existing TLS > connection established on UA registratioin for incoming calls to UA? > > > as an example... i have extensions 1002 (uses TLS) and 1009 (uses SIP/UDP). > Both extensions are registered to the same FS. I can place calls from 1002 > everywhere but 1002 cannot get any calls. > > http://pastebin.freeswitch.org/19575 -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/5d83e858/attachment.html From brian at freeswitch.org Mon Jul 23 22:02:32 2012 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Jul 2012 13:02:32 -0500 Subject: [Freeswitch-users] ZRTP - SAS-strings not the same In-Reply-To: <20120719140122.17800@gmx.net> References: <20120719140122.17800@gmx.net> Message-ID: <1DCC1BDA-EB23-4B44-820C-445910E7EAF6@freeswitch.org> Technically you dont even need --enable-zrtp if you just want end to end ZRTP and requires you to have a client that has the zrtp-hash in the sdp and 200ok from the far side. Otherwise Trusted MiTM gets in the way. If either end isn't enrolled the SAS will not match. /b On Jul 19, 2012, at 9:01 AM, Finn H?ck wrote: > Hi, > I got freeswitch working perfectly on CentOS 6.3 by following the guide on http://wiki.freeswitch.org/wiki/Installation_Guide. Then I followed the steps listet at http://wiki.freeswitch.org/wiki/ZRTP to get ZRTP working. Now, when I do calls from a zrtp-enabled client to another, communication still works BUT the sas-strings showing up on the clients aren't the same. Now my question is: why? Does anyone have some troubleshooting-tips for me? What possible reasons are there? > > Thanks in advance for any help. > Greetings, Finn -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/c6191518/attachment.html From mario_fs at mgtech.com Mon Jul 23 22:16:40 2012 From: mario_fs at mgtech.com (Mario G) Date: Mon, 23 Jul 2012 11:16:40 -0700 Subject: [Freeswitch-users] Bria iOS does SRTP wrong. In-Reply-To: <28D6D8F1-8758-4EEC-AFCA-B971BAD1CCAA@freeswitch.org> References: <28D6D8F1-8758-4EEC-AFCA-B971BAD1CCAA@freeswitch.org> Message-ID: <1473A507-8090-4BA8-B575-5D7CD0FB52A2@mgtech.com> Will do, BTW, I already have 4-5 other issues open with them. Also, note that Bria for iPad 2 and iPad 3 have different problems. Bria is really nice when it works though. But the reliability (or lack of) does not allow it to replace desk phones yet. Mario G On Jul 23, 2012, at 10:37 AM, Brian West wrote: > Dear FreeSWITCHologists, > Anyone that uses Bria on iOS, Please open a support case with CounterPath. I've tried, and they need to not offer a=crypto in an RTP/AVP and offer two m= lines one for RTP/AVP and RTP/SAVP in order of preference. They seem to think its not a critical thing to do SRTP the right way. (Guessing its not just SRTP LULZ!) > > Anyone see the refer to rfc3711 CRIT message? > > Why do I care so much... lolz > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST > iNUM: +883 5100 1286 0410 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120723/9d00826f/attachment.html From fluixab at bellsouth.net Mon Jul 23 22:16:36 2012 From: fluixab at bellsouth.net (Bernard Fluixa) Date: Mon, 23 Jul 2012 14:16:36 -0400 Subject: [Freeswitch-users] mod_avmd In-Reply-To: References: <4211B6CA-C067-4B6D-B271-CC1BB82C0206@bellsouth.net> Message-ID: <0C16CD66-3751-4946-90EC-8AEB03F6F06F@bellsouth.net> OK. I'm clear now. Thanks again. On Jul 23, 2012, at 1:17 PM, Michael Collins wrote: > Precisely. > -MC > > On Mon, Jul 23, 2012 at 9:59 AM, Bernard Fluixa wrote: > Michael, > > Thank you or your response. So it is my responsibility to do whatever needs to be done while mod_avmd attempts to detect a beep and to manually stop it after a beep as been detected or after a certain timeout. Correct? > > Bernard > > > > > On Jul 23, 2012, at 11:59 AM, Michael Collins wrote: > >> Bernard, >> >> Keep in mind that avmd is non-blocking, that is, it won't cause your dialplan or script to pause while it is attempting to detect the beep. Your dialplan will keep doing what it normally does, and if avmd detects a beep then it will throw an event which you catch and handle in your onInput function. >> >> -MC >> >> On Mon, Jul 23, 2012 at 6:47 AM, BF wrote: >> Hello, >> >> I'm trying to use it from a Lua script. My understanding is that mod_avmd detects beep from voicemail systems and is CPU intensive, please correct me if I'm wrong. >> >> The Lua example at FreeSWITCH Wiki is >> local human_detected = false; >> local voicemail_detected = false; >> >> function onInput(session, type, obj) >> if type == "dtmf" and obj['digit'] == '1' and human_detected == false then >> human_detected = true; >> return "break"; >> end >> >> if type == "event" and voicemail_detected == false then >> voicemail_detected = true; >> return "break"; >> end >> end >> >> session:setInputCallback("onInput"); >> session:execute("avmd","start"); >> In order to implement this example, the script must wait for the beep to be detected or not to process the case accordingly. What is no beep is detected? How can I prevent called party to hear only silence while potential beep detection is being executed? >> >> Thank you >> >> Bernard >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120723/6202e723/attachment-0001.html From fiorix at gmail.com Mon Jul 23 22:33:05 2012 From: fiorix at gmail.com (Alexandre Fiori) Date: Mon, 23 Jul 2012 14:33:05 -0400 Subject: [Freeswitch-users] Bria iOS does SRTP wrong. In-Reply-To: <28D6D8F1-8758-4EEC-AFCA-B971BAD1CCAA@freeswitch.org> References: <28D6D8F1-8758-4EEC-AFCA-B971BAD1CCAA@freeswitch.org> Message-ID: <3962BF16-E197-4799-BF80-DF5148DF6953@gmail.com> I use Bria iOS with SIP+TLS (sslv23 actually) and SRTP on a regular basis and never had any issues... But I never looked at captures hehe I'll open the ticket and I believe I can put a bit more pressure on it since I'm buying a large number of custom branded versions of Bria iOS. On 2012-07-23, at 1:37 PM, Brian West wrote: > Dear FreeSWITCHologists, > Anyone that uses Bria on iOS, Please open a support case with CounterPath. I've tried, and they need to not offer a=crypto in an RTP/AVP and offer two m= lines one for RTP/AVP and RTP/SAVP in order of preference. They seem to think its not a critical thing to do SRTP the right way. (Guessing its not just SRTP LULZ!) > > Anyone see the refer to rfc3711 CRIT message? > > Why do I care so much... lolz > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST > iNUM: +883 5100 1286 0410 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org - af -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/4d32bfbe/attachment.html From philq at qsystemsengineering.com Mon Jul 23 22:36:25 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Mon, 23 Jul 2012 14:36:25 -0400 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! Message-ID: <00a801cd6902$1390a4f0$3ab1eed0$@com> Thanks Anthony, I actually was about to file a Jira on this since my RTP port range is set from 16384 to 32768. I asked about the problem on the list before doing so because I wanted to see if anyone knew about it, since I had searched on Jira FIRST and found nothing. I think I can speak for everyone in saying that we all appreciate the hard work that goes into FS. For my part, I'll file a Jira in the future and follow-up on the mailing list if it looks like a bug. Regards, - Phil _____________________________________________ Anthony Minessale Mon Jul 23 20:37:09 MSD 2012 * Previous message: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! * Next message: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! * Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] commit 524468be7ba5ee0887d2e975e4f131171c601a54 Author: Anthony Minessale Date: Mon Jul 23 11:36:19 2012 -0500 FS-4373 --resolve this is probably the same issue nobody would file a bug about on the mailing list even after I begged them to from last weekend please please stop filing bugs on the mailing list. I will still work on them and try to fix them but you are making me do way more work. Unless everyone wants to pay for support I cannot keep this up. _____________________________________________ From: Phil Quesinberry Sent: Sunday, July 22, 2012 1:16 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: [CRIT] sofia_glue.c:1083 No RTP ports available! Recently through several of the latest git pulls, FS stops routing calls with the following error, usually within a few minutes of making a call through Google Voice, not sure if receiving a call through GV will cause the problem yet: sofia_glue.c:1083 No RTP ports available! Restarting FS fixes the problem. Before I file the requisite Jira, is anyone aware of this problem or a potential configuration issue that might not be compatible with the newer builds of FS? This wasn't happening until the last week or two. Current version we're running is: FreeSWITCH Version 1.2.0-rc2+git~20120719T223942Z~42f296de9b+unclean~20120719T235346Z Here's a quick snip from the console when attempting to dial out to show the context of the error: 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3948 Looking for zrtp-hash 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3926 Deciding whether to pass zrtp-hash between legs 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3928 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5041 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:3027 Set Codec sofia/internal/102 at 192.168.1.6:5060 PCMU/8000 20 ms 160 samples 64000 bits 2012-07-22 12:57:46.706132 [DEBUG] switch_core_codec.c:111 sofia/internal/102 at 192.168.1.6:5060 Original read codec set to PCMU:0 2012-07-22 12:57:46.706132 [DEBUG] sofia_glue.c:5173 No 2833 in SDP. Disable 2833 dtmf and switch to INFO 2012-07-22 12:57:46.706132 [CRIT] sofia_glue.c:1083 No RTP ports available! 2012-07-22 12:57:46.706132 [DEBUG] switch_core_session.c:778 Send signal sofia/internal/102 at 192.168.1.6:5060 [BREAK] 2012-07-22 12:57:46.706132 [DEBUG] switch_channel.c:2903 (sofia/internal/102 at 192.168.1.6:5060) Callstate Change RINGING -> HANGUP 2012-07-22 12:57:46.706132 [NOTICE] switch_channel.c:3392 Hangup sofia/internal/102 at 192.168.1.6:5060 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] Regards, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/6e17e43f/attachment-0001.html From brian at freeswitch.org Mon Jul 23 22:59:16 2012 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Jul 2012 13:59:16 -0500 Subject: [Freeswitch-users] Bria iOS does SRTP wrong. In-Reply-To: <3962BF16-E197-4799-BF80-DF5148DF6953@gmail.com> References: <28D6D8F1-8758-4EEC-AFCA-B971BAD1CCAA@freeswitch.org> <3962BF16-E197-4799-BF80-DF5148DF6953@gmail.com> Message-ID: I was doing that this weekend and they were never offering AVP/SRTP, but after further review they are actually going to check into it more now. On Jul 23, 2012, at 1:33 PM, Alexandre Fiori wrote: > > I use Bria iOS with SIP+TLS (sslv23 actually) and SRTP on a regular basis and never had any issues... > But I never looked at captures hehe > > I'll open the ticket and I believe I can put a bit more pressure on it since I'm buying a large number of custom branded versions of Bria iOS. > > > On 2012-07-23, at 1:37 PM, Brian West wrote: > >> Dear FreeSWITCHologists, >> Anyone that uses Bria on iOS, Please open a support case with CounterPath. I've tried, and they need to not offer a=crypto in an RTP/AVP and offer two m= lines one for RTP/AVP and RTP/SAVP in order of preference. They seem to think its not a critical thing to do SRTP the right way. (Guessing its not just SRTP LULZ!) >> >> Anyone see the refer to rfc3711 CRIT message? >> >> Why do I care so much... lolz > -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/5ea2c28f/attachment.html From brian at freeswitch.org Mon Jul 23 23:00:34 2012 From: brian at freeswitch.org (Brian West) Date: Mon, 23 Jul 2012 14:00:34 -0500 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! In-Reply-To: <00a801cd6902$1390a4f0$3ab1eed0$@com> References: <00a801cd6902$1390a4f0$3ab1eed0$@com> Message-ID: <99C9BFD9-BD2C-4470-9A3D-4E29493503C3@freeswitch.org> I would rather have a jira and need to close it as "This is fixed already!", vs "What bug?" :P On Jul 23, 2012, at 1:36 PM, Phil Quesinberry wrote: > Thanks Anthony, > > I actually was about to file a Jira on this since my RTP port range is set > from 16384 to 32768. I asked about the problem on the list before doing so > because I wanted to see if anyone knew about it, since I had searched on > Jira FIRST and found nothing. > > I think I can speak for everyone in saying that we all appreciate the hard > work that goes into FS. For my part, I'll file a Jira in the future and > follow-up on the mailing list if it looks like a bug. > > Regards, > > - Phil -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/1f421c21/attachment.html From toddb at toddbailey.net Mon Jul 23 23:20:25 2012 From: toddb at toddbailey.net (Todd) Date: Mon, 23 Jul 2012 12:20:25 -0700 Subject: [Freeswitch-users] No audio after upgrade to F17 Message-ID: <1343071225.4358.10.camel@mythtv> Well this is odd, after I installed F17, I don't get audio to or from the pstn line (via a spa3102 ata) once the call is answered. I do get ring tones, and Fs generated voice messages, and can leave or retrieve vm. I've tried a couple approaches so. When I installed F17, I copied the FS installation directly Booting back to F14, every this works as expected, so it's not the spa3102 or config files. Thinking there might be a incompatibility issues, I recompiled FS under F17, no change in operational behavior. Ideas where else to look ? From andrew at cassidywebservices.co.uk Mon Jul 23 23:27:08 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Mon, 23 Jul 2012 20:27:08 +0100 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! In-Reply-To: <99C9BFD9-BD2C-4470-9A3D-4E29493503C3@freeswitch.org> References: <00a801cd6902$1390a4f0$3ab1eed0$@com> <99C9BFD9-BD2C-4470-9A3D-4E29493503C3@freeswitch.org> Message-ID: Don't forget the extra ports needed for oob dtmf! On 23 July 2012 20:00, Brian West wrote: > I would rather have a jira and need to close it as "This is fixed > already!", vs "What bug?" > > :P > > On Jul 23, 2012, at 1:36 PM, Phil Quesinberry wrote: > > Thanks Anthony, > > I actually was about to file a Jira on this since my RTP port range is set > from 16384 to 32768. I asked about the problem on the list before doing so > > because I wanted to see if anyone knew about it, since I had searched on > Jira FIRST and found nothing. > > I think I can speak for everyone in saying that we all appreciate the hard > work that goes into FS. For my part, I'll file a Jira in the future and > follow-up on the mailing list if it looks like a bug. > > Regards, > > - Phil > > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST > iNUM: +883 5100 1286 0410 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/9d44d6a7/attachment-0001.html From peter.olsson at visionutveckling.se Tue Jul 24 00:03:04 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 23 Jul 2012 20:03:04 +0000 Subject: [Freeswitch-users] No audio after upgrade to F17 In-Reply-To: <1343071225.4358.10.camel@mythtv> References: <1343071225.4358.10.camel@mythtv> Message-ID: <1FFF97C269757C458224B7C895F35F15136EE2@cantor.std.visionutv.se> Local firewall maybe? ________________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] för Todd [toddb at toddbailey.net] Skickat: den 23 juli 2012 21:20 Till: freeswitch ?mne: [Freeswitch-users] No audio after upgrade to F17 Well this is odd, after I installed F17, I don't get audio to or from the pstn line (via a spa3102 ata) once the call is answered. I do get ring tones, and Fs generated voice messages, and can leave or retrieve vm. I've tried a couple approaches so. When I installed F17, I copied the FS installation directly Booting back to F14, every this works as expected, so it's not the spa3102 or config files. Thinking there might be a incompatibility issues, I recompiled FS under F17, no change in operational behavior. Ideas where else to look ? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:500da65e32766305730251! From msc at freeswitch.org Tue Jul 24 00:08:11 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Jul 2012 13:08:11 -0700 Subject: [Freeswitch-users] Important Information About Not Reporting Bugs on the Mailing List Message-ID: Hello all, We've had yet another thread where someone wasn't sure if there was a but so they asked on the mailing list. I've written up some simple guidelines and put them on the world-famous Bugs wiki page: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Important_Note.2C_Please_Read In short: not only should no one report bugs to the mailing list, no one should report symptoms to the mailing list and ask, "Is this a bug? Should I file a Jira?" If you have collected enough information to send an email to the list and ask whether or not to file a Jira then there's really no point in asking - just file the Jira. If you would like to discuss the symptoms on the mailing list then open the Jira *first* and then link to it in the message you send to the list. It's much better to open a Jira and find out later that it's not really a bug than it is to report a possible bug on the mailing list. Even if something is flagged as "not a bug" or "works as designed" there is still value in having the Jira ticket in the database for historical and research purposes. For those who have heard this discussion before, please be kind to the noobs and first-timers who may not appreciate the issues involved. Just point them to the bugs page. If someone repeatedly reports bugs to the mailing list then we will deliver the message with less delicacy. Thanks again for all your help. Using Jira will help keep Tony's life from become any more complicated than it already is. -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/2040e568/attachment.html From jlyons at quikvoip.net Tue Jul 24 00:51:58 2012 From: jlyons at quikvoip.net (James) Date: Mon, 23 Jul 2012 16:51:58 -0400 Subject: [Freeswitch-users] non-t38 fax fails between two local FS servers Message-ID: <20120723165158.611d7a0f@calico> I have two FS servers on the same LAN, one for receiving FAXIN and one for transmitting FAXOUT FAXOUT receives an originate via ESL to send a number that routes to FAXIN via a local FS GW. FAXIN directs call to python script via dialplan FAXIN drops call 10 secs after negotiation result with this error mod_spandsp_fax.c:500 Fax processing not successful - result (13) Unexpected message received. When I enable fax_enable_t38 and fax_enable_t38_request, this same call succeeds. This seems like bad configuration on FAXIN, but I receive some faxes from remote destinations without t38. -- James Lyons QuikVoIP, LLC. p: +1 (786) 369-5308 w: www.quikvoip.net From kris at kriskinc.com Tue Jul 24 01:25:29 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 23 Jul 2012 17:25:29 -0400 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! In-Reply-To: References: <00a801cd6902$1390a4f0$3ab1eed0$@com> <99C9BFD9-BD2C-4470-9A3D-4E29493503C3@freeswitch.org> Message-ID: What extra ports for out of band DTMF? On Mon, Jul 23, 2012 at 3:27 PM, Andrew Cassidy wrote: > > Don't forget the extra ports needed for oob dtmf! > -- Kristian Kielhofner From nickolayr at gmail.com Tue Jul 24 01:36:48 2012 From: nickolayr at gmail.com (Nikolay Rogoshchenkov) Date: Mon, 23 Jul 2012 17:36:48 -0400 Subject: [Freeswitch-users] make mod_rad_auth under FreeBSD 8.2 Message-ID: Hello, May be some-who knows how to compile mod_rad_auto under FreeBSD? This is what I'm got: =================================================================================================================================================== *dp# cd /usr/src/freeswitch/src/mod/applications/mod_rad_auth/* *You have new mail.* *dp# ./configure* *./configure: Command not found.* *dp# ls* *Makefile mod_rad_auth.c* *dp# make* *Compiling /usr/src/freeswitch/src/mod/applications/mod_rad_auth/mod_rad_auth.c...* *quiet_libtool: compile: gcc -I/usr/src/freeswitch/libs/curl/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/src/include -I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -I/usr/local/include -DHAVE_OPENSSL -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/applications/mod_rad_auth/mod_rad_auth.c -fPIC -DPIC -o .libs/mod_rad_auth.o* *In file included from /usr/src/freeswitch/src/mod/applications/mod_rad_auth/mod_rad_auth.c:34:* */usr/local/include/freeradius-client.h:23:20: error: ISO C does not permit named variadic macros* **** Error code 1* * * *Stop in /usr/src/freeswitch/src/mod/applications/mod_rad_auth.* **** Error code 1* * * *Stop in /usr/src/freeswitch/src/mod/applications/mod_rad_auth.* *dp# * =================================================================================================================================================== Used git repo with last commit at Wed May 16 20:35:15 2012. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/263bc1ac/attachment.html From curriegrad2004 at gmail.com Tue Jul 24 02:32:53 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Mon, 23 Jul 2012 15:32:53 -0700 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! In-Reply-To: References: <00a801cd6902$1390a4f0$3ab1eed0$@com> <99C9BFD9-BD2C-4470-9A3D-4E29493503C3@freeswitch.org> Message-ID: There is no extra ports for OOB dtmf. It's just sent over the RTP ports really On Mon, Jul 23, 2012 at 2:25 PM, Kristian Kielhofner wrote: > What extra ports for out of band DTMF? > > On Mon, Jul 23, 2012 at 3:27 PM, Andrew Cassidy > wrote: >> >> Don't forget the extra ports needed for oob dtmf! >> > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fluixab at bellsouth.net Tue Jul 24 02:38:16 2012 From: fluixab at bellsouth.net (Bernard Fluixa) Date: Mon, 23 Jul 2012 18:38:16 -0400 Subject: [Freeswitch-users] mod_avmd In-Reply-To: <0C16CD66-3751-4946-90EC-8AEB03F6F06F@bellsouth.net> References: <4211B6CA-C067-4B6D-B271-CC1BB82C0206@bellsouth.net> <0C16CD66-3751-4946-90EC-8AEB03F6F06F@bellsouth.net> Message-ID: <103F647B-4D6C-4E00-BED1-CB2ADB6CAAEA@bellsouth.net> Hello, I can start mod_avmd and it detects voicemail beep when calling my cellphone. However, when I pickup the phone and press a key to use IVR, I would like to stop mod_avmd. I copied the input callback example from FreeSWITCH Wiki. My issue is that my input handler is never triggered, neither when I key a DTMF nor when beep is detected. 1) My handler is: function onInput(s, type, obj) if (type == "dtmf") then avmd_result = 1; if ( debug_mode > 0 ) then print("onInput - Human detected"); end return "break"; end if (type == "event" ) then avmd_result = 2; if ( debug_mode > 0 ) then print("onInput - Voicemail detected"); end avmd_result = 2; return "break"; end end 2) To start mod_avmd, play file and collect digits: session:setInputCallback("onInput"); session:execute("avmd","start"); digits = session:playAndGetDigits(??.); 3) To stop mod_avmd: session:execute("avmd","stop"); 4) Output when keying DTMF (*3 in this case). Input handler does not kick off upon DTMF: EXECUTE sofia/external/+19543307528 avmd(start) 012-07-23 18:13:28.025981 [INFO] mod_avmd.c:538 <<< AVMD v=0.073060 f=0.900987 1147.172790Hz sma=0.090099 sqa=0.081178 >>> ... 012-07-23 18:13:30.492866 [INFO] mod_avmd.c:538 <<< AVMD v=0.052551 f=0.764130 972.920758Hz sma=0.076413 sqa=0.058389 >>> 2012-07-23 18:13:30.586854 [DEBUG] switch_rtp.c:3795 RTP RECV DTMF *:800 2012-07-23 18:13:30.605660 [INFO] mod_avmd.c:538 <<< AVMD v=0.067196 f=0.864073 1100.171521Hz sma=0.086407 sqa=0.074662 >>> ? 2012-07-23 18:13:30.998715 [INFO] mod_avmd.c:538 <<< AVMD v=0.030803 f=0.585023 744.874518Hz sma=0.058502 sqa=0.034225 >>> 2012-07-23 18:13:31.085676 [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 3:640 ... 012-07-23 18:13:36.645609 [INFO] mod_avmd.c:538 <<< AVMD v=0.030380 f=0.580996 739.747695Hz sma=0.058100 sqa=0.033756 >>> 5) Output when beep is detected - Mod_avmd stops output after beep detection but input handler does not kick off 2012-07-23 18:04:04.992856 [INFO] mod_avmd.c:538 <<< AVMD v=0.000048 f=0.812668 1034.721179Hz sma=0.811125 sqa=0.657971 >>> 2012-07-23 18:04:04.992856 [INFO] mod_avmd.c:561 <<< AVMD - Beep Detected >>> What am I missing? Thank you Bernard On Jul 23, 2012, at 2:16 PM, Bernard Fluixa wrote: > OK. I'm clear now. Thanks again. > On Jul 23, 2012, at 1:17 PM, Michael Collins wrote: > >> Precisely. >> -MC >> >> On Mon, Jul 23, 2012 at 9:59 AM, Bernard Fluixa wrote: >> Michael, >> >> Thank you or your response. So it is my responsibility to do whatever needs to be done while mod_avmd attempts to detect a beep and to manually stop it after a beep as been detected or after a certain timeout. Correct? >> >> Bernard >> >> >> >> >> On Jul 23, 2012, at 11:59 AM, Michael Collins wrote: >> >>> Bernard, >>> >>> Keep in mind that avmd is non-blocking, that is, it won't cause your dialplan or script to pause while it is attempting to detect the beep. Your dialplan will keep doing what it normally does, and if avmd detects a beep then it will throw an event which you catch and handle in your onInput function. >>> >>> -MC >>> >>> On Mon, Jul 23, 2012 at 6:47 AM, BF wrote: >>> Hello, >>> >>> I'm trying to use it from a Lua script. My understanding is that mod_avmd detects beep from voicemail systems and is CPU intensive, please correct me if I'm wrong. >>> >>> The Lua example at FreeSWITCH Wiki is >>> local human_detected = false; >>> local voicemail_detected = false; >>> >>> function onInput(session, type, obj) >>> if type == "dtmf" and obj['digit'] == '1' and human_detected == false then >>> human_detected = true; >>> return "break"; >>> end >>> >>> if type == "event" and voicemail_detected == false then >>> voicemail_detected = true; >>> return "break"; >>> end >>> end >>> >>> session:setInputCallback("onInput"); >>> session:execute("avmd","start"); >>> In order to implement this example, the script must wait for the beep to be detected or not to process the case accordingly. What is no beep is detected? How can I prevent called party to hear only silence while potential beep detection is being executed? >>> >>> Thank you >>> >>> Bernard >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120723/9bbb37a0/attachment-0001.html From msc at freeswitch.org Tue Jul 24 02:40:23 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Jul 2012 15:40:23 -0700 Subject: [Freeswitch-users] mod_avmd In-Reply-To: <103F647B-4D6C-4E00-BED1-CB2ADB6CAAEA@bellsouth.net> References: <4211B6CA-C067-4B6D-B271-CC1BB82C0206@bellsouth.net> <0C16CD66-3751-4946-90EC-8AEB03F6F06F@bellsouth.net> <103F647B-4D6C-4E00-BED1-CB2ADB6CAAEA@bellsouth.net> Message-ID: Can you pastebin the entire script? -MC On Mon, Jul 23, 2012 at 3:38 PM, Bernard Fluixa wrote: > Hello, > > I can start mod_avmd and it detects voicemail beep when calling my > cellphone. However, when I pickup the phone and press a key to use IVR, I > would like to stop mod_avmd. I copied the input callback example from > FreeSWITCH Wiki. My issue is that my input handler is never triggered, > neither when I key a DTMF nor when beep is detected. > > *1) My handler is: * > function onInput(s, type, obj) > if (type == "dtmf") then > avmd_result = 1; > if ( debug_mode > 0 ) then > print("onInput - Human detected"); > end > return "break"; > end > if (type == "event" ) then > avmd_result = 2; > if ( debug_mode > 0 ) then > print("onInput - Voicemail detected"); > end > avmd_result = 2; > return "break"; > end > end > > *2) To start mod_avmd, play file and collect digits:* > * session:setInputCallback("onInput");* > * session:execute("avmd","start");* > > digits = session:playAndGetDigits(??.); > > *3) To stop mod_avmd:* > session:execute("avmd","stop"); > > *4) Output when keying DTMF (*3 in this case). Input handler does not > kick off upon DTMF: * > EXECUTE sofia/external/+19543307528 avmd(start) > 012-07-23 18:13:28.025981 [INFO] mod_avmd.c:538 <<< AVMD v=0.073060 > f=0.900987 1147.172790Hz sma=0.090099 sqa=0.081178 >>> > ... > 012-07-23 18:13:30.492866 [INFO] mod_avmd.c:538 <<< AVMD v=0.052551 > f=0.764130 972.920758Hz sma=0.076413 sqa=0.058389 >>> > 2012-07-23 18:13:30.586854 [DEBUG] switch_rtp.c:3795 RTP RECV DTMF *:800 > 2012-07-23 18:13:30.605660 [INFO] mod_avmd.c:538 <<< AVMD v=0.067196 > f=0.864073 1100.171521Hz sma=0.086407 sqa=0.074662 >>> > ? > 2012-07-23 18:13:30.998715 [INFO] mod_avmd.c:538 <<< AVMD v=0.030803 > f=0.585023 744.874518Hz sma=0.058502 sqa=0.034225 >>> > 2012-07-23 18:13:31.085676 [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 3:640 > ... > 012-07-23 18:13:36.645609 [INFO] mod_avmd.c:538 <<< AVMD v=0.030380 > f=0.580996 739.747695Hz sma=0.058100 sqa=0.033756 >>> > > *5) Output when beep is detected - Mod_avmd stops output after beep > detection but input handler does not kick off* > 2012-07-23 18:04:04.992856 [INFO] mod_avmd.c:538 <<< AVMD v=0.000048 > f=0.812668 1034.721179Hz sma=0.811125 sqa=0.657971 >>> > 2012-07-23 18:04:04.992856 [INFO] mod_avmd.c:561 <<< AVMD - Beep Detected > >>> > > What am I missing? > > Thank you > > Bernard > > > > > On Jul 23, 2012, at 2:16 PM, Bernard Fluixa wrote: > > OK. I'm clear now. Thanks again. > On Jul 23, 2012, at 1:17 PM, Michael Collins wrote: > > Precisely. > -MC > > On Mon, Jul 23, 2012 at 9:59 AM, Bernard Fluixa wrote: > >> Michael, >> >> Thank you or your response. So it is my responsibility to do whatever >> needs to be done while mod_avmd attempts to detect a beep and to manually >> stop it after a beep as been detected or after a certain timeout. Correct? >> >> Bernard >> >> >> >> >> On Jul 23, 2012, at 11:59 AM, Michael Collins wrote: >> >> Bernard, >> >> Keep in mind that avmd is non-blocking, that is, it won't cause your >> dialplan or script to pause while it is attempting to detect the beep. Your >> dialplan will keep doing what it normally does, and if avmd detects a beep >> then it will throw an event which you catch and handle in your onInput >> function. >> >> -MC >> >> On Mon, Jul 23, 2012 at 6:47 AM, BF wrote: >> >>> Hello, >>> >>> I'm trying to use it from a Lua script. My understanding is that >>> mod_avmd detects beep from voicemail systems and is CPU intensive, please >>> correct me if I'm wrong. >>> >>> The Lua example at FreeSWITCH Wiki is >>> local human_detected = false; >>> local voicemail_detected = false; >>> >>> function onInput(session, type, obj) >>> if type == "dtmf" and obj['digit'] == '1' and human_detected == >>> false then >>> human_detected = true; >>> return "break"; >>> end >>> >>> if type == "event" and voicemail_detected == false then >>> voicemail_detected = true; >>> return "break"; >>> end >>> end >>> >>> session:setInputCallback("onInput"); >>> session:execute("avmd","start"); >>> In order to implement this example, the script must wait for the beep to >>> be detected or not to process the case accordingly. What is no beep is >>> detected? How can I prevent called party to hear only silence while >>> potential beep detection is being executed? >>> >>> Thank you >>> >>> Bernard >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/7e50a30e/attachment.html From kris at kriskinc.com Tue Jul 24 02:52:58 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 23 Jul 2012 18:52:58 -0400 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! In-Reply-To: References: <00a801cd6902$1390a4f0$3ab1eed0$@com> <99C9BFD9-BD2C-4470-9A3D-4E29493503C3@freeswitch.org> Message-ID: Agreed. On Mon, Jul 23, 2012 at 6:32 PM, curriegrad2004 wrote: > There is no extra ports for OOB dtmf. It's just sent over the RTP ports really > -- Kristian Kielhofner From msc at freeswitch.org Tue Jul 24 02:55:52 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Jul 2012 15:55:52 -0700 Subject: [Freeswitch-users] dialplan commands on playback In-Reply-To: <7AB73CC2-06A0-46E2-BA18-8603C208FADD@gmail.com> References: <5117A1BD-A549-4091-8F0F-577DE29C747C@gmail.com> <7AB73CC2-06A0-46E2-BA18-8603C208FADD@gmail.com> Message-ID: Well, if you have soxi installed (Linux, of course) then you could always use a system call to grab the file length. For kicks I whipped up a proof of concept dial plan to demonstrate how it could work. If anyone finds it useful please let me know and I'll wikify it. -MC On Mon, Jul 23, 2012 at 9:09 AM, Alexandre Fiori wrote: > > that's the problem, I don't know the length, but playback does :) > > thanks anyway > > On 2012-07-23, at 11:53 AM, Michael Collins wrote: > > If you know the length of the file then you could use sched_api to > schedule something. For example, if you know that your file is 12 seconds > long you could do something like this: > > sched_api +9 none uuid_broadcast :: bleg > > Check out both of these on the wiki - there's a lot that these can do. > -MC > > http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast > http://wiki.freeswitch.org/wiki/Mod_commands#sched_api > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > - af > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/7e78ecca/attachment.html From fiorix at gmail.com Tue Jul 24 03:01:13 2012 From: fiorix at gmail.com (Alexandre Fiori) Date: Mon, 23 Jul 2012 19:01:13 -0400 Subject: [Freeswitch-users] dialplan commands on playback In-Reply-To: References: <5117A1BD-A549-4091-8F0F-577DE29C747C@gmail.com> <7AB73CC2-06A0-46E2-BA18-8603C208FADD@gmail.com> Message-ID: Amazing. This is exactly what I was looking for, and couldn't figure out how to do. Thank you! On 2012-07-23, at 6:55 PM, Michael Collins wrote: > Well, if you have soxi installed (Linux, of course) then you could always use a system call to grab the file length. For kicks I whipped up a proof of concept dial plan to demonstrate how it could work. If anyone finds it useful please let me know and I'll wikify it. > -MC > > > > > > data="sound_file=${sounds_dir}/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav"/> > data="file_length=${system soxi -D ${sound_file} | tr -d '\\n' }"/> > > > > > > > > > > On Mon, Jul 23, 2012 at 9:09 AM, Alexandre Fiori wrote: > > that's the problem, I don't know the length, but playback does :) > > thanks anyway > > On 2012-07-23, at 11:53 AM, Michael Collins wrote: > >> If you know the length of the file then you could use sched_api to schedule something. For example, if you know that your file is 12 seconds long you could do something like this: >> >> sched_api +9 none uuid_broadcast :: bleg >> >> Check out both of these on the wiki - there's a lot that these can do. >> -MC >> >> http://wiki.freeswitch.org/wiki/Mod_commands#uuid_broadcast >> http://wiki.freeswitch.org/wiki/Mod_commands#sched_api >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > - af > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org - af -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/2467b6da/attachment-0001.html From bfmtl at hotmail.com Tue Jul 24 03:03:25 2012 From: bfmtl at hotmail.com (BF) Date: Mon, 23 Jul 2012 19:03:25 -0400 Subject: [Freeswitch-users] mod_avmd In-Reply-To: References: <4211B6CA-C067-4B6D-B271-CC1BB82C0206@bellsouth.net> <0C16CD66-3751-4946-90EC-8AEB03F6F06F@bellsouth.net> <103F647B-4D6C-4E00-BED1-CB2ADB6CAAEA@bellsouth.net> Message-ID: It's my very first pastebin. I hope this is what you're expecting: http://pastebin.freeswitch.org/19579 On Jul 23, 2012, at 6:40 PM, Michael Collins wrote: > Can you pastebin the entire script? > -MC > > On Mon, Jul 23, 2012 at 3:38 PM, Bernard Fluixa wrote: > Hello, > > I can start mod_avmd and it detects voicemail beep when calling my cellphone. However, when I pickup the phone and press a key to use IVR, I would like to stop mod_avmd. I copied the input callback example from FreeSWITCH Wiki. My issue is that my input handler is never triggered, neither when I key a DTMF nor when beep is detected. > > 1) My handler is: > function onInput(s, type, obj) > if (type == "dtmf") then > avmd_result = 1; > if ( debug_mode > 0 ) then > print("onInput - Human detected"); > end > return "break"; > end > if (type == "event" ) then > avmd_result = 2; > if ( debug_mode > 0 ) then > print("onInput - Voicemail detected"); > end > avmd_result = 2; > return "break"; > end > end > > 2) To start mod_avmd, play file and collect digits: > session:setInputCallback("onInput"); > session:execute("avmd","start"); > > digits = session:playAndGetDigits(??.); > > 3) To stop mod_avmd: > session:execute("avmd","stop"); > > 4) Output when keying DTMF (*3 in this case). Input handler does not kick off upon DTMF: > EXECUTE sofia/external/+19543307528 avmd(start) > 012-07-23 18:13:28.025981 [INFO] mod_avmd.c:538 <<< AVMD v=0.073060 f=0.900987 1147.172790Hz sma=0.090099 sqa=0.081178 >>> > ... > 012-07-23 18:13:30.492866 [INFO] mod_avmd.c:538 <<< AVMD v=0.052551 f=0.764130 972.920758Hz sma=0.076413 sqa=0.058389 >>> > 2012-07-23 18:13:30.586854 [DEBUG] switch_rtp.c:3795 RTP RECV DTMF *:800 > 2012-07-23 18:13:30.605660 [INFO] mod_avmd.c:538 <<< AVMD v=0.067196 f=0.864073 1100.171521Hz sma=0.086407 sqa=0.074662 >>> > ? > 2012-07-23 18:13:30.998715 [INFO] mod_avmd.c:538 <<< AVMD v=0.030803 f=0.585023 744.874518Hz sma=0.058502 sqa=0.034225 >>> > 2012-07-23 18:13:31.085676 [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 3:640 > ... > 012-07-23 18:13:36.645609 [INFO] mod_avmd.c:538 <<< AVMD v=0.030380 f=0.580996 739.747695Hz sma=0.058100 sqa=0.033756 >>> > > 5) Output when beep is detected - Mod_avmd stops output after beep detection but input handler does not kick off > 2012-07-23 18:04:04.992856 [INFO] mod_avmd.c:538 <<< AVMD v=0.000048 f=0.812668 1034.721179Hz sma=0.811125 sqa=0.657971 >>> > 2012-07-23 18:04:04.992856 [INFO] mod_avmd.c:561 <<< AVMD - Beep Detected >>> > > What am I missing? > > Thank you > > Bernard > > > > > On Jul 23, 2012, at 2:16 PM, Bernard Fluixa wrote: > >> OK. I'm clear now. Thanks again. >> On Jul 23, 2012, at 1:17 PM, Michael Collins wrote: >> >>> Precisely. >>> -MC >>> >>> On Mon, Jul 23, 2012 at 9:59 AM, Bernard Fluixa wrote: >>> Michael, >>> >>> Thank you or your response. So it is my responsibility to do whatever needs to be done while mod_avmd attempts to detect a beep and to manually stop it after a beep as been detected or after a certain timeout. Correct? >>> >>> Bernard >>> >>> >>> >>> >>> On Jul 23, 2012, at 11:59 AM, Michael Collins wrote: >>> >>>> Bernard, >>>> >>>> Keep in mind that avmd is non-blocking, that is, it won't cause your dialplan or script to pause while it is attempting to detect the beep. Your dialplan will keep doing what it normally does, and if avmd detects a beep then it will throw an event which you catch and handle in your onInput function. >>>> >>>> -MC >>>> >>>> On Mon, Jul 23, 2012 at 6:47 AM, BF wrote: >>>> Hello, >>>> >>>> I'm trying to use it from a Lua script. My understanding is that mod_avmd detects beep from voicemail systems and is CPU intensive, please correct me if I'm wrong. >>>> >>>> The Lua example at FreeSWITCH Wiki is >>>> local human_detected = false; >>>> local voicemail_detected = false; >>>> >>>> function onInput(session, type, obj) >>>> if type == "dtmf" and obj['digit'] == '1' and human_detected == false then >>>> human_detected = true; >>>> return "break"; >>>> end >>>> >>>> if type == "event" and voicemail_detected == false then >>>> voicemail_detected = true; >>>> return "break"; >>>> end >>>> end >>>> >>>> session:setInputCallback("onInput"); >>>> session:execute("avmd","start"); >>>> In order to implement this example, the script must wait for the beep to be detected or not to process the case accordingly. What is no beep is detected? How can I prevent called party to hear only silence while potential beep detection is being executed? >>>> >>>> Thank you >>>> >>>> Bernard >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120723/bfc4b5b4/attachment-0001.html From msc at freeswitch.org Tue Jul 24 03:11:05 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Jul 2012 16:11:05 -0700 Subject: [Freeswitch-users] dialplan commands on playback In-Reply-To: References: <5117A1BD-A549-4091-8F0F-577DE29C747C@gmail.com> <7AB73CC2-06A0-46E2-BA18-8603C208FADD@gmail.com> Message-ID: Excellent! Do you mind finding a spot up on the wiki for this? -MC On Mon, Jul 23, 2012 at 4:01 PM, Alexandre Fiori wrote: > > Amazing. This is exactly what I was looking for, and couldn't figure out > how to do. > Thank you! > > > On 2012-07-23, at 6:55 PM, Michael Collins wrote: > > Well, if you have soxi installed (Linux, of course) then you could always > use a system call to grab the file length. For kicks I whipped up a proof > of concept dial plan to demonstrate how it could work. If anyone finds it > useful please let me know and I'll wikify it. > -MC > > > > > > > data="sound_file=${sounds_dir}/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav"/> > data="file_length=${system soxi -D ${sound_file} | tr -d > '\\n' }"/> > > data="file_length=${expr(ceil(${file_length}))}"/> > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/dec7e116/attachment.html From anthony.minessale at gmail.com Tue Jul 24 03:14:45 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 23 Jul 2012 18:14:45 -0500 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! In-Reply-To: References: <00a801cd6902$1390a4f0$3ab1eed0$@com> <99C9BFD9-BD2C-4470-9A3D-4E29493503C3@freeswitch.org> Message-ID: Just remember to file jiras and tell everyone to come to cluecon and we can call it even. On Mon, Jul 23, 2012 at 5:52 PM, Kristian Kielhofner wrote: > Agreed. > > On Mon, Jul 23, 2012 at 6:32 PM, curriegrad2004 > wrote: >> There is no extra ports for OOB dtmf. It's just sent over the RTP ports really >> > > -- > Kristian Kielhofner > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From chris at opencsta.org Tue Jul 24 03:24:09 2012 From: chris at opencsta.org (Chris Mylonas) Date: Tue, 24 Jul 2012 09:24:09 +1000 Subject: [Freeswitch-users] [CRIT] sofia_glue.c:1083 No RTP ports available! In-Reply-To: References: <00a801cd6902$1390a4f0$3ab1eed0$@com> <99C9BFD9-BD2C-4470-9A3D-4E29493503C3@freeswitch.org> Message-ID: <106D6261-6B1E-4F74-B7AC-E3B4671A9ACC@opencsta.org> Would love to come to cluecon one year Tony. I need to update my code and do more stuff with FS before doing so ;) BTW, Tony's threat of not fixing bugs if it's not in JIRA is right on. I support this. If you haven't registered to FreeSWITCH's JIRA - do so today!!!! It takes SFA (slang) time to do so!!! Personally, when I have my dev hat on, I love to dev. When I have my SA hat on For the sys admins out there - do you know how good it feels to close tickets? Make Tony's day! /brown tongue On 24/07/2012, at 9:14 AM, Anthony Minessale wrote: > Just remember to file jiras and tell everyone to come to cluecon and > we can call it even. > > > On Mon, Jul 23, 2012 at 5:52 PM, Kristian Kielhofner wrote: >> Agreed. >> >> On Mon, Jul 23, 2012 at 6:32 PM, curriegrad2004 >> wrote: >>> There is no extra ports for OOB dtmf. It's just sent over the RTP ports really >>> >> >> -- >> Kristian Kielhofner >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Jul 24 03:37:22 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 23 Jul 2012 16:37:22 -0700 Subject: [Freeswitch-users] mod_avmd In-Reply-To: References: <4211B6CA-C067-4B6D-B271-CC1BB82C0206@bellsouth.net> <0C16CD66-3751-4946-90EC-8AEB03F6F06F@bellsouth.net> <103F647B-4D6C-4E00-BED1-CB2ADB6CAAEA@bellsouth.net> Message-ID: Well posted - nicely done for a self-proclaimed n00b. For better readability you might want to try selecting syntax highlighting, kinda like http://pastebin.freeswitch.org/19580. If I had to wager a guess as to what's going on I'd say you didn't get the FreeSWITCH 'bridge book' and read chapter 7. At first blush I'd say that your DTMFs are getting eaten by play_and_get_digits. Maybe try playback instead? And read chapter 7 - there's an example in there about reading digits w/ the input callback function. As far as the avmd stuff goes I'm not sure at this point. If I get a sec I'll tinker with it and see if I can replicate. -MC On Mon, Jul 23, 2012 at 4:03 PM, BF wrote: > It's my very first pastebin. I hope this is what you're expecting: > http://pastebin.freeswitch.org/19579 > > On Jul 23, 2012, at 6:40 PM, Michael Collins wrote: > > Can you pastebin the entire script? > -MC > > On Mon, Jul 23, 2012 at 3:38 PM, Bernard Fluixa wrote: > >> Hello, >> >> I can start mod_avmd and it detects voicemail beep when calling my >> cellphone. However, when I pickup the phone and press a key to use IVR, I >> would like to stop mod_avmd. I copied the input callback example from >> FreeSWITCH Wiki. My issue is that my input handler is never triggered, >> neither when I key a DTMF nor when beep is detected. >> >> *1) My handler is: * >> function onInput(s, type, obj) >> if (type == "dtmf") then >> avmd_result = 1; >> if ( debug_mode > 0 ) then >> print("onInput - Human detected"); >> end >> return "break"; >> end >> if (type == "event" ) then >> avmd_result = 2; >> if ( debug_mode > 0 ) then >> print("onInput - Voicemail detected"); >> end >> avmd_result = 2; >> return "break"; >> end >> end >> >> *2) To start mod_avmd, play file and collect digits:* >> * session:setInputCallback("onInput");* >> * session:execute("avmd","start");* >> >> digits = session:playAndGetDigits(??.); >> >> *3) To stop mod_avmd:* >> session:execute("avmd","stop"); >> >> *4) Output when keying DTMF (*3 in this case). Input handler does not >> kick off upon DTMF: * >> EXECUTE sofia/external/+19543307528 avmd(start) >> 012-07-23 18:13:28.025981 [INFO] mod_avmd.c:538 <<< AVMD v=0.073060 >> f=0.900987 1147.172790Hz sma=0.090099 sqa=0.081178 >>> >> ... >> 012-07-23 18:13:30.492866 [INFO] mod_avmd.c:538 <<< AVMD v=0.052551 >> f=0.764130 972.920758Hz sma=0.076413 sqa=0.058389 >>> >> 2012-07-23 18:13:30.586854 [DEBUG] switch_rtp.c:3795 RTP RECV DTMF *:800 >> 2012-07-23 18:13:30.605660 [INFO] mod_avmd.c:538 <<< AVMD v=0.067196 >> f=0.864073 1100.171521Hz sma=0.086407 sqa=0.074662 >>> >> ? >> 2012-07-23 18:13:30.998715 [INFO] mod_avmd.c:538 <<< AVMD v=0.030803 >> f=0.585023 744.874518Hz sma=0.058502 sqa=0.034225 >>> >> 2012-07-23 18:13:31.085676 [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 3:640 >> ... >> 012-07-23 18:13:36.645609 [INFO] mod_avmd.c:538 <<< AVMD v=0.030380 >> f=0.580996 739.747695Hz sma=0.058100 sqa=0.033756 >>> >> >> *5) Output when beep is detected - Mod_avmd stops output after beep >> detection but input handler does not kick off* >> 2012-07-23 18:04:04.992856 [INFO] mod_avmd.c:538 <<< AVMD v=0.000048 >> f=0.812668 1034.721179Hz sma=0.811125 sqa=0.657971 >>> >> 2012-07-23 18:04:04.992856 [INFO] mod_avmd.c:561 <<< AVMD - Beep Detected >> >>> >> >> What am I missing? >> >> Thank you >> >> Bernard >> >> >> >> >> On Jul 23, 2012, at 2:16 PM, Bernard Fluixa wrote: >> >> OK. I'm clear now. Thanks again. >> On Jul 23, 2012, at 1:17 PM, Michael Collins wrote: >> >> Precisely. >> -MC >> >> On Mon, Jul 23, 2012 at 9:59 AM, Bernard Fluixa wrote: >> >>> Michael, >>> >>> Thank you or your response. So it is my responsibility to do whatever >>> needs to be done while mod_avmd attempts to detect a beep and to manually >>> stop it after a beep as been detected or after a certain timeout. Correct? >>> >>> Bernard >>> >>> >>> >>> >>> On Jul 23, 2012, at 11:59 AM, Michael Collins wrote: >>> >>> Bernard, >>> >>> Keep in mind that avmd is non-blocking, that is, it won't cause your >>> dialplan or script to pause while it is attempting to detect the beep. Your >>> dialplan will keep doing what it normally does, and if avmd detects a beep >>> then it will throw an event which you catch and handle in your onInput >>> function. >>> >>> -MC >>> >>> On Mon, Jul 23, 2012 at 6:47 AM, BF wrote: >>> >>>> Hello, >>>> >>>> I'm trying to use it from a Lua script. My understanding is that >>>> mod_avmd detects beep from voicemail systems and is CPU intensive, please >>>> correct me if I'm wrong. >>>> >>>> The Lua example at FreeSWITCH Wiki is >>>> local human_detected = false; >>>> local voicemail_detected = false; >>>> >>>> function onInput(session, type, obj) >>>> if type == "dtmf" and obj['digit'] == '1' and human_detected == >>>> false then >>>> human_detected = true; >>>> return "break"; >>>> end >>>> >>>> if type == "event" and voicemail_detected == false then >>>> voicemail_detected = true; >>>> return "break"; >>>> end >>>> end >>>> >>>> session:setInputCallback("onInput"); >>>> session:execute("avmd","start"); >>>> In order to implement this example, the script must wait for the beep >>>> to be detected or not to process the case accordingly. What is no beep is >>>> detected? How can I prevent called party to hear only silence while >>>> potential beep detection is being executed? >>>> >>>> Thank you >>>> >>>> Bernard >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120723/cd53a3cb/attachment-0001.html From fiorix at gmail.com Tue Jul 24 05:11:24 2012 From: fiorix at gmail.com (Alexandre Fiori) Date: Mon, 23 Jul 2012 21:11:24 -0400 Subject: [Freeswitch-users] dialplan commands on playback In-Reply-To: References: <5117A1BD-A549-4091-8F0F-577DE29C747C@gmail.com> <7AB73CC2-06A0-46E2-BA18-8603C208FADD@gmail.com> Message-ID: Sure! I'll let you know Sent from the future On 2012-07-23, at 7:11 PM, Michael Collins wrote: > Excellent! Do you mind finding a spot up on the wiki for this? > -MC > > On Mon, Jul 23, 2012 at 4:01 PM, Alexandre Fiori wrote: > > Amazing. This is exactly what I was looking for, and couldn't figure out how to do. > Thank you! > > > On 2012-07-23, at 6:55 PM, Michael Collins wrote: > >> Well, if you have soxi installed (Linux, of course) then you could always use a system call to grab the file length. For kicks I whipped up a proof of concept dial plan to demonstrate how it could work. If anyone finds it useful please let me know and I'll wikify it. >> -MC >> >> >> >> >> >> > data="sound_file=${sounds_dir}/music/8000/danza-espanola-op-37-h-142-xii-arabesca.wav"/> >> > data="file_length=${system soxi -D ${sound_file} | tr -d '\\n' }"/> >> >> >> >> >> >> >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120723/050fcc16/attachment.html From curriegrad2004 at gmail.com Tue Jul 24 07:23:48 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Mon, 23 Jul 2012 20:23:48 -0700 Subject: [Freeswitch-users] Important Information About Not Reporting Bugs on the Mailing List In-Reply-To: References: Message-ID: We should auto send this email out every week or so... with different subject titles so people WON'T forget to file JIRAs if it is indeed confirmed as a bug On Mon, Jul 23, 2012 at 1:08 PM, Michael Collins wrote: > Hello all, > > We've had yet another thread where someone wasn't sure if there was a but so > they asked on the mailing list. I've written up some simple guidelines and > put them on the world-famous Bugs wiki page: > > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Important_Note.2C_Please_Read > > In short: not only should no one report bugs to the mailing list, no one > should report symptoms to the mailing list and ask, "Is this a bug? Should I > file a Jira?" If you have collected enough information to send an email to > the list and ask whether or not to file a Jira then there's really no point > in asking - just file the Jira. > > If you would like to discuss the symptoms on the mailing list then open the > Jira *first* and then link to it in the message you send to the list. It's > much better to open a Jira and find out later that it's not really a bug > than it is to report a possible bug on the mailing list. Even if something > is flagged as "not a bug" or "works as designed" there is still value in > having the Jira ticket in the database for historical and research purposes. > > For those who have heard this discussion before, please be kind to the noobs > and first-timers who may not appreciate the issues involved. Just point them > to the bugs page. If someone repeatedly reports bugs to the mailing list > then we will deliver the message with less delicacy. > > Thanks again for all your help. Using Jira will help keep Tony's life from > become any more complicated than it already is. > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From fluixab at bellsouth.net Tue Jul 24 07:30:11 2012 From: fluixab at bellsouth.net (Bernard Fluixa) Date: Mon, 23 Jul 2012 23:30:11 -0400 Subject: [Freeswitch-users] mod_avmd In-Reply-To: References: <4211B6CA-C067-4B6D-B271-CC1BB82C0206@bellsouth.net> <0C16CD66-3751-4946-90EC-8AEB03F6F06F@bellsouth.net> <103F647B-4D6C-4E00-BED1-CB2ADB6CAAEA@bellsouth.net> Message-ID: <0157A19E-77D2-4DAD-B9D6-8B9C1EA06985@bellsouth.net> Michael - I'll select syntax highlighting in my next pastebin. I replaced PlayAndGetDigits with session:streamFile and handler now kicks off for both DTMF and beep detection cases. Regarding chapter 7, the PlayAndGetDigits warning you're talking about is at page #139 of the book? Thanks for your help. Bernard On Jul 23, 2012, at 7:37 PM, Michael Collins wrote: > Well posted - nicely done for a self-proclaimed n00b. For better readability you might want to try selecting syntax highlighting, kinda like http://pastebin.freeswitch.org/19580. > > If I had to wager a guess as to what's going on I'd say you didn't get the FreeSWITCH 'bridge book' and read chapter 7. At first blush I'd say that your DTMFs are getting eaten by play_and_get_digits. Maybe try playback instead? And read chapter 7 - there's an example in there about reading digits w/ the input callback function. > > As far as the avmd stuff goes I'm not sure at this point. If I get a sec I'll tinker with it and see if I can replicate. > > -MC > > On Mon, Jul 23, 2012 at 4:03 PM, BF wrote: > It's my very first pastebin. I hope this is what you're expecting: http://pastebin.freeswitch.org/19579 > > On Jul 23, 2012, at 6:40 PM, Michael Collins wrote: > >> Can you pastebin the entire script? >> -MC >> >> On Mon, Jul 23, 2012 at 3:38 PM, Bernard Fluixa wrote: >> Hello, >> >> I can start mod_avmd and it detects voicemail beep when calling my cellphone. However, when I pickup the phone and press a key to use IVR, I would like to stop mod_avmd. I copied the input callback example from FreeSWITCH Wiki. My issue is that my input handler is never triggered, neither when I key a DTMF nor when beep is detected. >> >> 1) My handler is: >> function onInput(s, type, obj) >> if (type == "dtmf") then >> avmd_result = 1; >> if ( debug_mode > 0 ) then >> print("onInput - Human detected"); >> end >> return "break"; >> end >> if (type == "event" ) then >> avmd_result = 2; >> if ( debug_mode > 0 ) then >> print("onInput - Voicemail detected"); >> end >> avmd_result = 2; >> return "break"; >> end >> end >> >> 2) To start mod_avmd, play file and collect digits: >> session:setInputCallback("onInput"); >> session:execute("avmd","start"); >> >> digits = session:playAndGetDigits(??.); >> >> 3) To stop mod_avmd: >> session:execute("avmd","stop"); >> >> 4) Output when keying DTMF (*3 in this case). Input handler does not kick off upon DTMF: >> EXECUTE sofia/external/+19543307528 avmd(start) >> 012-07-23 18:13:28.025981 [INFO] mod_avmd.c:538 <<< AVMD v=0.073060 f=0.900987 1147.172790Hz sma=0.090099 sqa=0.081178 >>> >> ... >> 012-07-23 18:13:30.492866 [INFO] mod_avmd.c:538 <<< AVMD v=0.052551 f=0.764130 972.920758Hz sma=0.076413 sqa=0.058389 >>> >> 2012-07-23 18:13:30.586854 [DEBUG] switch_rtp.c:3795 RTP RECV DTMF *:800 >> 2012-07-23 18:13:30.605660 [INFO] mod_avmd.c:538 <<< AVMD v=0.067196 f=0.864073 1100.171521Hz sma=0.086407 sqa=0.074662 >>> >> ? >> 2012-07-23 18:13:30.998715 [INFO] mod_avmd.c:538 <<< AVMD v=0.030803 f=0.585023 744.874518Hz sma=0.058502 sqa=0.034225 >>> >> 2012-07-23 18:13:31.085676 [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 3:640 >> ... >> 012-07-23 18:13:36.645609 [INFO] mod_avmd.c:538 <<< AVMD v=0.030380 f=0.580996 739.747695Hz sma=0.058100 sqa=0.033756 >>> >> >> 5) Output when beep is detected - Mod_avmd stops output after beep detection but input handler does not kick off >> 2012-07-23 18:04:04.992856 [INFO] mod_avmd.c:538 <<< AVMD v=0.000048 f=0.812668 1034.721179Hz sma=0.811125 sqa=0.657971 >>> >> 2012-07-23 18:04:04.992856 [INFO] mod_avmd.c:561 <<< AVMD - Beep Detected >>> >> >> What am I missing? >> >> Thank you >> >> Bernard >> >> >> >> >> On Jul 23, 2012, at 2:16 PM, Bernard Fluixa wrote: >> >>> OK. I'm clear now. Thanks again. >>> On Jul 23, 2012, at 1:17 PM, Michael Collins wrote: >>> >>>> Precisely. >>>> -MC >>>> >>>> On Mon, Jul 23, 2012 at 9:59 AM, Bernard Fluixa wrote: >>>> Michael, >>>> >>>> Thank you or your response. So it is my responsibility to do whatever needs to be done while mod_avmd attempts to detect a beep and to manually stop it after a beep as been detected or after a certain timeout. Correct? >>>> >>>> Bernard >>>> >>>> >>>> >>>> >>>> On Jul 23, 2012, at 11:59 AM, Michael Collins wrote: >>>> >>>>> Bernard, >>>>> >>>>> Keep in mind that avmd is non-blocking, that is, it won't cause your dialplan or script to pause while it is attempting to detect the beep. Your dialplan will keep doing what it normally does, and if avmd detects a beep then it will throw an event which you catch and handle in your onInput function. >>>>> >>>>> -MC >>>>> >>>>> On Mon, Jul 23, 2012 at 6:47 AM, BF wrote: >>>>> Hello, >>>>> >>>>> I'm trying to use it from a Lua script. My understanding is that mod_avmd detects beep from voicemail systems and is CPU intensive, please correct me if I'm wrong. >>>>> >>>>> The Lua example at FreeSWITCH Wiki is >>>>> local human_detected = false; >>>>> local voicemail_detected = false; >>>>> >>>>> function onInput(session, type, obj) >>>>> if type == "dtmf" and obj['digit'] == '1' and human_detected == false then >>>>> human_detected = true; >>>>> return "break"; >>>>> end >>>>> >>>>> if type == "event" and voicemail_detected == false then >>>>> voicemail_detected = true; >>>>> return "break"; >>>>> end >>>>> end >>>>> >>>>> session:setInputCallback("onInput"); >>>>> session:execute("avmd","start"); >>>>> In order to implement this example, the script must wait for the beep to be detected or not to process the case accordingly. What is no beep is detected? How can I prevent called party to hear only silence while potential beep detection is being executed? >>>>> >>>>> Thank you >>>>> >>>>> Bernard >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> Join Us At ClueCon - Aug 7-9, 2012 >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> >>>>> -- >>>>> Michael S Collins >>>>> Twitter: @mercutioviz >>>>> http://www.FreeSWITCH.org >>>>> http://www.ClueCon.com >>>>> http://www.OSTAG.org >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> Join Us At ClueCon - Aug 7-9, 2012 >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120723/fb744dbc/attachment-0001.html From bdfoster at endigotech.com Tue Jul 24 08:09:28 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 24 Jul 2012 00:09:28 -0400 Subject: [Freeswitch-users] Important Information About Not Reporting Bugs on the Mailing List In-Reply-To: References: Message-ID: No need for an email, bkw just needs to step up his game as the ML cop. NEXT!!! Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 23, 2012 11:25 PM, "curriegrad2004" wrote: > We should auto send this email out every week or so... with different > subject titles so people WON'T forget to file JIRAs if it is indeed > confirmed as a bug > > On Mon, Jul 23, 2012 at 1:08 PM, Michael Collins > wrote: > > Hello all, > > > > We've had yet another thread where someone wasn't sure if there was a > but so > > they asked on the mailing list. I've written up some simple guidelines > and > > put them on the world-famous Bugs wiki page: > > > > > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Important_Note.2C_Please_Read > > > > In short: not only should no one report bugs to the mailing list, no one > > should report symptoms to the mailing list and ask, "Is this a bug? > Should I > > file a Jira?" If you have collected enough information to send an email > to > > the list and ask whether or not to file a Jira then there's really no > point > > in asking - just file the Jira. > > > > If you would like to discuss the symptoms on the mailing list then open > the > > Jira *first* and then link to it in the message you send to the list. > It's > > much better to open a Jira and find out later that it's not really a bug > > than it is to report a possible bug on the mailing list. Even if > something > > is flagged as "not a bug" or "works as designed" there is still value in > > having the Jira ticket in the database for historical and research > purposes. > > > > For those who have heard this discussion before, please be kind to the > noobs > > and first-timers who may not appreciate the issues involved. Just point > them > > to the bugs page. If someone repeatedly reports bugs to the mailing list > > then we will deliver the message with less delicacy. > > > > Thanks again for all your help. Using Jira will help keep Tony's life > from > > become any more complicated than it already is. > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120724/87ea8176/attachment.html From bdfoster at endigotech.com Tue Jul 24 08:11:56 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 24 Jul 2012 00:11:56 -0400 Subject: [Freeswitch-users] mod_avmd In-Reply-To: <0157A19E-77D2-4DAD-B9D6-8B9C1EA06985@bellsouth.net> References: <4211B6CA-C067-4B6D-B271-CC1BB82C0206@bellsouth.net> <0C16CD66-3751-4946-90EC-8AEB03F6F06F@bellsouth.net> <103F647B-4D6C-4E00-BED1-CB2ADB6CAAEA@bellsouth.net> <0157A19E-77D2-4DAD-B9D6-8B9C1EA06985@bellsouth.net> Message-ID: Bernard, welcome to FreeSWITCH! Can you fix my Uverse? Lol Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 23, 2012 11:31 PM, "Bernard Fluixa" wrote: > Michael - I'll select syntax highlighting in my next pastebin. I replaced > PlayAndGetDigits with session:streamFile and handler now kicks off for both > DTMF and beep detection cases. Regarding chapter 7, > the PlayAndGetDigits warning you're talking about is at page #139 of the > book? > > Thanks for your help. > > Bernard > > On Jul 23, 2012, at 7:37 PM, Michael Collins wrote: > > Well posted - nicely done for a self-proclaimed n00b. For better > readability you might want to try selecting syntax highlighting, kinda like > http://pastebin.freeswitch.org/19580. > > If I had to wager a guess as to what's going on I'd say > you didn't get the FreeSWITCH 'bridge book' and read > chapter 7. At first blush I'd say that your DTMFs are > getting eaten by play_and_get_digits. Maybe try playback instead? And read > chapter 7 - there's an example in there about reading digits w/ the input > callback function. > > As far as the avmd stuff goes I'm not sure at this point. If I get a sec > I'll tinker with it and see if I can replicate. > > -MC > > On Mon, Jul 23, 2012 at 4:03 PM, BF wrote: > >> It's my very first pastebin. I hope this is what you're expecting: >> http://pastebin.freeswitch.org/19579 >> >> On Jul 23, 2012, at 6:40 PM, Michael Collins wrote: >> >> Can you pastebin the entire script? >> -MC >> >> On Mon, Jul 23, 2012 at 3:38 PM, Bernard Fluixa wrote: >> >>> Hello, >>> >>> I can start mod_avmd and it detects voicemail beep when calling my >>> cellphone. However, when I pickup the phone and press a key to use IVR, I >>> would like to stop mod_avmd. I copied the input callback example from >>> FreeSWITCH Wiki. My issue is that my input handler is never triggered, >>> neither when I key a DTMF nor when beep is detected. >>> >>> *1) My handler is: * >>> function onInput(s, type, obj) >>> if (type == "dtmf") then >>> avmd_result = 1; >>> if ( debug_mode > 0 ) then >>> print("onInput - Human detected"); >>> end >>> return "break"; >>> end >>> if (type == "event" ) then >>> avmd_result = 2; >>> if ( debug_mode > 0 ) then >>> print("onInput - Voicemail detected"); >>> end >>> avmd_result = 2; >>> return "break"; >>> end >>> end >>> >>> *2) To start mod_avmd, play file and collect digits:* >>> * session:setInputCallback("onInput");* >>> * session:execute("avmd","start");* >>> >>> digits = session:playAndGetDigits(??.); >>> >>> *3) To stop mod_avmd:* >>> session:execute("avmd","stop"); >>> >>> *4) Output when keying DTMF (*3 in this case). Input handler does not >>> kick off upon DTMF: * >>> EXECUTE sofia/external/+19543307528 avmd(start) >>> 012-07-23 18:13:28.025981 [INFO] mod_avmd.c:538 <<< AVMD v=0.073060 >>> f=0.900987 1147.172790Hz sma=0.090099 sqa=0.081178 >>> >>> ... >>> 012-07-23 18:13:30.492866 [INFO] mod_avmd.c:538 <<< AVMD v=0.052551 >>> f=0.764130 972.920758Hz sma=0.076413 sqa=0.058389 >>> >>> 2012-07-23 18:13:30.586854 [DEBUG] switch_rtp.c:3795 RTP RECV DTMF *:800 >>> 2012-07-23 18:13:30.605660 [INFO] mod_avmd.c:538 <<< AVMD v=0.067196 >>> f=0.864073 1100.171521Hz sma=0.086407 sqa=0.074662 >>> >>> ? >>> 2012-07-23 18:13:30.998715 [INFO] mod_avmd.c:538 <<< AVMD v=0.030803 >>> f=0.585023 744.874518Hz sma=0.058502 sqa=0.034225 >>> >>> 2012-07-23 18:13:31.085676 [DEBUG] switch_rtp.c:3795 RTP RECV DTMF 3:640 >>> ... >>> 012-07-23 18:13:36.645609 [INFO] mod_avmd.c:538 <<< AVMD v=0.030380 >>> f=0.580996 739.747695Hz sma=0.058100 sqa=0.033756 >>> >>> >>> *5) Output when beep is detected - Mod_avmd stops output after beep >>> detection but input handler does not kick off* >>> 2012-07-23 18:04:04.992856 [INFO] mod_avmd.c:538 <<< AVMD v=0.000048 >>> f=0.812668 1034.721179Hz sma=0.811125 sqa=0.657971 >>> >>> 2012-07-23 18:04:04.992856 [INFO] mod_avmd.c:561 <<< AVMD - Beep >>> Detected >>> >>> >>> What am I missing? >>> >>> Thank you >>> >>> Bernard >>> >>> >>> >>> >>> On Jul 23, 2012, at 2:16 PM, Bernard Fluixa wrote: >>> >>> OK. I'm clear now. Thanks again. >>> On Jul 23, 2012, at 1:17 PM, Michael Collins wrote: >>> >>> Precisely. >>> -MC >>> >>> On Mon, Jul 23, 2012 at 9:59 AM, Bernard Fluixa wrote: >>> >>>> Michael, >>>> >>>> Thank you or your response. So it is my responsibility to do whatever >>>> needs to be done while mod_avmd attempts to detect a beep and to manually >>>> stop it after a beep as been detected or after a certain timeout. Correct? >>>> >>>> Bernard >>>> >>>> >>>> >>>> >>>> On Jul 23, 2012, at 11:59 AM, Michael Collins wrote: >>>> >>>> Bernard, >>>> >>>> Keep in mind that avmd is non-blocking, that is, it won't cause your >>>> dialplan or script to pause while it is attempting to detect the beep. Your >>>> dialplan will keep doing what it normally does, and if avmd detects a beep >>>> then it will throw an event which you catch and handle in your onInput >>>> function. >>>> >>>> -MC >>>> >>>> On Mon, Jul 23, 2012 at 6:47 AM, BF wrote: >>>> >>>>> Hello, >>>>> >>>>> I'm trying to use it from a Lua script. My understanding is that >>>>> mod_avmd detects beep from voicemail systems and is CPU intensive, please >>>>> correct me if I'm wrong. >>>>> >>>>> The Lua example at FreeSWITCH Wiki is >>>>> local human_detected = false; >>>>> local voicemail_detected = false; >>>>> >>>>> function onInput(session, type, obj) >>>>> if type == "dtmf" and obj['digit'] == '1' and human_detected == >>>>> false then >>>>> human_detected = true; >>>>> return "break"; >>>>> end >>>>> >>>>> if type == "event" and voicemail_detected == false then >>>>> voicemail_detected = true; >>>>> return "break"; >>>>> end >>>>> end >>>>> >>>>> session:setInputCallback("onInput"); >>>>> session:execute("avmd","start"); >>>>> In order to implement this example, the script must wait for the beep >>>>> to be detected or not to process the case accordingly. What is no beep is >>>>> detected? How can I prevent called party to hear only silence while >>>>> potential beep detection is being executed? >>>>> >>>>> Thank you >>>>> >>>>> Bernard >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> Join Us At ClueCon - Aug 7-9, 2012 >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> >>>> >>>> -- >>>> Michael S Collins >>>> Twitter: @mercutioviz >>>> http://www.FreeSWITCH.org >>>> http://www.ClueCon.com >>>> http://www.OSTAG.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120724/41ab6cc4/attachment-0001.html From fiorix at gmail.com Tue Jul 24 08:57:17 2012 From: fiorix at gmail.com (Alexandre Fiori) Date: Tue, 24 Jul 2012 00:57:17 -0400 Subject: [Freeswitch-users] Important Information About Not Reporting Bugs on the Mailing List In-Reply-To: References: Message-ID: <80D31FA8-4A7B-47A7-9CF4-84FC6607AF73@gmail.com> Stop spam Just let us know Sent from the future On 2012-07-24, at 12:09 AM, Brian Foster wrote: > No need for an email, bkw just needs to step up his game as the ML cop. NEXT!!! > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > > On Jul 23, 2012 11:25 PM, "curriegrad2004" wrote: > We should auto send this email out every week or so... with different > subject titles so people WON'T forget to file JIRAs if it is indeed > confirmed as a bug > > On Mon, Jul 23, 2012 at 1:08 PM, Michael Collins wrote: > > Hello all, > > > > We've had yet another thread where someone wasn't sure if there was a but so > > they asked on the mailing list. I've written up some simple guidelines and > > put them on the world-famous Bugs wiki page: > > > > http://wiki.freeswitch.org/wiki/Reporting_Bugs#Important_Note.2C_Please_Read > > > > In short: not only should no one report bugs to the mailing list, no one > > should report symptoms to the mailing list and ask, "Is this a bug? Should I > > file a Jira?" If you have collected enough information to send an email to > > the list and ask whether or not to file a Jira then there's really no point > > in asking - just file the Jira. > > > > If you would like to discuss the symptoms on the mailing list then open the > > Jira *first* and then link to it in the message you send to the list. It's > > much better to open a Jira and find out later that it's not really a bug > > than it is to report a possible bug on the mailing list. Even if something > > is flagged as "not a bug" or "works as designed" there is still value in > > having the Jira ticket in the database for historical and research purposes. > > > > For those who have heard this discussion before, please be kind to the noobs > > and first-timers who may not appreciate the issues involved. Just point them > > to the bugs page. If someone repeatedly reports bugs to the mailing list > > then we will deliver the message with less delicacy. > > > > Thanks again for all your help. Using Jira will help keep Tony's life from > > become any more complicated than it already is. > > > > -- > > Michael S Collins > > Twitter: @mercutioviz > > http://www.FreeSWITCH.org > > http://www.ClueCon.com > > http://www.OSTAG.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120724/cb1698c1/attachment.html From anton.jugatsu at gmail.com Tue Jul 24 10:19:27 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Tue, 24 Jul 2012 10:19:27 +0400 Subject: [Freeswitch-users] No audio after upgrade to F17 In-Reply-To: <1343071225.4358.10.camel@mythtv> References: <1343071225.4358.10.camel@mythtv> Message-ID: It defenitily firewall issue. Try to stop iptables service with service iptables stop. 2012/7/23 Todd > Well this is odd, after I installed F17, I don't get audio to or from > the pstn line (via a spa3102 ata) once the call is answered. > > I do get ring tones, and Fs generated voice messages, and can leave or > retrieve vm. > > I've tried a couple approaches so. > > When I installed F17, I copied the FS installation directly > > Booting back to F14, every this works as expected, so it's not the > spa3102 or config files. > > Thinking there might be a incompatibility issues, I recompiled FS under > F17, no change in operational behavior. > > Ideas where else to look ? > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120724/00dc5c9d/attachment.html From alex at thewinelake.com Tue Jul 24 13:03:06 2012 From: alex at thewinelake.com (Alex) Date: Tue, 24 Jul 2012 10:03:06 +0100 Subject: [Freeswitch-users] Setting built-in variables In-Reply-To: References: <5117A1BD-A549-4091-8F0F-577DE29C747C@gmail.com> <7AB73CC2-06A0-46E2-BA18-8603C208FADD@gmail.com> Message-ID: <500E64CA.2070407@thewinelake.com> I want to be able to sanitize the destination_number channel variable But it seems that setting it doesn't quite work (because it's a "built-in variable" or some misuse of inline=true Although it seems to be doing the right thing, it doesn't seem to be. Lower down the file, I have And when I place a call to +44123456789 the log records EXECUTE sofia/internal/0095301 at vrhnov11.dmclub.net set(destination_number=0044123456789) 2012-07-24 08:59:03.611635 [DEBUG] mod_dptools.c:1305 sofia/internal/0095301 at vrhnov11.dmclub.net SET [destination_number]=[0044123456789] Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net Action log(WARNING destination_number now ${destination_number}) Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net Regex (FAIL) [Outbound_call] ${default_ani_prefix}(00953010) =~ /^$/ break=never Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net Regex (FAIL) [Outbound_call] ${sip_auth_username}(0095301) =~ /^$/ break=never Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net Regex (FAIL) [Outbound_call] ${default_ani_prefix}(00953010) =~ /^$/ break=never Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net ANTI-Action set(ani_prefix=${default_ani_prefix}) INLINE EXECUTE sofia/internal/0095301 at vrhnov11.dmclub.net set(ani_prefix=00953010) 2012-07-24 08:59:03.611635 [DEBUG] mod_dptools.c:1305 sofia/internal/0095301 at vrhnov11.dmclub.net SET [ani_prefix]=[00953010] Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net Regex (FAIL) [Outbound_call] ${ani_prefix}(00953010) =~ /^(0)$/ break=never Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net Regex (FAIL) [Outbound_call] destination_number(+44123456789) =~ /^(\d{8}.*)/ break=on-false Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net ANTI-Action log(WARNING ${destination_number} not a long number) 2012-07-24 08:59:03.611635 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/0095301 at vrhnov11.dmclub.net) State Change CS_ROUTING -> CS_EXECUTE 2012-07-24 08:59:03.611635 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/0095301 at vrhnov11.dmclub.net [BREAK] 2012-07-24 08:59:03.611635 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/0095301 at vrhnov11.dmclub.net) State ROUTING going to sleep 2012-07-24 08:59:03.611635 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/0095301 at vrhnov11.dmclub.net) Running State Change CS_EXECUTE 2012-07-24 08:59:03.611635 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/0095301 at vrhnov11.dmclub.net) State EXECUTE 2012-07-24 08:59:03.611635 [DEBUG] mod_sofia.c:241 sofia/internal/0095301 at vrhnov11.dmclub.net SOFIA EXECUTE 2012-07-24 08:59:03.611635 [DEBUG] switch_core_state_machine.c:196 sofia/internal/0095301 at vrhnov11.dmclub.net Standard EXECUTE So something odd going on here - probably easy for the experts to see my mistake! From alex at thewinelake.com Tue Jul 24 13:11:23 2012 From: alex at thewinelake.com (Alex) Date: Tue, 24 Jul 2012 10:11:23 +0100 Subject: [Freeswitch-users] Setting built-in variables In-Reply-To: <500E64CA.2070407@thewinelake.com> References: <5117A1BD-A549-4091-8F0F-577DE29C747C@gmail.com> <7AB73CC2-06A0-46E2-BA18-8603C208FADD@gmail.com> <500E64CA.2070407@thewinelake.com> Message-ID: <500E66BB.8000202@thewinelake.com> Apologies about the formatting of that - quite horrible. Not sure why, though. Let's try again... Although it seems to be doing the right thing, it isn't. Lower down the file, I have And when I place a call to +44123456789 the log records EXECUTE sofia/internal/0095301 at vrhnov11.dmclub.net set(destination_number=0044123456789) 2012-07-24 08:59:03.611635 [DEBUG] mod_dptools.c:1305 sofia/internal/0095301 at vrhnov11.dmclub.net SET [destination_number]=[0044123456789] Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net Action log(WARNING destination_number now ${destination_number}) Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net Regex (FAIL) [Outbound_call] ${default_ani_prefix}(00953010) =~ /^$/ break=never Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net Regex (FAIL) [Outbound_call] ${sip_auth_username}(0095301) =~ /^$/ break=never Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net Regex (FAIL) [Outbound_call] ${default_ani_prefix}(00953010) =~ /^$/ break=never Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net ANTI-Action set(ani_prefix=${default_ani_prefix}) INLINE EXECUTE sofia/internal/0095301 at vrhnov11.dmclub.net set(ani_prefix=00953010) 2012-07-24 08:59:03.611635 [DEBUG] mod_dptools.c:1305 sofia/internal/0095301 at vrhnov11.dmclub.net SET [ani_prefix]=[00953010] Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net Regex (FAIL) [Outbound_call] ${ani_prefix}(00953010) =~ /^(0)$/ break=never Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net Regex (FAIL) [Outbound_call] destination_number(+44123456789) =~ /^(\d{8}.*)/ break=on-false Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net ANTI-Action log(WARNING ${destination_number} not a long number) 2012-07-24 08:59:03.611635 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/0095301 at vrhnov11.dmclub.net) State Change CS_ROUTING -> CS_EXECUTE 2012-07-24 08:59:03.611635 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/0095301 at vrhnov11.dmclub.net [BREAK] 2012-07-24 08:59:03.611635 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/0095301 at vrhnov11.dmclub.net) State ROUTING going to sleep 2012-07-24 08:59:03.611635 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/0095301 at vrhnov11.dmclub.net) Running State Change CS_EXECUTE 2012-07-24 08:59:03.611635 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/0095301 at vrhnov11.dmclub.net) State EXECUTE 2012-07-24 08:59:03.611635 [DEBUG] mod_sofia.c:241 sofia/internal/0095301 at vrhnov11.dmclub.net SOFIA EXECUTE 2012-07-24 08:59:03.611635 [DEBUG] switch_core_state_machine.c:196 sofia/internal/0095301 at vrhnov11.dmclub.net Standard EXECUTE So something odd going on here - probably easy for the experts to see my mistake! From alex at thewinelake.com Tue Jul 24 14:34:03 2012 From: alex at thewinelake.com (Alex) Date: Tue, 24 Jul 2012 11:34:03 +0100 Subject: [Freeswitch-users] Setting built-in variables In-Reply-To: <500E66BB.8000202@thewinelake.com> References: <5117A1BD-A549-4091-8F0F-577DE29C747C@gmail.com> <7AB73CC2-06A0-46E2-BA18-8603C208FADD@gmail.com> <500E64CA.2070407@thewinelake.com> <500E66BB.8000202@thewinelake.com> Message-ID: <500E7A1B.9060707@thewinelake.com> Just found this http://lists.freeswitch.org/pipermail/freeswitch-users/2011-May/072338.html Which seems a useful tip > Apologies about the formatting of that - quite horrible. Not sure why, > though. Let's try again... > > > > > inline="true"/> > > > > Although it seems to be doing the right thing, it isn't. Lower down the > file, I have > > > > data="[tenant_id=${tenant_id},a_ext=${a_ext},b_ext=${b_ext},origination_callee_id_number=$1,origination_caller_id_number=${ani_prefix}${ani}]sofia/internal/8980000000002$1 at 193.105.54.10"/> > > > > > > > And when I place a call to +44123456789 the log records > > > EXECUTE sofia/internal/0095301 at vrhnov11.dmclub.net > set(destination_number=0044123456789) > 2012-07-24 08:59:03.611635 [DEBUG] mod_dptools.c:1305 > sofia/internal/0095301 at vrhnov11.dmclub.net SET > [destination_number]=[0044123456789] > Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net Action log(WARNING > destination_number now ${destination_number}) > Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net Regex (FAIL) > [Outbound_call] ${default_ani_prefix}(00953010) =~ /^$/ break=never > Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net Regex (FAIL) > [Outbound_call] ${sip_auth_username}(0095301) =~ /^$/ break=never > Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net Regex (FAIL) > [Outbound_call] ${default_ani_prefix}(00953010) =~ /^$/ break=never > Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net ANTI-Action > set(ani_prefix=${default_ani_prefix}) INLINE > EXECUTE sofia/internal/0095301 at vrhnov11.dmclub.net set(ani_prefix=00953010) > 2012-07-24 08:59:03.611635 [DEBUG] mod_dptools.c:1305 > sofia/internal/0095301 at vrhnov11.dmclub.net SET [ani_prefix]=[00953010] > Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net Regex (FAIL) > [Outbound_call] ${ani_prefix}(00953010) =~ /^(0)$/ break=never > Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net Regex (FAIL) > [Outbound_call] destination_number(+44123456789) =~ /^(\d{8}.*)/ > break=on-false > Dialplan: sofia/internal/0095301 at vrhnov11.dmclub.net ANTI-Action > log(WARNING ${destination_number} not a long number) > 2012-07-24 08:59:03.611635 [DEBUG] switch_core_state_machine.c:154 > (sofia/internal/0095301 at vrhnov11.dmclub.net) State Change CS_ROUTING -> > CS_EXECUTE > 2012-07-24 08:59:03.611635 [DEBUG] switch_core_session.c:1228 Send > signal sofia/internal/0095301 at vrhnov11.dmclub.net [BREAK] > 2012-07-24 08:59:03.611635 [DEBUG] switch_core_state_machine.c:433 > (sofia/internal/0095301 at vrhnov11.dmclub.net) State ROUTING going to sleep > 2012-07-24 08:59:03.611635 [DEBUG] switch_core_state_machine.c:385 > (sofia/internal/0095301 at vrhnov11.dmclub.net) Running State Change > CS_EXECUTE > 2012-07-24 08:59:03.611635 [DEBUG] switch_core_state_machine.c:440 > (sofia/internal/0095301 at vrhnov11.dmclub.net) State EXECUTE > 2012-07-24 08:59:03.611635 [DEBUG] mod_sofia.c:241 > sofia/internal/0095301 at vrhnov11.dmclub.net SOFIA EXECUTE > 2012-07-24 08:59:03.611635 [DEBUG] switch_core_state_machine.c:196 > sofia/internal/0095301 at vrhnov11.dmclub.net Standard EXECUTE > > So something odd going on here - probably easy for the experts to see my > mistake! > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2196 / Virus Database: 2437/5143 - Release Date: 07/20/12 > > From christian.tausch at fh-koeln.de Tue Jul 24 13:20:10 2012 From: christian.tausch at fh-koeln.de (Christian Tausch) Date: Tue, 24 Jul 2012 11:20:10 +0200 Subject: [Freeswitch-users] error 403 registering at 1und1.de Message-ID: <500E68CA.1040300@fh-koeln.de> Hi list, i ran into a problem configuring the german voip-provider 1und1.de as a gateway. I have not completed my configuration yet, but in a first test the registration fails: The response to the second REGISTER message is "403: Contact User und Anrufernummer verschieden" (en: 403 Authentication User Name does not match account name). The visible difference between the REGISTER of freeswitch and the (working) SIP-client twinkle is indeed the user-string in the contact-header. Freeswitch: (gw + name) Contact: :5080;transport=udp;gw=1und1.de> Twinkle: (extension / login-user) Contact: ;transport=udp>;expires=3600 If this is the cause of the error message, is this expected behavior of a provider? And more important is it possible to configure freeswitch to set a specific contact-user? Thank you for your time. Christian Tausch -- Attachment (conf/directory/default/1und1.de.xml): From martyn at magiccow.co.uk Tue Jul 24 14:48:54 2012 From: martyn at magiccow.co.uk (Martyn Davies) Date: Tue, 24 Jul 2012 11:48:54 +0100 Subject: [Freeswitch-users] Testing iSAC Message-ID: I have an iSAC client that I'm testing against FreeSwitch. Currently I have my vars.xml with: global_codec_prefs=isac,PCMU,PCMA,GSM,G722 Which works, but only if the client specifies 32kHz mode. I want to test 16kHz mode, but FreeSwitch returns 'not acceptable here', and I can see that when Freeswitch offers iSAC in the SDP it says only isac/32000. So how to enable 16kHz mode for isac? I tried explicitly putting isac at 320000h30i, isac at 16000h30i into vars.xml, but this simply stops FreeSwitch from accepting anything. Hints gladly received. Regards, Martyn -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120724/1fcc2117/attachment.html From g.d.monnezza at tiscali.it Tue Jul 24 16:12:08 2012 From: g.d.monnezza at tiscali.it (VirteX) Date: Tue, 24 Jul 2012 05:12:08 -0700 (PDT) Subject: [Freeswitch-users] AutoNAT - Local Networks not excluded Message-ID: <34201844.post@talk.nabble.com> Hi guys. I appreciate so much the Auto-NAT for uPnP capable firewalls. But I'm experiencing an issue. I have a FreeSwitch server behind a NAT, but I can't find a way to avoid FreeSwitch using external IP (for SIP and RTP) for local networks (i.e. 192.168.0.0/16). In my sip profiles for various interfaces I have NOT set the . Anyway, the sofia status for all interfaces shows the EXT-RTP-IP and EXT-SIP-IP set (with my public gateway IP). That's ok, even if I didn' declard it with My SIP phones register from a network different from the server one, but still a local network. Then, SIP phones receive (from the server) the rtp and sip signalling with its external IP. This prevent any communication. How it is possible to tell FreeSwitch to NOT use ext IP for particular networks? Thanks to anyone who will point me in the right direction. g -- View this message in context: http://old.nabble.com/AutoNAT---Local-Networks-not-excluded-tp34201844p34201844.html Sent from the Freeswitch-users mailing list archive at Nabble.com. From frank at rosengart.de Tue Jul 24 18:49:29 2012 From: frank at rosengart.de (Frank Rosengart) Date: Tue, 24 Jul 2012 16:49:29 +0200 Subject: [Freeswitch-users] AOC for SIP Message-ID: <500EB5F9.6070206@rosengart.de> Hi, I'm wondering if anyone has implemented some kind of AOC (advice of charge) for SIP channels? This is either a frequent INFO message during the call or an extra to the BYE message. This includes pricing calculation based on call duration and destination (can be handled externally) and sending out the AOC message (either in the Snom-way or the Patton-dialect). Thanks! Frank http://wiki.snom.com/wiki/index.php/Advice_of_charge_%28AOC%29_in_SIP From bdfoster at endigotech.com Tue Jul 24 19:11:40 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 24 Jul 2012 11:11:40 -0400 Subject: [Freeswitch-users] error 403 registering at 1und1.de In-Reply-To: <500E68CA.1040300@fh-koeln.de> References: <500E68CA.1040300@fh-koeln.de> Message-ID: Take a peek in sip_profiles/external for a gateway example. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 24, 2012 9:26 AM, "Christian Tausch" wrote: > Hi list, > > i ran into a problem configuring the german voip-provider 1und1.de as a > gateway. I have not completed my configuration yet, but in a first test > the registration fails: > > The response to the second REGISTER message is "403: Contact User und > Anrufernummer verschieden" (en: 403 Authentication User Name does not > match account name). > > The visible difference between the REGISTER of freeswitch and the > (working) SIP-client twinkle is indeed the user-string in the > contact-header. > > Freeswitch: (gw + name) > Contact: :5080;transport=udp;gw=1und1.de> > > Twinkle: (extension / login-user) > Contact: ;transport=udp>;expires=3600 > > If this is the cause of the error message, is this expected behavior of > a provider? And more important is it possible to configure freeswitch to > set a specific contact-user? > > Thank you for your time. > > Christian Tausch > > -- > > Attachment (conf/directory/default/1und1.de.xml): > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120724/d6065fa1/attachment.html From toddb at toddbailey.net Tue Jul 24 20:21:23 2012 From: toddb at toddbailey.net (Todd Bailey) Date: Tue, 24 Jul 2012 09:21:23 -0700 Subject: [Freeswitch-users] No audio after upgrade to F17 In-Reply-To: <1343071225.4358.10.camel@mythtv> References: <1343071225.4358.10.camel@mythtv> Message-ID: <1343146883.22964.5.camel@mythtv> On Mon, 2012-07-23 at 12:20 -0700, Todd wrote: > Well this is odd, after I installed F17, I don't get audio to or from > the pstn line (via a spa3102 ata) once the call is answered. > > I do get ring tones, and Fs generated voice messages, and can leave or > retrieve vm. > > I've tried a couple approaches so. > > When I installed F17, I copied the FS installation directly > > Booting back to F14, every this works as expected, so it's not the > spa3102 or config files. > > Thinking there might be a incompatibility issues, I recompiled FS under > F17, no change in operational behavior. > > Ideas where else to look ? > > > > > From philq at qsystemsengineering.com Tue Jul 24 21:25:45 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Tue, 24 Jul 2012 13:25:45 -0400 Subject: [Freeswitch-users] Bypass media and NAT Traversal with Aastra Message-ID: <018f01cd69c1$5ef547b0$1cdfd710$@com> Hi, Is this a bug? (just kidding) As the number of extensions we serve increases, we're trying to avoid saturating our local bandwidth by proxying all of the media between endpoints. This seems to work fine with the Linksys devices but not with Aastra in all scenarios. The Aastras are configured with the NAT IP field populated appropriately and Rport (RFC 3581) enabled. The FS server is behind a NAT firewall with SIP and RTP port ranges forwarded to it. There are endpoints here on the same network as the switch, and there are remote endpoints behind their own NAT firewalls. Everything works fine in Proxy Media mode but in bypass mode we see the following behavior: Scenario 1: Remote endpoint places a call to PSTN, FS negotiates bypass media between remote endpoint and PSTN gateway. Call succeeds. Scenario 2: Remote endpoint places a call to another remote endpoint on same LAN. Call succeeds. Scenario 3: Local endpoint places a call to PSTN gateway. Call succeeds. Scenario 4: Remote endpoint places a call to local endpoint (or vice versa). Call fails with no audio with Aastra devices on both ends, but if one of the endpoints is a LinkSys, the call succeeds. I realize that I should be able to solve this problem with a local SBC like OpenSIPS/RTPproxy, but in the meantime I was hoping that there might be a setting that I was overlooking with the Aastra. When I look at the SIP traffic for the call, FS appears to be negotiating the addresses/ports properly so I suspect that this problem is related to Aastra's inability to deal with symmetric NAT. The Linksys devices seem to be able to deal with it with FS' help, presumably by doing some UDP hole-punching. Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120724/7effb36d/attachment.html From paul at cupis.co.uk Tue Jul 24 21:56:12 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Tue, 24 Jul 2012 18:56:12 +0100 Subject: [Freeswitch-users] No audio after upgrade to F17 In-Reply-To: <1343146883.22964.5.camel@mythtv> References: <1343071225.4358.10.camel@mythtv> <1343146883.22964.5.camel@mythtv> Message-ID: <500EE1BC.3090206@cupis.co.uk> On 24/07/12 17:21, Todd Bailey wrote: > On Mon, 2012-07-23 at 12:20 -0700, Todd wrote: >> When I installed F17, I copied the FS installation directly >> >> Booting back to F14, every this works as expected, so it's not the >> spa3102 or config files. >> >> Thinking there might be a incompatibility issues, I recompiled FS under >> F17, no change in operational behavior. >> >> Ideas where else to look ? Have a look at the "new features" in F15, F16 or F17, you'll probably find some security function (SElinux or similar) which may be blocking FreeSWITCH RTP traffic. Regards, From philq at qsystemsengineering.com Tue Jul 24 23:18:35 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Tue, 24 Jul 2012 15:18:35 -0400 Subject: [Freeswitch-users] AutoNAT - Local Networks not excluded Message-ID: <01df01cd69d1$231b3780$6951a680$@com> Set rtp-ip and sip-ip to your internal IP address. I believe that you should also be able to set it to: $${local_ip_v4} or $${bind_server_ip} as well. - Phil ---------- VirteX g.d.monnezza at tiscali.it Tue Jul 24 16:12:08 MSD 2012 Hi guys. I appreciate so much the Auto-NAT for uPnP capable firewalls. But I'm experiencing an issue. I have a FreeSwitch server behind a NAT, but I can't find a way to avoid FreeSwitch using external IP (for SIP and RTP) for local networks (i.e. 192.168.0.0/16). In my sip profiles for various interfaces I have NOT set the . Anyway, the sofia status for all interfaces shows the EXT-RTP-IP and EXT-SIP-IP set (with my public gateway IP). That's ok, even if I didn' declard it with My SIP phones register from a network different from the server one, but still a local network. Then, SIP phones receive (from the server) the rtp and sip signalling with its external IP. This prevent any communication. How it is possible to tell FreeSwitch to NOT use ext IP for particular networks? Thanks to anyone who will point me in the right direction. g -- View this message in context: http://old.nabble.com/AutoNAT---Local-Networks-not-excluded-tp34201844p34201 844.html Sent from the Freeswitch-users mailing list archive at Nabble.com. Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120724/68923fa1/attachment.html From bdfoster at endigotech.com Tue Jul 24 23:27:21 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 24 Jul 2012 15:27:21 -0400 Subject: [Freeswitch-users] Bypass media and NAT Traversal with Aastra In-Reply-To: <018f01cd69c1$5ef547b0$1cdfd710$@com> References: <018f01cd69c1$5ef547b0$1cdfd710$@com> Message-ID: don't proxy media when dealing with NAT. Set up a profile specifically for remote endpoints that doesn't bypass media. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 24, 2012 1:28 PM, "Phil Quesinberry" wrote: > ** > > Hi, > > Is this a bug? (just kidding) > > As the number of extensions we serve increases, we?re trying to avoid saturating > our local bandwidth by proxying all of the media between endpoints. This > seems to work fine with the Linksys devices but not with Aastra in all > scenarios. The Aastras are configured with the NAT IP field populatedappropriatelyand Rport (RFC 3581) enabled. > > The FS server is behind a NAT firewall with SIP and RTP port rangesforwarded to it. There are endpoints here on > the same network as the switch, and there are remote endpoints behind their > own NAT firewalls. Everything works fine in Proxy Media mode but in > bypass mode we see the following behavior: > > Scenario 1: > > Remote endpoint places a call to PSTN, FS negotiates bypass media between > remote endpoint and PSTN gateway. Call succeeds. > > Scenario 2: > > Remote endpoint places a call to another remote endpoint on same LAN. > Call succeeds. > > Scenario 3: > > Local endpoint places a call to PSTN gateway. Call succeeds. > > Scenario 4: > > Remote endpoint places a call to local endpoint (or vice versa). Callfails with no audiowith Aastra deviceson both ends, > but if one of the endpoints is a LinkSys, the call succeeds. > > I realize that I should be able to solve this problem with a local SBClike OpenSIPS/RTPproxy > , but in the meantime I was hoping that there might be a setting that I > was overlooking with the Aastra. When I look at the SIP traffic for the > call, FS appears to be negotiating the addresses/ports properly so Isuspect that this problem is related to > Aastra?s inability to deal with symmetric NAT. The Linksys devices seem > to be able to deal with it with FS? help, presumably by doing some UDP > hole-punching. > > *******Phil Quesinberry* > > Q Systems Engineering, Inc. > > Electronic Controls and Embedded Systems Development > > (410) 969-8002 > > *****http://www.qsystemsengineering.com* > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120724/f8607166/attachment-0001.html From chris at opencsta.org Wed Jul 25 03:23:47 2012 From: chris at opencsta.org (Chris Mylonas) Date: Wed, 25 Jul 2012 09:23:47 +1000 Subject: [Freeswitch-users] No audio after upgrade to F17 In-Reply-To: <500EE1BC.3090206@cupis.co.uk> References: <1343071225.4358.10.camel@mythtv> <1343146883.22964.5.camel@mythtv> <500EE1BC.3090206@cupis.co.uk> Message-ID: Further to Paul's comments In a terminal tail -f /var/log/messages In another terminal tail -f /path/to/your/freeswitch/log/freeswitch.log In another terminal, start freeswitch If it is an SELinux problem, which is probably suspected, there will be a kernel message stating something along the lines of a context is not allowed to do something. I had similar issues with openvpn and SELinux - this blog post may be useful in the process of keeping SELinux running and customising it's profile for running freeswitch. http://www.mrvoip.com.au/blog/selinux-openvpn Alternatively, look at ways of disabling SELinux - on other redhat related distros, it's a matter of editing a setting in a file to "disabled" and rebooting so the kernel doesn't load SELinux stuff. Another alternative is to change SELinux's mode from "enforcing" to "permissive" Cheers Chris On 25/07/2012, at 3:56 AM, Paul Cupis wrote: > On 24/07/12 17:21, Todd Bailey wrote: >> On Mon, 2012-07-23 at 12:20 -0700, Todd wrote: >>> When I installed F17, I copied the FS installation directly >>> >>> Booting back to F14, every this works as expected, so it's not the >>> spa3102 or config files. >>> >>> Thinking there might be a incompatibility issues, I recompiled FS under >>> F17, no change in operational behavior. >>> >>> Ideas where else to look ? > > Have a look at the "new features" in F15, F16 or F17, you'll probably > find some security function (SElinux or similar) which may be blocking > FreeSWITCH RTP traffic. > > Regards, > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120725/fddc5ff4/attachment.html From curriegrad2004 at gmail.com Wed Jul 25 07:45:34 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 24 Jul 2012 20:45:34 -0700 Subject: [Freeswitch-users] Testing iSAC In-Reply-To: References: Message-ID: You might want to report this issue in the JIRA as a feature request. Iirc it doesn't support that yet. On Jul 24, 2012 6:26 AM, "Martyn Davies" wrote: > I have an iSAC client that I'm testing against FreeSwitch. > > Currently I have my vars.xml with: > > global_codec_prefs=isac,PCMU,PCMA,GSM,G722 > > Which works, but only if the client specifies 32kHz mode. I want to test > 16kHz mode, but FreeSwitch returns 'not acceptable here', and I can see > that when Freeswitch offers iSAC in the SDP it says only isac/32000. > > So how to enable 16kHz mode for isac? > > I tried explicitly putting isac at 320000h30i, isac at 16000h30i into vars.xml, > but this simply stops FreeSwitch from accepting anything. > > Hints gladly received. > > Regards, > Martyn > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120724/46f4e325/attachment.html From evgeniy at bestnet.kharkov.ua Wed Jul 25 11:12:18 2012 From: evgeniy at bestnet.kharkov.ua (Evgeniy Movlyan) Date: Wed, 25 Jul 2012 10:12:18 +0300 Subject: [Freeswitch-users] mod_spandsp compilation error Message-ID: <500F9C52.4020503@bestnet.kharkov.ua> Hi to all, i made make current command today, and got this: making all mod_spandsp make[5]: Entering directory `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' Creating mod_spandsp_la-mod_spandsp.lo quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../../../../src/include -I../../../../libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT mod_spandsp_la-mod_spandsp.lo -MD -MP -MF .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o .libs/mod_spandsp_la-mod_spandsp.o In file included from /usr/local/src/freeswitch/libs/spandsp/src/spandsp/expose.h:83, from /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:141, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:54: error: expected specifier-qualifier-list before ?lab_params_t? /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:72: error: expected specifier-qualifier-list before ?lab_params_t? make[5]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1 make[5]: Leaving directory `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' make[4]: *** [mod_spandsp-all] Error 1 make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/local/src/freeswitch/src' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/local/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [current] Error 2 Any ideas? -- Evgeniy Movlyan, BestNet Ltd. From peter.olsson at visionutveckling.se Wed Jul 25 11:42:30 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 25 Jul 2012 07:42:30 +0000 Subject: [Freeswitch-users] mod_spandsp compilation error In-Reply-To: <500F9C52.4020503@bestnet.kharkov.ua> References: <500F9C52.4020503@bestnet.kharkov.ua> Message-ID: <09E40FBA-F194-478D-83B4-28E3B306E340@visionutveckling.se> 'make spandsp-reconf'. If it doesn't help, file a Jira. /Peter 25 jul 2012 kl. 09:19 skrev "Evgeniy Movlyan" : > Hi to all, > i made make current command today, and got this: > > making all mod_spandsp > make[5]: Entering directory > `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' > Creating mod_spandsp_la-mod_spandsp.lo > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. > -I../../../../src/include -I../../../../libs/xmlrpc-c > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g > -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic > -Wdeclaration-after-statement > -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff > -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT > mod_spandsp_la-mod_spandsp.lo -MD -MP -MF > .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o > .libs/mod_spandsp_la-mod_spandsp.o > In file included from > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/expose.h:83, > from > /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:141, > from mod_spandsp.h:50, > from mod_spandsp.c:39: > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:54: > error: expected specifier-qualifier-list before ?lab_params_t? > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:72: > error: expected specifier-qualifier-list before ?lab_params_t? > make[5]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1 > make[5]: Leaving directory > `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' > make[4]: *** [mod_spandsp-all] Error 1 > make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/usr/local/src/freeswitch/src' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/usr/local/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/usr/local/src/freeswitch' > make: *** [current] Error 2 > > Any ideas? > > -- > Evgeniy Movlyan, > BestNet Ltd. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:500f9ab132761590495690! > From rdmitry0911 at gmail.com Wed Jul 25 11:34:28 2012 From: rdmitry0911 at gmail.com (Dmitry R) Date: Wed, 25 Jul 2012 11:34:28 +0400 Subject: [Freeswitch-users] mod_spandsp compilation error In-Reply-To: <500F9C52.4020503@bestnet.kharkov.ua> Message-ID: I got the same problem. Solved after git pull && ./bootstrap.sh && ./configure && make -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Evgeniy Movlyan Sent: Wednesday, July 25, 2012 11:12 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] mod_spandsp compilation error Hi to all, i made make current command today, and got this: making all mod_spandsp make[5]: Entering directory `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' Creating mod_spandsp_la-mod_spandsp.lo quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../../../../src/include -I../../../../libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT mod_spandsp_la-mod_spandsp.lo -MD -MP -MF .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o .libs/mod_spandsp_la-mod_spandsp.o In file included from /usr/local/src/freeswitch/libs/spandsp/src/spandsp/expose.h:83, from /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:141, from mod_spandsp.h:50, from mod_spandsp.c:39: /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:54: error: expected specifier-qualifier-list before 'lab_params_t' /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:72: error: expected specifier-qualifier-list before 'lab_params_t' make[5]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1 make[5]: Leaving directory `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' make[4]: *** [mod_spandsp-all] Error 1 make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/local/src/freeswitch/src' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/local/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [current] Error 2 Any ideas? -- Evgeniy Movlyan, BestNet Ltd. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From alex at thewinelake.com Wed Jul 25 12:06:53 2012 From: alex at thewinelake.com (Alex) Date: Wed, 25 Jul 2012 09:06:53 +0100 Subject: [Freeswitch-users] leg_delay_start does not work in bridge enterprise In-Reply-To: <458A0010-EC8C-4ACC-B08D-A02E0AD26077@mgtech.com> References: <9D0E55F4-768E-4C9D-B1DF-9D074E44ADD7@mgtech.com> <458A0010-EC8C-4ACC-B08D-A02E0AD26077@mgtech.com> Message-ID: <500FA91D.1080503@thewinelake.com> Also suffering from this problem. I guess what you're saying is that it only works in single-threaded mode due to the way in which the legs communicate with eachother. Any workarounds, I wonder? I guess we'll have to get a lua script written for this. > I thought that might be the case. OK then we need an enhancement request to FreeSwitch I guess. Thanks! > > On Jul 19, 2012, at 12:02 PM, Mitch Capper wrote: > >> leg_delay_start is meant to not wait if there are no other legs it is >> waiting on, this was designed to try a local user for example before >> trying their cell phone, but if they are not registered it goes right >> to the secondary leg. >> >> ~Mitch >> >> On Thu, Jul 19, 2012 at 1:01 PM, Mario G wrote: >>> leg_delay_start is ignored when using the enterprise syntax below. I need >>> the second target delayed and there appears no other way to do this. Both >>> targets are called at the same time. >>> >>> Mario G >>> >>> >> data="{originate_timeout=60}${group_call(main@${domain_name}+A)}:_:{leg_delay_start=20,leg_timeout=23}sofia/gateway/${dial_gateway}/19161234567"/> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2196 / Virus Database: 2437/5142 - Release Date: 07/19/12 > > From alex at thewinelake.com Wed Jul 25 12:09:30 2012 From: alex at thewinelake.com (Alex) Date: Wed, 25 Jul 2012 09:09:30 +0100 Subject: [Freeswitch-users] leg_delay_start does not work in bridge enterprise In-Reply-To: <458A0010-EC8C-4ACC-B08D-A02E0AD26077@mgtech.com> References: <9D0E55F4-768E-4C9D-B1DF-9D074E44ADD7@mgtech.com> <458A0010-EC8C-4ACC-B08D-A02E0AD26077@mgtech.com> Message-ID: <500FA9BA.5030303@thewinelake.com> Also suffering from this problem and it's a biggie for us. I suppose the problem is due to the way in which the legs communicate. Looks like using enterprise originate was just a little too optimistic! Wish I'd known about this sooner and could have got on with writing our own lua script that can do the job better. If anyone's done this already and prepared to share, do get in touch. From evgeniy at bestnet.kharkov.ua Wed Jul 25 12:47:01 2012 From: evgeniy at bestnet.kharkov.ua (Evgeniy Movlyan) Date: Wed, 25 Jul 2012 11:47:01 +0300 Subject: [Freeswitch-users] mod_spandsp compilation error In-Reply-To: <09E40FBA-F194-478D-83B4-28E3B306E340@visionutveckling.se> References: <500F9C52.4020503@bestnet.kharkov.ua> <09E40FBA-F194-478D-83B4-28E3B306E340@visionutveckling.se> Message-ID: <500FB285.7070605@bestnet.kharkov.ua> Thanks, Peter. My problem is solved. 25.07.2012 10:42, Peter Olsson ?????: > 'make spandsp-reconf'. If it doesn't help, file a Jira. > > /Peter > > 25 jul 2012 kl. 09:19 skrev "Evgeniy Movlyan": > >> Hi to all, >> i made make current command today, and got this: >> >> making all mod_spandsp >> make[5]: Entering directory >> `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' >> Creating mod_spandsp_la-mod_spandsp.lo >> quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. >> -I../../../../src/include -I../../../../libs/xmlrpc-c >> -I/usr/local/src/freeswitch/libs/curl/include >> -I/usr/local/src/freeswitch/src/include >> -I/usr/local/src/freeswitch/src/include >> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC >> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 >> -I/usr/local/src/freeswitch/libs/curl/include >> -I/usr/local/src/freeswitch/src/include >> -I/usr/local/src/freeswitch/src/include >> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror >> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g >> -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic >> -Wdeclaration-after-statement >> -I/usr/local/src/freeswitch/libs/spandsp/src >> -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff >> -I/usr/local/src/freeswitch/libs/spandsp/src >> -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT >> mod_spandsp_la-mod_spandsp.lo -MD -MP -MF >> .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o >> .libs/mod_spandsp_la-mod_spandsp.o >> In file included from >> /usr/local/src/freeswitch/libs/spandsp/src/spandsp/expose.h:83, >> from >> /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:141, >> from mod_spandsp.h:50, >> from mod_spandsp.c:39: >> /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:54: >> error: expected specifier-qualifier-list before ?lab_params_t? >> /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:72: >> error: expected specifier-qualifier-list before ?lab_params_t? >> make[5]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1 >> make[5]: Leaving directory >> `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' >> make[4]: *** [mod_spandsp-all] Error 1 >> make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' >> make[3]: *** [all-recursive] Error 1 >> make[3]: Leaving directory `/usr/local/src/freeswitch/src' >> make[2]: *** [all-recursive] Error 1 >> make[2]: Leaving directory `/usr/local/src/freeswitch' >> make[1]: *** [all] Error 2 >> make[1]: Leaving directory `/usr/local/src/freeswitch' >> make: *** [current] Error 2 >> >> Any ideas? >> >> -- >> Evgeniy Movlyan, >> BestNet Ltd. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> !DSPAM:500f9ab132761590495690! >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Evgeniy Movlyan, BestNet Ltd. From alex at thewinelake.com Wed Jul 25 15:13:35 2012 From: alex at thewinelake.com (Alex) Date: Wed, 25 Jul 2012 12:13:35 +0100 Subject: [Freeswitch-users] String Manipulation In-Reply-To: References: Message-ID: <500FD4DF.10209@thewinelake.com> Is there a neat way within a dialplan script to strip all non-digits from a string? The obvious (non-neat) way is to write a lua function for it... From gerald.weber at besharp.at Wed Jul 25 16:14:06 2012 From: gerald.weber at besharp.at (Gerald Weber) Date: Wed, 25 Jul 2012 12:14:06 +0000 Subject: [Freeswitch-users] HTTP Request from within the core Message-ID: Hi, are there any ready-to-use core commands (to use in a c module ) to make http requests ? regards, gerald weber -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120725/bfee22e6/attachment.html From avi at avimarcus.net Wed Jul 25 16:25:36 2012 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 25 Jul 2012 15:25:36 +0300 Subject: [Freeswitch-users] HTTP Request from within the core In-Reply-To: References: Message-ID: You can use the API for curl or just check how the codebase for curl works: http://wiki.freeswitch.org/wiki/Mod_curl -- src/mod/applications/mod_curl/ Or look at http://wiki.freeswitch.org/wiki/Mod_http_cache -- src/mod/applications/mod_http_cache/ -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120725/922e6d2e/attachment.html From edwin.hoexum at office.ziggo.nl Wed Jul 25 16:54:42 2012 From: edwin.hoexum at office.ziggo.nl (Hoexum, Edwin) Date: Wed, 25 Jul 2012 14:54:42 +0200 Subject: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a In-Reply-To: References: <2C18241C6768CF469C7E432A25327AD513529739DE@MAIL-WPV01.office.intern> <4FED4DA1.10304@belrosbank.by> <1341617252.56013.YahooMailNeo@web39301.mail.mud.yahoo.com> Message-ID: <2C18241C6768CF469C7E432A25327AD51352D07199@MAIL-WPV01.office.intern> Gents, Thank you for the discussion I will try to test on a box without virtualization. And play with de Ethernet cards (intel ) to divide the traffic maybe some bonding of interfaces (interrupts stuff) etc.. I will let you know the results. mvg, Edwin [cid:image001.gif at 01CD6A4C.8EC89060] Edwin Hoexum | Senior Systeeem Specialist | Voice Management _________________________________________________________________ Dr. van Deenweg 120, Zwolle | Postbus 9501 | 9703 LM Groningen | www.ziggo.nl Tel: +31887172699 | Mobiel: +31655148679| e-mailadres: Edwin.hoexum at office.ziggo.nl Werk Locaties: Maandag en Woensdag locatie Utrecht Dinsdag, Donderdag en Vrijdag locatie Zwolle Van: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Namens A E G Verzonden: zaterdag 7 juli 2012 3:58 Aan: FreeSWITCH Users Help Onderwerp: Re: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a On Fri, Jul 6, 2012 at 7:27 PM, Stanislav Sinyagin > wrote: OpenVZ and XEN VPS'es are usually fine for quick prototyping and feature tests. Also they would be fine for private PBXes with low call volumes. For hundreds of simultaneous calls, it's probably cheaper to rent dedicated hardware servers. On the other side, if you have some smart load balancing, probably you can build a scalable design with VPS'es and eventually migrate to physical servers when volumes are high enough. also for a calling card business, probably the most critical CPU resource would be needed for the IVR and real-time billing, but not for the calls themselves. Thanks Anthony and Stanislav. I think I have enough warning along with some encouraging words to build a strategy. Thanks again., Now the search for the right IaaS provider begins. ________________________________ From: A E G > To: FreeSWITCH Users Help > Sent: Saturday, July 7, 2012 12:23 AM Subject: Re: [Freeswitch-users] RTP delay > 100ms by performance testing with freeswitch as trancoding device SILK to G711a On Fri, Jul 6, 2012 at 5:43 PM, Anthony Minessale > wrote: I would be insane to endorse using it in a virtual world. The support expectations would consume the rest of my life. I try to make the code work well on real boxes and if it works on virtual ones then that's a bonus =D We have good luck with openVZ since its a single real kernel and virtual runtimes but either way we do not recommend it so its a use at your own risk kind of thing. Yup agree and acknowledge that you wouldn't endorse or offer to support it, and nor should you. Simply trying to pick your (and anyone else who wants to chime in) brain on the topic. Totally PoC'ing it at my own risk :) So, in that spirit, just one last message about it. Just like sharing the experience with OpenVZ, let's just say that I am insane and I still want to try it in a private cloud. Would you say the resource consumption of FS (assuming it's just receiving the call, having it routed by a custom routing engine via ESL, and switching it) be more than that in the "real" world for the same call volume barring any clock sync/skew/jitter issues? Would something like Cloudstack on hardware (like the one you recommended earlier) with a FS VM with 12 vcpu cores, 24GB RAM and DAS be able to handle something like 300-400 concurrent calls or am I out of my mind? Thx On Fri, Jul 6, 2012 at 3:59 PM, A E G > wrote: > Yup got that, and are you saying the same is applicable and true even if one > was to try and run FreeSWITCH on a private cloud or in a virtual > environment? assuming of course that those tips would then apply to the > machine on which the Hypervisor will run. > > Just trying to get a handle on whether running FreeSWITCH to do something > like wholesale or calling card traffic in a purely virtual environment is / > has proven to work. You probably know the most in terms of all different > environments people are running FS in, and if you (or they) have pointers > specific to it being run in a virtual environment. > > Thanks so much > > On Fri, Jul 6, 2012 at 12:54 PM, Anthony Minessale > > wrote: >> >> Like I said "Really Really Nice Motherboard", as many CPU as >> possible/affordable, and a good chunk of RAM. >> Motherboard is the most important, a cheap motherboard with a ton of >> cores is a waste. >> ________________________________ This message is confidential and may be privileged. Any review, retransmission, dissemination or other use of, taking any action with reference to this information by persons other then the intended recipient is prohibited. If you receive this message in error, please notify the sender by reply e-mail and delete this message from all computers. Please note that e-mails are susceptible to change. The sender will not accept liability for the improper or incomplete transmission of the information contained in this message. Spaar het milieu door deze e-mail niet te printen/Please consider the environment before printing this email. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120725/c3d20a15/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: image001.gif Type: image/gif Size: 8196 bytes Desc: image001.gif Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120725/c3d20a15/attachment-0001.gif From gerald.weber at besharp.at Wed Jul 25 17:03:05 2012 From: gerald.weber at besharp.at (Gerald Weber) Date: Wed, 25 Jul 2012 13:03:05 +0000 Subject: [Freeswitch-users] HTTP Request from within the core In-Reply-To: References: Message-ID: Thanks, looks very easy to implement. Just on more question: I need to get the IP address of a registered user. E.g. user/2000 is a SNOM with ip 192.168.20.219 I guess a query against the registrations table ist the fastest way to do this ? Or is there any core api command ? Thanks Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Avi Marcus Gesendet: Mittwoch, 25. Juli 2012 14:26 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] HTTP Request from within the core You can use the API for curl or just check how the codebase for curl works: http://wiki.freeswitch.org/wiki/Mod_curl -- src/mod/applications/mod_curl/ Or look at http://wiki.freeswitch.org/wiki/Mod_http_cache -- src/mod/applications/mod_http_cache/ -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120725/fe5df05f/attachment.html From bdfoster at endigotech.com Wed Jul 25 17:10:28 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 25 Jul 2012 09:10:28 -0400 Subject: [Freeswitch-users] HTTP Request from within the core In-Reply-To: References: Message-ID: sofia_contact might be what you are looking for. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 25, 2012 9:04 AM, "Gerald Weber" wrote: > Thanks, looks very easy to implement.**** > > ** ** > > Just on more question:**** > > I need to get the IP address of a registered user.**** > > E.g. user/2000 is a SNOM with ip 192.168.20.219**** > > I guess a query against the registrations table ist the fastest way to do > this ?**** > > Or is there any core api command ? **** > > ** ** > > Thanks**** > > ** ** > > ** ** > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Avi Marcus > *Gesendet:* Mittwoch, 25. Juli 2012 14:26 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] HTTP Request from within the core**** > > ** ** > > You can use the API for curl or just check how the codebase for curl works: > **** > > http://wiki.freeswitch.org/wiki/Mod_curl -- src/mod/applications/mod_curl/ > **** > > Or look at http://wiki.freeswitch.org/wiki/Mod_http_cache > -- src/mod/applications/mod_http_cache/**** > > ** ** > > -Avi**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120725/ba10bcfd/attachment.html From martyn at magiccow.co.uk Wed Jul 25 14:58:57 2012 From: martyn at magiccow.co.uk (Martyn Davies) Date: Wed, 25 Jul 2012 11:58:57 +0100 Subject: [Freeswitch-users] Testing iSAC Message-ID: I've now got: global_codec_prefs=isac at 16000h,isac,PCMU,PCMA,GSM,G722 and I can connect in 16 and 32 modes (inbound to the freeswitch). I haven't got outbound iSAC working from the FreeSwitch yet, but I'm guessing that the client doesn't like the CNG SDP option (13). I've added this to the dialplan: and restarted, but still see the 13 option going out. How to disable this? Regards, Martyn On 25 July 2012 08:45, wrote: > > 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: No audio after upgrade to F17 (Chris Mylonas) > 2. Re: Testing iSAC (curriegrad2004) > 3. mod_spandsp compilation error (Evgeniy Movlyan) > 4. Re: mod_spandsp compilation error (Peter Olsson) > 5. Re: mod_spandsp compilation error (Dmitry R) > > > ---------- Forwarded message ---------- > From: Chris Mylonas > To: FreeSWITCH Users Help > Cc: > Date: Wed, 25 Jul 2012 09:23:47 +1000 > Subject: Re: [Freeswitch-users] No audio after upgrade to F17 > Further to Paul's comments > > In a terminal > tail -f /var/log/messages > > In another terminal > tail -f /path/to/your/freeswitch/log/freeswitch.log > > In another terminal, start freeswitch > > > If it is an SELinux problem, which is probably suspected, there will be a kernel message stating something along the lines of a context is not allowed to do something. > I had similar issues with openvpn and SELinux - this blog post may be useful in the process of keeping SELinux running and customising it's profile for running freeswitch. > http://www.mrvoip.com.au/blog/selinux-openvpn > > Alternatively, look at ways of disabling SELinux - on other redhat related distros, it's a matter of editing a setting in a file to "disabled" and rebooting so the kernel doesn't load SELinux stuff. Another alternative is to change SELinux's mode from "enforcing" to "permissive" > > > Cheers > Chris > > > > > > On 25/07/2012, at 3:56 AM, Paul Cupis wrote: > > On 24/07/12 17:21, Todd Bailey wrote: > > On Mon, 2012-07-23 at 12:20 -0700, Todd wrote: > > When I installed F17, I copied the FS installation directly > > > Booting back to F14, every this works as expected, so it's not the > > spa3102 or config files. > > > Thinking there might be a incompatibility issues, I recompiled FS under > > F17, no change in operational behavior. > > > Ideas where else to look ? > > > Have a look at the "new features" in F15, F16 or F17, you'll probably > find some security function (SElinux or similar) which may be blocking > FreeSWITCH RTP traffic. > > Regards, > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > ---------- Forwarded message ---------- > From: curriegrad2004 > To: FreeSWITCH Users Help > Cc: > Date: Tue, 24 Jul 2012 20:45:34 -0700 > Subject: Re: [Freeswitch-users] Testing iSAC > > You might want to report this issue in the JIRA as a feature request. > > Iirc it doesn't support that yet. > > On Jul 24, 2012 6:26 AM, "Martyn Davies" wrote: >> >> I have an iSAC client that I'm testing against FreeSwitch. >> >> Currently I have my vars.xml with: >> >> global_codec_prefs=isac,PCMU,PCMA,GSM,G722 >> >> Which works, but only if the client specifies 32kHz mode. I want to test 16kHz mode, but FreeSwitch returns 'not acceptable here', and I can see that when Freeswitch offers iSAC in the SDP it says only isac/32000. >> >> So how to enable 16kHz mode for isac? >> >> I tried explicitly putting isac at 320000h30i, isac at 16000h30i into vars.xml, but this simply stops FreeSwitch from accepting anything. >> >> Hints gladly received. >> >> Regards, >> Martyn >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > ---------- Forwarded message ---------- > From: Evgeniy Movlyan > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Wed, 25 Jul 2012 10:12:18 +0300 > Subject: [Freeswitch-users] mod_spandsp compilation error > Hi to all, > i made make current command today, and got this: > > making all mod_spandsp > make[5]: Entering directory `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' > Creating mod_spandsp_la-mod_spandsp.lo > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. -I../../../../src/include -I../../../../libs/xmlrpc-c -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -I/usr/local/src/freeswitch/libs/curl/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/src/include -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I/usr/local/src/freeswitch/libs/spandsp/src -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT mod_spandsp_la-mod_spandsp.lo -MD -MP -MF .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o .libs/mod_spandsp_la-mod_spandsp.o > In file included from /usr/local/src/freeswitch/libs/spandsp/src/spandsp/expose.h:83, > from /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:141, > from mod_spandsp.h:50, > from mod_spandsp.c:39: > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:54: error: expected specifier-qualifier-list before ?lab_params_t? > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:72: error: expected specifier-qualifier-list before ?lab_params_t? > make[5]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1 > make[5]: Leaving directory `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' > make[4]: *** [mod_spandsp-all] Error 1 > make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/usr/local/src/freeswitch/src' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/usr/local/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/usr/local/src/freeswitch' > make: *** [current] Error 2 > > Any ideas? > > -- > Evgeniy Movlyan, > BestNet Ltd. > > > > > ---------- Forwarded message ---------- > From: Peter Olsson > To: FreeSWITCH Users Help > Cc: > Date: Wed, 25 Jul 2012 07:42:30 +0000 > Subject: Re: [Freeswitch-users] mod_spandsp compilation error > 'make spandsp-reconf'. If it doesn't help, file a Jira. > > /Peter > > 25 jul 2012 kl. 09:19 skrev "Evgeniy Movlyan" : > > > Hi to all, > > i made make current command today, and got this: > > > > making all mod_spandsp > > make[5]: Entering directory > > `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' > > Creating mod_spandsp_la-mod_spandsp.lo > > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. > > -I../../../../src/include -I../../../../libs/xmlrpc-c > > -I/usr/local/src/freeswitch/libs/curl/include > > -I/usr/local/src/freeswitch/src/include > > -I/usr/local/src/freeswitch/src/include > > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC > > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 > > -I/usr/local/src/freeswitch/libs/curl/include > > -I/usr/local/src/freeswitch/src/include > > -I/usr/local/src/freeswitch/src/include > > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g > > -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic > > -Wdeclaration-after-statement > > -I/usr/local/src/freeswitch/libs/spandsp/src > > -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff > > -I/usr/local/src/freeswitch/libs/spandsp/src > > -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT > > mod_spandsp_la-mod_spandsp.lo -MD -MP -MF > > .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o > > .libs/mod_spandsp_la-mod_spandsp.o > > In file included from > > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/expose.h:83, > > from > > /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:141, > > from mod_spandsp.h:50, > > from mod_spandsp.c:39: > > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:54: > > error: expected specifier-qualifier-list before ?lab_params_t? > > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:72: > > error: expected specifier-qualifier-list before ?lab_params_t? > > make[5]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1 > > make[5]: Leaving directory > > `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' > > make[4]: *** [mod_spandsp-all] Error 1 > > make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' > > make[3]: *** [all-recursive] Error 1 > > make[3]: Leaving directory `/usr/local/src/freeswitch/src' > > make[2]: *** [all-recursive] Error 1 > > make[2]: Leaving directory `/usr/local/src/freeswitch' > > make[1]: *** [all] Error 2 > > make[1]: Leaving directory `/usr/local/src/freeswitch' > > make: *** [current] Error 2 > > > > Any ideas? > > > > -- > > Evgeniy Movlyan, > > BestNet Ltd. > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > !DSPAM:500f9ab132761590495690! > > > > > > > ---------- Forwarded message ---------- > From: "Dmitry R" > To: "'FreeSWITCH Users Help'" > Cc: > Date: Wed, 25 Jul 2012 11:34:28 +0400 > Subject: Re: [Freeswitch-users] mod_spandsp compilation error > I got the same problem. Solved after git pull && ./bootstrap.sh && > ./configure && make > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Evgeniy > Movlyan > Sent: Wednesday, July 25, 2012 11:12 AM > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] mod_spandsp compilation error > > Hi to all, > i made make current command today, and got this: > > making all mod_spandsp > make[5]: Entering directory > `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' > Creating mod_spandsp_la-mod_spandsp.lo > quiet_libtool: compile: gcc -DHAVE_CONFIG_H -I. > -I../../../../src/include -I../../../../libs/xmlrpc-c > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -fvisibility=hidden > -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 > -I/usr/local/src/freeswitch/libs/curl/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb > -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement > -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff > -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -I. -g -O2 -MT > mod_spandsp_la-mod_spandsp.lo -MD -MP -MF > .deps/mod_spandsp_la-mod_spandsp.Tpo -c mod_spandsp.c -fPIC -DPIC -o > .libs/mod_spandsp_la-mod_spandsp.o > In file included from > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/expose.h:83, > from > /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:141, > from mod_spandsp.h:50, > from mod_spandsp.c:39: > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:54: > error: expected specifier-qualifier-list before 'lab_params_t' > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/private/t42.h:72: > error: expected specifier-qualifier-list before 'lab_params_t' > make[5]: *** [mod_spandsp_la-mod_spandsp.lo] Error 1 > make[5]: Leaving directory > `/usr/local/src/freeswitch/src/mod/applications/mod_spandsp' > make[4]: *** [mod_spandsp-all] Error 1 > make[4]: Leaving directory `/usr/local/src/freeswitch/src/mod' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/usr/local/src/freeswitch/src' > make[2]: *** [all-recursive] Error 1 > make[2]: Leaving directory `/usr/local/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/usr/local/src/freeswitch' > make: *** [current] Error 2 > > Any ideas? > > -- > Evgeniy Movlyan, > BestNet Ltd. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From gerald.weber at besharp.at Wed Jul 25 17:47:53 2012 From: gerald.weber at besharp.at (Gerald Weber) Date: Wed, 25 Jul 2012 13:47:53 +0000 Subject: [Freeswitch-users] HTTP Request from within the core In-Reply-To: References: Message-ID: Thats it, muchas gracias ! Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Brian Foster Gesendet: Mittwoch, 25. Juli 2012 15:10 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] HTTP Request from within the core sofia_contact might be what you are looking for. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 25, 2012 9:04 AM, "Gerald Weber" > wrote: Thanks, looks very easy to implement. Just on more question: I need to get the IP address of a registered user. E.g. user/2000 is a SNOM with ip 192.168.20.219 I guess a query against the registrations table ist the fastest way to do this ? Or is there any core api command ? Thanks Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Avi Marcus Gesendet: Mittwoch, 25. Juli 2012 14:26 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] HTTP Request from within the core You can use the API for curl or just check how the codebase for curl works: http://wiki.freeswitch.org/wiki/Mod_curl -- src/mod/applications/mod_curl/ Or look at http://wiki.freeswitch.org/wiki/Mod_http_cache -- src/mod/applications/mod_http_cache/ -Avi _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120725/801f2aac/attachment.html From msc at freeswitch.org Wed Jul 25 20:07:41 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Jul 2012 09:07:41 -0700 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today Message-ID: Hey there, We are having our conference call in less than an hour. The agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2012_07_25 See you soon. -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120725/63e999fd/attachment.html From msc at freeswitch.org Wed Jul 25 20:39:07 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Jul 2012 09:39:07 -0700 Subject: [Freeswitch-users] String Manipulation In-Reply-To: <500FD4DF.10209@thewinelake.com> References: <500FD4DF.10209@thewinelake.com> Message-ID: >From what I can tell, the PCRE interface does pattern matching but not all the fancy modifiers, like /g to continually repeat an operation. (Please correct me if I'm wrong and I will wikify the specifics.) The good news is that a simple Lua script to do something like this isn't very resource intensive. -MC On Wed, Jul 25, 2012 at 4:13 AM, Alex wrote: > Is there a neat way within a dialplan script to strip all non-digits > from a string? > The obvious (non-neat) way is to write a lua function for it... > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120725/9d7ed01f/attachment.html From asaad2 at gmail.com Wed Jul 25 20:39:44 2012 From: asaad2 at gmail.com (BookBag) Date: Wed, 25 Jul 2012 11:39:44 -0500 Subject: [Freeswitch-users] Voice broadcasting Message-ID: Hello all, can anyone direct to some information on how I can setup a voice broadcast on FS. I tried looking in the wiki but it justs redirects you to heading "Robocalls", which in turn redirect you "voice broadcast" but there is no information under it. Basically I just want to call about 100 people or so and play a recording and then hangup. If anyone could help me , i'd really appreciate it. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120725/d5d1347f/attachment.html From msc at freeswitch.org Wed Jul 25 20:48:24 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Jul 2012 09:48:24 -0700 Subject: [Freeswitch-users] Voice broadcasting In-Reply-To: References: Message-ID: >From the fs_cli: originate {ignore_early_media=true}sofia/gateway/your_gw_name/18005551212 &playback("/path/to/file.wav") Please don't abuse your new found power! :) -MC On Wed, Jul 25, 2012 at 9:39 AM, BookBag wrote: > Hello all, can anyone direct to some information on how I can setup a > voice broadcast on FS. I tried looking in the wiki but it justs redirects > you to heading "Robocalls", which in turn redirect you "voice broadcast" > but there is no information under it. > > Basically I just want to call about 100 people or so and play a recording > and then hangup. > > > If anyone could help me , i'd really appreciate it. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120725/4dcc763a/attachment.html From asaad2 at gmail.com Wed Jul 25 20:56:50 2012 From: asaad2 at gmail.com (BookBag) Date: Wed, 25 Jul 2012 11:56:50 -0500 Subject: [Freeswitch-users] Voice broadcasting In-Reply-To: References: Message-ID: thanx lol :-) But is there a way to have it call from a per-determined list On Wed, Jul 25, 2012 at 11:48 AM, Michael Collins wrote: > >From the fs_cli: > originate {ignore_early_media=true}sofia/gateway/your_gw_name/18005551212&playback("/path/to/file.wav") > > Please don't abuse your new found power! :) > -MC > > > On Wed, Jul 25, 2012 at 9:39 AM, BookBag wrote: > >> Hello all, can anyone direct to some information on how I can setup a >> voice broadcast on FS. I tried looking in the wiki but it justs redirects >> you to heading "Robocalls", which in turn redirect you "voice broadcast" >> but there is no information under it. >> >> Basically I just want to call about 100 people or so and play a recording >> and then hangup. >> >> >> If anyone could help me , i'd really appreciate it. >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120725/10320b91/attachment-0001.html From krice at freeswitch.org Wed Jul 25 21:12:21 2012 From: krice at freeswitch.org (Ken Rice) Date: Wed, 25 Jul 2012 12:12:21 -0500 Subject: [Freeswitch-users] Voice broadcasting In-Reply-To: Message-ID: You need a queue manager and all sorts of other instrumentation that is outside the scope of FreeSWITCH itself. The short answer, yes you can do it, the not so short answer, theres a lot of work to do to turn FreeSWITCH into a dialer. You might dig around in google to find some things that do robocalls on freeswitch already... On 7/25/12 11:56 AM, "BookBag" wrote: > thanx lol :-) > > But is there a way to have it call from a per-determined list > > On Wed, Jul 25, 2012 at 11:48 AM, Michael Collins wrote: >>> >From the fs_cli: >> originate {ignore_early_media=true}sofia/gateway/your_gw_name/18005551212 >> &playback("/path/to/file.wav") >> >> Please don't abuse your new found power! :) >> -MC >> >> >> On Wed, Jul 25, 2012 at 9:39 AM, BookBag wrote: >>> Hello all, can anyone direct to some information on how I can setup a voice >>> broadcast on FS. I tried looking in the wiki but it justs redirects you to >>> heading "Robocalls", which in turn redirect you "voice broadcast" but there >>> is no information under it. >>> >>> Basically I just want to call about 100 people or so and play a recording >>> and then hangup. >>> >>> >>> If anyone could help me , i'd really appreciate it. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120725/9f54d086/attachment.html From anton.jugatsu at gmail.com Wed Jul 25 21:46:42 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Wed, 25 Jul 2012 21:46:42 +0400 Subject: [Freeswitch-users] Google Inc. DOES use FreeSWITCH for call center telephony routing system. So, why dont' we invite those guys to FS weekly conference :) Message-ID: Guys, according to those slides ( http://www.slideshare.net/OReillyOSCON/using-and-building-open-source-in-google-corporate-engineering-justin-mcwilliams slide number 10) from Justin McWilliams ( https://plus.google.com/u/0/107742494065843953483/about ), Google does use FS for internal call center. So, I think that it would be great to invite some folks from Google to participate @ weekly FS conference. We can contact Justin via G+. For me it was really surprise, because I've never seen any guys from google committing something or discussing something @ ML. The agenda for that conference would be quite simple: to describe some use cases and the decision why did they choose to use FreeSWITCH. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120725/da4daa4d/attachment.html From gabe at gundy.org Wed Jul 25 22:10:52 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 25 Jul 2012 12:10:52 -0600 Subject: [Freeswitch-users] Google Inc. DOES use FreeSWITCH for call center telephony routing system. So, why dont' we invite those guys to FS weekly conference :) In-Reply-To: References: Message-ID: On Wed, Jul 25, 2012 at 11:46 AM, Anton Kvashenkin wrote: > Guys, according to those slides (*LINK* slide number 10). FYI, it looks like it slide #9. I also think this would make a killer weekly FS conference call. Even better, I'd love to hear about it in ClueCon :) Best, Gabe From bdfoster at endigotech.com Wed Jul 25 22:11:24 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 25 Jul 2012 14:11:24 -0400 Subject: [Freeswitch-users] Voice broadcasting In-Reply-To: References: Message-ID: Is newfies dialer a good fit for this case? Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 25, 2012 1:14 PM, "Ken Rice" wrote: > You need a queue manager and all sorts of other instrumentation that is > outside the scope of FreeSWITCH itself. > > The short answer, yes you can do it, the not so short answer, theres a lot > of work to do to turn FreeSWITCH into a dialer. > > You might dig around in google to find some things that do robocalls on > freeswitch already... > > > On 7/25/12 11:56 AM, "BookBag" wrote: > > thanx lol :-) > > But is there a way to have it call from a per-determined list > > On Wed, Jul 25, 2012 at 11:48 AM, Michael Collins > wrote: > > >From the fs_cli: > originate {ignore_early_media=true}sofia/gateway/your_gw_name/18005551212 18005551212> &playback("/path/to/file.wav") > > Please don't abuse your new found power! :) > -MC > > > On Wed, Jul 25, 2012 at 9:39 AM, BookBag wrote: > > Hello all, can anyone direct to some information on how I can setup a > voice broadcast on FS. I tried looking in the wiki but it justs redirects > you to heading "Robocalls", which in turn redirect you "voice broadcast" > but there is no information under it. > > Basically I just want to call about 100 people or so and play a recording > and then hangup. > > > If anyone could help me , i'd really appreciate it. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120725/dc165d14/attachment-0001.html From msc at freeswitch.org Wed Jul 25 22:16:20 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Jul 2012 11:16:20 -0700 Subject: [Freeswitch-users] Voice broadcasting In-Reply-To: References: Message-ID: Yep. On Wed, Jul 25, 2012 at 11:11 AM, Brian Foster wrote: > Is newfies dialer a good fit for this case? > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > On Jul 25, 2012 1:14 PM, "Ken Rice" wrote: > >> You need a queue manager and all sorts of other instrumentation that is >> outside the scope of FreeSWITCH itself. >> >> The short answer, yes you can do it, the not so short answer, theres a >> lot of work to do to turn FreeSWITCH into a dialer. >> >> You might dig around in google to find some things that do robocalls on >> freeswitch already... >> >> >> On 7/25/12 11:56 AM, "BookBag" wrote: >> >> thanx lol :-) >> >> But is there a way to have it call from a per-determined list >> >> On Wed, Jul 25, 2012 at 11:48 AM, Michael Collins >> wrote: >> >> >From the fs_cli: >> originate {ignore_early_media=true}sofia/gateway/your_gw_name/18005551212> 18005551212> &playback("/path/to/file.wav") >> >> Please don't abuse your new found power! :) >> -MC >> >> >> On Wed, Jul 25, 2012 at 9:39 AM, BookBag wrote: >> >> Hello all, can anyone direct to some information on how I can setup a >> voice broadcast on FS. I tried looking in the wiki but it justs redirects >> you to heading "Robocalls", which in turn redirect you "voice broadcast" >> but there is no information under it. >> >> Basically I just want to call about 100 people or so and play a recording >> and then hangup. >> >> >> If anyone could help me , i'd really appreciate it. >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> ------------------------------ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120725/d88eab7e/attachment.html From msc at freeswitch.org Wed Jul 25 22:17:02 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Jul 2012 11:17:02 -0700 Subject: [Freeswitch-users] Google Inc. DOES use FreeSWITCH for call center telephony routing system. So, why dont' we invite those guys to FS weekly conference :) In-Reply-To: References: Message-ID: I think maybe in 2013 we can get Google to come talk to us. It would be even better if they sponsored us. ;) -MC On Wed, Jul 25, 2012 at 11:10 AM, Gabriel Gunderson wrote: > On Wed, Jul 25, 2012 at 11:46 AM, Anton Kvashenkin > wrote: > > Guys, according to those slides (*LINK* slide number 10). > > FYI, it looks like it slide #9. > > I also think this would make a killer weekly FS conference call. Even > better, I'd love to hear about it in ClueCon :) > > > Best, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120725/7f42bd57/attachment.html From anton.jugatsu at gmail.com Wed Jul 25 22:23:33 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Wed, 25 Jul 2012 22:23:33 +0400 Subject: [Freeswitch-users] Google Inc. DOES use FreeSWITCH for call center telephony routing system. So, why dont' we invite those guys to FS weekly conference :) In-Reply-To: References: Message-ID: Michael, it sucks, I don't want to wait one freaking year :) Maybe upcoming weekly FS conference call :) 2012/7/25 Michael Collins > I think maybe in 2013 we can get Google to come talk to us. It would be > even better if they sponsored us. ;) > -MC > > > On Wed, Jul 25, 2012 at 11:10 AM, Gabriel Gunderson wrote: > >> On Wed, Jul 25, 2012 at 11:46 AM, Anton Kvashenkin >> wrote: >> > Guys, according to those slides (*LINK* slide number 10). >> >> FYI, it looks like it slide #9. >> >> I also think this would make a killer weekly FS conference call. Even >> better, I'd love to hear about it in ClueCon :) >> >> >> Best, >> Gabe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120725/293ebe8b/attachment-0001.html From anthony.minessale at gmail.com Wed Jul 25 22:31:59 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Jul 2012 13:31:59 -0500 Subject: [Freeswitch-users] Google Inc. DOES use FreeSWITCH for call center telephony routing system. So, why dont' we invite those guys to FS weekly conference :) In-Reply-To: References: Message-ID: There is still time, CC is in 2 weeks and we have the call every wed. =p On Wed, Jul 25, 2012 at 1:23 PM, Anton Kvashenkin wrote: > Michael, it sucks, I don't want to wait one freaking year :) Maybe upcoming > weekly FS conference call :) > > > 2012/7/25 Michael Collins >> >> I think maybe in 2013 we can get Google to come talk to us. It would be >> even better if they sponsored us. ;) >> -MC >> >> >> On Wed, Jul 25, 2012 at 11:10 AM, Gabriel Gunderson >> wrote: >>> >>> On Wed, Jul 25, 2012 at 11:46 AM, Anton Kvashenkin >>> wrote: >>> > Guys, according to those slides (*LINK* slide number 10). >>> >>> FYI, it looks like it slide #9. >>> >>> I also think this would make a killer weekly FS conference call. Even >>> better, I'd love to hear about it in ClueCon :) >>> >>> >>> Best, >>> Gabe >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From asaad2 at gmail.com Wed Jul 25 22:50:30 2012 From: asaad2 at gmail.com (BookBag) Date: Wed, 25 Jul 2012 13:50:30 -0500 Subject: [Freeswitch-users] Voice broadcasting In-Reply-To: References: Message-ID: thanks everyone, I'll give it a try On Wed, Jul 25, 2012 at 1:11 PM, Brian Foster wrote: > Is newfies dialer a good fit for this case? > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > On Jul 25, 2012 1:14 PM, "Ken Rice" wrote: > >> You need a queue manager and all sorts of other instrumentation that is >> outside the scope of FreeSWITCH itself. >> >> The short answer, yes you can do it, the not so short answer, theres a >> lot of work to do to turn FreeSWITCH into a dialer. >> >> You might dig around in google to find some things that do robocalls on >> freeswitch already... >> >> >> On 7/25/12 11:56 AM, "BookBag" wrote: >> >> thanx lol :-) >> >> But is there a way to have it call from a per-determined list >> >> On Wed, Jul 25, 2012 at 11:48 AM, Michael Collins >> wrote: >> >> >From the fs_cli: >> originate {ignore_early_media=true}sofia/gateway/your_gw_name/18005551212> 18005551212> &playback("/path/to/file.wav") >> >> Please don't abuse your new found power! :) >> -MC >> >> >> On Wed, Jul 25, 2012 at 9:39 AM, BookBag wrote: >> >> Hello all, can anyone direct to some information on how I can setup a >> voice broadcast on FS. I tried looking in the wiki but it justs redirects >> you to heading "Robocalls", which in turn redirect you "voice broadcast" >> but there is no information under it. >> >> Basically I just want to call about 100 people or so and play a recording >> and then hangup. >> >> >> If anyone could help me , i'd really appreciate it. >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> ------------------------------ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120725/9a9bd21d/attachment.html From dvl36.ripe.nick at gmail.com Wed Jul 25 23:54:28 2012 From: dvl36.ripe.nick at gmail.com (dvl36) Date: Wed, 25 Jul 2012 12:54:28 -0700 (PDT) Subject: [Freeswitch-users] libs/spandsp woes in compilation In-Reply-To: References: <1342889501930-7581017.post@n2.nabble.com> Message-ID: <1343246068084-7581135.post@n2.nabble.com> I confirm. Tried to compile at device itself(ARMv5/Marvell Kirkwood/Seagate Goflex Home/Debian Squeezy) w/o success. Reverted to git HEAD 9fe08675a1d3f0a8ba9c777befed7d6cc8a921f9, as mazilo suggested and compilation is OK. I will try to reproduce and post to Jira. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/libs-spandsp-woes-in-compilation-tp7581017p7581135.html Sent from the freeswitch-users mailing list archive at Nabble.com. From bdfoster at endigotech.com Thu Jul 26 01:48:27 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 25 Jul 2012 17:48:27 -0400 Subject: [Freeswitch-users] Google Inc. DOES use FreeSWITCH for call center telephony routing system. So, why dont' we invite those guys to FS weekly conference :) In-Reply-To: References: Message-ID: Anthony, I bet you feel proud, eh? Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 25, 2012 2:34 PM, "Anthony Minessale" wrote: > There is still time, CC is in 2 weeks and we have the call every wed. > =p > > > On Wed, Jul 25, 2012 at 1:23 PM, Anton Kvashenkin > wrote: > > Michael, it sucks, I don't want to wait one freaking year :) Maybe > upcoming > > weekly FS conference call :) > > > > > > 2012/7/25 Michael Collins > >> > >> I think maybe in 2013 we can get Google to come talk to us. It would be > >> even better if they sponsored us. ;) > >> -MC > >> > >> > >> On Wed, Jul 25, 2012 at 11:10 AM, Gabriel Gunderson > >> wrote: > >>> > >>> On Wed, Jul 25, 2012 at 11:46 AM, Anton Kvashenkin > >>> wrote: > >>> > Guys, according to those slides (*LINK* slide number 10). > >>> > >>> FYI, it looks like it slide #9. > >>> > >>> I also think this would make a killer weekly FS conference call. Even > >>> better, I'd love to hear about it in ClueCon :) > >>> > >>> > >>> Best, > >>> Gabe > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> consulting at freeswitch.org > >>> http://www.freeswitchsolutions.com > >>> > >>> > >>> > >>> > >>> Official FreeSWITCH Sites > >>> http://www.freeswitch.org > >>> http://wiki.freeswitch.org > >>> http://www.cluecon.com > >>> > >>> Join Us At ClueCon - Aug 7-9, 2012 > >>> > >>> FreeSWITCH-users mailing list > >>> FreeSWITCH-users at lists.freeswitch.org > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >> > >> > >> > >> > >> -- > >> Michael S Collins > >> Twitter: @mercutioviz > >> http://www.FreeSWITCH.org > >> http://www.ClueCon.com > >> http://www.OSTAG.org > >> > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120725/342c186f/attachment-0001.html From onyeagbaikenna04 at gmail.com Thu Jul 26 01:42:22 2012 From: onyeagbaikenna04 at gmail.com (Ikenna Onyeagba) Date: Wed, 25 Jul 2012 22:42:22 +0100 Subject: [Freeswitch-users] freeSWITCH on windows Message-ID: I am having problems installing freeSWITCH on windows. i get to the point where i have to run freeswitch.exe from the debug directory but i cannot find the .exe file, please help urgently From msc at freeswitch.org Thu Jul 26 02:04:44 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Jul 2012 15:04:44 -0700 Subject: [Freeswitch-users] Google Inc. DOES use FreeSWITCH for call center telephony routing system. So, why dont' we invite those guys to FS weekly conference :) In-Reply-To: References: Message-ID: Of course, I'd say the crowning achievement would be FreeSWITCH being used at Digium HQ. :P -MC On Wed, Jul 25, 2012 at 2:48 PM, Brian Foster wrote: > Anthony, > > I bet you feel proud, eh? > > Brian Foster > Endigo Computer LLC > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120725/b65a3c78/attachment.html From msc at freeswitch.org Thu Jul 26 02:05:17 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Jul 2012 15:05:17 -0700 Subject: [Freeswitch-users] freeSWITCH on windows In-Reply-To: References: Message-ID: Are you downloading the binaries or are you building from scratch? -MC On Wed, Jul 25, 2012 at 2:42 PM, Ikenna Onyeagba wrote: > I am having problems installing freeSWITCH on windows. i get to the > point where i have to run freeswitch.exe from the debug directory but > i cannot find the .exe file, please help urgently > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120725/c3ea4d47/attachment.html From onyeagbaikenna04 at gmail.com Thu Jul 26 02:16:15 2012 From: onyeagbaikenna04 at gmail.com (Ikenna Onyeagba) Date: Wed, 25 Jul 2012 23:16:15 +0100 Subject: [Freeswitch-users] freeSWITCH on windows In-Reply-To: References: Message-ID: am building from the scratch. On Wed, Jul 25, 2012 at 11:05 PM, Michael Collins wrote: > Are you downloading the binaries or are you building from scratch? > -MC > > > On Wed, Jul 25, 2012 at 2:42 PM, Ikenna Onyeagba > wrote: >> >> I am having problems installing freeSWITCH on windows. i get to the >> point where i have to run freeswitch.exe from the debug directory but >> i cannot find the .exe file, please help urgently > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Jul 26 02:22:20 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Jul 2012 15:22:20 -0700 Subject: [Freeswitch-users] freeSWITCH on windows In-Reply-To: References: Message-ID: You may want to try the binaries, just to make sure your system is okay to run FreeSWITCH. They're quick and easy and won't affect your build from source. -MC http://wiki.freeswitch.org/wiki/Installation_for_Windows#Precompiled_Binaries On Wed, Jul 25, 2012 at 3:16 PM, Ikenna Onyeagba wrote: > am building from the scratch. > > > On Wed, Jul 25, 2012 at 11:05 PM, Michael Collins > wrote: > > Are you downloading the binaries or are you building from scratch? > > -MC > > > > > > On Wed, Jul 25, 2012 at 2:42 PM, Ikenna Onyeagba > > wrote: > >> > >> I am having problems installing freeSWITCH on windows. i get to the > >> point where i have to run freeswitch.exe from the debug directory but > >> i cannot find the .exe file, please help urgently > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120725/49e939aa/attachment.html From peter.olsson at visionutveckling.se Thu Jul 26 02:23:04 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 25 Jul 2012 22:23:04 +0000 Subject: [Freeswitch-users] freeSWITCH on windows In-Reply-To: References: , Message-ID: <92C8E590-263F-450E-92BE-9530F50AE7A8@visionutveckling.se> If it's built on VS2010 the name is FreeSwitchConsole.exe /Peter 26 jul 2012 kl. 00:21 skrev "Ikenna Onyeagba" : > am building from the scratch. > > > On Wed, Jul 25, 2012 at 11:05 PM, Michael Collins wrote: >> Are you downloading the binaries or are you building from scratch? >> -MC >> >> >> On Wed, Jul 25, 2012 at 2:42 PM, Ikenna Onyeagba >> wrote: >>> >>> I am having problems installing freeSWITCH on windows. i get to the >>> point where i have to run freeswitch.exe from the debug directory but >>> i cannot find the .exe file, please help urgently >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > !DSPAM:50106e0a32761380217421! > From onyeagbaikenna04 at gmail.com Thu Jul 26 02:27:15 2012 From: onyeagbaikenna04 at gmail.com (Ikenna Onyeagba) Date: Wed, 25 Jul 2012 23:27:15 +0100 Subject: [Freeswitch-users] freeSWITCH on windows In-Reply-To: <92C8E590-263F-450E-92BE-9530F50AE7A8@visionutveckling.se> References: <92C8E590-263F-450E-92BE-9530F50AE7A8@visionutveckling.se> Message-ID: thanks peter but where can i locate the FreeSwitchConsole.exe cos its not in the debug registry On Wed, Jul 25, 2012 at 11:23 PM, Peter Olsson wrote: > If it's built on VS2010 the name is FreeSwitchConsole.exe > > /Peter > > 26 jul 2012 kl. 00:21 skrev "Ikenna Onyeagba" : > >> am building from the scratch. >> >> >> On Wed, Jul 25, 2012 at 11:05 PM, Michael Collins wrote: >>> Are you downloading the binaries or are you building from scratch? >>> -MC >>> >>> >>> On Wed, Jul 25, 2012 at 2:42 PM, Ikenna Onyeagba >>> wrote: >>>> >>>> I am having problems installing freeSWITCH on windows. i get to the >>>> point where i have to run freeswitch.exe from the debug directory but >>>> i cannot find the .exe file, please help urgently >>> >>> >>> >>> -- >>> Michael S Collins >>> Twitter: @mercutioviz >>> http://www.FreeSWITCH.org >>> http://www.ClueCon.com >>> http://www.OSTAG.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> !DSPAM:50106e0a32761380217421! >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From onyeagbaikenna04 at gmail.com Thu Jul 26 02:41:24 2012 From: onyeagbaikenna04 at gmail.com (Ikenna Onyeagba) Date: Wed, 25 Jul 2012 23:41:24 +0100 Subject: [Freeswitch-users] freeSWITCH on windows In-Reply-To: References: Message-ID: am new to freeSWITCH, please forgive me if i ask alot of questions. what are the steps to start building from binary. On Wed, Jul 25, 2012 at 11:22 PM, Michael Collins wrote: > You may want to try the binaries, just to make sure your system is okay to > run FreeSWITCH. They're quick and easy and won't affect your build from > source. > -MC > http://wiki.freeswitch.org/wiki/Installation_for_Windows#Precompiled_Binaries > > On Wed, Jul 25, 2012 at 3:16 PM, Ikenna Onyeagba > wrote: >> >> am building from the scratch. >> >> >> On Wed, Jul 25, 2012 at 11:05 PM, Michael Collins >> wrote: >> > Are you downloading the binaries or are you building from scratch? >> > -MC >> > >> > >> > On Wed, Jul 25, 2012 at 2:42 PM, Ikenna Onyeagba >> > wrote: >> >> >> >> I am having problems installing freeSWITCH on windows. i get to the >> >> point where i have to run freeswitch.exe from the debug directory but >> >> i cannot find the .exe file, please help urgently >> > >> > >> > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Jul 26 03:30:58 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 25 Jul 2012 16:30:58 -0700 Subject: [Freeswitch-users] freeSWITCH on windows In-Reply-To: References: Message-ID: The link I gave you has all the instructions, however here's a really short version: go to http://files-sync.freeswitch.org/windows/installer/ choose x86 or x64 (whichever platform you have) download and run freeswitch.msi look in "Program Files" for the "FreeSWITCH" folder launch FreeSwitchConsole.exe application -MC On Wed, Jul 25, 2012 at 3:41 PM, Ikenna Onyeagba wrote: > am new to freeSWITCH, please forgive me if i ask alot of questions. > what are the steps to start building from binary. > > On Wed, Jul 25, 2012 at 11:22 PM, Michael Collins > wrote: > > You may want to try the binaries, just to make sure your system is okay > to > > run FreeSWITCH. They're quick and easy and won't affect your build from > > source. > > -MC > > > http://wiki.freeswitch.org/wiki/Installation_for_Windows#Precompiled_Binaries > > > > On Wed, Jul 25, 2012 at 3:16 PM, Ikenna Onyeagba > > wrote: > >> > >> am building from the scratch. > >> > >> > >> On Wed, Jul 25, 2012 at 11:05 PM, Michael Collins > >> wrote: > >> > Are you downloading the binaries or are you building from scratch? > >> > -MC > >> > > >> > > >> > On Wed, Jul 25, 2012 at 2:42 PM, Ikenna Onyeagba > >> > wrote: > >> >> > >> >> I am having problems installing freeSWITCH on windows. i get to the > >> >> point where i have to run freeswitch.exe from the debug directory but > >> >> i cannot find the .exe file, please help urgently > >> > > >> > > >> > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120725/c475b079/attachment.html From onyeagbaikenna04 at gmail.com Thu Jul 26 03:37:07 2012 From: onyeagbaikenna04 at gmail.com (onyeagbaikenna04 at gmail.com) Date: Wed, 25 Jul 2012 23:37:07 +0000 Subject: [Freeswitch-users] freeSWITCH on windows In-Reply-To: References: Message-ID: <603309988-1343259381-cardhu_decombobulator_blackberry.rim.net-112402843-@b13.c16.bise7.blackberry> Thanks Sent from my BlackBerry smartphone from Virgin Media -----Original Message----- From: Michael Collins Sender: freeswitch-users-bounces at lists.freeswitch.org Date: Wed, 25 Jul 2012 16:30:58 To: FreeSWITCH Users Help Reply-To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] freeSWITCH on windows _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From bdfoster at endigotech.com Thu Jul 26 04:04:30 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Wed, 25 Jul 2012 20:04:30 -0400 Subject: [Freeswitch-users] Google Inc. DOES use FreeSWITCH for call center telephony routing system. So, why dont' we invite those guys to FS weekly conference :) In-Reply-To: References: Message-ID: It's only a matter of time :) Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 25, 2012 6:05 PM, "Michael Collins" wrote: > Of course, I'd say the crowning achievement would be FreeSWITCH being used > at Digium HQ. :P > -MC > > On Wed, Jul 25, 2012 at 2:48 PM, Brian Foster wrote: > >> Anthony, >> >> I bet you feel proud, eh? >> >> Brian Foster >> Endigo Computer LLC >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120725/8cded761/attachment.html From anthony.minessale at gmail.com Thu Jul 26 04:11:46 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Jul 2012 19:11:46 -0500 Subject: [Freeswitch-users] Google Inc. DOES use FreeSWITCH for call center telephony routing system. So, why dont' we invite those guys to FS weekly conference :) In-Reply-To: References: Message-ID: I'm mostly curious for any feedback or if they made any mods. On Wed, Jul 25, 2012 at 7:04 PM, Brian Foster wrote: > It's only a matter of time :) > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > > On Jul 25, 2012 6:05 PM, "Michael Collins" wrote: >> >> Of course, I'd say the crowning achievement would be FreeSWITCH being used >> at Digium HQ. :P >> -MC >> >> On Wed, Jul 25, 2012 at 2:48 PM, Brian Foster >> wrote: >>> >>> Anthony, >>> >>> I bet you feel proud, eh? >>> >>> Brian Foster >>> Endigo Computer LLC >>> >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From anthony.minessale at gmail.com Thu Jul 26 04:22:54 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 25 Jul 2012 19:22:54 -0500 Subject: [Freeswitch-users] leg_delay_start does not work in bridge enterprise In-Reply-To: <500FA9BA.5030303@thewinelake.com> References: <9D0E55F4-768E-4C9D-B1DF-9D074E44ADD7@mgtech.com> <458A0010-EC8C-4ACC-B08D-A02E0AD26077@mgtech.com> <500FA9BA.5030303@thewinelake.com> Message-ID: You can use it in each sub originate between :_: but not at the top level, that would require some kind of feature bounty enhancement. each string sep by the :_: is its own entire function originate, so it really would never work how you expect once you clear that up. On Wed, Jul 25, 2012 at 3:09 AM, Alex wrote: > Also suffering from this problem and it's a biggie for us. > I suppose the problem is due to the way in which the legs communicate. > Looks like using enterprise originate was just a little too optimistic! > Wish I'd known about this sooner and could have got on with writing our > own lua script that can do the job better. > If anyone's done this already and prepared to share, do get in touch. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From sdevoy at bizfocused.com Thu Jul 26 05:07:52 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Wed, 25 Jul 2012 21:07:52 -0400 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL Message-ID: <057401cd6acb$14f83a90$3ee8afb0$@bizfocused.com> I really hope this does not blow up into an ideological jihad, but I am curious what people think and if they have any evidence to support their claims. I am in fact a long time windows developer (anyone remember windows 2.2 - that was hard shit). I have chosen what may be a unique approach to FS "Configuration, Command and Control". I started using FS by building it on Centos 5.n. Fighting, scratching and clawing it into a working multi-tenant switch. Looking back, everything I needed to do was in the email tree, I just didn't know what to ask. I have now moved FS on to a "production" VPS server for $30 a month with amazing success. See it here: http://www.synapseglobal.com/voip_services.php It comes prebuilt with Centos and FS latest build compiled and ready to go. That's not why I am writing though. Their tech support is adamant that it can handle up 12 CONCURRENT CALLS at the base configuration. I learned about IPTABLES and FAIL2BAN and like them very much. However, I still work better/faster/surer in my Windows environment. So, I have taken what some might think is the worst possible approach: Configure and Control my Centos FS Server from my own ASP.NET Web Application (hosted elsewhere). My approach is to use the socket interface to send commands and use programmatic SFTP to the SHH shell for XML file exchange. I am about 85% done with version 1.0 and very pleased with it. I hope to have customers be able to login and modify their own configurations (call routes, IVRs, extensions mapping to devices line keys, Cisco spa504g provisioning, etc). Other device provisioning is in the pipe, but we have all 504Gs here and the provisioning code has been a tremendous help. Anyway, that is how I got to the odd work configuration, now I would like a discussion: My belief is that the "slim profile" of Centos will allow FS to handle greater load on a given hardware profile than could be handled by FS on Windows with the same hardware. I would like to AVOID the issues of security for this discussion, I firmly believe that you will provide better security on the platform that you understand the best. Let's just talk RAM, MIPS, NICs and FS performance and other issues I might be missing. Should a Windows guy go with FS on Windows or do you really get more bang for your buck in a Unix environment? I look forward to reading your thoughts. Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120725/cab94dfd/attachment-0001.html From roger.castaldo at gmail.com Thu Jul 26 06:38:05 2012 From: roger.castaldo at gmail.com (Roger Castaldo) Date: Wed, 25 Jul 2012 22:38:05 -0400 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: <057401cd6acb$14f83a90$3ee8afb0$@bizfocused.com> References: <057401cd6acb$14f83a90$3ee8afb0$@bizfocused.com> Message-ID: Given that an idle Linux machine properly configured will always consume less CPU and ram than any widows machine I have ever used my vote is for Linux On Jul 25, 2012 8:08 PM, "Sean Devoy" wrote: > I really hope this does not blow up into an ideological jihad, but I am > curious what people think and if they have any evidence to support their > claims. I am in fact a long time windows developer (anyone remember > windows 2.2 ? that was hard shit). I have chosen what may be a unique > approach to FS ?Configuration, Command and Control?.**** > > ** ** > > I started using FS by building it on Centos 5.n. Fighting, scratching and > clawing it into a working multi-tenant switch. Looking back, everything I > needed to do was in the email tree, I just didn?t know what to ask. I have > now moved FS on to a ?production? VPS server for $30 a month with amazing > success. See it here: http://www.synapseglobal.com/voip_services.php It > comes prebuilt with Centos and FS latest build compiled and ready to go. > That?s not why I am writing though. Their tech support is adamant that it > can handle up 12 CONCURRENT CALLS at the base configuration.**** > > ** ** > > I learned about IPTABLES and FAIL2BAN and like them very much. However, I > still work better/faster/surer in my Windows environment. So, I have taken > what some might think is the worst possible approach: Configure and Control > my Centos FS Server from my own ASP.NET Web Application (hosted > elsewhere). My approach is to use the socket interface to send commands > and use programmatic SFTP to the SHH shell for XML file exchange. I am > about 85% done with version 1.0 and very pleased with it. I hope to have > customers be able to login and modify their own configurations (call > routes, IVRs, extensions mapping to devices line keys, Cisco spa504g > provisioning, etc). Other device provisioning is in the pipe, but we have > all 504Gs here and the provisioning code has been a tremendous help.**** > > ** ** > > Anyway, that is how I got to the odd work configuration, now I would like > a discussion:**** > > My belief is that the ?slim profile? of Centos will allow FS to handle > greater load on a given hardware profile than could be handled by FS on > Windows with the same hardware. I would like to AVOID the issues of > security for this discussion, I firmly believe that you will provide better > security on the platform that you understand the best. Let?s just talk > RAM, MIPS, NICs and FS performance and other issues I might be missing.*** > * > > ** ** > > Should a Windows guy go with FS on Windows or do you really get more bang > for your buck in a Unix environment?**** > > ** ** > > I look forward to reading your thoughts.**** > > ** ** > > Sean**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120725/12964e7f/attachment.html From gabe at gundy.org Thu Jul 26 06:49:44 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 25 Jul 2012 20:49:44 -0600 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: <057401cd6acb$14f83a90$3ee8afb0$@bizfocused.com> References: <057401cd6acb$14f83a90$3ee8afb0$@bizfocused.com> Message-ID: On Wed, Jul 25, 2012 at 7:07 PM, Sean Devoy wrote: > Their tech support is adamant that it can handle up 12 CONCURRENT CALLS at > the base configuration. That's a pretty low number of concurrent calls. I have no reason to disbelieve him. > My belief is that the ?slim profile? of Centos will allow FS to handle > greater load on a given hardware profile than could be handled by FS on > Windows with the same hardware. I would like to AVOID the issues of > security for this discussion, I firmly believe that you will provide better > security on the platform that you understand the best. Let?s just talk RAM, > MIPS, NICs and FS performance and other issues I might be missing. The CentOS install will require less overhead for the OS when compared to Windows. However, with a beefy box, it wouldn't even matter. FS is said to run nicely on Windows and there is no doubt that it performs well on Linux. I would only recommend Linux to someone who already knows Windows when dealing with *minimally* configured servers. In those situations, it becomes obvious (to me) that Linux is the better fit. > Should a Windows guy go with FS on Windows or do you really get more bang > for your buck in a Unix environment? Generally speaking, yes. More bang for your buck. I would go with what makes you feel comfortable for now. But, I would also encourage you to deepen your Linux know-how. All of the power, flexibility, stability, hackability, malleability, and reliability that you find in FreeSWITCH can often be found for free in the Linux ecosystem. You can expect to invest some time learning it --just like learning FreeSWITCH took time (heck, you've been learning Windows for nearly 25 years). Summary: go with what you feel good about today. Don't cheat yourself by shying away from Linux tomorrow. Good luck, Gabe From gabe at gundy.org Thu Jul 26 06:58:47 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 25 Jul 2012 20:58:47 -0600 Subject: [Freeswitch-users] Google Inc. DOES use FreeSWITCH for call center telephony routing system. So, why dont' we invite those guys to FS weekly conference :) In-Reply-To: References: Message-ID: On Wed, Jul 25, 2012 at 12:10 PM, Gabriel Gunderson wrote: > On Wed, Jul 25, 2012 at 11:46 AM, Anton Kvashenkin > wrote: >> Guys, according to those slides (*LINK* slide number 10). > > FYI, it looks like it slide #9. http://bit.ly/GOOGandFS That link takes you right to the slide... share it with your Asterisk and Avaya buddies ;) Gabe From ifoundthetao at gmail.com Thu Jul 26 07:03:54 2012 From: ifoundthetao at gmail.com (Timothy Bolton) Date: Wed, 25 Jul 2012 22:03:54 -0500 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: References: <057401cd6acb$14f83a90$3ee8afb0$@bizfocused.com> Message-ID: <5010B39A.8060202@gmail.com> Typically, when you're working with Linux, and in this case CentOS Linux, you can get into the nitty gritty with "relative" ease. CentOS is an OS on a conservative release schedule (with CentOS 6 to be EOL'ed in 2020). Which means, while bug-fixes and patches and updates are implemented, newer features generally aren't. So what that does, is offer you a very predictable platform on which you can run your system. Practical, predictable, and when set up properly, pretty low for resource consumption. With CentOS, I'm typically "Go CLI or go home". I also don't set up email on these servers, things which heavily poll will bog down your performance. This also adds a learning curve, if you're not used to working with the CLI (command line interface). I'd also like to piggy-back onto what Gabriel said, but in a different light. If you're working with a customer base, you really want to make sure you know what you're doing if it hits the fan. For instance, if you set up a funky iptables rule which won't allow anything through, or something along those lines. Not that it should (or would) direct you away from Linux for your FS environment, it's just something to be aware of. That being said, I think that FreeBSD would be the best bang for your buck. A verrry rough comparison of Linux vs BSD would be: Linux is "home grown", while BSD is "engineered". 'We who cut mere stones must always be envisioning cathedrals.' Quarry Worker's Creed On 7/25/2012 9:49 PM, Gabriel Gunderson wrote: > On Wed, Jul 25, 2012 at 7:07 PM, Sean Devoy wrote: >> Their tech support is adamant that it can handle up 12 CONCURRENT CALLS at >> the base configuration. > That's a pretty low number of concurrent calls. I have no reason to > disbelieve him. > > >> My belief is that the ?slim profile? of Centos will allow FS to handle >> greater load on a given hardware profile than could be handled by FS on >> Windows with the same hardware. I would like to AVOID the issues of >> security for this discussion, I firmly believe that you will provide better >> security on the platform that you understand the best. Let?s just talk RAM, >> MIPS, NICs and FS performance and other issues I might be missing. > The CentOS install will require less overhead for the OS when compared > to Windows. However, with a beefy box, it wouldn't even matter. FS is > said to run nicely on Windows and there is no doubt that it performs > well on Linux. I would only recommend Linux to someone who already > knows Windows when dealing with *minimally* configured servers. In > those situations, it becomes obvious (to me) that Linux is the better > fit. > > >> Should a Windows guy go with FS on Windows or do you really get more bang >> for your buck in a Unix environment? > Generally speaking, yes. More bang for your buck. > > I would go with what makes you feel comfortable for now. But, I would > also encourage you to deepen your Linux know-how. All of the power, > flexibility, stability, hackability, malleability, and reliability > that you find in FreeSWITCH can often be found for free in the Linux > ecosystem. You can expect to invest some time learning it --just like > learning FreeSWITCH took time (heck, you've been learning Windows for > nearly 25 years). > > Summary: go with what you feel good about today. Don't cheat yourself > by shying away from Linux tomorrow. > > > Good luck, > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From gabe at gundy.org Thu Jul 26 07:34:17 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 25 Jul 2012 21:34:17 -0600 Subject: [Freeswitch-users] SSD Tuning for Linux now on wiki In-Reply-To: <545AC920-E5A3-4570-A980-234C80E4A15B@mgtech.com> References: <545AC920-E5A3-4570-A980-234C80E4A15B@mgtech.com> Message-ID: On Mon, Jul 16, 2012 at 11:36 AM, Mario G wrote: > As promised, small but lot's of work when into this, everything was actually tested and in use. During the past few months I found many misconceptions and errors on the web about SSDs so this should be a time saver. Enjoy! Thanks for the contribution! Gabe From curriegrad2004 at gmail.com Thu Jul 26 08:52:48 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 25 Jul 2012 21:52:48 -0700 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: <5010B39A.8060202@gmail.com> References: <057401cd6acb$14f83a90$3ee8afb0$@bizfocused.com> <5010B39A.8060202@gmail.com> Message-ID: CentOS 6 should be avoided at all costs at the time of writing this. There are performance issues inside of it which Ken Rice hasn't had the time or energy to chase down. If you're really looking for help with FreeSWITCH on a Windows platform, you might want to talk to Dave Kompel about it. He's the resident Windows guru here ;) On Wed, Jul 25, 2012 at 8:03 PM, Timothy Bolton wrote: > Typically, when you're working with Linux, and in this case CentOS > Linux, you can get into the nitty gritty with "relative" ease. CentOS > is an OS on a conservative release schedule (with CentOS 6 to be EOL'ed > in 2020). Which means, while bug-fixes and patches and updates are > implemented, newer features generally aren't. So what that does, is > offer you a very predictable platform on which you can run your system. > Practical, predictable, and when set up properly, pretty low for > resource consumption. > > With CentOS, I'm typically "Go CLI or go home". I also don't set up > email on these servers, things which heavily poll will bog down your > performance. This also adds a learning curve, if you're not used to > working with the CLI (command line interface). > > I'd also like to piggy-back onto what Gabriel said, but in a different > light. If you're working with a customer base, you really want to make > sure you know what you're doing if it hits the fan. For instance, if > you set up a funky iptables rule which won't allow anything through, or > something along those lines. Not that it should (or would) direct you > away from Linux for your FS environment, it's just something to be aware of. > > That being said, I think that FreeBSD would be the best bang for your > buck. A verrry rough comparison of Linux vs BSD would be: Linux is > "home grown", while BSD is "engineered". > > > 'We who cut mere stones must always be envisioning cathedrals.' > Quarry Worker's Creed > > On 7/25/2012 9:49 PM, Gabriel Gunderson wrote: >> On Wed, Jul 25, 2012 at 7:07 PM, Sean Devoy wrote: >>> Their tech support is adamant that it can handle up 12 CONCURRENT CALLS at >>> the base configuration. >> That's a pretty low number of concurrent calls. I have no reason to >> disbelieve him. >> >> >>> My belief is that the ?slim profile? of Centos will allow FS to handle >>> greater load on a given hardware profile than could be handled by FS on >>> Windows with the same hardware. I would like to AVOID the issues of >>> security for this discussion, I firmly believe that you will provide better >>> security on the platform that you understand the best. Let?s just talk RAM, >>> MIPS, NICs and FS performance and other issues I might be missing. >> The CentOS install will require less overhead for the OS when compared >> to Windows. However, with a beefy box, it wouldn't even matter. FS is >> said to run nicely on Windows and there is no doubt that it performs >> well on Linux. I would only recommend Linux to someone who already >> knows Windows when dealing with *minimally* configured servers. In >> those situations, it becomes obvious (to me) that Linux is the better >> fit. >> >> >>> Should a Windows guy go with FS on Windows or do you really get more bang >>> for your buck in a Unix environment? >> Generally speaking, yes. More bang for your buck. >> >> I would go with what makes you feel comfortable for now. But, I would >> also encourage you to deepen your Linux know-how. All of the power, >> flexibility, stability, hackability, malleability, and reliability >> that you find in FreeSWITCH can often be found for free in the Linux >> ecosystem. You can expect to invest some time learning it --just like >> learning FreeSWITCH took time (heck, you've been learning Windows for >> nearly 25 years). >> >> Summary: go with what you feel good about today. Don't cheat yourself >> by shying away from Linux tomorrow. >> >> >> Good luck, >> Gabe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mgg at giagnocavo.net Thu Jul 26 09:00:47 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Thu, 26 Jul 2012 05:00:47 +0000 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: <057401cd6acb$14f83a90$3ee8afb0$@bizfocused.com> References: <057401cd6acb$14f83a90$3ee8afb0$@bizfocused.com> Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B612F81D9B@BLUPRD0711MB413.namprd07.prod.outlook.com> I've run FreeSWITCH under Linux and Windows. On Windows, I had no problems sustaining hundreds of calls/sec, over 2000 sessions (without media). And that was in a virtualized (Hyper-V) system with a Q6600 processor; I think about 2 cores were used. One big caveat for 32-bit FS on Windows. FS is...liberal...when it comes to managing threads. It doesn't use lightweight threads or continuations, it really just spawns threads left and right. On Windows, the default stack is 1MB, so you can quickly run out of memory addresses on a 32-bit process. So, either run x64 FS, or adjust the stack size. Unless you're really getting into high-end performance, then I highly doubt the OS matters. And even then, I haven't seen any benchmarks showing Linux with much higher performance for FreeSWITCH. The more important stuff is the rest of the stack, like fail2ban, OpenSIPS, iptables versus the not-so-amazing Windows Firewall, etc. As others have said, definitely go with whatever makes you more comfortable. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean Devoy Sent: Wednesday, July 25, 2012 7:08 PM To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] What's better Unix ro Windows? LOL I really hope this does not blow up into an ideological jihad, but I am curious what people think and if they have any evidence to support their claims. I am in fact a long time windows developer (anyone remember windows 2.2 - that was hard shit). I have chosen what may be a unique approach to FS "Configuration, Command and Control". I started using FS by building it on Centos 5.n. Fighting, scratching and clawing it into a working multi-tenant switch. Looking back, everything I needed to do was in the email tree, I just didn't know what to ask. I have now moved FS on to a "production" VPS server for $30 a month with amazing success. See it here: http://www.synapseglobal.com/voip_services.php It comes prebuilt with Centos and FS latest build compiled and ready to go. That's not why I am writing though. Their tech support is adamant that it can handle up 12 CONCURRENT CALLS at the base configuration. I learned about IPTABLES and FAIL2BAN and like them very much. However, I still work better/faster/surer in my Windows environment. So, I have taken what some might think is the worst possible approach: Configure and Control my Centos FS Server from my own ASP.NET Web Application (hosted elsewhere). My approach is to use the socket interface to send commands and use programmatic SFTP to the SHH shell for XML file exchange. I am about 85% done with version 1.0 and very pleased with it. I hope to have customers be able to login and modify their own configurations (call routes, IVRs, extensions mapping to devices line keys, Cisco spa504g provisioning, etc). Other device provisioning is in the pipe, but we have all 504Gs here and the provisioning code has been a tremendous help. Anyway, that is how I got to the odd work configuration, now I would like a discussion: My belief is that the "slim profile" of Centos will allow FS to handle greater load on a given hardware profile than could be handled by FS on Windows with the same hardware. I would like to AVOID the issues of security for this discussion, I firmly believe that you will provide better security on the platform that you understand the best. Let's just talk RAM, MIPS, NICs and FS performance and other issues I might be missing. Should a Windows guy go with FS on Windows or do you really get more bang for your buck in a Unix environment? I look forward to reading your thoughts. Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120726/938e935d/attachment.html From krice at freeswitch.org Thu Jul 26 09:19:13 2012 From: krice at freeswitch.org (Ken Rice) Date: Thu, 26 Jul 2012 00:19:13 -0500 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B612F81D9B@BLUPRD0711MB413.namprd07.prod.outlook.com> Message-ID: Tony demonstrated FreeSWITCH running 1000cps 30sec call duration with media for a total of 30K concurrent calls @ ClueCon last year (just a few weeks shy of a year ago) That being said, does that mean you need to choose Linux over Windows? No it does, use the the tools that fit your specific deployment requirements... For some people that means Windows, for others that means Linux or BSD or ${something_else_entirely}... But as Michael just said you are better off using the 64bit builds for more then just stack size reasons... On 7/26/12 12:00 AM, "Michael Giagnocavo" wrote: > I?ve run FreeSWITCH under Linux and Windows. On Windows, I had no problems > sustaining hundreds of calls/sec, over 2000 sessions (without media). And that > was in a virtualized (Hyper-V) system with a Q6600 processor; I think about 2 > cores were used. > > One big caveat for 32-bit FS on Windows. FS is?liberal?when it comes to > managing threads. It doesn?t use lightweight threads or continuations, it > really just spawns threads left and right. On Windows, the default stack is > 1MB, so you can quickly run out of memory addresses on a 32-bit process. So, > either run x64 FS, or adjust the stack size. > > Unless you?re really getting into high-end performance, then I highly doubt > the OS matters. And even then, I haven?t seen any benchmarks showing Linux > with much higher performance for FreeSWITCH. The more important stuff is the > rest of the stack, like fail2ban, OpenSIPS, iptables versus the not-so-amazing > Windows Firewall, etc. > > As others have said, definitely go with whatever makes you more comfortable. > > -Michael > > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean Devoy > Sent: Wednesday, July 25, 2012 7:08 PM > To: FreeSWITCH-users at lists.freeswitch.org > Subject: [Freeswitch-users] What's better Unix ro Windows? LOL > > I really hope this does not blow up into an ideological jihad, but I am > curious what people think and if they have any evidence to support their > claims. I am in fact a long time windows developer (anyone remember windows > 2.2 ? that was hard shit). I have chosen what may be a unique approach to FS > ?Configuration, Command and Control?. > > I started using FS by building it on Centos 5.n. Fighting, scratching and > clawing it into a working multi-tenant switch. Looking back, everything I > needed to do was in the email tree, I just didn?t know what to ask. I have > now moved FS on to a ?production? VPS server for $30 a month with amazing > success. See it here: http://www.synapseglobal.com/voip_services.php It comes > prebuilt with Centos and FS latest build compiled and ready to go. That?s not > why I am writing though. Their tech support is adamant that it can handle up > 12 CONCURRENT CALLS at the base configuration. > > I learned about IPTABLES and FAIL2BAN and like them very much. However, I > still work better/faster/surer in my Windows environment. So, I have taken > what some might think is the worst possible approach: Configure and Control my > Centos FS Server from my own ASP.NET Web Application (hosted elsewhere). My > approach is to use the socket interface to send commands and use programmatic > SFTP to the SHH shell for XML file exchange. I am about 85% done with version > 1.0 and very pleased with it. I hope to have customers be able to login and > modify their own configurations (call routes, IVRs, extensions mapping to > devices line keys, Cisco spa504g provisioning, etc). Other device > provisioning is in the pipe, but we have all 504Gs here and the provisioning > code has been a tremendous help. > > Anyway, that is how I got to the odd work configuration, now I would like a > discussion: > My belief is that the ?slim profile? of Centos will allow FS to handle greater > load on a given hardware profile than could be handled by FS on Windows with > the same hardware. I would like to AVOID the issues of security for this > discussion, I firmly believe that you will provide better security on the > platform that you understand the best. Let?s just talk RAM, MIPS, NICs and FS > performance and other issues I might be missing. > > Should a Windows guy go with FS on Windows or do you really get more bang for > your buck in a Unix environment? > > I look forward to reading your thoughts. > > Sean > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120726/7beb7901/attachment-0001.html From ssinyagin at yahoo.com Thu Jul 26 10:50:13 2012 From: ssinyagin at yahoo.com (Stanislav Sinyagin) Date: Wed, 25 Jul 2012 23:50:13 -0700 (PDT) Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: <057401cd6acb$14f83a90$3ee8afb0$@bizfocused.com> References: <057401cd6acb$14f83a90$3ee8afb0$@bizfocused.com> Message-ID: <1343285413.95347.YahooMailNeo@web39303.mail.mud.yahoo.com> you want a server platform on your server, in the first place. Also you don't want a desktop GUI to manage your server. [/sarcasm] >________________________________ > From: Sean Devoy >To: FreeSWITCH-users at lists.freeswitch.org >Sent: Thursday, July 26, 2012 3:07 AM >Subject: [Freeswitch-users] What's better Unix ro Windows? LOL > > >I really hope this does not blow up into an ideological jihad, but I am curious what people think and if they have any evidence to support their claims.? I am in fact a long time windows developer (anyone remember windows 2.2 ? that was hard shit).? I have chosen what may be a unique approach to FS ?Configuration, Command and Control?. >? >I started using FS by building it on Centos 5.n.? Fighting, scratching and clawing it into a working multi-tenant switch. ??Looking back, everything I needed to do was in the email tree, I just didn?t know what to ask.? I have now moved FS on to a ?production? VPS server for $30 a month with amazing success. See it here: http://www.synapseglobal.com/voip_services.php ?It comes prebuilt with Centos and FS latest build compiled and ready to go. ?That?s not why I am writing though.? Their tech support is adamant that it can handle up 12 CONCURRENT CALLS at the base configuration. >? >I learned about IPTABLES and FAIL2BAN and like them very much.? However, I still work better/faster/surer in my Windows environment.? So, I have taken what some might think is the worst possible approach: Configure and Control my Centos FS Server from my own ASP.NET Web Application (hosted elsewhere).? My approach is to use the socket interface to send commands and use programmatic SFTP to the SHH shell for XML file exchange.? I am about 85% done with version 1.0 and very pleased with it.? I hope to have customers be able to login and modify their own configurations (call routes, IVRs, extensions mapping to devices line keys, Cisco spa504g provisioning, etc).? Other device provisioning is in the pipe, but we have all 504Gs here and the provisioning code has been a tremendous help. >? >Anyway, that is how I got to the odd work configuration, now I would like a discussion: >My belief is that the ?slim profile? of Centos will allow FS to handle greater load on a given hardware profile than could be handled by FS on Windows with the same hardware.? I would like to AVOID the issues of security for this discussion, I firmly believe that you will provide better security on the platform that you understand the best.? Let?s just talk RAM, MIPS, NICs and FS performance and other issues I might be missing. >? >Should a Windows guy go with FS on Windows or do you really get more bang for your buck in a Unix environment? >? >I look forward to reading your thoughts. >? >Sean >? >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >Join Us At ClueCon - Aug 7-9, 2012 > >FreeSWITCH-users mailing list >FreeSWITCH-users at 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/20120725/69141803/attachment.html From chris at opencsta.org Thu Jul 26 10:50:52 2012 From: chris at opencsta.org (Chris Mylonas) Date: Thu, 26 Jul 2012 16:50:52 +1000 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: References: <057401cd6acb$14f83a90$3ee8afb0$@bizfocused.com> <5010B39A.8060202@gmail.com> Message-ID: <7EDA4871-BD37-4FFB-ABFA-1B65D6EC19BD@opencsta.org> > CentOS 6 should be avoided at all costs at the time of writing this. > There are performance issues inside of it which Ken Rice hasn't had > the time or energy to chase down. Interestingly some bsd folks warned me about CentOS 5 timing problems in the kernel when it was released. Is it a similar sort of thing with RedHat putting in their own kernel stuff? Are we talking a high load machine when mentioning performance problems? This is the first I've heard of CentOS 6 performance problems. Thanks for the heads up From krice at freeswitch.org Thu Jul 26 11:03:38 2012 From: krice at freeswitch.org (Ken Rice) Date: Thu, 26 Jul 2012 02:03:38 -0500 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: <7EDA4871-BD37-4FFB-ABFA-1B65D6EC19BD@opencsta.org> Message-ID: On FS, centos5 actually greatly outperforms centos6... Somethings not right but I havnet been able to put my finger on it... Once of these days I'll oprofile this thing and see whats up On 7/26/12 1:50 AM, "Chris Mylonas" wrote: > > >> CentOS 6 should be avoided at all costs at the time of writing this. >> There are performance issues inside of it which Ken Rice hasn't had >> the time or energy to chase down. > > Interestingly some bsd folks warned me about CentOS 5 timing problems in the > kernel when it was released. Is it a similar sort of thing with RedHat > putting in their own kernel stuff? > > Are we talking a high load machine when mentioning performance problems? > > This is the first I've heard of CentOS 6 performance problems. > > Thanks for the heads up > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From peter.olsson at visionutveckling.se Thu Jul 26 11:16:32 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 26 Jul 2012 07:16:32 +0000 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: <63B00DD1DA6A364E9F64A3A0BD2FE7B612F81D9B@BLUPRD0711MB413.namprd07.prod.outlook.com> References: <057401cd6acb$14f83a90$3ee8afb0$@bizfocused.com>, <63B00DD1DA6A364E9F64A3A0BD2FE7B612F81D9B@BLUPRD0711MB413.namprd07.prod.outlook.com> Message-ID: <46CD25E7-FE45-4D9F-B153-D366E1FD1B10@visionutveckling.se> I believe the stack size in FS is automatically set to 240k for new threads, even on Windows. However, x64 is of course a much better choice anyway. /Peter 26 jul 2012 kl. 07:06 skrev "Michael Giagnocavo" >: I?ve run FreeSWITCH under Linux and Windows. On Windows, I had no problems sustaining hundreds of calls/sec, over 2000 sessions (without media). And that was in a virtualized (Hyper-V) system with a Q6600 processor; I think about 2 cores were used. One big caveat for 32-bit FS on Windows. FS is?liberal?when it comes to managing threads. It doesn?t use lightweight threads or continuations, it really just spawns threads left and right. On Windows, the default stack is 1MB, so you can quickly run out of memory addresses on a 32-bit process. So, either run x64 FS, or adjust the stack size. Unless you?re really getting into high-end performance, then I highly doubt the OS matters. And even then, I haven?t seen any benchmarks showing Linux with much higher performance for FreeSWITCH. The more important stuff is the rest of the stack, like fail2ban, OpenSIPS, iptables versus the not-so-amazing Windows Firewall, etc. As others have said, definitely go with whatever makes you more comfortable. -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean Devoy Sent: Wednesday, July 25, 2012 7:08 PM To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] What's better Unix ro Windows? LOL I really hope this does not blow up into an ideological jihad, but I am curious what people think and if they have any evidence to support their claims. I am in fact a long time windows developer (anyone remember windows 2.2 ? that was hard shit). I have chosen what may be a unique approach to FS ?Configuration, Command and Control?. I started using FS by building it on Centos 5.n. Fighting, scratching and clawing it into a working multi-tenant switch. Looking back, everything I needed to do was in the email tree, I just didn?t know what to ask. I have now moved FS on to a ?production? VPS server for $30 a month with amazing success. See it here: http://www.synapseglobal.com/voip_services.php It comes prebuilt with Centos and FS latest build compiled and ready to go. That?s not why I am writing though. Their tech support is adamant that it can handle up 12 CONCURRENT CALLS at the base configuration. I learned about IPTABLES and FAIL2BAN and like them very much. However, I still work better/faster/surer in my Windows environment. So, I have taken what some might think is the worst possible approach: Configure and Control my Centos FS Server from my own ASP.NET Web Application (hosted elsewhere). My approach is to use the socket interface to send commands and use programmatic SFTP to the SHH shell for XML file exchange. I am about 85% done with version 1.0 and very pleased with it. I hope to have customers be able to login and modify their own configurations (call routes, IVRs, extensions mapping to devices line keys, Cisco spa504g provisioning, etc). Other device provisioning is in the pipe, but we have all 504Gs here and the provisioning code has been a tremendous help. Anyway, that is how I got to the odd work configuration, now I would like a discussion: My belief is that the ?slim profile? of Centos will allow FS to handle greater load on a given hardware profile than could be handled by FS on Windows with the same hardware. I would like to AVOID the issues of security for this discussion, I firmly believe that you will provide better security on the platform that you understand the best. Let?s just talk RAM, MIPS, NICs and FS performance and other issues I might be missing. Should a Windows guy go with FS on Windows or do you really get more bang for your buck in a Unix environment? I look forward to reading your thoughts. Sean !DSPAM:5010cd0632761292282155! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:5010cd0632761292282155! From peter.olsson at visionutveckling.se Thu Jul 26 11:28:36 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 26 Jul 2012 07:28:36 +0000 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: <057401cd6acb$14f83a90$3ee8afb0$@bizfocused.com> References: <057401cd6acb$14f83a90$3ee8afb0$@bizfocused.com> Message-ID: I've been running FS both on Windows and Linux, mostly on Windows though. I believe the best choice is the OS you know the best. FS performs well on both platforms, so I think you are more likely to get the Windows box running better then a Linux box, since you are more familiar with it. I do recommend at least Windows 2008 though, since older versions doesn't have as exact timing in the kernel. Windows 2008 R2 with 64-bit FS i the best choice. /Peter 26 jul 2012 kl. 03:17 skrev "Sean Devoy" >: I really hope this does not blow up into an ideological jihad, but I am curious what people think and if they have any evidence to support their claims. I am in fact a long time windows developer (anyone remember windows 2.2 ? that was hard shit). I have chosen what may be a unique approach to FS ?Configuration, Command and Control?. I started using FS by building it on Centos 5.n. Fighting, scratching and clawing it into a working multi-tenant switch. Looking back, everything I needed to do was in the email tree, I just didn?t know what to ask. I have now moved FS on to a ?production? VPS server for $30 a month with amazing success. See it here: http://www.synapseglobal.com/voip_services.php It comes prebuilt with Centos and FS latest build compiled and ready to go. That?s not why I am writing though. Their tech support is adamant that it can handle up 12 CONCURRENT CALLS at the base configuration. I learned about IPTABLES and FAIL2BAN and like them very much. However, I still work better/faster/surer in my Windows environment. So, I have taken what some might think is the worst possible approach: Configure and Control my Centos FS Server from my own ASP.NET Web Application (hosted elsewhere). My approach is to use the socket interface to send commands and use programmatic SFTP to the SHH shell for XML file exchange. I am about 85% done with version 1.0 and very pleased with it. I hope to have customers be able to login and modify their own configurations (call routes, IVRs, extensions mapping to devices line keys, Cisco spa504g provisioning, etc). Other device provisioning is in the pipe, but we have all 504Gs here and the provisioning code has been a tremendous help. Anyway, that is how I got to the odd work configuration, now I would like a discussion: My belief is that the ?slim profile? of Centos will allow FS to handle greater load on a given hardware profile than could be handled by FS on Windows with the same hardware. I would like to AVOID the issues of security for this discussion, I firmly believe that you will provide better security on the platform that you understand the best. Let?s just talk RAM, MIPS, NICs and FS performance and other issues I might be missing. Should a Windows guy go with FS on Windows or do you really get more bang for your buck in a Unix environment? I look forward to reading your thoughts. Sean !DSPAM:5010972032761429566116! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:5010972032761429566116! From alex at thewinelake.com Thu Jul 26 13:04:35 2012 From: alex at thewinelake.com (Alex) Date: Thu, 26 Jul 2012 10:04:35 +0100 Subject: [Freeswitch-users] leg_delay_start does not work in bridge enterprise In-Reply-To: References: <9D0E55F4-768E-4C9D-B1DF-9D074E44ADD7@mgtech.com> <458A0010-EC8C-4ACC-B08D-A02E0AD26077@mgtech.com> <500FA9BA.5030303@thewinelake.com> Message-ID: <50110823.2060604@thewinelake.com> I did wonder if there would be a cunning way to workaround this by having each enterprise destination that needed it to have a fake first destination eg :_:,[leg_delay_start=10]:_:,[leg_delay_start=20] The fake destination would never answer, but would have to be of the right type to avoid the problem that non-enterprise origin has (which is that it doesn't work with multiple SIP registrations) > You can use it in each sub originate between :_: but not at the top > level, that would require some kind of feature bounty enhancement. > each string sep by the :_: is its own entire function originate, so it > really would never work how you expect once you clear that up. > > > > On Wed, Jul 25, 2012 at 3:09 AM, Alex wrote: >> Also suffering from this problem and it's a biggie for us. >> I suppose the problem is due to the way in which the legs communicate. >> Looks like using enterprise originate was just a little too optimistic! >> Wish I'd known about this sooner and could have got on with writing our >> own lua script that can do the job better. >> If anyone's done this already and prepared to share, do get in touch. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > From g.d.monnezza at tiscali.it Thu Jul 26 02:12:30 2012 From: g.d.monnezza at tiscali.it (g) Date: Thu, 26 Jul 2012 00:12:30 +0200 Subject: [Freeswitch-users] AutoNAT - Local Networks not excluded In-Reply-To: <01df01cd69d1$231b3780$6951a680$@com> References: <01df01cd69d1$231b3780$6951a680$@com> Message-ID: <2078136.sHTGoXbj17@virtex> Thanks Phil. I'll try again, but my rtp-ip and sip-ip (the internal) are already correctly set (ie 192.168.1.200). The problem is that if the phone registered has belongs to a different (but still local) subnet (ie 192.168.2.143), then FS uses the external IP (ie 85.43.25.64) And no audio at all Any more suggestion? giuliano On Tuesday 24 July 2012 15:18:35 Phil Quesinberry wrote: > Set rtp-ip and sip-ip to your internal IP address. I believe that you > should also be able to set it to: $${local_ip_v4} or $${bind_server_ip} > as well. > > - Phil > > ---------- > VirteX g.d.monnezza at tiscali.it > 5D%20%20AutoNAT%20-%20Local%20Networks%20not%20excluded&In-Reply-To=> Tue > Jul 24 16:12:08 MSD 2012 > > > Hi guys. I appreciate so much the Auto-NAT for uPnP capable firewalls. But > I'm experiencing an issue. > I have a FreeSwitch server behind a NAT, but I can't find a way to avoid > FreeSwitch using external IP (for SIP and RTP) for local networks (i.e. > 192.168.0.0/16). > In my sip profiles for various interfaces I have NOT set the . > Anyway, the sofia status for all interfaces shows the EXT-RTP-IP and > EXT-SIP-IP set (with my public gateway IP). That's ok, even if I didn' > declard it with > My SIP phones register from a network different from the server one, but > still a local network. Then, SIP phones receive (from the server) the rtp > and sip signalling with its external IP. This prevent any communication. > How it is possible to tell FreeSwitch to NOT use ext IP for particular > networks? > Thanks to anyone who will point me in the right direction. > g From support at sping.nl Thu Jul 26 12:25:29 2012 From: support at sping.nl (Systeembeheer) Date: Thu, 26 Jul 2012 10:25:29 +0200 Subject: [Freeswitch-users] 'Dialed number' field does not contain true dialed number? In-Reply-To: References: <500CFD68.8060603@sping.nl> Message-ID: <5010FEF9.8030103@sping.nl> Thank you, that did the trick! Would it be an option to alter 'sip_profiles/external/example.xml' and add the 'auto_to_user' tip, like in the wiki? steveN On 07/23/2012 01:31 PM, Avi Marcus wrote: > You want: in your > gateway config. > > Via: http://wiki.freeswitch.org/wiki/Clarification:gateways > > > > -Avi > BestFone > > > On Mon, Jul 23, 2012 at 10:29 AM, Systeembeheer > wrote: > > Hi, > > Trying to get freeswitch to answer to a trunk with 10 different phone > numbers. With all inbound calls however, the 'destination number' in > conditions in the dialplan contains the login name of the gateway > (where > the trunk comes from), not the dialed number. The variable > 'sip_req_user' does contain the true dialed number, but why > doesn't the > 'dialed number' field? > > Thanks! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120726/371e003e/attachment.html From avi at avimarcus.net Thu Jul 26 19:00:40 2012 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 26 Jul 2012 18:00:40 +0300 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: References: <057401cd6acb$14f83a90$3ee8afb0$@bizfocused.com> Message-ID: tl;dr: I find maintenance, compiling, and low-level system information much easier in *nix. I'd just like to add a few things: I developed all my "tech expertise" on windows until the age of ~23. I've only been using Linux a year, but I can't imagine doing most of this stuff as easily on windows. Even on my Ubuntu desktop, I'm constantly in the terminal wget'ing or git cloning some source code, make'ing it, rsync'ing from/to my servers, or just popping up a terminal windows to manage things on my server... e.g. checking ram or CPU usage is just a few characters away ('free -m') A friend was bugging me to use windows and after installing it, I was having a really hard time even with cygwin to get all the same terminal options. There's just so many sys-admin tasks that are just easier with the CLI (and then it's great to have the same flexibility to do that on your desktop, too.) I used to use sshfs heavily to mount my server over ssh as a local drive, but there was a bit of lag when getting folder lists or saving -- which I remember is better than sftp anyway. I now have git repos of everything I usually manage, and use Salt-stack for configuration management (again, config fed via git), so to make most changes I just edit a local git repo and git push to my server over SSH. I know you can replicate this on windows, but not nearly as easily... -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120726/42837ad6/attachment.html From spencer at 5ninesolutions.com Thu Jul 26 21:02:08 2012 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Thu, 26 Jul 2012 10:02:08 -0700 Subject: [Freeswitch-users] mod_loopback pre answer Message-ID: <28C0A2BE-6990-4011-94FD-7D48F85A394D@5ninesolutions.com> Hello, I have a small multi-tenant installation where I need a hunt group where a user can do server side call forwarding on a per extension basis. I'd also like the huntgroup to send a 180 instead of 183 with SDP when ringing. I've currently solved this problem by constructing the bridge in a lua script like this: if session:ready() then session:execute("bridge", "[leg_timeout=30]user/1002@"..domain_name..",[leg_timeout=30]user/1003@"..domain_name..",[leg_timeout=30]user/1004@"..domain_name..",[leg_timeout=30]user/1005@"..domain_name..",[leg_timeout=25]loopback/1001"); --timeout originate_disposition = session:getVariable("originate_disposition"); if originate_disposition ~= "SUCCESS" then session:execute("transfer", "900"); end This allows extension 1001 to forward their calls by inserting the appropriate dialplan entry prior to the bridge to the local user. My problem is that when using loopback, ring_ready does not work because of a hard coded pre answer in mod loopback at line 969: switch_channel_pre_answer(channel);. So I have a few questions: Is there a better way to construct this kind of call forwarding/follow me without loopback at all? i.e. is there any way to set user defined call forwarding when do a bridge to a local user? If not, what are the implications of commenting this line in the code? Would I then be able to send a 180 w/o SDP when using loopback? Thanks for your help! Spencer From steve at teltechcorp.com Thu Jul 26 21:06:29 2012 From: steve at teltechcorp.com (Stephen Corsen) Date: Thu, 26 Jul 2012 13:06:29 -0400 Subject: [Freeswitch-users] (no subject) Message-ID: Hi, I'm trying to use mod_ladspa in freeswitch. I'd like to do the equivalent of the following via the socket connection using the 'api' command: (the above is take from /src/mod/applications/mod_ladspa/conf/dialplan/00_ladspa.xml) Given defined values for the parameters (AT_TUNE, AT_FIXED, etc), what should the 'api' command be (in particular, what is the syntax for a mod_ladspa command)? Thanks From krice at freeswitch.org Thu Jul 26 21:12:15 2012 From: krice at freeswitch.org (Ken Rice) Date: Thu, 26 Jul 2012 12:12:15 -0500 Subject: [Freeswitch-users] Just a Reminder on asking for help on a problem... Message-ID: Hey Guys, I know there are a pile of us that answer questions on the mailing list regularly and there are some of us that work with or on the core dev team that are very well versed in FS and can probably answer questions on ${random_problem_you_are_trying_to_figure_out}, but please don?t just email one guy idk manon the dev team directly for help unless this is something they have explicitly asked you to do (or unless you are offering them some sort of compensation for helping you (read $$$$)). We have a Mailing list so that everyone can benefit from the free help, and so you can get answers faster. (how is it faster? If ${developer} isnt around maybe ${other_user} can help you out with your question. This frees up the devs to continue doing what they do best which is coding and fixing issues, and it just makes the community stronger. Also the mailing list is archived and heavily indexed by Google and other search engines which means maybe someone else can find they answer they are looking for later without even asking... Thanks for understand! See you at ClueCon! K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120726/081e907b/attachment.html From avi at avimarcus.net Thu Jul 26 21:19:05 2012 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 26 Jul 2012 20:19:05 +0300 Subject: [Freeswitch-users] Just a Reminder on asking for help on a problem... In-Reply-To: References: Message-ID: And also... if you have an issue that either you spend time figuring out or get help with, *please wikify it *. It's quite easy to get an account and make changes -- that way the information is at hand in a somewhat organized format. (Although the install information has gotten awfully scattered). -Avi Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120726/34889410/attachment.html From philq at qsystemsengineering.com Thu Jul 26 21:58:48 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Thu, 26 Jul 2012 13:58:48 -0400 Subject: [Freeswitch-users] Issues after enabling mod_memcache Message-ID: <01a901cd6b58$51fcf9c0$f5f6ed40$@com> After enabling mod_memcache (I didn't realize it was disabled by default), I get the following error when a cid lookup is performed: 2012-07-26 13:38:48.201415 [ERR] mod_memcache.c:350 Error while running command get: SYSTEM ERROR 2012-07-26 13:38:48.201415 [ERR] mod_memcache.c:350 Error while running command get: SYSTEM ERROR 2012-07-26 13:38:48.201415 [ERR] mod_memcache.c:350 Error while running command get: SYSTEM ERROR 2012-07-26 13:38:48.841598 [ERR] mod_memcache.c:350 Error while running command set: SYSTEM ERROR Am I missing something? I restarted FS after modifying modules.conf.xml and mod_memcache is present and uncomment in modules.conf. Not sure what else I should be doing to make this work. Thanks, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120726/26086884/attachment.html From darcy at Vex.Net Thu Jul 26 21:33:31 2012 From: darcy at Vex.Net (D'Arcy Cain) Date: Thu, 26 Jul 2012 13:33:31 -0400 Subject: [Freeswitch-users] Just a Reminder on asking for help on a problem... In-Reply-To: References: Message-ID: <20120726133331.fd260e6425873db9a3b5ac92@Vex.Net> On Thu, 26 Jul 2012 12:12:15 -0500 Ken Rice wrote: > We have a Mailing list so that everyone can benefit from the free help, and > so you can get answers faster. (how is it faster? If ${developer} isnt > around maybe ${other_user} can help you out with your question. This frees > up the devs to continue doing what they do best which is coding and fixing > issues, and it just makes the community stronger. Also the mailing list is http://www.catb.org/~esr/faqs/smart-questions.html is a good resource to check out before asking questions as well. -- D'Arcy J.M. Cain System Administrator, Vex.Net http://www.Vex.Net/ IM:darcy at Vex.Net From gabe at gundy.org Thu Jul 26 22:06:55 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Thu, 26 Jul 2012 12:06:55 -0600 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: References: <057401cd6acb$14f83a90$3ee8afb0$@bizfocused.com> Message-ID: On Thu, Jul 26, 2012 at 9:00 AM, Avi Marcus wrote: > I now have git repos of everything I usually manage, and use Salt-stack for > configuration management (again, config fed via git), so to make most > changes I just edit a local git repo and git push to my server over SSH. Another Salt user, eh? That's some good stuff. We abandoned our homegrown solution and moved to Salt. It's wicked nice for the kind of setups that we typically deploy FreeSWITCH in. Best, Gabe From aksrini at hotmail.com Thu Jul 26 22:11:02 2012 From: aksrini at hotmail.com (Srini K) Date: Thu, 26 Jul 2012 11:11:02 -0700 Subject: [Freeswitch-users] No reINVITE when bridging two sessions from mod_managed with bypass_media_after_bridge=true In-Reply-To: <20120721181514.c5444fbc@mail.tritonwest.net> References: , <20120721181514.c5444fbc@mail.tritonwest.net> Message-ID: Thanks Dave, for pointing out mistakes. It did help and I can bypass media. The issue with Session.Execute("bridge","sofia/gateway...") is not getting a finer control over outbound leg before bridge. For eg, I need to play a prompt on bLeg(outbound) before bridging. How do I do it? Is there a way? I tried Session.Execute("bridge","{bypass_media_after_bridge=true, group_confirm_file=/path/to/prompt.wav}sofia/gateway/...") and also Session.Execute("bridge","{bypass_media_after_bridge=true, playback=/path/to/prompt.wav}sofia/gateway/...") without any success. I was able to do it with ease using new ManagedSession. -SriniFrom: drk at drkngs.net To: freeswitch-users at lists.freeswitch.org Date: Sat, 21 Jul 2012 11:15:14 -0700 Subject: Re: [Freeswitch-users] No reINVITE when bridging two sessions from mod_managed with bypass_media_after_bridge=true Andrew, There are a bunch of things wrong with your example for what you're trying to do. The first has nothing to do w/ mod_managed, which is that you are trying to set variables on the a_leg, that are used by the bridge application, after the b_leg is up. The media variables that you set on the a_leg, are arguments to the dial plan app "bridge". Since you are never using it to place the outbound call for the b_leg, they're not going to work. Another problem is that mod_managed doesen't really work right to place an outbound call with the "new ManagedSession". That is left over from old days. What's wrong with doing a "Session.Execute("bridge","sofia/gateway...")"? If you do this then any flags that are used as parameters by the bridge applicaiton will work, since you are just calling it as you would from the dialplan. --Dave From: Andrew Cassidy [mailto:andrew at cassidywebservices.co.uk] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Sat, 21 Jul 2012 10:12:06 -0700 Subject: Re: [Freeswitch-users] No reINVITE when bridging two sessions from mod_managed with bypass_media_after_bridge=true Try using uuid_media after the bridge instaed, see if that helps? http://wiki.freeswitch.org/wiki/Mod_commands#uuid_media On 20 July 2012 21:30, Srini K wrote: Hi, Iam trying to bypass media from FS after two call legs are bridged using mod_managed. The code looks like public void Run(AppContext context) { var fsApi = new FreeSWITCH.Native.Api(); var aLegSession = context.Session; // Answer the incoming call aLegSession.Answer(); // Play the prompt aLegSession.StreamFile("ivr/ThankYou.wav", 0); // Create outBound session var bLegSession = new ManagedSession("sofia/gateway/95/4151230000"); // Bypass Media aLegSession.SetVariable("bypass_media_after_bridge", "true"); bLegSession.SetVariable("bypass_media_after_bridge", "true"); fsApi.ExecuteString(string.Format("uuid_bridge {0} {1}", aLegSession.GetUuid(), bLegSession.GetUuid())); } I don't see FreeSWITCH sending re-Invite after the call is bridged. What I've already tried and did not succeed: 1) set bypass_media=true, on A leg only, on B leg only, on both legs 2) set bypass_media_after_bridge=true, on A leg only, on B leg only, on both legs When I tried without using mod_managed using only dialplan, FS sends re-Invite. Whether Iam doing anything stupid in mod_managed? Regards Srini _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Andrew Cassidy BSc (Hons) MBCS SSCA Managing Director T 03300 100 960 F 03300 100 961 E andrew at cassidywebservices.co.uk W www.cassidywebservices.co.uk _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120726/b51bb373/attachment-0001.html From gabe at gundy.org Thu Jul 26 22:12:40 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Thu, 26 Jul 2012 12:12:40 -0600 Subject: [Freeswitch-users] mod_ladspa Was: (no subject) Message-ID: On Thu, Jul 26, 2012 at 11:06 AM, Stephen Corsen wrote: > I'm trying to use mod_ladspa in freeswitch. I'd like to do the > equivalent of the following via the socket connection using the 'api' > command: You're going to want to use a subject in the email. Not doing so almost guarantees that it gets overlooked. This becomes more important as you start looking for help in the areas of FreeSWITCH that are less widely used (as I suspect mod_ladspa is). Good luck, Gabe From drk at drkngs.net Thu Jul 26 22:49:24 2012 From: drk at drkngs.net (Dave R. Kompel) Date: Thu, 26 Jul 2012 11:49:24 -0700 Subject: [Freeswitch-users] =?iso-8859-1?q?What=27s_better_Unix_ro_Windows?= =?iso-8859-1?q?=3F_LOL?= In-Reply-To: Message-ID: <20120726184924.3e844d2d@mail.tritonwest.net> A few things you should also note: If you're running a low end system, that is not heavy on ram and CPU, forget about windows. This is also coming from a windows person. However if you are using a high end system, then you should have no problem running FS under windows, and depending on your application, it may out perform the *ix box. With that said, Windows8/Server 2012 are about 1 week away from RTM. FS running on Server 2012RC handles about 25% more load then the equivlent box running Server 2008 R2. Also much better support for Server Core (No GUI) on 2012, then 2008 R2. FS can be installed on 2008 R2 core, however it's a real pain in the ass, where it's simple on 2012 Core. I also have a reduced memory footprint on 2012Core and a system processing about 2000 sessions, at 50 CPS, running my softswitch module, and SQL installed on the same box runs in about 880mb of ram. So if you're going to try it on windows, you may want to hold out for a week, till the official release of Server 2012. --Dave _____ From: Avi Marcus [mailto:avi at avimarcus.net] To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] Sent: Thu, 26 Jul 2012 08:00:40 -0700 Subject: Re: [Freeswitch-users] What's better Unix ro Windows? LOL tl;dr: I find maintenance, compiling, and low-level system information much easier in *nix. I'd just like to add a few things: I developed all my "tech expertise" on windows until the age of ~23. I've only been using Linux a year, but I can't imagine doing most of this stuff as easily on windows. Even on my Ubuntu desktop, I'm constantly in the terminal wget'ing or git cloning some source code, make'ing it, rsync'ing from/to my servers, or just popping up a terminal windows to manage things on my server... e.g. checking ram or CPU usage is just a few characters away ('free -m') A friend was bugging me to use windows and after installing it, I was having a really hard time even with cygwin to get all the same terminal options. There's just so many sys-admin tasks that are just easier with the CLI (and then it's great to have the same flexibility to do that on your desktop, too.) I used to use sshfs heavily to mount my server over ssh as a local drive, but there was a bit of lag when getting folder lists or saving -- which I remember is better than sftp anyway. I now have git repos of everything I usually manage, and use Salt-stack for configuration management (again, config fed via git), so to make most changes I just edit a local git repo and git push to my server over SSH. I know you can replicate this on windows, but not nearly as easily... -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120726/f5369285/attachment.html From bdfoster at endigotech.com Thu Jul 26 23:06:17 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 26 Jul 2012 15:06:17 -0400 Subject: [Freeswitch-users] Issues after enabling mod_memcache In-Reply-To: <01a901cd6b58$51fcf9c0$f5f6ed40$@com> References: <01a901cd6b58$51fcf9c0$f5f6ed40$@com> Message-ID: apt-get install memcached Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 26, 2012 2:01 PM, "Phil Quesinberry" wrote: > ** > > After enabling mod_memcache (I didn?t realize it was disabled by default), > I get the following error when a cid lookup is performed: > > 2012-07-26 13:38:48.201415 [ERR] mod_memcache.c:350 Error while running > command get: SYSTEM ERROR > > 2012-07-26 13:38:48.201415 [ERR] mod_memcache.c:350 Error while running > command get: SYSTEM ERROR > > 2012-07-26 13:38:48.201415 [ERR] mod_memcache.c:350 Error while running > command get: SYSTEM ERROR > > 2012-07-26 13:38:48.841598 [ERR] mod_memcache.c:350 Error while running > command set: SYSTEM ERROR > > Am I missing something? I restarted FS after modifying modules.conf.xml > and mod_memcache is present and uncomment in modules.conf. Not sure what > else I should be doing to make this work. > > Thanks, > > *******Phil Quesinberry* > > Q Systems Engineering, Inc. > > Electronic Controls and Embedded Systems Development > > (410) 969-8002 > > *****http://www.qsystemsengineering.com* > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120726/2c7e72b8/attachment.html From philq at qsystemsengineering.com Thu Jul 26 23:35:03 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Thu, 26 Jul 2012 15:35:03 -0400 Subject: [Freeswitch-users] Issues after enabling mod_memcache Message-ID: <021501cd6b65$c4187ae0$4c4970a0$@com> Oh geez. I didn't realize that was a Linux package/service I needed to install. My inexperience with Linux is showing and my face is red. Starting the service after doing a yum install memcached, followed by a 'chkconfig --level 345 memcached on' did the trick (CentOS). Thank you! - Phil apt-get install memcached Brian Foster Endigo Computer LLC _____________________________________________ From: Phil Quesinberry Sent: Thursday, July 26, 2012 1:59 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: Issues after enabling mod_memcache After enabling mod_memcache (I didn't realize it was disabled by default), I get the following error when a cid lookup is performed: 2012-07-26 13:38:48.201415 [ERR] mod_memcache.c:350 Error while running command get: SYSTEM ERROR 2012-07-26 13:38:48.201415 [ERR] mod_memcache.c:350 Error while running command get: SYSTEM ERROR 2012-07-26 13:38:48.201415 [ERR] mod_memcache.c:350 Error while running command get: SYSTEM ERROR 2012-07-26 13:38:48.841598 [ERR] mod_memcache.c:350 Error while running command set: SYSTEM ERROR Am I missing something? I restarted FS after modifying modules.conf.xml and mod_memcache is present and uncomment in modules.conf. Not sure what else I should be doing to make this work. Thanks, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120726/15c5765c/attachment-0001.html From philq at qsystemsengineering.com Thu Jul 26 23:59:05 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Thu, 26 Jul 2012 15:59:05 -0400 Subject: [Freeswitch-users] Setting effecting_caller_id_name Message-ID: <024801cd6b69$1cfaf7c0$56f0e740$@com> And while I'm asking dumb questions. When doing CNAM dips from opencnam.com, often you get a result of "Currently running a lookup for phone 'xxxxxxxxxx'. on incoming calls, typically for wireless or other unknown name callers and I wanted to change that to "Wireless/Unknown" Since caller_id_name is apparently read-only, I am attempting to set effective_caller_id_name. I put the following in public.xml right below the "fix_cidnam_plus" entry, in other words after a CNAM lookup has been performed. If I crafted my regex properly, then it should be matching on the first part of the string and setting the variable appropriately. Is 'effective_caller_id_name' the variable I should be setting? Many thanks, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120726/2da91371/attachment.html From prasd.d.b at gmail.com Fri Jul 27 01:17:03 2012 From: prasd.d.b at gmail.com (Prasd D) Date: Thu, 26 Jul 2012 14:17:03 -0700 Subject: [Freeswitch-users] Push server and push capability Message-ID: I was searching and came across this amazing jira request for push server capability. This is exactly what I felt too, except didn't delve or earlier understand how to achieve this. http://jira.freeswitch.org/browse/FS-4347 This guy has identified very much what is really needed to be able to use it on smartphones. And suggests that Freeswitch and Csipsimple together show this capability where an incoming call or voicemail to a user using Csipsimple is notified through push notification. That starts Csipsimple (which is otherwise turned off) on his phone. This is a huge huge and amazingly useful thing since no one keeps their voip client on their phones on due to power consumption and battery drain ! This I think is the future of VOIP since smartphones are going to become soon huge % of voip clients. From bdfoster at endigotech.com Fri Jul 27 01:28:53 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 26 Jul 2012 17:28:53 -0400 Subject: [Freeswitch-users] Setting effecting_caller_id_name In-Reply-To: <024801cd6b69$1cfaf7c0$56f0e740$@com> References: <024801cd6b69$1cfaf7c0$56f0e740$@com> Message-ID: You need a $ after 'lookup' for it to be a regex. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 26, 2012 4:01 PM, "Phil Quesinberry" wrote: > ** > > And while I?m asking dumb questions? > > When doing CNAM dips from opencnam.com, often you get a result of ?Currently > running a lookup for phone ?xxxxxxxxxx?? on incoming calls, typically for > wireless or other unknown name callers and I wanted to change that to ? > Wireless/Unknown? Since caller_id_name is apparently read-only, I am > attempting to set effective_caller_id_name. I put the following in > public.xml right below the ?fix_cidnam_plus? entry, in other words after > a CNAM lookup has been performed. > > > > > > data="effective_caller_id_name=Wireless/Unknown"/> > > > > > > If I crafted my regex properly, then it should be matching on the first > part of the string and setting the variable appropriately. Is ? > effective_caller_id_name? the variable I should be setting? > > Many thanks, > > *******Phil Quesinberry* > > Q Systems Engineering, Inc. > > Electronic Controls and Embedded Systems Development > > (410) 969-8002 > > *****http://www.qsystemsengineering.com* > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120726/3e708b3d/attachment.html From anthony.minessale at gmail.com Fri Jul 27 01:37:23 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Jul 2012 16:37:23 -0500 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: <20120726184924.3e844d2d@mail.tritonwest.net> References: <20120726184924.3e844d2d@mail.tritonwest.net> Message-ID: If everything you are doing is via ESL and remote control, it won't matter too much to you which one you run. If you already have windows stuff you feel comfortable deploying its fine. I lean to Linux just because I am most agile there and I tend to do most of my dev on ssh terms and I have a ton of boxes I can control and jump between. Also I can get to them from nearly any pc around me without slow rdp type stuff (nxwin is pretty fricken fast tho) Raw performance, I would say windows might have a minor disadvantage because there is a lot of compatibility layers between top level calls and resulting operations. What might be a native call in Linux might be a slower drop-in replacement in Win. We can only control so much of this since we have tons of depends that all use their own method for portability. I think the call volume and stress on the server needed to measure the difference is fairly high in the many hundreds of calls. So bottom line, try it. It should work fine, if it doesn't ask here, just make sure you mention Windows in the subject to attract the right responses. On Thu, Jul 26, 2012 at 1:49 PM, Dave R. Kompel wrote: > A few things you should also note: > > If you're running a low end system, that is not heavy on ram and CPU, forget > about windows. This is also coming from a windows person. However if you are > using a high end system, then you should have no problem running FS under > windows, and depending on your application, it may out perform the *ix box. > > With that said, Windows8/Server 2012 are about 1 week away from RTM. FS > running on Server 2012RC handles about 25% more load then the equivlent box > running Server 2008 R2. Also much better support for Server Core (No GUI) on > 2012, then 2008 R2. FS can be installed on 2008 R2 core, however it's a real > pain in the ass, where it's simple on 2012 Core. > > I also have a reduced memory footprint on 2012Core and a system processing > about 2000 sessions, at 50 CPS, running my softswitch module, and SQL > installed on the same box runs in about 880mb of ram. > > So if you're going to try it on windows, you may want to hold out for a > week, till the official release of Server 2012. > > --Dave > > ________________________________ > From: Avi Marcus [mailto:avi at avimarcus.net] > To: FreeSWITCH Users Help [mailto:freeswitch-users at lists.freeswitch.org] > Sent: Thu, 26 Jul 2012 08:00:40 -0700 > Subject: Re: [Freeswitch-users] What's better Unix ro Windows? LOL > > > tl;dr: I find maintenance, compiling, and low-level system information much > easier in *nix. > > I'd just like to add a few things: > I developed all my "tech expertise" on windows until the age of ~23. > I've only been using Linux a year, but I can't imagine doing most of this > stuff as easily on windows. Even on my Ubuntu desktop, I'm constantly in the > terminal wget'ing or git cloning some source code, make'ing it, rsync'ing > from/to my servers, or just popping up a terminal windows to manage things > on my server... e.g. checking ram or CPU usage is just a few characters away > ('free -m') > > A friend was bugging me to use windows and after installing it, I was having > a really hard time even with cygwin to get all the same terminal options. > There's just so many sys-admin tasks that are just easier with the CLI (and > then it's great to have the same flexibility to do that on your desktop, > too.) > > I used to use sshfs heavily to mount my server over ssh as a local drive, > but there was a bit of lag when getting folder lists or saving -- which I > remember is better than sftp anyway. > I now have git repos of everything I usually manage, and use Salt-stack for > configuration management (again, config fed via git), so to make most > changes I just edit a local git repo and git push to my server over SSH. I > know you can replicate this on windows, but not nearly as easily... > > -Avi > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From jrichey at itltd.net Fri Jul 27 01:42:38 2012 From: jrichey at itltd.net (JRichey) Date: Thu, 26 Jul 2012 14:42:38 -0700 Subject: [Freeswitch-users] Setting effecting_caller_id_name Message-ID: <6ECAF1527329364583AB525CF34ABF950B31A6B6@ms.kallback.com> '$' is the 'end of line' match and not a requirement for a regex. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org]On Behalf Of Brian Foster Sent: Thursday, July 26, 2012 2:29 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Setting effecting_caller_id_name You need a $ after 'lookup' for it to be a regex. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 26, 2012 4:01 PM, "Phil Quesinberry" < philq at qsystemsengineering.com > wrote: And while I?m asking dumb questions? When doing CNAM dips from opencnam.com , often you get a result of ?Currently running a lookup for phone ?xxxxxxxxxx?? on incoming calls, typically for wireless or other unknown name callers and I wanted to change that to ?Wireless/Unknown? Since caller_id_name is apparently read-only, I am attempting to set effective_caller_id_name. I put the following in public.xml right below the ?fix_cidnam_plus? entry, in other words after a CNAM lookup has been performed. If I crafted my regex properly, then it should be matching on the first part of the string and setting the variable appropriately. Is ?effective_caller_id_name? the variable I should be setting? Many thanks, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com <> Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120726/6e2ccf34/attachment-0001.html From philq at qsystemsengineering.com Fri Jul 27 01:46:00 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Thu, 26 Jul 2012 17:46:00 -0400 Subject: [Freeswitch-users] Setting effecting_caller_id_name Message-ID: <000c01cd6b78$0c5f4470$251dcd50$@com> If you put the $ at the end then it will try to match the entire string instead of just the beginning of it, which won't work in this case. Is there a way to match just the beginning of the string in FS? Thanks, - Phil You need a $ after 'lookup' for it to be a regex. Brian Foster Endigo Computer LLC _____________________________________________ From: Phil Quesinberry Sent: Thursday, July 26, 2012 3:59 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: Setting effecting_caller_id_name And while I'm asking dumb questions. When doing CNAM dips from opencnam.com, often you get a result of "Currently running a lookup for phone 'xxxxxxxxxx'. on incoming calls, typically for wireless or other unknown name callers and I wanted to change that to "Wireless/Unknown" Since caller_id_name is apparently read-only, I am attempting to set effective_caller_id_name. I put the following in public.xml right below the "fix_cidnam_plus" entry, in other words after a CNAM lookup has been performed. If I crafted my regex properly, then it should be matching on the first part of the string and setting the variable appropriately. Is 'effective_caller_id_name' the variable I should be setting? Many thanks, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120726/1142a778/attachment.html From s at star2star.com Fri Jul 27 00:56:47 2012 From: s at star2star.com (Shawn Solomon) Date: Thu, 26 Jul 2012 16:56:47 -0400 Subject: [Freeswitch-users] mod_native_file and mod_file_string interoperability Message-ID: Hello, Hopefully someone can pass along a tip to get mod_native_file to process files from mod_file_string, or something equally awesome. My goal is to have something like: without having to present extensions, thus avoiding transcoding. Thanks, S -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120726/61fcd131/attachment.html From yufei.tao at redembedded.com Fri Jul 27 02:04:13 2012 From: yufei.tao at redembedded.com (yufei.tao) Date: Thu, 26 Jul 2012 23:04:13 +0100 Subject: [Freeswitch-users] H264 transcoding Message-ID: <5011BEDD.8070509@redembedded.com> Hi I am trying to decide if it is feasible to let FS do transcoding between different H264 formats for live video calls. This is because I've got SIP clients that both use H264 but with different formats and one (with a bad H264 decoder) has problems decoding H264 stream from the other. But each of these two clients communicate fine using H264 with a third client that uses ffmpeg. I'm thinking of adding a module which uses ffmpeg, so that it will transcode H264 between different parameters. I've got a few questions: 1. Is this feasible? I'm not looking at supporting many simultaneous calls. 2. What is involved in transcoding real-time video stream? 3. Anyone's done anything like this before? I'm new to FS and any suggestions would be very much appreciated! Yufei From krice at freeswitch.org Fri Jul 27 03:53:54 2012 From: krice at freeswitch.org (Ken Rice) Date: Thu, 26 Jul 2012 18:53:54 -0500 Subject: [Freeswitch-users] H264 transcoding In-Reply-To: <5011BEDD.8070509@redembedded.com> Message-ID: Is it possible sure... Is ot probably to happen anytime soon? Not until the patents run out... On 7/26/12 5:04 PM, "yufei.tao" wrote: > Hi > > I am trying to decide if it is feasible to let FS do transcoding between > different H264 formats for live video calls. This is because I've got > SIP clients that both use H264 but with different formats and one (with > a bad H264 decoder) has problems decoding H264 stream from the other. > But each of these two clients communicate fine using H264 with a third > client that uses ffmpeg. I'm thinking of adding a module which uses > ffmpeg, so that it will transcode H264 between different parameters. > > I've got a few questions: > > 1. Is this feasible? I'm not looking at supporting many simultaneous calls. > 2. What is involved in transcoding real-time video stream? > 3. Anyone's done anything like this before? > > I'm new to FS and any suggestions would be very much appreciated! > > Yufei > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 opencsta.org Fri Jul 27 04:01:33 2012 From: chris at opencsta.org (Chris Mylonas) Date: Fri, 27 Jul 2012 10:01:33 +1000 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: References: <20120726184924.3e844d2d@mail.tritonwest.net> Message-ID: <2A416971-0F73-41EC-AAC2-5B11AAE5CB76@opencsta.org> > I tend to do most of my dev on ssh terms v.interesting. how the hell do you manage that? emacs/vim? From bdfoster at endigotech.com Fri Jul 27 04:11:45 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 26 Jul 2012 20:11:45 -0400 Subject: [Freeswitch-users] Setting effecting_caller_id_name In-Reply-To: <000c01cd6b78$0c5f4470$251dcd50$@com> References: <000c01cd6b78$0c5f4470$251dcd50$@com> Message-ID: You can just not use a regex. Do you need to escape the spaces? Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 26, 2012 5:47 PM, "Phil Quesinberry" wrote: > ** > > If you put the $ at the end then it will try to match the entire string > instead of just the beginning of it, which won?t work in this case. Is there > a way to match just the beginning of the string in FS? > > Thanks, > > - Phil > > You need a $ after 'lookup' for it to be a regex. > > Brian Foster > > Endigo Computer LLC > > _____________________________________________ > *****From:* Phil Quesinberry > *****Sent:* Thursday, July 26, 2012 3:59 PM > *****To:* 'freeswitch-users at lists.freeswitch.org' > *****Subject:* Setting effecting_caller_id_name > > And while I?m asking dumb questions? > > When doing CNAM dips from opencnam.com, often you get a result of > ?Currently running a lookup for phone ?xxxxxxxxxx?? on incoming calls, > typically for wireless or other unknown name callers and I wanted to change > that to ?Wireless/Unknown? Since caller_id_name is apparently read-only, I > am attempting to set effective_caller_id_name. I put the following in > public.xml right below the ?fix_cidnam_plus? entry, in other words after a > CNAM lookup has been performed. > > > > > > data="effective_caller_id_name=Wireless/Unknown"/> > > > > > > If I crafted my regex properly, then it should be matching on the first > part of the string and setting the variable appropriately. Is > ?effective_caller_id_name? the variable I should be setting? > > Many thanks, > > *******Phil Quesinberry* > > Q Systems Engineering, Inc. > > Electronic Controls and Embedded Systems Development > > (410) 969-8002 > > ***http://www.qsystemsengineering.com* > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120726/267ba779/attachment-0001.html From mario_fs at mgtech.com Fri Jul 27 04:32:35 2012 From: mario_fs at mgtech.com (Mario G) Date: Thu, 26 Jul 2012 17:32:35 -0700 Subject: [Freeswitch-users] leg_delay_start does not work in bridge enterprise In-Reply-To: <50110823.2060604@thewinelake.com> References: <9D0E55F4-768E-4C9D-B1DF-9D074E44ADD7@mgtech.com> <458A0010-EC8C-4ACC-B08D-A02E0AD26077@mgtech.com> <500FA9BA.5030303@thewinelake.com> <50110823.2060604@thewinelake.com> Message-ID: This is why I had to bridge to internal extensions, then after a timeout another bridge to all extensions plus the cells. And this is why if you pickup exactly between extensions you have to hangup and pickup to get the call. A real pain. Dropping enterprise fixes it but caused other issues. I am interested if your trick will work, but the even if it does the potential problem is that FS may change and this "trick" might stop working. I had that happen when I upgraded FS this year. Had situations were sometimes the caller was actually one of the targets (don't ask why), worked from 2010 to spring 2012 and the bridge stopped working. Something changed and I had to execute another extension so I could test the caller (can't do nested if/then/else). So the moral of the story is that unless the technique is officially supported, it may not work in the future. On Jul 26, 2012, at 2:04 AM, Alex wrote: > I did wonder if there would be a cunning way to workaround this by > having each enterprise destination that needed it to have a fake first > destination > > > eg > > :_:,[leg_delay_start=10]:_:,[leg_delay_start=20] > > The fake destination would never answer, but would have to be of the > right type to avoid the problem that non-enterprise origin has (which is > that it doesn't work with multiple SIP registrations) >> You can use it in each sub originate between :_: but not at the top >> level, that would require some kind of feature bounty enhancement. >> each string sep by the :_: is its own entire function originate, so it >> really would never work how you expect once you clear that up. >> >> >> >> On Wed, Jul 25, 2012 at 3:09 AM, Alex wrote: >>> Also suffering from this problem and it's a biggie for us. >>> I suppose the problem is due to the way in which the legs communicate. >>> Looks like using enterprise originate was just a little too optimistic! >>> Wish I'd known about this sooner and could have got on with writing our >>> own lua script that can do the job better. >>> If anyone's done this already and prepared to share, do get in touch. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From bdfoster at endigotech.com Fri Jul 27 04:32:49 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 26 Jul 2012 20:32:49 -0400 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: <2A416971-0F73-41EC-AAC2-5B11AAE5CB76@opencsta.org> References: <20120726184924.3e844d2d@mail.tritonwest.net> <2A416971-0F73-41EC-AAC2-5B11AAE5CB76@opencsta.org> Message-ID: Or pussies like me use nano. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 26, 2012 8:02 PM, "Chris Mylonas" wrote: > > I tend to do most of my dev on ssh terms > > > v.interesting. > how the hell do you manage that? > > emacs/vim? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120726/95a39965/attachment.html From krice at freeswitch.org Fri Jul 27 04:38:46 2012 From: krice at freeswitch.org (Ken Rice) Date: Thu, 26 Jul 2012 19:38:46 -0500 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: <2A416971-0F73-41EC-AAC2-5B11AAE5CB76@opencsta.org> References: <20120726184924.3e844d2d@mail.tritonwest.net> <2A416971-0F73-41EC-AAC2-5B11AAE5CB76@opencsta.org> Message-ID: most of freeswitch has been developedin an ssh terminal using screen and emacs... Ken Sent from my iPad On Jul 26, 2012, at 7:01 PM, Chris Mylonas wrote: >> I tend to do most of my dev on ssh terms > > > v.interesting. > how the hell do you manage that? > > emacs/vim? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From terry at digital-outpost.com Fri Jul 27 04:45:04 2012 From: terry at digital-outpost.com (Terry Barnum) Date: Thu, 26 Jul 2012 17:45:04 -0700 Subject: [Freeswitch-users] H264 transcoding In-Reply-To: References: Message-ID: Use x264? http://en.wikipedia.org/wiki/X264 On Jul 26, 2012, at 4:53 PM, Ken Rice wrote: > Is it possible sure... Is ot probably to happen anytime soon? Not until the > patents run out... > > > On 7/26/12 5:04 PM, "yufei.tao" wrote: > >> Hi >> >> I am trying to decide if it is feasible to let FS do transcoding between >> different H264 formats for live video calls. This is because I've got >> SIP clients that both use H264 but with different formats and one (with >> a bad H264 decoder) has problems decoding H264 stream from the other. >> But each of these two clients communicate fine using H264 with a third >> client that uses ffmpeg. I'm thinking of adding a module which uses >> ffmpeg, so that it will transcode H264 between different parameters. >> >> I've got a few questions: >> >> 1. Is this feasible? I'm not looking at supporting many simultaneous calls. >> 2. What is involved in transcoding real-time video stream? >> 3. Anyone's done anything like this before? >> >> I'm new to FS and any suggestions would be very much appreciated! >> >> Yufei >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From gabe at gundy.org Fri Jul 27 04:48:09 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Thu, 26 Jul 2012 18:48:09 -0600 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: References: <20120726184924.3e844d2d@mail.tritonwest.net> <2A416971-0F73-41EC-AAC2-5B11AAE5CB76@opencsta.org> Message-ID: On Thu, Jul 26, 2012 at 6:38 PM, Ken Rice wrote: > most of freeswitch has been developedin an ssh terminal using screen and emacs... I love screen, and well... that's all ;) :wq From curriegrad2004 at gmail.com Fri Jul 27 05:05:32 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Thu, 26 Jul 2012 18:05:32 -0700 Subject: [Freeswitch-users] H264 transcoding In-Reply-To: References: Message-ID: x264's license is incompatible with FreeSWITCH. As it stands for now, GPL is incompatible with MPL. On Thu, Jul 26, 2012 at 5:45 PM, Terry Barnum wrote: > Use x264? http://en.wikipedia.org/wiki/X264 > > On Jul 26, 2012, at 4:53 PM, Ken Rice wrote: > >> Is it possible sure... Is ot probably to happen anytime soon? Not until the >> patents run out... >> >> >> On 7/26/12 5:04 PM, "yufei.tao" wrote: >> >>> Hi >>> >>> I am trying to decide if it is feasible to let FS do transcoding between >>> different H264 formats for live video calls. This is because I've got >>> SIP clients that both use H264 but with different formats and one (with >>> a bad H264 decoder) has problems decoding H264 stream from the other. >>> But each of these two clients communicate fine using H264 with a third >>> client that uses ffmpeg. I'm thinking of adding a module which uses >>> ffmpeg, so that it will transcode H264 between different parameters. >>> >>> I've got a few questions: >>> >>> 1. Is this feasible? I'm not looking at supporting many simultaneous calls. >>> 2. What is involved in transcoding real-time video stream? >>> 3. Anyone's done anything like this before? >>> >>> I'm new to FS and any suggestions would be very much appreciated! >>> >>> Yufei >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From philq at qsystemsengineering.com Fri Jul 27 06:28:45 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Thu, 26 Jul 2012 22:28:45 -0400 Subject: [Freeswitch-users] Setting effecting_caller_id_name Message-ID: <009f01cd6b9f$8eb103b0$ac130b10$@com> The reply is different each time, depending upon the number being looked up. So, I just want to look at the first part of the string. If FS can't do a regex match without the trailing $, I'm guessing there's a way to just do it in XML. I'll try and see what I can find after the storm passes unless you have a better idea, I need to shut this computer down right now. Thanks, - Phil You can just not use a regex. Do you need to escape the spaces? Brian Foster Endigo Computer LLC _____________________________________________ From: Phil Quesinberry Sent: Thursday, July 26, 2012 5:46 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: RE: Setting effecting_caller_id_name If you put the $ at the end then it will try to match the entire string instead of just the beginning of it, which won't work in this case. Is there a way to match just the beginning of the string in FS? Thanks, - Phil You need a $ after 'lookup' for it to be a regex. Brian Foster Endigo Computer LLC _____________________________________________ From: Phil Quesinberry Sent: Thursday, July 26, 2012 3:59 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: Setting effecting_caller_id_name And while I'm asking dumb questions. When doing CNAM dips from opencnam.com, often you get a result of "Currently running a lookup for phone 'xxxxxxxxxx'. on incoming calls, typically for wireless or other unknown name callers and I wanted to change that to "Wireless/Unknown" Since caller_id_name is apparently read-only, I am attempting to set effective_caller_id_name. I put the following in public.xml right below the "fix_cidnam_plus" entry, in other words after a CNAM lookup has been performed. If I crafted my regex properly, then it should be matching on the first part of the string and setting the variable appropriately. Is 'effective_caller_id_name' the variable I should be setting? Many thanks, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120726/9cabd6ea/attachment-0001.html From krice at freeswitch.org Fri Jul 27 07:19:30 2012 From: krice at freeswitch.org (Ken Rice) Date: Thu, 26 Jul 2012 22:19:30 -0500 Subject: [Freeswitch-users] H264 transcoding In-Reply-To: References: Message-ID: <5E0B0409-B5A4-4722-ACDA-8F6140EE49C7@freeswitch.org> we can not and will not use GPL software, the license is not compatible with the GPL and would polute the codebase with additional restrictions that are not wanted or needed. now if someone could get them to change the license or atleast give us a license under better terms such as the LGPL or the MPL then the license issue would be null Ken Sent from my iPad On Jul 26, 2012, at 7:45 PM, Terry Barnum wrote: > Use x264? http://en.wikipedia.org/wiki/X264 > > On Jul 26, 2012, at 4:53 PM, Ken Rice wrote: > >> Is it possible sure... Is ot probably to happen anytime soon? Not until the >> patents run out... >> >> >> On 7/26/12 5:04 PM, "yufei.tao" wrote: >> >>> Hi >>> >>> I am trying to decide if it is feasible to let FS do transcoding between >>> different H264 formats for live video calls. This is because I've got >>> SIP clients that both use H264 but with different formats and one (with >>> a bad H264 decoder) has problems decoding H264 stream from the other. >>> But each of these two clients communicate fine using H264 with a third >>> client that uses ffmpeg. I'm thinking of adding a module which uses >>> ffmpeg, so that it will transcode H264 between different parameters. >>> >>> I've got a few questions: >>> >>> 1. Is this feasible? I'm not looking at supporting many simultaneous calls. >>> 2. What is involved in transcoding real-time video stream? >>> 3. Anyone's done anything like this before? >>> >>> I'm new to FS and any suggestions would be very much appreciated! >>> >>> Yufei >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From curriegrad2004 at gmail.com Fri Jul 27 07:41:13 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Thu, 26 Jul 2012 20:41:13 -0700 Subject: [Freeswitch-users] H264 transcoding In-Reply-To: <5E0B0409-B5A4-4722-ACDA-8F6140EE49C7@freeswitch.org> References: <5E0B0409-B5A4-4722-ACDA-8F6140EE49C7@freeswitch.org> Message-ID: Ken, If you think those guys over at x264 will ever change the license from GPL to LGPL, you're just dreaming the pie in the sky... In short, don't even think about it ;P On Thu, Jul 26, 2012 at 8:19 PM, Ken Rice wrote: > we can not and will not use GPL software, the license is not compatible with the GPL and would polute the codebase with additional restrictions that are not wanted or needed. now if someone could get them to change the license or atleast give us a license under better terms such as the LGPL or the MPL then the license issue would be null > > Ken > Sent from my iPad > > On Jul 26, 2012, at 7:45 PM, Terry Barnum wrote: > >> Use x264? http://en.wikipedia.org/wiki/X264 >> >> On Jul 26, 2012, at 4:53 PM, Ken Rice wrote: >> >>> Is it possible sure... Is ot probably to happen anytime soon? Not until the >>> patents run out... >>> >>> >>> On 7/26/12 5:04 PM, "yufei.tao" wrote: >>> >>>> Hi >>>> >>>> I am trying to decide if it is feasible to let FS do transcoding between >>>> different H264 formats for live video calls. This is because I've got >>>> SIP clients that both use H264 but with different formats and one (with >>>> a bad H264 decoder) has problems decoding H264 stream from the other. >>>> But each of these two clients communicate fine using H264 with a third >>>> client that uses ffmpeg. I'm thinking of adding a module which uses >>>> ffmpeg, so that it will transcode H264 between different parameters. >>>> >>>> I've got a few questions: >>>> >>>> 1. Is this feasible? I'm not looking at supporting many simultaneous calls. >>>> 2. What is involved in transcoding real-time video stream? >>>> 3. Anyone's done anything like this before? >>>> >>>> I'm new to FS and any suggestions would be very much appreciated! >>>> >>>> Yufei >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Jul 27 07:46:00 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 26 Jul 2012 22:46:00 -0500 Subject: [Freeswitch-users] H264 transcoding In-Reply-To: References: <5E0B0409-B5A4-4722-ACDA-8F6140EE49C7@freeswitch.org> Message-ID: you would probably need to do something like make a mod for ffmpeg that protects you from the gpl then allow the user to build that lib on his own and choose at compile time to install patented or adverse licensed components. No license rules prohibit an end user from combining code only distributors. but even then we need a bunch of code to write. On Thu, Jul 26, 2012 at 10:41 PM, curriegrad2004 wrote: > Ken, > > If you think those guys over at x264 will ever change the license from > GPL to LGPL, you're just dreaming the pie in the sky... > > In short, don't even think about it ;P > > On Thu, Jul 26, 2012 at 8:19 PM, Ken Rice wrote: >> we can not and will not use GPL software, the license is not compatible with the GPL and would polute the codebase with additional restrictions that are not wanted or needed. now if someone could get them to change the license or atleast give us a license under better terms such as the LGPL or the MPL then the license issue would be null >> >> Ken >> Sent from my iPad >> >> On Jul 26, 2012, at 7:45 PM, Terry Barnum wrote: >> >>> Use x264? http://en.wikipedia.org/wiki/X264 >>> >>> On Jul 26, 2012, at 4:53 PM, Ken Rice wrote: >>> >>>> Is it possible sure... Is ot probably to happen anytime soon? Not until the >>>> patents run out... >>>> >>>> >>>> On 7/26/12 5:04 PM, "yufei.tao" wrote: >>>> >>>>> Hi >>>>> >>>>> I am trying to decide if it is feasible to let FS do transcoding between >>>>> different H264 formats for live video calls. This is because I've got >>>>> SIP clients that both use H264 but with different formats and one (with >>>>> a bad H264 decoder) has problems decoding H264 stream from the other. >>>>> But each of these two clients communicate fine using H264 with a third >>>>> client that uses ffmpeg. I'm thinking of adding a module which uses >>>>> ffmpeg, so that it will transcode H264 between different parameters. >>>>> >>>>> I've got a few questions: >>>>> >>>>> 1. Is this feasible? I'm not looking at supporting many simultaneous calls. >>>>> 2. What is involved in transcoding real-time video stream? >>>>> 3. Anyone's done anything like this before? >>>>> >>>>> I'm new to FS and any suggestions would be very much appreciated! >>>>> >>>>> Yufei >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> Join Us At ClueCon - Aug 7-9, 2012 >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From ben at langfeld.co.uk Fri Jul 27 09:46:15 2012 From: ben at langfeld.co.uk (Ben Langfeld) Date: Fri, 27 Jul 2012 07:46:15 +0200 Subject: [Freeswitch-users] Push server and push capability In-Reply-To: References: Message-ID: You could, and probably should, implement this as an inbound event socket listener, rather than in FS core. You could do it in....ruby? http://github.com/adhearsion/ruby_fs Happy hacking. Regards, Ben Langfeld On 26 July 2012 23:17, Prasd D wrote: > I was searching and came across this amazing jira request for push > server capability. > This is exactly what I felt too, except didn't delve or earlier > understand how to achieve this. > > http://jira.freeswitch.org/browse/FS-4347 > > This guy has identified very much what is really needed to be able to > use it on smartphones. > And suggests that Freeswitch and Csipsimple together show this > capability where an incoming call or voicemail to a user using > Csipsimple is notified through push notification. That starts > Csipsimple (which is otherwise turned off) on his phone. > > This is a huge huge and amazingly useful thing since no one keeps > their voip client on their phones on due to power consumption and > battery drain ! > > This I think is the future of VOIP since smartphones are going to > become soon huge % of voip clients. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120727/65512112/attachment.html From william.king at quentustech.com Fri Jul 27 10:43:08 2012 From: william.king at quentustech.com (William King) Date: Thu, 26 Jul 2012 23:43:08 -0700 Subject: [Freeswitch-users] H264 transcoding In-Reply-To: References: <5E0B0409-B5A4-4722-ACDA-8F6140EE49C7@freeswitch.org> Message-ID: <5012387C.8050903@quentustech.com> libvlc is LGPL http://www.videolan.org/press/lgpl.html and there is now a mod_vlc(though it doesn't yet support video streams). The user can choose to build vlc with only the LGPL components or add the more 'adverse' modules. In none of the LGPL packages of libvlc is ffmpeg enabled, but there is a module for libvlc for ffmpeg. http://wiki.videolan.org/FFmpeg The only pieces now may just be the FS side of things for video. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 07/26/2012 08:46 PM, Anthony Minessale wrote: > you would probably need to do something like make a mod for ffmpeg > that protects you from the gpl then allow the user to build that lib > on his own and choose at compile time to install patented or adverse > licensed components. No license rules prohibit an end user from > combining code only distributors. > > but even then we need a bunch of code to write. > > > On Thu, Jul 26, 2012 at 10:41 PM, curriegrad2004 > wrote: >> Ken, >> >> If you think those guys over at x264 will ever change the license from >> GPL to LGPL, you're just dreaming the pie in the sky... >> >> In short, don't even think about it ;P >> >> On Thu, Jul 26, 2012 at 8:19 PM, Ken Rice wrote: >>> we can not and will not use GPL software, the license is not compatible with the GPL and would polute the codebase with additional restrictions that are not wanted or needed. now if someone could get them to change the license or atleast give us a license under better terms such as the LGPL or the MPL then the license issue would be null >>> >>> Ken >>> Sent from my iPad >>> >>> On Jul 26, 2012, at 7:45 PM, Terry Barnum wrote: >>> >>>> Use x264? http://en.wikipedia.org/wiki/X264 >>>> >>>> On Jul 26, 2012, at 4:53 PM, Ken Rice wrote: >>>> >>>>> Is it possible sure... Is ot probably to happen anytime soon? Not until the >>>>> patents run out... >>>>> >>>>> >>>>> On 7/26/12 5:04 PM, "yufei.tao" wrote: >>>>> >>>>>> Hi >>>>>> >>>>>> I am trying to decide if it is feasible to let FS do transcoding between >>>>>> different H264 formats for live video calls. This is because I've got >>>>>> SIP clients that both use H264 but with different formats and one (with >>>>>> a bad H264 decoder) has problems decoding H264 stream from the other. >>>>>> But each of these two clients communicate fine using H264 with a third >>>>>> client that uses ffmpeg. I'm thinking of adding a module which uses >>>>>> ffmpeg, so that it will transcode H264 between different parameters. >>>>>> >>>>>> I've got a few questions: >>>>>> >>>>>> 1. Is this feasible? I'm not looking at supporting many simultaneous calls. >>>>>> 2. What is involved in transcoding real-time video stream? >>>>>> 3. Anyone's done anything like this before? >>>>>> >>>>>> I'm new to FS and any suggestions would be very much appreciated! >>>>>> >>>>>> Yufei >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> Join Us At ClueCon - Aug 7-9, 2012 >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>>> http://www.freeswitch.org >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> Join Us At ClueCon - Aug 7-9, 2012 >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> Join Us At ClueCon - Aug 7-9, 2012 >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20120726/2308bfcb/attachment-0001.html From william.king at quentustech.com Fri Jul 27 10:53:59 2012 From: william.king at quentustech.com (William King) Date: Thu, 26 Jul 2012 23:53:59 -0700 Subject: [Freeswitch-users] HTTP Request from within the core In-Reply-To: References: Message-ID: <50123B07.6080604@quentustech.com> It sounds like you might be doing something similar to what mod_snom started with. I've worked heavily with phone automation and I would be curious which phones you're working with and your experiences with them. Here's an example I was working on to automate the testing of Polycom BLF lights and call scenarios: http://imgur.com/a/9JfcR William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 07/25/2012 06:47 AM, Gerald Weber wrote: > > Thats it, muchas gracias ! > > *Von:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von > *Brian Foster > *Gesendet:* Mittwoch, 25. Juli 2012 15:10 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] HTTP Request from within the core > > sofia_contact might be what you are looking for. > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > > On Jul 25, 2012 9:04 AM, "Gerald Weber" > wrote: > > Thanks, looks very easy to implement. > > Just on more question: > > I need to get the IP address of a registered user. > > E.g. user/2000 is a SNOM with ip 192.168.20.219 > > I guess a query against the registrations table ist the fastest way to > do this ? > > Or is there any core api command ? > > Thanks > > *Von:*freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *Im Auftrag > von *Avi Marcus > *Gesendet:* Mittwoch, 25. Juli 2012 14:26 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] HTTP Request from within the core > > You can use the API for curl or just check how the codebase for curl > works: > > http://wiki.freeswitch.org/wiki/Mod_curl -- src/mod/applications/mod_curl/ > > Or look at > http://wiki.freeswitch.org/wiki/Mod_http_cache -- src/mod/applications/mod_http_cache/ > > -Avi > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120726/87cc994c/attachment.html From peter.olsson at visionutveckling.se Fri Jul 27 12:09:11 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 27 Jul 2012 08:09:11 +0000 Subject: [Freeswitch-users] Setting effecting_caller_id_name In-Reply-To: <000c01cd6b78$0c5f4470$251dcd50$@com> References: <000c01cd6b78$0c5f4470$251dcd50$@com> Message-ID: <53CCE98C-9E1B-4501-953A-AC17C3D473C2@visionutveckling.se> Is fix_cidnam_plus setting it's data inline? I think it must update the caller_id_name var inline, to make it possible to do a regex on that variable right afterwards in the dialplan. /Peter 26 jul 2012 kl. 23:51 skrev "Phil Quesinberry" >: If you put the $ at the end then it will try to match the entire string instead of just the beginning of it, which won?t work in this case. Is there a way to match just the beginning of the string in FS? Thanks, - Phil You need a $ after 'lookup' for it to be a regex. Brian Foster Endigo Computer LLC _____________________________________________ From: Phil Quesinberry Sent: Thursday, July 26, 2012 3:59 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: Setting effecting_caller_id_name And while I?m asking dumb questions? When doing CNAM dips from opencnam.com, often you get a result of ?Currently running a lookup for phone ?xxxxxxxxxx?? on incoming calls, typically for wireless or other unknown name callers and I wanted to change that to ?Wireless/Unknown? Since caller_id_name is apparently read-only, I am attempting to set effective_caller_id_name. I put the following in public.xml right below the ?fix_cidnam_plus? entry, in other words after a CNAM lookup has been performed. If I crafted my regex properly, then it should be matching on the first part of the string and setting the variable appropriately. Is ?effective_caller_id_name? the variable I should be setting? Many thanks, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com !DSPAM:5011b86432763191115303! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org !DSPAM:5011b86432763191115303! From avi at avimarcus.net Fri Jul 27 13:55:06 2012 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 27 Jul 2012 12:55:06 +0300 Subject: [Freeswitch-users] Setting effecting_caller_id_name In-Reply-To: <53CCE98C-9E1B-4501-953A-AC17C3D473C2@visionutveckling.se> References: <000c01cd6b78$0c5f4470$251dcd50$@com> <53CCE98C-9E1B-4501-953A-AC17C3D473C2@visionutveckling.se> Message-ID: If this is still an issue, please paste a log of FS going over this extension. This entire thread has been only speculation. -Avi On Fri, Jul 27, 2012 at 11:09 AM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > Is fix_cidnam_plus setting it's data inline? I think it must update the > caller_id_name var inline, to make it possible to do a regex on that > variable right afterwards in the dialplan. > > /Peter > > 26 jul 2012 kl. 23:51 skrev "Phil Quesinberry" < > philq at qsystemsengineering.com>: > > > If you put the $ at the end then it will try to match the entire string > instead of just the beginning of it, which won?t work in this case. Is > there a way to match just the beginning of the string in FS? > > Thanks, > > - Phil > > You need a $ after 'lookup' for it to be a regex. > > Brian Foster > > Endigo Computer LLC > > _____________________________________________ > From: Phil Quesinberry > Sent: Thursday, July 26, 2012 3:59 PM > To: 'freeswitch-users at lists.freeswitch.org freeswitch-users at lists.freeswitch.org>' > Subject: Setting effecting_caller_id_name > > And while I?m asking dumb questions? > > When doing CNAM dips from opencnam.com, often you > get a result of ?Currently running a lookup for phone ?xxxxxxxxxx?? on > incoming calls, typically for wireless or other unknown name callers and I > wanted to change that to ?Wireless/Unknown? Since caller_id_name is > apparently read-only, I am attempting to set effective_caller_id_name. I > put the following in public.xml right below the ?fix_cidnam_plus? entry, in > other words after a CNAM lookup has been performed. > > > > > > data="effective_caller_id_name=Wireless/Unknown"/> > > > > > > If I crafted my regex properly, then it should be matching on the first > part of the string and setting the variable appropriately. Is > ?effective_caller_id_name? the variable I should be setting? > > Many thanks, > > Phil Quesinberry > > Q Systems Engineering, Inc. > > Electronic Controls and Embedded Systems Development > > (410) 969-8002 > > http://www.qsystemsengineering.com > > !DSPAM:5011b86432763191115303! > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org FreeSWITCH-users at lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > !DSPAM:5011b86432763191115303! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120727/6cbc2ffd/attachment-0001.html From gerald.weber at besharp.at Fri Jul 27 14:23:25 2012 From: gerald.weber at besharp.at (Gerald Weber) Date: Fri, 27 Jul 2012 10:23:25 +0000 Subject: [Freeswitch-users] HTTP Request from within the core In-Reply-To: <50123B07.6080604@quentustech.com> References: <50123B07.6080604@quentustech.com> Message-ID: Wow,cool idea to test the leds this way ! But my goal is a different: I need a way to tell a snom phone to pickup or hangup a call using a (agent) webinterface. With PhonerLite this is easy, just send uuid_phone_event uuid talk and you are done, but snom doesn't accept a NOTIFY talk event. They offer a way to simulate a keypress using http requests (e.g. "/command.htm?key=ENTER" ). My idea is to patch this functionality into mod_snom. The other way: Let uuid_phone_event decide what to send by looking up the user_agent in the sip_registration table. (I dont like this one, because this means i have to patch every phone specific handling into the core. But it would be a nicer way for me to build my webinterface.) gw Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von William King Gesendet: Freitag, 27. Juli 2012 08:54 An: freeswitch-users at lists.freeswitch.org Betreff: Re: [Freeswitch-users] HTTP Request from within the core It sounds like you might be doing something similar to what mod_snom started with. I've worked heavily with phone automation and I would be curious which phones you're working with and your experiences with them. Here's an example I was working on to automate the testing of Polycom BLF lights and call scenarios: http://imgur.com/a/9JfcR William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 07/25/2012 06:47 AM, Gerald Weber wrote: Thats it, muchas gracias ! Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Brian Foster Gesendet: Mittwoch, 25. Juli 2012 15:10 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] HTTP Request from within the core sofia_contact might be what you are looking for. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 25, 2012 9:04 AM, "Gerald Weber" > wrote: Thanks, looks very easy to implement. Just on more question: I need to get the IP address of a registered user. E.g. user/2000 is a SNOM with ip 192.168.20.219 I guess a query against the registrations table ist the fastest way to do this ? Or is there any core api command ? Thanks Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Avi Marcus Gesendet: Mittwoch, 25. Juli 2012 14:26 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] HTTP Request from within the core You can use the API for curl or just check how the codebase for curl works: http://wiki.freeswitch.org/wiki/Mod_curl -- src/mod/applications/mod_curl/ Or look at http://wiki.freeswitch.org/wiki/Mod_http_cache -- src/mod/applications/mod_http_cache/ -Avi _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120727/aabb6f95/attachment.html From nathandownes at hotmail.com Fri Jul 27 15:16:47 2012 From: nathandownes at hotmail.com (Mr Nathan Downes) Date: Fri, 27 Jul 2012 21:16:47 +1000 Subject: [Freeswitch-users] multi-tenant setup - call goes to wrong place Message-ID: Hi, I have a multi tenant setup with fusionpbx, for one domain I use extensions that don't register just for voicemail i.e 201 at domain1.com , when someone from xx at domain1.com calls 201 everything appears to go correctly, domain is set and call goes to 201 at domain1.com but when it first becomes an ip it sends to 201 at domain2.ip.address. It doesn't seem to happen with all extensions numbered the same in different domains.. Any idea what to look for?? Log as follows. 2012-07-26 09:37:35.178096 [NOTICE] switch_channel.c:926 New Channel sofia/internal/306 at domain1.com [b62db898-d6b1-11e1-be15-73f389528c7f] 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/306 at domain1.com) Running State Change CS_NEW 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:403 (sofia/internal/306 at domain1.com) State NEW 2012-07-26 09:37:35.178096 [DEBUG] sofia.c:5838 Channel sofia/internal/306 at domain1.com entering state [received][100] 2012-07-26 09:37:35.178096 [DEBUG] sofia.c:5849 Remote SDP: v=0 o=- 60255882 60255882 IN IP4 115.64.93.203 s=- c=IN IP4 115.64.93.203 t=0 0 m=audio 55938 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:3941 Looking for zrtp-hash 2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:3919 Deciding whether to pass zrtp-hash between legs 2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:3921 CF_ZRTP_PASSTHRU_REQ not set, so not propagating zrtp-hash 2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:5034 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] 2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:3020 Set Codec sofia/internal/306 at domain1.com PCMA/8000 20 ms 160 samples 64000 bits 2012-07-26 09:37:35.178096 [DEBUG] switch_core_codec.c:111 sofia/internal/306 at domain1.com Original read codec set to PCMA:8 2012-07-26 09:37:35.178096 [DEBUG] sofia_glue.c:5155 Set 2833 dtmf send/recv payload to 101 2012-07-26 09:37:35.178096 [DEBUG] sofia.c:6077 (sofia/internal/306 at domain1.com) State Change CS_NEW -> CS_INIT 2012-07-26 09:37:35.178096 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/306 at domain1.com [BREAK] 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/306 at domain1.com) Running State Change CS_INIT 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:424 (sofia/internal/306 at domain1.com) State INIT 2012-07-26 09:37:35.178096 [DEBUG] mod_sofia.c:85 sofia/internal/306 at domain1.com SOFIA INIT 2012-07-26 09:37:35.178096 [DEBUG] mod_sofia.c:125 (sofia/internal/306 at domain1.com) State Change CS_INIT -> CS_ROUTING 2012-07-26 09:37:35.178096 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/306 at domain1.com [BREAK] 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:424 (sofia/internal/306 at domain1.com) State INIT going to sleep 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/306 at domain1.com) Running State Change CS_ROUTING 2012-07-26 09:37:35.178096 [DEBUG] switch_channel.c:1919 (sofia/internal/306 at domain1.com) Callstate Change DOWN -> RINGING 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/306 at domain1.com) State ROUTING 2012-07-26 09:37:35.178096 [DEBUG] mod_sofia.c:148 sofia/internal/306 at domain1.com SOFIA ROUTING 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:104 sofia/internal/306 at domain1.com Standard ROUTING 2012-07-26 09:37:35.178096 [INFO] mod_dialplan_xml.c:485 Processing 0286249343 <306>->201 in context domain1.com Dialplan: sofia/internal/306 at domain1.com parsing [domain1.com->unloop] continue=false Dialplan: sofia/internal/306 at domain1.com parsing [domain1.com->Local_Extension] continue=false Dialplan: sofia/internal/306 at domain1.com Regex (PASS) [Local_Extension] destination_number(201) =~ /(^\d{2,7}$)/ break=on-false Dialplan: sofia/internal/306 at domain1.com Action set(dialed_extension=201) Dialplan: sofia/internal/306 at domain1.com Action export(dialed_extension=201) Dialplan: sofia/internal/306 at domain1.com Action limit(hash ${domain_name} 201 ${limit_max} ${limit_destination}) Dialplan: sofia/internal/306 at domain1.com Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/306 at domain1.com Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/archive/${strftime(%Y)}/${s trftime(%b)}/${strftime(%d)}/${uuid}.wav) Dialplan: sofia/internal/306 at domain1.com Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/306 at domain1.com Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: sofia/internal/306 at domain1.com Action set(ringback=${us-ring}) Dialplan: sofia/internal/306 at domain1.com Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/306 at domain1.com Action set(call_timeout=30) Dialplan: sofia/internal/306 at domain1.com Action set(hangup_after_bridge=true) Dialplan: sofia/internal/306 at domain1.com Action set(continue_on_fail=true) Dialplan: sofia/internal/306 at domain1.com Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_numbe r}) Dialplan: sofia/internal/306 at domain1.com Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/306 at domain1.com Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/306 at domain1.com Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/306 at domain1.com Action bridge(user/${user_data(${destination_number}@${domain_name} attr id)}@${domain_name}) Dialplan: sofia/internal/306 at domain1.com Action answer() Dialplan: sofia/internal/306 at domain1.com Action sleep(1000) Dialplan: sofia/internal/306 at domain1.com Action voicemail(default ${domain_name} ${dialed_extension}) 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/306 at domain1.com) State Change CS_ROUTING -> CS_EXECUTE 2012-07-26 09:37:35.178096 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/306 at domain1.com [BREAK] 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/306 at domain1.com) State ROUTING going to sleep 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/306 at domain1.com) Running State Change CS_EXECUTE 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/306 at domain1.com) State EXECUTE 2012-07-26 09:37:35.178096 [DEBUG] mod_sofia.c:241 sofia/internal/306 at domain1.com SOFIA EXECUTE 2012-07-26 09:37:35.178096 [DEBUG] switch_core_state_machine.c:196 sofia/internal/306 at domain1.com Standard EXECUTE EXECUTE sofia/internal/306 at domain1.com set(call_direction=local) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306 at domain1.com SET [call_direction]=[local] EXECUTE sofia/internal/306 at domain1.com set(open=true) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306 at domain1.com SET [open]=[true] EXECUTE sofia/internal/306 at domain1.com hash(insert/domain1.com-spymap/306/b62db898-d6b1-11e1-be15-73f389528c7f) EXECUTE sofia/internal/306 at domain1.com hash(insert/domain1.com-last_dial/306/201) EXECUTE sofia/internal/306 at domain1.com hash(insert/domain1.com-last_dial/global/b62db898-d6b1-11e1-be15-73f389528c7 f) EXECUTE sofia/internal/306 at domain1.com set(RFC2822_DATE=Thu, 26 Jul 2012 09:37:35 +1000) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306 at domain1.com SET [RFC2822_DATE]=[Thu, 26 Jul 2012 09:37:35 +1000] EXECUTE sofia/internal/306 at domain1.com set(dialed_extension=201) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306 at domain1.com SET [dialed_extension]=[201] EXECUTE sofia/internal/306 at domain1.com export(dialed_extension=201) 2012-07-26 09:37:35.178096 [DEBUG] switch_channel.c:1093 EXPORT (export_vars) [dialed_extension]=[201] EXECUTE sofia/internal/306 at domain1.com limit(hash domain1.com 201 2 !BUSY) 2012-07-26 09:37:35.178096 [INFO] switch_limit.c:126 incr called: domain1.com_201 max:2, interval:0 2012-07-26 09:37:35.178096 [INFO] mod_hash.c:202 Usage for domain1.com_201 is now 1/2 EXECUTE sofia/internal/306 at domain1.com bind_meta_app(1 b s execute_extension::dx XML features) 2012-07-26 09:37:35.178096 [INFO] switch_ivr_async.c:3328 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE sofia/internal/306 at domain1.com bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/archive/2012/Jul/26/b62db89 8-d6b1-11e1-be15-73f389528c7f.wav) 2012-07-26 09:37:35.178096 [INFO] switch_ivr_async.c:3328 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/archive/2012/Jul/26/b62db89 8-d6b1-11e1-be15-73f389528c7f.wav EXECUTE sofia/internal/306 at domain1.com bind_meta_app(3 b s execute_extension::cf XML features) 2012-07-26 09:37:35.178096 [INFO] switch_ivr_async.c:3328 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE sofia/internal/306 at domain1.com bind_meta_app(4 b s execute_extension::att_xfer XML features) 2012-07-26 09:37:35.178096 [INFO] switch_ivr_async.c:3328 Bound B-Leg: *4 execute_extension::att_xfer XML features EXECUTE sofia/internal/306 at domain1.com set(ringback=%(2000, 4000, 440.0, 480.0)) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306 at domain1.com SET [ringback]=[%(2000, 4000, 440.0, 480.0)] EXECUTE sofia/internal/306 at domain1.com set(transfer_ringback=local_stream://moh) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306 at domain1.com SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/306 at domain1.com set(call_timeout=30) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306 at domain1.com SET [call_timeout]=[30] EXECUTE sofia/internal/306 at domain1.com set(hangup_after_bridge=true) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306 at domain1.com SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/306 at domain1.com set(continue_on_fail=true) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306 at domain1.com SET [continue_on_fail]=[true] EXECUTE sofia/internal/306 at domain1.com hash(insert/domain1.com-call_return/201/306) EXECUTE sofia/internal/306 at domain1.com hash(insert/domain1.com-last_dial_ext/201/b62db898-d6b1-11e1-be15-73f389528c 7f) EXECUTE sofia/internal/306 at domain1.com set(called_party_callgroup=) 2012-07-26 09:37:35.178096 [DEBUG] mod_dptools.c:1305 sofia/internal/306 at domain1.com SET [called_party_callgroup]=[UNDEF] EXECUTE sofia/internal/306 at domain1.com hash(insert/domain1.com-last_dial//b62db898-d6b1-11e1-be15-73f389528c7f) EXECUTE sofia/internal/306 at domain1.com bridge(user/201 at domain1.com) 2012-07-26 09:37:35.178096 [DEBUG] switch_channel.c:1047 sofia/internal/306 at domain1.com EXPORTING[export_vars] [domain_name]=[domain1.com] to event 2012-07-26 09:37:35.178096 [DEBUG] switch_channel.c:1047 sofia/internal/306 at domain1.com EXPORTING[export_vars] [dialed_extension]=[201] to event 2012-07-26 09:37:35.178096 [DEBUG] switch_ivr_originate.c:1958 Parsing global variables 2012-07-26 09:37:35.198097 [DEBUG] switch_channel.c:1047 sofia/internal/306 at domain1.com EXPORTING[export_vars] [domain_name]=[domain1.com] to event 2012-07-26 09:37:35.198097 [DEBUG] switch_channel.c:1047 sofia/internal/306 at domain1.com EXPORTING[export_vars] [dialed_extension]=[201] to event 2012-07-26 09:37:35.198097 [DEBUG] switch_ivr_originate.c:1958 Parsing global variables 2012-07-26 09:37:35.198097 [DEBUG] switch_event.c:1470 Parsing variable [sip_invite_domain]=[domain1.com] 2012-07-26 09:37:35.198097 [DEBUG] switch_event.c:1470 Parsing variable [presence_id]=[201 at domain1.com] 2012-07-26 09:37:35.198097 [NOTICE] switch_channel.c:926 New Channel sofia/internal/sip:201 at 110.143.31.101:1178 [b630218c-d6b1-11e1-be1d-73f389528c7f] 2012-07-26 09:37:35.198097 [DEBUG] mod_sofia.c:4734 (sofia/internal/sip:201 at 110.143.31.101:1178) State Change CS_NEW -> CS_INIT 2012-07-26 09:37:35.198097 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/sip:201 at 110.143.31.101:1178 [BREAK] 2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:201 at 110.143.31.101:1178) Running State Change CS_INIT 2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:424 (sofia/internal/sip:201 at 110.143.31.101:1178) State INIT 2012-07-26 09:37:35.198097 [DEBUG] mod_sofia.c:85 sofia/internal/sip:201 at 110.143.31.101:1178 SOFIA INIT 2012-07-26 09:37:35.198097 [DEBUG] sofia_glue.c:2602 Local SDP: v=0 o=FreeSWITCH 1343234313 1343234314 IN IP4 203.174.163.226 s=FreeSWITCH c=IN IP4 203.174.163.226 t=0 0 m=audio 25142 RTP/AVP 8 0 3 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 a=sendrecv 2012-07-26 09:37:35.198097 [DEBUG] mod_sofia.c:125 (sofia/internal/sip:201 at 110.143.31.101:1178) State Change CS_INIT -> CS_ROUTING 2012-07-26 09:37:35.198097 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/sip:201 at 110.143.31.101:1178 [BREAK] 2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:424 (sofia/internal/sip:201 at 110.143.31.101:1178) State INIT going to sleep 2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:201 at 110.143.31.101:1178) Running State Change CS_ROUTING 2012-07-26 09:37:35.198097 [DEBUG] switch_channel.c:1919 (sofia/internal/sip:201 at 110.143.31.101:1178) Callstate Change DOWN -> RINGING 2012-07-26 09:37:35.198097 [DEBUG] switch_core_session.c:923 Send signal sofia/internal/sip:201 at 110.143.31.101:1178 [BREAK] 2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/sip:201 at 110.143.31.101:1178) State ROUTING 2012-07-26 09:37:35.198097 [DEBUG] mod_sofia.c:148 sofia/internal/sip:201 at 110.143.31.101:1178 SOFIA ROUTING 2012-07-26 09:37:35.198097 [DEBUG] switch_ivr_originate.c:67 (sofia/internal/sip:201 at 110.143.31.101:1178) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-07-26 09:37:35.198097 [DEBUG] switch_core_session.c:1228 Send signal sofia/internal/sip:201 at 110.143.31.101:1178 [BREAK] 2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:433 (sofia/internal/sip:201 at 110.143.31.101:1178) State ROUTING going to sleep 2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:385 (sofia/internal/sip:201 at 110.143.31.101:1178) Running State Change CS_CONSUME_MEDIA 2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:452 (sofia/internal/sip:201 at 110.143.31.101:1178) State CONSUME_MEDIA 2012-07-26 09:37:35.198097 [DEBUG] switch_core_state_machine.c:452 (sofia/internal/sip:201 at 110.143.31.101:1178) State CONSUME_MEDIA going to sleep 2012-07-26 09:37:35.198097 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:201 at 110.143.31.101:1178 entering state [calling][0] 2012-07-26 09:37:35.318096 [DEBUG] switch_core_session.c:923 Send signal sofia/internal/sip:201 at 110.143.31.101:1178 [BREAK] 2012-07-26 09:37:35.318096 [DEBUG] switch_core_session.c:923 Send signal sofia/internal/sip:201 at 110.143.31.101:1178 [BREAK] 2012-07-26 09:37:35.338097 [DEBUG] sofia.c:5838 Channel sofia/internal/sip:201 at 110.143.31.101:1178 entering state [proceeding][180] 2012-07-26 09:37:35.338097 [NOTICE] sofia.c:5930 Ring-Ready sofia/internal/sip:201 at 110.143.31.101:1178! 2012-07-26 09:37:35.338097 [INFO] switch_ivr_originate.c:1156 Sending early media 2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3269 AUDIO RTP [sofia/internal/306 at domain1.com] 203.174.163.226 port 26698 -> 115.64.93.203 port 55938 codec: 8 ms: 20 2012-07-26 09:37:35.338097 [DEBUG] switch_rtp.c:1680 Starting timer [soft] 160 bytes per 20ms 2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3491 Setting Jitterbuffer to 60ms (3 frames) 2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3533 Set 2833 dtmf send payload to 101 2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3539 Set 2833 dtmf receive payload to 101 2012-07-26 09:37:35.338097 [DEBUG] sofia_glue.c:3566 sofia/internal/306 at domain1.com Set rtp dtmf delay to 40 2012-07-26 09:37:35.338097 [DEBUG] mod_sofia.c:2606 Ring SDP: v=0 o=FreeSWITCH 1343232757 1343232758 IN IP4 203.174.163.226 s=FreeSWITCH c=IN IP4 203.174.163.226 t=0 0 m=audio 26698 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv 2012-07-26 09:37:35.338097 [NOTICE] mod_sofia.c:2609 Pre-Answer sofia/internal/306 at domain1.com! 2012-07-26 09:37:35.338097 [DEBUG] switch_channel.c:3042 (sofia/internal/306 at domain1.com) Callstate Change RINGING -> EARLY 2012-07-26 09:37:35.338097 [DEBUG] switch_core_session.c:777 Send signal sofia/internal/306 at domain1.com [BREAK] -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120727/c378da6f/attachment-0001.html From alex at thewinelake.com Fri Jul 27 16:01:39 2012 From: alex at thewinelake.com (Alex) Date: Fri, 27 Jul 2012 13:01:39 +0100 Subject: [Freeswitch-users] PAGD 0 digits In-Reply-To: <50110823.2060604@thewinelake.com> References: <9D0E55F4-768E-4C9D-B1DF-9D074E44ADD7@mgtech.com> <458A0010-EC8C-4ACC-B08D-A02E0AD26077@mgtech.com> <500FA9BA.5030303@thewinelake.com> <50110823.2060604@thewinelake.com> Message-ID: <50128323.7000008@thewinelake.com> If I want to allow someone to enter an empty string of digits, how is it done? I've tried various things... My latest effort (that still doesn't work!) is keypress = session:playAndGetDigits(1, 1, 3, 5000, "*#", what_to_play, "", "\\d{0,1}|\\#|\\*") terminator = session:getVariable("read_terminator_used") I note that the minimum value for the first param is 1 - so maybe it can't be done? From g.d.monnezza at tiscali.it Fri Jul 27 15:49:13 2012 From: g.d.monnezza at tiscali.it (g) Date: Fri, 27 Jul 2012 13:49:13 +0200 Subject: [Freeswitch-users] AutoNAT - Local Networks not excluded In-Reply-To: References: <34201844.post@talk.nabble.com> Message-ID: <3724094.f3e3ozOioZ@virtex> Thanks for suggestion, Prasd. Unluckely, my phones (Grandstream GXP1105) ave not NAT on/off choice. I can only manually set the IP of the NAT (public IP of the phone's gateway) and is currently empty. So I consider no NAT is enabled. The problem, in my opinion, is not on the phones. They present themselves correctly to FS server (them aren't NATted). Is, indeed, FS server presenting itself with wrong IP (the external one) to the phones, believing them are outside the LAN just because them have IP subnet address different from the subnet of the server. The question is how to tell FS some neworks are "internal" and let local IP is used like in the case of its own network. I'll try again google'ing for answer ... g On Thursday 26 July 2012 14:04:03 you wrote: > Did you try disabling NAT and STUN on your local / LAN SIP phones ? > > On 7/24/12, VirteX wrote: > > Hi guys. I appreciate so much the Auto-NAT for uPnP capable firewalls. > > But I'm experiencing an issue. > > I have a FreeSwitch server behind a NAT, but I can't find a way to avoid > > FreeSwitch using external IP (for SIP and RTP) for local networks (i.e. > > 192.168.0.0/16). > > In my sip profiles for various interfaces I have NOT set the . > > Anyway, the sofia status for all interfaces shows the EXT-RTP-IP and > > EXT-SIP-IP set (with my public gateway IP). That's ok, even if I didn' > > declard it with > > My SIP phones register from a network different from the server one, but > > still a local network. Then, SIP phones receive (from the server) the > > rtp > > and sip signalling with its external IP. This prevent any communication. > > How it is possible to tell FreeSwitch to NOT use ext IP for particular > > networks? > > Thanks to anyone who will point me in the right direction. > > g > > -- > > View this message in context: > > http://old.nabble.com/AutoNAT---Local-Networks-not-excluded-tp34201844p3 > > 4201844.html Sent from the Freeswitch-users mailing list archive at > > Nabble.com. > > > > > > ________________________________________________________________________ > > _ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org From g.d.monnezza at tiscali.it Fri Jul 27 16:30:30 2012 From: g.d.monnezza at tiscali.it (g) Date: Fri, 27 Jul 2012 14:30:30 +0200 Subject: [Freeswitch-users] AutoNAT - Local Networks not excluded In-Reply-To: <34201844.post@talk.nabble.com> References: <34201844.post@talk.nabble.com> Message-ID: <2234341.IrPRE0hISf@virtex> I found someone talking about similar problems. I read that "... some lines of code in sofia_reg.c if (is_nat && profile->local_network && switch_check_network_list_ip(network_ip, profile->local_network)) { if (profile->debug) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "IP %s is on local network, not seting NAT mode.\n", network_ip); } is_nat = NULL; } " So I think there is the possibility to set which are local networks. Also I found in my sip_profiles for nat-mode contain (as it should be) the ext- IP declaration: but not the line May be this line solve my problem. I'll try as soon as possible, but all my FS servers ara actually in production environments :( If someone has the chance to test it successfully, please report it. g On Tuesday 24 July 2012 05:12:08 VirteX wrote: > Hi guys. I appreciate so much the Auto-NAT for uPnP capable firewalls. But > I'm experiencing an issue. > I have a FreeSwitch server behind a NAT, but I can't find a way to avoid > FreeSwitch using external IP (for SIP and RTP) for local networks (i.e. > 192.168.0.0/16). > In my sip profiles for various interfaces I have NOT set the . > Anyway, the sofia status for all interfaces shows the EXT-RTP-IP and > EXT-SIP-IP set (with my public gateway IP). That's ok, even if I didn' > declard it with > My SIP phones register from a network different from the server one, but > still a local network. Then, SIP phones receive (from the server) the rtp > and sip signalling with its external IP. This prevent any communication. > How it is possible to tell FreeSwitch to NOT use ext IP for particular > networks? > Thanks to anyone who will point me in the right direction. > g From philq at qsystemsengineering.com Fri Jul 27 18:23:49 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Fri, 27 Jul 2012 10:23:49 -0400 Subject: [Freeswitch-users] AutoNAT - Local Networks not excluded Message-ID: <010201cd6c03$73b0a2f0$5b11e8d0$@com> G, Are you registering your phones to the internal sip profile? Do you have anything like aggressive NAT detection enabled for that profile? For the extensions, are you rewriting the contact IP/port (is NDLB-connectile-dysfuncion or NDLB-tls-connectile-dysfunction specified for sip-force-contact)? Do a 'show registrations' from the fs_cli as well as a 'sofia status profile internal reg' and post the results here (you may want to partially obscure any external IP addresses shown before posting) to give us more of an idea of what's going on. - Phil _____________________________________________ From: Phil Quesinberry [mailto:philq at qsystemsengineering.com] Sent: Tuesday, July 24, 2012 3:19 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: re: AutoNAT - Local Networks not excluded Set rtp-ip and sip-ip to your internal IP address. I believe that you should also be able to set it to: $${local_ip_v4} or $${bind_server_ip} as well. - Phil ---------- VirteX g.d.monnezza at tiscali.it Tue Jul 24 16:12:08 MSD 2012 Hi guys. I appreciate so much the Auto-NAT for uPnP capable firewalls. But I'm experiencing an issue. I have a FreeSwitch server behind a NAT, but I can't find a way to avoid FreeSwitch using external IP (for SIP and RTP) for local networks (i.e. 192.168.0.0/16). In my sip profiles for various interfaces I have NOT set the . Anyway, the sofia status for all interfaces shows the EXT-RTP-IP and EXT-SIP-IP set (with my public gateway IP). That's ok, even if I didn' declard it with My SIP phones register from a network different from the server one, but still a local network. Then, SIP phones receive (from the server) the rtp and sip signalling with its external IP. This prevent any communication. How it is possible to tell FreeSwitch to NOT use ext IP for particular networks? Thanks to anyone who will point me in the right direction. g -- View this message in context: http://old.nabble.com/AutoNAT---Local-Networks-not-excluded-tp34201844p34201 844.html Sent from the Freeswitch-users mailing list archive at Nabble.com. Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120727/44c992d7/attachment.html From philq at qsystemsengineering.com Fri Jul 27 18:31:19 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Fri, 27 Jul 2012 10:31:19 -0400 Subject: [Freeswitch-users] AutoNAT - Local Networks not excluded Message-ID: <010701cd6c04$7d5f29b0$781d7d10$@com> One other thing comes to mind. A lot of routers (especially SOHO routers) have ALG functionality that can break the SIP signaling, even when the ALG functionality is supposedly turned off. You can usually get around this by changing the SIP port to something other than 5060. If the phones connect via TLS (usually on port 5061) then this shouldn't be a problem, as they can't mess with the encrypted traffic. - Phil _____________________________________________ From: Phil Quesinberry [mailto:philq at qsystemsengineering.com] Sent: Friday, July 27, 2012 10:24 AM To: 'freeswitch-users at lists.freeswitch.org' Subject: RE: re: AutoNAT - Local Networks not excluded G, Are you registering your phones to the internal sip profile? Do you have anything like aggressive NAT detection enabled for that profile? For the extensions, are you rewriting the contact IP/port (is NDLB-connectile-dysfuncion or NDLB-tls-connectile-dysfunction specified for sip-force-contact)? Do a 'show registrations' from the fs_cli as well as a 'sofia status profile internal reg' and post the results here (you may want to partially obscure any external IP addresses shown before posting) to give us more of an idea of what's going on. - Phil _____________________________________________ From: Phil Quesinberry [mailto:philq at qsystemsengineering.com] Sent: Tuesday, July 24, 2012 3:19 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: re: AutoNAT - Local Networks not excluded Set rtp-ip and sip-ip to your internal IP address. I believe that you should also be able to set it to: $${local_ip_v4} or $${bind_server_ip} as well. - Phil ---------- VirteX g.d.monnezza at tiscali.it Tue Jul 24 16:12:08 MSD 2012 Hi guys. I appreciate so much the Auto-NAT for uPnP capable firewalls. But I'm experiencing an issue. I have a FreeSwitch server behind a NAT, but I can't find a way to avoid FreeSwitch using external IP (for SIP and RTP) for local networks (i.e. 192.168.0.0/16). In my sip profiles for various interfaces I have NOT set the . Anyway, the sofia status for all interfaces shows the EXT-RTP-IP and EXT-SIP-IP set (with my public gateway IP). That's ok, even if I didn' declard it with My SIP phones register from a network different from the server one, but still a local network. Then, SIP phones receive (from the server) the rtp and sip signalling with its external IP. This prevent any communication. How it is possible to tell FreeSwitch to NOT use ext IP for particular networks? Thanks to anyone who will point me in the right direction. g -- View this message in context: http://old.nabble.com/AutoNAT---Local-Networks-not-excluded-tp34201844p34201 844.html Sent from the Freeswitch-users mailing list archive at Nabble.com. Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120727/298e997d/attachment-0001.html From philq at qsystemsengineering.com Fri Jul 27 18:49:09 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Fri, 27 Jul 2012 10:49:09 -0400 Subject: [Freeswitch-users] AutoNAT - Local Networks not excluded Message-ID: <014201cd6c06$fdee2a20$f9ca7e60$@com> With that in mind - in my working configuration with phones both on the local LAN with FS as well as remote natted networks, I have: ./sip_profiles/internal.xml: (All extensions are registered to the internal profile) In most cases, it was necessary to have FS rewrite the contact IP and port for remote extensions. - Phil I found someone talking about similar problems. I read that "... some lines of code in sofia_reg.c if (is_nat && profile->local_network && switch_check_network_list_ip(network_ip, profile->local_network)) { if (profile->debug) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "IP %s is on local network, not seting NAT mode.\n", network_ip); } is_nat = NULL; } " So I think there is the possibility to set which are local networks. Also I found in my sip_profiles for nat-mode contain (as it should be) the ext- IP declaration: but not the line May be this line solve my problem. I'll try as soon as possible, but all my FS servers ara actually in production environments :( If someone has the chance to test it successfully, please report it. g _____________________________________________ From: Phil Quesinberry Sent: Friday, July 27, 2012 10:31 AM To: 'freeswitch-users at lists.freeswitch.org' Subject: RE: re: AutoNAT - Local Networks not excluded One other thing comes to mind. A lot of routers (especially SOHO routers) have ALG functionality that can break the SIP signaling, even when the ALG functionality is supposedly turned off. You can usually get around this by changing the SIP port to something other than 5060. If the phones connect via TLS (usually on port 5061) then this shouldn't be a problem, as they can't mess with the encrypted traffic. - Phil _____________________________________________ From: Phil Quesinberry Sent: Friday, July 27, 2012 10:24 AM To: 'freeswitch-users at lists.freeswitch.org' Subject: RE: re: AutoNAT - Local Networks not excluded G, Are you registering your phones to the internal sip profile? Do you have anything like aggressive NAT detection enabled for that profile? For the extensions, are you rewriting the contact IP/port (is NDLB-connectile-dysfuncion or NDLB-tls-connectile-dysfunction specified for sip-force-contact)? Do a 'show registrations' from the fs_cli as well as a 'sofia status profile internal reg' and post the results here (you may want to partially obscure any external IP addresses shown before posting) to give us more of an idea of what's going on. - Phil _____________________________________________ From: Phil Quesinberry Sent: Tuesday, July 24, 2012 3:19 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: re: AutoNAT - Local Networks not excluded Set rtp-ip and sip-ip to your internal IP address. I believe that you should also be able to set it to: $${local_ip_v4} or $${bind_server_ip} as well. - Phil ---------- VirteX g.d.monnezza at tiscali.it Tue Jul 24 16:12:08 MSD 2012 Hi guys. I appreciate so much the Auto-NAT for uPnP capable firewalls. But I'm experiencing an issue. I have a FreeSwitch server behind a NAT, but I can't find a way to avoid FreeSwitch using external IP (for SIP and RTP) for local networks (i.e. 192.168.0.0/16). In my sip profiles for various interfaces I have NOT set the . Anyway, the sofia status for all interfaces shows the EXT-RTP-IP and EXT-SIP-IP set (with my public gateway IP). That's ok, even if I didn' declard it with My SIP phones register from a network different from the server one, but still a local network. Then, SIP phones receive (from the server) the rtp and sip signalling with its external IP. This prevent any communication. How it is possible to tell FreeSwitch to NOT use ext IP for particular networks? Thanks to anyone who will point me in the right direction. g -- View this message in context: http://old.nabble.com/AutoNAT---Local-Networks-not-excluded-tp34201844p34201 844.html Sent from the Freeswitch-users mailing list archive at Nabble.com. Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120727/9b37e051/attachment.html From krice at freeswitch.org Fri Jul 27 19:02:50 2012 From: krice at freeswitch.org (Ken Rice) Date: Fri, 27 Jul 2012 10:02:50 -0500 Subject: [Freeswitch-users] Setting effecting_caller_id_name In-Reply-To: Message-ID: This exmaple should be correct provided that caller_id_name containts ?Currently runnint a lookup? with this pinned to that start of the field... On that extension in your dialplan toss a If your regex isnt matching on the caller_id_name field the above will show you whats in there (along with every thing else too heh) K On 7/26/12 7:11 PM, "Brian Foster" wrote: >> ?? >> >> ????? >> >> ??????? > data="effective_caller_id_name=Wireless/Unknown"/> >> >> ????? >> >> ??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120727/b370f87a/attachment-0001.html From anthony.minessale at gmail.com Fri Jul 27 19:51:09 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Jul 2012 10:51:09 -0500 Subject: [Freeswitch-users] AutoNAT - Local Networks not excluded In-Reply-To: <014201cd6c06$fdee2a20$f9ca7e60$@com> References: <014201cd6c06$fdee2a20$f9ca7e60$@com> Message-ID: for more complex networks make your own acl in acl.conf.xml and use it in place of localnet.auto in your sofia profile config. On Fri, Jul 27, 2012 at 9:49 AM, Phil Quesinberry wrote: > With that in mind - in my working configuration with phones both on the > local LAN with FS as well as remote natted networks, I have: > > ./sip_profiles/internal.xml: value="nat.auto"/> (All extensions are registered to the internal > profile) > > In most cases, it was necessary to have FS rewrite the contact IP and port > for remote extensions. > > - Phil > > I found someone talking about similar problems. I read that > > "... some lines of code in sofia_reg.c > > if (is_nat && profile->local_network && > > switch_check_network_list_ip(network_ip, profile->local_network)) { > > if (profile->debug) { > > switch_log_printf(SWITCH_CHANNEL_LOG, > > SWITCH_LOG_DEBUG, "IP %s is on local network, not seting NAT mode.\n", > > network_ip); > > } > > is_nat = NULL; > > } > > " > > So I think there is the possibility to set which are local networks. > > Also I found in my sip_profiles for nat-mode contain (as it should be) the > ext- > > IP declaration: > > > > > > but not the line > > > > May be this line solve my problem. > > I'll try as soon as possible, but all my FS servers ara actually in > production > > environments :( > > If someone has the chance to test it successfully, please report it. > > g > > _____________________________________________ > From: Phil Quesinberry > Sent: Friday, July 27, 2012 10:31 AM > > To: 'freeswitch-users at lists.freeswitch.org' > Subject: RE: re: AutoNAT - Local Networks not excluded > > One other thing comes to mind. A lot of routers (especially SOHO routers) > have ALG functionality that can break the SIP signaling, even when the ALG > functionality is supposedly turned off. You can usually get around this by > changing the SIP port to something other than 5060. If the phones connect > via TLS (usually on port 5061) then this shouldn?t be a problem, as they > can?t mess with the encrypted traffic. > > - Phil > > _____________________________________________ > From: Phil Quesinberry > > Sent: Friday, July 27, 2012 10:24 AM > To: 'freeswitch-users at lists.freeswitch.org' > Subject: RE: re: AutoNAT - Local Networks not excluded > > G, > > Are you registering your phones to the internal sip profile? Do you have > anything like aggressive NAT detection enabled for that profile? For the > extensions, are you rewriting the contact IP/port (is > NDLB-connectile-dysfuncion or NDLB-tls-connectile-dysfunction specified for > sip-force-contact)? > > Do a ?show registrations? from the fs_cli as well as a ?sofia status profile > internal reg? and post the results here (you may want to partially obscure > any external IP addresses shown before posting) to give us more of an idea > of what?s going on. > > - Phil > > _____________________________________________ > From: Phil Quesinberry > > Sent: Tuesday, July 24, 2012 3:19 PM > To: 'freeswitch-users at lists.freeswitch.org' > Subject: re: AutoNAT - Local Networks not excluded > > Set rtp-ip and sip-ip to your internal IP address. I believe that you > should also be able to set it to: $${local_ip_v4} or $${bind_server_ip} > as well. > > - Phil > > ---------- > > VirteX g.d.monnezza at tiscali.it > Tue Jul 24 16:12:08 MSD 2012 > > Hi guys. I appreciate so much the Auto-NAT for uPnP capable firewalls. But > > I'm experiencing an issue. > > I have a FreeSwitch server behind a NAT, but I can't find a way to avoid > > FreeSwitch using external IP (for SIP and RTP) for local networks (i.e. > > 192.168.0.0/16). > > In my sip profiles for various interfaces I have NOT set the . > > Anyway, the sofia status for all interfaces shows the EXT-RTP-IP and > > EXT-SIP-IP set (with my public gateway IP). That's ok, even if I didn' > > declard it with > > My SIP phones register from a network different from the server one, but > > still a local network. Then, SIP phones receive (from the server) the rtp > > and sip signalling with its external IP. This prevent any communication. > > How it is possible to tell FreeSwitch to NOT use ext IP for particular > > networks? > > Thanks to anyone who will point me in the right direction. > > g > > -- > > View this message in context: > http://old.nabble.com/AutoNAT---Local-Networks-not-excluded-tp34201844p34201844.html > > Sent from the Freeswitch-users mailing list archive at Nabble.com. > > > Phil Quesinberry > > Q Systems Engineering, Inc. > > Electronic Controls and Embedded Systems Development > > (410) 969-8002 > > http://www.qsystemsengineering.com > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From yufei.tao at redembedded.com Fri Jul 27 19:17:13 2012 From: yufei.tao at redembedded.com (Yufei Tao) Date: Fri, 27 Jul 2012 16:17:13 +0100 Subject: [Freeswitch-users] H264 transcodin In-Reply-To: References: Message-ID: <5012B0F9.5010201@redembedded.com> Thanks everyone for the responses! If I understand it correctly, if I installed ffmpeg on itself separately from FS, I could write a module for FS, in which I just call the ffmpeg program by running a command line. This way would it be classified as "not combine them into a larger work", thus free from license incompatibility problem? Not sure if that'll work for real-time transcoding of x264 though? Thanks very much for you opinions! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ From share2010.cn at gmail.com Fri Jul 27 19:23:50 2012 From: share2010.cn at gmail.com (dolphincn) Date: Fri, 27 Jul 2012 08:23:50 -0700 (PDT) Subject: [Freeswitch-users] ssl and sips and srtp configure issue. Message-ID: <1343402630186-7581215.post@n2.nabble.com> there have some problem when sip client and sips client call each other. 1, auto hangup in one minute. 2, sip client call sips client ,sips client can not pickup the call sometimes, and sips client with srtp mode can not pick up the call or sip client only hear hold music. 3, sips client using srtp could not use irv navigation and bind digit action, like press number 0 will not transfer to operator, no response. used softs: freeswitch x64 1.2rc for windows. 3cxphone client for windows ssl configure: var.xml internal.xml other used default setting. sip client configured sip server is local ip address (local_ip_v4 like 100.100.100.20). sips client configured sip server is external ip address (fs.share.com). ca files: conf/ssl/agent.pem conf/ssl/cafile.pem client installed ca file is cafile.pem. firewall open 5060-5061 for tcp/udp, 16384-32768 for udp, other closed. ssl work well, call 9664 can hear is secured call. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ssl-and-sips-and-srtp-configure-issue-tp7581215.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/20120727/4156b500/attachment.html From share2010.cn at gmail.com Fri Jul 27 19:34:22 2012 From: share2010.cn at gmail.com (dolphincn) Date: Fri, 27 Jul 2012 08:34:22 -0700 (PDT) Subject: [Freeswitch-users] ssl and sips and srtp calling problem. In-Reply-To: <1343402630186-7581215.post@n2.nabble.com> References: <1343402630186-7581215.post@n2.nabble.com> Message-ID: <1343403262725-7581216.post@n2.nabble.com> internal.xml -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/ssl-and-sips-and-srtp-calling-problem-tp7581215p7581216.html Sent from the freeswitch-users mailing list archive at Nabble.com. From g.d.monnezza at tiscali.it Fri Jul 27 19:43:15 2012 From: g.d.monnezza at tiscali.it (g) Date: Fri, 27 Jul 2012 17:43:15 +0200 Subject: [Freeswitch-users] AutoNAT - Local Networks not excluded In-Reply-To: <014201cd6c06$fdee2a20$f9ca7e60$@com> References: <014201cd6c06$fdee2a20$f9ca7e60$@com> Message-ID: <2393191.1xfJt1jI3L@virtex> Many interesting suggestions on you reply. Thanks. I'm now more confused :S ... But I have multiple path to solution of my issue :) I think the most interesting is the I'm missing, but I can't try to add it now. Anyway, here below some details of my installation. With this setup, on FS eth0 interface I clearly see trafic from FS going out to phone with addresses: 192.168.50.250 (FS) -> 192.168.40.103 (phone) but no trafic coming bak from the phone. Dumping network interface of the network gateway I see RTP packets trying to go out from phone to the internet address XX.YY.ZZ.KK: 192.168.40.103 (phone) -> XX.YY.ZZ.KK (Public IP of the gateway) So, is clear that FS presents itself to the phone with external IP, and the phone replies to that IP :( If the phone lives in same network as FS (i.e. 192.168.50.0/24) everything works. So FS has a "brain" determining what is to NAT and what is not. ____ Configuration details Interface profile settings: Interface details: sofia status profile sipinterface_1 ================================================================================================= Name sipinterface_1 Domain Name N/A Auto-NAT true DBName sofia_reg_sipinterface_1 Pres Hosts Dialplan XML Context multitenant_routing_context Challenge Realm auto_to RTP-IP 192.168.50.250 Ext-RTP-IP XX.YY.ZZ.KK SIP-IP 192.168.50.250 Ext-SIP-IP XX.YY.ZZ.KK URL sip:mod_sofia at 192.168.50.250:5060 BIND-URL sip:mod_sofia at 192.168.50.250:5060 HOLD-MUSIC N/A OUTBOUND-PROXY N/A CODECS IN GSM,PCMU,PCMA CODECS OUT GSM,PCMU,PCMA TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false ZRTP-PASSTHRU false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 REGISTRATIONS 12 Phone registration details: Call-ID: 1180386363-5060-1 at BJC.BGI.EA.BAD User: 101 at 192.168.200.250 Contact: "user" Agent: Grandstream GXP1105 1.0.4.9 Status: Registered(UDP)(unknown) EXP(2012-07-27 18:09:08) EXPSECS(3476) Host: microsrv IP: 192.168.40.103 Port: 5060 Auth-User: 101 Auth-Realm: 192.168.50.250 MWI-Account: 101 at voicemail_1 On Friday 27 July 2012 10:49:09 Phil Quesinberry wrote: > With that in mind - in my working configuration with phones both on the > local LAN with FS as well as remote natted networks, I have: > ./sip_profiles/internal.xml: value="nat.auto"/> (All extensions are registered to the internal > profile) > > In most cases, it was necessary to have FS rewrite the contact IP and port > for remote extensions. > > - Phil > > > I found someone talking about similar problems. I read that > "... some lines of code in sofia_reg.c > > if (is_nat && profile->local_network && > switch_check_network_list_ip(network_ip, profile->local_network)) { > if (profile->debug) { > switch_log_printf(SWITCH_CHANNEL_LOG, > SWITCH_LOG_DEBUG, "IP %s is on local network, not seting NAT mode.\n", > network_ip); > } > is_nat = NULL; > } > " > > So I think there is the possibility to set which are local networks. > > Also I found in my sip_profiles for nat-mode contain (as it should be) the > ext- > IP declaration: > > > but not the line > > May be this line solve my problem. > I'll try as soon as possible, but all my FS servers ara actually in > production > environments :( > If someone has the chance to test it successfully, please report it. > g > > _____________________________________________ > From: Phil Quesinberry > Sent: Friday, July 27, 2012 10:31 AM > To: 'freeswitch-users at lists.freeswitch.org' > Subject: RE: re: AutoNAT - Local Networks not excluded > > > One other thing comes to mind. A lot of routers (especially SOHO routers) > have ALG functionality that can break the SIP signaling, even when the ALG > functionality is supposedly turned off. You can usually get around this by > changing the SIP port to something other than 5060. If the phones connect > via TLS (usually on port 5061) then this shouldn't be a problem, as they > can't mess with the encrypted traffic. > > - Phil > _____________________________________________ > From: Phil Quesinberry > Sent: Friday, July 27, 2012 10:24 AM > To: 'freeswitch-users at lists.freeswitch.org' > Subject: RE: re: AutoNAT - Local Networks not excluded > > > G, > > Are you registering your phones to the internal sip profile? Do you have > anything like aggressive NAT detection enabled for that profile? For the > extensions, are you rewriting the contact IP/port (is > NDLB-connectile-dysfuncion or NDLB-tls-connectile-dysfunction specified for > sip-force-contact)? > > Do a 'show registrations' from the fs_cli as well as a 'sofia status profile > internal reg' and post the results here (you may want to partially obscure > any external IP addresses shown before posting) to give us more of an idea > of what's going on. > > - Phil > _____________________________________________ > From: Phil Quesinberry > Sent: Tuesday, July 24, 2012 3:19 PM > To: 'freeswitch-users at lists.freeswitch.org' > Subject: re: AutoNAT - Local Networks not excluded > > > Set rtp-ip and sip-ip to your internal IP address. I believe that you > should also be able to set it to: $${local_ip_v4} or $${bind_server_ip} > as well. > > - Phil > > ---------- > VirteX g.d.monnezza at tiscali.it > 5D%20%20AutoNAT%20-%20Local%20Networks%20not%20excluded&In-Reply-To=> Tue > Jul 24 16:12:08 MSD 2012 > > > Hi guys. I appreciate so much the Auto-NAT for uPnP capable firewalls. But > I'm experiencing an issue. > I have a FreeSwitch server behind a NAT, but I can't find a way to avoid > FreeSwitch using external IP (for SIP and RTP) for local networks (i.e. > 192.168.0.0/16). > In my sip profiles for various interfaces I have NOT set the . > Anyway, the sofia status for all interfaces shows the EXT-RTP-IP and > EXT-SIP-IP set (with my public gateway IP). That's ok, even if I didn' > declard it with > My SIP phones register from a network different from the server one, but > still a local network. Then, SIP phones receive (from the server) the rtp > and sip signalling with its external IP. This prevent any communication. > How it is possible to tell FreeSwitch to NOT use ext IP for particular > networks? > Thanks to anyone who will point me in the right direction. > g From sdevoy at bizfocused.com Fri Jul 27 21:06:18 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Fri, 27 Jul 2012 13:06:18 -0400 Subject: [Freeswitch-users] SIP to make phone reboot or resync Message-ID: <07b501cd6c1a$238b3e90$6aa1bbb0$@bizfocused.com> HI All, Does anyone have an code that will cause Freeswitch to send a SIP message to a CISCO SPA5xx phone that will cause it to reboot or resync (aka re-provision)? I know about the URLs that cause the phone to do this, but they are NATed, so I need to use SIP to hit them. Assume I am programming language omni-lingual for this request! Thanks, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120727/eeb32315/attachment-0001.html From anthony.minessale at gmail.com Fri Jul 27 21:20:30 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Jul 2012 12:20:30 -0500 Subject: [Freeswitch-users] SIP to make phone reboot or resync In-Reply-To: <07b501cd6c1a$238b3e90$6aa1bbb0$@bizfocused.com> References: <07b501cd6c1a$238b3e90$6aa1bbb0$@bizfocused.com> Message-ID: sofia profile check_sync [call_id] // no call_id means all sofia profile flush_inbound_reg [call_id] [reboot] On Fri, Jul 27, 2012 at 12:06 PM, Sean Devoy wrote: > HI All, > > > > Does anyone have an code that will cause Freeswitch to send a SIP message to > a CISCO SPA5xx phone that will cause it to reboot or resync (aka > re-provision)? I know about the URLs that cause the phone to do this, but > they are NATed, so I need to use SIP to hit them. > > > > Assume I am programming language omni-lingual for this request! > > > > Thanks, > > Sean > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From ajohnston at blimessaging.com Fri Jul 27 23:34:43 2012 From: ajohnston at blimessaging.com (Adam Johnston) Date: Fri, 27 Jul 2012 15:34:43 -0400 Subject: [Freeswitch-users] FreeSWITCH dying with no core file Message-ID: Hi all, I use FreeSWITCH exclusively for high-volume fax, both sending and receiving, and I've been running into a sporadic issue where the FreeSWITCH process will die with nothing in the logs to indicate an issue and no core file anywhere on the system. I'm setting "ulimit -c unlimited" in my init script, so I would expect to see a core file. Version: FreeSWITCH Version 1.2.0-rc2+git~20120713T162602Z~0afd7318bd+unclean~20120713T182401Z is where I've seen this occur most frequently. I would update to latest git and retry, but as I said, this issue is sporadic and by the time it happens again I'll have yet *another* git to update to. Right now I'm just trying to collect more information, as all I can tell is that crashes occur when there are dozens-to-hundreds of sending faxes, which is almost all the time during business hours. I was running a version from February, 2012 for months and months and I can't ever remember seeing this happen. Is there a way of obtaining more crash information I'm overlooking? Is my only option running gcore and crossing my fingers? Many thanks, Adam Johnston -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120727/f72a84d6/attachment.html From admin at blindi.net Fri Jul 27 23:55:58 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Fri, 27 Jul 2012 21:55:58 +0200 (CEST) Subject: [Freeswitch-users] problem: fs block public extension for inbound calling In-Reply-To: References: Message-ID: Hi all, i have setup a voicechat. The extension name is: dorf. my Domain: telco01.blindi.net I like to call: dorf at telco01.blindi.net from outside for all unregistered users. then i call this extension, Fs bring the errormessage: 2012-07-27 21:46:09.295221 [DEBUG] sofia.c:8069 IP 188.193.24.39 Rejected by acl "domains". Falling back to Digest auth. 2012-07-27 21:46:09.295221 [WARNING] sofia_reg.c:1474 SIP auth challenge (INVITE ) on sofia profile 'internal' for [dorf at telco01.blindi.net] from ip 188.193.24.3 9 2012-07-27 21:46:09.329205 [DEBUG] sofia.c:8069 IP 188.193.24.39 Rejected by acl "domains". Falling back to Digest auth. 2012-07-27 21:46:09.329205 [WARNING] sofia_reg.c:2442 Can't find user [699 at telco01.blindi.net] You must define a domain called 'telco01.blindi.net' in your directory and add a user with the id="699" attribute and you must configure your device to use the proper domain in it's authenticati on credentials. 2012-07-27 21:46:09.329205 [WARNING] sofia_reg.c:1419 SIP auth failure (INVITE) on sofia profile 'internal' for [dorf at telco01.blindi.net] from ip 188.193.24.39 Fail2ban block my Ip. How can i setup these extension for public inbound correctly please? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From stkn at freeswitch.org Sat Jul 28 00:59:24 2012 From: stkn at freeswitch.org (Stefan Knoblich) Date: Fri, 27 Jul 2012 22:59:24 +0200 Subject: [Freeswitch-users] FreeSWITCH dying with no core file In-Reply-To: References: Message-ID: <5013012C.9050003@freeswitch.org> -----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 On 07/27/12 21:34, Adam Johnston wrote: > I use FreeSWITCH exclusively for high-volume fax, both sending and receiving, and I've been running into a sporadic issue where the FreeSWITCH process will die with nothing in the logs to > indicate an issue and no core file anywhere on the system. I'm setting "ulimit -c unlimited" in my init script, so I would expect to see a core file. Version: FreeSWITCH Version > 1.2.0-rc2+git~20120713T162602Z~0afd7318bd+unclean~20120713T182401Z You need to set /proc/sys/fs/suid_dumpable to a non-zero value (see man 5 proc), to enable core dumping for processes that switch their UID on startup (such as freeswitch with -u option). By default, the freeswitch process will try to put the core file into the CWD (current work directory), so make sure the UID it is running as can write there (or "cd" into a different directory before starting it). You might also want to check out man 5 core, for all the other options (e.g. piping core dumps into a custom handler, with core_pattern). -----BEGIN PGP SIGNATURE----- Version: GnuPG v2.0.19 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ iEYEARECAAYFAlATASwACgkQjiIIAK4rYUrjvQCgh1grtcBfPVhbZNw8tdQpsyda 614Amwb1pKDFL93dU33cy2wj1FNYYnzJ =0Xqk -----END PGP SIGNATURE----- From anthony.minessale at gmail.com Sat Jul 28 01:45:07 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Jul 2012 16:45:07 -0500 Subject: [Freeswitch-users] HTTP Request from within the core In-Reply-To: References: <50123B07.6080604@quentustech.com> Message-ID: Ya and seven is working on that a bit. If we build some momentum we can get it working. On Jul 27, 2012 5:24 AM, "Gerald Weber" wrote: > Wow,cool idea to test the leds this way !**** > > ** ** > > But my goal is a different:**** > > I need a way to tell a snom phone to pickup or hangup a call using a > (agent) webinterface.**** > > With PhonerLite this is easy, just send uuid_phone_event uuid talk and you > are done, but snom doesn?t accept a NOTIFY talk event.**** > > They offer a way to simulate a keypress using http requests (e.g. ?/command.htm?key=ENTER? ). My idea is to patch this functionality into mod_snom.**** > > ** ** > > The other way:**** > > Let uuid_phone_event decide what to send by looking up the user_agent in the sip_registration table.**** > > (I dont like this one, because this means i have to patch every phone specific handling into the core. But it would be a nicer way for me to build my webinterface.)**** > > ** ** > > gw**** > > ** ** > > ** ** > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *William > King > *Gesendet:* Freitag, 27. Juli 2012 08:54 > *An:* freeswitch-users at lists.freeswitch.org > *Betreff:* Re: [Freeswitch-users] HTTP Request from within the core**** > > ** ** > > It sounds like you might be doing something similar to what mod_snom > started with. I've worked heavily with phone automation and I would be > curious which phones you're working with and your experiences with them. > > Here's an example I was working on to automate the testing of Polycom BLF > lights and call scenarios: http://imgur.com/a/9JfcR > > > **** > > William King**** > > Senior Engineer**** > > Quentus Technologies, INC**** > > 1037 NE 65th St Suite 273**** > > Seattle, WA 98115**** > > Main: (877) 211-9337**** > > Office: (206) 388-4772**** > > Cell: (253) 686-5518**** > > william.king at quentustech.com **** > > > On 07/25/2012 06:47 AM, Gerald Weber wrote: **** > > Thats it, muchas gracias !**** > > **** > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [ > mailto:freeswitch-users-bounces at lists.freeswitch.org] > *Im Auftrag von *Brian Foster > *Gesendet:* Mittwoch, 25. Juli 2012 15:10 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] HTTP Request from within the core**** > > **** > > sofia_contact might be what you are looking for. **** > > Brian Foster > Endigo Computer LLC**** > > Sent from a mobile device.**** > > On Jul 25, 2012 9:04 AM, "Gerald Weber" wrote:** > ** > > Thanks, looks very easy to implement.**** > > **** > > Just on more question:**** > > I need to get the IP address of a registered user.**** > > E.g. user/2000 is a SNOM with ip 192.168.20.219**** > > I guess a query against the registrations table ist the fastest way to do > this ?**** > > Or is there any core api command ? **** > > **** > > Thanks**** > > **** > > **** > > *Von:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *Im Auftrag von *Avi Marcus > *Gesendet:* Mittwoch, 25. Juli 2012 14:26 > *An:* FreeSWITCH Users Help > *Betreff:* Re: [Freeswitch-users] HTTP Request from within the core**** > > **** > > You can use the API for curl or just check how the codebase for curl works: > **** > > http://wiki.freeswitch.org/wiki/Mod_curl -- src/mod/applications/mod_curl/ > **** > > Or look at http://wiki.freeswitch.org/wiki/Mod_http_cache > -- src/mod/applications/mod_http_cache/**** > > **** > > -Avi**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > > > > **** > > _________________________________________________________________________**** > > Professional FreeSWITCH Consulting Services:**** > > consulting at freeswitch.org**** > > http://www.freeswitchsolutions.com**** > > ** ** > > **** > > **** > > ** ** > > Official FreeSWITCH Sites**** > > http://www.freeswitch.org**** > > http://wiki.freeswitch.org**** > > http://www.cluecon.com**** > > ** ** > > Join Us At ClueCon - Aug 7-9, 2012**** > > ** ** > > FreeSWITCH-users mailing list**** > > FreeSWITCH-users at lists.freeswitch.org**** > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users**** > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users**** > > http://www.freeswitch.org**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120727/351030fa/attachment-0001.html From anthony.minessale at gmail.com Sat Jul 28 01:48:48 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Jul 2012 16:48:48 -0500 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: References: <20120726184924.3e844d2d@mail.tritonwest.net> <2A416971-0F73-41EC-AAC2-5B11AAE5CB76@opencsta.org> Message-ID: On Jul 26, 2012 7:40 PM, "Ken Rice" wrote: > > most of freeswitch has been developedin an ssh terminal using screen and emacs... Yep, and it still is every day! > > Ken > Sent from my iPad > > On Jul 26, 2012, at 7:01 PM, Chris Mylonas wrote: > > >> I tend to do most of my dev on ssh terms > > > > > > v.interesting. > > how the hell do you manage that? > > > > emacs/vim? > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120727/808554ab/attachment.html From curriegrad2004 at gmail.com Sat Jul 28 01:54:42 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 27 Jul 2012 14:54:42 -0700 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: References: <20120726184924.3e844d2d@mail.tritonwest.net> <2A416971-0F73-41EC-AAC2-5B11AAE5CB76@opencsta.org> Message-ID: Then again, some parts were fixed in MSVC 2010 :P On Fri, Jul 27, 2012 at 2:48 PM, Anthony Minessale wrote: > > On Jul 26, 2012 7:40 PM, "Ken Rice" wrote: >> >> most of freeswitch has been developedin an ssh terminal using screen and >> emacs... > > Yep, and it still is every day! > > >> >> Ken >> Sent from my iPad >> >> On Jul 26, 2012, at 7:01 PM, Chris Mylonas wrote: >> >> >> I tend to do most of my dev on ssh terms >> > >> > >> > v.interesting. >> > how the hell do you manage that? >> > >> > emacs/vim? >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From sdevoy at bizfocused.com Sat Jul 28 02:03:13 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Fri, 27 Jul 2012 18:03:13 -0400 Subject: [Freeswitch-users] SIP to make phone reboot or resync In-Reply-To: References: <07b501cd6c1a$238b3e90$6aa1bbb0$@bizfocused.com> Message-ID: <094e01cd6c43$9e44cc90$dace65b0$@bizfocused.com> Thank Anthony, BUT.. Error: 2012-07-27 17:58:55.118642 [ERR] sofia_reg.c:2165 Cannot locate any authentication credentials to complete an authentication request for realm '"fs_bfis.bizfocused.com"' 1 - where do I specify the credentials? 2 - Are these Freeswitch credentials or phone credentials? I suspect the latter. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Friday, July 27, 2012 1:21 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync sofia profile check_sync [call_id] // no call_id means all sofia profile flush_inbound_reg [call_id] [reboot] On Fri, Jul 27, 2012 at 12:06 PM, Sean Devoy wrote: > HI All, > > > > Does anyone have an code that will cause Freeswitch to send a SIP > message to a CISCO SPA5xx phone that will cause it to reboot or resync > (aka re-provision)? I know about the URLs that cause the phone to do > this, but they are NATed, so I need to use SIP to hit them. > > > > Assume I am programming language omni-lingual for this request! > > > > Thanks, > > Sean > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.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 Sat Jul 28 02:09:45 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Jul 2012 17:09:45 -0500 Subject: [Freeswitch-users] No reINVITE when bridging two sessions from mod_managed with bypass_media_after_bridge=true In-Reply-To: References: Message-ID: Do you exit the app after you call uuid bridge? Your code is running in the session thread so you need to exit so it can change states. On Jul 20, 2012 3:32 PM, "Srini K" wrote: > Hi, > Iam trying to bypass media from FS after two call legs are bridged using > mod_managed. > The code looks like > public void Run(AppContext context) > { > var fsApi = new FreeSWITCH.Native.Api(); > var aLegSession = context.Session; > // Answer the incoming call > aLegSession.Answer(); > // Play the prompt > aLegSession.StreamFile("ivr/ThankYou.wav", 0); > // Create outBound session > var bLegSession = new > ManagedSession("sofia/gateway/95/4151230000"); > > // Bypass Media > aLegSession.SetVariable("bypass_media_after_bridge", "true"); > bLegSession.SetVariable("bypass_media_after_bridge", "true"); > fsApi.ExecuteString(string.Format("uuid_bridge {0} {1}", > aLegSession.GetUuid(), bLegSession.GetUuid())); > } > > I don't see FreeSWITCH sending re-Invite after the call is bridged. > What I've already tried and did not succeed: > 1) set bypass_media=true, on A leg only, on B leg only, on both legs > 2) set bypass_media_after_bridge=true, on A leg only, on B leg only, on > both legs > > When I tried without using mod_managed using only dialplan, FS sends > re-Invite. > > > > > > > Whether Iam doing anything stupid in mod_managed? > > Regards > Srini > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120727/428739af/attachment.html From anthony.minessale at gmail.com Sat Jul 28 02:11:53 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Jul 2012 17:11:53 -0500 Subject: [Freeswitch-users] SIP to make phone reboot or resync In-Reply-To: <094e01cd6c43$9e44cc90$dace65b0$@bizfocused.com> References: <07b501cd6c1a$238b3e90$6aa1bbb0$@bizfocused.com> <094e01cd6c43$9e44cc90$dace65b0$@bizfocused.com> Message-ID: Make a gateway with no reg and the credentials and name it after the realm in the challenge. On Jul 27, 2012 5:05 PM, "Sean Devoy" wrote: > Thank Anthony, BUT.. Error: > 2012-07-27 17:58:55.118642 [ERR] sofia_reg.c:2165 Cannot locate any > authentication credentials to complete an authentication request for realm > '"fs_bfis.bizfocused.com"' > > 1 - where do I specify the credentials? > 2 - Are these Freeswitch credentials or phone credentials? I suspect the > latter. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony > Minessale > Sent: Friday, July 27, 2012 1:21 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync > > sofia profile check_sync [call_id] // no call_id means all > sofia profile flush_inbound_reg [call_id] [reboot] > > > > On Fri, Jul 27, 2012 at 12:06 PM, Sean Devoy > wrote: > > HI All, > > > > > > > > Does anyone have an code that will cause Freeswitch to send a SIP > > message to a CISCO SPA5xx phone that will cause it to reboot or resync > > (aka re-provision)? I know about the URLs that cause the phone to do > > this, but they are NATed, so I need to use SIP to hit them. > > > > > > > > Assume I am programming language omni-lingual for this request! > > > > > > > > Thanks, > > > > Sean > > > > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users 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 > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120727/82023233/attachment-0001.html From admin at blindi.net Sat Jul 28 02:33:22 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Sat, 28 Jul 2012 00:33:22 +0200 (CEST) Subject: [Freeswitch-users] The voice is hacked after connecting from internal extension In-Reply-To: References: Message-ID: Hi Guy, the problem is: for example: i have e ivr extension 1020 my outboundextension is: 1000 i dial from 1000 to 1001 the firstprompt is hacked "welcome to ivr". the secondprompt playd correctly. I have the last git. The System is a core i7, 24mb ram. debian squeeze amd64. I think this is a timing problem. I don.t no. the voice play very slowly, clearly, and hacked. Can you help please? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From lists.jj at googlemail.com Sat Jul 28 00:58:44 2012 From: lists.jj at googlemail.com (Johannes Jakob) Date: Fri, 27 Jul 2012 22:58:44 +0200 Subject: [Freeswitch-users] Problems with multi-FS ODBC cluster Message-ID: Hi, I'm operating a cluster of three FS boxes "behind" an OpenSIPs balancer/proxy that's not in the media path, but proxies everything SIP related in- and outbound. The central brain is a mysql master/slave setup and I'm using a combination of NFS and rsync to keep configs and voicemail synced. I want to balance registrations and calls to all three boxes, but because of the following issues, currently only one FS is active, the others on hot standby. The things that are still troubling me are: 1) Group intercept (*8) isn't working 2) Presence subscriptions aren't working 3) mod_callcenter is misbehaving badly - nonworking. I was thinking, that with everything moved to the database, the single boxes should be aware of callstates and uuids on the other boxes, like they know about which user is registered to which FS. Now I'm not sure if I've forgotten something, or if a multie-active setup would be a trade-off between redundancy and loss of features? All FreeSWITCHes are up-to-date, odbc-dsn has been set for all important modules: /usr/local/freeswitch/conf/autoload_configs$ grep dsn * | grep -v '!--' callcenter.conf.xml: db.conf.xml: switch.conf.xml: voicemail.conf.xml: and for both profiles: $ grep dsn /usr/local/freeswitch/conf/sip_profiles/* | grep -v '!--' /usr/local/freeswitch/conf/sip_profiles/external.xml: /usr/local/freeswitch/conf/sip_profiles/internal.xml: track-calls is also set for both of them, even though I guess this isn't relevant for my problems as long as I don't use recovery? $ grep track /usr/local/freeswitch/conf/sip_profiles/* | grep -v '!--' /usr/local/freeswitch/conf/sip_profiles/external.xml: /usr/local/freeswitch/conf/sip_profiles/internal.xml: As for group intercept, the log says (yes, I stripped most of it, you all know the default dialplan I guess ;) --- Dialplan: sofia/internal/1001 at sip.example.org Regex (PASS) [group-intercept] destination_number(*8) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/1001 at sip.example.org Action answer() Dialplan: sofia/internal/1001 at sip.example.org Action set(intercept_unanswered_only=true) Dialplan: sofia/internal/1001 at sip.example.org Action intercept(${hash(select/${domain_name}-last_dial/${call_group})}) Dialplan: sofia/internal/1001 at sip.example.org Action sleep(2000) EXECUTE sofia/internal/1001 at sip.example.org set(call_direction=local) EXECUTE sofia/internal/1001 at sip.example.org answer() 2012-07-27 22:31:35.693292 [NOTICE] mod_dptools.c:1148 Channel [sofia/internal/1001 at sip.example.org] has been answered EXECUTE sofia/internal/1001 at sip.example.org set(intercept_unanswered_only=true) EXECUTE sofia/internal/1001 at sip.example.org intercept() 2012-07-27 22:31:35.693292 [ERR] mod_dptools.c:714 Usage: [-bleg] --- in extension local_extension, the needed information gets stored in this hash: and this is working just fine, when enabling only one of the FS boxes. Did I miss something to share the content of hash's memory? What could I have missed, that subscriptions aren't working cross FS? (sip_subscriptions is filled pretty good, but BLF isn't working anyway) Any ideas if mod_callcenter should be working at all? I'm not quite sure how to interpret this note in the config: does that mean, mod_callcenter doesn't support multi FS setups or does it mean, that XML config isn't supported for multi FS environments, but it should be working fine with odbc? All of it's config exists in the database (callcenter_config works and confirms the config on all of the FS boxes), but it doesn't work since the moment more than one FS box accessed the database... Well, I know this is quite a beast of an email, but I'd really appreciate any pointers in the right direction! Thanks! From sos at sokhapkin.dyndns.org Sat Jul 28 03:22:43 2012 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Fri, 27 Jul 2012 19:22:43 -0400 Subject: [Freeswitch-users] What's better Unix ro Windows? LOL In-Reply-To: References: Message-ID: <4122395.tzcKGAtc0f@sos> I prefer GNU Midnight Commander editor in screen terminal window over ssh (can't kill DOS Norton Commander circa 1988 inside me :-) On Friday 27 July 2012 16:48:48 Anthony Minessale wrote: > On Jul 26, 2012 7:40 PM, "Ken Rice" wrote: > > most of freeswitch has been developedin an ssh terminal using screen and > > emacs... > > Yep, and it still is every day! > > > Ken > > Sent from my iPad > > > > On Jul 26, 2012, at 7:01 PM, Chris Mylonas wrote: > > >> I tend to do most of my dev on ssh terms > > > > > > v.interesting. > > > how the hell do you manage that? > > > > > > emacs/vim? > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.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 Sat Jul 28 04:44:18 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 27 Jul 2012 19:44:18 -0500 Subject: [Freeswitch-users] 10 Days till ClueCon! Message-ID: Countdown has started! Anyone who is going who has not registered, do it now so we can expedite the process for you and get the most accurate food totals. Anyone who is not going, Drat.... You still have 10 days to change your mind. Keeping in the tradition, BKW is always willing to sell your employer on the idea! -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From dujinfang at gmail.com Sat Jul 28 04:45:27 2012 From: dujinfang at gmail.com (Seven Du) Date: Sat, 28 Jul 2012 08:45:27 +0800 Subject: [Freeswitch-users] H264 transcodin In-Reply-To: <5012B0F9.5010201@redembedded.com> References: <5012B0F9.5010201@redembedded.com> Message-ID: <709B8F9F1687401094A356C3FB77D32D@gmail.com> Real-time encoding with statically linked x264 lib works fine for me from QCIF to D1 resolution, 720p is slow and discarding frames on a Xeon Quad core CPU. I haven't look how to use the GPU, or if possible. It is working in my lab and I have the same question with Yufei Tao when going to production or deliver to customer. Based on http://lists.freeswitch.org/pipermail/freeswitch-dev/2010-September/004227.html , In my understanding, compile and link and use by my self should be fine and, if I deliver to a customer, it should be fine if I provide the code and help the customer to compile on their own server? I'd like to open source the code to public later, but, I'd like to know is it a MUST or MAY? If you pipe to ffmpeg or x264 command line it's not been treated as combine into a large work, and I'm not sure if realtime transcoding will be smooth. 7. On Friday, July 27, 2012 at 11:17 PM, Yufei Tao wrote: > Thanks everyone for the responses! > > If I understand it correctly, if I installed ffmpeg on itself separately > from FS, I could write a module for FS, in which I just call the ffmpeg > program by running a command line. This way would it be classified as > "not combine them into a larger work", thus free from license > incompatibility problem? > > Not sure if that'll work for real-time transcoding of x264 though? > > Thanks very much for you opinions! > Yufei > > -- > Yufei Tao > Red Embedded > > This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. > > You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. > > Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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/20120728/81bf9928/attachment.html From dujinfang at gmail.com Sat Jul 28 05:10:10 2012 From: dujinfang at gmail.com (Seven Du) Date: Sat, 28 Jul 2012 09:10:10 +0800 Subject: [Freeswitch-users] H264 transcoding In-Reply-To: <5012387C.8050903@quentustech.com> References: <5E0B0409-B5A4-4722-ACDA-8F6140EE49C7@freeswitch.org> <5012387C.8050903@quentustech.com> Message-ID: I patched libvlc and it can now decode any video vlc supported and encode with x264 and send to any sip phone. Specifically I'm trying to use it to play a 1080p stream and resize to CIF or D1 so a video phone will accept(Sending 1080p will cause some phones to reboot :( ). Still need a lot of code to make it working neatly, however, I might can do a demo on ClueCon and @William if you'd like to review the code and merge into tree I'll happy to contribute that later. On Friday, July 27, 2012 at 2:43 PM, William King wrote: > libvlc is LGPL http://www.videolan.org/press/lgpl.html and there is now a mod_vlc(though it doesn't yet support video streams). The user can choose to build vlc with only the LGPL components or add the more 'adverse' modules. In none of the LGPL packages of libvlc is ffmpeg enabled, but there is a module for libvlc for ffmpeg. http://wiki.videolan.org/FFmpeg > > The only pieces now may just be the FS side of things for video. > William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com (mailto:william.king at quentustech.com) > On 07/26/2012 08:46 PM, Anthony Minessale wrote: > > you would probably need to do something like make a mod for ffmpeg that protects you from the gpl then allow the user to build that lib on his own and choose at compile time to install patented or adverse licensed components. No license rules prohibit an end user from combining code only distributors. but even then we need a bunch of code to write. On Thu, Jul 26, 2012 at 10:41 PM, curriegrad2004 (mailto:curriegrad2004 at gmail.com) wrote: > > > Ken, If you think those guys over at x264 will ever change the license from GPL to LGPL, you're just dreaming the pie in the sky... In short, don't even think about it ;P On Thu, Jul 26, 2012 at 8:19 PM, Ken Rice (mailto:krice at freeswitch.org) wrote: > > > > we can not and will not use GPL software, the license is not compatible with the GPL and would polute the codebase with additional restrictions that are not wanted or needed. now if someone could get them to change the license or atleast give us a license under better terms such as the LGPL or the MPL then the license issue would be null Ken Sent from my iPad On Jul 26, 2012, at 7:45 PM, Terry Barnum (mailto:terry at digital-outpost.com) wrote: > > > > > Use x264? http://en.wikipedia.org/wiki/X264 On Jul 26, 2012, at 4:53 PM, Ken Rice wrote: > > > > > > Is it possible sure... Is ot probably to happen anytime soon? Not until the patents run out... On 7/26/12 5:04 PM, "yufei.tao" (mailto:yufei.tao at redembedded.com) wrote: > > > > > > > Hi I am trying to decide if it is feasible to let FS do transcoding between different H264 formats for live video calls. This is because I've got SIP clients that both use H264 but with different formats and one (with a bad H264 decoder) has problems decoding H264 stream from the other. But each of these two clients communicate fine using H264 with a third client that uses ffmpeg. I'm thinking of adding a module which uses ffmpeg, so that it will transcode H264 between different parameters. I've got a few questions: 1. Is this feasible? I'm not looking at supporting many simultaneous calls. 2. What is involved in transcoding real-time video stream? 3. Anyone's done anything like this before? I'm new to FS and any suggestions would be very much appreciated! Yufei _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org (mailto:consulting at freeswitch.org) http://www.freeswitchsolutions. com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org (mailto:consulting at freeswitch.org) http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org > > > > > > > > > > > > > > > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org (mailto:consulting at freeswitch.org) http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org > > > > > > > > > > > > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org (mailto:consulting at freeswitch.org) http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org > > > > > > > > > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org (mailto:consulting at freeswitch.org) http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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/20120728/f93ac3e4/attachment-0001.html From mgg at giagnocavo.net Sat Jul 28 09:42:03 2012 From: mgg at giagnocavo.net (Michael Giagnocavo) Date: Sat, 28 Jul 2012 05:42:03 +0000 Subject: [Freeswitch-users] No reINVITE when bridging two sessions from mod_managed with bypass_media_after_bridge=true In-Reply-To: References: Message-ID: <63B00DD1DA6A364E9F64A3A0BD2FE7B612F890DC@BLUPRD0711MB413.namprd07.prod.outlook.com> So how for instance would you do logic like: set rate=0.01 bridge to foo if failure { Set rate = 0.02 Bridge to bar } If you have to exit after each bridge? Set continue after bridge to true, then re-enter your app after stashing away state? -Michael From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Friday, July 27, 2012 4:10 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] No reINVITE when bridging two sessions from mod_managed with bypass_media_after_bridge=true Do you exit the app after you call uuid bridge? Your code is running in the session thread so you need to exit so it can change states. On Jul 20, 2012 3:32 PM, "Srini K" > wrote: Hi, Iam trying to bypass media from FS after two call legs are bridged using mod_managed. The code looks like public void Run(AppContext context) { var fsApi = new FreeSWITCH.Native.Api(); var aLegSession = context.Session; // Answer the incoming call aLegSession.Answer(); // Play the prompt aLegSession.StreamFile("ivr/ThankYou.wav", 0); // Create outBound session var bLegSession = new ManagedSession("sofia/gateway/95/4151230000"); // Bypass Media aLegSession.SetVariable("bypass_media_after_bridge", "true"); bLegSession.SetVariable("bypass_media_after_bridge", "true"); fsApi.ExecuteString(string.Format("uuid_bridge {0} {1}", aLegSession.GetUuid(), bLegSession.GetUuid())); } I don't see FreeSWITCH sending re-Invite after the call is bridged. What I've already tried and did not succeed: 1) set bypass_media=true, on A leg only, on B leg only, on both legs 2) set bypass_media_after_bridge=true, on A leg only, on B leg only, on both legs When I tried without using mod_managed using only dialplan, FS sends re-Invite. Whether Iam doing anything stupid in mod_managed? Regards Srini _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120728/ed6d67d9/attachment.html From prasd.d.b at gmail.com Sat Jul 28 10:27:55 2012 From: prasd.d.b at gmail.com (Prasd D) Date: Fri, 27 Jul 2012 23:27:55 -0700 Subject: [Freeswitch-users] Push server and push capability In-Reply-To: References: Message-ID: I agree its best done as a module as opposed to core. It says, "RubyFS does not provide any features beyond connection management and protocol parsing. " It has to somehow use c2dm or (something close to that) in order to support it. It would be great to really see that FS + Csipsimple together have this working. On 7/26/12, Ben Langfeld wrote: > You could, and probably should, implement this as an inbound event socket > listener, rather than in FS core. You could do it in....ruby? > http://github.com/adhearsion/ruby_fs > > Happy hacking. > > Regards, > Ben Langfeld > > > On 26 July 2012 23:17, Prasd D wrote: > >> I was searching and came across this amazing jira request for push >> server capability. >> This is exactly what I felt too, except didn't delve or earlier >> understand how to achieve this. >> >> http://jira.freeswitch.org/browse/FS-4347 >> >> This guy has identified very much what is really needed to be able to >> use it on smartphones. >> And suggests that Freeswitch and Csipsimple together show this >> capability where an incoming call or voicemail to a user using >> Csipsimple is notified through push notification. That starts >> Csipsimple (which is otherwise turned off) on his phone. >> >> This is a huge huge and amazingly useful thing since no one keeps >> their voip client on their phones on due to power consumption and >> battery drain ! >> >> This I think is the future of VOIP since smartphones are going to >> become soon huge % of voip clients. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > -- Thanks, Prasd From anton.jugatsu at gmail.com Sat Jul 28 12:04:55 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Sat, 28 Jul 2012 12:04:55 +0400 Subject: [Freeswitch-users] Google Inc. DOES use FreeSWITCH for call center telephony routing system. So, why dont' we invite those guys to FS weekly conference :) In-Reply-To: References: Message-ID: We got response from Jeff Bates, Open Source IT Program Manager @ Google! It would be awesome to invite Jeff to FreeSWITCH weekly conference scheldued on the first august. All we got to do is just ping Jeff via e-mail: batesj [at] you can guess the company name. 2012/7/26 Gabriel Gunderson > On Wed, Jul 25, 2012 at 12:10 PM, Gabriel Gunderson > wrote: > > On Wed, Jul 25, 2012 at 11:46 AM, Anton Kvashenkin > > wrote: > >> Guys, according to those slides (*LINK* slide number 10). > > > > FYI, it looks like it slide #9. > > http://bit.ly/GOOGandFS > > That link takes you right to the slide... share it with your Asterisk > and Avaya buddies ;) > > > Gabe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120728/65f53d48/attachment.html From ben at langfeld.co.uk Sat Jul 28 12:25:58 2012 From: ben at langfeld.co.uk (Ben Langfeld) Date: Sat, 28 Jul 2012 10:25:58 +0200 Subject: [Freeswitch-users] Push server and push capability In-Reply-To: References: Message-ID: Prasd, RubyFS is for driving IES from Ruby. The Android push notification part is your problem ;) For that, this may help: https://rubygems.org/gems/c2dm Regards, Ben Langfeld On 28 July 2012 08:27, Prasd D wrote: > I agree its best done as a module as opposed to core. > > It says, "RubyFS does not provide any features beyond connection > management and protocol parsing. " > > It has to somehow use c2dm or (something close to that) in order to > support it. > > It would be great to really see that FS + Csipsimple together have this > working. > > > On 7/26/12, Ben Langfeld wrote: > > You could, and probably should, implement this as an inbound event socket > > listener, rather than in FS core. You could do it in....ruby? > > http://github.com/adhearsion/ruby_fs > > > > Happy hacking. > > > > Regards, > > Ben Langfeld > > > > > > On 26 July 2012 23:17, Prasd D wrote: > > > >> I was searching and came across this amazing jira request for push > >> server capability. > >> This is exactly what I felt too, except didn't delve or earlier > >> understand how to achieve this. > >> > >> http://jira.freeswitch.org/browse/FS-4347 > >> > >> This guy has identified very much what is really needed to be able to > >> use it on smartphones. > >> And suggests that Freeswitch and Csipsimple together show this > >> capability where an incoming call or voicemail to a user using > >> Csipsimple is notified through push notification. That starts > >> Csipsimple (which is otherwise turned off) on his phone. > >> > >> This is a huge huge and amazingly useful thing since no one keeps > >> their voip client on their phones on due to power consumption and > >> battery drain ! > >> > >> This I think is the future of VOIP since smartphones are going to > >> become soon huge % of voip clients. > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > -- > Thanks, > Prasd > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120728/ef589ff3/attachment-0001.html From chrisbware at yahoo.it Sat Jul 28 13:39:34 2012 From: chrisbware at yahoo.it (Chris B. Ware) Date: Sat, 28 Jul 2012 10:39:34 +0100 (BST) Subject: [Freeswitch-users] Fifo reparse Message-ID: <1343468374.37656.YahooMailNeo@web132306.mail.ird.yahoo.com> Hi guys, if I have calls in fifos and try to add another fifo through XML curl using "fifo reparse", does it impact on active calls ? Thank you for your help, Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120728/153db17f/attachment.html From dujinfang at gmail.com Sat Jul 28 17:13:48 2012 From: dujinfang at gmail.com (Seven Du) Date: Sat, 28 Jul 2012 21:13:48 +0800 Subject: [Freeswitch-users] Fifo reparse In-Reply-To: <1343468374.37656.YahooMailNeo@web132306.mail.ird.yahoo.com> References: <1343468374.37656.YahooMailNeo@web132306.mail.ird.yahoo.com> Message-ID: <070354AA63F64088B2676FFE9E4F6A4C@gmail.com> It should not, can you try to confirm? On Saturday, July 28, 2012 at 5:39 PM, Chris B. Ware wrote: > Hi guys, > > if I have calls in fifos and try to add another fifo through XML curl using "fifo reparse", does it impact on active calls ? > > > Thank you for your help, > Chris > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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/20120728/6eaabd8f/attachment.html From govoiper at gmail.com Sat Jul 28 17:30:39 2012 From: govoiper at gmail.com (SamyGo) Date: Sat, 28 Jul 2012 18:30:39 +0500 Subject: [Freeswitch-users] Bind Sofia external profile to Static private IP Message-ID: Hello, Probably an easy thing but my luck I'm stuck in this awkward situation here, I want my sofia external profile to bind to my private IP: 192.168.15.21 , so for the purpose I've edited the vars.xml and sip_profiles/external.xml but still everytime I restart freeswitch it detects the Public IP and binds thye external-sip-ip and external-rtp-ip to the Public IP ? I'm very confused at this and followed all the previous mailing lists threads and to-dos as well but in vain. Waiting for a savior here. -- Thanks Sammy Go. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120728/c9efd3f8/attachment.html From brian at freeswitch.org Sat Jul 28 17:34:05 2012 From: brian at freeswitch.org (Brian West) Date: Sat, 28 Jul 2012 08:34:05 -0500 Subject: [Freeswitch-users] Bind Sofia external profile to Static private IP In-Reply-To: References: Message-ID: <4664115619900296997@unknownmsgid> Open the profile directly and edit it, vars.xml is for ease not flexibility. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 On Jul 28, 2012, at 8:31 AM, SamyGo wrote: > Hello, > Probably an easy thing but my luck I'm stuck in this awkward situation here, I want my sofia external profile to bind to my private IP: 192.168.15.21 , so for the purpose I've edited the vars.xml and sip_profiles/external.xml but still everytime I restart freeswitch it detects the Public IP and binds thye external-sip-ip and external-rtp-ip to the Public IP ? > > I'm very confused at this and followed all the previous mailing lists threads and to-dos as well but in vain. > > Waiting for a savior here. > -- > Thanks > Sammy Go. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Jul 28 17:35:04 2012 From: brian at freeswitch.org (Brian West) Date: Sat, 28 Jul 2012 08:35:04 -0500 Subject: [Freeswitch-users] Fifo reparse In-Reply-To: <1343468374.37656.YahooMailNeo@web132306.mail.ird.yahoo.com> References: <1343468374.37656.YahooMailNeo@web132306.mail.ird.yahoo.com> Message-ID: <-3636852024308739286@unknownmsgid> No, but usually faster to try before asking. ;) -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 On Jul 28, 2012, at 4:41 AM, "Chris B. Ware" wrote: Hi guys, if I have calls in fifos and try to add another fifo through XML curl using "fifo reparse", does it impact on active calls ? Thank you for your help, Chris _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120728/ae619844/attachment.html From brian at freeswitch.org Sat Jul 28 17:39:03 2012 From: brian at freeswitch.org (Brian West) Date: Sat, 28 Jul 2012 08:39:03 -0500 Subject: [Freeswitch-users] Push server and push capability In-Reply-To: References: Message-ID: <1299420284849059083@unknownmsgid> Seems like a very specific feature to have, needs to be a module. Also who is writing it? -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 On Jul 28, 2012, at 3:27 AM, Ben Langfeld wrote: Prasd, RubyFS is for driving IES from Ruby. The Android push notification part is your problem ;) For that, this may help: https://rubygems.org/gems/c2dm Regards, Ben Langfeld On 28 July 2012 08:27, Prasd D wrote: > I agree its best done as a module as opposed to core. > > It says, "RubyFS does not provide any features beyond connection > management and protocol parsing. " > > It has to somehow use c2dm or (something close to that) in order to > support it. > > It would be great to really see that FS + Csipsimple together have this > working. > > > On 7/26/12, Ben Langfeld wrote: > > You could, and probably should, implement this as an inbound event socket > > listener, rather than in FS core. You could do it in....ruby? > > http://github.com/adhearsion/ruby_fs > > > > Happy hacking. > > > > Regards, > > Ben Langfeld > > > > > > On 26 July 2012 23:17, Prasd D wrote: > > > >> I was searching and came across this amazing jira request for push > >> server capability. > >> This is exactly what I felt too, except didn't delve or earlier > >> understand how to achieve this. > >> > >> http://jira.freeswitch.org/browse/FS-4347 > >> > >> This guy has identified very much what is really needed to be able to > >> use it on smartphones. > >> And suggests that Freeswitch and Csipsimple together show this > >> capability where an incoming call or voicemail to a user using > >> Csipsimple is notified through push notification. That starts > >> Csipsimple (which is otherwise turned off) on his phone. > >> > >> This is a huge huge and amazingly useful thing since no one keeps > >> their voip client on their phones on due to power consumption and > >> battery drain ! > >> > >> This I think is the future of VOIP since smartphones are going to > >> become soon huge % of voip clients. > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > -- > Thanks, > Prasd > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120728/04427695/attachment-0001.html From brian at freeswitch.org Sat Jul 28 17:42:37 2012 From: brian at freeswitch.org (Brian West) Date: Sat, 28 Jul 2012 08:42:37 -0500 Subject: [Freeswitch-users] H264 transcoding In-Reply-To: <5012387C.8050903@quentustech.com> References: <5E0B0409-B5A4-4722-ACDA-8F6140EE49C7@freeswitch.org> <5012387C.8050903@quentustech.com> Message-ID: <8254643968774186641@unknownmsgid> X264 is NOT. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 On Jul 27, 2012, at 1:44 AM, William King wrote: libvlc is LGPL http://www.videolan.org/press/lgpl.html and there is now a mod_vlc(though it doesn't yet support video streams). The user can choose to build vlc with only the LGPL components or add the more 'adverse' modules. In none of the LGPL packages of libvlc is ffmpeg enabled, but there is a module for libvlc for ffmpeg. http://wiki.videolan.org/FFmpeg The only pieces now may just be the FS side of things for video. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518william.king at quentustech.com On 07/26/2012 08:46 PM, Anthony Minessale wrote: you would probably need to do something like make a mod for ffmpeg that protects you from the gpl then allow the user to build that lib on his own and choose at compile time to install patented or adverse licensed components. No license rules prohibit an end user from combining code only distributors. but even then we need a bunch of code to write. On Thu, Jul 26, 2012 at 10:41 PM, curriegrad2004 wrote: Ken, If you think those guys over at x264 will ever change the license from GPL to LGPL, you're just dreaming the pie in the sky... In short, don't even think about it ;P On Thu, Jul 26, 2012 at 8:19 PM, Ken Rice wrote: we can not and will not use GPL software, the license is not compatible with the GPL and would polute the codebase with additional restrictions that are not wanted or needed. now if someone could get them to change the license or atleast give us a license under better terms such as the LGPL or the MPL then the license issue would be null Ken Sent from my iPad On Jul 26, 2012, at 7:45 PM, Terry Barnum wrote: Use x264? http://en.wikipedia.org/wiki/X264 On Jul 26, 2012, at 4:53 PM, Ken Rice wrote: Is it possible sure... Is ot probably to happen anytime soon? Not until the patents run out... On 7/26/12 5:04 PM, "yufei.tao" wrote: Hi I am trying to decide if it is feasible to let FS do transcoding between different H264 formats for live video calls. This is because I've got SIP clients that both use H264 but with different formats and one (with a bad H264 decoder) has problems decoding H264 stream from the other. But each of these two clients communicate fine using H264 with a third client that uses ffmpeg. I'm thinking of adding a module which uses ffmpeg, so that it will transcode H264 between different parameters. I've got a few questions: 1. Is this feasible? I'm not looking at supporting many simultaneous calls. 2. What is involved in transcoding real-time video stream? 3. Anyone's done anything like this before? I'm new to FS and any suggestions would be very much appreciated! Yufei _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel Communication Server Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120728/d73b4670/attachment.html From brian at freeswitch.org Sat Jul 28 17:46:14 2012 From: brian at freeswitch.org (Brian West) Date: Sat, 28 Jul 2012 08:46:14 -0500 Subject: [Freeswitch-users] Setting effecting_caller_id_name In-Reply-To: References: Message-ID: <4854635138739576167@unknownmsgid> Sounds like your doing openCNAM, check the cnam.cgi in tree, you should be checking the response code and not the text returned. I built the wrapper to clean up the input before I query because of this. -- Brian West brian at freeswitch.org FreeSWITCH Solutions, LLC PO BOX PO BOX 2531 Brookfield, WI 53008-2531 Twitter: @FreeSWITCH_Wire T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST iNUM: +883 5100 1286 0410 On Jul 27, 2012, at 10:06 AM, Ken Rice wrote: Re: [Freeswitch-users] Setting effecting_caller_id_name This exmaple should be correct provided that caller_id_name containts ?Currently runnint a lookup? with this pinned to that start of the field... On that extension in your dialplan toss a If your regex isnt matching on the caller_id_name field the above will show you whats in there (along with every thing else too heh) K On 7/26/12 7:11 PM, "Brian Foster" wrote: _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120728/488ef150/attachment-0001.html From govoiper at gmail.com Sat Jul 28 18:34:37 2012 From: govoiper at gmail.com (SamyGo) Date: Sat, 28 Jul 2012 19:34:37 +0500 Subject: [Freeswitch-users] Bind Sofia external profile to Static private IP In-Reply-To: <4664115619900296997@unknownmsgid> References: <4664115619900296997@unknownmsgid> Message-ID: yes I tried that too edited the profile external.xml directly but in vain..so I was reading another thread here of a person who had totally inverse issue and Michael asked if he is starting freeswitch with -nonat , applying that switch while starting freeswitch seems to have worked here. On Sat, Jul 28, 2012 at 6:34 PM, Brian West wrote: > Open the profile directly and edit it, vars.xml is for ease not > flexibility. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST > iNUM: +883 5100 1286 0410 > > On Jul 28, 2012, at 8:31 AM, SamyGo wrote: > > > Hello, > > Probably an easy thing but my luck I'm stuck in this awkward situation > here, I want my sofia external profile to bind to my private IP: > 192.168.15.21 , so for the purpose I've edited the vars.xml and > sip_profiles/external.xml but still everytime I restart freeswitch it > detects the Public IP and binds thye external-sip-ip and external-rtp-ip to > the Public IP ? > > > > I'm very confused at this and followed all the previous mailing lists > threads and to-dos as well but in vain. > > > > Waiting for a savior here. > > -- > > Thanks > > Sammy Go. > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120728/b8b84038/attachment.html From govoiper at gmail.com Sat Jul 28 18:59:26 2012 From: govoiper at gmail.com (SamyGo) Date: Sat, 28 Jul 2012 19:59:26 +0500 Subject: [Freeswitch-users] LUA session:Bridge not actually bridging calls ~ Message-ID: Hello, I wanted to make a lua script which just dials out two different numbers via some external gateway and when both calls are answered they are just bridged. For this a very impressive Lua example http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control is copied and all I had to do was change the dialA and dialB strings and its working great as far as the SIP signalling is concerned. execute this string and I get calls on two different number but things get interesting when Freeswitch bridge() the two legs. No AUDIO..not even one-way. I could see on my own gateway that RTPs for both the legs are actually forwarded to Freeswitch ! On my sip pcap traces analyzing on wireshark I could actually hear the two persons saying Hello but neither could hear anything. The above example lua call_control script is used as it is. Please suggest. Regards Sammy Go. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120728/9742b1be/attachment.html From govoiper at gmail.com Sat Jul 28 19:18:56 2012 From: govoiper at gmail.com (SamyGo) Date: Sat, 28 Jul 2012 20:18:56 +0500 Subject: [Freeswitch-users] LUA session:Bridge not actually bridging calls ~ In-Reply-To: References: Message-ID: Here are the FS console logs: http://pastebin.freeswitch.org/19595 Please suggest what am I missing here. On Sat, Jul 28, 2012 at 7:59 PM, SamyGo wrote: > Hello, > I wanted to make a lua script which just dials out two different numbers > via some external gateway and when both calls are answered they are just > bridged. For this a very impressive Lua example > http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control is copied > and all I had to do was change the dialA and dialB strings and its working > great as far as the SIP signalling is concerned. > > execute this string and I get calls on two different number but things get > interesting when Freeswitch bridge() the two legs. No AUDIO..not even > one-way. I could see on my own gateway that RTPs for both the legs are > actually forwarded to Freeswitch ! > > On my sip pcap traces analyzing on wireshark I could actually hear the two > persons saying Hello but neither could hear anything. > > The above example lua call_control script is used as it is. > Please suggest. > > Regards > Sammy Go. > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120728/ddbdc3fa/attachment.html From andrew at cassidywebservices.co.uk Sat Jul 28 19:24:31 2012 From: andrew at cassidywebservices.co.uk (Andrew Cassidy) Date: Sat, 28 Jul 2012 16:24:31 +0100 Subject: [Freeswitch-users] FreeSWITCH dying with no core file In-Reply-To: <5013012C.9050003@freeswitch.org> References: <5013012C.9050003@freeswitch.org> Message-ID: I'll also give that a go as I have similar random restarts. On 27 July 2012 21:59, Stefan Knoblich wrote: > -----BEGIN PGP SIGNED MESSAGE----- > Hash: SHA1 > > On 07/27/12 21:34, Adam Johnston wrote: > > I use FreeSWITCH exclusively for high-volume fax, both sending and > receiving, and I've been running into a sporadic issue where the FreeSWITCH > process will die with nothing in the logs to > > indicate an issue and no core file anywhere on the system. I'm setting > "ulimit -c unlimited" in my init script, so I would expect to see a core > file. Version: FreeSWITCH Version > > 1.2.0-rc2+git~20120713T162602Z~0afd7318bd+unclean~20120713T182401Z > > You need to set /proc/sys/fs/suid_dumpable to a non-zero value (see man 5 > proc), to enable core dumping for processes > that switch their UID on startup (such as freeswitch with -u option). > > By default, the freeswitch process will try to put the core file into the > CWD (current work directory), > so make sure the UID it is running as can write there (or "cd" into a > different directory before starting it). > You might also want to check out man 5 core, for all the other options > (e.g. piping core dumps into a custom handler, with core_pattern). > > -----BEGIN PGP SIGNATURE----- > Version: GnuPG v2.0.19 (GNU/Linux) > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ > > iEYEARECAAYFAlATASwACgkQjiIIAK4rYUrjvQCgh1grtcBfPVhbZNw8tdQpsyda > 614Amwb1pKDFL93dU33cy2wj1FNYYnzJ > =0Xqk > -----END PGP SIGNATURE----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- *Andrew Cassidy BSc (Hons) MBCS SSCA* Managing Director *T *03300 100 960 *F *03300 100 961 *E *andrew at cassidywebservices.co.uk *W *www.cassidywebservices.co.uk -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120728/6852528c/attachment.html From Tim.Meade at Millicorp.com Sat Jul 28 19:24:36 2012 From: Tim.Meade at Millicorp.com (Tim Meade) Date: Sat, 28 Jul 2012 15:24:36 +0000 Subject: [Freeswitch-users] ODBC, FreeTDS, Microsoft SQL Message-ID: <804D48104511D4468F0D60DF9D310035095C33BC@MAILBOX.millicorp.com> We wanted to run some ODBC tests using a MSSQL backend. We setup the ODBC with MySQL and everything worked great. Then installed FreeTDS and configured it instead of MySQL. When FS restarted, it created some of the required tables. Calls, complete, interfaces, nat, sip_presence, sip_recovery, sip_shared_apperance_subscriptions. What it did not create was any of the registrations tables. When a device tries to register, FS shows in the CLI: 012-07-28 09:43:48.892312 [INFO] switch_odbc.c:285 The connection has been re-established 2012-07-28 09:43:49.912319 [ERR] switch_odbc.c:494 ERR: [create index sr_sip_host on sip_registrations (sip_host)] [STATE: 42000 CODE 1088 ERROR: [unixODBC][FreeTDS][SQL Server]Cannot find the object "sip_registrations" because it does not exist or you do not have permissions. ] 2012-07-28 09:43:49.912319 [ERR] switch_core_sqldb.c:487 SQL ERR [STATE: 42000 CODE 1088 ERROR: [unixODBC][FreeTDS][SQL Server]Cannot find the object "sip_registrations" because it does not exist or you do not have permissions. ] create index sr_sip_host on sip_registrations (sip_host) 2012-07-28 09:43:49.912319 [CRIT] switch_odbc.c:280 The sql server is not responding for DSN freeswitch [STATE: 42000 CODE 1088 ERROR: [unixODBC][FreeTDS][SQL Server]Cannot find the object "sip_registrations" because it does not exist or you do not have permissions. ][244] Seems the issue is that FS didn't create all the appropriate tables on start. Has anyone else seen something like this? I cannot find much on using FS with FreeTDS and MSSQL. Thanks Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120728/ea235ea8/attachment-0001.html From curriegrad2004 at gmail.com Sat Jul 28 20:09:34 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sat, 28 Jul 2012 09:09:34 -0700 Subject: [Freeswitch-users] H264 transcoding In-Reply-To: <8254643968774186641@unknownmsgid> References: <5E0B0409-B5A4-4722-ACDA-8F6140EE49C7@freeswitch.org> <5012387C.8050903@quentustech.com> <8254643968774186641@unknownmsgid> Message-ID: What do you mean it's not? On Sat, Jul 28, 2012 at 6:42 AM, Brian West wrote: > X264 is NOT. > > -- > Brian West > brian at freeswitch.org > FreeSWITCH Solutions, LLC > PO BOX PO BOX 2531 > Brookfield, WI 53008-2531 > Twitter: @FreeSWITCH_Wire > T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST > iNUM: +883 5100 1286 0410 > > On Jul 27, 2012, at 1:44 AM, William King > wrote: > > libvlc is LGPL http://www.videolan.org/press/lgpl.html and there is now a > mod_vlc(though it doesn't yet support video streams). The user can choose to > build vlc with only the LGPL components or add the more 'adverse' modules. > In none of the LGPL packages of libvlc is ffmpeg enabled, but there is a > module for libvlc for ffmpeg. http://wiki.videolan.org/FFmpeg > > The only pieces now may just be the FS side of things for video. > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > > On 07/26/2012 08:46 PM, Anthony Minessale wrote: > > you would probably need to do something like make a mod for ffmpeg > that protects you from the gpl then allow the user to build that lib > on his own and choose at compile time to install patented or adverse > licensed components. No license rules prohibit an end user from > combining code only distributors. > > but even then we need a bunch of code to write. > > > On Thu, Jul 26, 2012 at 10:41 PM, curriegrad2004 > wrote: > > Ken, > > If you think those guys over at x264 will ever change the license from > GPL to LGPL, you're just dreaming the pie in the sky... > > In short, don't even think about it ;P > > On Thu, Jul 26, 2012 at 8:19 PM, Ken Rice wrote: > > we can not and will not use GPL software, the license is not compatible with > the GPL and would polute the codebase with additional restrictions that are > not wanted or needed. now if someone could get them to change the license or > atleast give us a license under better terms such as the LGPL or the MPL > then the license issue would be null > > Ken > Sent from my iPad > > On Jul 26, 2012, at 7:45 PM, Terry Barnum wrote: > > Use x264? http://en.wikipedia.org/wiki/X264 > > On Jul 26, 2012, at 4:53 PM, Ken Rice wrote: > > Is it possible sure... Is ot probably to happen anytime soon? Not until the > patents run out... > > > On 7/26/12 5:04 PM, "yufei.tao" wrote: > > Hi > > I am trying to decide if it is feasible to let FS do transcoding between > different H264 formats for live video calls. This is because I've got > SIP clients that both use H264 but with different formats and one (with > a bad H264 decoder) has problems decoding H264 stream from the other. > But each of these two clients communicate fine using H264 with a third > client that uses ffmpeg. I'm thinking of adding a module which uses > ffmpeg, so that it will transcode H264 between different parameters. > > I've got a few questions: > > 1. Is this feasible? I'm not looking at supporting many simultaneous calls. > 2. What is involved in transcoding real-time video stream? > 3. Anyone's done anything like this before? > > I'm new to FS and any suggestions would be very much appreciated! > > Yufei > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From govoiper at gmail.com Sat Jul 28 20:23:16 2012 From: govoiper at gmail.com (SamyGo) Date: Sat, 28 Jul 2012 21:23:16 +0500 Subject: [Freeswitch-users] LUA session:Bridge not actually bridging calls ~ In-Reply-To: References: Message-ID: Hi again, So, It was a very minor change in configuration and it was working. Basically FreeSwicth was bridging the two legs BUT there was a codec issue. All I had to do was in Asterisk (serving as my gateway) to allow only ulaw and alaw i.e disallow=all allow=ulaw allow=alaw What I really really wish to know is that why there was no indication of codec mismatch or ptime mismatch or sample rate mismatch while transcoding or anything. It will be fine if none replies but it will be great to know the real reason behind this and from where in logs can I verify this !! Thanks Sammy On Sat, Jul 28, 2012 at 8:18 PM, SamyGo wrote: > Here are the FS console logs: > http://pastebin.freeswitch.org/19595 > > Please suggest what am I missing here. > > > On Sat, Jul 28, 2012 at 7:59 PM, SamyGo wrote: > >> Hello, >> I wanted to make a lua script which just dials out two different numbers >> via some external gateway and when both calls are answered they are just >> bridged. For this a very impressive Lua example >> http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control is copied >> and all I had to do was change the dialA and dialB strings and its working >> great as far as the SIP signalling is concerned. >> >> execute this string and I get calls on two different number but things >> get interesting when Freeswitch bridge() the two legs. No AUDIO..not even >> one-way. I could see on my own gateway that RTPs for both the legs are >> actually forwarded to Freeswitch ! >> >> On my sip pcap traces analyzing on wireshark I could actually hear the >> two persons saying Hello but neither could hear anything. >> >> The above example lua call_control script is used as it is. >> Please suggest. >> >> Regards >> Sammy Go. >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120728/703a6d6f/attachment.html From sdevoy at bizfocused.com Sat Jul 28 20:40:28 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sat, 28 Jul 2012 12:40:28 -0400 Subject: [Freeswitch-users] SIP to make phone reboot or resync In-Reply-To: References: <07b501cd6c1a$238b3e90$6aa1bbb0$@bizfocused.com> <094e01cd6c43$9e44cc90$dace65b0$@bizfocused.com> Message-ID: <0c4001cd6cdf$b2603a10$1720ae30$@bizfocused.com> HI all, I am close, but I still don't understand what credentials are required. The response is: SIP/2.0 401 Unauthorized I have tried the web admin credentials for the phone, I don't k now what FS credentials I could pass. Any ideas? Thanks, Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Friday, July 27, 2012 6:12 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync Make a gateway with no reg and the credentials and name it after the realm in the challenge. On Jul 27, 2012 5:05 PM, "Sean Devoy" wrote: Thank Anthony, BUT.. Error: 2012-07-27 17:58:55.118642 [ERR] sofia_reg.c:2165 Cannot locate any authentication credentials to complete an authentication request for realm '"fs_bfis.bizfocused.com"' 1 - where do I specify the credentials? 2 - Are these Freeswitch credentials or phone credentials? I suspect the latter. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Friday, July 27, 2012 1:21 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync sofia profile check_sync [call_id] // no call_id means all sofia profile flush_inbound_reg [call_id] [reboot] On Fri, Jul 27, 2012 at 12:06 PM, Sean Devoy wrote: > HI All, > > > > Does anyone have an code that will cause Freeswitch to send a SIP > message to a CISCO SPA5xx phone that will cause it to reboot or resync > (aka re-provision)? I know about the URLs that cause the phone to do > this, but they are NATed, so I need to use SIP to hit them. > > > > Assume I am programming language omni-lingual for this request! > > > > Thanks, > > Sean > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120728/331a45e6/attachment-0001.html From vallimamod.abdullah at imtelecom.fr Sat Jul 28 21:04:57 2012 From: vallimamod.abdullah at imtelecom.fr (Vallimamod ABDULLAH) Date: Sat, 28 Jul 2012 19:04:57 +0200 Subject: [Freeswitch-users] SIP to make phone reboot or resync In-Reply-To: <0c4001cd6cdf$b2603a10$1720ae30$@bizfocused.com> References: <07b501cd6c1a$238b3e90$6aa1bbb0$@bizfocused.com> <094e01cd6c43$9e44cc90$dace65b0$@bizfocused.com> <0c4001cd6cdf$b2603a10$1720ae30$@bizfocused.com> Message-ID: Hi, It's normally the sip account credentials. There is also an option in the admin interface to disable this authentication ("Auth Resync-Reboot" option if I recall correctly in Ext tab) Best Regards, Vallimamod Abdullah . On Jul 28, 2012, at 6:40 PM, Sean Devoy wrote: > HI all, > > I am close, but I still don?t understand what credentials are required. The response is: > SIP/2.0 401 Unauthorized > > I have tried the web admin credentials for the phone, I don?t k now what FS credentials I could pass. > > Any ideas? > > Thanks, > Sean > > From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale > Sent: Friday, July 27, 2012 6:12 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync > > Make a gateway with no reg and the credentials and name it after the realm in the challenge. > > On Jul 27, 2012 5:05 PM, "Sean Devoy" wrote: > Thank Anthony, BUT.. Error: > 2012-07-27 17:58:55.118642 [ERR] sofia_reg.c:2165 Cannot locate any > authentication credentials to complete an authentication request for realm > '"fs_bfis.bizfocused.com"' > > 1 - where do I specify the credentials? > 2 - Are these Freeswitch credentials or phone credentials? I suspect the > latter. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony > Minessale > Sent: Friday, July 27, 2012 1:21 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync > > sofia profile check_sync [call_id] // no call_id means all > sofia profile flush_inbound_reg [call_id] [reboot] > > > > On Fri, Jul 27, 2012 at 12:06 PM, Sean Devoy wrote: > > HI All, > > > > > > > > Does anyone have an code that will cause Freeswitch to send a SIP > > message to a CISCO SPA5xx phone that will cause it to reboot or resync > > (aka re-provision)? I know about the URLs that cause the phone to do > > this, but they are NATed, so I need to use SIP to hit them. > > > > > > > > Assume I am programming language omni-lingual for this request! > > > > > > > > Thanks, > > > > Sean > > > > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users 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 > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From mi.ke at null.net Sat Jul 28 22:12:48 2012 From: mi.ke at null.net (Mi Ke) Date: Sat, 28 Jul 2012 14:12:48 -0400 Subject: [Freeswitch-users] hunting in failover mode Message-ID: <20120728181248.154600@gmx.com> worked for me in combination with 1|2|3 construction ----- Original Message ----- From: Mi Ke Sent: 07/20/12 09:03 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] hunting in failover mode Michael, thanks for answering - I already tried this but I need to set continue_on_fail to TRUE because that seems to be the only way to pass original hangup code from leg B to A when bridge completes. And when COF is set to TRUE it behaves like it's related to the "whole" bridge i.e. until all 3 endpoints are tried entire bridge is not considered as failed of succeded. Re separate approach - I'm not sure how can I do that as the number of EPs in my case is dynamic, bridge app receives them as a 1|2|3 dialstring from rouing lua app. Thanks / Mike ----- Original Message ----- From: Michael Collins Sent: 07/20/12 08:39 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] hunting in failover mode Perhaps you could use this? http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail Specify the conditions on which the bridge should keep going. Note: you may need to separate these out into individual bridge apps with a single endpoint. I haven't tried it lately, so be sure to report back what you find out. -MC On Fri, Jul 20, 2012 at 10:30 AM, Mi Ke < mi.ke at null.net > wrote: Hi All, When I do a failover using bridge app and ep1|ep2|ep3 construstion - how do I stop further hunting if any of already hunted endpoints returned e.g. USER_BUSY ? Thanks / Mike _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120728/346a09e2/attachment.html From Tim.Meade at Millicorp.com Sat Jul 28 23:52:18 2012 From: Tim.Meade at Millicorp.com (Tim Meade) Date: Sat, 28 Jul 2012 19:52:18 +0000 Subject: [Freeswitch-users] ODBC, FreeTDS, Microsoft SQL In-Reply-To: <804D48104511D4468F0D60DF9D310035095C33BC@MAILBOX.millicorp.com> References: <804D48104511D4468F0D60DF9D310035095C33BC@MAILBOX.millicorp.com> Message-ID: <804D48104511D4468F0D60DF9D310035095C563F@MAILBOX.millicorp.com> I've found a work around using the Microsoft SQL drivers instead of FreeTDS. Following most of this: http://msdn.microsoft.com/en-us/library/hh568454.aspx and download here: http://www.microsoft.com/en-us/download/details.aspx?id=28160 Great install instructions: http://blog.nhaslam.com/2011/12/12/sql-server-odbc-on-linux/ and setting the obc.ini to: [freeswitch] Driver = SQL Server Native Client 11.0 Description = FS SQL Tester Trace = No Server = 192.168.5.10 Port = 1433 Database = FSTester All the tables are created properly. Tim From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim Meade Sent: Saturday, July 28, 2012 11:25 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] ODBC, FreeTDS, Microsoft SQL We wanted to run some ODBC tests using a MSSQL backend. We setup the ODBC with MySQL and everything worked great. Then installed FreeTDS and configured it instead of MySQL. When FS restarted, it created some of the required tables. Calls, complete, interfaces, nat, sip_presence, sip_recovery, sip_shared_apperance_subscriptions. What it did not create was any of the registrations tables. When a device tries to register, FS shows in the CLI: 012-07-28 09:43:48.892312 [INFO] switch_odbc.c:285 The connection has been re-established 2012-07-28 09:43:49.912319 [ERR] switch_odbc.c:494 ERR: [create index sr_sip_host on sip_registrations (sip_host)] [STATE: 42000 CODE 1088 ERROR: [unixODBC][FreeTDS][SQL Server]Cannot find the object "sip_registrations" because it does not exist or you do not have permissions. ] 2012-07-28 09:43:49.912319 [ERR] switch_core_sqldb.c:487 SQL ERR [STATE: 42000 CODE 1088 ERROR: [unixODBC][FreeTDS][SQL Server]Cannot find the object "sip_registrations" because it does not exist or you do not have permissions. ] create index sr_sip_host on sip_registrations (sip_host) 2012-07-28 09:43:49.912319 [CRIT] switch_odbc.c:280 The sql server is not responding for DSN freeswitch [STATE: 42000 CODE 1088 ERROR: [unixODBC][FreeTDS][SQL Server]Cannot find the object "sip_registrations" because it does not exist or you do not have permissions. ][244] Seems the issue is that FS didn't create all the appropriate tables on start. Has anyone else seen something like this? I cannot find much on using FS with FreeTDS and MSSQL. Thanks Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120728/5500fdab/attachment-0001.html From covici at ccs.covici.com Sun Jul 29 03:55:24 2012 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Sat, 28 Jul 2012 19:55:24 -0400 Subject: [Freeswitch-users] FreeSWITCH dying with no core file In-Reply-To: References: <5013012C.9050003@freeswitch.org> Message-ID: <28307.1343519724@ccs.covici.com> Andrew Cassidy wrote: > I'll also give that a go as I have similar random restarts. > > On 27 July 2012 21:59, Stefan Knoblich wrote: > > > -----BEGIN PGP SIGNED MESSAGE----- > > Hash: SHA1 > > > > On 07/27/12 21:34, Adam Johnston wrote: > > > I use FreeSWITCH exclusively for high-volume fax, both sending and > > receiving, and I've been running into a sporadic issue where the FreeSWITCH > > process will die with nothing in the logs to > > > indicate an issue and no core file anywhere on the system. I'm setting > > "ulimit -c unlimited" in my init script, so I would expect to see a core > > file. Version: FreeSWITCH Version > > > 1.2.0-rc2+git~20120713T162602Z~0afd7318bd+unclean~20120713T182401Z > > > > You need to set /proc/sys/fs/suid_dumpable to a non-zero value (see man 5 > > proc), to enable core dumping for processes > > that switch their UID on startup (such as freeswitch with -u option). > > > > By default, the freeswitch process will try to put the core file into the > > CWD (current work directory), > > so make sure the UID it is running as can write there (or "cd" into a > > different directory before starting it). > > You might also want to check out man 5 core, for all the other options > > (e.g. piping core dumps into a custom handler, with core_pattern). > > > > -----BEGIN PGP SIGNATURE----- > > Version: GnuPG v2.0.19 (GNU/Linux) > > Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org/ > > > > iEYEARECAAYFAlATASwACgkQjiIIAK4rYUrjvQCgh1grtcBfPVhbZNw8tdQpsyda > > 614Amwb1pKDFL93dU33cy2wj1FNYYnzJ > > =0Xqk > > -----END PGP SIGNATURE----- > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > Thanks much for that tip -- I had never heard of that parameter, maybe it should be in the wiki in the section about getting a backtrace. -- Your life is like a penny. You're going to lose it. The question is: How do you spend it? John Covici covici at ccs.covici.com From william.suffill at gmail.com Sun Jul 29 05:44:53 2012 From: william.suffill at gmail.com (William Suffill) Date: Sat, 28 Jul 2012 21:44:53 -0400 Subject: [Freeswitch-users] H264 transcoding In-Reply-To: References: <5E0B0409-B5A4-4722-ACDA-8F6140EE49C7@freeswitch.org> <5012387C.8050903@quentustech.com> <8254643968774186641@unknownmsgid> Message-ID: Probably that x264 isn't licensed LGPL since it's under a GPL/Commercial licensing AFAIK. From sdevoy at bizfocused.com Sun Jul 29 06:11:10 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sat, 28 Jul 2012 22:11:10 -0400 Subject: [Freeswitch-users] ODBC, FreeTDS, Microsoft SQL In-Reply-To: <804D48104511D4468F0D60DF9D310035095C563F@MAILBOX.millicorp.com> References: <804D48104511D4468F0D60DF9D310035095C33BC@MAILBOX.millicorp.com> <804D48104511D4468F0D60DF9D310035095C563F@MAILBOX.millicorp.com> Message-ID: <0dcb01cd6d2f$6c608dd0$4521a970$@bizfocused.com> For what it's worth, in my past experiences porting either direction Sql Server <==> MySQL the 2 problem areas were Date vs Time vs DSateTime fields and "AutoIncrement" columns (aka surrogate keys). HTH Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim Meade Sent: Saturday, July 28, 2012 3:52 PM To: FreeSWITCH Users Help Cc: Matt Kerper; Chris St. Clair Subject: Re: [Freeswitch-users] ODBC, FreeTDS, Microsoft SQL I've found a work around using the Microsoft SQL drivers instead of FreeTDS. Following most of this: http://msdn.microsoft.com/en-us/library/hh568454.aspx and download here: http://www.microsoft.com/en-us/download/details.aspx?id=28160 Great install instructions: http://blog.nhaslam.com/2011/12/12/sql-server-odbc-on-linux/ and setting the obc.ini to: [freeswitch] Driver = SQL Server Native Client 11.0 Description = FS SQL Tester Trace = No Server = 192.168.5.10 Port = 1433 Database = FSTester All the tables are created properly. Tim From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim Meade Sent: Saturday, July 28, 2012 11:25 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] ODBC, FreeTDS, Microsoft SQL We wanted to run some ODBC tests using a MSSQL backend. We setup the ODBC with MySQL and everything worked great. Then installed FreeTDS and configured it instead of MySQL. When FS restarted, it created some of the required tables. Calls, complete, interfaces, nat, sip_presence, sip_recovery, sip_shared_apperance_subscriptions. What it did not create was any of the registrations tables. When a device tries to register, FS shows in the CLI: 012-07-28 09:43:48.892312 [INFO] switch_odbc.c:285 The connection has been re-established 2012-07-28 09:43:49.912319 [ERR] switch_odbc.c:494 ERR: [create index sr_sip_host on sip_registrations (sip_host)] [STATE: 42000 CODE 1088 ERROR: [unixODBC][FreeTDS][SQL Server]Cannot find the object "sip_registrations" because it does not exist or you do not have permissions. ] 2012-07-28 09:43:49.912319 [ERR] switch_core_sqldb.c:487 SQL ERR [STATE: 42000 CODE 1088 ERROR: [unixODBC][FreeTDS][SQL Server]Cannot find the object "sip_registrations" because it does not exist or you do not have permissions. ] create index sr_sip_host on sip_registrations (sip_host) 2012-07-28 09:43:49.912319 [CRIT] switch_odbc.c:280 The sql server is not responding for DSN freeswitch [STATE: 42000 CODE 1088 ERROR: [unixODBC][FreeTDS][SQL Server]Cannot find the object "sip_registrations" because it does not exist or you do not have permissions. ][244] Seems the issue is that FS didn't create all the appropriate tables on start. Has anyone else seen something like this? I cannot find much on using FS with FreeTDS and MSSQL. Thanks Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120728/483c14cc/attachment.html From sdevoy at bizfocused.com Sun Jul 29 08:05:35 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sun, 29 Jul 2012 00:05:35 -0400 Subject: [Freeswitch-users] SIP to make phone reboot or resync In-Reply-To: References: <07b501cd6c1a$238b3e90$6aa1bbb0$@bizfocused.com> <094e01cd6c43$9e44cc90$dace65b0$@bizfocused.com> <0c4001cd6cdf$b2603a10$1720ae30$@bizfocused.com> Message-ID: <0deb01cd6d3f$67e43a30$37acae90$@bizfocused.com> Thank you sir. That did help, I can resync my phone test either with user/pass using any proxy user/pass from the lines on THAT phone. I can also disable Auth Resync Reboot and let anyone send reboots - which might be fun, but seems like a bad plan! This is not very useful for me since the phone's proxy user/pass must be specified in the gateway XML for whole realm in the the SIP profile. I hoped to be able to re-provision any phone on demand. It seems impractical to have to restart/rescan my sofia gateway every time I want to issue a resync to a different phone. Am I missing something? Sean -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vallimamod ABDULLAH Sent: Saturday, July 28, 2012 1:05 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync Hi, It's normally the sip account credentials. There is also an option in the admin interface to disable this authentication ("Auth Resync-Reboot" option if I recall correctly in Ext tab) Best Regards, Vallimamod Abdullah . On Jul 28, 2012, at 6:40 PM, Sean Devoy wrote: > HI all, > > I am close, but I still don't understand what credentials are required. The response is: > SIP/2.0 401 Unauthorized > > I have tried the web admin credentials for the phone, I don't k now what FS credentials I could pass. > > Any ideas? > > Thanks, > Sean > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony Minessale > Sent: Friday, July 27, 2012 6:12 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync > > Make a gateway with no reg and the credentials and name it after the realm in the challenge. > > On Jul 27, 2012 5:05 PM, "Sean Devoy" wrote: > Thank Anthony, BUT.. Error: > 2012-07-27 17:58:55.118642 [ERR] sofia_reg.c:2165 Cannot locate any > authentication credentials to complete an authentication request for > realm '"fs_bfis.bizfocused.com"' > > 1 - where do I specify the credentials? > 2 - Are these Freeswitch credentials or phone credentials? I suspect > the latter. > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Anthony Minessale > Sent: Friday, July 27, 2012 1:21 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync > > sofia profile check_sync [call_id] // no call_id means > all sofia profile flush_inbound_reg [call_id] [reboot] > > > > On Fri, Jul 27, 2012 at 12:06 PM, Sean Devoy wrote: > > HI All, > > > > > > > > Does anyone have an code that will cause Freeswitch to send a SIP > > message to a CISCO SPA5xx phone that will cause it to reboot or > > resync (aka re-provision)? I know about the URLs that cause the > > phone to do this, but they are NATed, so I need to use SIP to hit them. > > > > > > > > Assume I am programming language omni-lingual for this request! > > > > > > > > Thanks, > > > > Sean > > > > > > ____________________________________________________________________ > > __ ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u > > se > > rs > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users 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 > > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users 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 > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From curriegrad2004 at gmail.com Sun Jul 29 11:58:31 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sun, 29 Jul 2012 00:58:31 -0700 Subject: [Freeswitch-users] Google Inc. DOES use FreeSWITCH for call center telephony routing system. So, why dont' we invite those guys to FS weekly conference :) In-Reply-To: References: Message-ID: Let's invite him over this thread and see what his reactions are like after we jumped the gun on him :) On Sat, Jul 28, 2012 at 1:04 AM, Anton Kvashenkin wrote: > We got response from Jeff Bates, Open Source IT Program Manager @ Google! > > It would be awesome to invite Jeff to FreeSWITCH weekly conference scheldued > on the first august. All we got to do is just ping Jeff via e-mail: batesj > [at] you can guess the company name. > > > 2012/7/26 Gabriel Gunderson >> >> On Wed, Jul 25, 2012 at 12:10 PM, Gabriel Gunderson >> wrote: >> > On Wed, Jul 25, 2012 at 11:46 AM, Anton Kvashenkin >> > wrote: >> >> Guys, according to those slides (*LINK* slide number 10). >> > >> > FYI, it looks like it slide #9. >> >> http://bit.ly/GOOGandFS >> >> That link takes you right to the slide... share it with your Asterisk >> and Avaya buddies ;) >> >> >> Gabe >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From vallimamod.abdullah at imtelecom.fr Sun Jul 29 14:32:48 2012 From: vallimamod.abdullah at imtelecom.fr (Vallimamod ABDULLAH) Date: Sun, 29 Jul 2012 12:32:48 +0200 Subject: [Freeswitch-users] SIP to make phone reboot or resync In-Reply-To: <0deb01cd6d3f$67e43a30$37acae90$@bizfocused.com> References: <07b501cd6c1a$238b3e90$6aa1bbb0$@bizfocused.com> <094e01cd6c43$9e44cc90$dace65b0$@bizfocused.com> <0c4001cd6cdf$b2603a10$1720ae30$@bizfocused.com> <0deb01cd6d3f$67e43a30$37acae90$@bizfocused.com> Message-ID: I don't know if there is a simple way to do this with freeswitch without reload. You can either disable the auth (Remember that as your phone is behind a NAT, only the proxy to which it is registered is able to send him a resync event from outside the lan.) You can also use sipsak to send a hand crafted notify event, it is able to handle authentication with -a option. -- Best Regards, Vallimamod Abdullah . On Jul 29, 2012, at 6:05 AM, Sean Devoy wrote: > Thank you sir. That did help, I can resync my phone test either with > user/pass using any proxy user/pass from the lines on THAT phone. I can > also disable Auth Resync Reboot and let anyone send reboots - which might be > fun, but seems like a bad plan! > > This is not very useful for me since the phone's proxy user/pass must be > specified in the gateway XML for whole realm in the the SIP profile. I > hoped to be able to re-provision any phone on demand. It seems impractical > to have to restart/rescan my sofia gateway every time I want to issue a > resync to a different phone. > > Am I missing something? > > Sean > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Vallimamod ABDULLAH > Sent: Saturday, July 28, 2012 1:05 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync > > Hi, > > It's normally the sip account credentials. There is also an option in the > admin interface to disable this authentication ("Auth Resync-Reboot" option > if I recall correctly in Ext tab) > > Best Regards, > Vallimamod Abdullah > . > > > On Jul 28, 2012, at 6:40 PM, Sean Devoy wrote: > >> HI all, >> >> I am close, but I still don't understand what credentials are required. > The response is: >> SIP/2.0 401 Unauthorized >> >> I have tried the web admin credentials for the phone, I don't k now what > FS credentials I could pass. >> >> Any ideas? >> >> Thanks, >> Sean >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Anthony Minessale >> Sent: Friday, July 27, 2012 6:12 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync >> >> Make a gateway with no reg and the credentials and name it after the realm > in the challenge. >> >> On Jul 27, 2012 5:05 PM, "Sean Devoy" wrote: >> Thank Anthony, BUT.. Error: >> 2012-07-27 17:58:55.118642 [ERR] sofia_reg.c:2165 Cannot locate any >> authentication credentials to complete an authentication request for >> realm '"fs_bfis.bizfocused.com"' >> >> 1 - where do I specify the credentials? >> 2 - Are these Freeswitch credentials or phone credentials? I suspect >> the latter. >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Anthony Minessale >> Sent: Friday, July 27, 2012 1:21 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync >> >> sofia profile check_sync [call_id] // no call_id means >> all sofia profile flush_inbound_reg [call_id] [reboot] >> >> >> >> On Fri, Jul 27, 2012 at 12:06 PM, Sean Devoy > wrote: >>> HI All, >>> >>> >>> >>> Does anyone have an code that will cause Freeswitch to send a SIP >>> message to a CISCO SPA5xx phone that will cause it to reboot or >>> resync (aka re-provision)? I know about the URLs that cause the >>> phone to do this, but they are NATed, so I need to use SIP to hit them. >>> >>> >>> >>> Assume I am programming language omni-lingual for this request! >>> >>> >>> >>> Thanks, >>> >>> Sean >>> >>> >>> ____________________________________________________________________ >>> __ ___ Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u >>> se >>> rs >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users 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 >> >> >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users 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 >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From cogs66 at gmail.com Sun Jul 29 14:33:32 2012 From: cogs66 at gmail.com (cogs66) Date: Sun, 29 Jul 2012 03:33:32 -0700 (PDT) Subject: [Freeswitch-users] No call_uuid in ringing state unix odbc for PostgreSQL Message-ID: <1343558012157-7581267.post@n2.nabble.com> Hello all I use ODBC in the core with PostgreSQL and have an intercept application that looks for the call_uuid when in a ringing state. The issue i have is that only 'null' is displayed not the uuid so the application fails. Once the call has been answered the call_uuid appears as it should. I have tested on other FreeSWITCH installations and everything works great so i know it is to do with this rogue setup. I ran a FreeSWITCH upgrade a few days ago in case this was the issue. Using: Ubuntu 10.04LTS PostgreSQL 9.1 FreeSWITCH Version 1.2.0-rc2+git~20120726T181739Z~696fb9c28b (1.2.0-rc2; git at commit 696fb9c28b on Thu, 26 Jul 2012 18:17:39 Z) I wonder if you are able to shed some light please. Thanks Andy -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/No-call-uuid-in-ringing-state-unix-odbc-for-PostgreSQL-tp7581267.html Sent from the freeswitch-users mailing list archive at Nabble.com. From fieldpeak at gmail.com Sun Jul 29 16:25:33 2012 From: fieldpeak at gmail.com (fieldpeak) Date: Sun, 29 Jul 2012 20:25:33 +0800 Subject: [Freeswitch-users] Help!! FS -TLS interworking issue, How to config to allow "gentls_cert" to generate a root certificate with more longer valid-period ? Message-ID: Hi Masters, I'm testing the TLS on FS to work with softphone. followed the wiki (http://wiki.freeswitch.org/wiki/Tls#EyeBeam.2FBria_Setup ), I generated the CA (root) certificate by below command, however, when i install the root certificate on windows,* it prompt me that the valid period is for only one month*. I tried to change the "DAYS=2190" inside the "gentls_cert" script, but it only effect on server certificate(agent.pem) but not root certificate *(cafile.pem*), Could anyone please help me, appreciated for your any advise!! *./gentls_cert setup -cn fs.audiocodes.com.cn -alt DNS:fs.audiocodes.com.cn-org audiocodes.com.cn* * * *below is from wiki. * *This will create CA certificate and key along with in conf/ssl/CA directory(cacert.pem, cakey.pem) and certificate in the conf/ssl folder(cafile.pem).* * [ Note: The name given for -cn and -alt should be the same as the DNS name of your freeswitch installation and used as the registrar name on the phone (at least on Polycoms). ] You can change the "DAYS=2190" line in the gentls_cert file to make the certificate valid for longer time. However making it too long has some wrap around problem, it appears.* *To short, I want to change the valid period longer for the cafile.pem , thanks!!** * -- Regards, Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120729/1e420a9e/attachment.html From admin at blindi.net Sun Jul 29 16:49:07 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Sun, 29 Jul 2012 14:49:07 +0200 (CEST) Subject: [Freeswitch-users] how can i setup guests incoming calls? In-Reply-To: References: Message-ID: Hi guys, i have setup a fs. my extension is 1000 This extension is accessible by registered users. But unfortunately, not from outside. I call the extension from the world: 1000 at telco01.blindi.net The errormessage is: 2012-07-29 14:43:52.261937 [DEBUG] sofia.c:8071 IP 84.200.210.158 Rejected by ac l "domains". Falling back to Digest auth. 2012-07-29 14:43:52.261937 [WARNING] sofia_reg.c:1474 SIP auth challenge (INVITE ) on sofia profile 'internal' for [1000 at telco01.blindi.net] from ip 84.200.210.1 58 What cahn i do please? I like to reach my extension 1000 from any. Thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From asaad2 at gmail.com Sun Jul 29 17:09:22 2012 From: asaad2 at gmail.com (BookBag) Date: Sun, 29 Jul 2012 09:09:22 -0400 Subject: [Freeswitch-users] how can i setup guests incoming calls? In-Reply-To: References: Message-ID: Seems like you need to edit your acl list to allow direct contact from that ip address. On Sun, Jul 29, 2012 at 8:49 AM, Thomas Hoellriegel wrote: > Hi guys, > i have setup a fs. my extension is 1000 > This extension is accessible by registered users. > But unfortunately, not from outside. > I call the extension from the world: > 1000 at telco01.blindi.net > The errormessage is: > 2012-07-29 14:43:52.261937 [DEBUG] sofia.c:8071 IP 84.200.210.158 Rejected > by ac > l "domains". Falling back to Digest auth. > 2012-07-29 14:43:52.261937 [WARNING] sofia_reg.c:1474 SIP auth challenge > (INVITE > ) on sofia profile 'internal' for [1000 at telco01.blindi.net] from ip > 84.200.210.1 > 58 > > What cahn i do please? > I like to reach my extension 1000 from any. > Thanks. > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120729/2b49fc00/attachment-0001.html From Tim.Meade at Millicorp.com Sun Jul 29 17:36:13 2012 From: Tim.Meade at Millicorp.com (Tim Meade) Date: Sun, 29 Jul 2012 13:36:13 +0000 Subject: [Freeswitch-users] ODBC, FreeTDS, Microsoft SQL In-Reply-To: <0dcb01cd6d2f$6c608dd0$4521a970$@bizfocused.com> References: <804D48104511D4468F0D60DF9D310035095C33BC@MAILBOX.millicorp.com> <804D48104511D4468F0D60DF9D310035095C563F@MAILBOX.millicorp.com> <0dcb01cd6d2f$6c608dd0$4521a970$@bizfocused.com> Message-ID: <804D48104511D4468F0D60DF9D310035095C8514@MAILBOX.millicorp.com> Thanks Sean, With the Microsoft drivers, FS does create all the tables. The registrations still error out. We have shelved it for now. It was just a what if project. Tim From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean Devoy Sent: Saturday, July 28, 2012 10:11 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] ODBC, FreeTDS, Microsoft SQL For what it's worth, in my past experiences porting either direction Sql Server <==> MySQL the 2 problem areas were Date vs Time vs DSateTime fields and "AutoIncrement" columns (aka surrogate keys). HTH Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim Meade Sent: Saturday, July 28, 2012 3:52 PM To: FreeSWITCH Users Help Cc: Matt Kerper; Chris St. Clair Subject: Re: [Freeswitch-users] ODBC, FreeTDS, Microsoft SQL I've found a work around using the Microsoft SQL drivers instead of FreeTDS. Following most of this: http://msdn.microsoft.com/en-us/library/hh568454.aspx and download here: http://www.microsoft.com/en-us/download/details.aspx?id=28160 Great install instructions: http://blog.nhaslam.com/2011/12/12/sql-server-odbc-on-linux/ and setting the obc.ini to: [freeswitch] Driver = SQL Server Native Client 11.0 Description = FS SQL Tester Trace = No Server = 192.168.5.10 Port = 1433 Database = FSTester All the tables are created properly. Tim From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Tim Meade Sent: Saturday, July 28, 2012 11:25 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] ODBC, FreeTDS, Microsoft SQL We wanted to run some ODBC tests using a MSSQL backend. We setup the ODBC with MySQL and everything worked great. Then installed FreeTDS and configured it instead of MySQL. When FS restarted, it created some of the required tables. Calls, complete, interfaces, nat, sip_presence, sip_recovery, sip_shared_apperance_subscriptions. What it did not create was any of the registrations tables. When a device tries to register, FS shows in the CLI: 012-07-28 09:43:48.892312 [INFO] switch_odbc.c:285 The connection has been re-established 2012-07-28 09:43:49.912319 [ERR] switch_odbc.c:494 ERR: [create index sr_sip_host on sip_registrations (sip_host)] [STATE: 42000 CODE 1088 ERROR: [unixODBC][FreeTDS][SQL Server]Cannot find the object "sip_registrations" because it does not exist or you do not have permissions. ] 2012-07-28 09:43:49.912319 [ERR] switch_core_sqldb.c:487 SQL ERR [STATE: 42000 CODE 1088 ERROR: [unixODBC][FreeTDS][SQL Server]Cannot find the object "sip_registrations" because it does not exist or you do not have permissions. ] create index sr_sip_host on sip_registrations (sip_host) 2012-07-28 09:43:49.912319 [CRIT] switch_odbc.c:280 The sql server is not responding for DSN freeswitch [STATE: 42000 CODE 1088 ERROR: [unixODBC][FreeTDS][SQL Server]Cannot find the object "sip_registrations" because it does not exist or you do not have permissions. ][244] Seems the issue is that FS didn't create all the appropriate tables on start. Has anyone else seen something like this? I cannot find much on using FS with FreeTDS and MSSQL. Thanks Tim -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120729/fdb75eea/attachment.html From errotan at elder.hu Sun Jul 29 18:39:02 2012 From: errotan at elder.hu (=?ISO-8859-1?Q?Pusk=E1s_Zsolt?=) Date: Sun, 29 Jul 2012 16:39:02 +0200 Subject: [Freeswitch-users] SIP to make phone reboot or resync In-Reply-To: <07b501cd6c1a$238b3e90$6aa1bbb0$@bizfocused.com> References: <07b501cd6c1a$238b3e90$6aa1bbb0$@bizfocused.com> Message-ID: <50154B06.2030106@elder.hu> You can use the sendevent command to reboot or resync the profile of your phone. http://wiki.freeswitch.org/wiki/Event_Socket#sendevent Example: sendevent NOTIFY profile: internal event-string: resync;profile=http://10.20.30.40/profile.xml user: 1000 host: 10.20.30.40 content-type: application/simple-message-summary 2012-07-27 19:06 keltez?ssel, Sean Devoy ?rta: > > HI All, > > Does anyone have an code that will cause Freeswitch to send a SIP > message to a CISCO SPA5xx phone that will cause it to reboot or resync > (aka re-provision)? I know about the URLs that cause the phone to do > this, but they are NATed, so I need to use SIP to hit them. > > Assume I am programming language omni-lingual for this request! > > Thanks, > > Sean > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120729/bc0dcef4/attachment-0001.html From mi.ke at null.net Sun Jul 29 21:05:08 2012 From: mi.ke at null.net (Mi Ke) Date: Sun, 29 Jul 2012 13:05:08 -0400 Subject: [Freeswitch-users] how to force sofia to use digest auth first? Message-ID: <20120729170508.154600@gmx.com> Hi All, I use the following bridge params to originate call: {sip_auth_username=xxx,sip_auth_password=yyy,effective_caller_id_name=xxx,effective_caller_id_number=xxx}sofia/external/1111 at 1.1.1.1 Since my remote carrier supports only digest authentication, it replies with 401 (Unauthorized) for my first INVITE, and then FS falls back to digest auth and my calls goes OK. Is it possible to change auth type priority for sip profile/globally so digest auth will be used first or disable plain auth completely ? WBR / Mike -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120729/fe38b2fc/attachment.html From avi at avimarcus.net Sun Jul 29 21:39:31 2012 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 29 Jul 2012 20:39:31 +0300 Subject: [Freeswitch-users] how to force sofia to use digest auth first? In-Reply-To: <20120729170508.154600@gmx.com> References: <20120729170508.154600@gmx.com> Message-ID: It's not what you think... part of the 401 unauthorized attempt is that it sends back a nonce key that you use to encrypt your password. That way, SIP has built-in protection against replay attacks even over unencrypted connections. (If someone is watching though, they can try to brute force your passwords based on the nonce key...) -Avi On Sun, Jul 29, 2012 at 8:05 PM, Mi Ke wrote: > Hi All, > > I use the following bridge params to originate call: > > > {sip_auth_username=xxx,sip_auth_password=yyy,effective_caller_id_name=xxx,effective_caller_id_number=xxx}sofia/external/ > 1111 at 1.1.1.1 > > Since my remote carrier supports only digest authentication, it replies > with 401 (Unauthorized) for my first INVITE, and then FS falls back to > digest auth and my calls goes OK. > > Is it possible to change auth type priority for sip profile/globally so > digest auth will be used first or disable plain auth completely ? > > WBR / Mike > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120729/5174e05e/attachment.html From mi.ke at null.net Sun Jul 29 21:59:42 2012 From: mi.ke at null.net (Mi Ke) Date: Sun, 29 Jul 2012 13:59:42 -0400 Subject: [Freeswitch-users] how to force sofia to use digest auth first? Message-ID: <20120729175942.154580@gmx.com> I see... thank you very much for explaining ----- Original Message ----- From: Avi Marcus Sent: 07/29/12 08:39 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] how to force sofia to use digest auth first? It's not what you think... part of the 401 unauthorized attempt is that it sends back a nonce key that you use to encrypt your password. That way, SIP has built-in protection against replay attacks even over unencrypted connections. (If someone is watching though, they can try to brute force your passwords based on the nonce key...) -Avi On Sun, Jul 29, 2012 at 8:05 PM, Mi Ke < mi.ke at null.net > wrote: Hi All, I use the following bridge params to originate call: {sip_auth_username=xxx,sip_auth_password=yyy,effective_caller_id_name=xxx,effective_caller_id_number=xxx}sofia/external/ 1111 at 1.1.1.1 Since my remote carrier supports only digest authentication, it replies with 401 (Unauthorized) for my first INVITE, and then FS falls back to digest auth and my calls goes OK. Is it possible to change auth type priority for sip profile/globally so digest auth will be used first or disable plain auth completely ? WBR / Mike _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120729/41793afe/attachment.html From mitch.capper at gmail.com Sun Jul 29 22:23:00 2012 From: mitch.capper at gmail.com (Mitch Capper) Date: Sun, 29 Jul 2012 11:23:00 -0700 Subject: [Freeswitch-users] Help!! FS -TLS interworking issue, How to config to allow "gentls_cert" to generate a root certificate with more longer valid-period ? In-Reply-To: References: Message-ID: Try running head from GIT and let us know if you still have a problem. ~mitch On Sun, Jul 29, 2012 at 5:25 AM, fieldpeak wrote: > Hi Masters, > > > > I'm testing the TLS on FS to work with softphone. > > > > followed the wiki > (http://wiki.freeswitch.org/wiki/Tls#EyeBeam.2FBria_Setup), > > > > I generated the CA (root) certificate by below command, however, when i > install the root certificate on windows, it prompt me that the valid period > is for only one month. I tried to change the "DAYS=2190" inside the > "gentls_cert" script, but it only effect on server certificate(agent.pem) > but not root certificate (cafile.pem), Could anyone please help me, > appreciated for your any advise!! > > > > ./gentls_cert setup -cn fs.audiocodes.com.cn -alt DNS:fs.audiocodes.com.cn > -org audiocodes.com.cn > > > > below is from wiki. > > This will create CA certificate and key along with in conf/ssl/CA > directory(cacert.pem, cakey.pem) and certificate in the conf/ssl > folder(cafile.pem). > > [ Note: The name given for -cn and -alt should be the same as the DNS name > of your freeswitch installation and used as the registrar name on the phone > (at least on Polycoms). ] You can change the "DAYS=2190" line in the > gentls_cert file to make the certificate valid for longer time. However > making it too long has some wrap around problem, it appears. > > > To short, I want to change the valid period longer for the cafile.pem , > thanks!! > > > -- > Regards, > Charles > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From curriegrad2004 at gmail.com Mon Jul 30 00:55:38 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sun, 29 Jul 2012 13:55:38 -0700 Subject: [Freeswitch-users] Help!! FS -TLS interworking issue, How to config to allow "gentls_cert" to generate a root certificate with more longer valid-period ? In-Reply-To: References: Message-ID: You do realize that if your org does have an existing PKI infrastructure, you can in theory skip all of that. You just need to know the right values to configure and that's about it. (wink, wink, nudge, nudge) :) On Sun, Jul 29, 2012 at 11:23 AM, Mitch Capper wrote: > Try running head from GIT and let us know if you still have a problem. > > ~mitch > > On Sun, Jul 29, 2012 at 5:25 AM, fieldpeak wrote: >> Hi Masters, >> >> >> >> I'm testing the TLS on FS to work with softphone. >> >> >> >> followed the wiki >> (http://wiki.freeswitch.org/wiki/Tls#EyeBeam.2FBria_Setup), >> >> >> >> I generated the CA (root) certificate by below command, however, when i >> install the root certificate on windows, it prompt me that the valid period >> is for only one month. I tried to change the "DAYS=2190" inside the >> "gentls_cert" script, but it only effect on server certificate(agent.pem) >> but not root certificate (cafile.pem), Could anyone please help me, >> appreciated for your any advise!! >> >> >> >> ./gentls_cert setup -cn fs.audiocodes.com.cn -alt DNS:fs.audiocodes.com.cn >> -org audiocodes.com.cn >> >> >> >> below is from wiki. >> >> This will create CA certificate and key along with in conf/ssl/CA >> directory(cacert.pem, cakey.pem) and certificate in the conf/ssl >> folder(cafile.pem). >> >> [ Note: The name given for -cn and -alt should be the same as the DNS name >> of your freeswitch installation and used as the registrar name on the phone >> (at least on Polycoms). ] You can change the "DAYS=2190" line in the >> gentls_cert file to make the certificate valid for longer time. However >> making it too long has some wrap around problem, it appears. >> >> >> To short, I want to change the valid period longer for the cafile.pem , >> thanks!! >> >> >> -- >> Regards, >> Charles >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From prasd.d.b at gmail.com Mon Jul 30 02:25:39 2012 From: prasd.d.b at gmail.com (Prasd D) Date: Sun, 29 Jul 2012 15:25:39 -0700 Subject: [Freeswitch-users] Push server and push capability In-Reply-To: References: <501400AE.8040007@gmail.com> <50140122.40209@gmail.com> Message-ID: Forgot ccs On 7/29/12, Prasd D wrote: > Hi Regis, > > Thanks for throwing light on this. > > I fully agree with you on your first reason, and the initial general > comment. > >> AFAIK, what is done in C2DM is not magic. It should not be impossible to >> reach same thing on a third party app. > > I agree with you that we can have two sockets open. > > This is what I gather from what you put down above and have questions : > > - Presently push has only one socket open. 100% agree with you it > should not rely on a specific push provider > - A 3rd party app or something within csipsimple or a small component > of it will keep a socket open (while rest of csipsimple is shut off > for power purposes) through which it will receive a notification. This > notification will then start csipsimple to take an incoming call for > example. > Advantage of having a 3rd party app is that it can support push > notifications for other purposes also. However then again we are > relying on some other 3rd party push service provider. Instead if it > is done with the same SIP provider (achieved by using Freeswitch > hopefully it supports this as a module soon), then its even better. > I take it that this push service is then applicable only for this and > keeping a socket open just for this is not as optimal. > > > I didn't understand "solve the root cause on the app". I am not sure > if you are referring to "There is indeed known problems to keep the > connection alive on some devices with CSipSimple." when you are > talking about root cause. > But let's say csipsimple works fine when it is running. Then I guess > there is no "root cause" then? > > My main and only concern is saving power and battery and not having to > run csipsimple app running all the time just to receive incoming > calls. > So in this scenario (and no connection alive problems with > csipsimple), there is no "root cause" and the solution is just to use > push service. Is that correct ? > > Alternative is that csipsimple has components or plugins and while all > of it shuts off, one small component keeps running with a socket open > I guess to get simplied notification (rather than complex sip > messages) JUST to tell incoming call, whereafter the main csipsimple > starts. The Freeswitch server then waits for csipsimple to start and > send the invite (I am not sure how it will know that csipsimple woke > up, but I guess the job of the plugin is that. Perhaps it sens a push > notification and then few invites until it gets a response). > > Please clarify above. Would be great if you can provide a solution > together with freeswitch ! > > I think this is the future and as one of the developers indicated in > that jira bug, there are many companies now doing this (like > acrobits). > > > On 7/28/12, R?gis Montoya wrote: >> Hi Prasd, >> >> As I said many times to you about C2DM for invites... I still think >> that's a bad idea. >> >> There is indeed known problems to keep the connection alive on some >> devices with CSipSimple. >> But solving the issue by delegating to another solution hosted by google >> (or to any proprietary network) is not a good solution. If you want to >> develop something you should rather try to help on CSipSimple/pjsip for >> android to find out a reliable way for your devices. >> >> Several reason for that : >> * You become dependant of a push service provider... in this case >> google. Somebody using android should not be forced to use google. -and >> it also doesn't apply to private/enterprise networks-. >> * The C2DM doesn't ensure for reliability. Read the google docs, they >> tell that they don't assure that it will be delivered in time and for >> sure. It's the same for Apple push notification. >> * AFAIK, what is done in C2DM is not magic. It should not be >> impossible to reach same thing on a third party app. >> * Your sip provider becomes dependant of the push notification service >> provider : there is a mechanism to ensure your server is correctly >> allowed to publish on the push service provider (both for C2DM and Apple >> notifications). >> * Your application (in this case the CSipSimple plugin for your C2DM >> deployment), is also linked to the same mechanism and will be linked to >> certificates of the server (so also one app per service...) >> >> The only pro argument would be to have only one socket open, but in my >> opinion it's not a big deal. >> The other not receivable argument is laziness... (even if deployment of >> push notification can lead to headaches). >> >> If you really want to go this way feel free. The api provided by >> CSipSimple to other applications on android should allow you to do what >> you plan a clean and separated way. (I mean by a plugin). >> >> And I think that approach on FS is the same than mine : if you want such >> a thing, it should be addressed by a plugin. >> But my opinion remains the same, it's not a real solution to the >> problem. And I encourage any developer / sip provider facing this >> problem to help to focus their efforts to solve the root cause on the >> app ;). >> >> Best regards, >> R?gis >> > > > -- > Thanks, > Prasd > -- Thanks, Prasd From fieldpeak at gmail.com Mon Jul 30 04:57:03 2012 From: fieldpeak at gmail.com (fieldpeak) Date: Mon, 30 Jul 2012 08:57:03 +0800 Subject: [Freeswitch-users] Help!! FS -TLS interworking issue, How to config to allow "gentls_cert" to generate a root certificate with more longer valid-period ? In-Reply-To: References: Message-ID: Could you please help advise more details? how should i do to implement your advise... i have very few knowledge about pki, certificate and tls, thanks a lot! :) BR,Charles ? 2012-7-30 ??4:57?"curriegrad2004" ??? > You do realize that if your org does have an existing PKI > infrastructure, you can in theory skip all of that. You just need to > know the right values to configure and that's about it. (wink, wink, > nudge, nudge) :) > > On Sun, Jul 29, 2012 at 11:23 AM, Mitch Capper > wrote: > > Try running head from GIT and let us know if you still have a problem. > > > > ~mitch > > > > On Sun, Jul 29, 2012 at 5:25 AM, fieldpeak wrote: > >> Hi Masters, > >> > >> > >> > >> I'm testing the TLS on FS to work with softphone. > >> > >> > >> > >> followed the wiki > >> (http://wiki.freeswitch.org/wiki/Tls#EyeBeam.2FBria_Setup), > >> > >> > >> > >> I generated the CA (root) certificate by below command, however, when i > >> install the root certificate on windows, it prompt me that the valid > period > >> is for only one month. I tried to change the "DAYS=2190" inside the > >> "gentls_cert" script, but it only effect on server > certificate(agent.pem) > >> but not root certificate (cafile.pem), Could anyone please help me, > >> appreciated for your any advise!! > >> > >> > >> > >> ./gentls_cert setup -cn fs.audiocodes.com.cn -alt DNS: > fs.audiocodes.com.cn > >> -org audiocodes.com.cn > >> > >> > >> > >> below is from wiki. > >> > >> This will create CA certificate and key along with in conf/ssl/CA > >> directory(cacert.pem, cakey.pem) and certificate in the conf/ssl > >> folder(cafile.pem). > >> > >> [ Note: The name given for -cn and -alt should be the same as the DNS > name > >> of your freeswitch installation and used as the registrar name on the > phone > >> (at least on Polycoms). ] You can change the "DAYS=2190" line in the > >> gentls_cert file to make the certificate valid for longer time. However > >> making it too long has some wrap around problem, it appears. > >> > >> > >> To short, I want to change the valid period longer for the cafile.pem , > >> thanks!! > >> > >> > >> -- > >> Regards, > >> Charles > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120730/32e259b6/attachment.html From anton.jugatsu at gmail.com Mon Jul 30 08:55:08 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Mon, 30 Jul 2012 08:55:08 +0400 Subject: [Freeswitch-users] how can i setup guests incoming calls? In-Reply-To: References: Message-ID: To receive inbound calls to internal profile (port 5060) you need to use acl for incoming ip address (auth-call=true, apply-inbound-acl=domains) or better way to use external profile (port 5080, auth-calls=false) to receive inbound calls. Just edit public context to match 1000 destination number. 2012/7/29 BookBag > Seems like you need to edit your acl list to allow direct contact from > that ip address. > > On Sun, Jul 29, 2012 at 8:49 AM, Thomas Hoellriegel wrote: > >> Hi guys, >> i have setup a fs. my extension is 1000 >> This extension is accessible by registered users. >> But unfortunately, not from outside. >> I call the extension from the world: >> 1000 at telco01.blindi.net >> The errormessage is: >> 2012-07-29 14:43:52.261937 [DEBUG] sofia.c:8071 IP 84.200.210.158 >> Rejected by ac >> l "domains". Falling back to Digest auth. >> 2012-07-29 14:43:52.261937 [WARNING] sofia_reg.c:1474 SIP auth challenge >> (INVITE >> ) on sofia profile 'internal' for [1000 at telco01.blindi.net] from ip >> 84.200.210.1 >> 58 >> >> What cahn i do please? >> I like to reach my extension 1000 from any. >> Thanks. >> >> --------------- >> Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: >> http://www.blindi.net/callback >> homepage: http://www.blindi.net >> blinde-misc mailingliste f?r blinde. anmeldung unter: >> http://www.blindi.net/mailman/**listinfo/blinde-misc >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120730/d0173522/attachment.html From admin at blindi.net Mon Jul 30 11:06:27 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Mon, 30 Jul 2012 09:06:27 +0200 (CEST) Subject: [Freeswitch-users] how can i setup guests incoming calls? In-Reply-To: References: Message-ID: Hi BookBag Thanks for your help, the extension works --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From oseslija at gmail.com Mon Jul 30 11:08:47 2012 From: oseslija at gmail.com (Ognjen Seslija) Date: Mon, 30 Jul 2012 09:08:47 +0200 Subject: [Freeswitch-users] SIP to make phone reboot or resync In-Reply-To: <0deb01cd6d3f$67e43a30$37acae90$@bizfocused.com> References: <07b501cd6c1a$238b3e90$6aa1bbb0$@bizfocused.com> <094e01cd6c43$9e44cc90$dace65b0$@bizfocused.com> <0c4001cd6cdf$b2603a10$1720ae30$@bizfocused.com> <0deb01cd6d3f$67e43a30$37acae90$@bizfocused.com> Message-ID: Just disable auth for resync/reboot on the phone. It's proprietary in any way, will work only with SPA8000 pbx afaik. On Sun, Jul 29, 2012 at 6:05 AM, Sean Devoy wrote: > Thank you sir. That did help, I can resync my phone test either with > user/pass using any proxy user/pass from the lines on THAT phone. I can > also disable Auth Resync Reboot and let anyone send reboots - which might > be > fun, but seems like a bad plan! > > This is not very useful for me since the phone's proxy user/pass must be > specified in the gateway XML for whole realm in the the SIP profile. I > hoped to be able to re-provision any phone on demand. It seems impractical > to have to restart/rescan my sofia gateway every time I want to issue a > resync to a different phone. > > Am I missing something? > > Sean > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Vallimamod ABDULLAH > Sent: Saturday, July 28, 2012 1:05 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync > > Hi, > > It's normally the sip account credentials. There is also an option in the > admin interface to disable this authentication ("Auth Resync-Reboot" option > if I recall correctly in Ext tab) > > Best Regards, > Vallimamod Abdullah > . > > > On Jul 28, 2012, at 6:40 PM, Sean Devoy wrote: > > > HI all, > > > > I am close, but I still don't understand what credentials are required. > The response is: > > SIP/2.0 401 Unauthorized > > > > I have tried the web admin credentials for the phone, I don't k now what > FS credentials I could pass. > > > > Any ideas? > > > > Thanks, > > Sean > > > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > Anthony Minessale > > Sent: Friday, July 27, 2012 6:12 PM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync > > > > Make a gateway with no reg and the credentials and name it after the > realm > in the challenge. > > > > On Jul 27, 2012 5:05 PM, "Sean Devoy" wrote: > > Thank Anthony, BUT.. Error: > > 2012-07-27 17:58:55.118642 [ERR] sofia_reg.c:2165 Cannot locate any > > authentication credentials to complete an authentication request for > > realm '"fs_bfis.bizfocused.com"' > > > > 1 - where do I specify the credentials? > > 2 - Are these Freeswitch credentials or phone credentials? I suspect > > the latter. > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > Anthony Minessale > > Sent: Friday, July 27, 2012 1:21 PM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync > > > > sofia profile check_sync [call_id] // no call_id means > > all sofia profile flush_inbound_reg [call_id] [reboot] > > > > > > > > On Fri, Jul 27, 2012 at 12:06 PM, Sean Devoy > wrote: > > > HI All, > > > > > > > > > > > > Does anyone have an code that will cause Freeswitch to send a SIP > > > message to a CISCO SPA5xx phone that will cause it to reboot or > > > resync (aka re-provision)? I know about the URLs that cause the > > > phone to do this, but they are NATed, so I need to use SIP to hit them. > > > > > > > > > > > > Assume I am programming language omni-lingual for this request! > > > > > > > > > > > > Thanks, > > > > > > Sean > > > > > > > > > ____________________________________________________________________ > > > __ ___ Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u > > > se > > > rs > > > http://www.freeswitch.org > > > > > > > > > > > -- > > Anthony Minessale II > > > > FreeSWITCH http://www.freeswitch.org/ > > ClueCon http://www.cluecon.com/ > > Twitter: http://twitter.com/FreeSWITCH_wire > > > > AIM: anthm > > MSN:anthony_minessale at hotmail.com > > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > > IRC: irc.freenode.net #freeswitch > > > > FreeSWITCH Developer Conference > > sip:888 at conference.freeswitch.org > > googletalk:conf+888 at conference.freeswitch.org > > pstn:+19193869900 > > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users 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 > > > > > > > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users 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 > > ______________________________________________________________________ > > ___ Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120730/e42afd76/attachment-0001.html From oseslija at gmail.com Mon Jul 30 11:09:32 2012 From: oseslija at gmail.com (Ognjen Seslija) Date: Mon, 30 Jul 2012 09:09:32 +0200 Subject: [Freeswitch-users] SIP to make phone reboot or resync In-Reply-To: References: <07b501cd6c1a$238b3e90$6aa1bbb0$@bizfocused.com> <094e01cd6c43$9e44cc90$dace65b0$@bizfocused.com> <0c4001cd6cdf$b2603a10$1720ae30$@bizfocused.com> <0deb01cd6d3f$67e43a30$37acae90$@bizfocused.com> Message-ID: Err SPA9000...I always mix the two On Mon, Jul 30, 2012 at 9:08 AM, Ognjen Seslija wrote: > Just disable auth for resync/reboot on the phone. It's proprietary in any > way, will work only with SPA8000 pbx afaik. > > > On Sun, Jul 29, 2012 at 6:05 AM, Sean Devoy wrote: > >> Thank you sir. That did help, I can resync my phone test either with >> user/pass using any proxy user/pass from the lines on THAT phone. I can >> also disable Auth Resync Reboot and let anyone send reboots - which might >> be >> fun, but seems like a bad plan! >> >> This is not very useful for me since the phone's proxy user/pass must be >> specified in the gateway XML for whole realm in the the SIP profile. I >> hoped to be able to re-provision any phone on demand. It seems impractical >> to have to restart/rescan my sofia gateway every time I want to issue a >> resync to a different phone. >> >> Am I missing something? >> >> Sean >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Vallimamod ABDULLAH >> Sent: Saturday, July 28, 2012 1:05 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync >> >> Hi, >> >> It's normally the sip account credentials. There is also an option in the >> admin interface to disable this authentication ("Auth Resync-Reboot" >> option >> if I recall correctly in Ext tab) >> >> Best Regards, >> Vallimamod Abdullah >> . >> >> >> On Jul 28, 2012, at 6:40 PM, Sean Devoy wrote: >> >> > HI all, >> > >> > I am close, but I still don't understand what credentials are required. >> The response is: >> > SIP/2.0 401 Unauthorized >> > >> > I have tried the web admin credentials for the phone, I don't k now what >> FS credentials I could pass. >> > >> > Any ideas? >> > >> > Thanks, >> > Sean >> > >> > From: freeswitch-users-bounces at lists.freeswitch.org >> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> > Anthony Minessale >> > Sent: Friday, July 27, 2012 6:12 PM >> > To: FreeSWITCH Users Help >> > Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync >> > >> > Make a gateway with no reg and the credentials and name it after the >> realm >> in the challenge. >> > >> > On Jul 27, 2012 5:05 PM, "Sean Devoy" wrote: >> > Thank Anthony, BUT.. Error: >> > 2012-07-27 17:58:55.118642 [ERR] sofia_reg.c:2165 Cannot locate any >> > authentication credentials to complete an authentication request for >> > realm '"fs_bfis.bizfocused.com"' >> > >> > 1 - where do I specify the credentials? >> > 2 - Are these Freeswitch credentials or phone credentials? I suspect >> > the latter. >> > >> > -----Original Message----- >> > From: freeswitch-users-bounces at lists.freeswitch.org >> > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> > Anthony Minessale >> > Sent: Friday, July 27, 2012 1:21 PM >> > To: FreeSWITCH Users Help >> > Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync >> > >> > sofia profile check_sync [call_id] // no call_id means >> > all sofia profile flush_inbound_reg [call_id] [reboot] >> > >> > >> > >> > On Fri, Jul 27, 2012 at 12:06 PM, Sean Devoy >> wrote: >> > > HI All, >> > > >> > > >> > > >> > > Does anyone have an code that will cause Freeswitch to send a SIP >> > > message to a CISCO SPA5xx phone that will cause it to reboot or >> > > resync (aka re-provision)? I know about the URLs that cause the >> > > phone to do this, but they are NATed, so I need to use SIP to hit >> them. >> > > >> > > >> > > >> > > Assume I am programming language omni-lingual for this request! >> > > >> > > >> > > >> > > Thanks, >> > > >> > > Sean >> > > >> > > >> > > ____________________________________________________________________ >> > > __ ___ Professional FreeSWITCH Consulting Services: >> > > consulting at freeswitch.org >> > > http://www.freeswitchsolutions.com >> > > >> > > >> > > >> > > >> > > Official FreeSWITCH Sites >> > > http://www.freeswitch.org >> > > http://wiki.freeswitch.org >> > > http://www.cluecon.com >> > > >> > > Join Us At ClueCon - Aug 7-9, 2012 >> > > >> > > FreeSWITCH-users mailing list >> > > FreeSWITCH-users at lists.freeswitch.org >> > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u >> > > se >> > > rs >> > > http://www.freeswitch.org >> > > >> > >> > >> > >> > -- >> > Anthony Minessale II >> > >> > FreeSWITCH http://www.freeswitch.org/ >> > ClueCon http://www.cluecon.com/ >> > Twitter: http://twitter.com/FreeSWITCH_wire >> > >> > AIM: anthm >> > MSN:anthony_minessale at hotmail.com >> > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> > IRC: irc.freenode.net #freeswitch >> > >> > FreeSWITCH Developer Conference >> > sip:888 at conference.freeswitch.org >> > googletalk:conf+888 at conference.freeswitch.org >> > pstn:+19193869900 >> > >> > ______________________________________________________________________ >> > ___ Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users 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 >> > >> > >> > >> > ______________________________________________________________________ >> > ___ Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users 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 >> > ______________________________________________________________________ >> > ___ Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20120730/980b3ab4/attachment.html From admin at blindi.net Mon Jul 30 11:22:07 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Mon, 30 Jul 2012 09:22:07 +0200 (CEST) Subject: [Freeswitch-users] Question how can i get variables in lua from xml dialplan? In-Reply-To: References: Message-ID: Hi guys, I like to import variables from xml dialplan in lua. I found: session:getVariable. the problem: i can only get variables from the lua wiki page. But different variables for example: strftime or from: vars.xml : sound_prefix can.t be get. Or timeconditions don.t working in lua. Can your help please? thanks --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From dujinfang at gmail.com Mon Jul 30 11:39:56 2012 From: dujinfang at gmail.com (Seven Du) Date: Mon, 30 Jul 2012 15:39:56 +0800 Subject: [Freeswitch-users] Question how can i get variables in lua from xml dialplan? In-Reply-To: References: Message-ID: try variable_sound_prefix http://wiki.freeswitch.org/wiki/Channel_Variables On Monday, July 30, 2012 at 3:22 PM, Thomas Hoellriegel wrote: > Hi guys, > I like to import variables from xml dialplan in lua. > I found: > session:getVariable. > the problem: > i can only get variables from the lua wiki page. > But different variables for example: > strftime > or from: vars.xml : sound_prefix can.t be get. > > Or timeconditions don.t working in lua. > > Can your help please? > thanks > > > > > > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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/20120730/5cbc7a90/attachment-0001.html From chrisbware at yahoo.it Mon Jul 30 14:51:51 2012 From: chrisbware at yahoo.it (Chris B. Ware) Date: Mon, 30 Jul 2012 11:51:51 +0100 (BST) Subject: [Freeswitch-users] Fifo importance Message-ID: <1343645511.18520.YahooMailNeo@web132302.mail.ird.yahoo.com> Hi, Just another question on fifos (thanks for your answers on 'reparse'): what's the meaning of ?'importance' parameter on fifo configuration? The same question is on wiki but with no answers. Thanks, Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/be1e53d8/attachment.html From alex at thewinelake.com Mon Jul 30 16:52:18 2012 From: alex at thewinelake.com (Alex) Date: Mon, 30 Jul 2012 13:52:18 +0100 Subject: [Freeswitch-users] Sending calls to multiple destinations (with and without Enterprise Originate) In-Reply-To: References: <000c01cd6b78$0c5f4470$251dcd50$@com> Message-ID: <50168382.8020600@thewinelake.com> I've been having problems with some aspects of Enterprise Origin (notably call_leg_delay) and am wondering if I need to be using EO at all. Trouble is that even pretty simple multi-destination dialplans just don't seem to be working for me with multiple SIP registrations. eg. The idea here is that the user can have a SIP phone as well as a PSTN phone. The above script works OK with a single SIP registration against user 0095301. But when one tries to connect more than one SIP handset, It doesn't work (and the warning "[WARNING] switch_ivr_originate.c:2351 Only calling the first element in the list in this mode." comes up). (However, the pstn phone does kick in after 10s, which is nice). That's why we went for Enterprise Originate - which works well for multiple SIP registrations, except for the fact that we no longer have control over delayed leg-starting. Now I'm wondering what that warning is all about - eg. what is "this mode"??? Just in case you're wondering - we do have the line : in the /conf/sip_profiles/internal.xml (Although I confess to not quite understanding why it's using a profile called "internal" for this) *** Question: What does "this mode" mean? A variant on this is to not use multiple SIP registrations to the same user, but actually create another user (so we have a and b): This is close to achieving our aim, although it would be better if it could also support multiple SIP regsitrations. From gerald.weber at besharp.at Mon Jul 30 16:56:53 2012 From: gerald.weber at besharp.at (Gerald Weber) Date: Mon, 30 Jul 2012 12:56:53 +0000 Subject: [Freeswitch-users] sofia_contact result Message-ID: Hi, can i rely on the structure of the result string i get back from e.g.: freeswitch at default> sofia_contact 2000 (or sofia_contact user/2000) sofia/internal/sip:2000 at 192.168.20.219:5060 thanks gw -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/16374e9a/attachment.html From anthony.minessale at gmail.com Mon Jul 30 17:10:33 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Jul 2012 08:10:33 -0500 Subject: [Freeswitch-users] sofia_contact result In-Reply-To: References: Message-ID: Depends on what you mean by rely? On Mon, Jul 30, 2012 at 7:56 AM, Gerald Weber wrote: > Hi, > > > > can i rely on the structure of the result string i get back from e.g.: > > > > freeswitch at default> sofia_contact 2000 (or sofia_contact user/2000) > > sofia/internal/sip:2000 at 192.168.20.219:5060 > > > > thanks > > gw > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From anthony.minessale at gmail.com Mon Jul 30 17:22:09 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Jul 2012 08:22:09 -0500 Subject: [Freeswitch-users] LUA session:Bridge not actually bridging calls ~ In-Reply-To: References: Message-ID: This script is actually a bit over-complicated dialA = "sofia/gateway/fs1/9903" dialB = "user/1001" legA = freeswitch.Session(dialA) if (legA:ready()) { legA:execute("bridge", dialB) } I think the problem is that we don't support dealing with many codecs in the 200ok They send gsm and ulaw and then choose ulaw and we go with the first one gsm On Sat, Jul 28, 2012 at 11:23 AM, SamyGo wrote: > Hi again, > > So, It was a very minor change in configuration and it was working. > Basically FreeSwicth was bridging the two legs BUT there was a codec issue. > All I had to do was in Asterisk (serving as my gateway) to allow only ulaw > and alaw > i.e > > disallow=all > allow=ulaw > allow=alaw > > What I really really wish to know is that why there was no indication of > codec mismatch or ptime mismatch or sample rate mismatch while transcoding > or anything. > > It will be fine if none replies but it will be great to know the real > reason behind this and from where in logs can I verify this !! > > Thanks > Sammy > > > On Sat, Jul 28, 2012 at 8:18 PM, SamyGo wrote: >> >> Here are the FS console logs: >> http://pastebin.freeswitch.org/19595 >> >> Please suggest what am I missing here. >> >> >> On Sat, Jul 28, 2012 at 7:59 PM, SamyGo wrote: >>> >>> Hello, >>> I wanted to make a lua script which just dials out two different numbers >>> via some external gateway and when both calls are answered they are just >>> bridged. For this a very impressive Lua example >>> http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control is copied and >>> all I had to do was change the dialA and dialB strings and its working great >>> as far as the SIP signalling is concerned. >>> >>> execute this string and I get calls on two different number but things >>> get interesting when Freeswitch bridge() the two legs. No AUDIO..not even >>> one-way. I could see on my own gateway that RTPs for both the legs are >>> actually forwarded to Freeswitch ! >>> >>> On my sip pcap traces analyzing on wireshark I could actually hear the >>> two persons saying Hello but neither could hear anything. >>> >>> The above example lua call_control script is used as it is. >>> Please suggest. >>> >>> Regards >>> Sammy Go. >>> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From gerald.weber at besharp.at Mon Jul 30 17:25:48 2012 From: gerald.weber at besharp.at (Gerald Weber) Date: Mon, 30 Jul 2012 13:25:48 +0000 Subject: [Freeswitch-users] sofia_contact result In-Reply-To: References: Message-ID: With rely i mean, ist he host/ip always between '@' and ':' or are there any other combinations ? sofia//sip:@: Background: I'm currenlty working on mod_snom to implement the KEYPRESS feature. To do so, i need the ip of the user, 2 choices came on my mind: 1. use sofia_contact as suggested by Brian Foster and parse the output. 2. do a select network_ip from registrations where reg_user = 'xxxx' sofia_contact looks better because i can pass the whole contact string from mod_callcenter using switch_api_execute without all the fiddling with domain, user, etc. Thats why i'm interested in the structure. ps: I created a feature request at snom to support notify talk last november, but still no response :( -----Urspr?ngliche Nachricht----- Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale Gesendet: Montag, 30. Juli 2012 15:11 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] sofia_contact result Depends on what you mean by rely? On Mon, Jul 30, 2012 at 7:56 AM, Gerald Weber wrote: > Hi, > > > > can i rely on the structure of the result string i get back from e.g.: > > > > freeswitch at default> sofia_contact 2000 (or sofia_contact user/2000) > > sofia/internal/sip:2000 at 192.168.20.219:5060 > > > > thanks > > gw > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users 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 > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From andrew.paul85 at gmail.com Mon Jul 30 18:01:34 2012 From: andrew.paul85 at gmail.com (Andrew Paul) Date: Mon, 30 Jul 2012 19:31:34 +0530 Subject: [Freeswitch-users] CPU Usage Message-ID: Hai all, How can i reduce cpu high usage in freeswitch. I am using freeswitch version of 1.2.0. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/85d29acf/attachment.html From anthony.minessale at gmail.com Mon Jul 30 18:26:26 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Jul 2012 09:26:26 -0500 Subject: [Freeswitch-users] H264 transcoding In-Reply-To: References: <5E0B0409-B5A4-4722-ACDA-8F6140EE49C7@freeswitch.org> <5012387C.8050903@quentustech.com> Message-ID: cool On Fri, Jul 27, 2012 at 8:10 PM, Seven Du wrote: > I patched libvlc and it can now decode any video vlc supported and encode > with x264 and send to any sip phone. Specifically I'm trying to use it to > play a 1080p stream and resize to CIF or D1 so a video phone will > accept(Sending 1080p will cause some phones to reboot :( ). > > Still need a lot of code to make it working neatly, however, I might can do > a demo on ClueCon and @William if you'd like to review the code and merge > into tree I'll happy to contribute that later. > > On Friday, July 27, 2012 at 2:43 PM, William King wrote: > > libvlc is LGPL http://www.videolan.org/press/lgpl.html and there is now a > mod_vlc(though it doesn't yet support video streams). The user can choose to > build vlc with only the LGPL components or add the more 'adverse' modules. > In none of the LGPL packages of libvlc is ffmpeg enabled, but there is a > module for libvlc for ffmpeg. http://wiki.videolan.org/FFmpeg > > The only pieces now may just be the FS side of things for video. > > William King > Senior Engineer > Quentus Technologies, INC > 1037 NE 65th St Suite 273 > Seattle, WA 98115 > Main: (877) 211-9337 > Office: (206) 388-4772 > Cell: (253) 686-5518 > william.king at quentustech.com > > > On 07/26/2012 08:46 PM, Anthony Minessale wrote: > > you would probably need to do something like make a mod for ffmpeg > that protects you from the gpl then allow the user to build that lib > on his own and choose at compile time to install patented or adverse > licensed components. No license rules prohibit an end user from > combining code only distributors. > > but even then we need a bunch of code to write. > > > On Thu, Jul 26, 2012 at 10:41 PM, curriegrad2004 > wrote: > > Ken, > > If you think those guys over at x264 will ever change the license from > GPL to LGPL, you're just dreaming the pie in the sky... > > In short, don't even think about it ;P > > On Thu, Jul 26, 2012 at 8:19 PM, Ken Rice wrote: > > we can not and will not use GPL software, the license is not compatible with > the GPL and would polute the codebase with additional restrictions that are > not wanted or needed. now if someone could get them to change the license or > atleast give us a license under better terms such as the LGPL or the MPL > then the license issue would be null > > Ken > Sent from my iPad > > On Jul 26, 2012, at 7:45 PM, Terry Barnum wrote: > > Use x264? http://en.wikipedia.org/wiki/X264 > > On Jul 26, 2012, at 4:53 PM, Ken Rice wrote: > > Is it possible sure... Is ot probably to happen anytime soon? Not until the > patents run out... > > > On 7/26/12 5:04 PM, "yufei.tao" wrote: > > Hi > > I am trying to decide if it is feasible to let FS do transcoding between > different H264 formats for live video calls. This is because I've got > SIP clients that both use H264 but with different formats and one (with > a bad H264 decoder) has problems decoding H264 stream from the other. > But each of these two clients communicate fine using H264 with a third > client that uses ffmpeg. I'm thinking of adding a module which uses > ffmpeg, so that it will transcode H264 between different parameters. > > I've got a few questions: > > 1. Is this feasible? I'm not looking at supporting many simultaneous calls. > 2. What is involved in transcoding real-time video stream? > 3. Anyone's done anything like this before? > > I'm new to FS and any suggestions would be very much appreciated! > > Yufei > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From anton.jugatsu at gmail.com Mon Jul 30 18:28:42 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Mon, 30 Jul 2012 18:28:42 +0400 Subject: [Freeswitch-users] CPU Usage In-Reply-To: References: Message-ID: http://wiki.freeswitch.org/wiki/Reporting_Bugs#CPU_Usage 2012/7/30 Andrew Paul > Hai all, > How can i reduce cpu high usage in freeswitch. I am using freeswitch > version of 1.2.0. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120730/49994ee8/attachment.html From alex at thewinelake.com Mon Jul 30 18:33:32 2012 From: alex at thewinelake.com (Alex) Date: Mon, 30 Jul 2012 15:33:32 +0100 Subject: [Freeswitch-users] Sending calls to multiple destinations (with and without Enterprise Originate) In-Reply-To: <50168382.8020600@thewinelake.com> References: <000c01cd6b78$0c5f4470$251dcd50$@com> <50168382.8020600@thewinelake.com> Message-ID: <50169B3C.1030108@thewinelake.com> This issue is now closed (the problem is with "user/0095301@${domain_name}" - if that is changed to "${sofia_contact(0095301)}" then all is well). Thanks to Anthony for that one.... > I've been having problems with some aspects of Enterprise Origin > (notably call_leg_delay) and am wondering if I need to be using EO at all. > > Trouble is that even pretty simple multi-destination dialplans just > don't seem to be working for me with multiple SIP registrations. > > eg. > > > > > > > > > data="leg_common=sess_ref=${uuid},tenant_id=0095,b_ext=301,group_confirm_file=lua > confirmcall.lua,group_confirm_key=exec,whisper_msg=/home/pabx/004-0095/20120730-100551-00001/recordings/tts/301_whisper.wav,accept_msg=/home/pabx/004-0095/20120730-100551-00001/recordings/tts/accept1.wav"/> > data="leg_sip=[accept_mode=Direct,sip_h_sb_routing=${sb_routing}&441223900481_301_pers_Fred,origination_caller_id_name=WizzCo > x301 Fred,leg_timeout=60]user/0095301@${domain_name}"/> > data="leg_pstn=[leg_delay_start=10,accept_mode=Direct,leg_timeout=60,origination_caller_id_number=00953010${ani}]sofia/internal/898000000000202074904992 at x.y.z.a"/> > data="<${leg_common}>${leg_sip},${leg_pstn}"/> > > > > > The idea here is that the user can have a SIP phone as well as a PSTN > phone. > > The above script works OK with a single SIP registration against user > 0095301. > > But when one tries to connect more than one SIP handset, It doesn't work > (and the warning "[WARNING] switch_ivr_originate.c:2351 Only calling the > first element in the list in this mode." comes up). > (However, the pstn phone does kick in after 10s, which is nice). > > That's why we went for Enterprise Originate - which works well for > multiple SIP registrations, except for the fact that we no longer have > control over delayed leg-starting. > > Now I'm wondering what that warning is all about - eg. what is "this > mode"??? > Just in case you're wondering - we do have the line : name="multiple-registrations" value="true"/> in the > /conf/sip_profiles/internal.xml > (Although I confess to not quite understanding why it's using a profile > called "internal" for this) > > *** Question: What does "this mode" mean? > > > A variant on this is to not use multiple SIP registrations to the same > user, but actually create another user (so we have a and b): > > data="leg_sipa=[accept_mode=Direct,sip_h_sb_routing=${sb_routing}&441223900481_301_pers_Fred,leg_timeout=60]user/0095301a@${domain_name}"/> > data="leg_sipb=[accept_mode=Direct,sip_h_sb_routing=${sb_routing}&441223900481_301_pers_Fred,leg_timeout=60]user/0095301b@${domain_name}"/> > data="leg_pstn=[leg_delay_start=10,accept_mode=Direct,leg_timeout=60,origination_caller_id_number=00953010${ani}]sofia/internal/898000000000202074904992 at x.y.z.a"/> > data="<${leg_common}>${leg_sipa},${leg_sipb},${leg_pstn}"/> > > This is close to achieving our aim, although it would be better if it > could also support multiple SIP regsitrations. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ----- > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2196 / Virus Database: 2437/5143 - Release Date: 07/20/12 > Internal Virus Database is out of date. > > From bdfoster at endigotech.com Mon Jul 30 18:41:16 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 30 Jul 2012 10:41:16 -0400 Subject: [Freeswitch-users] CPU Usage In-Reply-To: References: Message-ID: Also, as statated on the link referenced, this is best diagnosed in real time on IRC (irc.freenode.net/#freeswitch). If this turns out to be a bug, please use JIRA (jira.freeswitch.org). Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 30, 2012 10:03 AM, "Andrew Paul" wrote: > Hai all, > How can i reduce cpu high usage in freeswitch. I am using freeswitch > version of 1.2.0. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120730/0c52eb5e/attachment.html From g.d.monnezza at tiscali.it Mon Jul 30 13:13:33 2012 From: g.d.monnezza at tiscali.it (g) Date: Mon, 30 Jul 2012 11:13:33 +0200 Subject: [Freeswitch-users] Bind Sofia external profile to Static private IP In-Reply-To: References: <4664115619900296997@unknownmsgid> Message-ID: <201207301113.33629.g.d.monnezza@tiscali.it> Hi Brian. Try playing with ext-rtp-ip directive. i,e. force in your sip profile Or, if you prefer, start Freeswitch with -nonat option. A more sophisticated approach is to play with auto-nat. The autoNAT engine in Freeswitch is powerful. It works with uPnP, like skype does. So, working with most of the internet gateways (like home adsl routers) it is capable to open a port to communicate on the public network. Hope it helps g On Saturday 28 July 2012 16:34:37 SamyGo wrote: > yes I tried that too edited the profile external.xml directly but in > vain..so I was reading another thread here of a person who had totally > inverse issue and Michael asked if he is starting freeswitch with -nonat , > applying that switch while starting freeswitch seems to have worked here. > > On Sat, Jul 28, 2012 at 6:34 PM, Brian West wrote: > > Open the profile directly and edit it, vars.xml is for ease not > > flexibility. > > > > -- > > Brian West > > brian at freeswitch.org > > FreeSWITCH Solutions, LLC > > PO BOX PO BOX 2531 > > Brookfield, WI 53008-2531 > > Twitter: @FreeSWITCH_Wire > > T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST > > iNUM: +883 5100 1286 0410 > > > > On Jul 28, 2012, at 8:31 AM, SamyGo wrote: > > > Hello, > > > Probably an easy thing but my luck I'm stuck in this awkward situation > > > > here, I want my sofia external profile to bind to my private IP: > > 192.168.15.21 , so for the purpose I've edited the vars.xml and > > sip_profiles/external.xml but still everytime I restart freeswitch it > > detects the Public IP and binds thye external-sip-ip and external-rtp-ip > > to the Public IP ? > > > > > I'm very confused at this and followed all the previous mailing lists > > > > threads and to-dos as well but in vain. > > > > > Waiting for a savior here. > > > -- > > > Thanks > > > Sammy Go. > > > > > > > > > _______________________________________________________________________ > > > __ Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-user > > > s http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org From yufei.tao at redembedded.com Mon Jul 30 16:33:21 2012 From: yufei.tao at redembedded.com (Yufei Tao) Date: Mon, 30 Jul 2012 13:33:21 +0100 Subject: [Freeswitch-users] H264 transcoding In-Reply-To: References: Message-ID: <50167F11.9010808@redembedded.com> Hi Seven Thanks very much for your replies! I thought I should be very surprised if I was the first one wanting to do this. It'd be very helpful if you could share your code with x264, or libvlc, to spare me, and possibly others of reinventing the wheels. I'll compile the code myself of course which should be free of the licensing troubles I guess. I'd be very grateful if you could share the code and give some instructions. Thank you very much! Yufei Subject: Re: [Freeswitch-users] H264 transcodin From: Seven Du Date: 28/07/12 01:45 To: FreeSWITCH Users Help Real-time encoding with statically linked x264 lib works fine for me from QCIF to D1 resolution, 720p is slow and discarding frames on a Xeon Quad core CPU. I haven't look how to use the GPU, or if possible. It is working in my lab and I have the same question with Yufei Tao when going to production or deliver to customer. Based on http://lists.freeswitch.org/pipermail/freeswitch-dev/2010-September/004227.html , In my understanding, compile and link and use by my self should be fine and, if I deliver to a customer, it should be fine if I provide the code and help the customer to compile on their own server? I'd like to open source the code to public later, but, I'd like to know is it a MUST or MAY? If you pipe to ffmpeg or x264 command line it's not been treated as combine into a large work, and I'm not sure if realtime transcoding will be smooth. 7. On Friday, July 27, 2012 at 11:17 PM, Yufei Tao wrote: Thanks everyone for the responses! If I understand it correctly, if I installed ffmpeg on itself separately from FS, I could write a module for FS, in which I just call the ffmpeg program by running a command line. This way would it be classified as "not combine them into a larger work", thus free from license incompatibility problem? Not sure if that'll work for real-time transcoding of x264 though? Thanks very much for you opinions! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ForwardedMessage.eml Subject: Re: [Freeswitch-users] H264 transcoding From: Seven Du Date: 28/07/12 02:10 To: FreeSWITCH Users Help I patched libvlc and it can now decode any video vlc supported and encode with x264 and send to any sip phone. Specifically I'm trying to use it to play a 1080p stream and resize to CIF or D1 so a video phone will accept(Sending 1080p will cause some phones to reboot :( ). Still need a lot of code to make it working neatly, however, I might can do a demo on ClueCon and @William if you'd like to review the code and merge into tree I'll happy to contribute that later. On Friday, July 27, 2012 at 2:43 PM, William King wrote: libvlc is LGPL http://www.videolan.org/press/lgpl.html and there is now a mod_vlc(though it doesn't yet support video streams). The user can choose to build vlc with only the LGPL components or add the more 'adverse' modules. In none of the LGPL packages of libvlc is ffmpeg enabled, but there is a module for libvlc for ffmpeg. http://wiki.videolan.org/FFmpeg The only pieces now may just be the FS side of things for video. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 07/26/2012 08:46 PM, Anthony Minessale wrote: you would probably need to do something like make a mod for ffmpeg that protects you from the gpl then allow the user to build that lib on his own and choose at compile time to install patented or adverse licensed components. No license rules prohibit an end user from combining code only distributors. but even then we need a bunch of code to write. On Thu, Jul 26, 2012 at 10:41 PM, curriegrad2004 wrote: Ken, If you think those guys over at x264 will ever change the license from GPL to LGPL, you're just dreaming the pie in the sky... In short, don't even think about it ;P On Thu, Jul 26, 2012 at 8:19 PM, Ken Rice wrote: we can not and will not use GPL software, the license is not compatible with the GPL and would polute the codebase with additional restrictions that are not wanted or needed. now if someone could get them to change the license or atleast give us a license under better terms such as the LGPL or the MPL then the license issue would be null Ken Sent from my iPad On Jul 26, 2012, at 7:45 PM, Terry Barnum wrote: Use x264? http://en.wikipedia.org/wiki/X264 On Jul 26, 2012, at 4:53 PM, Ken Rice wrote: Is it possible sure... Is ot probably to happen anytime soon? Not until the patents run out... On 7/26/12 5:04 PM, "yufei.tao" wrote: Hi I am trying to decide if it is feasible to let FS do transcoding between different H264 formats for live video calls. This is because I've got SIP clients that both use H264 but with different formats and one (with a bad H264 decoder) has problems decoding H264 stream from the other. But each of these two clients communicate fine using H264 with a third client that uses ffmpeg. I'm thinking of adding a module which uses ffmpeg, so that it will transcode H264 between different parameters. I've got a few questions: 1. Is this feasible? I'm not looking at supporting many simultaneous calls. 2. What is involved in transcoding real-time video stream? 3. Anyone's done anything like this before? I'm new to FS and any suggestions would be very much appreciated! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/2c949fa7/attachment-0001.html From sveen88 at hotmail.com Mon Jul 30 12:57:54 2012 From: sveen88 at hotmail.com (Andre Sveen) Date: Mon, 30 Jul 2012 10:57:54 +0200 Subject: [Freeswitch-users] Trouble registering Cisco 7942 Message-ID: Hi. I am using firmware 8.5(4) SIP. I see the register message on the FreeSwitch machine using ngrep. Everything seems ok, but I get unauthorized. Then I thought to myseld that this might be NAT issue. So I tried setting up a Asterisk on the same subnet as the telephone is sitting on. Same result. I have taken some info out of security reasons. Theese are marked in red text. The ngrep log below is from the FreeSwitch machine. I have also read from zero to hero here. http://www.freeswitch.org/node/401 The two first link give description of how to setup a TFTP server and soforth. I already have this. Important to mention that the Cisco 7942 is a newer phone than in theese first two links so to get the relevant config look at the third link. Below is the third link from this page. Have used lots of info from this page. http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP Best regards Andre U public_ip_office_network:3197 -> ip_of_the_pbx_public_ip:5060 REGISTER sip:dns_name_of_pbx SIP/2.0. Via: SIP/2.0/UDP 192.168.1.240:5060;branch=z9hG4bK3fcf18ed. From: ;tag=58bfea206ae00003efc5fd41-f09664b5. To: . Call-ID: 58bfea20-6ae00002-2dbab34e-95e84e4d at 192.168.1.240. Max-Forwards: 70. Date: Wed, 16 Dec 2009 08:44:00 GMT. CSeq: 102 REGISTER. User-Agent: Cisco-CP7942G/8.5.3. Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="434". Supported: (null),X-cisco-xsi-7.0.1. Content-Length: 0. Reason: SIP;cause=200;text="cisco-alarm:12 Name=SEP[mac address of phone] Load=term42.default Last=cm-reset-tcp". Expires: 3600. . U ip_of_the_public_ip:5060 -> public_ip_office_network:38398 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.240:5060;branch=z9hG4bK3fcf18ed;received=public_ip_of_office_network;rport=38398. From: ;tag=58bfea206ae00003efc5fd41-f09664b5. To: ;tag=02KZmrmjpDyHD. Call-ID: 58bfea20-6ae00002-2dbab34e-95e84e4d at 192.168.1.240. CSeq: 102 REGISTER. User-Agent: XXXX internal. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. WWW-Authenticate: Digest realm="centrex.iplink.no", nonce="b91fe311-c1d7-e111-bd36-001e0bc2ead2", algorithm=MD5, qop="auth". Content-Length: 0. Then it repeats itself. Config file phone SEP[macaddress].cnf.xml this is downloaded to phone via TFTP. SIP username password D-M-Y Central Europe Standard/Daylight Time Unicast 2000 5060 5061 dns_name_of_pbx true true x--serviceuri-cfwdall x-cisco-serviceuri-pickup x-cisco-serviceuri-opickup x-cisco-serviceuri-gpickup x-cisco-serviceuri-meetme x-cisco-serviceuri-abbrdial false 2 true true 2 2 0 true 6 10 180 3600 5 120 120 5 500 4000 70 false None 1 false true false false g711a 101 3 avt false false 3 false external IP address Bilutleie 0 false 10 false 16384 32766 9 3000 3000 3000 3000 dns_name_pbx 5060 2 3 3000 secret false 1 *97 4 5 true false false true 5060 184 0 dialplan.xml true 2 SIP42.8-5-4S false false 0 1 0 0 0 0 0 1,2,3,4,5,6,7 00:00 00:00 00:00 1 1 96 0 96 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/f98b8d05/attachment-0001.html From sveen88 at hotmail.com Mon Jul 30 15:07:54 2012 From: sveen88 at hotmail.com (Andre Sveen) Date: Mon, 30 Jul 2012 13:07:54 +0200 Subject: [Freeswitch-users] Trouble registering Cisco 7942 In-Reply-To: References: Message-ID: Hi. I am using firmware 8.5(4) SIP. I see the register message on the FreeSwitch machine using ngrep. Everything seems ok, but I get unauthorized. Then I thought to myseld that this might be NAT issue. So I tried setting up a Asterisk on the same subnet as the telephone is sitting on. Same result. I have taken some info out of security reasons. Theese are marked in red text. The ngrep log below is from the FreeSwitch machine. I have also read from zero to hero here. http://www.freeswitch.org/node/401 The two first link give description of how to setup a TFTP server and soforth. I already have this. Important to mention that the Cisco 7942 is a newer phone than in theese first two links so to get the relevant config look at the third link. Below is the third link from this page. Have used lots of info from this page. http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP Best regards Andre U public_ip_office_network:3197 -> ip_of_the_pbx_public_ip:5060 REGISTER sip:dns_name_of_pbx SIP/2.0. Via: SIP/2.0/UDP 192.168.1.240:5060;branch=z9hG4bK3fcf18ed. From: ;tag=58bfea206ae00003efc5fd41-f09664b5. To: . Call-ID: 58bfea20-6ae00002-2dbab34e-95e84e4d at 192.168.1.240. Max-Forwards: 70. Date: Wed, 16 Dec 2009 08:44:00 GMT. CSeq: 102 REGISTER. User-Agent: Cisco-CP7942G/8.5.3. Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="434". Supported: (null),X-cisco-xsi-7.0.1. Content-Length: 0. Reason: SIP;cause=200;text="cisco-alarm:12 Name=SEP[mac address of phone] Load=term42.default Last=cm-reset-tcp". Expires: 3600. . U ip_of_the_public_ip:5060 -> public_ip_office_network:38398 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.240:5060;branch=z9hG4bK3fcf18ed;received=public_ip_of_office_network;rport=38398. From: ;tag=58bfea206ae00003efc5fd41-f09664b5. To: ;tag=02KZmrmjpDyHD. Call-ID: 58bfea20-6ae00002-2dbab34e-95e84e4d at 192.168.1.240. CSeq: 102 REGISTER. User-Agent: XXXX internal. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. WWW-Authenticate: Digest realm="centrex.iplink.no", nonce="b91fe311-c1d7-e111-bd36-001e0bc2ead2", algorithm=MD5, qop="auth". Content-Length: 0. Then it repeats itself. Config file phone SEP[macaddress].cnf.xml this is downloaded to phone via TFTP. SIP username password D-M-Y Central Europe Standard/Daylight Time Unicast 2000 5060 5061 dns_name_of_pbx true true x--serviceuri-cfwdall x-cisco-serviceuri-pickup x-cisco-serviceuri-opickup x-cisco-serviceuri-gpickup x-cisco-serviceuri-meetme x-cisco-serviceuri-abbrdial false 2 true true 2 2 0 true 6 10 180 3600 5 120 120 5 500 4000 70 false None 1 false true false false g711a 101 3 avt false false 3 false external IP address Bilutleie 0 false 10 false 16384 32766 9 3000 3000 3000 3000 dns_name_pbx 5060 2 3 3000 secret false 1 *97 4 5 true false false true 5060 184 0 dialplan.xml true 2 SIP42.8-5-4S false false 0 1 0 0 0 0 0 1,2,3,4,5,6,7 00:00 00:00 00:00 1 1 96 0 96 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/4f4170db/attachment-0001.html From sveen88 at hotmail.com Mon Jul 30 17:33:46 2012 From: sveen88 at hotmail.com (Andre Sveen) Date: Mon, 30 Jul 2012 15:33:46 +0200 Subject: [Freeswitch-users] Trouble registering Cisco 7942 In-Reply-To: References: , Message-ID: Hi. I am using firmware 8.5(4) SIP. I see the register message on the FreeSwitch machine using ngrep. Everything seems ok, but I get unauthorized. Then I thought to myseld that this might be NAT issue. So I tried setting up a Asterisk on the same subnet as the telephone is sitting on. Same result. I have taken some info out of security reasons. Theese are marked in red text. The ngrep log below is from the FreeSwitch machine. I have also read from zero to hero here. http://www.freeswitch.org/node/401 The two first link give description of how to setup a TFTP server and soforth. I already have this. Important to mention that the Cisco 7942 is a newer phone than in theese first two links so to get the relevant config look at the third link. Below is the third link from this page. Have used lots of info from this page. http://www.voip-info.org/wiki/view/Asterisk+phone+cisco+79x1+xml+configuration+files+for+SIP A little snippet from the console log on the phone. 977: ERR 09:43:30.785766 JVM: dns_gethostbysrv 3 h_errno 978: ERR 09:43:30.786378 JVM: sipTransportGetServerIPAddr: Error: sipTransportGetServerAddrPort returned error. 979: ERR 09:43:30.803646 JVM: dns_gethostbysrv 3 h_errno 980: ERR 09:43:30.804270 JVM: sipTransportGetServerIPAddr: Error: sipTransportGetServerAddrPort returned error. 981: ERR 09:43:30.821237 JVM: dns_gethostbysrv 3 h_errno 982: ERR 09:43:30.821830 JVM: sipTransportGetServerIPAddr: Error: sipTransportGetServerAddrPort returned error. 983: ERR 09:43:30.838442 JVM: dns_gethostbysrv 3 h_errno 984: ERR 09:43:30.839066 JVM: sipTransportGetServerIPAddr: Error: sipTransportGetServerAddrPort returned error. 985: ERR 09:43:30.856124 JVM: dns_gethostbysrv 3 h_errno 986: ERR 09:43:30.856715 JVM: sipTransportGetServerIPAddr: Error: sipTransportGetServerAddrPort returned error. 987: ERR 09:43:30.874024 JVM: dns_gethostbysrv 3 h_errno 988: ERR 09:43:30.874615 JVM: sipTransportGetServerIPAddr: Error: sipTransportGetServerAddrPort returned error. 989: ERR 09:43:30.891639 JVM: dns_gethostbysrv 3 h_errno 990: ERR 09:43:30.892265 JVM: sipTransportGetServerIPAddr: Error: sipTransportGetServerAddrPort returned error. 991: ERR 09:43:30.908952 JVM: dns_gethostbysrv 3 h_errno 992: ERR 09:43:30.909538 JVM: sipTransportGetServerIPAddr: Error: sipTransportGetServerAddrPort returned error. 993: NOT 09:43:30.985443 JVM: Startup Module Loader|cip.midp.midletsuite.InstallerModule:? - propertyChanged - device.callagent.messages.0 value=0 994: NOT 09:43:30.988054 JVM: Startup Module Loader|cip.midp.midletsuite.InstallerModule:? - propertyChanged - device.callagent.messages.0 value=0 995: ERR 09:43:31.002746 JVM: dns_gethostbysrv 3 h_errno Best regards Andre U public_ip_office_network:3197 -> ip_of_the_pbx_public_ip:5060 REGISTER sip:dns_name_of_pbx SIP/2.0. Via: SIP/2.0/UDP 192.168.1.240:5060;branch=z9hG4bK3fcf18ed. From: ;tag=58bfea206ae00003efc5fd41-f09664b5. To: . Call-ID: 58bfea20-6ae00002-2dbab34e-95e84e4d at 192.168.1.240. Max-Forwards: 70. Date: Wed, 16 Dec 2009 08:44:00 GMT. CSeq: 102 REGISTER. User-Agent: Cisco-CP7942G/8.5.3. Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="434". Supported: (null),X-cisco-xsi-7.0.1. Content-Length: 0. Reason: SIP;cause=200;text="cisco-alarm:12 Name=SEP[mac address of phone] Load=term42.default Last=cm-reset-tcp". Expires: 3600. . U ip_of_the_public_ip:5060 -> public_ip_office_network:38398 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.240:5060;branch=z9hG4bK3fcf18ed;received=public_ip_of_office_network;rport=38398. From: ;tag=58bfea206ae00003efc5fd41-f09664b5. To: ;tag=02KZmrmjpDyHD. Call-ID: 58bfea20-6ae00002-2dbab34e-95e84e4d at 192.168.1.240. CSeq: 102 REGISTER. User-Agent: XXXX internal. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE. Supported: timer, precondition, path, replaces. WWW-Authenticate: Digest realm="centrex.iplink.no", nonce="b91fe311-c1d7-e111-bd36-001e0bc2ead2", algorithm=MD5, qop="auth". Content-Length: 0. Then it repeats itself. Config file phone SEP[macaddress].cnf.xml this is downloaded to phone via TFTP. SIP username password D-M-Y Central Europe Standard/Daylight Time Unicast 2000 5060 5061 dns_name_of_pbx true true x--serviceuri-cfwdall x-cisco-serviceuri-pickup x-cisco-serviceuri-opickup x-cisco-serviceuri-gpickup x-cisco-serviceuri-meetme x-cisco-serviceuri-abbrdial false 2 true true 2 2 0 true 6 10 180 3600 5 120 120 5 500 4000 70 false None 1 false true false false g711a 101 3 avt false false 3 false external IP address Bilutleie 0 false 10 false 16384 32766 9 3000 3000 3000 3000 dns_name_pbx 5060 2 3 3000 secret false 1 *97 4 5 true false false true 5060 184 0 dialplan.xml true 2 SIP42.8-5-4S false false 0 1 0 0 0 0 0 1,2,3,4,5,6,7 00:00 00:00 00:00 1 1 96 0 96 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/d76a72fd/attachment.html From anthony.minessale at gmail.com Mon Jul 30 19:26:01 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Jul 2012 10:26:01 -0500 Subject: [Freeswitch-users] sofia_contact result In-Reply-To: References: Message-ID: its basically "sofia/" + + "/" On Mon, Jul 30, 2012 at 8:25 AM, Gerald Weber wrote: > With rely i mean, ist he host/ip always between '@' and ':' or are there any other combinations ? > sofia//sip:@: > > Background: > I'm currenlty working on mod_snom to implement the KEYPRESS feature. > To do so, i need the ip of the user, 2 choices came on my mind: > > 1. use sofia_contact as suggested by Brian Foster and parse the output. > 2. do a select network_ip from registrations where reg_user = 'xxxx' > > sofia_contact looks better because i can pass the whole contact string from mod_callcenter using switch_api_execute without all the fiddling with domain, user, etc. > > Thats why i'm interested in the structure. > > ps: I created a feature request at snom to support notify talk last november, but still no response :( > > > > -----Urspr?ngliche Nachricht----- > Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Anthony Minessale > Gesendet: Montag, 30. Juli 2012 15:11 > An: FreeSWITCH Users Help > Betreff: Re: [Freeswitch-users] sofia_contact result > > Depends on what you mean by rely? > > > On Mon, Jul 30, 2012 at 7:56 AM, Gerald Weber wrote: >> Hi, >> >> >> >> can i rely on the structure of the result string i get back from e.g.: >> >> >> >> freeswitch at default> sofia_contact 2000 (or sofia_contact user/2000) >> >> sofia/internal/sip:2000 at 192.168.20.219:5060 >> >> >> >> thanks >> >> gw >> >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users 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 >> > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From mi.ke at null.net Mon Jul 30 19:11:07 2012 From: mi.ke at null.net (Mi Ke) Date: Mon, 30 Jul 2012 11:11:07 -0400 Subject: [Freeswitch-users] CPU Usage Message-ID: <20120730151107.154620@gmx.com> try running it with -heavy-timer as recommended here: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-CPU-usage-td5443765.html ----- Original Message ----- From: Andrew Paul Sent: 07/30/12 05:01 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] CPU Usage Hai all, How can i reduce cpu high usage in freeswitch. I am using freeswitch version of 1.2.0. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/b16a63ee/attachment.html From nbhatti at gmail.com Mon Jul 30 19:32:34 2012 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Mon, 30 Jul 2012 18:32:34 +0300 Subject: [Freeswitch-users] CPU Usage In-Reply-To: References: Message-ID: Which OS is this? If CentOS 6.x, it is not supported and you will face issues. See http://jira.freeswitch.org/browse/FS-4316 On Mon, Jul 30, 2012 at 5:01 PM, Andrew Paul wrote: > Hai all, > How can i reduce cpu high usage in freeswitch. I am using freeswitch > version of 1.2.0. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120730/f5ad5fd8/attachment.html From peter.olsson at visionutveckling.se Mon Jul 30 19:37:21 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 30 Jul 2012 15:37:21 +0000 Subject: [Freeswitch-users] CPU Usage Message-ID: <1FFF97C269757C458224B7C895F35F15139DD5@cantor.std.visionutv.se> First of all, make sure you are not using CentOS 6.x ? since there are known (CPU) issues with CentOS 6 and FS. The original question doesn?t say much though... I guess another recommendation could be to make sure to bypass media, which will make FS use less CPU. However, it?s kind of impossible to know what to answer on a general question like that. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Mi Ke Skickat: den 30 juli 2012 17:11 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] CPU Usage try running it with -heavy-timer as recommended here: http://freeswitch-users.2379917.n2.nabble.com/freeswitch-CPU-usage-td5443765.html ----- Original Message ----- From: Andrew Paul Sent: 07/30/12 05:01 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] CPU Usage Hai all, How can i reduce cpu high usage in freeswitch. I am using freeswitch version of 1.2.0. !DSPAM:5016a5e332761623080383! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/b762c72b/attachment.html From dujinfang at gmail.com Mon Jul 30 20:06:03 2012 From: dujinfang at gmail.com (Seven Du) Date: Tue, 31 Jul 2012 00:06:03 +0800 Subject: [Freeswitch-users] H264 transcoding In-Reply-To: <50167F11.9010808@redembedded.com> References: <50167F11.9010808@redembedded.com> Message-ID: <9367DFD0AC5249B19594D89A578B4890@gmail.com> Cluecon is coming and I think we could wait a few days and we still have some issues to figure out ... On Monday, July 30, 2012 at 8:33 PM, Yufei Tao wrote: > Hi Seven > > Thanks very much for your replies! I thought I should be very surprised if I was the first one wanting to do this. It'd be very helpful if you could share your code with x264, or libvlc, to spare me, and possibly others of reinventing the wheels. I'll compile the code myself of course which should be free of the licensing troubles I guess. > > I'd be very grateful if you could share the code and give some instructions. Thank you very much! > > Yufei > > > Subject: Re: [Freeswitch-users] H264 transcodin > From: > Seven Du (mailto:dujinfang at gmail.com) > > Date: > 28/07/12 01:45 > > > > To: > FreeSWITCH Users Help (mailto:freeswitch-users at lists.freeswitch.org) > > > > > Real-time encoding with statically linked x264 lib works fine for me from QCIF to D1 resolution, 720p is slow and discarding frames on a Xeon Quad core CPU. I haven't look how to use the GPU, or if possible. > > It is working in my lab and I have the same question with Yufei Tao when going to production or deliver to customer. Based on http://lists.freeswitch.org/pipermail/freeswitch-dev/2010-September/004227.html , In my understanding, compile and link and use by my self should be fine and, if I deliver to a customer, it should be fine if I provide the code and help the customer to compile on their own server? I'd like to open source the code to public later, but, I'd like to know is it a MUST or MAY? > > If you pipe to ffmpeg or x264 command line it's not been treated as combine into a large work, and I'm not sure if realtime transcoding will be smooth. > > 7. > > On Friday, July 27, 2012 at 11:17 PM, Yufei Tao wrote: > > > Thanks everyone for the responses! > > > > If I understand it correctly, if I installed ffmpeg on itself separately > > from FS, I could write a module for FS, in which I just call the ffmpeg > > program by running a command line. This way would it be classified as > > "not combine them into a larger work", thus free from license > > incompatibility problem? > > > > Not sure if that'll work for real-time transcoding of x264 though? > > > > Thanks very much for you opinions! > > Yufei > > > > -- > > Yufei Tao > > Red Embedded > > > > This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. > > > > You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. > > > > Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > ForwardedMessage.eml > Subject: > Re: [Freeswitch-users] H264 transcoding > > From: > Seven Du (mailto:dujinfang at gmail.com) > > Date: > 28/07/12 02:10 > > > > To: > FreeSWITCH Users Help (mailto:freeswitch-users at lists.freeswitch.org) > > > > > I patched libvlc and it can now decode any video vlc supported and encode with x264 and send to any sip phone. Specifically I'm trying to use it to play a 1080p stream and resize to CIF or D1 so a video phone will accept(Sending 1080p will cause some phones to reboot :( ). > > Still need a lot of code to make it working neatly, however, I might can do a demo on ClueCon and @William if you'd like to review the code and merge into tree I'll happy to contribute that later. > > > On Friday, July 27, 2012 at 2:43 PM, William King wrote: > > > libvlc is LGPL http://www.videolan.org/press/lgpl.html and there is now a mod_vlc(though it doesn't yet support video streams). The user can choose to build vlc with only the LGPL components or add the more 'adverse' modules. In none of the LGPL packages of libvlc is ffmpeg enabled, but there is a module for libvlc for ffmpeg. http://wiki.videolan.org/FFmpeg > > > > The only pieces now may just be the FS side of things for video. > > William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com (mailto:william.king at quentustech.com) > > On 07/26/2012 08:46 PM, Anthony Minessale wrote: > > > you would probably need to do something like make a mod for ffmpeg that protects you from the gpl then allow the user to build that lib on his own and choose at compile time to install patented or adverse licensed components. No license rules prohibit an end user from combining code only distributors. but even then we need a bunch of code to write. On Thu, Jul 26, 2012 at 10:41 PM, curriegrad2004 (mailto:curriegrad2004 at gmail.com) wrote: > > > > Ken, If you think those guys over at x264 will ever change the license from GPL to LGPL, you're just dreaming the pie in the sky... In short, don't even think about it ;P On Thu, Jul 26, 2012 at 8:19 PM, Ken Rice (mailto:krice at freeswitch.org) wrote: > > > > > we can not and will not use GPL software, the license is not compatible with the GPL and would polute the codebase with additional restrictions that are not wanted or needed. now if someone could get them to change the license or atleast give us a license under better terms such as the LGPL or the MPL then the license issue would be null Ken Sent from my iPad On Jul 26, 2012, at 7:45 PM, Terry Barnum (mailto:terry at digital-outpost.com) wrote: > > > > > > Use x264? http://en.wikipedia.org/wiki/X264 On Jul 26, 2012, at 4:53 PM, Ken Rice wrote: > > > > > > > Is it possible sure... Is ot probably to happen anytime soon? Not until the patents run out... On 7/26/12 5:04 PM, "yufei.tao" (mailto:yufei.tao at redembedded.com) wrote: > > > > > > > > Hi I am trying to decide if it is feasible to let FS do transcoding between different H264 formats for live video calls. This is because I've got SIP clients that both use H264 but with different formats and one (with a bad H264 decoder) has problems decoding H264 stream from the other. But each of these two clients communicate fine using H264 with a third client that uses ffmpeg. I'm thinking of adding a module which uses ffmpeg, so that it will transcode H264 between different parameters. I've got a few questions: 1. Is this feasible? I'm not looking at supporting many simultaneous calls. 2. What is involved in transcoding real-time video stream? 3. Anyone's done anything like this before? I'm new to FS and any suggestions would be very much appreciated! Yufei > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > Yufei Tao > Red Embedded > This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. > You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. > Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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/20120731/a17aab26/attachment-0001.html From dujinfang at gmail.com Mon Jul 30 20:17:54 2012 From: dujinfang at gmail.com (Seven Du) Date: Tue, 31 Jul 2012 00:17:54 +0800 Subject: [Freeswitch-users] H264 transcoding In-Reply-To: <9367DFD0AC5249B19594D89A578B4890@gmail.com> References: <50167F11.9010808@redembedded.com> <9367DFD0AC5249B19594D89A578B4890@gmail.com> Message-ID: <607832F5D20D464DA8435263D397C24F@gmail.com> And Yufei, my solution won't solve your problem at this time, I only did encoding and for your transcoder to work there need a lot more work?. On Tuesday, July 31, 2012 at 12:06 AM, Seven Du wrote: > Cluecon is coming and I think we could wait a few days and we still have some issues to figure out ... > > > On Monday, July 30, 2012 at 8:33 PM, Yufei Tao wrote: > > > Hi Seven > > > > Thanks very much for your replies! I thought I should be very surprised if I was the first one wanting to do this. It'd be very helpful if you could share your code with x264, or libvlc, to spare me, and possibly others of reinventing the wheels. I'll compile the code myself of course which should be free of the licensing troubles I guess. > > > > I'd be very grateful if you could share the code and give some instructions. Thank you very much! > > > > Yufei > > > > > > Subject: Re: [Freeswitch-users] H264 transcodin > > From: > > Seven Du (mailto:dujinfang at gmail.com) > > > > Date: > > 28/07/12 01:45 > > > > > > > > To: > > FreeSWITCH Users Help (mailto:freeswitch-users at lists.freeswitch.org) > > > > > > > > > > Real-time encoding with statically linked x264 lib works fine for me from QCIF to D1 resolution, 720p is slow and discarding frames on a Xeon Quad core CPU. I haven't look how to use the GPU, or if possible. > > > > It is working in my lab and I have the same question with Yufei Tao when going to production or deliver to customer. Based on http://lists.freeswitch.org/pipermail/freeswitch-dev/2010-September/004227.html , In my understanding, compile and link and use by my self should be fine and, if I deliver to a customer, it should be fine if I provide the code and help the customer to compile on their own server? I'd like to open source the code to public later, but, I'd like to know is it a MUST or MAY? > > > > If you pipe to ffmpeg or x264 command line it's not been treated as combine into a large work, and I'm not sure if realtime transcoding will be smooth. > > > > 7. > > > > On Friday, July 27, 2012 at 11:17 PM, Yufei Tao wrote: > > > > > Thanks everyone for the responses! > > > > > > If I understand it correctly, if I installed ffmpeg on itself separately > > > from FS, I could write a module for FS, in which I just call the ffmpeg > > > program by running a command line. This way would it be classified as > > > "not combine them into a larger work", thus free from license > > > incompatibility problem? > > > > > > Not sure if that'll work for real-time transcoding of x264 though? > > > > > > Thanks very much for you opinions! > > > Yufei > > > > > > -- > > > Yufei Tao > > > Red Embedded > > > > > > This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. > > > > > > You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. > > > > > > Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > ForwardedMessage.eml > > Subject: > > Re: [Freeswitch-users] H264 transcoding > > > > From: > > Seven Du (mailto:dujinfang at gmail.com) > > > > Date: > > 28/07/12 02:10 > > > > > > > > To: > > FreeSWITCH Users Help (mailto:freeswitch-users at lists.freeswitch.org) > > > > > > > > > > I patched libvlc and it can now decode any video vlc supported and encode with x264 and send to any sip phone. Specifically I'm trying to use it to play a 1080p stream and resize to CIF or D1 so a video phone will accept(Sending 1080p will cause some phones to reboot :( ). > > > > Still need a lot of code to make it working neatly, however, I might can do a demo on ClueCon and @William if you'd like to review the code and merge into tree I'll happy to contribute that later. > > > > > > On Friday, July 27, 2012 at 2:43 PM, William King wrote: > > > > > libvlc is LGPL http://www.videolan.org/press/lgpl.html and there is now a mod_vlc(though it doesn't yet support video streams). The user can choose to build vlc with only the LGPL components or add the more 'adverse' modules. In none of the LGPL packages of libvlc is ffmpeg enabled, but there is a module for libvlc for ffmpeg. http://wiki.videolan.org/FFmpeg > > > > > > The only pieces now may just be the FS side of things for video. > > > William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com (mailto:william.king at quentustech.com) > > > On 07/26/2012 08:46 PM, Anthony Minessale wrote: > > > > you would probably need to do something like make a mod for ffmpeg that protects you from the gpl then allow the user to build that lib on his own and choose at compile time to install patented or adverse licensed components. No license rules prohibit an end user from combining code only distributors. but even then we need a bunch of code to write. On Thu, Jul 26, 2012 at 10:41 PM, curriegrad2004 (mailto:curriegrad2004 at gmail.com) wrote: > > > > > Ken, If you think those guys over at x264 will ever change the license from GPL to LGPL, you're just dreaming the pie in the sky... In short, don't even think about it ;P On Thu, Jul 26, 2012 at 8:19 PM, Ken Rice (mailto:krice at freeswitch.org) wrote: > > > > > > we can not and will not use GPL software, the license is not compatible with the GPL and would polute the codebase with additional restrictions that are not wanted or needed. now if someone could get them to change the license or atleast give us a license under better terms such as the LGPL or the MPL then the license issue would be null Ken Sent from my iPad On Jul 26, 2012, at 7:45 PM, Terry Barnum (mailto:terry at digital-outpost.com) wrote: > > > > > > > Use x264? http://en.wikipedia.org/wiki/X264 On Jul 26, 2012, at 4:53 PM, Ken Rice wrote: > > > > > > > > Is it possible sure... Is ot probably to happen anytime soon? Not until the patents run out... On 7/26/12 5:04 PM, "yufei.tao" (mailto:yufei.tao at redembedded.com) wrote: > > > > > > > > > Hi I am trying to decide if it is feasible to let FS do transcoding between different H264 formats for live video calls. This is because I've got SIP clients that both use H264 but with different formats and one (with a bad H264 decoder) has problems decoding H264 stream from the other. But each of these two clients communicate fine using H264 with a third client that uses ffmpeg. I'm thinking of adding a module which uses ffmpeg, so that it will transcode H264 between different parameters. I've got a few questions: 1. Is this feasible? I'm not looking at supporting many simultaneous calls. 2. What is involved in transcoding real-time video stream? 3. Anyone's done anything like this before? I'm new to FS and any suggestions would be very much appreciated! Yufei > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > > Yufei Tao > > Red Embedded > > This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. > > You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. > > Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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/20120731/eca3716d/attachment.html From robert.hadley at teotech.com Mon Jul 30 20:28:36 2012 From: robert.hadley at teotech.com (Robert Hadley) Date: Mon, 30 Jul 2012 16:28:36 +0000 Subject: [Freeswitch-users] Help!! FS -TLS interworking issue, How to config to allow "gentls_cert" to generate a root certificate with more longer valid-period ? In-Reply-To: References: Message-ID: <71943DD5C22943448A24B7C5CDC238070FF1A549@SN2PRD0410MB396.namprd04.prod.outlook.com> Hi Charles, Try the changes in this attached freeswitch/scripts/gentls_cert.in file. There were a few typos in the original script. Regards, Robert From: fieldpeak [mailto:fieldpeak at gmail.com] Sent: Sunday, July 29, 2012 5:57 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help!! FS -TLS interworking issue, How to config to allow "gentls_cert" to generate a root certificate with more longer valid-period ? Could you please help advise more details? how should i do to implement your advise... i have very few knowledge about pki, certificate and tls, thanks a lot! :) BR,Charles ? 2012-7-30 ??4:57?"curriegrad2004" >??? You do realize that if your org does have an existing PKI infrastructure, you can in theory skip all of that. You just need to know the right values to configure and that's about it. (wink, wink, nudge, nudge) :) On Sun, Jul 29, 2012 at 11:23 AM, Mitch Capper > wrote: > Try running head from GIT and let us know if you still have a problem. > > ~mitch > > On Sun, Jul 29, 2012 at 5:25 AM, fieldpeak > wrote: >> Hi Masters, >> >> >> >> I'm testing the TLS on FS to work with softphone. >> >> >> >> followed the wiki >> (http://wiki.freeswitch.org/wiki/Tls#EyeBeam.2FBria_Setup), >> >> >> >> I generated the CA (root) certificate by below command, however, when i >> install the root certificate on windows, it prompt me that the valid period >> is for only one month. I tried to change the "DAYS=2190" inside the >> "gentls_cert" script, but it only effect on server certificate(agent.pem) >> but not root certificate (cafile.pem), Could anyone please help me, >> appreciated for your any advise!! >> >> >> >> ./gentls_cert setup -cn fs.audiocodes.com.cn -alt DNS:fs.audiocodes.com.cn >> -org audiocodes.com.cn >> >> >> >> below is from wiki. >> >> This will create CA certificate and key along with in conf/ssl/CA >> directory(cacert.pem, cakey.pem) and certificate in the conf/ssl >> folder(cafile.pem). >> >> [ Note: The name given for -cn and -alt should be the same as the DNS name >> of your freeswitch installation and used as the registrar name on the phone >> (at least on Polycoms). ] You can change the "DAYS=2190" line in the >> gentls_cert file to make the certificate valid for longer time. However >> making it too long has some wrap around problem, it appears. >> >> >> To short, I want to change the valid period longer for the cafile.pem , >> thanks!! >> >> >> -- >> Regards, >> Charles >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120730/4afcfada/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: gentls_cert.in Type: application/octet-stream Size: 4785 bytes Desc: gentls_cert.in Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/4afcfada/attachment-0001.obj From msc at freeswitch.org Mon Jul 30 20:51:38 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Jul 2012 09:51:38 -0700 Subject: [Freeswitch-users] FreeSWITCH dying with no core file In-Reply-To: <28307.1343519724@ccs.covici.com> References: <5013012C.9050003@freeswitch.org> <28307.1343519724@ccs.covici.com> Message-ID: > Thanks much for that tip -- I had never heard of that parameter, maybe > it should be in the wiki in the section about getting a backtrace. > > Crude but effective: http://wiki.freeswitch.org/wiki/Reporting_Bugs#Important_Note -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/e5e8fdc6/attachment.html From msc at freeswitch.org Mon Jul 30 21:07:28 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Jul 2012 10:07:28 -0700 Subject: [Freeswitch-users] Help!! FS -TLS interworking issue, How to config to allow "gentls_cert" to generate a root certificate with more longer valid-period ? In-Reply-To: <71943DD5C22943448A24B7C5CDC238070FF1A549@SN2PRD0410MB396.namprd04.prod.outlook.com> References: <71943DD5C22943448A24B7C5CDC238070FF1A549@SN2PRD0410MB396.namprd04.prod.outlook.com> Message-ID: On Mon, Jul 30, 2012 at 9:28 AM, Robert Hadley wrote: > Hi Charles,**** > > ** ** > > Try the changes in this attached freeswitch/scripts/gentls_cert.in file. > There were a few typos in the original script.**** > > ** ** > > Regards,**** > > Robert > I'd like to verify that those typos are indeed really typos and are really fixed. If anyone has input on them please let me know and I will see about getting the gentls_cert.in file updated. I definitely would like to see this tested before we make any updates. Thanks, MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/8c1a8a02/attachment.html From marketing at cluecon.com Mon Jul 30 22:04:28 2012 From: marketing at cluecon.com (Michael Collins) Date: Mon, 30 Jul 2012 11:04:28 -0700 Subject: [Freeswitch-users] FreeSWITCH Weekly News and Notes Message-ID: Greetings from the FreeSWITCH team! Things have been busy with all of the ClueCon preparations! We hope to see you in Chicago next week. In the meantime we are still doing our weekly conference calls. This past Wednesday we enjoyed a very nice demonstration by Dave Kompel (IRC: drk__) on how to implement Fail2Ban -like functionality in a Windows environment. Dave's code is simple and elegant. If you missed the presentation then please download it hereand listen to it. The source code is also available . This week we will be having a community discussion on several subjects, not the least of which is the interesting news that Google uses FreeSWITCH internally . This is the last weekly news and notes before ClueCon. Next Monday I'll be sending this out *from Chicago**!* To the scores of people who have registered and sponsored us this year I would like to give a well-deserved thanks. We are looking forward to seeing everyone next week. If you haven't yet registered there is still time, but please hurry! The sooner we know you're coming, the better prepared we will be to take care of you when you arrive. Have a great week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE cc12-0730 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/06f12166/attachment.html From msc at freeswitch.org Mon Jul 30 22:42:28 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Jul 2012 11:42:28 -0700 Subject: [Freeswitch-users] Question how can i get variables in lua from xml dialplan? In-Reply-To: References: Message-ID: The following code worked for me: myvar = session:getVariable('sound_prefix') freeswitch.consoleLog('WARNING',"The sound_prefix variable contains '" .. myvar .. "'\n") with dialplan: -MC On Mon, Jul 30, 2012 at 12:22 AM, Thomas Hoellriegel wrote: > Hi guys, > I like to import variables from xml dialplan in lua. > I found: > session:getVariable. > the problem: > i can only get variables from the lua wiki page. > But different variables for example: > strftime > or from: vars.xml : sound_prefix can.t be get. > > Or timeconditions don.t working in lua. > > Can your help please? > thanks > > > > > > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/93c9db04/attachment-0001.html From leonardo at daitangroup.com Mon Jul 30 22:55:03 2012 From: leonardo at daitangroup.com (Leonardo) Date: Mon, 30 Jul 2012 15:55:03 -0300 Subject: [Freeswitch-users] How to use TTS/flite via mod_socket Message-ID: <5016D887.5060806@daitangroup.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/c115576e/attachment.html From nasida at live.ru Mon Jul 30 23:00:13 2012 From: nasida at live.ru (Yuriy Nasida) Date: Mon, 30 Jul 2012 23:00:13 +0400 Subject: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf Message-ID: Hello guys, I change my voice via 'soundtouch' and I try to record this via simple 'record' command. I have to stop recording after '#' but can not make this because I must recieve DTMF inband. I.e. 'soundtouch' changes dtmf as well and FS doesn't recognise dtmf.How can I stop recording? Any ideas? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/570cc1d2/attachment.html From Hector.Geraldino at ipsoft.com Mon Jul 30 23:08:30 2012 From: Hector.Geraldino at ipsoft.com (Hector Geraldino) Date: Mon, 30 Jul 2012 15:08:30 -0400 Subject: [Freeswitch-users] How to use TTS/flite via mod_socket In-Reply-To: <5016D887.5060806@daitangroup.com> References: <5016D887.5060806@daitangroup.com> Message-ID: <6A6B4C284AD15042B429EB9D904544AD02304F5C38@NY1-EXMB-01.ip-soft.net> I don't know what API you're using to handle ESL connections, but in general all you need to do is to send an execute command with sendmsg: sendmsg call-command: execute execute-app-name: speak execute-app-arg: flite|kal|Hello world From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Leonardo Sent: Monday, July 30, 2012 2:55 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] How to use TTS/flite via mod_socket Hi. I compiled the mod_flite and it is working on my Freeswitch. I added the following dialplan: Is there a way to "speak" in a "connected" call based on UUID and using a mod_event_socket? Thanks, Leo -- Leonardo N. S. Pereira, Software Engineer T:+55.19.3112-1200 ext. 1283|F:+55.19.3207-1437 DaitanGroup|www.daitangroup.com|Highly Reliable Outsourcing. Value Added Services Worldwide. Privileged and confidential. If this message has been received in error, please notify sender and delete it immediately. Conte?do confidencial. Se esta mensagem foi recebida por engano, favor avisar o remetente e apag?-la imediatamente. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/54508bb4/attachment.html From peter.olsson at visionutveckling.se Mon Jul 30 23:23:09 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 30 Jul 2012 19:23:09 +0000 Subject: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf In-Reply-To: References: Message-ID: <1FFF97C269757C458224B7C895F35F15139F23@cantor.std.visionutv.se> If it's inband you need to execute "start_dtmf" application first. Then it should detect DTMF's for you. /Peter ________________________________ Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Yuriy Nasida [nasida at live.ru] Skickat: den 30 juli 2012 21:00 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf Hello guys, I change my voice via 'soundtouch' and I try to record this via simple 'record' command. I have to stop recording after '#' but can not make this because I must recieve DTMF inband. I.e. 'soundtouch' changes dtmf as well and FS doesn't recognise dtmf. How can I stop recording? Any ideas? Thanks. !DSPAM:5016d77b32761519715207! From nasida at live.ru Mon Jul 30 23:38:34 2012 From: nasida at live.ru (Yuriy Nasida) Date: Mon, 30 Jul 2012 23:38:34 +0400 Subject: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf In-Reply-To: <1FFF97C269757C458224B7C895F35F15139F23@cantor.std.visionutv.se> References: , <1FFF97C269757C458224B7C895F35F15139F23@cantor.std.visionutv.se> Message-ID: Peter, yes I have "start_dtmf" and I can detect DTMF. But! DTMF is not being recognised if I use I think soundtouch module changes my dtmf tones (which is inband).I start thinking about bribge + loopback. This way I will able to use soundtouch on b-leg only. Thanks. > From: peter.olsson at visionutveckling.se > To: freeswitch-users at lists.freeswitch.org > Date: Mon, 30 Jul 2012 19:23:09 +0000 > Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf > > If it's inband you need to execute "start_dtmf" application first. Then it should detect DTMF's for you. > > /Peter > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Yuriy Nasida [nasida at live.ru] > Skickat: den 30 juli 2012 21:00 > Till: freeswitch-users at lists.freeswitch.org > ?mne: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf > > Hello guys, > > I change my voice via 'soundtouch' and I try to record this via simple 'record' command. > > > > I have to stop recording after '#' but can not make this because I must recieve DTMF inband. > I.e. 'soundtouch' changes dtmf as well and FS doesn't recognise dtmf. > How can I stop recording? > > Any ideas? > Thanks. > !DSPAM:5016d77b32761519715207! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120730/4520cb46/attachment-0001.html From nasida at live.ru Tue Jul 31 00:25:02 2012 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 31 Jul 2012 00:25:02 +0400 Subject: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf In-Reply-To: References: , , <1FFF97C269757C458224B7C895F35F15139F23@cantor.std.visionutv.se>, Message-ID: bribge + loopback doesn't work because I have to set ON B-LEG which is affected by soundtouch.Hm... Any ideas ? From: nasida at live.ru To: freeswitch-users at lists.freeswitch.org Date: Mon, 30 Jul 2012 23:38:34 +0400 Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf Peter, yes I have "start_dtmf" and I can detect DTMF. But! DTMF is not being recognised if I use I think soundtouch module changes my dtmf tones (which is inband).I start thinking about bribge + loopback. This way I will able to use soundtouch on b-leg only. Thanks. > From: peter.olsson at visionutveckling.se > To: freeswitch-users at lists.freeswitch.org > Date: Mon, 30 Jul 2012 19:23:09 +0000 > Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf > > If it's inband you need to execute "start_dtmf" application first. Then it should detect DTMF's for you. > > /Peter > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Yuriy Nasida [nasida at live.ru] > Skickat: den 30 juli 2012 21:00 > Till: freeswitch-users at lists.freeswitch.org > ?mne: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf > > Hello guys, > > I change my voice via 'soundtouch' and I try to record this via simple 'record' command. > > > > I have to stop recording after '#' but can not make this because I must recieve DTMF inband. > I.e. 'soundtouch' changes dtmf as well and FS doesn't recognise dtmf. > How can I stop recording? > > Any ideas? > Thanks. > !DSPAM:5016d77b32761519715207! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120731/50f26f97/attachment.html From X.Liu at hw.ac.uk Tue Jul 31 01:38:11 2012 From: X.Liu at hw.ac.uk (Liu, Xingkun) Date: Mon, 30 Jul 2012 22:38:11 +0100 Subject: [Freeswitch-users] Am I using play_and_detect_speech correctly? Message-ID: Hello, I am using play_and_detect_speech with Java ESL in my IVR applications. Previously I call it again each time after I receive any recognition event, like recognition complete, no-input-timeout, or recognition-timeout, it seems to work fine. Now I have changed my app to issue play_and_detect_speech command based on my available system utterances as well as the speech event. I.e., I use a separate thread to constantly check if there is a system utterance coming in from another component of my application, if there is any utterance I issue the command which will speak the new utterance and listen to user input no matter whether or not previous command has finished. And if there is any speech event (recognition result, timeout etc.) the play_and_detect_speech command is also issued but with playing silence. Obviously the new command will stop the utterance speaking of the previous command if it is not finished. My question is will the new play_and_detect_speech command also stop the previous ASR listening or will there be many ASR listening channel and sending speech data (or silence) to ASR server? Do I need to explicitly issue a "stop" commnad before issuing a new play_and_detect_speech? If yes, how to do that, by "detect_speech stop"? Recently there is a network traffice problem (lots of connections /data transportation to the ASR server machine) when I am running my application. I am not sure if this is because of other issues or because of my new changes to the way of using play_and_detect_speech. Please any one could shed a light on this? Many thanks! Xing -- Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 We invite research leaders and ambitious early career researchers to join us in leading and driving research in key inter-disciplinary themes. Please see www.hw.ac.uk/researchleaders for further information and how to apply. Heriot-Watt University is a Scottish charity registered under charity number SC000278. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/12261399/attachment.html From anthony.minessale at gmail.com Tue Jul 31 01:43:56 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 30 Jul 2012 16:43:56 -0500 Subject: [Freeswitch-users] SIP to make phone reboot or resync In-Reply-To: <0deb01cd6d3f$67e43a30$37acae90$@bizfocused.com> References: <07b501cd6c1a$238b3e90$6aa1bbb0$@bizfocused.com> <094e01cd6c43$9e44cc90$dace65b0$@bizfocused.com> <0c4001cd6cdf$b2603a10$1720ae30$@bizfocused.com> <0deb01cd6d3f$67e43a30$37acae90$@bizfocused.com> Message-ID: reverse-auth-user and reverse-auth-pass params in the params section of the xml tag in the user directory can be configured instead of a gateway and it will use those credentials instead. On Sat, Jul 28, 2012 at 11:05 PM, Sean Devoy wrote: > Thank you sir. That did help, I can resync my phone test either with > user/pass using any proxy user/pass from the lines on THAT phone. I can > also disable Auth Resync Reboot and let anyone send reboots - which might be > fun, but seems like a bad plan! > > This is not very useful for me since the phone's proxy user/pass must be > specified in the gateway XML for whole realm in the the SIP profile. I > hoped to be able to re-provision any phone on demand. It seems impractical > to have to restart/rescan my sofia gateway every time I want to issue a > resync to a different phone. > > Am I missing something? > > Sean > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Vallimamod ABDULLAH > Sent: Saturday, July 28, 2012 1:05 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync > > Hi, > > It's normally the sip account credentials. There is also an option in the > admin interface to disable this authentication ("Auth Resync-Reboot" option > if I recall correctly in Ext tab) > > Best Regards, > Vallimamod Abdullah > . > > > On Jul 28, 2012, at 6:40 PM, Sean Devoy wrote: > >> HI all, >> >> I am close, but I still don't understand what credentials are required. > The response is: >> SIP/2.0 401 Unauthorized >> >> I have tried the web admin credentials for the phone, I don't k now what > FS credentials I could pass. >> >> Any ideas? >> >> Thanks, >> Sean >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Anthony Minessale >> Sent: Friday, July 27, 2012 6:12 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync >> >> Make a gateway with no reg and the credentials and name it after the realm > in the challenge. >> >> On Jul 27, 2012 5:05 PM, "Sean Devoy" wrote: >> Thank Anthony, BUT.. Error: >> 2012-07-27 17:58:55.118642 [ERR] sofia_reg.c:2165 Cannot locate any >> authentication credentials to complete an authentication request for >> realm '"fs_bfis.bizfocused.com"' >> >> 1 - where do I specify the credentials? >> 2 - Are these Freeswitch credentials or phone credentials? I suspect >> the latter. >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Anthony Minessale >> Sent: Friday, July 27, 2012 1:21 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync >> >> sofia profile check_sync [call_id] // no call_id means >> all sofia profile flush_inbound_reg [call_id] [reboot] >> >> >> >> On Fri, Jul 27, 2012 at 12:06 PM, Sean Devoy > wrote: >> > HI All, >> > >> > >> > >> > Does anyone have an code that will cause Freeswitch to send a SIP >> > message to a CISCO SPA5xx phone that will cause it to reboot or >> > resync (aka re-provision)? I know about the URLs that cause the >> > phone to do this, but they are NATed, so I need to use SIP to hit them. >> > >> > >> > >> > Assume I am programming language omni-lingual for this request! >> > >> > >> > >> > Thanks, >> > >> > Sean >> > >> > >> > ____________________________________________________________________ >> > __ ___ Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u >> > se >> > rs >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users 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 >> >> >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users 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 >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 From msc at freeswitch.org Tue Jul 31 02:02:35 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Jul 2012 15:02:35 -0700 Subject: [Freeswitch-users] SIP to make phone reboot or resync In-Reply-To: References: <07b501cd6c1a$238b3e90$6aa1bbb0$@bizfocused.com> <094e01cd6c43$9e44cc90$dace65b0$@bizfocused.com> <0c4001cd6cdf$b2603a10$1720ae30$@bizfocused.com> <0deb01cd6d3f$67e43a30$37acae90$@bizfocused.com> Message-ID: On Mon, Jul 30, 2012 at 2:43 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > reverse-auth-user and reverse-auth-pass params in the params section > of the xml tag in the user directory can be configured instead > of a gateway and it will use those credentials instead. > > FYI this is actually on the wiki (thanks kn0x): http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#reverse_authentication It might be helpful to make it easier to find, perhaps by linking to it from another page. Sean, can you tell us where on the wiki you were looking for this? That will give me an idea of where exactly I should add the link & info. -MC > On Sat, Jul 28, 2012 at 11:05 PM, Sean Devoy > wrote: > > Thank you sir. That did help, I can resync my phone test either with > > user/pass using any proxy user/pass from the lines on THAT phone. I can > > also disable Auth Resync Reboot and let anyone send reboots - which > might be > > fun, but seems like a bad plan! > > > > This is not very useful for me since the phone's proxy user/pass must be > > specified in the gateway XML for whole realm in the the SIP profile. I > > hoped to be able to re-provision any phone on demand. It seems > impractical > > to have to restart/rescan my sofia gateway every time I want to issue a > > resync to a different phone. > > > > Am I missing something? > > > > Sean > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > Vallimamod ABDULLAH > > Sent: Saturday, July 28, 2012 1:05 PM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync > > > > Hi, > > > > It's normally the sip account credentials. There is also an option in the > > admin interface to disable this authentication ("Auth Resync-Reboot" > option > > if I recall correctly in Ext tab) > > > > Best Regards, > > Vallimamod Abdullah > > . > > > > > > On Jul 28, 2012, at 6:40 PM, Sean Devoy wrote: > > > >> HI all, > >> > >> I am close, but I still don't understand what credentials are required. > > The response is: > >> SIP/2.0 401 Unauthorized > >> > >> I have tried the web admin credentials for the phone, I don't k now what > > FS credentials I could pass. > >> > >> Any ideas? > >> > >> Thanks, > >> Sean > >> > >> From: freeswitch-users-bounces at lists.freeswitch.org > >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > >> Anthony Minessale > >> Sent: Friday, July 27, 2012 6:12 PM > >> To: FreeSWITCH Users Help > >> Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync > >> > >> Make a gateway with no reg and the credentials and name it after the > realm > > in the challenge. > >> > >> On Jul 27, 2012 5:05 PM, "Sean Devoy" wrote: > >> Thank Anthony, BUT.. Error: > >> 2012-07-27 17:58:55.118642 [ERR] sofia_reg.c:2165 Cannot locate any > >> authentication credentials to complete an authentication request for > >> realm '"fs_bfis.bizfocused.com"' > >> > >> 1 - where do I specify the credentials? > >> 2 - Are these Freeswitch credentials or phone credentials? I suspect > >> the latter. > >> > >> -----Original Message----- > >> From: freeswitch-users-bounces at lists.freeswitch.org > >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > >> Anthony Minessale > >> Sent: Friday, July 27, 2012 1:21 PM > >> To: FreeSWITCH Users Help > >> Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync > >> > >> sofia profile check_sync [call_id] // no call_id means > >> all sofia profile flush_inbound_reg [call_id] [reboot] > >> > >> > >> > >> On Fri, Jul 27, 2012 at 12:06 PM, Sean Devoy > > wrote: > >> > HI All, > >> > > >> > > >> > > >> > Does anyone have an code that will cause Freeswitch to send a SIP > >> > message to a CISCO SPA5xx phone that will cause it to reboot or > >> > resync (aka re-provision)? I know about the URLs that cause the > >> > phone to do this, but they are NATed, so I need to use SIP to hit > them. > >> > > >> > > >> > > >> > Assume I am programming language omni-lingual for this request! > >> > > >> > > >> > > >> > Thanks, > >> > > >> > Sean > >> > > >> > > >> > ____________________________________________________________________ > >> > __ ___ Professional FreeSWITCH Consulting Services: > >> > consulting at freeswitch.org > >> > http://www.freeswitchsolutions.com > >> > > >> > > >> > > >> > > >> > Official FreeSWITCH Sites > >> > http://www.freeswitch.org > >> > http://wiki.freeswitch.org > >> > http://www.cluecon.com > >> > > >> > Join Us At ClueCon - Aug 7-9, 2012 > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u > >> > se > >> > rs > >> > http://www.freeswitch.org > >> > > >> > >> > >> > >> -- > >> Anthony Minessale II > >> > >> FreeSWITCH http://www.freeswitch.org/ > >> ClueCon http://www.cluecon.com/ > >> Twitter: http://twitter.com/FreeSWITCH_wire > >> > >> AIM: anthm > >> MSN:anthony_minessale at hotmail.com > >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > >> IRC: irc.freenode.net #freeswitch > >> > >> FreeSWITCH Developer Conference > >> sip:888 at conference.freeswitch.org > >> googletalk:conf+888 at conference.freeswitch.org > >> pstn:+19193869900 > >> > >> ______________________________________________________________________ > >> ___ Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users 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 > >> > >> > >> > >> ______________________________________________________________________ > >> ___ Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users 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 > >> ______________________________________________________________________ > >> ___ Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> Join Us At ClueCon - Aug 7-9, 2012 > >> > >> FreeSWITCH-users 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 > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/4a6278a9/attachment-0001.html From msc at freeswitch.org Tue Jul 31 02:05:00 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Jul 2012 15:05:00 -0700 Subject: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15139F23@cantor.std.visionutv.se> Message-ID: Do you have a way to send DTMFs via 2833 or SIP INFO? -MC On Mon, Jul 30, 2012 at 1:25 PM, Yuriy Nasida wrote: > bribge + loopback doesn't work because I have to set application="set" data="playback_terminators=#"/> ON B-LEG which is > affected by soundtouch. > Hm... Any ideas ? > > ------------------------------ > From: nasida at live.ru > To: freeswitch-users at lists.freeswitch.org > Date: Mon, 30 Jul 2012 23:38:34 +0400 > > Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + > inbound dtmf > > Peter, yes I have "start_dtmf" and I can detect DTMF. But! DTMF is not > being recognised if I use > I think soundtouch module changes my dtmf tones (which is inband). > I start thinking about bribge + loopback. This way I will able to use > soundtouch on b-leg only. > > Thanks. > > > From: peter.olsson at visionutveckling.se > > To: freeswitch-users at lists.freeswitch.org > > Date: Mon, 30 Jul 2012 19:23:09 +0000 > > Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + > inbound dtmf > > > > If it's inband you need to execute "start_dtmf" application first. Then > it should detect DTMF's for you. > > > > /Peter > > ________________________________ > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] f?r Yuriy Nasida [ > nasida at live.ru] > > Skickat: den 30 juli 2012 21:00 > > Till: freeswitch-users at lists.freeswitch.org > > ?mne: [Freeswitch-users] soundtouch with playback_terminators + inbound > dtmf > > > > Hello guys, > > > > I change my voice via 'soundtouch' and I try to record this via simple > 'record' command. > > > > > > > > I have to stop recording after '#' but can not make this because I must > recieve DTMF inband. > > I.e. 'soundtouch' changes dtmf as well and FS doesn't recognise dtmf. > > How can I stop recording? > > > > Any ideas? > > Thanks. > > !DSPAM:5016d77b32761519715207! > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/23f2225a/attachment.html From msc at freeswitch.org Tue Jul 31 02:10:23 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Jul 2012 15:10:23 -0700 Subject: [Freeswitch-users] How to use TTS/flite via mod_socket In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD02304F5C38@NY1-EXMB-01.ip-soft.net> References: <5016D887.5060806@daitangroup.com> <6A6B4C284AD15042B429EB9D904544AD02304F5C38@NY1-EXMB-01.ip-soft.net> Message-ID: On Mon, Jul 30, 2012 at 12:08 PM, Hector Geraldino < Hector.Geraldino at ipsoft.com> wrote: > I don?t know what API you?re using to handle ESL connections, but in > general all you need to do is to send an execute command with sendmsg:**** > > ** ** > > sendmsg **** > > call-command: execute**** > > execute-app-name: speak**** > > execute-app-arg: flite|kal|Hello world**** > > ** ** > > ** > The method Hector suggests is good if you are doing a raw outbound event socket connection. I'd recommend using ESL and the programming language of your choice to help abstract away some of the unnecessary details. Also, if for some reason you need to execute the command on an arbitrary UUID then you can use uuid_broadcast. From fs_cli you can test it: uuid_broadcast speak::flite|kal|'hello word' bleg Replace 'bleg' with 'aleg' or 'both' depending on exactly you want to accomplish, i.e. play on the inbound leg, outbound leg, or both. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/72918aa7/attachment.html From sgaleotti at ition.com.ar Tue Jul 31 02:12:39 2012 From: sgaleotti at ition.com.ar (Sergio Galeotti (ITION)) Date: Mon, 30 Jul 2012 19:12:39 -0300 Subject: [Freeswitch-users] Skypeopen Message-ID: <501706D7.3040403@ition.com.ar> Hi; I'm new in FS user list. There is someone who is using the Skype_Open module for incoming and outgoing? I would like to contact someone who is experiencing as it does not find much information about it and I have many doubts about the operation. I have the platform installed and configured, but there are some things if they are not bugs, limitations or poor configuration on my part thanks!! Sergio From nasida at live.ru Tue Jul 31 02:16:14 2012 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 31 Jul 2012 02:16:14 +0400 Subject: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf In-Reply-To: References: , <1FFF97C269757C458224B7C895F35F15139F23@cantor.std.visionutv.se>, , , Message-ID: Unfortunately not. My inbound carrier can not do this.May be I will able to detect '#' which was changed by 'soundtouch' by means of 'tone_detect'.http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detectBut not sure. This looks not good. Date: Mon, 30 Jul 2012 15:05:00 -0700 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf Do you have a way to send DTMFs via 2833 or SIP INFO? -MC On Mon, Jul 30, 2012 at 1:25 PM, Yuriy Nasida wrote: bribge + loopback doesn't work because I have to set ON B-LEG which is affected by soundtouch. Hm... Any ideas ? From: nasida at live.ru To: freeswitch-users at lists.freeswitch.org Date: Mon, 30 Jul 2012 23:38:34 +0400 Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf Peter, yes I have "start_dtmf" and I can detect DTMF. But! DTMF is not being recognised if I use I think soundtouch module changes my dtmf tones (which is inband). I start thinking about bribge + loopback. This way I will able to use soundtouch on b-leg only. Thanks. > From: peter.olsson at visionutveckling.se > To: freeswitch-users at lists.freeswitch.org > Date: Mon, 30 Jul 2012 19:23:09 +0000 > Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf > > If it's inband you need to execute "start_dtmf" application first. Then it should detect DTMF's for you. > > /Peter > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Yuriy Nasida [nasida at live.ru] > Skickat: den 30 juli 2012 21:00 > Till: freeswitch-users at lists.freeswitch.org > ?mne: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf > > Hello guys, > > I change my voice via 'soundtouch' and I try to record this via simple 'record' command. > > > > I have to stop recording after '#' but can not make this because I must recieve DTMF inband. > I.e. 'soundtouch' changes dtmf as well and FS doesn't recognise dtmf. > How can I stop recording? > > Any ideas? > Thanks. > !DSPAM:5016d77b32761519715207! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120731/bcb5de46/attachment-0001.html From msc at freeswitch.org Tue Jul 31 02:25:59 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Jul 2012 15:25:59 -0700 Subject: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15139F23@cantor.std.visionutv.se> Message-ID: Question: what is the application here? Are you modifying the user's voice? It sounds to me like you are stuck. I suppose you could run the audio of a DTMF key through sound touch and then open it in a spectrum analyzer to see what happens. If it's predictable then maybe you could detect the resulting tones with tone_detect, but that just reeks of nasty hackiness. A better solution would be a carrier who can deliver 2833 DTMFs. -MC On Mon, Jul 30, 2012 at 3:16 PM, Yuriy Nasida wrote: > Unfortunately not. My inbound carrier can not do this. > May be I will able to detect '#' which was changed by 'soundtouch' by > means of 'tone_detect'. > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect > But not sure. This looks not good. > > ------------------------------ > Date: Mon, 30 Jul 2012 15:05:00 -0700 > From: msc at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + > inbound dtmf > > Do you have a way to send DTMFs via 2833 or SIP INFO? > -MC > > On Mon, Jul 30, 2012 at 1:25 PM, Yuriy Nasida wrote: > > bribge + loopback doesn't work because I have to set application="set" data="playback_terminators=#"/> ON B-LEG which is > affected by soundtouch. > Hm... Any ideas ? > > ------------------------------ > From: nasida at live.ru > To: freeswitch-users at lists.freeswitch.org > Date: Mon, 30 Jul 2012 23:38:34 +0400 > > Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + > inbound dtmf > > Peter, yes I have "start_dtmf" and I can detect DTMF. But! DTMF is not > being recognised if I use > I think soundtouch module changes my dtmf tones (which is inband). > I start thinking about bribge + loopback. This way I will able to use > soundtouch on b-leg only. > > Thanks. > > > From: peter.olsson at visionutveckling.se > > To: freeswitch-users at lists.freeswitch.org > > Date: Mon, 30 Jul 2012 19:23:09 +0000 > > Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + > inbound dtmf > > > > If it's inband you need to execute "start_dtmf" application first. Then > it should detect DTMF's for you. > > > > /Peter > > ________________________________ > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] f?r Yuriy Nasida [ > nasida at live.ru] > > Skickat: den 30 juli 2012 21:00 > > Till: freeswitch-users at lists.freeswitch.org > > ?mne: [Freeswitch-users] soundtouch with playback_terminators + inbound > dtmf > > > > Hello guys, > > > > I change my voice via 'soundtouch' and I try to record this via simple > 'record' command. > > > > > > > > I have to stop recording after '#' but can not make this because I must > recieve DTMF inband. > > I.e. 'soundtouch' changes dtmf as well and FS doesn't recognise dtmf. > > How can I stop recording? > > > > Any ideas? > > Thanks. > > !DSPAM:5016d77b32761519715207! > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/e705d006/attachment.html From marketing at cluecon.com Tue Jul 31 02:49:23 2012 From: marketing at cluecon.com (Michael Collins) Date: Mon, 30 Jul 2012 15:49:23 -0700 Subject: [Freeswitch-users] IMPORTANT: Last chance to book at the Hyatt Message-ID: Greetings! I wanted to let you know that the Hyatt now has less than 10 rooms available for ClueCon 2012 this year! If you plan on booking please do so * immediately*. You can call or click the link on our ClueCon hotels page . See you next week! -- Michael S Collins ClueCon Team http://www.cluecon.com 877-7-4ACLUE -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/70056da3/attachment.html From msc at freeswitch.org Tue Jul 31 02:58:27 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Jul 2012 15:58:27 -0700 Subject: [Freeswitch-users] PAGD 0 digits In-Reply-To: <50128323.7000008@thewinelake.com> References: <9D0E55F4-768E-4C9D-B1DF-9D074E44ADD7@mgtech.com> <458A0010-EC8C-4ACC-B08D-A02E0AD26077@mgtech.com> <500FA9BA.5030303@thewinelake.com> <50110823.2060604@thewinelake.com> <50128323.7000008@thewinelake.com> Message-ID: After the timeout it simply moves on to the next application. Would that not suffice? -MC On Fri, Jul 27, 2012 at 5:01 AM, Alex wrote: > If I want to allow someone to enter an empty string of digits, how is it > done? > > I've tried various things... > > My latest effort (that still doesn't work!) is > > keypress = session:playAndGetDigits(1, 1, 3, 5000, "*#", > what_to_play, "", "\\d{0,1}|\\#|\\*") > terminator = session:getVariable("read_terminator_used") > > I note that the minimum value for the first param is 1 - so maybe it > can't be done? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/f3d227cf/attachment-0001.html From nasida at live.ru Tue Jul 31 03:26:02 2012 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 31 Jul 2012 03:26:02 +0400 Subject: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf In-Reply-To: References: , <1FFF97C269757C458224B7C895F35F15139F23@cantor.std.visionutv.se>, , , , , Message-ID: It will just the funny application which will allow customers to record his changed voice which will sounds like Darth Vader for example:) and to send to the destination number. It is easy but I can not create method for stop of recording from user side by means of DTMF because they is inband and 'soundtouch' modifies dtmf as well.Yep you are correct. 'tone_detect' looks like nasty hackiness in this case but probably I will have to create this. Date: Mon, 30 Jul 2012 15:25:59 -0700 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf Question: what is the application here? Are you modifying the user's voice? It sounds to me like you are stuck. I suppose you could run the audio of a DTMF key through sound touch and then open it in a spectrum analyzer to see what happens. If it's predictable then maybe you could detect the resulting tones with tone_detect, but that just reeks of nasty hackiness. A better solution would be a carrier who can deliver 2833 DTMFs. -MC On Mon, Jul 30, 2012 at 3:16 PM, Yuriy Nasida wrote: Unfortunately not. My inbound carrier can not do this.May be I will able to detect '#' which was changed by 'soundtouch' by means of 'tone_detect'.http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect But not sure. This looks not good. Date: Mon, 30 Jul 2012 15:05:00 -0700 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf Do you have a way to send DTMFs via 2833 or SIP INFO? -MC On Mon, Jul 30, 2012 at 1:25 PM, Yuriy Nasida wrote: bribge + loopback doesn't work because I have to set ON B-LEG which is affected by soundtouch. Hm... Any ideas ? From: nasida at live.ru To: freeswitch-users at lists.freeswitch.org Date: Mon, 30 Jul 2012 23:38:34 +0400 Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf Peter, yes I have "start_dtmf" and I can detect DTMF. But! DTMF is not being recognised if I use I think soundtouch module changes my dtmf tones (which is inband). I start thinking about bribge + loopback. This way I will able to use soundtouch on b-leg only. Thanks. > From: peter.olsson at visionutveckling.se > To: freeswitch-users at lists.freeswitch.org > Date: Mon, 30 Jul 2012 19:23:09 +0000 > Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf > > If it's inband you need to execute "start_dtmf" application first. Then it should detect DTMF's for you. > > /Peter > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Yuriy Nasida [nasida at live.ru] > Skickat: den 30 juli 2012 21:00 > Till: freeswitch-users at lists.freeswitch.org > ?mne: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf > > Hello guys, > > I change my voice via 'soundtouch' and I try to record this via simple 'record' command. > > > > I have to stop recording after '#' but can not make this because I must recieve DTMF inband. > I.e. 'soundtouch' changes dtmf as well and FS doesn't recognise dtmf. > How can I stop recording? > > Any ideas? > Thanks. > !DSPAM:5016d77b32761519715207! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120731/b5f832c5/attachment.html From msc at freeswitch.org Tue Jul 31 03:32:48 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 30 Jul 2012 16:32:48 -0700 Subject: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15139F23@cantor.std.visionutv.se> Message-ID: Either that or have them hit the mute button and wait several seconds. Hopefully silence still works. :P You can even use sox to post-process and remove trailing silence from the recording. -MC On Mon, Jul 30, 2012 at 4:26 PM, Yuriy Nasida wrote: > It will just the funny application which will allow customers to record > his changed voice which will sounds like Darth Vader for example:) and to > send to the destination number. It is easy but I can not create method for > stop of recording from user side by means of DTMF because they is inband > and 'soundtouch' modifies dtmf as well. > Yep you are correct. 'tone_detect' looks like nasty hackiness in this case > but probably I will have to create this. > > ------------------------------ > Date: Mon, 30 Jul 2012 15:25:59 -0700 > From: msc at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + > inbound dtmf > > Question: what is the application here? Are you modifying the user's > voice? It sounds to me like you are stuck. I suppose you could run the > audio of a DTMF key through sound touch and then open it in a spectrum > analyzer to see what happens. If it's predictable then maybe you could > detect the resulting tones with tone_detect, but that just reeks of nasty > hackiness. A better solution would be a carrier who can deliver 2833 DTMFs. > > -MC > > On Mon, Jul 30, 2012 at 3:16 PM, Yuriy Nasida wrote: > > Unfortunately not. My inbound carrier can not do this. > May be I will able to detect '#' which was changed by 'soundtouch' by > means of 'tone_detect'. > http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect > But not sure. This looks not good. > > ------------------------------ > Date: Mon, 30 Jul 2012 15:05:00 -0700 > From: msc at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + > inbound dtmf > > Do you have a way to send DTMFs via 2833 or SIP INFO? > -MC > > On Mon, Jul 30, 2012 at 1:25 PM, Yuriy Nasida wrote: > > bribge + loopback doesn't work because I have to set application="set" data="playback_terminators=#"/> ON B-LEG which is > affected by soundtouch. > Hm... Any ideas ? > > ------------------------------ > From: nasida at live.ru > To: freeswitch-users at lists.freeswitch.org > Date: Mon, 30 Jul 2012 23:38:34 +0400 > > Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + > inbound dtmf > > Peter, yes I have "start_dtmf" and I can detect DTMF. But! DTMF is not > being recognised if I use > I think soundtouch module changes my dtmf tones (which is inband). > I start thinking about bribge + loopback. This way I will able to use > soundtouch on b-leg only. > > Thanks. > > > From: peter.olsson at visionutveckling.se > > To: freeswitch-users at lists.freeswitch.org > > Date: Mon, 30 Jul 2012 19:23:09 +0000 > > Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + > inbound dtmf > > > > If it's inband you need to execute "start_dtmf" application first. Then > it should detect DTMF's for you. > > > > /Peter > > ________________________________ > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] f?r Yuriy Nasida [ > nasida at live.ru] > > Skickat: den 30 juli 2012 21:00 > > Till: freeswitch-users at lists.freeswitch.org > > ?mne: [Freeswitch-users] soundtouch with playback_terminators + inbound > dtmf > > > > Hello guys, > > > > I change my voice via 'soundtouch' and I try to record this via simple > 'record' command. > > > > > > > > I have to stop recording after '#' but can not make this because I must > recieve DTMF inband. > > I.e. 'soundtouch' changes dtmf as well and FS doesn't recognise dtmf. > > How can I stop recording? > > > > Any ideas? > > Thanks. > > !DSPAM:5016d77b32761519715207! > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/637ca4ba/attachment-0001.html From cmrienzo at gmail.com Tue Jul 31 03:53:02 2012 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Mon, 30 Jul 2012 19:53:02 -0400 Subject: [Freeswitch-users] Am I using play_and_detect_speech correctly? In-Reply-To: References: Message-ID: You are using that APP in a way that hasn't been tested. Try to do "speech_detect pause" and see if that helps. It will reserve your ASR session for reuse while "speech_detect stop" will tear it down completely. If that doesn't work, open a jira ticket and attach a full call trace. It's difficult to understand exactly what is happening from your description. Chris On Mon, Jul 30, 2012 at 5:38 PM, Liu, Xingkun wrote: > ** > > Hello, > > I am using play_and_detect_speech with Java ESL in my IVR applications. > > Previously I call it again each time after I receive any recognition event, > like recognition complete, no-input-timeout, or recognition-timeout, > it seems to work fine. > > Now I have changed my app to issue play_and_detect_speech command based on > my available system utterances as well as the speech event. > > I.e., I use a separate thread to constantly check if there is a system > utterance coming in > from another component of my application, if there is any utterance I > issue the command which will > speak the new utterance and listen to user input no matter whether or not > previous > command has finished. And if there is any speech event (recognition > result, timeout etc.) > the play_and_detect_speech command is also issued but with playing silence. > > Obviously the new command will stop the utterance speaking of the previous > command if it is not finished. > > My question is > > will the new play_and_detect_speech command also stop the previous ASR > listening > or will there be many ASR listening channel and sending speech data (or > silence) to ASR server? > > Do I need to explicitly issue a "stop" commnad before issuing a new > play_and_detect_speech? > If yes, how to do that, by "detect_speech stop"? > > Recently there is a network traffice problem (lots of connections /data > transportation to the ASR server machine) > when I am running my application. I am not sure if this is because of > other issues > or because of my new changes to the way of using play_and_detect_speech. > > Please any one could shed a light on this? > > Many thanks! > > Xing > > > ------------------------------ > > *Heriot-Watt University is the Sunday Times Scottish University of the > Year 2011-2012.* > > We invite research leaders and ambitious early career researchers to join > us in leading and driving research in key inter-disciplinary themes. Please > see www.hw.ac.uk/researchleaders for further information and how to > apply. > > Heriot-Watt University is a Scottish charity registered under charity > number SC000278. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120730/87ec9b1f/attachment.html From cmrienzo at gmail.com Tue Jul 31 04:00:40 2012 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Mon, 30 Jul 2012 20:00:40 -0400 Subject: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf In-Reply-To: References: <1FFF97C269757C458224B7C895F35F15139F23@cantor.std.visionutv.se> Message-ID: Did you execute start_dtmf BEFORE soundtouch like Peter recommended? Assuming the bugs are executed in order of creation, the dtmf detector should read the clean frame before soundtouch mangles it. Chris On Mon, Jul 30, 2012 at 4:25 PM, Yuriy Nasida wrote: > bribge + loopback doesn't work because I have to set application="set" data="playback_terminators=#"/> ON B-LEG which is > affected by soundtouch. > Hm... Any ideas ? > > ------------------------------ > From: nasida at live.ru > To: freeswitch-users at lists.freeswitch.org > Date: Mon, 30 Jul 2012 23:38:34 +0400 > > Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + > inbound dtmf > > Peter, yes I have "start_dtmf" and I can detect DTMF. But! DTMF is not > being recognised if I use > I think soundtouch module changes my dtmf tones (which is inband). > I start thinking about bribge + loopback. This way I will able to use > soundtouch on b-leg only. > > Thanks. > > > From: peter.olsson at visionutveckling.se > > To: freeswitch-users at lists.freeswitch.org > > Date: Mon, 30 Jul 2012 19:23:09 +0000 > > Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + > inbound dtmf > > > > If it's inband you need to execute "start_dtmf" application first. Then > it should detect DTMF's for you. > > > > /Peter > > ________________________________ > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] f?r Yuriy Nasida [ > nasida at live.ru] > > Skickat: den 30 juli 2012 21:00 > > Till: freeswitch-users at lists.freeswitch.org > > ?mne: [Freeswitch-users] soundtouch with playback_terminators + inbound > dtmf > > > > Hello guys, > > > > I change my voice via 'soundtouch' and I try to record this via simple > 'record' command. > > > > > > > > I have to stop recording after '#' but can not make this because I must > recieve DTMF inband. > > I.e. 'soundtouch' changes dtmf as well and FS doesn't recognise dtmf. > > How can I stop recording? > > > > Any ideas? > > Thanks. > > !DSPAM:5016d77b32761519715207! > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: consulting at freeswitch.org > http://www.freeswitchsolutions.com FreeSWITCH-powered IP PBX: The CudaTel > Communication Server Official FreeSWITCH Sites > http://www.freeswitch.org http://wiki.freeswitch.org > http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 > FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120730/bdda0351/attachment.html From nasida at live.ru Tue Jul 31 04:03:53 2012 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 31 Jul 2012 04:03:53 +0400 Subject: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf In-Reply-To: References: , <1FFF97C269757C458224B7C895F35F15139F23@cantor.std.visionutv.se>, , , , , , , Message-ID: Yeah I thought about mute button but I suppose customers will not happy with such instruction :DThanks anyway! Date: Mon, 30 Jul 2012 16:32:48 -0700 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf Either that or have them hit the mute button and wait several seconds. Hopefully silence still works. :P You can even use sox to post-process and remove trailing silence from the recording. -MC On Mon, Jul 30, 2012 at 4:26 PM, Yuriy Nasida wrote: It will just the funny application which will allow customers to record his changed voice which will sounds like Darth Vader for example:) and to send to the destination number. It is easy but I can not create method for stop of recording from user side by means of DTMF because they is inband and 'soundtouch' modifies dtmf as well. Yep you are correct. 'tone_detect' looks like nasty hackiness in this case but probably I will have to create this. Date: Mon, 30 Jul 2012 15:25:59 -0700 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf Question: what is the application here? Are you modifying the user's voice? It sounds to me like you are stuck. I suppose you could run the audio of a DTMF key through sound touch and then open it in a spectrum analyzer to see what happens. If it's predictable then maybe you could detect the resulting tones with tone_detect, but that just reeks of nasty hackiness. A better solution would be a carrier who can deliver 2833 DTMFs. -MC On Mon, Jul 30, 2012 at 3:16 PM, Yuriy Nasida wrote: Unfortunately not. My inbound carrier can not do this.May be I will able to detect '#' which was changed by 'soundtouch' by means of 'tone_detect'.http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect But not sure. This looks not good. Date: Mon, 30 Jul 2012 15:05:00 -0700 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf Do you have a way to send DTMFs via 2833 or SIP INFO? -MC On Mon, Jul 30, 2012 at 1:25 PM, Yuriy Nasida wrote: bribge + loopback doesn't work because I have to set ON B-LEG which is affected by soundtouch. Hm... Any ideas ? From: nasida at live.ru To: freeswitch-users at lists.freeswitch.org Date: Mon, 30 Jul 2012 23:38:34 +0400 Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf Peter, yes I have "start_dtmf" and I can detect DTMF. But! DTMF is not being recognised if I use I think soundtouch module changes my dtmf tones (which is inband). I start thinking about bribge + loopback. This way I will able to use soundtouch on b-leg only. Thanks. > From: peter.olsson at visionutveckling.se > To: freeswitch-users at lists.freeswitch.org > Date: Mon, 30 Jul 2012 19:23:09 +0000 > Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf > > If it's inband you need to execute "start_dtmf" application first. Then it should detect DTMF's for you. > > /Peter > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Yuriy Nasida [nasida at live.ru] > Skickat: den 30 juli 2012 21:00 > Till: freeswitch-users at lists.freeswitch.org > ?mne: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf > > Hello guys, > > I change my voice via 'soundtouch' and I try to record this via simple 'record' command. > > > > I have to stop recording after '#' but can not make this because I must recieve DTMF inband. > I.e. 'soundtouch' changes dtmf as well and FS doesn't recognise dtmf. > How can I stop recording? > > Any ideas? > Thanks. > !DSPAM:5016d77b32761519715207! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120731/5ca43926/attachment-0001.html From nasida at live.ru Tue Jul 31 04:39:29 2012 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 31 Jul 2012 04:39:29 +0400 Subject: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf In-Reply-To: References: , <1FFF97C269757C458224B7C895F35F15139F23@cantor.std.visionutv.se>, , , Message-ID: Yes I have done this first of all. I invoke soundtouch before 'record' only and I stop soundtouch after 'record'. I use 'start_dtmf' in the beginning of my lua script. I see that my inband dtmf doesn't work only while record process. I have good dtmf recognition for all other part of my menu. Unfortunately the order of creation is indifferent. Date: Mon, 30 Jul 2012 20:00:40 -0400 From: cmrienzo at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf Did you execute start_dtmf BEFORE soundtouch like Peter recommended? Assuming the bugs are executed in order of creation, the dtmf detector should read the clean frame before soundtouch mangles it. Chris On Mon, Jul 30, 2012 at 4:25 PM, Yuriy Nasida wrote: bribge + loopback doesn't work because I have to set ON B-LEG which is affected by soundtouch. Hm... Any ideas ? From: nasida at live.ru To: freeswitch-users at lists.freeswitch.org Date: Mon, 30 Jul 2012 23:38:34 +0400 Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf Peter, yes I have "start_dtmf" and I can detect DTMF. But! DTMF is not being recognised if I use I think soundtouch module changes my dtmf tones (which is inband). I start thinking about bribge + loopback. This way I will able to use soundtouch on b-leg only. Thanks. > From: peter.olsson at visionutveckling.se > To: freeswitch-users at lists.freeswitch.org > Date: Mon, 30 Jul 2012 19:23:09 +0000 > Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf > > If it's inband you need to execute "start_dtmf" application first. Then it should detect DTMF's for you. > > /Peter > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Yuriy Nasida [nasida at live.ru] > Skickat: den 30 juli 2012 21:00 > Till: freeswitch-users at lists.freeswitch.org > ?mne: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf > > Hello guys, > > I change my voice via 'soundtouch' and I try to record this via simple 'record' command. > > > > I have to stop recording after '#' but can not make this because I must recieve DTMF inband. > I.e. 'soundtouch' changes dtmf as well and FS doesn't recognise dtmf. > How can I stop recording? > > Any ideas? > Thanks. > !DSPAM:5016d77b32761519715207! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120731/9d312eec/attachment.html From nasida at live.ru Tue Jul 31 04:46:25 2012 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 31 Jul 2012 04:46:25 +0400 Subject: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf In-Reply-To: References: , <1FFF97C269757C458224B7C895F35F15139F23@cantor.std.visionutv.se>, , , , Message-ID: Also I hear in wav file that dtmf tones are changed. They began to sound like Darth Vader's dtmf too ... :) From: nasida at live.ru To: freeswitch-users at lists.freeswitch.org Subject: RE: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf Date: Tue, 31 Jul 2012 04:39:29 +0400 Yes I have done this first of all. I invoke soundtouch before 'record' only and I stop soundtouch after 'record'. I use 'start_dtmf' in the beginning of my lua script. I see that my inband dtmf doesn't work only while record process. I have good dtmf recognition for all other part of my menu. Unfortunately the order of creation is indifferent. Date: Mon, 30 Jul 2012 20:00:40 -0400 From: cmrienzo at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf Did you execute start_dtmf BEFORE soundtouch like Peter recommended? Assuming the bugs are executed in order of creation, the dtmf detector should read the clean frame before soundtouch mangles it. Chris On Mon, Jul 30, 2012 at 4:25 PM, Yuriy Nasida wrote: bribge + loopback doesn't work because I have to set ON B-LEG which is affected by soundtouch. Hm... Any ideas ? From: nasida at live.ru To: freeswitch-users at lists.freeswitch.org Date: Mon, 30 Jul 2012 23:38:34 +0400 Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf Peter, yes I have "start_dtmf" and I can detect DTMF. But! DTMF is not being recognised if I use I think soundtouch module changes my dtmf tones (which is inband). I start thinking about bribge + loopback. This way I will able to use soundtouch on b-leg only. Thanks. > From: peter.olsson at visionutveckling.se > To: freeswitch-users at lists.freeswitch.org > Date: Mon, 30 Jul 2012 19:23:09 +0000 > Subject: Re: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf > > If it's inband you need to execute "start_dtmf" application first. Then it should detect DTMF's for you. > > /Peter > ________________________________ > Fr?n: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] f?r Yuriy Nasida [nasida at live.ru] > Skickat: den 30 juli 2012 21:00 > Till: freeswitch-users at lists.freeswitch.org > ?mne: [Freeswitch-users] soundtouch with playback_terminators + inbound dtmf > > Hello guys, > > I change my voice via 'soundtouch' and I try to record this via simple 'record' command. > > > > I have to stop recording after '#' but can not make this because I must recieve DTMF inband. > I.e. 'soundtouch' changes dtmf as well and FS doesn't recognise dtmf. > How can I stop recording? > > Any ideas? > Thanks. > !DSPAM:5016d77b32761519715207! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at 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/20120731/bb289870/attachment-0001.html From sdevoy at bizfocused.com Tue Jul 31 06:09:50 2012 From: sdevoy at bizfocused.com (Sean Devoy) Date: Mon, 30 Jul 2012 22:09:50 -0400 Subject: [Freeswitch-users] SIP to make phone reboot or resync In-Reply-To: References: <07b501cd6c1a$238b3e90$6aa1bbb0$@bizfocused.com> <094e01cd6c43$9e44cc90$dace65b0$@bizfocused.com> <0c4001cd6cdf$b2603a10$1720ae30$@bizfocused.com> <0deb01cd6d3f$67e43a30$37acae90$@bizfocused.com> Message-ID: <184e01cd6ec1$915f0740$b41d15c0$@bizfocused.com> Michael, I Googled "freeswitch sip re-provision phone" which came up with almost nothing. This thread is now the TOP 5 hits on that query! I also Googled "freeswitch sip reboot phone" which comes up with several links that appear to be polycom specific. In fact if I had read more I may have gotten what I needed with some brute force trial and error. I must say I was expecting to find sendevent code to achieve this, not something as simple as a SOFIA command! Here again, as you can see from my follow up question, I had no idea what to search for with regard to reverse-auth-user and reverse-auth-pass params. You guys rock. Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Monday, July 30, 2012 6:03 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync On Mon, Jul 30, 2012 at 2:43 PM, Anthony Minessale wrote: reverse-auth-user and reverse-auth-pass params in the params section of the xml tag in the user directory can be configured instead of a gateway and it will use those credentials instead. FYI this is actually on the wiki (thanks kn0x): http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide#reverse_authenticat ion It might be helpful to make it easier to find, perhaps by linking to it from another page. Sean, can you tell us where on the wiki you were looking for this? That will give me an idea of where exactly I should add the link & info. -MC On Sat, Jul 28, 2012 at 11:05 PM, Sean Devoy wrote: > Thank you sir. That did help, I can resync my phone test either with > user/pass using any proxy user/pass from the lines on THAT phone. I can > also disable Auth Resync Reboot and let anyone send reboots - which might be > fun, but seems like a bad plan! > > This is not very useful for me since the phone's proxy user/pass must be > specified in the gateway XML for whole realm in the the SIP profile. I > hoped to be able to re-provision any phone on demand. It seems impractical > to have to restart/rescan my sofia gateway every time I want to issue a > resync to a different phone. > > Am I missing something? > > Sean > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > Vallimamod ABDULLAH > Sent: Saturday, July 28, 2012 1:05 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync > > Hi, > > It's normally the sip account credentials. There is also an option in the > admin interface to disable this authentication ("Auth Resync-Reboot" option > if I recall correctly in Ext tab) > > Best Regards, > Vallimamod Abdullah > . > > > On Jul 28, 2012, at 6:40 PM, Sean Devoy wrote: > >> HI all, >> >> I am close, but I still don't understand what credentials are required. > The response is: >> SIP/2.0 401 Unauthorized >> >> I have tried the web admin credentials for the phone, I don't k now what > FS credentials I could pass. >> >> Any ideas? >> >> Thanks, >> Sean >> >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Anthony Minessale >> Sent: Friday, July 27, 2012 6:12 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync >> >> Make a gateway with no reg and the credentials and name it after the realm > in the challenge. >> >> On Jul 27, 2012 5:05 PM, "Sean Devoy" wrote: >> Thank Anthony, BUT.. Error: >> 2012-07-27 17:58:55.118642 [ERR] sofia_reg.c:2165 Cannot locate any >> authentication credentials to complete an authentication request for >> realm '"fs_bfis.bizfocused.com"' >> >> 1 - where do I specify the credentials? >> 2 - Are these Freeswitch credentials or phone credentials? I suspect >> the latter. >> >> -----Original Message----- >> From: freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of >> Anthony Minessale >> Sent: Friday, July 27, 2012 1:21 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] SIP to make phone reboot or resync >> >> sofia profile check_sync [call_id] // no call_id means >> all sofia profile flush_inbound_reg [call_id] [reboot] >> >> >> >> On Fri, Jul 27, 2012 at 12:06 PM, Sean Devoy > wrote: >> > HI All, >> > >> > >> > >> > Does anyone have an code that will cause Freeswitch to send a SIP >> > message to a CISCO SPA5xx phone that will cause it to reboot or >> > resync (aka re-provision)? I know about the URLs that cause the >> > phone to do this, but they are NATed, so I need to use SIP to hit them. >> > >> > >> > >> > Assume I am programming language omni-lingual for this request! >> > >> > >> > >> > Thanks, >> > >> > Sean >> > >> > >> > ____________________________________________________________________ >> > __ ___ Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u >> > se >> > rs >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users 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 >> >> >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users 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 >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ Twitter: http://twitter.com/FreeSWITCH_wire AIM: anthm MSN:anthony_minessale at hotmail.com GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888 at conference.freeswitch.org googletalk:conf+888 at conference.freeswitch.org pstn:+19193869900 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120730/0faa09da/attachment-0001.html From govoiper at gmail.com Tue Jul 31 08:36:45 2012 From: govoiper at gmail.com (SamyGo) Date: Tue, 31 Jul 2012 09:36:45 +0500 Subject: [Freeswitch-users] Bind Sofia external profile to Static private IP In-Reply-To: <201207301113.33629.g.d.monnezza@tiscali.it> References: <4664115619900296997@unknownmsgid> <201207301113.33629.g.d.monnezza@tiscali.it> Message-ID: Hi, Thanks for this, Yes I initially did start FS with -no-nat and it seemed to fix it, and after your this email I plugged in "ext-*" params in the profile and started with autoNAT and this also works. The autoNAT engine in Freeswitch is > powerful. It works with uPnP, like skype does. So, working with most of > the internet gateways (like > home adsl routers) it is capable to open a port to communicate on the > public network ^ Is definitely a very useful piece of info to be remembered. Thanks for this :) Thanks, Sammy On Mon, Jul 30, 2012 at 2:13 PM, g wrote: > Hi Brian. Try playing with ext-rtp-ip directive. > i,e. force in your sip profile > > > > Or, if you prefer, start Freeswitch with -nonat option. > > A more sophisticated approach is to play with auto-nat. The autoNAT engine > in Freeswitch is > powerful. It works with uPnP, like skype does. So, working with most of > the internet gateways (like > home adsl routers) it is capable to open a port to communicate on the > public network. > Hope it helps > > g > > > On Saturday 28 July 2012 16:34:37 SamyGo wrote: > > yes I tried that too edited the profile external.xml directly but in > > vain..so I was reading another thread here of a person who had totally > > inverse issue and Michael asked if he is starting freeswitch with -nonat > , > > applying that switch while starting freeswitch seems to have worked here. > > > > On Sat, Jul 28, 2012 at 6:34 PM, Brian West > wrote: > > > Open the profile directly and edit it, vars.xml is for ease not > > > flexibility. > > > > > > -- > > > Brian West > > > brian at freeswitch.org > > > FreeSWITCH Solutions, LLC > > > PO BOX PO BOX 2531 > > > Brookfield, WI 53008-2531 > > > Twitter: @FreeSWITCH_Wire > > > T: +1.213.286.0410 | F: +1.213.286.0401 | M: +1.918.424.WEST > > > iNUM: +883 5100 1286 0410 > > > > > > On Jul 28, 2012, at 8:31 AM, SamyGo wrote: > > > > Hello, > > > > Probably an easy thing but my luck I'm stuck in this awkward > situation > > > > > > here, I want my sofia external profile to bind to my private IP: > > > 192.168.15.21 , so for the purpose I've edited the vars.xml and > > > sip_profiles/external.xml but still everytime I restart freeswitch it > > > detects the Public IP and binds thye external-sip-ip and > external-rtp-ip > > > to the Public IP ? > > > > > > > I'm very confused at this and followed all the previous mailing lists > > > > > > threads and to-dos as well but in vain. > > > > > > > Waiting for a savior here. > > > > -- > > > > Thanks > > > > Sammy Go. > > > > > > > > > > > > > _______________________________________________________________________ > > > > __ Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-user > > > > s http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120731/dc471aab/attachment.html From govoiper at gmail.com Tue Jul 31 08:40:24 2012 From: govoiper at gmail.com (SamyGo) Date: Tue, 31 Jul 2012 09:40:24 +0500 Subject: [Freeswitch-users] LUA session:Bridge not actually bridging calls ~ In-Reply-To: References: Message-ID: Though the problem is solved but is there any corresponding WARN or anything in the logs so that anyone can pick the problem right away. This fixing of codecs took alot of time and I was never expecting that this could be the cause Thinking that its Bridge() which is not doing its part. Thanks a lot for your time on this. Regards, Sammy On Mon, Jul 30, 2012 at 6:22 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > This script is actually a bit over-complicated > > dialA = "sofia/gateway/fs1/9903" > dialB = "user/1001" > legA = freeswitch.Session(dialA) > > if (legA:ready()) { > legA:execute("bridge", dialB) > } > > I think the problem is that we don't support dealing with many codecs > in the 200ok > They send gsm and ulaw and then choose ulaw and we go with the first one > gsm > > > > On Sat, Jul 28, 2012 at 11:23 AM, SamyGo wrote: > > Hi again, > > > > So, It was a very minor change in configuration and it was working. > > Basically FreeSwicth was bridging the two legs BUT there was a codec > issue. > > All I had to do was in Asterisk (serving as my gateway) to allow only > ulaw > > and alaw > > i.e > > > > disallow=all > > allow=ulaw > > allow=alaw > > > > What I really really wish to know is that why there was no indication of > > codec mismatch or ptime mismatch or sample rate mismatch while > transcoding > > or anything. > > > > It will be fine if none replies but it will be great to know the real > > reason behind this and from where in logs can I verify this !! > > > > Thanks > > Sammy > > > > > > On Sat, Jul 28, 2012 at 8:18 PM, SamyGo wrote: > >> > >> Here are the FS console logs: > >> http://pastebin.freeswitch.org/19595 > >> > >> Please suggest what am I missing here. > >> > >> > >> On Sat, Jul 28, 2012 at 7:59 PM, SamyGo wrote: > >>> > >>> Hello, > >>> I wanted to make a lua script which just dials out two different > numbers > >>> via some external gateway and when both calls are answered they are > just > >>> bridged. For this a very impressive Lua example > >>> http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control is > copied and > >>> all I had to do was change the dialA and dialB strings and its working > great > >>> as far as the SIP signalling is concerned. > >>> > >>> execute this string and I get calls on two different number but things > >>> get interesting when Freeswitch bridge() the two legs. No AUDIO..not > even > >>> one-way. I could see on my own gateway that RTPs for both the legs are > >>> actually forwarded to Freeswitch ! > >>> > >>> On my sip pcap traces analyzing on wireshark I could actually hear the > >>> two persons saying Hello but neither could hear anything. > >>> > >>> The above example lua call_control script is used as it is. > >>> Please suggest. > >>> > >>> Regards > >>> Sammy Go. > >>> > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Anthony Minessale II > > FreeSWITCH http://www.freeswitch.org/ > ClueCon http://www.cluecon.com/ > Twitter: http://twitter.com/FreeSWITCH_wire > > AIM: anthm > MSN:anthony_minessale at hotmail.com > GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com > IRC: irc.freenode.net #freeswitch > > FreeSWITCH Developer Conference > sip:888 at conference.freeswitch.org > googletalk:conf+888 at conference.freeswitch.org > pstn:+19193869900 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120731/42e2255b/attachment-0001.html From gmaruzz at gmail.com Tue Jul 31 10:33:11 2012 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Tue, 31 Jul 2012 08:33:11 +0200 Subject: [Freeswitch-users] Skypeopen In-Reply-To: <501706D7.3040403@ition.com.ar> References: <501706D7.3040403@ition.com.ar> Message-ID: have a look at: http://wiki.freeswitch.org/wiki/Skypopen On Tue, Jul 31, 2012 at 12:12 AM, Sergio Galeotti (ITION) < sgaleotti at ition.com.ar> wrote: > Hi; > I'm new in FS user list. > There is someone who is using the Skype_Open module for incoming and > outgoing? > I would like to contact someone who is experiencing as it does not find > much information about it and I have many doubts about the operation. > I have the platform installed and configured, but there are some things > if they are not bugs, limitations or poor configuration on my part > thanks!! > > Sergio > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/81f795dd/attachment.html From virbhati at gmail.com Tue Jul 31 10:48:50 2012 From: virbhati at gmail.com (virendra bhati) Date: Tue, 31 Jul 2012 12:18:50 +0530 Subject: [Freeswitch-users] Nibble billing is not deducted balance Message-ID: Hi Team, I have installed and configure Freeswitch. I want to used Mod_nibblebill for realtime billing with freeswitch. I have configure and installed freeswtich with Mod_nibblebill but balance is not deducted still. Any help will be appreciated.... /******************************************************************/ /******************************************************************/ *FS CLI * 2012-07-31 02:30:02.268212 [DEBUG] switch_ivr_originate.c:1947 Parsing global variables 2012-07-31 02:30:02.268212 [DEBUG] switch_event.c:1519 Parsing variable [plivo_request_uuid]=[28a9ae1a-dad9-11e1-b432-00163e908c9f] 2012-07-31 02:30:02.268212 [DEBUG] switch_event.c:1519 Parsing variable [plivo_answer_url]=[http://46.252.152.160:8008/api/v1/answercall/] 2012-07-31 02:30:02.268212 [DEBUG] switch_event.c:1519 Parsing variable [plivo_hangup_url]=[http://46.252.152.160:8008/api/v1/hangupcall/] 2012-07-31 02:30:02.268212 [DEBUG] switch_event.c:1519 Parsing variable [origination_caller_id_number]=[0498722120] 2012-07-31 02:30:02.268212 [DEBUG] switch_event.c:1519 Parsing variable [bridge_early_media]=[true] 2012-07-31 02:30:02.268212 [DEBUG] switch_event.c:1519 Parsing variable [hangup_after_bridge]=[true] 2012-07-31 02:30:02.268212 [DEBUG] switch_event.c:1519 Parsing variable [hangup_after_bridge]=[true] 2012-07-31 02:30:02.268212 [DEBUG] switch_event.c:1519 Parsing variable [nibble_account]=[97183008] 2012-07-31 02:30:02.268212 [DEBUG] switch_event.c:1519 Parsing variable [plivo_from]=[0498722120] 2012-07-31 02:30:02.268212 [DEBUG] switch_event.c:1519 Parsing variable [plivo_to]=[919718300881] 2012-07-31 02:30:02.268212 [DEBUG] switch_event.c:1519 Parsing variable [plivo_app]=[true] 2012-07-31 02:30:02.268212 [DEBUG] switch_event.c:1519 Parsing variable [absolute_codec_string]=[PCMA] 2012-07-31 02:30:02.268212 [DEBUG] switch_event.c:1519 Parsing variable [originate_timeout]=[60] 2012-07-31 02:30:02.268212 [DEBUG] switch_event.c:1519 Parsing variable [ignore_early_media]=[true] 2012-07-31 02:30:02.268212 [NOTICE] switch_channel.c:926 New Channel sofia/external/919718300881 [28aa5810-dad9-11e1-b60b-1782e99edc0a] 2012-07-31 02:30:02.268212 [DEBUG] mod_sofia.c:4705 (sofia/external/919718300881) State Change CS_NEW -> CS_INIT 2012-07-31 02:30:02.268212 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 02:30:02.268212 [DEBUG] switch_core_state_machine.c:385 (sofia/external/919718300881) Running State Change CS_INIT 2012-07-31 02:30:02.268212 [DEBUG] switch_core_state_machine.c:424 (sofia/external/919718300881) State INIT 2012-07-31 02:30:02.268212 [DEBUG] mod_sofia.c:85 sofia/external/919718300881 SOFIA INIT 2012-07-31 02:30:02.268212 [DEBUG] mod_sofia.c:125 (sofia/external/919718300881) State Change CS_INIT -> CS_ROUTING 2012-07-31 02:30:02.268212 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 02:30:02.268212 [DEBUG] switch_core_state_machine.c:424 (sofia/external/919718300881) State INIT going to sleep 2012-07-31 02:30:02.268212 [DEBUG] switch_core_state_machine.c:385 (sofia/external/919718300881) Running State Change CS_ROUTING 2012-07-31 02:30:02.268212 [DEBUG] switch_channel.c:1919 (sofia/external/919718300881) Callstate Change DOWN -> RINGING 2012-07-31 02:30:02.268212 [DEBUG] switch_core_state_machine.c:433 (sofia/external/919718300881) State ROUTING 2012-07-31 02:30:02.268212 [DEBUG] mod_sofia.c:148 sofia/external/919718300881 SOFIA ROUTING 2012-07-31 02:30:02.268212 [DEBUG] switch_ivr_originate.c:67 (sofia/external/919718300881) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-07-31 02:30:02.268212 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 02:30:02.268212 [DEBUG] mod_nibblebill.c:383 Doing lookup query [SELECT cash AS nibble_balance FROM accounts WHERE id='97183008'] 2012-07-31 02:30:02.268212 [DEBUG] switch_core_session.c:919 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 02:30:02.268212 [DEBUG] mod_nibblebill.c:393 Retrieved current balance for account 97183008 (balance = 500.500000) 2012-07-31 02:30:02.268212 [DEBUG] switch_core_state_machine.c:433 (sofia/external/919718300881) State ROUTING going to sleep 2012-07-31 02:30:02.268212 [DEBUG] switch_core_state_machine.c:385 (sofia/external/919718300881) Running State Change CS_CONSUME_MEDIA 2012-07-31 02:30:02.268212 [DEBUG] switch_core_state_machine.c:452 (sofia/external/919718300881) State CONSUME_MEDIA 2012-07-31 02:30:02.268212 [DEBUG] mod_nibblebill.c:383 Doing lookup query [SELECT cash AS nibble_balance FROM accounts WHERE id='97183008'] 2012-07-31 02:30:02.268212 [DEBUG] mod_nibblebill.c:393 Retrieved current balance for account 97183008 (balance = 500.500000) 2012-07-31 02:30:02.268212 [DEBUG] switch_core_state_machine.c:452 (sofia/external/919718300881) State CONSUME_MEDIA going to sleep 2012-07-31 02:30:02.268212 [DEBUG] sofia.c:5745 Channel sofia/external/919718300881 entering state [calling][0] 2012-07-31 02:30:02.308197 [DEBUG] switch_core_session.c:919 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 02:30:02.308197 [DEBUG] switch_core_session.c:919 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 02:30:02.308197 [DEBUG] sofia.c:5745 Channel sofia/external/919718300881 entering state [calling][0] 2012-07-31 02:30:05.488328 [DEBUG] switch_core_session.c:919 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 02:30:05.488328 [DEBUG] switch_core_session.c:919 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 02:30:05.488328 [DEBUG] sofia.c:5745 Channel sofia/external/919718300881 entering state [proceeding][183] 2012-07-31 02:30:05.488328 [DEBUG] sofia.c:5756 Remote SDP: v=0 o=root 3607 3607 IN IP4 31.216.132.13 s=session c=IN IP4 31.216.132.13 t=0 0 m=audio 14460 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2012-07-31 02:30:05.488328 [DEBUG] sofia_glue.c:4915 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] 2012-07-31 02:30:05.488328 [DEBUG] sofia_glue.c:2995 Set Codec sofia/external/919718300881 PCMA/8000 20 ms 160 samples 64000 bits 2012-07-31 02:30:05.488328 [DEBUG] switch_core_codec.c:111 sofia/external/919718300881 Original read codec set to PCMA:8 2012-07-31 02:30:05.488328 [DEBUG] sofia_glue.c:5029 Set 2833 dtmf send payload to 101 2012-07-31 02:30:05.488328 [DEBUG] sofia_glue.c:3244 AUDIO RTP [sofia/external/919718300881] 46.252.152.160 port 20512 -> 31.216.132.13 port 14460 codec: 8 ms: 20 2012-07-31 02:30:05.488328 [DEBUG] switch_rtp.c:1676 Starting timer [soft] 160 bytes per 20ms 2012-07-31 02:30:05.508186 [DEBUG] sofia_glue.c:3508 Set 2833 dtmf send payload to 101 2012-07-31 02:30:05.508186 [DEBUG] sofia_glue.c:3514 Set 2833 dtmf receive payload to 101 2012-07-31 02:30:05.508186 [DEBUG] sofia_glue.c:3541 sofia/external/919718300881 Set rtp dtmf delay to 40 2012-07-31 02:30:05.508186 [NOTICE] sofia_glue.c:4052 Pre-Answer sofia/external/919718300881! 2012-07-31 02:30:05.508186 [DEBUG] switch_channel.c:3019 (sofia/external/919718300881) Callstate Change RINGING -> EARLY 2012-07-31 02:31:02.008196 [DEBUG] switch_channel.c:2882 (sofia/external/919718300881) Callstate Change EARLY -> HANGUP 2012-07-31 02:31:02.008196 [NOTICE] switch_ivr_originate.c:3246 Hangup sofia/external/919718300881 [CS_CONSUME_MEDIA] [NO_ANSWER] 2012-07-31 02:31:02.008196 [DEBUG] switch_channel.c:2905 Send signal sofia/external/919718300881 [KILL] 2012-07-31 02:31:02.008196 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 02:31:02.008196 [DEBUG] switch_core_state_machine.c:385 (sofia/external/919718300881) Running State Change CS_HANGUP 2012-07-31 02:31:02.008196 [DEBUG] switch_core_state_machine.c:625 (sofia/external/919718300881) State HANGUP 2012-07-31 02:31:02.008196 [DEBUG] mod_sofia.c:469 Channel sofia/external/919718300881 hanging up, cause: NO_ANSWER 2012-07-31 02:31:02.008196 [DEBUG] mod_sofia.c:527 Sending CANCEL to sofia/external/919718300881 2012-07-31 02:31:02.008196 [DEBUG] mod_nibblebill.c:383 Doing lookup query [SELECT cash AS nibble_balance FROM accounts WHERE id='97183008'] 2012-07-31 02:31:02.008196 [DEBUG] mod_nibblebill.c:393 Retrieved current balance for account 97183008 (balance = 500.500000) 2012-07-31 02:31:02.008196 [DEBUG] switch_core_state_machine.c:47 sofia/external/919718300881 Standard HANGUP, cause: NO_ANSWER 2012-07-31 02:31:02.008196 [DEBUG] switch_core_state_machine.c:625 (sofia/external/919718300881) State HANGUP going to sleep 2012-07-31 02:31:02.008196 [DEBUG] switch_core_state_machine.c:416 (sofia/external/919718300881) State Change CS_HANGUP -> CS_REPORTING 2012-07-31 02:31:02.008196 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 02:31:02.008196 [DEBUG] switch_core_state_machine.c:385 (sofia/external/919718300881) Running State Change CS_REPORTING 2012-07-31 02:31:02.008196 [DEBUG] switch_core_state_machine.c:685 (sofia/external/919718300881) State REPORTING 2012-07-31 02:31:02.008196 [DEBUG] switch_core_state_machine.c:79 sofia/external/919718300881 Standard REPORTING, cause: NO_ANSWER 2012-07-31 02:31:02.008196 [DEBUG] switch_core_state_machine.c:685 (sofia/external/919718300881) State REPORTING going to sleep 2012-07-31 02:31:02.008196 [DEBUG] switch_core_state_machine.c:410 (sofia/external/919718300881) State Change CS_REPORTING -> CS_DESTROY 2012-07-31 02:31:02.008196 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 02:31:02.008196 [DEBUG] switch_core_session.c:1424 Session 104 (sofia/external/919718300881) Locked, Waiting on external entities 2012-07-31 02:31:02.008196 [NOTICE] switch_core_session.c:1442 Session 104 (sofia/external/919718300881) Ended 2012-07-31 02:31:02.008196 [NOTICE] switch_core_session.c:1444 Close Channel sofia/external/919718300881 [CS_DESTROY] 2012-07-31 02:31:02.008196 [DEBUG] switch_core_state_machine.c:514 (sofia/external/919718300881) Callstate Change HANGUP -> DOWN 2012-07-31 02:31:02.008196 [DEBUG] switch_core_state_machine.c:517 (sofia/external/919718300881) Running State Change CS_DESTROY 2012-07-31 02:31:02.008196 [DEBUG] switch_core_state_machine.c:527 (sofia/external/919718300881) State DESTROY 2012-07-31 02:31:02.008196 [DEBUG] mod_sofia.c:374 sofia/external/919718300881 SOFIA DESTROY 2012-07-31 02:31:02.008196 [DEBUG] switch_core_state_machine.c:86 sofia/external/919718300881 Standard DESTROY 2012-07-31 02:31:02.008196 [DEBUG] switch_core_state_machine.c:527 (sofia/external/919718300881) State DESTROY going to sleep freeswitch at internal> -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbhati at gmail.com Skype id:- virbhati2 New Delhi(India) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/2c930cc6/attachment-0001.html From virbhati at gmail.com Tue Jul 31 11:35:24 2012 From: virbhati at gmail.com (virendra bhati) Date: Tue, 31 Jul 2012 13:05:24 +0530 Subject: [Freeswitch-users] Nibble Billing deducted balance from account even call doesn't come to my mobile number Message-ID: Hi Team, Strange issue with Nibblebill module. My setup is like that I am using newfies-dialer for making call from server to any number. I am using Nibblebill for realtime billing. Call don't come to my mobile and status at Freeswitch changed to Answer and nibblebill start deduction of my balance from define tables...... *Below is the FS_CLI output:* 2012-07-31 03:10:09.668193 [DEBUG] switch_ivr_originate.c:1947 Parsing global variables 2012-07-31 03:10:09.668193 [DEBUG] switch_event.c:1519 Parsing variable [plivo_request_uuid]=[c395799a-dade-11e1-b432-00163e908c9f] 2012-07-31 03:10:09.668193 [DEBUG] switch_event.c:1519 Parsing variable [plivo_answer_url]=[http://46.252.152.160:8008/api/v1/answercall/] 2012-07-31 03:10:09.668193 [DEBUG] switch_event.c:1519 Parsing variable [plivo_hangup_url]=[http://46.252.152.160:8008/api/v1/hangupcall/] 2012-07-31 03:10:09.668193 [DEBUG] switch_event.c:1519 Parsing variable [origination_caller_id_number]=[0498722120] 2012-07-31 03:10:09.668193 [DEBUG] switch_event.c:1519 Parsing variable [bridge_early_media]=[true] 2012-07-31 03:10:09.668193 [DEBUG] switch_event.c:1519 Parsing variable [hangup_after_bridge]=[true] 2012-07-31 03:10:09.668193 [DEBUG] switch_event.c:1519 Parsing variable [hangup_after_bridge]=[true] 2012-07-31 03:10:09.668193 [DEBUG] switch_event.c:1519 Parsing variable [nibble_account]=[97183008] 2012-07-31 03:10:09.668193 [DEBUG] switch_event.c:1519 Parsing variable [nibble_rate]=[5] 2012-07-31 03:10:09.668193 [DEBUG] switch_event.c:1519 Parsing variable [nibble_increment]=[30] 2012-07-31 03:10:09.668193 [DEBUG] switch_event.c:1519 Parsing variable [plivo_from]=[0498722120] 2012-07-31 03:10:09.668193 [DEBUG] switch_event.c:1519 Parsing variable [plivo_to]=[919718300881] 2012-07-31 03:10:09.668193 [DEBUG] switch_event.c:1519 Parsing variable [plivo_app]=[true] 2012-07-31 03:10:09.668193 [DEBUG] switch_event.c:1519 Parsing variable [absolute_codec_string]=[PCMA] 2012-07-31 03:10:09.668193 [DEBUG] switch_event.c:1519 Parsing variable [originate_timeout]=[60] 2012-07-31 03:10:09.668193 [DEBUG] switch_event.c:1519 Parsing variable [ignore_early_media]=[true] 2012-07-31 03:10:09.668193 [NOTICE] switch_channel.c:926 New Channel sofia/external/919718300881 [c3961efe-dade-11e1-b60f-1782e99edc0a] 2012-07-31 03:10:09.668193 [DEBUG] mod_sofia.c:4705 (sofia/external/919718300881) State Change CS_NEW -> CS_INIT 2012-07-31 03:10:09.668193 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:09.668193 [DEBUG] switch_core_state_machine.c:385 (sofia/external/919718300881) Running State Change CS_INIT 2012-07-31 03:10:09.668193 [DEBUG] switch_core_state_machine.c:424 (sofia/external/919718300881) State INIT 2012-07-31 03:10:09.668193 [DEBUG] mod_sofia.c:85 sofia/external/919718300881 SOFIA INIT 2012-07-31 03:10:09.668193 [DEBUG] mod_sofia.c:125 (sofia/external/919718300881) State Change CS_INIT -> CS_ROUTING 2012-07-31 03:10:09.668193 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:09.668193 [DEBUG] switch_core_state_machine.c:424 (sofia/external/919718300881) State INIT going to sleep 2012-07-31 03:10:09.668193 [DEBUG] switch_core_state_machine.c:385 (sofia/external/919718300881) Running State Change CS_ROUTING 2012-07-31 03:10:09.668193 [DEBUG] switch_channel.c:1919 (sofia/external/919718300881) Callstate Change DOWN -> RINGING 2012-07-31 03:10:09.668193 [DEBUG] switch_core_state_machine.c:433 (sofia/external/919718300881) State ROUTING 2012-07-31 03:10:09.668193 [DEBUG] mod_sofia.c:148 sofia/external/919718300881 SOFIA ROUTING 2012-07-31 03:10:09.668193 [DEBUG] switch_ivr_originate.c:67 (sofia/external/919718300881) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-07-31 03:10:09.668193 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:09.668193 [DEBUG] mod_nibblebill.c:453 Attempting to bill at $5 per minute to account 97183008 2012-07-31 03:10:09.668193 [DEBUG] mod_nibblebill.c:465 Not billing 97183008 - call is not in answered state 2012-07-31 03:10:09.668193 [DEBUG] mod_nibblebill.c:383 Doing lookup query [SELECT cash AS nibble_balance FROM accounts WHERE id='97183008'] 2012-07-31 03:10:09.668193 [DEBUG] switch_core_session.c:919 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:09.668193 [DEBUG] mod_nibblebill.c:393 Retrieved current balance for account 97183008 (balance = 500.500000) 2012-07-31 03:10:09.668193 [DEBUG] mod_nibblebill.c:469 Comparing 500.500000 to hangup balance of 0.000000 2012-07-31 03:10:09.668193 [DEBUG] mod_nibblebill.c:383 Doing lookup query [SELECT cash AS nibble_balance FROM accounts WHERE id='97183008'] 2012-07-31 03:10:09.668193 [DEBUG] mod_nibblebill.c:393 Retrieved current balance for account 97183008 (balance = 500.500000) 2012-07-31 03:10:09.668193 [DEBUG] switch_core_state_machine.c:433 (sofia/external/919718300881) State ROUTING going to sleep 2012-07-31 03:10:09.668193 [DEBUG] switch_core_state_machine.c:385 (sofia/external/919718300881) Running State Change CS_CONSUME_MEDIA 2012-07-31 03:10:09.668193 [DEBUG] switch_core_state_machine.c:452 (sofia/external/919718300881) State CONSUME_MEDIA 2012-07-31 03:10:09.668193 [DEBUG] mod_nibblebill.c:453 Attempting to bill at $5 per minute to account 97183008 2012-07-31 03:10:09.668193 [DEBUG] mod_nibblebill.c:465 Not billing 97183008 - call is not in answered state 2012-07-31 03:10:09.668193 [DEBUG] mod_nibblebill.c:383 Doing lookup query [SELECT cash AS nibble_balance FROM accounts WHERE id='97183008'] 2012-07-31 03:10:09.668193 [DEBUG] mod_nibblebill.c:393 Retrieved current balance for account 97183008 (balance = 500.500000) 2012-07-31 03:10:09.668193 [DEBUG] mod_nibblebill.c:469 Comparing 500.500000 to hangup balance of 0.000000 2012-07-31 03:10:09.668193 [DEBUG] mod_nibblebill.c:383 Doing lookup query [SELECT cash AS nibble_balance FROM accounts WHERE id='97183008'] 2012-07-31 03:10:09.668193 [DEBUG] mod_nibblebill.c:393 Retrieved current balance for account 97183008 (balance = 500.500000) 2012-07-31 03:10:09.668193 [INFO] switch_core_session.c:1376 sofia/external/919718300881 setting session heartbeat to 30 second(s). 2012-07-31 03:10:09.668193 [DEBUG] switch_core_state_machine.c:452 (sofia/external/919718300881) State CONSUME_MEDIA going to sleep 2012-07-31 03:10:09.668193 [DEBUG] sofia.c:5745 Channel sofia/external/919718300881 entering state [calling][0] 2012-07-31 03:10:09.708195 [DEBUG] switch_core_session.c:919 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:09.708195 [DEBUG] switch_core_session.c:919 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:09.708195 [DEBUG] sofia.c:5745 Channel sofia/external/919718300881 entering state [calling][0] 2012-07-31 03:10:20.808198 [DEBUG] switch_core_session.c:919 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:20.808198 [DEBUG] switch_core_session.c:919 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:20.808198 [DEBUG] sofia.c:5745 Channel sofia/external/919718300881 entering state [proceeding][183] 2012-07-31 03:10:20.808198 [DEBUG] sofia.c:5756 Remote SDP: v=0 o=root 3607 3607 IN IP4 31.216.132.13 s=session c=IN IP4 31.216.132.13 t=0 0 m=audio 19974 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2012-07-31 03:10:20.808198 [DEBUG] sofia_glue.c:4915 Audio Codec Compare [PCMA:8:8000:20:64000]/[PCMA:8:8000:20:64000] 2012-07-31 03:10:20.808198 [DEBUG] sofia_glue.c:2995 Set Codec sofia/external/919718300881 PCMA/8000 20 ms 160 samples 64000 bits 2012-07-31 03:10:20.808198 [DEBUG] switch_core_codec.c:111 sofia/external/919718300881 Original read codec set to PCMA:8 2012-07-31 03:10:20.808198 [DEBUG] sofia_glue.c:5029 Set 2833 dtmf send payload to 101 2012-07-31 03:10:20.808198 [DEBUG] sofia_glue.c:3244 AUDIO RTP [sofia/external/919718300881] 46.252.152.160 port 26726 -> 31.216.132.13 port 19974 codec: 8 ms: 20 2012-07-31 03:10:20.808198 [DEBUG] switch_rtp.c:1676 Starting timer [soft] 160 bytes per 20ms 2012-07-31 03:10:20.808198 [DEBUG] sofia_glue.c:3508 Set 2833 dtmf send payload to 101 2012-07-31 03:10:20.808198 [DEBUG] sofia_glue.c:3514 Set 2833 dtmf receive payload to 101 2012-07-31 03:10:20.808198 [DEBUG] sofia_glue.c:3541 sofia/external/919718300881 Set rtp dtmf delay to 40 2012-07-31 03:10:20.808198 [NOTICE] sofia_glue.c:4052 Pre-Answer sofia/external/919718300881! 2012-07-31 03:10:20.808198 [DEBUG] switch_channel.c:3019 (sofia/external/919718300881) Callstate Change RINGING -> EARLY 2012-07-31 03:10:48.328196 [DEBUG] switch_core_session.c:919 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:48.328196 [DEBUG] switch_core_session.c:919 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:48.328196 [DEBUG] sofia.c:5745 Channel sofia/external/919718300881 entering state [completing][200] 2012-07-31 03:10:48.328196 [DEBUG] sofia.c:5756 Remote SDP: v=0 o=root 3607 3608 IN IP4 31.216.132.13 s=session c=IN IP4 31.216.132.13 t=0 0 m=audio 19974 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 2012-07-31 03:10:48.328196 [DEBUG] switch_core_session.c:919 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:48.328196 [DEBUG] switch_core_session.c:919 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:48.328196 [DEBUG] sofia.c:5745 Channel sofia/external/919718300881 entering state [ready][200] 2012-07-31 03:10:48.328196 [DEBUG] switch_channel.c:3278 (sofia/external/919718300881) Callstate Change EARLY -> ACTIVE 2012-07-31 03:10:48.328196 [DEBUG] switch_ivr_originate.c:3330 Originate Resulted in Success: [sofia/external/919718300881] 2012-07-31 03:10:48.328196 [NOTICE] sofia.c:6420 Channel [sofia/external/919718300881] has been answered 2012-07-31 03:10:48.328196 [INFO] switch_channel.c:2744 sofia/external/919718300881 Flipping CID from "" <0498722120> to "Outbound Call" <919718300881> 2012-07-31 03:10:48.328196 [DEBUG] mod_commands.c:3616 (sofia/external/919718300881) State Change CS_CONSUME_MEDIA -> CS_EXECUTE 2012-07-31 03:10:48.328196 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:48.328196 [DEBUG] switch_core_state_machine.c:385 (sofia/external/919718300881) Running State Change CS_EXECUTE 2012-07-31 03:10:48.328196 [DEBUG] switch_core_state_machine.c:440 (sofia/external/919718300881) State EXECUTE 2012-07-31 03:10:48.328196 [DEBUG] mod_sofia.c:241 sofia/external/919718300881 SOFIA EXECUTE 2012-07-31 03:10:48.328196 [INFO] switch_core_session.c:1376 sofia/external/919718300881 setting session heartbeat to 30 second(s). 2012-07-31 03:10:48.328196 [DEBUG] switch_core_state_machine.c:196 sofia/external/919718300881 Standard EXECUTE EXECUTE sofia/external/919718300881 socket(127.0.0.1:8084 async full) 2012-07-31 03:10:48.328196 [DEBUG] mod_nibblebill.c:612 Received request via SESSION_HEARTBEAT! 2012-07-31 03:10:48.328196 [DEBUG] mod_nibblebill.c:453 Attempting to bill at $5 per minute to account 97183008 2012-07-31 03:10:48.328196 [INFO] mod_nibblebill.c:505 Beginning new billing on c3961efe-dade-11e1-b60f-1782e99edc0a 2012-07-31 03:10:48.328196 [DEBUG] mod_nibblebill.c:511 0 seconds passed since last bill time of 2012-07-31 03:10:48 2012-07-31 03:10:48.328196 [DEBUG] mod_nibblebill.c:528 Billing $2.500000 to 97183008 (Call: c3961efe-dade-11e1-b60f-1782e99edc0a / 0.000000 so far) 2012-07-31 03:10:48.328196 [DEBUG] mod_nibblebill.c:338 Doing update query [UPDATE accounts SET cash=cash-30 WHERE id='97183008'] 2012-07-31 03:10:48.328196 [DEBUG] mod_nibblebill.c:383 Doing lookup query [SELECT cash AS nibble_balance FROM accounts WHERE id='97183008'] 2012-07-31 03:10:48.328196 [DEBUG] mod_nibblebill.c:393 Retrieved current balance for account 97183008 (balance = 470.500000) 2012-07-31 03:10:48.348194 [DEBUG] switch_core_session.c:1056 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:48.348194 [DEBUG] switch_core_session.c:1056 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:48.368196 [DEBUG] switch_rtp.c:3253 Correct ip/port confirmed. 2012-07-31 03:10:48.368196 [DEBUG] switch_ivr.c:598 sofia/external/919718300881 Command Execute set(plivo_app=true) EXECUTE sofia/external/919718300881 set(plivo_app=true) 2012-07-31 03:10:48.368196 [DEBUG] mod_dptools.c:1305 sofia/external/919718300881 SET [plivo_app]=[true] 2012-07-31 03:10:48.408195 [DEBUG] switch_ivr.c:598 sofia/external/919718300881 Command Execute set(hangup_after_bridge=false) EXECUTE sofia/external/919718300881 set(hangup_after_bridge=false) 2012-07-31 03:10:48.408195 [DEBUG] mod_dptools.c:1305 sofia/external/919718300881 SET [hangup_after_bridge]=[false] 2012-07-31 03:10:48.468630 [DEBUG] switch_core_session.c:1056 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:48.488187 [DEBUG] switch_ivr.c:598 sofia/external/919718300881 Command Execute speak(flite|slt|Hello World this is demo IVR message) EXECUTE sofia/external/919718300881 speak(flite|slt|Hello World this is demo IVR message) 2012-07-31 03:10:48.488187 [DEBUG] switch_ivr_play_say.c:2473 OPEN TTS flite 2012-07-31 03:10:48.488187 [DEBUG] switch_ivr_play_say.c:2482 Raw Codec Activated 2012-07-31 03:10:48.608191 [DEBUG] switch_ivr_play_say.c:2164 Speaking text: Hello World this is demo IVR message 2012-07-31 03:10:51.428188 [DEBUG] switch_ivr_play_say.c:2361 done speaking text 2012-07-31 03:10:51.428188 [DEBUG] switch_core_session.c:1056 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:51.428188 [DEBUG] mod_event_socket.c:2644 (sofia/external/919718300881) State Change CS_EXECUTE -> CS_RESET 2012-07-31 03:10:51.428188 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:51.448191 [DEBUG] switch_ivr.c:598 sofia/external/919718300881 Command Execute hangup() EXECUTE sofia/external/919718300881 hangup() 2012-07-31 03:10:51.448191 [DEBUG] switch_channel.c:2882 (sofia/external/919718300881) Callstate Change ACTIVE -> HANGUP 2012-07-31 03:10:51.448191 [NOTICE] mod_dptools.c:1134 Hangup sofia/external/919718300881 [CS_RESET] [NORMAL_CLEARING] 2012-07-31 03:10:51.448191 [DEBUG] switch_channel.c:2905 Send signal sofia/external/919718300881 [KILL] 2012-07-31 03:10:51.448191 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:51.448191 [DEBUG] switch_core_session.c:2329 sofia/external/919718300881 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-07-31 03:10:51.448191 [DEBUG] switch_core_session.c:2329 sofia/external/919718300881 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-07-31 03:10:51.448191 [DEBUG] switch_core_state_machine.c:440 (sofia/external/919718300881) State EXECUTE going to sleep 2012-07-31 03:10:51.448191 [DEBUG] switch_core_state_machine.c:385 (sofia/external/919718300881) Running State Change CS_HANGUP 2012-07-31 03:10:51.448191 [DEBUG] switch_core_state_machine.c:625 (sofia/external/919718300881) State HANGUP 2012-07-31 03:10:51.448191 [DEBUG] mod_sofia.c:469 Channel sofia/external/919718300881 hanging up, cause: NORMAL_CLEARING 2012-07-31 03:10:51.448191 [DEBUG] mod_sofia.c:517 Sending BYE to sofia/external/919718300881 2012-07-31 03:10:51.448191 [DEBUG] mod_nibblebill.c:453 Attempting to bill at $5 per minute to account 97183008 2012-07-31 03:10:51.448191 [DEBUG] mod_nibblebill.c:511 -26 seconds passed since last bill time of 2012-07-31 03:11:18 2012-07-31 03:10:51.448191 [DEBUG] mod_nibblebill.c:383 Doing lookup query [SELECT cash AS nibble_balance FROM accounts WHERE id='97183008'] 2012-07-31 03:10:51.448191 [DEBUG] mod_nibblebill.c:393 Retrieved current balance for account 97183008 (balance = 470.500000) 2012-07-31 03:10:51.448191 [DEBUG] switch_core_state_machine.c:47 sofia/external/919718300881 Standard HANGUP, cause: NORMAL_CLEARING 2012-07-31 03:10:51.448191 [DEBUG] switch_core_state_machine.c:625 (sofia/external/919718300881) State HANGUP going to sleep 2012-07-31 03:10:51.448191 [DEBUG] switch_core_state_machine.c:416 (sofia/external/919718300881) State Change CS_HANGUP -> CS_REPORTING 2012-07-31 03:10:51.448191 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:51.448191 [DEBUG] switch_core_state_machine.c:385 (sofia/external/919718300881) Running State Change CS_REPORTING 2012-07-31 03:10:51.448191 [DEBUG] switch_core_state_machine.c:685 (sofia/external/919718300881) State REPORTING 2012-07-31 03:10:51.448191 [DEBUG] switch_core_state_machine.c:79 sofia/external/919718300881 Standard REPORTING, cause: NORMAL_CLEARING 2012-07-31 03:10:51.448191 [DEBUG] switch_core_state_machine.c:685 (sofia/external/919718300881) State REPORTING going to sleep 2012-07-31 03:10:51.448191 [DEBUG] switch_core_state_machine.c:410 (sofia/external/919718300881) State Change CS_REPORTING -> CS_DESTROY 2012-07-31 03:10:51.448191 [DEBUG] switch_core_session.c:1224 Send signal sofia/external/919718300881 [BREAK] 2012-07-31 03:10:51.448191 [DEBUG] switch_core_session.c:1424 Session 105 (sofia/external/919718300881) Locked, Waiting on external entities 2012-07-31 03:10:51.448191 [NOTICE] switch_core_session.c:1442 Session 105 (sofia/external/919718300881) Ended 2012-07-31 03:10:51.448191 [NOTICE] switch_core_session.c:1444 Close Channel sofia/external/919718300881 [CS_DESTROY] 2012-07-31 03:10:51.448191 [DEBUG] switch_core_state_machine.c:514 (sofia/external/919718300881) Callstate Change HANGUP -> DOWN 2012-07-31 03:10:51.448191 [DEBUG] switch_core_state_machine.c:517 (sofia/external/919718300881) Running State Change CS_DESTROY 2012-07-31 03:10:51.448191 [DEBUG] switch_core_state_machine.c:527 (sofia/external/919718300881) State DESTROY 2012-07-31 03:10:51.448191 [DEBUG] mod_sofia.c:374 sofia/external/919718300881 SOFIA DESTROY 2012-07-31 03:10:51.448191 [DEBUG] switch_core_state_machine.c:86 sofia/external/919718300881 Standard DESTROY 2012-07-31 03:10:51.448191 [DEBUG] switch_core_state_machine.c:527 (sofia/external/919718300881) State DESTROY going to sleep if anything is required from my end then please let me know if will provide here.... -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbhati at gmail.com Skype id:- virbhati2 New Delhi(India) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/388580e9/attachment-0001.html From evgeniy at bestnet.kharkov.ua Tue Jul 31 11:48:07 2012 From: evgeniy at bestnet.kharkov.ua (Evgeniy Movlyan) Date: Tue, 31 Jul 2012 10:48:07 +0300 Subject: [Freeswitch-users] Nibble Billing deducted balance from account even call doesn't come to my mobile number In-Reply-To: References: Message-ID: <50178DB7.7000606@bestnet.kharkov.ua> I have the same problem. When i am calling from one my extension to another all is ok, but when i am calling to external number i got this message: "mod_nibblebill.c:465 Not billing XXXXXXXXXX - call is not in answered state". 31.07.2012 10:35, virendra bhati ?????: > mod_nibblebill.c:465 Not billing > 97183008 - call is not in answered state -- Evgeniy Movlyan, BestNet Ltd. From x.liu at hw.ac.uk Tue Jul 31 11:59:37 2012 From: x.liu at hw.ac.uk (x.liu) Date: Tue, 31 Jul 2012 08:59:37 +0100 Subject: [Freeswitch-users] Am I using play_and_detect_speech correctly? In-Reply-To: References: Message-ID: <50179069.6020700@hw.ac.uk> Hi Chris, Okay, thanks! I will have tries to see how it works. For the normal use of the app: play_and_detect_speech, will it resume previous ASR session, or will it stop previous session and restart a new session each time when I issue it via ESL? For the app of detect_speech, I can first to start an ASR session by "detect_speech speechMod grammar grammarPath", then I can do "detect_speech pause", or "detect_speech resume" or "detect_speech stop". But for play_and_detect_speech, I am not sure how it works in terms of starting, pause, resume/restart sequence. The play_and_detect_speech is a very good, useful app as it supports the barge-in. So we definitely need to use it. Thanks! Xing On 07/31/2012 12:53 AM, Christopher Rienzo wrote: > You are using that APP in a way that hasn't been tested. Try to do > "speech_detect pause" and see if that helps. It will reserve your ASR > session for reuse while "speech_detect stop" will tear it down > completely. If that doesn't work, open a jira ticket and attach a > full call trace. It's difficult to understand exactly what is > happening from your description. > > > Chris > > > > On Mon, Jul 30, 2012 at 5:38 PM, Liu, Xingkun > wrote: > > Hello, > > I am using play_and_detect_speech with Java ESL in my IVR > applications. > > Previously I call it again each time after I receive any > recognition event, > like recognition complete, no-input-timeout, or recognition-timeout, > it seems to work fine. > > Now I have changed my app to issue play_and_detect_speech command > based on > my available system utterances as well as the speech event. > > I.e., I use a separate thread to constantly check if there is a > system utterance coming in > from another component of my application, if there is any > utterance I issue the command which will > speak the new utterance and listen to user input no matter whether > or not previous > command has finished. And if there is any speech event > (recognition result, timeout etc.) > the play_and_detect_speech command is also issued but with playing > silence. > > Obviously the new command will stop the utterance speaking of the > previous command if it is not finished. > > My question is > > will the new play_and_detect_speech command also stop the previous > ASR listening > or will there be many ASR listening channel and sending speech > data (or silence) to ASR server? > > Do I need to explicitly issue a "stop" commnad before issuing a > new play_and_detect_speech? > If yes, how to do that, by "detect_speech stop"? > > Recently there is a network traffice problem (lots of connections > /data transportation to the ASR server machine) > when I am running my application. I am not sure if this is because > of other issues > or because of my new changes to the way of using > play_and_detect_speech. > > Please any one could shed a light on this? > > Many thanks! > > Xing > > > ------------------------------------------------------------------------ > > *Heriot-Watt University is the Sunday Times Scottish University of > the Year 2011-2012.* > > We invite research leaders and ambitious early career researchers > to join us in leading and driving research in key > inter-disciplinary themes. Please see www.hw.ac.uk/researchleaders > for further information and > how to apply. > > Heriot-Watt University is a Scottish charity registered under > charity number SC000278. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 We invite research leaders and ambitious early career researchers to join us in leading and driving research in key inter-disciplinary themes. Please see www.hw.ac.uk/researchleaders for further information and how to apply. Heriot-Watt University is a Scottish charity registered under charity number SC000278. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/75e3f241/attachment.html From govoiper at gmail.com Tue Jul 31 12:06:05 2012 From: govoiper at gmail.com (SamyGo) Date: Tue, 31 Jul 2012 13:06:05 +0500 Subject: [Freeswitch-users] Nibble Billing deducted balance from account even call doesn't come to my mobile number In-Reply-To: <50178DB7.7000606@bestnet.kharkov.ua> References: <50178DB7.7000606@bestnet.kharkov.ua> Message-ID: Hi, Well its works perfect for me, do you guys have ignore_early_media set in your outbound string, if no then set it and then see what happens. On Jul 31, 2012 12:50 PM, "Evgeniy Movlyan" wrote: > I have the same problem. When i am calling from one my extension to > another all is ok, but when i am calling to external number i got this > message: "mod_nibblebill.c:465 Not billing XXXXXXXXXX - call is not in > answered state". > > 31.07.2012 10:35, virendra bhati ?????: > > mod_nibblebill.c:465 Not billing > > 97183008 - call is not in answered state > > -- > Evgeniy Movlyan, > BestNet Ltd. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120731/24502a0a/attachment.html From admin at blindi.net Tue Jul 31 12:48:52 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Tue, 31 Jul 2012 10:48:52 +0200 (CEST) Subject: [Freeswitch-users] Question how can i get variables in lua from xml dialplan? In-Reply-To: References: Message-ID: Hi Seven, Thanks! its works --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From yufei.tao at redembedded.com Tue Jul 31 12:55:10 2012 From: yufei.tao at redembedded.com (Yufei Tao) Date: Tue, 31 Jul 2012 09:55:10 +0100 Subject: [Freeswitch-users] H264 transcoding In-Reply-To: References: Message-ID: <50179D6E.2010101@redembedded.com> Hi Seven I think something that both decodes and encodes H264, would be a good starting point. And your work on top of libvlc sounds like the closest resource so far for my transcoder. Is that right? Please bear with me as I'm new to FS and all these libs and would appreciate it very much pointed at the right direction. Yufei Subject: Re: [Freeswitch-users] H264 transcoding From: Seven Du Date: 30/07/12 17:17 To: FreeSWITCH Users Help And Yufei, my solution won't solve your problem at this time, I only did encoding and for your transcoder to work there need a lot more work.... On Tuesday, July 31, 2012 at 12:06 AM, Seven Du wrote: Cluecon is coming and I think we could wait a few days and we still have some issues to figure out ... On Monday, July 30, 2012 at 8:33 PM, Yufei Tao wrote: Hi Seven Thanks very much for your replies! I thought I should be very surprised if I was the first one wanting to do this. It'd be very helpful if you could share your code with x264, or libvlc, to spare me, and possibly others of reinventing the wheels. I'll compile the code myself of course which should be free of the licensing troubles I guess. I'd be very grateful if you could share the code and give some instructions. Thank you very much! Yufei Subject: Re: [Freeswitch-users] H264 transcodin From: Seven Du Date: 28/07/12 01:45 To: FreeSWITCH Users Help Real-time encoding with statically linked x264 lib works fine for me from QCIF to D1 resolution, 720p is slow and discarding frames on a Xeon Quad core CPU. I haven't look how to use the GPU, or if possible. It is working in my lab and I have the same question with Yufei Tao when going to production or deliver to customer. Based on http://lists.freeswitch.org/pipermail/freeswitch-dev/2010-September/004227.html , In my understanding, compile and link and use by my self should be fine and, if I deliver to a customer, it should be fine if I provide the code and help the customer to compile on their own server? I'd like to open source the code to public later, but, I'd like to know is it a MUST or MAY? If you pipe to ffmpeg or x264 command line it's not been treated as combine into a large work, and I'm not sure if realtime transcoding will be smooth. 7. On Friday, July 27, 2012 at 11:17 PM, Yufei Tao wrote: Thanks everyone for the responses! If I understand it correctly, if I installed ffmpeg on itself separately from FS, I could write a module for FS, in which I just call the ffmpeg program by running a command line. This way would it be classified as "not combine them into a larger work", thus free from license incompatibility problem? Not sure if that'll work for real-time transcoding of x264 though? Thanks very much for you opinions! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com Join Us At ClueCon - Aug 7-9, 2012 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ForwardedMessage.eml Subject: Re: [Freeswitch-users] H264 transcoding From: Seven Du Date: 28/07/12 02:10 To: FreeSWITCH Users Help I patched libvlc and it can now decode any video vlc supported and encode with x264 and send to any sip phone. Specifically I'm trying to use it to play a 1080p stream and resize to CIF or D1 so a video phone will accept(Sending 1080p will cause some phones to reboot :( ). Still need a lot of code to make it working neatly, however, I might can do a demo on ClueCon and @William if you'd like to review the code and merge into tree I'll happy to contribute that later. On Friday, July 27, 2012 at 2:43 PM, William King wrote: libvlc is LGPL http://www.videolan.org/press/lgpl.html and there is now a mod_vlc(though it doesn't yet support video streams). The user can choose to build vlc with only the LGPL components or add the more 'adverse' modules. In none of the LGPL packages of libvlc is ffmpeg enabled, but there is a module for libvlc for ffmpeg. http://wiki.videolan.org/FFmpeg The only pieces now may just be the FS side of things for video. William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com On 07/26/2012 08:46 PM, Anthony Minessale wrote: you would probably need to do something like make a mod for ffmpeg that protects you from the gpl then allow the user to build that lib on his own and choose at compile time to install patented or adverse licensed components. No license rules prohibit an end user from combining code only distributors. but even then we need a bunch of code to write. On Thu, Jul 26, 2012 at 10:41 PM, curriegrad2004 wrote: Ken, If you think those guys over at x264 will ever change the license from GPL to LGPL, you're just dreaming the pie in the sky... In short, don't even think about it ;P On Thu, Jul 26, 2012 at 8:19 PM, Ken Rice wrote: we can not and will not use GPL software, the license is not compatible with the GPL and would polute the codebase with additional restrictions that are not wanted or needed. now if someone could get them to change the license or atleast give us a license under better terms such as the LGPL or the MPL then the license issue would be null Ken Sent from my iPad On Jul 26, 2012, at 7:45 PM, Terry Barnum wrote: Use x264? http://en.wikipedia.org/wiki/X264 On Jul 26, 2012, at 4:53 PM, Ken Rice wrote: Is it possible sure... Is ot probably to happen anytime soon? Not until the patents run out... On 7/26/12 5:04 PM, "yufei.tao" wrote: Hi I am trying to decide if it is feasible to let FS do transcoding between different H264 formats for live video calls. This is because I've got SIP clients that both use H264 but with different formats and one (with a bad H264 decoder) has problems decoding H264 stream from the other. But each of these two clients communicate fine using H264 with a third client that uses ffmpeg. I'm thinking of adding a module which uses ffmpeg, so that it will transcode H264 between different parameters. I've got a few questions: 1. Is this feasible? I'm not looking at supporting many simultaneous calls. 2. What is involved in transcoding real-time video stream? 3. Anyone's done anything like this before? I'm new to FS and any suggestions would be very much appreciated! Yufei -- Yufei Tao Red Embedded This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/fe139aec/attachment.html From admin at blindi.net Tue Jul 31 13:12:28 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Tue, 31 Jul 2012 11:12:28 +0200 (CEST) Subject: [Freeswitch-users] Lua problem there are only two arguments evaluated In-Reply-To: References: <50178DB7.7000606@bestnet.kharkov.ua> Message-ID: hi guy, I have simply written a voicemailrecording in lua. My dialplan extension is: The luascript: -- simply mailbox recording -- parameters: -- mailbox, sender, message mailbox = argv[1]; from_box = argv[2]; msg_file = argv[3]; snd = "/usr/local/freeswitch/sounds/dorf"; file, errMsg = io.open( mailbox, "r" ); if not file then session:streamFile("/usr/local/freeswitch/sounds/dorf/mailbox_not.alaw") session:execute("transfer","dorf_now XML dorf_now"); else freeswitch.consoleLog("info", "msg_file is: " .. msg_file .. "\n"); session:execute("playback",mailbox); session:streamFile("/usr/local/freeswitch/sounds/ivr/beep.alaw") session:execute("record","/tmp/" .. msg_file .. " 180 200 3 "); session:execute("transfer","dorf_send XML dorf_send"); end I become the error: [ERR] mod_lua.cpp:198 /usr/local/freeswitch/scripts/recordmsg.lua:13 : attempt to concatenate global 'msg_file' (a nil value)M I have 3 arguments, only 2 arguments will be works. Can your help please? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From govoiper at gmail.com Tue Jul 31 13:23:06 2012 From: govoiper at gmail.com (SamyGo) Date: Tue, 31 Jul 2012 14:23:06 +0500 Subject: [Freeswitch-users] Lua problem there are only two arguments evaluated In-Reply-To: References: <50178DB7.7000606@bestnet.kharkov.ua> Message-ID: Hi, 1st Argument: $${dorf_mailbox_prefix}/${**mailbox_to}/greeting.alaw 2nd Argument: ${mailbox} 3rd Argument: ${msgtime} Can you verify that the ${msgtime} is properly calculated and is not NULL at the taime of execution of application=lua ? Regards, Sammy On Tue, Jul 31, 2012 at 2:12 PM, Thomas Hoellriegel wrote: > hi guy, > I have simply written a voicemailrecording in lua. > My dialplan extension is: > > expression="^44[0-9][0-9][0-9]**[0-9]\d*$"> > > > > > > The luascript: > > -- simply mailbox recording > -- parameters: > -- mailbox, sender, message > mailbox = argv[1]; > from_box = argv[2]; > msg_file = argv[3]; > snd = "/usr/local/freeswitch/sounds/**dorf"; > file, errMsg = io.open( mailbox, "r" ); > if not file then > session:streamFile("/usr/**local/freeswitch/sounds/dorf/** > mailbox_not.alaw") > session:execute("transfer","**dorf_now XML dorf_now"); > else > freeswitch.consoleLog("info", "msg_file is: " .. msg_file .. "\n"); > session:execute("playback",**mailbox); > session:streamFile("/usr/**local/freeswitch/sounds/ivr/**beep.alaw") > session:execute("record","/**tmp/" .. msg_file .. " 180 200 3 "); > session:execute("transfer","**dorf_send XML dorf_send"); > end > > I become the error: > [ERR] mod_lua.cpp:198 /usr/local/freeswitch/scripts/**recordmsg.lua:13 > : attempt to concatenate global 'msg_file' (a nil value)M > > > I have 3 arguments, only 2 arguments will be works. > Can your help please? > thanks. > > > > > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120731/41bfed01/attachment-0001.html From admin at blindi.net Tue Jul 31 13:37:23 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Tue, 31 Jul 2012 11:37:23 +0200 (CEST) Subject: [Freeswitch-users] Lua problem there are only two arguments evaluated In-Reply-To: References: <50178DB7.7000606@bestnet.kharkov.ua> Message-ID: Hi SamyGo > 1st Argument: $${dorf_mailbox_prefix}/${**mailbox_to}/greeting.alaw > 2nd Argument: ${mailbox} > 3rd Argument: ${msgtime} > > Can you verify that the ${msgtime} is properly calculated and is not NULL > at the taime of execution of application=lua ? I like to save a message in the timestamp format: for example: msg_2012_31_07_15_32_31.alaw I read the current date and time from: ${msgtime} But i have no output. What is wrong please? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From slava at tangramltd.com Tue Jul 31 13:55:34 2012 From: slava at tangramltd.com (Viacheslav Dubrovskyi) Date: Tue, 31 Jul 2012 12:55:34 +0300 Subject: [Freeswitch-users] Callcenter and FIFO Mod Question In-Reply-To: <06502C073AD9394AADB3CA7FD94931BC0971F6B9@okc1x1.Logixcom.com> References: <06502C073AD9394AADB3CA7FD94931BC0971F6B9@okc1x1.Logixcom.com> Message-ID: <5017AB96.5010506@tangramltd.com> 17.07.2012 02:49, Joshua Foshee ?????: > > Does the Callcenter have any chime feature to play announces while a > user is on hold? > No, but you can make own local_stream with chime. And change it by set cc_moh_override for example from local_stream.conf.xml > > > > I know the FIFO does have this option with chime but adding on that > does it have ability to setup a IVR while playing the chime? > > -- WBR, Viacheslav Dubrovskyi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/8aa50c4b/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: smime.p7s Type: application/pkcs7-signature Size: 4931 bytes Desc: ?????????????????????????????????? ?????????????? S/MIME Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/8aa50c4b/attachment.bin From govoiper at gmail.com Tue Jul 31 14:08:05 2012 From: govoiper at gmail.com (SamyGo) Date: Tue, 31 Jul 2012 15:08:05 +0500 Subject: [Freeswitch-users] Lua problem there are only two arguments evaluated In-Reply-To: References: <50178DB7.7000606@bestnet.kharkov.ua> Message-ID: If you don't get an output on : > > msgtime=msg_${strftime(%**Y_%m_%d_%H_%M_%S)}_${mailbox}.**alaw ${strftime(%**Y_%m_%d_%H_%M_%S)} I think the function strftime don't like the "_" Captures all the fields of the strftime separately and then concatenate into your required format formt he temporary variables. i.e year=${strftime(%**Y)} month=${strftime(%**M)} day=${strftime(%d)} and then string= ${year}_${month}_${day} On Tue, Jul 31, 2012 at 2:37 PM, Thomas Hoellriegel wrote: > Hi SamyGo > > 1st Argument: $${dorf_mailbox_prefix}/${****mailbox_to}/greeting.alaw >> >> 2nd Argument: ${mailbox} >> 3rd Argument: ${msgtime} >> >> Can you verify that the ${msgtime} is properly calculated and is not NULL >> at the taime of execution of application=lua ? >> > > I like to save a message in the timestamp format: > for example: > msg_2012_31_07_15_32_31.alaw > I read the current date and time from: > ${msgtime} > But i have no output. > What is wrong please? > > thanks. > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120731/96282ad1/attachment.html From virbhati at gmail.com Tue Jul 31 14:58:53 2012 From: virbhati at gmail.com (virendra bhati) Date: Tue, 31 Jul 2012 16:28:53 +0530 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 73, Issue 228 In-Reply-To: References: Message-ID: Yes ignore_early_media=true no improvement On Tue, Jul 31, 2012 at 2:19 PM, < freeswitch-users-request at lists.freeswitch.org> wrote: > 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: Nibble Billing deducted balance from account even call > doesn't come to my mobile number (Evgeniy Movlyan) > 2. Re: Am I using play_and_detect_speech correctly? (x.liu) > 3. Re: Nibble Billing deducted balance from account even call > doesn't come to my mobile number (SamyGo) > 4. Re: Question how can i get variables in lua from xml > dialplan? (Thomas Hoellriegel) > > > ---------- Forwarded message ---------- > From: Evgeniy Movlyan > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Tue, 31 Jul 2012 10:48:07 +0300 > Subject: Re: [Freeswitch-users] Nibble Billing deducted balance from > account even call doesn't come to my mobile number > I have the same problem. When i am calling from one my extension to > another all is ok, but when i am calling to external number i got this > message: "mod_nibblebill.c:465 Not billing XXXXXXXXXX - call is not in > answered state". > > 31.07.2012 10:35, virendra bhati ?????: > >> mod_nibblebill.c:465 Not billing >> 97183008 - call is not in answered state >> > > -- > Evgeniy Movlyan, > BestNet Ltd. > > > > > ---------- Forwarded message ---------- > From: "x.liu" > To: freeswitch-users at lists.freeswitch.org > Cc: > Date: Tue, 31 Jul 2012 08:59:37 +0100 > Subject: Re: [Freeswitch-users] Am I using play_and_detect_speech > correctly? > Hi Chris, > > Okay, thanks! I will have tries to see how it works. > > For the normal use of the app: play_and_detect_speech, will it resume > previous ASR session, > or will it stop previous session and restart a new session each time when > I issue it via ESL? > > For the app of detect_speech, I can first to start an ASR session by > "detect_speech speechMod grammar grammarPath", > then I can do "detect_speech pause", or "detect_speech resume" or > "detect_speech stop". > But for play_and_detect_speech, I am not sure how it works in terms of > starting, pause, resume/restart sequence. > > The play_and_detect_speech is a very good, useful app as it supports the > barge-in. So we definitely need to use it. > > Thanks! > > Xing > > > > > On 07/31/2012 12:53 AM, Christopher Rienzo wrote: > > You are using that APP in a way that hasn't been tested. Try to do > "speech_detect pause" and see if that helps. It will reserve your ASR > session for reuse while "speech_detect stop" will tear it down completely. > If that doesn't work, open a jira ticket and attach a full call trace. > It's difficult to understand exactly what is happening from your > description. > > > Chris > > > > On Mon, Jul 30, 2012 at 5:38 PM, Liu, Xingkun wrote: > >> Hello, >> >> I am using play_and_detect_speech with Java ESL in my IVR applications. >> >> Previously I call it again each time after I receive any recognition >> event, >> like recognition complete, no-input-timeout, or recognition-timeout, >> it seems to work fine. >> >> Now I have changed my app to issue play_and_detect_speech command based on >> my available system utterances as well as the speech event. >> >> I.e., I use a separate thread to constantly check if there is a system >> utterance coming in >> from another component of my application, if there is any utterance I >> issue the command which will >> speak the new utterance and listen to user input no matter whether or not >> previous >> command has finished. And if there is any speech event (recognition >> result, timeout etc.) >> the play_and_detect_speech command is also issued but with playing >> silence. >> >> Obviously the new command will stop the utterance speaking of the >> previous command if it is not finished. >> >> My question is >> >> will the new play_and_detect_speech command also stop the previous ASR >> listening >> or will there be many ASR listening channel and sending speech data (or >> silence) to ASR server? >> >> Do I need to explicitly issue a "stop" commnad before issuing a new >> play_and_detect_speech? >> If yes, how to do that, by "detect_speech stop"? >> >> Recently there is a network traffice problem (lots of connections /data >> transportation to the ASR server machine) >> when I am running my application. I am not sure if this is because of >> other issues >> or because of my new changes to the way of using play_and_detect_speech. >> >> Please any one could shed a light on this? >> >> Many thanks! >> >> Xing >> >> >> ------------------------------ >> >> *Heriot-Watt University is the Sunday Times Scottish University of the >> Year 2011-2012.* >> >> We invite research leaders and ambitious early career researchers to >> join us in leading and driving research in key inter-disciplinary themes. >> Please see www.hw.ac.uk/researchleaders for further information and how >> to apply. >> >> Heriot-Watt University is a Scottish charity registered under charity >> number SC000278. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > ------------------------------ > > *Heriot-Watt University is the Sunday Times Scottish University of the > Year 2011-2012.* > > We invite research leaders and ambitious early career researchers to join > us in leading and driving research in key inter-disciplinary themes. Please > see www.hw.ac.uk/researchleaders for further information and how to > apply. > > Heriot-Watt University is a Scottish charity registered under charity > number SC000278. > > > ---------- Forwarded message ---------- > From: SamyGo > To: FreeSWITCH Users Help > Cc: > Date: Tue, 31 Jul 2012 13:06:05 +0500 > Subject: Re: [Freeswitch-users] Nibble Billing deducted balance from > account even call doesn't come to my mobile number > > Hi, > Well its works perfect for me, do you guys have ignore_early_media set in > your outbound string, if no then set it and then see what happens. > On Jul 31, 2012 12:50 PM, "Evgeniy Movlyan" > wrote: > >> I have the same problem. When i am calling from one my extension to >> another all is ok, but when i am calling to external number i got this >> message: "mod_nibblebill.c:465 Not billing XXXXXXXXXX - call is not in >> answered state". >> >> 31.07.2012 10:35, virendra bhati ?????: >> > mod_nibblebill.c:465 Not billing >> > 97183008 - call is not in answered state >> >> -- >> Evgeniy Movlyan, >> BestNet Ltd. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > ---------- Forwarded message ---------- > From: Thomas Hoellriegel > To: FreeSWITCH Users Help > Cc: > Date: Tue, 31 Jul 2012 10:48:52 +0200 (CEST) > Subject: Re: [Freeswitch-users] Question how can i get variables in lua > from xml dialplan? > Hi Seven, > Thanks! its works > > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Thanks and regards Virendra Bhati +91-9718300881 Asterisk Developer E-mail-: virbhati at gmail.com Skype id:- virbhati2 New Delhi(India) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/866f369e/attachment-0001.html From govoiper at gmail.com Tue Jul 31 15:14:47 2012 From: govoiper at gmail.com (SamyGo) Date: Tue, 31 Jul 2012 16:14:47 +0500 Subject: [Freeswitch-users] Nibble Billing deducted balance from account even call doesn't come to my mobile number In-Reply-To: References: <50178DB7.7000606@bestnet.kharkov.ua> Message-ID: Hi, I see that you've also turned on the bridge_early_media, for me that means if 183 comes from the b-leg then bridge it. And as soon the legs get bridged nibblebill session gets in action. Can you try setting it off/false ? http://wiki.freeswitch.org/wiki/Variable_bridge_early_media BR Sammy On Tue, Jul 31, 2012 at 1:06 PM, SamyGo wrote: > Hi, > Well its works perfect for me, do you guys have ignore_early_media set in > your outbound string, if no then set it and then see what happens. > On Jul 31, 2012 12:50 PM, "Evgeniy Movlyan" > wrote: > >> I have the same problem. When i am calling from one my extension to >> another all is ok, but when i am calling to external number i got this >> message: "mod_nibblebill.c:465 Not billing XXXXXXXXXX - call is not in >> answered state". >> >> 31.07.2012 10:35, virendra bhati ?????: >> > mod_nibblebill.c:465 Not billing >> > 97183008 - call is not in answered state >> >> -- >> Evgeniy Movlyan, >> BestNet Ltd. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20120731/f77059e1/attachment.html From lists at telefaks.de Tue Jul 31 15:47:13 2012 From: lists at telefaks.de (Peter Steinbach) Date: Tue, 31 Jul 2012 13:47:13 +0200 Subject: [Freeswitch-users] EVENT_TRAP: IP change detected In-Reply-To: <4F59430B.8040004@tiendalinux.com> References: <4F4D5F07.4040301@tiendalinux.com> <4F59430B.8040004@tiendalinux.com> Message-ID: <5017C5C1.7020602@telefaks.de> I currently have the same problem. 2012-07-31 08:52:02.167397 [INFO] mod_sofia.c:5173 EVENT_TRAP: IP change detected 2012-07-31 08:52:02.167397 [INFO] mod_sofia.c:5174 IP change detected [10.2.2.11]->[192.168.1.1] []->[] and then the call gets hungup up due to missing rtp. That means: for the internal profile the ip 10.2.2.11 changed to 192.168.1.1 which is the IP of the external interface There is also a commented parameter in sofia.conf.xml Is it advisable to also change this to ? -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de On 03/09/12 00:38, Nestor A Diaz wrote: > On 02/28/2012 11:26 PM, Anthony Minessale wrote: >> it happens when the route to the internet changes to another device. >> you an disable this: >> >> edit your sofia profile file internal.xml >> >> and comment this: >> >> >> > Hi Anthony , but the route didn't change, i just added aliases to the > external interface (mistake: i wrote internal interface on the first > mail but it was the external interface), the funny thing is that no > matter if i disabled the aliases, after some calls the freeswitch > trigger the event: > > 2012-03-08 16:04:56.186572 [INFO] mod_sofia.c:5155 EVENT_TRAP: IP change > detected > > and no internal profile anymore, i have already disabled the aliases > when this happened. > > The external interface was a network card with native vlan and the > internal interface was a vlan over the same hardware interface. (just > because i used to separate voice from data on different vlans) > > I just put your fix on the internal profile, I hope the problem does not > happen again. > > Hope that will give you some hints. > > Thanks for such awesome piece of software ! > > Slds. > From anthony.minessale at gmail.com Tue Jul 31 16:18:41 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 31 Jul 2012 07:18:41 -0500 Subject: [Freeswitch-users] LUA session:Bridge not actually bridging calls ~ In-Reply-To: References: Message-ID: Not at the moment. If everyone is going to start doing this we may have to change something. On Jul 31, 2012 12:41 AM, "SamyGo" wrote: > Though the problem is solved but is there any corresponding WARN or > anything in the logs so that anyone can pick the problem right away. > This fixing of codecs took alot of time and I was never expecting that > this could be the cause Thinking that its Bridge() which is not doing its > part. > > Thanks a lot for your time on this. > > Regards, > Sammy > > > On Mon, Jul 30, 2012 at 6:22 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> This script is actually a bit over-complicated >> >> dialA = "sofia/gateway/fs1/9903" >> dialB = "user/1001" >> legA = freeswitch.Session(dialA) >> >> if (legA:ready()) { >> legA:execute("bridge", dialB) >> } >> >> I think the problem is that we don't support dealing with many codecs >> in the 200ok >> They send gsm and ulaw and then choose ulaw and we go with the first one >> gsm >> >> >> >> On Sat, Jul 28, 2012 at 11:23 AM, SamyGo wrote: >> > Hi again, >> > >> > So, It was a very minor change in configuration and it was working. >> > Basically FreeSwicth was bridging the two legs BUT there was a codec >> issue. >> > All I had to do was in Asterisk (serving as my gateway) to allow only >> ulaw >> > and alaw >> > i.e >> > >> > disallow=all >> > allow=ulaw >> > allow=alaw >> > >> > What I really really wish to know is that why there was no indication of >> > codec mismatch or ptime mismatch or sample rate mismatch while >> transcoding >> > or anything. >> > >> > It will be fine if none replies but it will be great to know the real >> > reason behind this and from where in logs can I verify this !! >> > >> > Thanks >> > Sammy >> > >> > >> > On Sat, Jul 28, 2012 at 8:18 PM, SamyGo wrote: >> >> >> >> Here are the FS console logs: >> >> http://pastebin.freeswitch.org/19595 >> >> >> >> Please suggest what am I missing here. >> >> >> >> >> >> On Sat, Jul 28, 2012 at 7:59 PM, SamyGo wrote: >> >>> >> >>> Hello, >> >>> I wanted to make a lua script which just dials out two different >> numbers >> >>> via some external gateway and when both calls are answered they are >> just >> >>> bridged. For this a very impressive Lua example >> >>> http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control is >> copied and >> >>> all I had to do was change the dialA and dialB strings and its >> working great >> >>> as far as the SIP signalling is concerned. >> >>> >> >>> execute this string and I get calls on two different number but things >> >>> get interesting when Freeswitch bridge() the two legs. No AUDIO..not >> even >> >>> one-way. I could see on my own gateway that RTPs for both the legs >> are >> >>> actually forwarded to Freeswitch ! >> >>> >> >>> On my sip pcap traces analyzing on wireshark I could actually hear the >> >>> two persons saying Hello but neither could hear anything. >> >>> >> >>> The above example lua call_control script is used as it is. >> >>> Please suggest. >> >>> >> >>> Regards >> >>> Sammy Go. >> >>> >> >> >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120731/4c0881cf/attachment.html From anthony.minessale at gmail.com Tue Jul 31 16:24:10 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 31 Jul 2012 07:24:10 -0500 Subject: [Freeswitch-users] EVENT_TRAP: IP change detected In-Reply-To: <5017C5C1.7020602@telefaks.de> References: <4F4D5F07.4040301@tiendalinux.com> <4F59430B.8040004@tiendalinux.com> <5017C5C1.7020602@telefaks.de> Message-ID: If the path to the internet changes by doing a bind and getting the primary ip of the interface, this fires. If its causing problems, you can disable it in sofia.conf.xml On Jul 31, 2012 7:48 AM, "Peter Steinbach" wrote: > I currently have the same problem. > > 2012-07-31 08:52:02.167397 [INFO] mod_sofia.c:5173 EVENT_TRAP: IP change > detected > 2012-07-31 08:52:02.167397 [INFO] mod_sofia.c:5174 IP change detected > [10.2.2.11]->[192.168.1.1] []->[] > and then the call gets hungup up due to missing rtp. > That means: for the internal profile the ip 10.2.2.11 changed to > 192.168.1.1 which is the IP of the external interface > > There is also a commented parameter in sofia.conf.xml > > Is it advisable to also change this to value="false"/>? > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet: www.telefaks.de > > > > On 03/09/12 00:38, Nestor A Diaz wrote: > > On 02/28/2012 11:26 PM, Anthony Minessale wrote: > >> it happens when the route to the internet changes to another device. > >> you an disable this: > >> > >> edit your sofia profile file internal.xml > >> > >> and comment this: > >> > >> > >> > > Hi Anthony , but the route didn't change, i just added aliases to the > > external interface (mistake: i wrote internal interface on the first > > mail but it was the external interface), the funny thing is that no > > matter if i disabled the aliases, after some calls the freeswitch > > trigger the event: > > > > 2012-03-08 16:04:56.186572 [INFO] mod_sofia.c:5155 EVENT_TRAP: IP change > > detected > > > > and no internal profile anymore, i have already disabled the aliases > > when this happened. > > > > The external interface was a network card with native vlan and the > > internal interface was a vlan over the same hardware interface. (just > > because i used to separate voice from data on different vlans) > > > > I just put your fix on the internal profile, I hope the problem does not > > happen again. > > > > Hope that will give you some hints. > > > > Thanks for such awesome piece of software ! > > > > Slds. > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120731/6a24fb53/attachment-0001.html From cmrienzo at gmail.com Tue Jul 31 16:44:50 2012 From: cmrienzo at gmail.com (Christopher Rienzo) Date: Tue, 31 Jul 2012 08:44:50 -0400 Subject: [Freeswitch-users] Am I using play_and_detect_speech correctly? In-Reply-To: <50179069.6020700@hw.ac.uk> References: <50179069.6020700@hw.ac.uk> Message-ID: I looked over the code again last night and verified that play_and_detect_speech should try to stop detection before loading the grammar. However, that scenario is not really tested, so try the workaround first. If that doesn't work, open a jira ticket with detailed logs so it can be fixed. You can use the detect_speech APP to do everything over ESL. It is a lot more work, but once you figure it out, you have the freedom to do whatever you want, including handling barge-in. play_and_detect_speech is designed to handle the typical use case. Chris On Tue, Jul 31, 2012 at 3:59 AM, x.liu wrote: > Hi Chris, > > Okay, thanks! I will have tries to see how it works. > > For the normal use of the app: play_and_detect_speech, will it resume > previous ASR session, > or will it stop previous session and restart a new session each time when > I issue it via ESL? > > For the app of detect_speech, I can first to start an ASR session by > "detect_speech speechMod grammar grammarPath", > then I can do "detect_speech pause", or "detect_speech resume" or > "detect_speech stop". > But for play_and_detect_speech, I am not sure how it works in terms of > starting, pause, resume/restart sequence. > > The play_and_detect_speech is a very good, useful app as it supports the > barge-in. So we definitely need to use it. > > Thanks! > > Xing > > > > > > On 07/31/2012 12:53 AM, Christopher Rienzo wrote: > > You are using that APP in a way that hasn't been tested. Try to do > "speech_detect pause" and see if that helps. It will reserve your ASR > session for reuse while "speech_detect stop" will tear it down completely. > If that doesn't work, open a jira ticket and attach a full call trace. > It's difficult to understand exactly what is happening from your > description. > > > Chris > > > > On Mon, Jul 30, 2012 at 5:38 PM, Liu, Xingkun wrote: > >> Hello, >> >> I am using play_and_detect_speech with Java ESL in my IVR applications. >> >> Previously I call it again each time after I receive any recognition >> event, >> like recognition complete, no-input-timeout, or recognition-timeout, >> it seems to work fine. >> >> Now I have changed my app to issue play_and_detect_speech command based on >> my available system utterances as well as the speech event. >> >> I.e., I use a separate thread to constantly check if there is a system >> utterance coming in >> from another component of my application, if there is any utterance I >> issue the command which will >> speak the new utterance and listen to user input no matter whether or not >> previous >> command has finished. And if there is any speech event (recognition >> result, timeout etc.) >> the play_and_detect_speech command is also issued but with playing >> silence. >> >> Obviously the new command will stop the utterance speaking of the >> previous command if it is not finished. >> >> My question is >> >> will the new play_and_detect_speech command also stop the previous ASR >> listening >> or will there be many ASR listening channel and sending speech data (or >> silence) to ASR server? >> >> Do I need to explicitly issue a "stop" commnad before issuing a new >> play_and_detect_speech? >> If yes, how to do that, by "detect_speech stop"? >> >> Recently there is a network traffice problem (lots of connections /data >> transportation to the ASR server machine) >> when I am running my application. I am not sure if this is because of >> other issues >> or because of my new changes to the way of using play_and_detect_speech. >> >> Please any one could shed a light on this? >> >> Many thanks! >> >> Xing >> >> >> ------------------------------ >> >> *Heriot-Watt University is the Sunday Times Scottish University of the >> Year 2011-2012.* >> >> We invite research leaders and ambitious early career researchers to >> join us in leading and driving research in key inter-disciplinary themes. >> Please see www.hw.ac.uk/researchleaders for further information and how >> to apply. >> >> Heriot-Watt University is a Scottish charity registered under charity >> number SC000278. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > ------------------------------ > > *Heriot-Watt University is the Sunday Times Scottish University of the > Year 2011-2012.* > > We invite research leaders and ambitious early career researchers to join > us in leading and driving research in key inter-disciplinary themes. Please > see www.hw.ac.uk/researchleaders for further information and how to > apply. > > Heriot-Watt University is a Scottish charity registered under charity > number SC000278. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120731/41d5c9ac/attachment.html From fieldpeak at gmail.com Tue Jul 31 16:44:54 2012 From: fieldpeak at gmail.com (fieldpeak) Date: Tue, 31 Jul 2012 20:44:54 +0800 Subject: [Freeswitch-users] Help!! FS -TLS interworking issue, How to config to allow "gentls_cert" to generate a root certificate with more longer valid-period ? In-Reply-To: References: <71943DD5C22943448A24B7C5CDC238070FF1A549@SN2PRD0410MB396.namprd04.prod.outlook.com> Message-ID: *Hi All, gentls_cert.in works well, it resolved this problem. Thank you all very much!! [?] Best Regards, Charles * 2012/7/31 Michael Collins > > > On Mon, Jul 30, 2012 at 9:28 AM, Robert Hadley wrote: > >> Hi Charles,**** >> >> ** ** >> >> Try the changes in this attached freeswitch/scripts/gentls_cert.in file. >> There were a few typos in the original script.**** >> >> ** ** >> >> Regards,**** >> >> Robert >> > > I'd like to verify that those typos are indeed really typos and are really > fixed. If anyone has input on them please let me know and I will see about > getting the gentls_cert.in file updated. I definitely would like to see > this tested before we make any updates. > > Thanks, > MC > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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, Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/74b6cca1/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 125 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/74b6cca1/attachment.gif From ronmccar at gmail.com Tue Jul 31 16:59:49 2012 From: ronmccar at gmail.com (Ron McCarthy) Date: Tue, 31 Jul 2012 05:59:49 -0700 Subject: [Freeswitch-users] 482 sent on serial forking Message-ID: Hi, I have a OpenSIPs box running drouting that is sending calls to freeswitch for CDR and B2BUA purposes. When the first INVITE is sent to freeswitch the call routes correct with no issues, but if the first attempt fails for whatever reason then OpenSIPs will pick the next gateway ID and send the call to freeswitch, at that point a 482 is sent back. I lowered the T4 timer in freeswitch to 250ms but still get this error, I saw another post that freeswitch thinks the dialog is still active and thus causing the 482. Any ideals on this? I am using append_branch on the openSIPs side but I don't think that matters and the dialog on freeswitch is done by callID from what I can tell. Im pretty sure this is a common setup so I think it's something easy im missing. Thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/75f58125/attachment-0001.html From bdfoster at endigotech.com Tue Jul 31 17:10:03 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 31 Jul 2012 09:10:03 -0400 Subject: [Freeswitch-users] Skypeopen In-Reply-To: References: <501706D7.3040403@ition.com.ar> Message-ID: It works and works well... if you follow the instructions. Btw Giovanni is the developer. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 31, 2012 2:35 AM, "Giovanni Maruzzelli" wrote: > have a look at: > http://wiki.freeswitch.org/wiki/Skypopen > > > On Tue, Jul 31, 2012 at 12:12 AM, Sergio Galeotti (ITION) < > sgaleotti at ition.com.ar> wrote: > >> Hi; >> I'm new in FS user list. >> There is someone who is using the Skype_Open module for incoming and >> outgoing? >> I would like to contact someone who is experiencing as it does not find >> much information about it and I have many doubts about the operation. >> I have the platform installed and configured, but there are some things >> if they are not bugs, limitations or poor configuration on my part >> thanks!! >> >> Sergio >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120731/b18bdbc6/attachment.html From lloyd.aloysius at gmail.com Tue Jul 31 17:37:21 2012 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Tue, 31 Jul 2012 09:37:21 -0400 Subject: [Freeswitch-users] xml cdr help Message-ID: Hi All: I implement the xml cdr. Now For a Incoming call -> IVR - > Dial Extension create two records see below. Basically both records should say inbound call . But one say as outbound. What is the reason the xml cdr module behave this way. Also for a one incoming call to a destination why showing the two records? Also I notice when we make a outbound call the direction field say inbound. But it should be outbound. Any help is appreciated. callerid number , destination number , uuid ,bridge uuid , date ,time 1416471234 144 outbound *7f055bf3-6622-4a47-8a86-e92c7683d74d*898cc58c-8447-4a51-b559-3d1f2a063014 2012/07/30 15:37:41 00:00:12 1416471234 144 inbound 898cc58c-8447-4a51-b559-3d1f2a063014 * 7f055bf3-6622-4a47-8a86-e92c7683d74d* 2012/07/301 5:37:23 00:00:30 Thanks Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/4c18d9ce/attachment.html From ramesh_mind at yahoo.com Tue Jul 31 17:40:01 2012 From: ramesh_mind at yahoo.com (ramesh) Date: Tue, 31 Jul 2012 06:40:01 -0700 (PDT) Subject: [Freeswitch-users] Error While Dialing Two Numbers either serially or simultaneously Message-ID: <1343742001770-7581362.post@n2.nabble.com> Hi Team, Can any one help me by looking into the below lua script and tel me whats wrong in this! Am getting error for the past few days session:execute("bridge","{ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7,originate_continue_on_timeout=true}sofia/gateway/bandwidth.com/+919458595856,{ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7}sofia/gateway/bandwidth.com/+919923920339"); actually this script was working before, below is the error i get EXECUTE sofia/internal/+15595158870 at 4.55.21.99:5060 lua(/lua/freeswitch/scripts/8887129962_dialplan.lua f6b2ef40-db12-11e1-b609-5bed86da9fbc 8887129962 +15595158870 2012-07-31 13:23:49) EXECUTE sofia/internal/+15595158870 at 4.55.21.99:5060 bridge({ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7,originate_continue_on_timeout=true}sofia/gateway/bandwidth.com/+919884978816,{ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7}sofia/gateway/bandwidth.com/+919677080275) 2012-07-31 13:23:49.252695 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-07-31 13:23:49.252695 [DEBUG] switch_event.c:1521 Parsing variable [ignore_early_media]=[true] 2012-07-31 13:23:49.252695 [DEBUG] switch_event.c:1521 Parsing variable [monitor_early_media_fail]=[user_busy:1:480+620!destination_out_of_order:2:1776.7] 2012-07-31 13:23:49.252695 [DEBUG] switch_event.c:1521 Parsing variable [originate_continue_on_timeout]=[true] 2012-07-31 13:23:49.252695 [NOTICE] switch_channel.c:930 New Channel sofia/internal/+919884978816 [f6b57b20-db12-11e1-b60d-5bed86da9fbc] 2012-07-31 13:23:49.252695 [DEBUG] mod_sofia.c:4670 (sofia/internal/+919884978816) State Change CS_NEW -> CS_INIT 2012-07-31 13:23:49.252695 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/+919884978816 [BREAK] 2012-07-31 13:23:49.252695 [ERR] switch_core_session.c:427 *Could not locate channel type {ignore_early_media=true 2012-07-31 13:23:49.252695 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [{ignore_early_media=true] cause: [CHAN_NOT_IMPLEMENTED] 2012-07-31 13:23:49.252695 [ERR] switch_core_session.c:427 Could not locate channel type monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7}sofia 2012-07-31 13:23:49.252695 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7}sofia] cause: [CHAN_NOT_IMPLEMENTED] * i dont know what i missed!!!! Any Help or Suggestion would be really helpful! Thanks Ramesh -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-While-Dialing-Two-Numbers-either-serially-or-simultaneously-tp7581362.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/20120731/5a1ca3a5/attachment.html From ramesh_mind at yahoo.com Tue Jul 31 17:43:39 2012 From: ramesh_mind at yahoo.com (ramesh) Date: Tue, 31 Jul 2012 06:43:39 -0700 (PDT) Subject: [Freeswitch-users] Unable To Dial Two Numbers either serially or simultaneously Message-ID: <1343742219348-7581365.post@n2.nabble.com> Hi Team , I got a script which will dial a numbers either serially or simultaneously , but for the past few days i was getting error , can anyone hint what is wrong in the following script . EXECUTE sofia/internal/+15595158870 at 4.55.21.99:5060 lua(/lua/freeswitch/scripts/8887129962_dialplan.lua f6b2ef40-db12-11e1-b609-5bed86da9fbc 8887129962 +15595158870 2012-07-31 13:23:49) EXECUTE sofia/internal/+15595158870 at 4.55.21.99:5060 bridge({ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7,originate_continue_on_timeout=true}sofia/gateway/bandwidth.com/+919884978816,{ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7}sofia/gateway/bandwidth.com/+919677080275) 2012-07-31 13:23:49.252695 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-07-31 13:23:49.252695 [DEBUG] switch_event.c:1521 Parsing variable [ignore_early_media]=[true] 2012-07-31 13:23:49.252695 [DEBUG] switch_event.c:1521 Parsing variable [monitor_early_media_fail]=[user_busy:1:480+620!destination_out_of_order:2:1776.7] 2012-07-31 13:23:49.252695 [DEBUG] switch_event.c:1521 Parsing variable [originate_continue_on_timeout]=[true] 2012-07-31 13:23:49.252695 [NOTICE] switch_channel.c:930 New Channel sofia/internal/+919884978816 [f6b57b20-db12-11e1-b60d-5bed86da9fbc] 2012-07-31 13:23:49.252695 [DEBUG] mod_sofia.c:4670 (sofia/internal/+919884978816) State Change CS_NEW -> CS_INIT 2012-07-31 13:23:49.252695 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/+919884978816 [BREAK] 2012-07-31 13:23:49.252695 [ERR] switch_core_session.c:427 *Could not locate channel type {ignore_early_media=true 2012-07-31 13:23:49.252695 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [{ignore_early_media=true] cause: [CHAN_NOT_IMPLEMENTED] 2012-07-31 13:23:49.252695 [ERR] switch_core_session.c:427 Could not locate channel type monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7}sofia 2012-07-31 13:23:49.252695 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7}sofia] cause: [CHAN_NOT_IMPLEMENTED]* I dont know what i missed Any Help or Suggestion will help me a lot Thanks Ramesh -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Unable-To-Dial-Two-Numbers-either-serially-or-simultaneously-tp7581365.html Sent from the freeswitch-users mailing list archive at Nabble.com. From bdfoster at endigotech.com Tue Jul 31 17:43:17 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 31 Jul 2012 09:43:17 -0400 Subject: [Freeswitch-users] xml cdr help In-Reply-To: References: Message-ID: Freeswitch is a B2BUA. When it says inbound or outbound it is relative to the switch. When you dial another extension you are starting another call leg, going outbound to the phone. This isn't a bug or a configuration issue. It's what makes sense. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 31, 2012 9:39 AM, "Lloyd Aloysius" wrote: > Hi All: > > I implement the xml cdr. Now For a Incoming call -> IVR - > Dial Extension > create two records see below. Basically both records should say inbound > call . But one say as outbound. What is the reason the xml cdr module > behave this way. Also for a one incoming call to a destination why showing > the two records? > > > Also I notice when we make a outbound call the direction field say > inbound. But it should be outbound. Any help is appreciated. > > callerid number , destination number , uuid ,bridge uuid , date ,time > > 1416471234 144 outbound *7f055bf3-6622-4a47-8a86-e92c7683d74d*898cc58c-8447-4a51-b559-3d1f2a063014 2012/07/30 15:37:41 00:00:12 > > 1416471234 144 inbound 898cc58c-8447-4a51-b559-3d1f2a063014 * > 7f055bf3-6622-4a47-8a86-e92c7683d74d* 2012/07/301 5:37:23 00:00:30 > > > Thanks > Lloyd > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120731/ea061f62/attachment-0001.html From alex at thewinelake.com Tue Jul 31 17:45:00 2012 From: alex at thewinelake.com (Alex) Date: Tue, 31 Jul 2012 14:45:00 +0100 Subject: [Freeswitch-users] PAGD 0 digits In-Reply-To: References: <9D0E55F4-768E-4C9D-B1DF-9D074E44ADD7@mgtech.com> <458A0010-EC8C-4ACC-B08D-A02E0AD26077@mgtech.com> <500FA9BA.5030303@thewinelake.com> <50110823.2060604@thewinelake.com> <50128323.7000008@thewinelake.com> Message-ID: <5017E15C.9060707@thewinelake.com> Not really. But I discovered that 0 IS a valid value for max digits. > After the timeout it simply moves on to the next application. Would > that not suffice? > -MC > > On Fri, Jul 27, 2012 at 5:01 AM, Alex > wrote: > > If I want to allow someone to enter an empty string of digits, how > is it > done? > > I've tried various things... > > My latest effort (that still doesn't work!) is > > keypress = session:playAndGetDigits(1, 1, 3, 5000, "*#", > what_to_play, "", "\\d{0,1}|\\#|\\*") > terminator = session:getVariable("read_terminator_used") > > I note that the minimum value for the first param is 1 - so maybe it > can't be done? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2196 / Virus Database: 2437/5166 - Release Date: 07/30/12 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/01ce522b/attachment.html From bdfoster at endigotech.com Tue Jul 31 17:55:52 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 31 Jul 2012 09:55:52 -0400 Subject: [Freeswitch-users] Error While Dialing Two Numbers either serially or simultaneously In-Reply-To: <1343742001770-7581362.post@n2.nabble.com> References: <1343742001770-7581362.post@n2.nabble.com> Message-ID: Probably a } or two. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 31, 2012 9:41 AM, "ramesh" wrote: > Hi Team, Can any one help me by looking into the below lua script and tel > me whats wrong in this! Am getting error for the past few days > session:execute("bridge","{ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7,originate_continue_on_timeout=true}sofia/gateway/ > bandwidth.com/+919458595856 > ,{ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7}sofia/gateway/ > bandwidth.com/+919923920339"); actually this script was working before, > below is the error i get EXECUTE sofia/internal/+ > 15595158870 at 4.55.21.99:5060lua(/lua/freeswitch/scripts/8887129962_dialplan.lua > f6b2ef40-db12-11e1-b609-5bed86da9fbc 8887129962 +15595158870 2012-07-31 > 13:23:49) EXECUTE sofia/internal/+15595158870 at 4.55.21.99:5060bridge({ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7,originate_continue_on_timeout=true}sofia/gateway/ > bandwidth.com/+919884978816,{ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7}sofia/gateway/bandwidth.com/+919677080275) > 2012-07-31 13:23:49.252695 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables 2012-07-31 13:23:49.252695 [DEBUG] switch_event.c:1521 > Parsing variable [ignore_early_media]=[true] 2012-07-31 13:23:49.252695 > [DEBUG] switch_event.c:1521 Parsing variable > [monitor_early_media_fail]=[user_busy:1:480+620!destination_out_of_order:2:1776.7] > 2012-07-31 13:23:49.252695 [DEBUG] switch_event.c:1521 Parsing variable > [originate_continue_on_timeout]=[true] 2012-07-31 13:23:49.252695 [NOTICE] > switch_channel.c:930 New Channel sofia/internal/+919884978816 > [f6b57b20-db12-11e1-b60d-5bed86da9fbc] 2012-07-31 13:23:49.252695 [DEBUG] > mod_sofia.c:4670 (sofia/internal/+919884978816) State Change CS_NEW -> > CS_INIT 2012-07-31 13:23:49.252695 [DEBUG] switch_core_session.c:1180 Send > signal sofia/internal/+919884978816 [BREAK] 2012-07-31 13:23:49.252695 > [ERR] switch_core_session.c:427 *Could not locate channel type > {ignore_early_media=true 2012-07-31 13:23:49.252695 [NOTICE] > switch_ivr_originate.c:2459 Cannot create outgoing channel of type > [{ignore_early_media=true] cause: [CHAN_NOT_IMPLEMENTED] 2012-07-31 > 13:23:49.252695 [ERR] switch_core_session.c:427 Could not locate channel > type > monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7}sofia > 2012-07-31 13:23:49.252695 [NOTICE] switch_ivr_originate.c:2459 Cannot > create outgoing channel of type > [monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7}sofia] > cause: [CHAN_NOT_IMPLEMENTED] * i dont know what i missed!!!! Any Help or > Suggestion would be really helpful! Thanks Ramesh > ------------------------------ > View this message in context: Error While Dialing Two Numbers either > serially or simultaneously > Sent from the freeswitch-users mailing list archiveat Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120731/cb709acf/attachment.html From bdfoster at endigotech.com Tue Jul 31 17:55:52 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 31 Jul 2012 09:55:52 -0400 Subject: [Freeswitch-users] PAGD 0 digits In-Reply-To: <5017E15C.9060707@thewinelake.com> References: <9D0E55F4-768E-4C9D-B1DF-9D074E44ADD7@mgtech.com> <458A0010-EC8C-4ACC-B08D-A02E0AD26077@mgtech.com> <500FA9BA.5030303@thewinelake.com> <50110823.2060604@thewinelake.com> <50128323.7000008@thewinelake.com> <5017E15C.9060707@thewinelake.com> Message-ID: In that case 0 means no maximum. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 31, 2012 9:45 AM, "Alex" wrote: > Not really. But I discovered that 0 IS a valid value for max digits. > > After the timeout it simply moves on to the next application. Would that > not suffice? > -MC > > On Fri, Jul 27, 2012 at 5:01 AM, Alex wrote: > >> If I want to allow someone to enter an empty string of digits, how is it >> done? >> >> I've tried various things... >> >> My latest effort (that still doesn't work!) is >> >> keypress = session:playAndGetDigits(1, 1, 3, 5000, "*#", >> what_to_play, "", "\\d{0,1}|\\#|\\*") >> terminator = session:getVariable("read_terminator_used") >> >> I note that the minimum value for the first param is 1 - so maybe it >> can't be done? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > -- > Michael S Collins > Twitter: @mercutioviz > http://www.FreeSWITCH.org > http://www.ClueCon.com > http://www.OSTAG.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2196 / Virus Database: 2437/5166 - Release Date: 07/30/12 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120731/c949f649/attachment-0001.html From ramesh_mind at yahoo.com Tue Jul 31 18:12:30 2012 From: ramesh_mind at yahoo.com (ramesh) Date: Tue, 31 Jul 2012 07:12:30 -0700 (PDT) Subject: [Freeswitch-users] Error While Dialing Two Numbers either serially or simultaneously In-Reply-To: References: <1343742001770-7581362.post@n2.nabble.com> Message-ID: <1343743950224-7581370.post@n2.nabble.com> Im sorry i dint get u , where should i use } , the variables should be enclosed withing {} and i made it tat way , can u tel me where i missed } -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-While-Dialing-Two-Numbers-either-serially-or-simultaneously-tp7581364p7581370.html Sent from the freeswitch-users mailing list archive at Nabble.com. From avi at avimarcus.net Tue Jul 31 18:15:32 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 31 Jul 2012 17:15:32 +0300 Subject: [Freeswitch-users] Error While Dialing Two Numbers either serially or simultaneously In-Reply-To: References: <1343742001770-7581362.post@n2.nabble.com> Message-ID: As Brian was saying, you can only use {} once -- it's for global variables. Since you want the same string on both, you can do simply: {ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7,originate_continue_on_timeout=true}sofia/gateway/bandwidth.com/+919458595856,sofia/gateway/bandwidth.com/+919923920339 Or encase each leg's variables in [] instead. By the way, this means " 'and' -- call both at the same time". Use a | (or) separator to only do the second once the first fails. -Avi On Tue, Jul 31, 2012 at 4:55 PM, Brian Foster wrote: > Probably a } or two. > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > > On Jul 31, 2012 9:41 AM, "ramesh" wrote: >> >> Hi Team, Can any one help me by looking into the below lua script and tel >> me whats wrong in this! Am getting error for the past few days >> session:execute("bridge","{ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7,originate_continue_on_timeout=true}sofia/gateway/bandwidth.com/+919458595856,{ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7}sofia/gateway/bandwidth.com/+919923920339"); >> actually this script was working before, below is the error i get EXECUTE >> sofia/internal/+15595158870 at 4.55.21.99:5060 >> lua(/lua/freeswitch/scripts/8887129962_dialplan.lua >> f6b2ef40-db12-11e1-b609-5bed86da9fbc 8887129962 +15595158870 2012-07-31 >> 13:23:49) EXECUTE sofia/internal/+15595158870 at 4.55.21.99:5060 >> bridge({ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7,originate_continue_on_timeout=true}sofia/gateway/bandwidth.com/+919884978816,{ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7}sofia/gateway/bandwidth.com/+919677080275) >> 2012-07-31 13:23:49.252695 [DEBUG] switch_ivr_originate.c:1884 Parsing >> global variables 2012-07-31 13:23:49.252695 [DEBUG] switch_event.c:1521 >> Parsing variable [ignore_early_media]=[true] 2012-07-31 13:23:49.252695 >> [DEBUG] switch_event.c:1521 Parsing variable >> [monitor_early_media_fail]=[user_busy:1:480+620!destination_out_of_order:2:1776.7] >> 2012-07-31 13:23:49.252695 [DEBUG] switch_event.c:1521 Parsing variable >> [originate_continue_on_timeout]=[true] 2012-07-31 13:23:49.252695 [NOTICE] >> switch_channel.c:930 New Channel sofia/internal/+919884978816 >> [f6b57b20-db12-11e1-b60d-5bed86da9fbc] 2012-07-31 13:23:49.252695 [DEBUG] >> mod_sofia.c:4670 (sofia/internal/+919884978816) State Change CS_NEW -> >> CS_INIT 2012-07-31 13:23:49.252695 [DEBUG] switch_core_session.c:1180 Send >> signal sofia/internal/+919884978816 [BREAK] 2012-07-31 13:23:49.252695 [ERR] >> switch_core_session.c:427 Could not locate channel type >> {ignore_early_media=true 2012-07-31 13:23:49.252695 [NOTICE] >> switch_ivr_originate.c:2459 Cannot create outgoing channel of type >> [{ignore_early_media=true] cause: [CHAN_NOT_IMPLEMENTED] 2012-07-31 >> 13:23:49.252695 [ERR] switch_core_session.c:427 Could not locate channel >> type >> monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7}sofia >> 2012-07-31 13:23:49.252695 [NOTICE] switch_ivr_originate.c:2459 Cannot >> create outgoing channel of type >> [monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7}sofia] >> cause: [CHAN_NOT_IMPLEMENTED] i dont know what i missed!!!! Any Help or >> Suggestion would be really helpful! Thanks Ramesh >> ________________________________ >> View this message in context: Error While Dialing Two Numbers either >> serially or simultaneously >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From ramesh_mind at yahoo.com Tue Jul 31 18:18:20 2012 From: ramesh_mind at yahoo.com (ramesh) Date: Tue, 31 Jul 2012 07:18:20 -0700 (PDT) Subject: [Freeswitch-users] Error While Dialing Two Numbers either serially or simultaneously In-Reply-To: <1343743950224-7581370.post@n2.nabble.com> References: <1343742001770-7581362.post@n2.nabble.com> <1343743950224-7581370.post@n2.nabble.com> Message-ID: <1343744300292-7581371.post@n2.nabble.com> The Point i dint mentioned is, the first number in the script got executed successfully , and the problem occurs while dialing second number , and when i remove variables from the second number , then the script is working without any error. am not able to figure out what would be the reason for it. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-While-Dialing-Two-Numbers-either-serially-or-simultaneously-tp7581364p7581371.html Sent from the freeswitch-users mailing list archive at Nabble.com. From leonardo at daitangroup.com Tue Jul 31 18:26:38 2012 From: leonardo at daitangroup.com (Leonardo) Date: Tue, 31 Jul 2012 11:26:38 -0300 Subject: [Freeswitch-users] How to use TTS/flite via mod_socket In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD02304F5C38@NY1-EXMB-01.ip-soft.net> References: <5016D887.5060806@daitangroup.com> <6A6B4C284AD15042B429EB9D904544AD02304F5C38@NY1-EXMB-01.ip-soft.net> Message-ID: <5017EB1E.3090102@daitangroup.com> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/28cb5b79/attachment-0001.html From ramesh_mind at yahoo.com Tue Jul 31 18:27:42 2012 From: ramesh_mind at yahoo.com (ramesh) Date: Tue, 31 Jul 2012 07:27:42 -0700 (PDT) Subject: [Freeswitch-users] Error While Dialing Two Numbers either serially or simultaneously In-Reply-To: <1343744300292-7581371.post@n2.nabble.com> References: <1343742001770-7581362.post@n2.nabble.com> <1343743950224-7581370.post@n2.nabble.com> <1343744300292-7581371.post@n2.nabble.com> Message-ID: <1343744862639-7581373.post@n2.nabble.com> Thank you for your reply , i want to dial both users simultaneously or serially based on some dynamic values, that why i used comma seperator to make simultaneous calls. wether the following will work? "[ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7,originate_continue_on_timeout=true]sofia/gateway/bandwidth.com/+919884978816,[ignore_early_media=true,monitor_early_media_fail=user_busy:1:480+620!destination_out_of_order:2:1776.7] sofia/gateway/bandwidth.com/+919677080275" caching the variables into the [] tags?, am an newbie sorry if this question is silly -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-While-Dialing-Two-Numbers-either-serially-or-simultaneously-tp7581364p7581373.html Sent from the freeswitch-users mailing list archive at Nabble.com. From krice at freeswitch.org Tue Jul 31 18:29:33 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 31 Jul 2012 09:29:33 -0500 Subject: [Freeswitch-users] xml cdr help In-Reply-To: Message-ID: The Inbound/Outbound is from the perspective of FreeSWITCH... Keep in mind that FreeSWITCH is a b2bua that means that 1 call that goes Endpoint -> FreeSWITCH -> EndPoint is really 2 calls bridged together by freeswitch A Leg = Endpoint -> FreeSwitch (inbound on your CDR) B Leg = FreeSWITCH -> Endpoint (Outbound tagged on your CDR) You?ll notice that the UUIDs are flipped, cause each leg has a unique UUID and the ?Bridge UUID? field is the link back to the other leg(s) associated with that call On 7/31/12 8:37 AM, "Lloyd Aloysius" wrote: > Hi All: > > I implement the xml cdr. Now For a?Incoming?call -> IVR - > Dial Extension > create two records see below. Basically both records should say inbound call . > But one say as outbound. What is the reason the xml cdr module behave this > way. Also for a one?incoming?call to a destination why showing the two > records? > > > Also I notice when we make a outbound call the direction field say inbound. > But it should be outbound. Any help is appreciated. > > callerid number , destination number , uuid ,bridge uuid , date ,time > > 1416471234 ?144 ?outbound ? 7f055bf3-6622-4a47-8a86-e92c7683d74d > 898cc58c-8447-4a51-b559-3d1f2a063014 ?2012/07/30 15:37:41 00:00:12? > > 1416471234 ?144 ?inbound ? ?898cc58c-8447-4a51-b559-3d1f2a063014 > 7f055bf3-6622-4a47-8a86-e92c7683d74d ?2012/07/301 5:37:23 00:00:30? > > > Thanks > Lloyd > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120731/e13fbe0e/attachment.html From dujinfang at gmail.com Tue Jul 31 18:40:23 2012 From: dujinfang at gmail.com (Seven Du) Date: Tue, 31 Jul 2012 22:40:23 +0800 Subject: [Freeswitch-users] H264 transcoding In-Reply-To: <50179D6E.2010101@redembedded.com> References: <50179D6E.2010101@redembedded.com> Message-ID: It would be better to fix your clients ... And if you really want video transcoding, check out how voice is working now in tree. If add video transcoding support like voice, it would be a big project. Talk with core developers on the freeswitch-dev list would be a good start I think. libvlc could automatically decode any "external videos", say mp4, http, ts stream, rtp/rtsp steam etc, but I'm also new to vlc and have no idea if it can do transcoding like what you want. My first purpose when answering your question was explaining it is possible to encoding using cpu and libx264. Anything beyond that would need more research. 7. On Tuesday, July 31, 2012 at 4:55 PM, Yufei Tao wrote: > Hi Seven > > I think something that both decodes and encodes H264, would be a good starting point. And your work on top of libvlc sounds like the closest resource so far for my transcoder. Is that right? Please bear with me as I'm new to FS and all these libs and would appreciate it very much pointed at the right direction. > > Yufei > > > Subject: Re: [Freeswitch-users] H264 transcoding > From: > Seven Du (mailto:dujinfang at gmail.com) > > Date: > 30/07/12 17:17 > > > > To: > FreeSWITCH Users Help (mailto:freeswitch-users at lists.freeswitch.org) > > > > > And Yufei, my solution won't solve your problem at this time, I only did encoding and for your transcoder to work there need a lot more work?. > > > On Tuesday, July 31, 2012 at 12:06 AM, Seven Du wrote: > > Cluecon is coming and I think we could wait a few days and we still have some issues to figure out ... > > > On Monday, July 30, 2012 at 8:33 PM, Yufei Tao wrote: > > > Hi Seven > > > > Thanks very much for your replies! I thought I should be very surprised if I was the first one wanting to do this. It'd be very helpful if you could share your code with x264, or libvlc, to spare me, and possibly others of reinventing the wheels. I'll compile the code myself of course which should be free of the licensing troubles I guess. > > > > I'd be very grateful if you could share the code and give some instructions. Thank you very much! > > > > Yufei > > > > > > Subject: Re: [Freeswitch-users] H264 transcodin > > From: > > Seven Du (mailto:dujinfang at gmail.com) > > > > Date: > > 28/07/12 01:45 > > > > > > > > To: > > FreeSWITCH Users Help (mailto:freeswitch-users at lists.freeswitch.org) > > > > > > > > > > Real-time encoding with statically linked x264 lib works fine for me from QCIF to D1 resolution, 720p is slow and discarding frames on a Xeon Quad core CPU. I haven't look how to use the GPU, or if possible. > > > > It is working in my lab and I have the same question with Yufei Tao when going to production or deliver to customer. Based on http://lists.freeswitch.org/pipermail/freeswitch-dev/2010-September/004227.html , In my understanding, compile and link and use by my self should be fine and, if I deliver to a customer, it should be fine if I provide the code and help the customer to compile on their own server? I'd like to open source the code to public later, but, I'd like to know is it a MUST or MAY? > > > > If you pipe to ffmpeg or x264 command line it's not been treated as combine into a large work, and I'm not sure if realtime transcoding will be smooth. > > > > 7. > > > > On Friday, July 27, 2012 at 11:17 PM, Yufei Tao wrote: > > > > > Thanks everyone for the responses! > > > > > > If I understand it correctly, if I installed ffmpeg on itself separately > > > from FS, I could write a module for FS, in which I just call the ffmpeg > > > program by running a command line. This way would it be classified as > > > "not combine them into a larger work", thus free from license > > > incompatibility problem? > > > > > > Not sure if that'll work for real-time transcoding of x264 though? > > > > > > Thanks very much for you opinions! > > > Yufei > > > > > > -- > > > Yufei Tao > > > Red Embedded > > > > > > This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. > > > > > > You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. > > > > > > Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > Join Us At ClueCon - Aug 7-9, 2012 > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at lists.freeswitch.org) > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > > > > ForwardedMessage.eml > > Subject: > > Re: [Freeswitch-users] H264 transcoding > > > > From: > > Seven Du (mailto:dujinfang at gmail.com) > > > > Date: > > 28/07/12 02:10 > > > > > > > > To: > > FreeSWITCH Users Help (mailto:freeswitch-users at lists.freeswitch.org) > > > > > > > > > > I patched libvlc and it can now decode any video vlc supported and encode with x264 and send to any sip phone. Specifically I'm trying to use it to play a 1080p stream and resize to CIF or D1 so a video phone will accept(Sending 1080p will cause some phones to reboot :( ). > > > > Still need a lot of code to make it working neatly, however, I might can do a demo on ClueCon and @William if you'd like to review the code and merge into tree I'll happy to contribute that later. > > > > > > On Friday, July 27, 2012 at 2:43 PM, William King wrote: > > > > > libvlc is LGPL http://www.videolan.org/press/lgpl.html and there is now a mod_vlc(though it doesn't yet support video streams). The user can choose to build vlc with only the LGPL components or add the more 'adverse' modules. In none of the LGPL packages of libvlc is ffmpeg enabled, but there is a module for libvlc for ffmpeg. http://wiki.videolan.org/FFmpeg > > > > > > The only pieces now may just be the FS side of things for video. > > > William King Senior Engineer Quentus Technologies, INC 1037 NE 65th St Suite 273 Seattle, WA 98115 Main: (877) 211-9337 Office: (206) 388-4772 Cell: (253) 686-5518 william.king at quentustech.com (mailto:william.king at quentustech.com) > > > On 07/26/2012 08:46 PM, Anthony Minessale wrote: > > > > you would probably need to do something like make a mod for ffmpeg that protects you from the gpl then allow the user to build that lib on his own and choose at compile time to install patented or adverse licensed components. No license rules prohibit an end user from combining code only distributors. but even then we need a bunch of code to write. On Thu, Jul 26, 2012 at 10:41 PM, curriegrad2004 (mailto:curriegrad2004 at gmail.com) wrote: > > > > > Ken, If you think those guys over at x264 will ever change the license from GPL to LGPL, you're just dreaming the pie in the sky... In short, don't even think about it ;P On Thu, Jul 26, 2012 at 8:19 PM, Ken Rice (mailto:krice at freeswitch.org) wrote: > > > > > > we can not and will not use GPL software, the license is not compatible with the GPL and would polute the codebase with additional restrictions that are not wanted or needed. now if someone could get them to change the license or atleast give us a license under better terms such as the LGPL or the MPL then the license issue would be null Ken Sent from my iPad On Jul 26, 2012, at 7:45 PM, Terry Barnum (mailto:terry at digital-outpost.com) wrote: > > > > > > > Use x264? http://en.wikipedia.org/wiki/X264 On Jul 26, 2012, at 4:53 PM, Ken Rice wrote: > > > > > > > > Is it possible sure... Is ot probably to happen anytime soon? Not until the patents run out... On 7/26/12 5:04 PM, "yufei.tao" (mailto:yufei.tao at redembedded.com) wrote: > > > > > > > > > Hi I am trying to decide if it is feasible to let FS do transcoding between different H264 formats for live video calls. This is because I've got SIP clients that both use H264 but with different formats and one (with a bad H264 decoder) has problems decoding H264 stream from the other. But each of these two clients communicate fine using H264 with a third client that uses ffmpeg. I'm thinking of adding a module which uses ffmpeg, so that it will transcode H264 between different parameters. I've got a few questions: 1. Is this feasible? I'm not looking at supporting many simultaneous calls. 2. What is involved in transcoding real-time video stream? 3. Anyone's done anything like this before? I'm new to FS and any suggestions would be very much appreciated! Yufei > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > -- > Yufei Tao > Red Embedded > This E-mail and any attachments hereto are strictly confidential and intended solely for the addressee. If you are not the intended addressee please notify the sender by return and delete the message. > You must not disclose, forward or copy this E-mail or attachments to any third party without the prior consent of the sender. > Red Embedded Design, Company Number 06688253 Registered in England: The Waterfront, Salts Mill Rd, Saltaire, BD17 7EZ > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org (mailto:FreeSWITCH-users at 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/20120731/43e524ea/attachment-0001.html From msc at freeswitch.org Tue Jul 31 18:49:20 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 31 Jul 2012 07:49:20 -0700 Subject: [Freeswitch-users] PAGD 0 digits In-Reply-To: <5017E15C.9060707@thewinelake.com> References: <9D0E55F4-768E-4C9D-B1DF-9D074E44ADD7@mgtech.com> <458A0010-EC8C-4ACC-B08D-A02E0AD26077@mgtech.com> <500FA9BA.5030303@thewinelake.com> <50110823.2060604@thewinelake.com> <50128323.7000008@thewinelake.com> <5017E15C.9060707@thewinelake.com> Message-ID: I must confess: I do not know how to enter no digits without using the timeout... -MC On Tue, Jul 31, 2012 at 6:45 AM, Alex wrote: > Not really. But I discovered that 0 IS a valid value for max digits. > > After the timeout it simply moves on to the next application. Would that > not suffice? > -MC > > On Fri, Jul 27, 2012 at 5:01 AM, Alex wrote: > >> If I want to allow someone to enter an empty string of digits, how is it >> done? >> >> I've tried various things... >> >> My latest effort (that still doesn't work!) is >> >> keypress = session:playAndGetDigits(1, 1, 3, 5000, "*#", >> what_to_play, "", "\\d{0,1}|\\#|\\*") >> terminator = session:getVariable("read_terminator_used") >> >> I note that the minimum value for the first param is 1 - so maybe it >> can't be done? >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/3c2eecd9/attachment.html From sgaleotti at ition.com.ar Tue Jul 31 17:09:30 2012 From: sgaleotti at ition.com.ar (Sergio Galeotti (ITION)) Date: Tue, 31 Jul 2012 10:09:30 -0300 Subject: [Freeswitch-users] Skypeopen In-Reply-To: References: <501706D7.3040403@ition.com.ar> Message-ID: <5017D90A.2070503@ition.com.ar> Hi Giovanni; This installation guide is very good and was the one used for the initial setup. While I share with you that is very stable, there are some things that are not resolved even if well, if someone is pending or if any configuration issue. For example, if I use the RR method to make channels available, would need to make sets of interfaces and I just want to take some users to take calls and not others (as in Asterisk trunk groups to make and use DAHDI/g0 or DAHDI / g1 to differentiate from each other) With respect to the incoming CALLS, I read that there was a way to channel configuration, to use a single user to receive a call over and through an internal mechanism of mod_skypeopen own, subsequent calls to the first user was transferred to the first free the rest of the interfaces. With soprpresa, I found that without this setting specifies, that's the same. If you have additional information about the compratas would appreciate me for so to keep experimenting. thanks Sergio On 7/31/2012 3:33 AM, Giovanni Maruzzelli wrote: > have a look at: > http://wiki.freeswitch.org/wiki/Skypopen > > > On Tue, Jul 31, 2012 at 12:12 AM, Sergio Galeotti (ITION) > > wrote: > > Hi; > I'm new in FS user list. > There is someone who is using the Skype_Open module for incoming and > outgoing? > I would like to contact someone who is experiencing as it does not > find > much information about it and I have many doubts about the operation. > I have the platform installed and configured, but there are some > things > if they are not bugs, limitations or poor configuration on my part > thanks!! > > Sergio > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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, > > Giovanni Maruzzelli > Cell : +39-347-2665618 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120731/b564ffd9/attachment.html From ramesh_mind at yahoo.com Tue Jul 31 19:04:31 2012 From: ramesh_mind at yahoo.com (ramesh) Date: Tue, 31 Jul 2012 08:04:31 -0700 (PDT) Subject: [Freeswitch-users] Error While Dialing Two Numbers either serially or simultaneously In-Reply-To: <1343742001770-7581362.post@n2.nabble.com> References: <1343742001770-7581362.post@n2.nabble.com> Message-ID: <1343747071588-7581379.post@n2.nabble.com> Thank You so much guys!!! its working now !!!! Thanks Ramesh -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-While-Dialing-Two-Numbers-either-serially-or-simultaneously-tp7581364p7581379.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Tue Jul 31 19:14:25 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 31 Jul 2012 08:14:25 -0700 Subject: [Freeswitch-users] SIP to make phone reboot or resync In-Reply-To: <184e01cd6ec1$915f0740$b41d15c0$@bizfocused.com> References: <07b501cd6c1a$238b3e90$6aa1bbb0$@bizfocused.com> <094e01cd6c43$9e44cc90$dace65b0$@bizfocused.com> <0c4001cd6cdf$b2603a10$1720ae30$@bizfocused.com> <0deb01cd6d3f$67e43a30$37acae90$@bizfocused.com> <184e01cd6ec1$915f0740$b41d15c0$@bizfocused.com> Message-ID: On Mon, Jul 30, 2012 at 7:09 PM, Sean Devoy wrote: > Michael,**** > > ** ** > > I Googled ?freeswitch sip re-provision phone? which came up with almost > nothing. This thread is now the TOP 5 hits on that query!**** > > ** ** > > I also Googled ?freeswitch sip reboot phone? which comes up with several > links that appear to be polycom specific. In fact if I had read more I may > have gotten what I needed with some brute force trial and error.**** > > ** ** > > I must say I was expecting to find sendevent code to achieve this, not > something as simple as a SOFIA command!**** > > ** ** > > Here again, as you can see from my follow up question, I had no idea what > to search for with regard to reverse-auth-user and reverse-auth-pass > params.**** > > ** ** > > You guys rock. > Do we rock so much that you'll be coming to ClueCon 2012?! :P -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/b47cfe62/attachment.html From alex at thewinelake.com Tue Jul 31 19:16:08 2012 From: alex at thewinelake.com (Alex) Date: Tue, 31 Jul 2012 16:16:08 +0100 Subject: [Freeswitch-users] PAGD 0 digits In-Reply-To: References: <9D0E55F4-768E-4C9D-B1DF-9D074E44ADD7@mgtech.com> <458A0010-EC8C-4ACC-B08D-A02E0AD26077@mgtech.com> <500FA9BA.5030303@thewinelake.com> <50110823.2060604@thewinelake.com> <50128323.7000008@thewinelake.com> <5017E15C.9060707@thewinelake.com> Message-ID: <5017F6B8.6080801@thewinelake.com> Just press # > I must confess: I do not know how to enter no digits without using the > timeout... > -MC > > On Tue, Jul 31, 2012 at 6:45 AM, Alex > wrote: > > Not really. But I discovered that 0 IS a valid value for max digits. >> After the timeout it simply moves on to the next application. >> Would that not suffice? >> -MC >> >> On Fri, Jul 27, 2012 at 5:01 AM, Alex > > wrote: >> >> If I want to allow someone to enter an empty string of >> digits, how is it >> done? >> >> I've tried various things... >> >> My latest effort (that still doesn't work!) is >> >> keypress = session:playAndGetDigits(1, 1, 3, 5000, "*#", >> what_to_play, "", "\\d{0,1}|\\#|\\*") >> terminator = session:getVariable("read_terminator_used") >> >> I note that the minimum value for the first param is 1 - so >> maybe it >> can't be done? >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2196 / Virus Database: 2437/5166 - Release Date: 07/30/12 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/40d61633/attachment-0001.html From msc at freeswitch.org Tue Jul 31 19:15:58 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 31 Jul 2012 08:15:58 -0700 Subject: [Freeswitch-users] Bind Sofia external profile to Static private IP In-Reply-To: References: <4664115619900296997@unknownmsgid> <201207301113.33629.g.d.monnezza@tiscali.it> Message-ID: On Mon, Jul 30, 2012 at 9:36 PM, SamyGo wrote: > Hi, > Thanks for this, Yes I initially did start FS with -no-nat and it seemed > to fix it, and after your this email I plugged in "ext-*" params in the > profile and started with autoNAT and this also works. > > The autoNAT engine in Freeswitch is >> powerful. It works with uPnP, like skype does. So, working with most of >> the internet gateways (like >> home adsl routers) it is capable to open a port to communicate on the >> public network > > ^ Is definitely a very useful piece of info to be remembered. Thanks for > this :) > > Thanks, > Sammy > Thank you for following up. Now, out of curiosity, what does the "BS" in "BS Telecom" stand for? I'm trying to keep my imagination in check... :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/6f4df682/attachment.html From alex at thewinelake.com Tue Jul 31 19:19:13 2012 From: alex at thewinelake.com (Alex) Date: Tue, 31 Jul 2012 16:19:13 +0100 Subject: [Freeswitch-users] PAGD 0 digits In-Reply-To: References: <9D0E55F4-768E-4C9D-B1DF-9D074E44ADD7@mgtech.com> <458A0010-EC8C-4ACC-B08D-A02E0AD26077@mgtech.com> <500FA9BA.5030303@thewinelake.com> <50110823.2060604@thewinelake.com> <50128323.7000008@thewinelake.com> <5017E15C.9060707@thewinelake.com> Message-ID: <5017F771.2010305@thewinelake.com> ...or is it a minimum? What I wanted was to ask the user to enter a number terminated with # - or press * to cancel. I was having the problem that if I had * as a terminator and 1 as a minimum length, the users kept getting asked to enter again. This problem has been solved, although possibly not in the most elegant way! > > In that case 0 means no maximum. > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > > On Jul 31, 2012 9:45 AM, "Alex" > wrote: > > Not really. But I discovered that 0 IS a valid value for max digits. >> After the timeout it simply moves on to the next application. >> Would that not suffice? >> -MC >> >> On Fri, Jul 27, 2012 at 5:01 AM, Alex > > wrote: >> >> If I want to allow someone to enter an empty string of >> digits, how is it >> done? >> >> I've tried various things... >> >> My latest effort (that still doesn't work!) is >> >> keypress = session:playAndGetDigits(1, 1, 3, 5000, "*#", >> what_to_play, "", "\\d{0,1}|\\#|\\*") >> terminator = session:getVariable("read_terminator_used") >> >> I note that the minimum value for the first param is 1 - so >> maybe it >> can't be done? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2196 / Virus Database: 2437/5166 - Release Date: >> 07/30/12 >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2196 / Virus Database: 2437/5166 - Release Date: 07/30/12 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/00078812/attachment.html From msc at freeswitch.org Tue Jul 31 19:22:37 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 31 Jul 2012 08:22:37 -0700 Subject: [Freeswitch-users] Lua problem there are only two arguments evaluated In-Reply-To: References: <50178DB7.7000606@bestnet.kharkov.ua> Message-ID: I believe you need to run this command "inline": Change it to this and try again: Let us know if it works or not. Also, your homework assignment will be to read this: http://wiki.freeswitch.org/wiki/Dialplan_XML#Inline_Actions And learn about hunting (or "parsing" phase) vs. execution phase of dialplan processing. :) -MC On Tue, Jul 31, 2012 at 2:12 AM, Thomas Hoellriegel wrote: > hi guy, > I have simply written a voicemailrecording in lua. > My dialplan extension is: > > expression="^44[0-9][0-9][0-9]**[0-9]\d*$"> > > > > > > The luascript: > > -- simply mailbox recording > -- parameters: > -- mailbox, sender, message > mailbox = argv[1]; > from_box = argv[2]; > msg_file = argv[3]; > snd = "/usr/local/freeswitch/sounds/**dorf"; > file, errMsg = io.open( mailbox, "r" ); > if not file then > session:streamFile("/usr/**local/freeswitch/sounds/dorf/** > mailbox_not.alaw") > session:execute("transfer","**dorf_now XML dorf_now"); > else > freeswitch.consoleLog("info", "msg_file is: " .. msg_file .. "\n"); > session:execute("playback",**mailbox); > session:streamFile("/usr/**local/freeswitch/sounds/ivr/**beep.alaw") > session:execute("record","/**tmp/" .. msg_file .. " 180 200 3 "); > session:execute("transfer","**dorf_send XML dorf_send"); > end > > I become the error: > [ERR] mod_lua.cpp:198 /usr/local/freeswitch/scripts/**recordmsg.lua:13 > : attempt to concatenate global 'msg_file' (a nil value)M > > > I have 3 arguments, only 2 arguments will be works. > Can your help please? > thanks. > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/186e6dc0/attachment-0001.html From bdfoster at endigotech.com Tue Jul 31 19:36:11 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Tue, 31 Jul 2012 11:36:11 -0400 Subject: [Freeswitch-users] PAGD 0 digits In-Reply-To: <5017F771.2010305@thewinelake.com> References: <9D0E55F4-768E-4C9D-B1DF-9D074E44ADD7@mgtech.com> <458A0010-EC8C-4ACC-B08D-A02E0AD26077@mgtech.com> <500FA9BA.5030303@thewinelake.com> <50110823.2060604@thewinelake.com> <50128323.7000008@thewinelake.com> <5017E15C.9060707@thewinelake.com> <5017F771.2010305@thewinelake.com> Message-ID: The elegant solution would probably require a LUA script. Brian Foster Endigo Computer LLC Sent from a mobile device. On Jul 31, 2012 11:20 AM, "Alex" wrote: > ...or is it a minimum? > What I wanted was to ask the user to enter a number terminated with # - or > press * to cancel. > I was having the problem that if I had * as a terminator and 1 as a > minimum length, the users kept getting asked to enter again. > This problem has been solved, although possibly not in the most elegant > way! > > In that case 0 means no maximum. > > Brian Foster > Endigo Computer LLC > > Sent from a mobile device. > On Jul 31, 2012 9:45 AM, "Alex" wrote: > >> Not really. But I discovered that 0 IS a valid value for max digits. >> >> After the timeout it simply moves on to the next application. Would that >> not suffice? >> -MC >> >> On Fri, Jul 27, 2012 at 5:01 AM, Alex wrote: >> >>> If I want to allow someone to enter an empty string of digits, how is it >>> done? >>> >>> I've tried various things... >>> >>> My latest effort (that still doesn't work!) is >>> >>> keypress = session:playAndGetDigits(1, 1, 3, 5000, "*#", >>> what_to_play, "", "\\d{0,1}|\\#|\\*") >>> terminator = session:getVariable("read_terminator_used") >>> >>> I note that the minimum value for the first param is 1 - so maybe it >>> can't be done? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> Join Us At ClueCon - Aug 7-9, 2012 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> -- >> Michael S Collins >> Twitter: @mercutioviz >> http://www.FreeSWITCH.org >> http://www.ClueCon.com >> http://www.OSTAG.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com >> >> >> >> Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org >> >> >> >> No virus found in this message. >> Checked by AVG - www.avg.com >> Version: 2012.0.2196 / Virus Database: 2437/5166 - Release Date: 07/30/12 >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing listFreeSWITCH-users at lists.freeswitch.orghttp://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-usershttp://www.freeswitch.org > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.2196 / Virus Database: 2437/5166 - Release Date: 07/30/12 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120731/699e26d6/attachment.html From richgration at gmail.com Tue Jul 31 19:34:56 2012 From: richgration at gmail.com (Richard Gration) Date: Tue, 31 Jul 2012 16:34:56 +0100 Subject: [Freeswitch-users] Using application="limit" with custom variable Message-ID: Hi all, I asked this in #freeswitch earlier, but didn't manage to find a resolution there, so I'm posting here in the hope my question gets a wider audience. (The dialplan I'm using is fetched with mod_xml_curl. An example of a call is here: http://pastebin.freeswitch.org/19616 ) I'm trying to limit concurrent calls on a per user basis. The problem I'm facing is that using the limit settings I have, freeswitch only counts connected calls, not calls in the setup phase (trying, ringing etc) towards the limit. So, in the example dialplan above, the account 99999 is being limited to one concurrent call. This works if there is a call up for that account, but in my tests I could make 2 or more calls, provided none of them was actually connected. I would like to know if there is a way to have freeswitch consider all connections for this account to count towards the limit total. If this *can* be done, then I expect that it is a setting in mod_dptools or perhaps fs core, rather than something I can return in my dialplan. Cheers, Rich -- Once our basic material needs are met - in my utopia, anyway - life becomes a perpetual celebration in which everyone has a talent to contribute. But we cannot levitate ourselves into that blessed condition by wishing it. We need to brace ourselves for a struggle against terrifying obstacles, both of our own making and imposed by the natural world. And the first step is to recover from the delusion that is positive thinking. -- Barbara Ehrenreich From philq at qsystemsengineering.com Tue Jul 31 20:19:36 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Tue, 31 Jul 2012 12:19:36 -0400 Subject: [Freeswitch-users] Setting effecting_caller_id_name Message-ID: <039901cd6f38$4a624ba0$df26e2e0$@com> Ok, the problem here is that the variable caller_id_name contains the Caller ID number instead of the CNAM that was looked up. Is there a variable to look at and change for the CNAM info? Info shows the CNAM info in Caller-Caller-ID-Name but attempts to match on variations of that have failed. Obviously I'm missing some basic piece of info here but I haven't been able to find it, even within the FreeSwitch book. I've pasted a small section of the relevant console output below. I should also mention that I'm doing this check within public.xml since I want it to apply to all incoming calls. Thanks, - Phil 2012-07-31 11:33:30.800571 [INFO] mod_dialplan_xml.c:485 Processing 4435551212 <4435551212>->4105551212 in context public Dialplan: sofia/external/4435551212 at 140.239.xx.x parsing [public->outside_call] continue=true Dialplan: sofia/external/4435551212 at 140.239.xx.x Absolute Condition [outside_call] Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (PASS) [outside_call] ${module_exists(mod_cidlookup)}(true) =~ /true/ break=on-false Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (PASS) [outside_call] caller_id_name(4435551212) =~ /^4435551212$|^$/ break=on-false Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (PASS) [outside_call] caller_id_number(4435551212) =~ /^1?([2-9]\d\d[2-9]\d{6})$/ break=on-false Dialplan: sofia/external/4435551212 at 140.239.xx.x Action cidlookup(4435551212) Dialplan: sofia/external/4435551212 at 140.239.xx.x parsing [public->fix_cidnam_plus] continue=true Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (FAIL) [fix_cidnam_plus] caller_id_name(4435551212) =~ /^\+1?([2-9]\d\d[2-9]\d{6})$/ break=on-false Dialplan: sofia/external/4435551212 at 140.239.xx.x parsing [public->currently_running] continue=true Dialplan: sofia/external/4435551212 at 140.239.xx.x Absolute Condition [currently_running] Dialplan: sofia/external/4435551212 at 140.239.xx.x Action info() Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (FAIL) [currently_running] caller_id_name(4435551212) =~ /^Currently running a lookup/ break=on-false Caller-Direction: [inbound] Caller-Username: [4435551212] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [SMITH,JOHN] Caller-Caller-ID-Number: [4435551212] Caller-Network-Addr: [140.239.xx.x] Caller-ANI: [4435551212] Caller-Destination-Number: [4105551212] _____________________________________________ From: Phil Quesinberry Sent: Thursday, July 26, 2012 10:29 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: RE: Setting effecting_caller_id_name The reply is different each time, depending upon the number being looked up. So, I just want to look at the first part of the string. If FS can't do a regex match without the trailing $, I'm guessing there's a way to just do it in XML. I'll try and see what I can find after the storm passes unless you have a better idea, I need to shut this computer down right now. Thanks, - Phil You can just not use a regex. Do you need to escape the spaces? Brian Foster Endigo Computer LLC _____________________________________________ From: Phil Quesinberry Sent: Thursday, July 26, 2012 5:46 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: RE: Setting effecting_caller_id_name If you put the $ at the end then it will try to match the entire string instead of just the beginning of it, which won't work in this case. Is there a way to match just the beginning of the string in FS? Thanks, - Phil You need a $ after 'lookup' for it to be a regex. Brian Foster Endigo Computer LLC _____________________________________________ From: Phil Quesinberry Sent: Thursday, July 26, 2012 3:59 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: Setting effecting_caller_id_name And while I'm asking dumb questions. When doing CNAM dips from opencnam.com, often you get a result of "Currently running a lookup for phone 'xxxxxxxxxx'. on incoming calls, typically for wireless or other unknown name callers and I wanted to change that to "Wireless/Unknown" Since caller_id_name is apparently read-only, I am attempting to set effective_caller_id_name. I put the following in public.xml right below the "fix_cidnam_plus" entry, in other words after a CNAM lookup has been performed. If I crafted my regex properly, then it should be matching on the first part of the string and setting the variable appropriately. Is 'effective_caller_id_name' the variable I should be setting? Many thanks, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/760dd4e8/attachment-0001.html From msc at freeswitch.org Tue Jul 31 20:44:00 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 31 Jul 2012 09:44:00 -0700 Subject: [Freeswitch-users] Setting effecting_caller_id_name In-Reply-To: <039901cd6f38$4a624ba0$df26e2e0$@com> References: <039901cd6f38$4a624ba0$df26e2e0$@com> Message-ID: Phil, I have a question about this line from your trace: Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (FAIL) [fix_cidnam_plus] caller_id_name(4435551212) =~ /^\+1?([2-9]\d\d[2-9]\d{6})$/ break=on-false Are you thinking that "4435551212" should match the regex in extension "fix_cidnam_plus"? The reason I ask is that I think your regex may have a small gotcha. It looks to my like you are making the leading "1" optional by having "1?" in front of the rest of the regex. However, it looks to me like the "+" is not optional, so unless the phone number is "+4435551212" it won't match that regex. If the + is meant to be optional then the easy way to test would be to add a "?" right after the "+": Of course, if you were banking on having the + in there then I'm all wet and you can ignore me... -MC On Tue, Jul 31, 2012 at 9:19 AM, Phil Quesinberry < philq at qsystemsengineering.com> wrote: > ** > > Ok, the problem here is that the variable caller_id_name contains the > Caller ID number instead of the CNAM that was looked up. Is there avariable to look at and change for > the CNAM info? Info shows the CNAM info in Caller-Caller-ID-Name but > attempts to match on variations of that have failed. Obviously I?m > missing some basic piece of info here but I haven?t been able to find it, > even within the FreeSwitch book. > > I?ve pasted a small section of the relevant console output below. > > I should also mention that I?m doing this check within public.xml since I > want it to apply to all incoming calls. > > Thanks, > > - Phil > > 2012-07-31 11:33:30.800571 [INFO] mod_dialplan_xml.c:485 Processing > 4435551212 <4435551212>->4105551212 in context public > > Dialplan: sofia/external/4435551212 at 140.239.xx.x parsing > [public->outside_call] continue=true > > Dialplan: sofia/external/4435551212 at 140.239.xx.x Absolute Condition > [outside_call] > > Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (PASS) > [outside_call] ${module_exists(mod_cidlookup)}(true) =~ /true/ > break=on-false > > Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (PASS) > [outside_call] caller_id_name(4435551212) =~ /^4435551212$|^$/ > break=on-false > > Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (PASS) > [outside_call] caller_id_number(4435551212) =~ > /^1?([2-9]\d\d[2-9]\d{6})$/ break=on-false > > Dialplan: sofia/external/4435551212 at 140.239.xx.x Action cidlookup(4435 > 551212) > > Dialplan: sofia/external/4435551212 at 140.239.xx.x parsing > [public->fix_cidnam_plus] continue=true > > Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (FAIL) > [fix_cidnam_plus] caller_id_name(4435551212) =~ > /^\+1?([2-9]\d\d[2-9]\d{6})$/ break=on-false > > Dialplan: sofia/external/4435551212 at 140.239.xx.x parsing > [public->currently_running] continue=true > > Dialplan: sofia/external/4435551212 at 140.239.xx.x Absolute Condition > [currently_running] > > Dialplan: sofia/external/4435551212 at 140.239.xx.x Action info() > > Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (FAIL) > [currently_running] caller_id_name(4435551212) =~ /^Currently running a > lookup/ break=on-false > > Caller-Direction: [inbound] > > Caller-Username: [4435551212] > > Caller-Dialplan: [XML] > > Caller-Caller-ID-Name: [SMITH,JOHN] > > Caller-Caller-ID-Number: [4435551212] > > Caller-Network-Addr: [140.239.xx.x] > > Caller-ANI: [4435551212] > > Caller-Destination-Number: [4105551212] > > _____________________________________________ > *****From:* Phil Quesinberry > *****Sent:* Thursday, July 26, 2012 10:29 PM > *****To:* 'freeswitch-users at lists.freeswitch.org' > *****Subject:* RE: Setting effecting_caller_id_name > > The reply is different each time, depending upon the number being looked > up. So, I just want to look at the first part of the string. If FS can?t > do a regex match without the trailing $, I?m guessing there?s a way to just > do it in XML. > > I?ll try and see what I can find after the storm passes unless you have a > better idea, I need to shut this computer down right now. > > Thanks, > > - Phil > > You can just not use a regex. > > Do you need to escape the spaces? > > Brian Foster > > Endigo Computer LLC > > _____________________________________________ > *****From:* Phil Quesinberry > > *****Sent:* Thursday, July 26, 2012 5:46 PM > *****To:* 'freeswitch-users at lists.freeswitch.org' > *****Subject:* RE: Setting effecting_caller_id_name > > If you put the $ at the end then it will try to match the entire string > instead of just the beginning of it, which won?t work in this case. Is > there a way to match just the beginning of the string in FS? > > Thanks, > > - Phil > > You need a $ after 'lookup' for it to be a regex. > > Brian Foster > > Endigo Computer LLC > > _____________________________________________ > *****From:* Phil Quesinberry > *****Sent:* Thursday, July 26, 2012 3:59 PM > *****To:* 'freeswitch-users at lists.freeswitch.org' > *****Subject:* Setting effecting_caller_id_name > > And while I?m asking dumb questions? > > When doing CNAM dips from opencnam.com, often you get a result of > ?Currently running a lookup for phone ?xxxxxxxxxx?? on incoming calls, > typically for wireless or other unknown name callers and I wanted to change > that to ?Wireless/Unknown? Since caller_id_name is apparently read-only, I > am attempting to set effective_caller_id_name. I put the following in > public.xml right below the ?fix_cidnam_plus? entry, in other words after a > CNAM lookup has been performed. > > > > > > data="effective_caller_id_name=Wireless/Unknown"/> > > > > > > If I crafted my regex properly, then it should be matching on the first > part of the string and setting the variable appropriately. Is > ?effective_caller_id_name? the variable I should be setting? > > Many thanks, > > *******Phil Quesinberry* > > Q Systems Engineering, Inc. > > Electronic Controls and Embedded Systems Development > > (410) 969-8002 > > ***http://www.qsystemsengineering.com* > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Michael S Collins Twitter: @mercutioviz http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/4d8edd60/attachment-0001.html From govoiper at gmail.com Tue Jul 31 20:47:35 2012 From: govoiper at gmail.com (SamyGo) Date: Tue, 31 Jul 2012 21:47:35 +0500 Subject: [Freeswitch-users] LUA session:Bridge not actually bridging calls ~ In-Reply-To: References: Message-ID: Thats understandable. Im cool with it, afterall its working. :) On Jul 31, 2012 9:40 AM, "SamyGo" wrote: > Though the problem is solved but is there any corresponding WARN or > anything in the logs so that anyone can pick the problem right away. > This fixing of codecs took alot of time and I was never expecting that > this could be the cause Thinking that its Bridge() which is not doing its > part. > > Thanks a lot for your time on this. > > Regards, > Sammy > > > On Mon, Jul 30, 2012 at 6:22 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> This script is actually a bit over-complicated >> >> dialA = "sofia/gateway/fs1/9903" >> dialB = "user/1001" >> legA = freeswitch.Session(dialA) >> >> if (legA:ready()) { >> legA:execute("bridge", dialB) >> } >> >> I think the problem is that we don't support dealing with many codecs >> in the 200ok >> They send gsm and ulaw and then choose ulaw and we go with the first one >> gsm >> >> >> >> On Sat, Jul 28, 2012 at 11:23 AM, SamyGo wrote: >> > Hi again, >> > >> > So, It was a very minor change in configuration and it was working. >> > Basically FreeSwicth was bridging the two legs BUT there was a codec >> issue. >> > All I had to do was in Asterisk (serving as my gateway) to allow only >> ulaw >> > and alaw >> > i.e >> > >> > disallow=all >> > allow=ulaw >> > allow=alaw >> > >> > What I really really wish to know is that why there was no indication of >> > codec mismatch or ptime mismatch or sample rate mismatch while >> transcoding >> > or anything. >> > >> > It will be fine if none replies but it will be great to know the real >> > reason behind this and from where in logs can I verify this !! >> > >> > Thanks >> > Sammy >> > >> > >> > On Sat, Jul 28, 2012 at 8:18 PM, SamyGo wrote: >> >> >> >> Here are the FS console logs: >> >> http://pastebin.freeswitch.org/19595 >> >> >> >> Please suggest what am I missing here. >> >> >> >> >> >> On Sat, Jul 28, 2012 at 7:59 PM, SamyGo wrote: >> >>> >> >>> Hello, >> >>> I wanted to make a lua script which just dials out two different >> numbers >> >>> via some external gateway and when both calls are answered they are >> just >> >>> bridged. For this a very impressive Lua example >> >>> http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control is >> copied and >> >>> all I had to do was change the dialA and dialB strings and its >> working great >> >>> as far as the SIP signalling is concerned. >> >>> >> >>> execute this string and I get calls on two different number but things >> >>> get interesting when Freeswitch bridge() the two legs. No AUDIO..not >> even >> >>> one-way. I could see on my own gateway that RTPs for both the legs >> are >> >>> actually forwarded to Freeswitch ! >> >>> >> >>> On my sip pcap traces analyzing on wireshark I could actually hear the >> >>> two persons saying Hello but neither could hear anything. >> >>> >> >>> The above example lua call_control script is used as it is. >> >>> Please suggest. >> >>> >> >>> Regards >> >>> Sammy Go. >> >>> >> >> >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > Join Us At ClueCon - Aug 7-9, 2012 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Anthony Minessale II >> >> FreeSWITCH http://www.freeswitch.org/ >> ClueCon http://www.cluecon.com/ >> Twitter: http://twitter.com/FreeSWITCH_wire >> >> AIM: anthm >> MSN:anthony_minessale at hotmail.com >> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com >> IRC: irc.freenode.net #freeswitch >> >> FreeSWITCH Developer Conference >> sip:888 at conference.freeswitch.org >> googletalk:conf+888 at conference.freeswitch.org >> pstn:+19193869900 >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> Join Us At ClueCon - Aug 7-9, 2012 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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/20120731/6bb0e794/attachment.html From david.villasmil.work at gmail.com Tue Jul 31 20:50:33 2012 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 31 Jul 2012 18:50:33 +0200 Subject: [Freeswitch-users] ESL problem In-Reply-To: References: <7aa29e790907272316m43671f92jf9f5442f6c7caf22@mail.gmail.com> <87f2f3b90907272353n142f268atb804509038147f0b@mail.gmail.com> Message-ID: Hello Bringing up a very old post, when making perlmod-install I get: install -m 755 ESL.so /usr/local/share/perl/5.14.2 install: cannot create regular file `/usr/local/share/perl/5.14.2': No such file or directory make: *** [install] Error 1 I assume that's because make is trying to install to "/usr/local/share/perl/5.14.2" which doesn't exists in my wheezy/sid, but should be installed in "/usr/share/perl/5.14.2" How can I change the destination directory? Thanks David On Tue, Jul 28, 2009 at 10:26 AM, Brian West wrote: > Don't forget you need to install libs/esl/perl/ESL.so and libs/esl/ > perl/ESL.pm into your system perl library path. > > /b > > On Jul 28, 2009, at 1:53 AM, Michael Collins wrote: > > > Make certain that you've built both libesl and the Perl mod. Change > > directory into /usr/src/freeswitch.trunk/libs/esl (or whatever your > > path is to libs/esl) and do these commands: > > make > > make perlmod > > > > Then give it another shot. > > -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/20120731/e81e9165/attachment.html From paul at cupis.co.uk Tue Jul 31 22:24:50 2012 From: paul at cupis.co.uk (Paul Cupis) Date: Tue, 31 Jul 2012 19:24:50 +0100 Subject: [Freeswitch-users] ESL problem In-Reply-To: References: <7aa29e790907272316m43671f92jf9f5442f6c7caf22@mail.gmail.com> <87f2f3b90907272353n142f268atb804509038147f0b@mail.gmail.com> Message-ID: <501822F2.8040603@cupis.co.uk> On 31/07/12 17:50, David Villasmil wrote: > Bringing up a very old post, when making perlmod-install I get: > > install -m 755 ESL.so /usr/local/share/perl/5.14.2 > install: cannot create regular file `/usr/local/share/perl/5.14.2': No > such file or directory > make: *** [install] Error 1 > > I assume that's because make is trying to install to > "/usr/local/share/perl/5.14.2" which doesn't exists in my wheezy/sid, > but should be installed in "/usr/share/perl/5.14.2" > > How can I change the destination directory? It should probably be installed in /usr/local, just run: mkdir -p /usr/local/share/perl/5.14.2 and then re-run your `make perlmod-install`. Regards, From david.villasmil.work at gmail.com Tue Jul 31 22:35:46 2012 From: david.villasmil.work at gmail.com (David Villasmil) Date: Tue, 31 Jul 2012 20:35:46 +0200 Subject: [Freeswitch-users] ESL problem In-Reply-To: <501822F2.8040603@cupis.co.uk> References: <7aa29e790907272316m43671f92jf9f5442f6c7caf22@mail.gmail.com> <87f2f3b90907272353n142f268atb804509038147f0b@mail.gmail.com> <501822F2.8040603@cupis.co.uk> Message-ID: Hello Paul, Yes, i thought of doing that of course, but I'm not sure I want to install it in a non-standard path (for my setup)... I'll do that anyway. Thanks David On Tue, Jul 31, 2012 at 8:24 PM, Paul Cupis wrote: > On 31/07/12 17:50, David Villasmil wrote: > > Bringing up a very old post, when making perlmod-install I get: > > > > install -m 755 ESL.so /usr/local/share/perl/5.14.2 > > install: cannot create regular file `/usr/local/share/perl/5.14.2': No > > such file or directory > > make: *** [install] Error 1 > > > > I assume that's because make is trying to install to > > "/usr/local/share/perl/5.14.2" which doesn't exists in my wheezy/sid, > > but should be installed in "/usr/share/perl/5.14.2" > > > > How can I change the destination directory? > > It should probably be installed in /usr/local, just run: > > mkdir -p /usr/local/share/perl/5.14.2 > > and then re-run your `make perlmod-install`. > > Regards, > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > Join Us At ClueCon - Aug 7-9, 2012 > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20120731/86f079a8/attachment-0001.html From philq at qsystemsengineering.com Tue Jul 31 22:56:33 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Tue, 31 Jul 2012 14:56:33 -0400 Subject: [Freeswitch-users] Setting effecting_caller_id_name Message-ID: <045001cd6f4e$36cb7c90$a46275b0$@com> Hi Michael, Good eye. I actually edited the fix_cidnam_plus entry a while back when it wasn't working correctly for a DID that sends the '+' at the beginning. The edited expression is below the original one in public.xml which is commented out. Now I don't remember if the original expression was in there by default or if I pasted it in from somewhere due to my lack of experience with FreeSWITCH at the time, but the original regex wasn't quite right since OpenCNAM does not accept the leading '1'. I was only worried about numbers with the leading '+' since the other case is handled by the previous entry. Your fix is certainly more elegant. When I start to feel like I'm becoming reasonably proficient with regexes, I subsequently wind up not doing anything with them for a long time and then almost forget what little I've learned. :) I've been learning a lot about Linux and FreeSWITCH at the same time in the past months, it's a lot of information for my aging mind to take in all at once. I really wish I could have done this 20 years ago! Cheers, - Phil Michael Collins Tue Jul 31 20:44:00 MSD 2012 Phil, I have a question about this line from your trace: Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (FAIL) [fix_cidnam_plus] caller_id_name(4435551212) =~ /^\+1?([2-9]\d\d[2-9]\d{6})$/ break=on-false Are you thinking that "4435551212" should match the regex in extension "fix_cidnam_plus"? The reason I ask is that I think your regex may have a small gotcha. It looks to my like you are making the leading "1" optional by having "1?" in front of the rest of the regex. However, it looks to me like the "+" is not optional, so unless the phone number is "+4435551212" it won't match that regex. If the + is meant to be optional then the easy way to test would be to add a "?" right after the "+": Of course, if you were banking on having the + in there then I'm all wet and you can ignore me... -MC _____________________________________________ From: Phil Quesinberry Sent: Tuesday, July 31, 2012 12:20 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: RE: Setting effecting_caller_id_name Ok, the problem here is that the variable caller_id_name contains the Caller ID number instead of the CNAM that was looked up. Is there a variable to look at and change for the CNAM info? Info shows the CNAM info in Caller-Caller-ID-Name but attempts to match on variations of that have failed. Obviously I'm missing some basic piece of info here but I haven't been able to find it, even within the FreeSwitch book. I've pasted a small section of the relevant console output below. I should also mention that I'm doing this check within public.xml since I want it to apply to all incoming calls. Thanks, - Phil 2012-07-31 11:33:30.800571 [INFO] mod_dialplan_xml.c:485 Processing 4435551212 <4435551212>->4105551212 in context public Dialplan: sofia/external/4435551212 at 140.239.xx.x parsing [public->outside_call] continue=true Dialplan: sofia/external/4435551212 at 140.239.xx.x Absolute Condition [outside_call] Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (PASS) [outside_call] ${module_exists(mod_cidlookup)}(true) =~ /true/ break=on-false Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (PASS) [outside_call] caller_id_name(4435551212) =~ /^4435551212$|^$/ break=on-false Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (PASS) [outside_call] caller_id_number(4435551212) =~ /^1?([2-9]\d\d[2-9]\d{6})$/ break=on-false Dialplan: sofia/external/4435551212 at 140.239.xx.x Action cidlookup(4435551212) Dialplan: sofia/external/4435551212 at 140.239.xx.x parsing [public->fix_cidnam_plus] continue=true Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (FAIL) [fix_cidnam_plus] caller_id_name(4435551212) =~ /^\+1?([2-9]\d\d[2-9]\d{6})$/ break=on-false Dialplan: sofia/external/4435551212 at 140.239.xx.x parsing [public->currently_running] continue=true Dialplan: sofia/external/4435551212 at 140.239.xx.x Absolute Condition [currently_running] Dialplan: sofia/external/4435551212 at 140.239.xx.x Action info() Dialplan: sofia/external/4435551212 at 140.239.xx.x Regex (FAIL) [currently_running] caller_id_name(4435551212) =~ /^Currently running a lookup/ break=on-false Caller-Direction: [inbound] Caller-Username: [4435551212] Caller-Dialplan: [XML] Caller-Caller-ID-Name: [SMITH,JOHN] Caller-Caller-ID-Number: [4435551212] Caller-Network-Addr: [140.239.xx.x] Caller-ANI: [4435551212] Caller-Destination-Number: [4105551212] _____________________________________________ From: Phil Quesinberry Sent: Thursday, July 26, 2012 10:29 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: RE: Setting effecting_caller_id_name The reply is different each time, depending upon the number being looked up. So, I just want to look at the first part of the string. If FS can't do a regex match without the trailing $, I'm guessing there's a way to just do it in XML. I'll try and see what I can find after the storm passes unless you have a better idea, I need to shut this computer down right now. Thanks, - Phil You can just not use a regex. Do you need to escape the spaces? Brian Foster Endigo Computer LLC _____________________________________________ From: Phil Quesinberry Sent: Thursday, July 26, 2012 5:46 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: RE: Setting effecting_caller_id_name If you put the $ at the end then it will try to match the entire string instead of just the beginning of it, which won't work in this case. Is there a way to match just the beginning of the string in FS? Thanks, - Phil You need a $ after 'lookup' for it to be a regex. Brian Foster Endigo Computer LLC _____________________________________________ From: Phil Quesinberry Sent: Thursday, July 26, 2012 3:59 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: Setting effecting_caller_id_name And while I'm asking dumb questions. When doing CNAM dips from opencnam.com, often you get a result of "Currently running a lookup for phone 'xxxxxxxxxx'. on incoming calls, typically for wireless or other unknown name callers and I wanted to change that to "Wireless/Unknown" Since caller_id_name is apparently read-only, I am attempting to set effective_caller_id_name. I put the following in public.xml right below the "fix_cidnam_plus" entry, in other words after a CNAM lookup has been performed. If I crafted my regex properly, then it should be matching on the first part of the string and setting the variable appropriately. Is 'effective_caller_id_name' the variable I should be setting? Many thanks, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120731/e2cb5f15/attachment-0001.html From darcyp at voice2net.ca Fri Jul 27 01:08:57 2012 From: darcyp at voice2net.ca (Darcy Primrose) Date: Thu, 26 Jul 2012 17:08:57 -0400 Subject: [Freeswitch-users] mwi Message-ID: <812FE464086D4006A397E0C8623B7C2C@owner397fa27d2> Is there a means to turn the mwi indicators on/off from fs_cli Darcy Primrose -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120726/a1dbf817/attachment.html