From msc at freeswitch.org Wed Feb 1 00:08:12 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 31 Jan 2012 13:08:12 -0800 Subject: [Freeswitch-users] exit lua script in hangup hook In-Reply-To: <1328040393457-7241003.post@n2.nabble.com> References: <1328040393457-7241003.post@n2.nabble.com> Message-ID: That's what I get for being 2 months behind on the changelog! :D If we have any Lua peeps out there who could test this and update the wiki I would be eternally grateful... -MC On Tue, Jan 31, 2012 at 12:06 PM, Jeff Lenk wrote: > Just to let you know mercutioviz that anthm did make some source code > changes > last week that should have enabled this to work. > > Commit:09ad887948f7513725ca8b53bdfe721d9008e73b > > * FS-3841 --resolve ok return the string "die" or "exit" from hanguphook to > cause an error or call s:destroy("any err message"); either should now halt > the script > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/exit-lua-script-in-hangup-hook-tp7223506p7241003.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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/fbb914d2/attachment-0001.html From msc at freeswitch.org Wed Feb 1 00:09:52 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 31 Jan 2012 13:09:52 -0800 Subject: [Freeswitch-users] Conference Auto Outcall to External Numbers In-Reply-To: <44E5C0A9D48A3246966A4AE04692014D132E145D@CH1PRD0610MB355.namprd06.prod.outlook.com> References: <44E5C0A9D48A3246966A4AE04692014D132E145D@CH1PRD0610MB355.namprd06.prod.outlook.com> Message-ID: What actually happens when you set the call forwarding? Console debug log w/ SIP trace would be helpful, as would any relevant configs you've got in place. -MC On Tue, Jan 31, 2012 at 12:46 PM, Ryan Watkins wrote: > Morning/Afternoon/Evening all? **** > > ** ** > > I?ve been trying to setup a conference call that would use the auto > outcall function, and it would call external numbers (like their cell). > I?ve tried to setup extensions for the individuals, and set the call > forwarding to their external number. Although the auto outcall function > will successfully call the extensions when they are set to an internal SIP > phone, when call forwarding is enabled the conference call will not call > the external number. Any ideas on how to accomplish this?**** > > ** ** > > Thanks a lot!**** > > ** ** > > *Ryan ***** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/7dde4dad/attachment.html From brett at davidson.gen.nz Wed Feb 1 00:33:33 2012 From: brett at davidson.gen.nz (Brett Davidson) Date: Wed, 1 Feb 2012 10:33:33 +1300 (NZDT) Subject: [Freeswitch-users] Newbie question. Where to install Freeswitch? Message-ID: <20120201103333.BOL00715@msgsrv.mail.isx.net.nz> I am about to build myself a new PfSense box at home as I want Gigabit wirespeed routing between two subnets and then found FreeSwitch. It appears you can install FreeSwitch on pfsense and that would make sense from a consolidation and power usage factor but it doesn't seem advisable to put your Telephony app on your firewall. That a real concern or am I being a little too paranoid (I work in IT). ;-) I understand that FreeSwitch could run on an Alix 2D3 (my current PfSense machine) but am wondering if it's a little underpowered. (AMD Geode at 500MHz with 256Mb RAM). I don't see much documented about resources and where to place FreeSwitch. Any recommendations? Cheers in advance, Brett. From mike at jerris.com Wed Feb 1 01:20:03 2012 From: mike at jerris.com (Michael Jerris) Date: Tue, 31 Jan 2012 17:20:03 -0500 Subject: [Freeswitch-users] valet parking with 2 FS servers In-Reply-To: References: Message-ID: <577F9061-361B-4DE5-B498-4CB178608FA0@jerris.com> There is no way to do that currently. You will need to direct all parking to one box currently. Mike On Jan 31, 2012, at 3:26 PM, Vik Killa wrote: > Hi, > Is there any way to share the valet parking lots across multiple FS > servers? I have a setup where opensips distributes calls to two FS > servers. When a call is parked and I run: > CLI>valet_info > The parked call only appears on one FS box. So if the call pickup goes > to the opposite FS server, the call will not be picked up. > Is there anyway to make both FS boxes aware of valet_info? Similar how > to "sofia status profile internal reg" is aware using ODBC From Ryan at ocens.com Wed Feb 1 01:28:01 2012 From: Ryan at ocens.com (Ryan Watkins) Date: Tue, 31 Jan 2012 22:28:01 +0000 Subject: [Freeswitch-users] Conference Auto Outcall to External Numbers In-Reply-To: References: <44E5C0A9D48A3246966A4AE04692014D132E145D@CH1PRD0610MB355.namprd06.prod.outlook.com> Message-ID: <44E5C0A9D48A3246966A4AE04692014D132E155E@CH1PRD0610MB355.namprd06.prod.outlook.com> The call forwarding function itself works, if I set call forwarding for a specific extension and then call that extension it will ring at the forwarded external phone. Also, with call forwarding disabled on the extensions the conference auto outcall does call the internal extensions. There just seems to be a disconnect between the conference outcall to a forwarded extension. But perhaps I should rephrase the question... The conference extension I'm needing to setup should auto outcall to a set of external numbers. I figured the way to do it would be to create internal extensions, and set the call forwarding to the external number. But perhaps there's a better way to approach this need? For example, the outcall user listing in the conference.xml is as follows: Is there a way to change the data syntax to instead direct the outbound auto call to an external number instead of an internal user? Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, January 31, 2012 1:10 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference Auto Outcall to External Numbers What actually happens when you set the call forwarding? Console debug log w/ SIP trace would be helpful, as would any relevant configs you've got in place. -MC On Tue, Jan 31, 2012 at 12:46 PM, Ryan Watkins > wrote: Morning/Afternoon/Evening all... I've been trying to setup a conference call that would use the auto outcall function, and it would call external numbers (like their cell). I've tried to setup extensions for the individuals, and set the call forwarding to their external number. Although the auto outcall function will successfully call the extensions when they are set to an internal SIP phone, when call forwarding is enabled the conference call will not call the external number. Any ideas on how to accomplish this? Thanks a lot! Ryan _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/fec9517f/attachment.html From craigesmith at gmail.com Wed Feb 1 03:11:23 2012 From: craigesmith at gmail.com (Craig Smith) Date: Tue, 31 Jan 2012 19:11:23 -0500 Subject: [Freeswitch-users] dialplan show? Message-ID: I asked this before, but I'm going to try one more time, because I guess I find it very curious that it doesn't exists. How do I check the dialplan in FreeSWITCH, like I do in Asterisk with the 'dialplan show' command? Thanks (again) Craig -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/9d67584f/attachment.html From sos at sokhapkin.dyndns.org Wed Feb 1 03:25:05 2012 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Tue, 31 Jan 2012 19:25:05 -0500 Subject: [Freeswitch-users] dialplan show? In-Reply-To: References: Message-ID: <147565084.7TVCtAV9zm@sos> "xml_locate dialplan" On Tuesday 31 January 2012 19:11:23 Craig Smith wrote: > I asked this before, but I'm going to try one more time, because I guess I > find it very curious that it doesn't exists. > > How do I check the dialplan in FreeSWITCH, like I do in Asterisk with the > 'dialplan show' command? > > Thanks (again) > > Craig From sherifomran2000 at yahoo.com Wed Feb 1 03:51:25 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Tue, 31 Jan 2012 16:51:25 -0800 (PST) Subject: [Freeswitch-users] ERR root tag missing In-Reply-To: <147565084.7TVCtAV9zm@sos> Message-ID: <1328057485.57316.YahooMailClassic@web110810.mail.gq1.yahoo.com> hello guys, what could be this error? 2012-02-01 00:41:24.765163 [CONSOLE] switch_core.c:1859 FreeSWITCH Version 1.0.head (git-ac7f94e 2012-01-05 12-43-17 -0600) Started. Max Sessions[1000] Session Rate[30] SQL [Enabled] 2012-02-01 00:41:24.767426 [ERR] switch_xml.c:1623 Error[[error near line 1]: root tag missing] thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/e0b10ecb/attachment.html From mike at jerris.com Wed Feb 1 04:14:56 2012 From: mike at jerris.com (Michael Jerris) Date: Tue, 31 Jan 2012 20:14:56 -0500 Subject: [Freeswitch-users] ERR root tag missing In-Reply-To: <1328057485.57316.YahooMailClassic@web110810.mail.gq1.yahoo.com> References: <1328057485.57316.YahooMailClassic@web110810.mail.gq1.yahoo.com> Message-ID: http://wiki.freeswitch.org/wiki/Debugging_Freeswitch#Chasing_down_XML_errors On Jan 31, 2012, at 7:51 PM, Sherif Omran wrote: > hello guys, > > what could be this error? > > 2012-02-01 00:41:24.765163 [CONSOLE] switch_core.c:1859 > FreeSWITCH Version 1.0.head (git-ac7f94e 2012-01-05 12-43-17 -0600) Started. > Max Sessions[1000] > Session Rate[30] > SQL [Enabled] > 2012-02-01 00:41:24.767426 [ERR] switch_xml.c:1623 Error[[error near line 1]: root tag missing] > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/0cf9a0cc/attachment.html From david.villasmil.work at gmail.com Wed Feb 1 04:19:54 2012 From: david.villasmil.work at gmail.com (David Villasmil) Date: Wed, 1 Feb 2012 02:19:54 +0100 Subject: [Freeswitch-users] freeswitch billing In-Reply-To: References: <5DA89746-B4E5-4107-93E5-9E7C4976DF85@gmx.de> <94D3B27C-6680-44FE-855C-6FE1037DF449@gmx.de> <65013512-02C2-4448-8835-82CE557A2496@gmx.de> Message-ID: Sherif, Send me your sofia.conf.xml please. Also, file_string I don't use it at all... for the "/fsxml/gateways.php" error, that php file is no used anymore. for I tought I removed those... didn't I?.... Yes. those lines are not in that file anymore... please download the LASTEST verison from the git. for "The installation of the webserver was in /var/www I think this should have been in /var/www/html" That's what the variable is for... ;) you can set that to your web server root directory, in my case it was "/var/www/", in yours can be anything... thanks for trying it! David On Wed, Feb 1, 2012 at 1:48 AM, Sherif Omran wrote: > Hi David, > > I changed the web folder to /var/www/html and reinstalled after adjusting > the modules > > It says now > > Creating default Sofia configuration > Creating default Distributors configuration > Creating default Dialplan configuration > > It is seems reachable now. However when i start freeswitch it says the > following. Please look on to the ERR: > > 2012-02-01 00:41:24.564947 [ERR] switch_xml.c:1623 Error[[error near > line 1]: root tag missing] > > This error is many times > > 2012-02-01 00:41:24.559934 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_console] > 2012-02-01 00:41:24.560293 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_logfile] > 2012-02-01 00:41:24.561055 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_enum] > 2012-02-01 00:41:24.562623 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_xml_curl] > 2012-02-01 00:41:24.564947 [ERR] switch_xml.c:1623 Error[[error near line > 1]: root tag missing] > 2012-02-01 00:41:24.565176 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_xml_cdr] > 2012-02-01 00:41:24.567012 [ERR] switch_xml.c:1623 Error[[error near line > 1]: root tag missing] > 2012-02-01 00:41:24.567444 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_cdr_csv] > 2012-02-01 00:41:24.567937 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_event_socket] > 2012-02-01 00:41:24.571180 [ERR] switch_xml.c:1623 Error[[error near line > 1]: root tag missing] > 2012-02-01 00:41:24.571309 [ERR] sofia.c:3566 Open of sofia.conf failed > 2012-02-01 00:41:24.571363 [CRIT] switch_loadable_module.c:1281 Error > Loading module /usr/local/freeswitch/mod/mod_sofia.so > **Module load routine returned an error** > 2012-02-01 00:41:24.571594 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_loopback] > 2012-02-01 00:41:24.640031 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_commands] > 2012-02-01 00:41:24.651262 [ERR] switch_xml.c:1623 Error[[error near line > 1]: root tag missing] > 2012-02-01 00:41:24.655463 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_db] > 2012-02-01 00:41:24.656327 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_dptools] > 2012-02-01 00:41:24.666206 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_expr] > 2012-02-01 00:41:24.668046 [ERR] switch_xml.c:1623 Error[[error near line > 1]: root tag missing] > 2012-02-01 00:41:24.672956 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_fifo] > 2012-02-01 00:41:24.678599 [ERR] switch_xml.c:1623 Error[[error near line > 1]: root tag missing] > 2012-02-01 00:41:24.678785 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_hash] > 2012-02-01 00:41:24.681265 [ERR] switch_xml.c:1623 Error[[error near line > 1]: root tag missing] > 2012-02-01 00:41:24.682817 [ERR] switch_xml.c:1623 Error[[error near line > 1]: root tag missing] > 2012-02-01 00:41:24.688323 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_voicemail] > 2012-02-01 00:41:24.691706 [ERR] switch_xml.c:1623 Error[[error near line > 1]: root tag missing] > 2012-02-01 00:41:24.691834 [ERR] mod_distributor.c:109 Open of > distributor.conf failed > 2012-02-01 00:41:24.691887 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_distributor] > 2012-02-01 00:41:24.692399 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_esf] > 2012-02-01 00:41:24.692809 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_fsv] > 2012-02-01 00:41:24.693259 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_cluechoo] > 2012-02-01 00:41:24.693769 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_valet_parking] > 2012-02-01 00:41:24.694197 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_dialplan_xml] > 2012-02-01 00:41:24.694606 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_dialplan_asterisk] > 2012-02-01 00:41:24.697755 [ERR] switch_xml.c:1623 Error[[error near line > 1]: root tag missing] > 2012-02-01 00:41:24.720513 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_spandsp] > 2012-02-01 00:41:24.725014 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_g723_1] > 2012-02-01 00:41:24.725416 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_g729] > 2012-02-01 00:41:24.726184 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_amr] > 2012-02-01 00:41:24.726558 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_ilbc] > 2012-02-01 00:41:24.727052 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_speex] > 2012-02-01 00:41:24.727435 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_h26x] > 2012-02-01 00:41:24.728311 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_siren] > 2012-02-01 00:41:24.729904 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_sndfile] > 2012-02-01 00:41:24.732573 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_native_file] > 2012-02-01 00:41:24.736664 [ERR] switch_xml.c:1623 Error[[error near line > 1]: root tag missing] > 2012-02-01 00:41:24.736990 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_local_stream] > 2012-02-01 00:41:24.737566 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_tone_stream] > 2012-02-01 00:41:24.741204 [CONSOLE] mod_local_stream.c:161 Can't open > directory: /usr/local/freeswitch/sounds/music/16000 > 2012-02-01 00:41:24.741397 [CONSOLE] mod_local_stream.c:161 Can't open > directory: /usr/local/freeswitch/sounds/music/32000 > 2012-02-01 00:41:24.743688 [ERR] switch_xml.c:1623 Error[[error near line > 1]: root tag missing] > 2012-02-01 00:41:24.746603 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_spidermonkey] > 2012-02-01 00:41:24.752619 [ERR] switch_xml.c:1623 Error[[error near line > 1]: root tag missing] > 2012-02-01 00:41:24.752921 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_lua] > 2012-02-01 00:41:24.753601 [CONSOLE] switch_loadable_module.c:1299 > Successfully Loaded [mod_say_en] > 2012-02-01 00:41:24.756170 [ERR] switch_xml.c:1623 Error[[error near line > 1]: root tag missing] > 2012-02-01 00:41:24.756254 [CONSOLE] switch_loadable_module.c:124 > Starting runtime thread for CORE_SOFTTIMER_MODULE > 2012-02-01 00:41:24.756320 [CONSOLE] switch_loadable_module.c:124 > Starting runtime thread for mod_event_socket > 2012-02-01 00:41:24.761878 [ERR] switch_xml.c:1623 Error[[error near line > 1]: root tag missing] > 2012-02-01 00:41:24.761959 [CONSOLE] switch_core.c:1139 Created ip list > net_list_1 default (deny) > 2012-02-01 00:41:24.762001 [CONSOLE] switch_core.c:1139 Created ip list > net_list_2 default (deny) > 2012-02-01 00:41:24.762036 [CONSOLE] switch_core.c:1139 Created ip list > net_list_3 default (deny) > 2012-02-01 00:41:24.762072 [CONSOLE] switch_core.c:1139 Created ip list > net_list_4 default (deny) > 2012-02-01 00:41:24.762107 [CONSOLE] switch_core.c:1139 Created ip list > net_list_5 default (deny) > 2012-02-01 00:41:24.762273 [CONSOLE] switch_core.c:1139 Created ip list > net_list_6 default (allow) > 2012-02-01 00:41:24.764023 [ERR] switch_xml.c:1623 Error[[error near line > 1]: root tag missing] > 2012-02-01 00:41:24.764921 [ERR] switch_xml.c:1623 Error[[error near line > 1]: root tag missing] > 2012-02-01 00:41:24.765105 [CONSOLE] switch_core.c:1856 > _____ ______ _____ _____ ____ _ _ > | ___| __ ___ ___/ ___\ \ / /_ _|_ _/ ___| | | | > | |_ | '__/ _ \/ _ \___ \\ \ /\ / / | | | || | | |_| | > | _|| | | __/ __/___) |\ V V / | | | || |___| _ | > |_| |_| \___|\___|____/ \_/\_/ |___| |_| \____|_| |_| > > ************************************************************ > * Anthony Minessale II, Michael Jerris, Brian West, Others * > * FreeSWITCH (http://www.freeswitch.org) * > * Paypal Donations Appreciated: paypal at freeswitch.org * > * Brought to you by ClueCon http://www.cluecon.com/ * > ************************************************************ > > 2012-02-01 00:41:24.765163 [CONSOLE] switch_core.c:1859 > FreeSWITCH Version 1.0.head (git-ac7f94e 2012-01-05 12-43-17 -0600) > Started. > Max Sessions[1000] > Session Rate[30] > SQL [Enabled] > 2012-02-01 00:41:24.767426 [ERR] switch_xml.c:1623 Error[[error near line > 1]: root tag missing] > > > > kind regards, > Sherif > > > > > On Jan 31, 2012, at 9:56 PM, David Villasmil wrote: > > Hello Sherif, > > "1- variables file > freeswitch path should be /usr/local/freeswitch/conf" > > No, it should be /usr/local/freeswitch/ as the script copies config files > as well as scripts. > > Ok, let me know > > David > > > On Tue, Jan 31, 2012 at 8:42 PM, Sherif Omran wrote: > >> Hi David, >> >> please edit the following in the installer >> >> 1- variables file >> freeswitch path should be /usr/local/freeswitch/conf >> >> 2- home >> /home/freeswitch >> >> 3- I get error when you execute mysql -u root -p passwd < filename >> it works for me this way mysql -u root -p > then it requests for the password >> >> 4- try to add a username and password for root in the variables >> >> I disabled copying the files to FS till i can merge it manually >> >> try adding chmod 777 to the installer to run >> >> I will let you know when i get 1 step futher >> >> regards >> Sherif >> >> >> >> On Jan 31, 2012, at 7:58 PM, Sherif Omran wrote: >> >> Hi David, >> >> I saw the install.pl script. I made the required changes manually to the >> freeswitch xml files. What is the best way to still use the install script? >> Note that my freeswitch xml files are not default. >> >> waiting your reply >> >> kind regards, >> Sherif >> >> >> >> On Jan 28, 2012, at 8:29 PM, David Villasmil wrote: >> >> No, I don't. >> >> On Sat, Jan 28, 2012 at 1:09 AM, Sherif Omran wrote: >> >>> Hi David, >>> >>> do you use mod_nibble? >>> >>> kind regards, >>> Sherif >>> >>> >>> On Jan 27, 2012, at 2:47 PM, David Villasmil wrote: >>> >>> hello Sherif, >>> >>> Help it is always welcome, even more because i'm not an expert on web >>> pages. >>> I think it is better if we setup a chat conference to talk about this >>> and any other questions you may have. >>> >>> The CDRs are posted via web with xml_curl. >>> The scripts in the gateway directoy are to be copied into the freeswitch >>> server. "gateway" is just a name i gave to the server. >>> >>> Hope this helps. there are some changes that you need to make before >>> making it work, but i can walk you through it. >>> >>> David >>> >>> On Thu, Jan 26, 2012 at 10:13 PM, Sherif Omran wrote: >>> >>>> Dear David, >>>> >>>> there is no folder called gateway in my freeswitch folder. the normal >>>> vars.xml file is located in usr/local/freeswitch/conf/var.xml >>>> >>>> I am a bit confused, do you mean one should create a new folder called >>>> gateway inside the conf folder? or should it be inside the freeswitch >>>> folder? >>>> >>>> Also can it work on 1 server? >>>> >>>> waiting your reply >>>> >>>> best regards, >>>> Sherif >>>> >>>> >>>> On Jan 9, 2012, at 6:35 PM, David Villasmil wrote: >>>> >>>> Sorry, >>>> >>>> It is on https://github.com/davidcsi/FreeSWITCH-Billing >>>> >>>> David >>>> >>>> On Mon, Jan 9, 2012 at 6:34 PM, David Villasmil < >>>> david.villasmil.work at gmail.com> wrote: >>>> >>>>> Hello Sheriff, >>>>> >>>>> I have been using it for more than a year on production. >>>>> >>>>> Enjoy >>>>> >>>>> David >>>>> >>>>> >>>>> On Sat, Jan 7, 2012 at 11:27 PM, Sherif Omran wrote: >>>>> >>>>>> Hi David >>>>>> >>>>>> Is it a full application ready to be used or still under development? >>>>>> Could you please give me the link to the download >>>>>> >>>>>> also please add your program here >>>>>> >>>>>> http://wiki.freeswitch.org/wiki/Billing >>>>>> >>>>>> thank you >>>>>> >>>>>> regards, >>>>>> Sherif >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> On Jan 1, 2012, at 10:48 PM, David Villasmil wrote: >>>>>> >>>>>> Hello Sherif, >>>>>> >>>>>> Unfortunately, I don't have a script yet. But the Readme explains how >>>>>> to install it. >>>>>> I don't have a online demo either as of now, but I will set it up >>>>>> shortly. >>>>>> >>>>>> Thanks for trying it >>>>>> >>>>>> David >>>>>> >>>>>> On Sun, Jan 1, 2012 at 2:27 PM, Sherif Omran wrote: >>>>>> >>>>>>> hi David >>>>>>> >>>>>>> I am interested in your module, do you have an installation script >>>>>>> for a running FS? >>>>>>> >>>>>>> Is there a demo online? >>>>>>> >>>>>>> best regards, >>>>>>> Sherif Omran >>>>>>> >>>>>>> >>>>>> >>>>>> >>>>> >>>> >>>> >>> >>> >> >> >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/dde03d97/attachment-0001.html From dujinfang at gmail.com Wed Feb 1 04:22:30 2012 From: dujinfang at gmail.com (Seven Du) Date: Wed, 1 Feb 2012 09:22:30 +0800 Subject: [Freeswitch-users] ERR root tag missing In-Reply-To: <1328057485.57316.YahooMailClassic@web110810.mail.gq1.yahoo.com> References: <1328057485.57316.YahooMailClassic@web110810.mail.gq1.yahoo.com> Message-ID: <49081B0480174C919A4EAACFCB546A04@gmail.com> check log/freeswitch.xml.fsxml or if you use mod_xml_curl check your webserver log and also check what it returns with xml_curl debug_on -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com) On Wednesday, February 1, 2012 at 8:51 AM, Sherif Omran wrote: > hello guys, > > what could be this error? > > 2012-02-01 00:41:24.765163 [CONSOLE] switch_core.c:1859 > FreeSWITCH Version 1.0.head (git-ac7f94e 2012-01-05 12-43-17 -0600) Started. > Max Sessions[1000] > Session Rate[30] > SQL [Enabled] > 2012-02-01 00:41:24.767426 [ERR] switch_xml.c:1623 Error[[error near line 1]: root tag missing] > > > thank you > > > > _________________________________________________________________________ > 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/20120201/0ad3c814/attachment.html From dujinfang at gmail.com Wed Feb 1 04:29:08 2012 From: dujinfang at gmail.com (Seven Du) Date: Wed, 1 Feb 2012 09:29:08 +0800 Subject: [Freeswitch-users] Skypopen - Skype Client on a different machine than Freeswitch with Skypopen module In-Reply-To: References: <7FB3A2EEA03E4DEB8D617B131C6FB9A7@gmail.com> Message-ID: <2436E7ED128349A7AC3B65AB02F41B9D@gmail.com> Sorry, Idapted was history and I left and I cannot access that server now. There's no secret in modules.conf but you should follow this page to make skypopen work on your box http://wiki.freeswitch.org/wiki/Skypiax On Tuesday, January 31, 2012 at 9:10 AM, A E [Gmail] wrote: > On Mon, Jan 30, 2012 at 6:12 PM, Seven Du wrote: > > > On Tuesday, January 31, 2012 at 2:20 AM, A E [Gmail] wrote: > > > Ok cool. Thank you so much. > > > > > > I found this as some very slimmed down modules.conf.xml figuring FS running on an embedded platform might give me the best case from which to build on. It's here: https://dev.openwrt.org/browser/packages/net/freeswitch/files/etc.minimal/autoload_configs/modules.conf.xml?order=date > > > > > > Apart from adding mod_skypopen to this, does this look good? What else needs to be added to it? > > > > > Add anything you need, see http://wiki.freeswitch.org/wiki/FreeSwitch_Modules for detail > > > > you still can remove some modules if you don't use them, e.g. mod_cdr_csv etc. > > Haha mmm yes, I do understand that but maybe I didn't phrase my question correctly...I'm talking strictly in reference to the FS-Skype box in your environment and what is the bare minimum I can have on that box to be able to successfully run it as a Skype gateway. Kinda trying to get hands on your modules.conf.xml but if that's an inappropriate request, then I apologize in advance. > > > > > > > > > Thanks so much for your help :) > > > > > > > > > On Mon, Jan 30, 2012 at 9:00 AM, Seven Du wrote: > > > > Yes, exactly, you could remove any unnecessary modules if you don't use them. > > > > > > > > to be clear, you could send calls from FS1 to FS2 and FS3 give you have proper dialplan set to route calls to the desired destination. > > > > that means you could just treat FS2 and FS3 as a third party sip gateway but they can routing calls to anywhere you want. > > > > > > > > FS1 (mod_sofia) ---------------------- (mod_sofia)FS2(mod_skypopen) -------------------skype > > > > \ > > > > \------------------ (mod_sofia)FS3(mod_dingaling) ---------------------gtalk > > > > > > > > That testimonial is completely following the wiki's sharing rule, but you can use the idea completely freely. > > > > > > > > On Monday, January 30, 2012 at 12:46 PM, A E [Gmail] wrote: > > > > > > > > > > > > > > > > > On Sun, Jan 29, 2012 at 2:14 AM, Seven Du wrote: > > > > > > skype clients should run with FS on the same box. > > > > > > > > > > > > You could run both FS and skype clients on another box just as a gateway if you want scale. We had run multi- FS instances on one box for scale/stable purpose, but you can get the idea from http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com > > > > > > > > > > > > -- > > > > > > About: http://about.me/dujinfang > > > > > > Blog: http://www.dujinfang.com > > > > > > Proj: http://www.freeswitch.org.cn > > > > > > > > > > > > Sent with Sparrow (http://www.sparrowmailapp.com) > > > > > > > > > > > Thanks Seven. I had a good read of that testimonial and it's basically saying what you just said here in a line. The question now is are the design docs for that setup proprietary or kinda sorta can be GPL'ed or "shared" in some form? :) > > > > > > > > > > If not, then just to clarify, the FS-Skype servers load their own mod_skypopen and is just a gateway to send calls to once it's determined by the FS in the center acting like a soft-switch/call-control platform? Is it completely possible to run/build an FS instance with just mod_skyopen or mod_dingaling (and anything else that's fundamentally needed by FS) loaded? I mean I could Google this but it might help if you could tell me what your modules.conf.xml looks like for Skype only instance? > > > > > > > > > > Thanks so much in advance, and apologies if I'm being unreasonable in requesting for the info :) > > _________________________________________________________________________ > 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/20120201/2347483e/attachment.html From covici at ccs.covici.com Wed Feb 1 04:36:45 2012 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Tue, 31 Jan 2012 20:36:45 -0500 Subject: [Freeswitch-users] dialplan show? In-Reply-To: References: Message-ID: <15194.1328060205@ccs.covici.com> There is no such command for freeswitch, but the complete XML file, with all the includes is in the file (assuming default directories) /usr/local/freeswitch/log/freeswitch.xml.fsxml . It does not change unless you change the xml files and do a reloadxml from the fs_cli console. Hope this helps. Craig Smith wrote: > I asked this before, but I'm going to try one more time, because I guess I > find it very curious that it doesn't exists. > > How do I check the dialplan in FreeSWITCH, like I do in Asterisk with the > 'dialplan show' command? > > Thanks (again) > > Craig > > ---------------------------------------------------- > Alternatives: > > ---------------------------------------------------- > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- 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 krice at freeswitch.org Wed Feb 1 08:48:07 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 31 Jan 2012 23:48:07 -0600 Subject: [Freeswitch-users] Looking for some old gear for testing Message-ID: Hey guys, Looking for an old AS5300 or such that?s rigged for T1s with atleast 24 voice resources for testing PRI and TDM interconnects to FreeSWITCH. If you can help me out please contact me off list... K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120131/dec92ae4/attachment-0001.html From admin at blindi.net Wed Feb 1 08:59:12 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Wed, 1 Feb 2012 06:59:12 +0100 (CET) Subject: [Freeswitch-users] Voicemail Password security bug? In-Reply-To: <15194.1328060205@ccs.covici.com> References: <15194.1328060205@ccs.covici.com> Message-ID: Hi guys, i have installed a fresh fs. The git is from today. After installing Fs, i remove all standard users in: conf/directory/default I restart FS: /etc/init.d/freeswitch restart I connect my Softphone to fs, and enter 4000 the voicemail login extension. After playing the loginprompt i entered: 1000 - 1019 these extensions do not exist anymore. But the voicemailsystem accept the extensionnumbers and same passwords. For example: 1000#1000 # or: 1010#1010 and so on. I have full access to the removed voicemailboxes. I have to manually create these new extensions. 1000 - 1019 And i change the Passwords, the voicemailsystem drops the logins with same passwords. 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 hynek.cihlar at gmail.com Wed Feb 1 10:29:20 2012 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Wed, 1 Feb 2012 08:29:20 +0100 Subject: [Freeswitch-users] sip_gateway dialplan variable In-Reply-To: References: Message-ID: Hello Mike and thanks for the reply. Unfortunately sip_req_host only gives me the sip host. My setup is probably not quite usual, I have a multihome system with many gateways, each gateway name assigned to a unique id. The unique id is simply a random string identifying the gateway in the whole system. The unique id is also persisted in the DB, based on the id additional call processing takes place. My initial idea was to get the gateway name in the dialplan and pass it to an ESL app. But it really looks like it is not possible to retrieve the gateway name at all, sadly. Hynek On Tue, Jan 24, 2012 at 8:42 AM, Michael Lutz wrote: > Hi Hynek, > > {sip_req_host} should give your own gateway (local), and {network_addr} > should give the remote gateway. > > Otherwise writing a cdr and look will help you with this. > > Regards, > Mike > > Op 24 jan. 2012 om 08:27 heeft Hynek Cihlar het > volgende geschreven: > > In one of the examples on wiki I found ${sip_gateway}, but that doesn't > seem to work. Also, there is ${sip_gateway_name}, but that works on > outbound calls only. > > Hynek > > > > On Mon, Jan 23, 2012 at 7:03 PM, Hynek Cihlar wrote: > >> Dear, is there a dialplan variable that would hold the gateway name an >> incoming call came through? Searching through the source code and wiki >> didn't yield any positive result. >> >> Thanks! >> >> Hynek >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/4cae6fdf/attachment.html From gerald.weber at besharp.at Wed Feb 1 11:48:53 2012 From: gerald.weber at besharp.at (Gerald Weber) Date: Wed, 1 Feb 2012 08:48:53 +0000 Subject: [Freeswitch-users] exit lua script in hangup hook In-Reply-To: References: <1328040393457-7241003.post@n2.nabble.com> Message-ID: Hello, Just tried all 4 cases (error(), return "exit", return "die", session:destroy("42")) All of them are working as expected. Thanks for implementing ! I just checked the wiki and the example for the setHangupHook Call (http://wiki.freeswitch.org/wiki/Mod_lua#session:setHangupHook) In the code, error() is used to exit the script - which is now working. So I guess the possibility to exit the script was already there, but not working until the patch from Anthony ? Anyways, thanks ! Now I have to figure out how to handle outgoing calls from lua. And how to control them. (seems like ESL is the only choice for fully asynchronous call handling ?) PS: I've updated the wiki page. Von: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] Im Auftrag von Michael Collins Gesendet: Dienstag, 31. J?nner 2012 22:08 An: FreeSWITCH Users Help Betreff: Re: [Freeswitch-users] exit lua script in hangup hook That's what I get for being 2 months behind on the changelog! :D If we have any Lua peeps out there who could test this and update the wiki I would be eternally grateful... -MC On Tue, Jan 31, 2012 at 12:06 PM, Jeff Lenk > wrote: Just to let you know mercutioviz that anthm did make some source code changes last week that should have enabled this to work. Commit:09ad887948f7513725ca8b53bdfe721d9008e73b * FS-3841 --resolve ok return the string "die" or "exit" from hanguphook to cause an error or call s:destroy("any err message"); either should now halt the script -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/exit-lua-script-in-hangup-hook-tp7223506p7241003.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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/cbaf16a6/attachment.html From hynek.cihlar at gmail.com Wed Feb 1 11:50:36 2012 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Wed, 1 Feb 2012 09:50:36 +0100 Subject: [Freeswitch-users] Method of checking if media is being played at a session In-Reply-To: References: Message-ID: I do it a little different, I wait for the EXECUTE event of the playback application and search for the correlation id. The reason for the correlation ID is to be able to react even in cases there are multiple unbridged calls and some IVR logic going on on both of them. Hynek On Mon, Jan 30, 2012 at 10:54 PM, Michael Lutz wrote: > Hi Hynek (or anyone else ;-) > > Thank you for your response. I see you also use the ESL to invoke the > playback? Is there a particulair for that other than I am using the > UUID_BROADCAST using the api? > I am trying to make it work through the ESL but it seems the initial > events are not received directly after the the broadcast, but it takes > a while. I will paste some example code of what I have done/am trying > to do. > > Any help apreciated! > > In this example I have created a couple of functions (explained in > code) and try t odo an async playback of an audio file, wait for the > ESL to confirm the PLAYBACK_START and then call another function which > waits for all PLAYBACK_STOP's (sum of plays) to return. Then I do > session:hangup. But it never gets the ESL event. > > In theory this should work. But in some way the events do not come in > right away. This code is testable. (only specify a valid filename in > the streamFile) > > Code: http://pastebin.com/ri0L7wmx > > Regards, > Michael Lutz. > > > 2012/1/30 Hynek Cihlar : > > Although I do it using ESL it may be helpful for your case. When issuing > the > > broadcast request I create a 'correlation ID' variable with a generated > > unique ID (can be any suitable string) for each file to be broadcast. > The ID > > is passed back to the ESL app after each particular file stops playing > back. > > > > In the ESl language I issue the command "execute > > playback file_string://myfile.mp3{CorrelationID=myuniqueID}" and then > wait > > for an event with the correct correlation ID. > > > > Hynek > > > > > > > > On Mon, Jan 30, 2012 at 5:40 PM, Michael Lutz > wrote: > >> > >> Hi All, > >> > >> I am dealing with a little problem... > >> > >> To play asynchronous audio in my Lua script I use the api function > >> "uuid_broadcast {uuid} {fileName} aleg". This all works well and audio > >> is being buffered if I call my function multiple times. > >> When I want to abort all plays and flush the buffer I use "uuid_break > >> {uuid} all". This all works excellent! > >> > >> My problem is I at certain points need to know if audio is being > >> played (or awaits in the buffer) and need to pause execution untill > >> all audio is beging played (or flushed). > >> > >> Is there a way of doing this in either Lua or apiFunctions? or a > >> channel variable perhaps? I have looked and tryed but cannot seem to > >> find a way doing this. > >> > >> > >> Any hints (or solution ;-) appreciated! > >> > >> > >> Regards, > >> Michael Lutz > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> 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 > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > 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 > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/2e606ab4/attachment-0001.html From brian at freeswitch.org Wed Feb 1 12:03:10 2012 From: brian at freeswitch.org (Brian West) Date: Wed, 1 Feb 2012 03:03:10 -0600 Subject: [Freeswitch-users] Call FW on phone In-Reply-To: <4F266228.9050008@softnet.si> References: <4F201657.9040005@softnet.si> <4F22571A.9010009@softnet.si> <4F266228.9050008@softnet.si> Message-ID: http://wiki.freeswitch.org/wiki/Dialplan_Handling_Incoming_Redirect#sip_redirect_dialstring This variable will contain the exact dial string to bridge to the contact in the 302. Remeber all these 302 rediret redirect handling variables MUST be set before the initial bridge to the endpoint that MAY or MAY NOT send a 302 (cuz you won't know really) cut to the chase bridge data="${sip_redirect_dialstring}" /b On Jan 30, 2012, at 3:26 AM, Miha Zoubek wrote: > Hi @Vitalie, > > I guess this will do it, but I just have one questione. As call FW number is not in destination_number variable I need a help how to make a bridge with a contact variable (Contact: ;reg-id=1) as in this variblae CFW number is hidden on which call should transfer. > > Thanks! > > Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/88598375/attachment.html From miha at softnet.si Wed Feb 1 12:19:39 2012 From: miha at softnet.si (Miha Zoubek) Date: Wed, 01 Feb 2012 10:19:39 +0100 Subject: [Freeswitch-users] Striping destination number Message-ID: <4F2903AB.3030905@softnet.si> Hi, how can I strip destination number in my dialplan if the destination number is not in condition option. I need to stip leading zero from ${sip_redirect_contact_user_0} variable. Thanks! Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. From miha at softnet.si Wed Feb 1 12:20:24 2012 From: miha at softnet.si (Miha Zoubek) Date: Wed, 01 Feb 2012 10:20:24 +0100 Subject: [Freeswitch-users] Call FW on phone In-Reply-To: References: <4F201657.9040005@softnet.si> <4F22571A.9010009@softnet.si> <4F266228.9050008@softnet.si> Message-ID: <4F2903D8.70804@softnet.si> Hi @brain, I figure it out:) Thanks again! Regards, Miha On 2/1/2012 10:03 AM, Brian West wrote: > http://wiki.freeswitch.org/wiki/Dialplan_Handling_Incoming_Redirect#sip_redirect_dialstring > > > This variable will contain the exact dial string to bridge to the > contact in the 302. Remeber all these 302 rediret redirect handling > variables MUST be set before the initial bridge to the endpoint that > MAY or MAY NOT send a 302 (cuz you won't know really) > > cut to the chase > > bridge data="${sip_redirect_dialstring}" > > > /b > > On Jan 30, 2012, at 3:26 AM, Miha Zoubek wrote: > >> Hi @Vitalie, >> >> I guess this will do it, but I just have one questione. As call FW >> number is not in destination_number variable I need a help how to >> make a bridge with a contact variable (Contact: >> ;reg-id=1) as in this variblae >> CFW number is hidden on which call should transfer. >> >> 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/18d482a7/attachment.html From vipkilla at gmail.com Wed Feb 1 16:04:18 2012 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 1 Feb 2012 08:04:18 -0500 Subject: [Freeswitch-users] valet parking with 2 FS servers In-Reply-To: <577F9061-361B-4DE5-B498-4CB178608FA0@jerris.com> References: <577F9061-361B-4DE5-B498-4CB178608FA0@jerris.com> Message-ID: Would this even be possible? I'm only asking because I may consider a bounty for such a feature. Also would a similar feature be possible with conferences? Thanks. On Tue, Jan 31, 2012 at 5:20 PM, Michael Jerris wrote: > There is no way to do that currently. ?You will need to direct all parking to one box currently. > > Mike > > On Jan 31, 2012, at 3:26 PM, Vik Killa wrote: > >> Hi, >> Is there any way to share the valet parking lots across multiple FS >> servers? I have a setup where opensips distributes calls to two FS >> servers. When a call is parked and I run: >> CLI>valet_info >> The parked call only appears on one FS box. So if the call pickup goes >> to the opposite FS server, the call will not be picked up. >> Is there anyway to make both FS boxes aware of valet_info? Similar how >> to "sofia status profile internal reg" is aware using ODBC > From mike at jerris.com Wed Feb 1 17:15:11 2012 From: mike at jerris.com (Michael Jerris) Date: Wed, 1 Feb 2012 09:15:11 -0500 Subject: [Freeswitch-users] valet parking with 2 FS servers In-Reply-To: References: <577F9061-361B-4DE5-B498-4CB178608FA0@jerris.com> Message-ID: <55DF5FBF-EA03-4440-8364-2A2E9D81D34B@jerris.com> its all software, anything is possible. On Feb 1, 2012, at 8:04 AM, Vik Killa wrote: > Would this even be possible? I'm only asking because I may consider a > bounty for such a feature. Also would a similar feature be possible > with conferences? > Thanks. > > On Tue, Jan 31, 2012 at 5:20 PM, Michael Jerris wrote: >> There is no way to do that currently. You will need to direct all parking to one box currently. >> >> Mike >> >> On Jan 31, 2012, at 3:26 PM, Vik Killa wrote: >> >>> Hi, >>> Is there any way to share the valet parking lots across multiple FS >>> servers? I have a setup where opensips distributes calls to two FS >>> servers. When a call is parked and I run: >>> CLI>valet_info >>> The parked call only appears on one FS box. So if the call pickup goes >>> to the opposite FS server, the call will not be picked up. >>> Is there anyway to make both FS boxes aware of valet_info? Similar how >>> to "sofia status profile internal reg" is aware using ODBC >> From kbdfck at gmail.com Wed Feb 1 17:48:16 2012 From: kbdfck at gmail.com (Dmitry Sytchev) Date: Wed, 1 Feb 2012 18:48:16 +0400 Subject: [Freeswitch-users] sofia_contact is now CaSe-sensitive? Message-ID: Hi all After upgrade to git-head I faced with problem - sofia_contact seems to be case-sensitive now. freeswitch at domain.com> sofia_contact test1 at domain.com error/user_not_registered freeswitch at domain.com> sofia_contact Test1 at domain.com sofia/local/sip:Test1 at 172.19.36.54:5060 ;fs_nat=yes;fs_path=sip%3ATest1%40172.19.36.54%3A5060 Is this intended change? How do I normalize my reg. contacts to match database values? -- Best regards, Dmitry Sytchev, IT Engineer -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/b94cf24b/attachment.html From all.eforums at gmail.com Wed Feb 1 19:16:47 2012 From: all.eforums at gmail.com (A E [Gmail]) Date: Wed, 1 Feb 2012 11:16:47 -0500 Subject: [Freeswitch-users] Skypopen - Skype Client on a different machine than Freeswitch with Skypopen module In-Reply-To: <2436E7ED128349A7AC3B65AB02F41B9D@gmail.com> References: <7FB3A2EEA03E4DEB8D617B131C6FB9A7@gmail.com> <2436E7ED128349A7AC3B65AB02F41B9D@gmail.com> Message-ID: Thanks Seven, Yes, I have been using that page all along. My problem is that I have Debian running on Sparc. The Skype client downloaded by the Installer in Skyopen is build for x86. I don't know if the Skype client has also been modified by Giovanni or he doesn't have access to its source code. Anyway, I'll try to work it out. Thanks again for your help. On Tue, Jan 31, 2012 at 8:29 PM, Seven Du wrote: > Sorry, Idapted was history and I left and I cannot access that server now. > > There's no secret in modules.conf but you should follow this page to make > skypopen work on your box http://wiki.freeswitch.org/wiki/Skypiax > > On Tuesday, January 31, 2012 at 9:10 AM, A E [Gmail] wrote: > > On Mon, Jan 30, 2012 at 6:12 PM, Seven Du wrote: > > > On Tuesday, January 31, 2012 at 2:20 AM, A E [Gmail] wrote: > > Ok cool. Thank you so much. > > I found this as some very slimmed down modules.conf.xml figuring FS > running on an embedded platform might give me the best case from which to > build on. It's here: > https://dev.openwrt.org/browser/packages/net/freeswitch/files/etc.minimal/autoload_configs/modules.conf.xml?order=date > > Apart from adding mod_skypopen to this, does this look good? What else > needs to be added to it? > > Add anything you need, see > http://wiki.freeswitch.org/wiki/FreeSwitch_Modules for detail > > you still can remove some modules if you don't use them, e.g. mod_cdr_csv > etc. > > > Haha mmm yes, I do understand that but maybe I didn't phrase my question > correctly...I'm talking strictly in reference to the FS-Skype box in your > environment and what is the bare minimum I can have on that box to be able > to successfully run it as a Skype gateway. Kinda trying to get hands on > your modules.conf.xml but if that's an inappropriate request, then I > apologize in advance. > > > > > Thanks so much for your help :) > > > On Mon, Jan 30, 2012 at 9:00 AM, Seven Du wrote: > > Yes, exactly, you could remove any unnecessary modules if you don't use > them. > > to be clear, you could send calls from FS1 to FS2 and FS3 give you have > proper dialplan set to route calls to the desired destination. > that means you could just treat FS2 and FS3 as a third party sip gateway > but they can routing calls to anywhere you want. > > FS1 (mod_sofia) ---------------------- (mod_sofia)FS2(mod_skypopen) > -------------------skype > \ > \------------------ > (mod_sofia)FS3(mod_dingaling) ---------------------gtalk > > That testimonial is completely following the wiki's sharing rule, but you > can use the idea completely freely. > > On Monday, January 30, 2012 at 12:46 PM, A E [Gmail] wrote: > > On Sun, Jan 29, 2012 at 2:14 AM, Seven Du wrote: > > skype clients should run with FS on the same box. > > You could run both FS and skype clients on another box just as a gateway > if you want scale. We had run multi- FS instances on one box for > scale/stable purpose, but you can get the idea from > http://wiki.freeswitch.org/wiki/FreeSWITCH_Testimonial_on_Idapted.com > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > Sent with Sparrow > > Thanks Seven. I had a good read of that testimonial and it's basically > saying what you just said here in a line. The question now is are the > design docs for that setup proprietary or kinda sorta can be GPL'ed or > "shared" in some form? :) > > If not, then just to clarify, the FS-Skype servers load their own > mod_skypopen and is just a gateway to send calls to once it's determined by > the FS in the center acting like a soft-switch/call-control platform? Is it > completely possible to run/build an FS instance with just mod_skyopen or > mod_dingaling (and anything else that's fundamentally needed by FS) loaded? > I mean I could Google this but it might help if you could tell me what your > modules.conf.xml looks like for Skype only instance? > > Thanks so much in advance, and apologies if I'm being unreasonable in > requesting for the info :) > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/85307784/attachment-0001.html From justlikeef at gmail.com Wed Feb 1 21:16:32 2012 From: justlikeef at gmail.com (Rob Hutton) Date: Wed, 1 Feb 2012 13:16:32 -0500 Subject: [Freeswitch-users] Don't want to display ringing BLF Message-ID: <201202011316.33130.justlikeef@gmail.com> We have a group of 5 phones currently configured in a ring group that all have BLF keys monitoring each other. The problem is that when a call comes in, all of the BLF lines flash and no one can tell who is actually on the phone. Can anyone think of a configuration where all five phone can ring on an incoming call, they can see if the other phones are in use, and ringing is ignored? Thanks, Rob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/c59aad4a/attachment.html From msc at freeswitch.org Wed Feb 1 21:25:47 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Feb 2012 10:25:47 -0800 Subject: [Freeswitch-users] Striping destination number In-Reply-To: <4F2903AB.3030905@softnet.si> References: <4F2903AB.3030905@softnet.si> Message-ID: If you know for a fact that it's only a leading zero then you can use the variable manipulation features: http://wiki.freeswitch.org/wiki/Manipulating_Channel_Variables -MC On Wed, Feb 1, 2012 at 1:19 AM, Miha Zoubek wrote: > Hi, > > how can I strip destination number in my dialplan if the destination > number is not in condition option. > > data="sofia/external/386${sip_redirect_contact_user_0}@xxx.xxx.xxx.xxx" /> > I need to stip leading zero from ${sip_redirect_contact_user_0} variable. > > Thanks! > Miha > > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/caea0fa7/attachment.html From ira at connectmevoice.com Wed Feb 1 22:38:18 2012 From: ira at connectmevoice.com (Ira Tessler) Date: Wed, 1 Feb 2012 14:38:18 -0500 Subject: [Freeswitch-users] Using Oracle as a database Message-ID: <02d4e1d1331f3dca74d67e89c52fec57@mail.gmail.com> I have odbc set up for Oracle on my FS Linux box. I can connect to Oracle using isql and run queies. I am trying to configure Freeswitch to using the database but when I start up Freeswitch I get the following error message: 2012-02-01 14:29:07.056793 [CRIT] switch_odbc.c:280 The sql server is not responding for DSN dbnj [STATE: HY000 CODE 923 ERROR: [Oracle][ODBC][Ora]ORA-00923: FROM keyword not found where expected Anyone have any ideas ? Thanks, Ira Tessler ConnectMe (732) 490-9007 x2 www.connectmevoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/2ef758b1/attachment.html From mario_fs at mgtech.com Wed Feb 1 22:58:16 2012 From: mario_fs at mgtech.com (Mario G) Date: Wed, 1 Feb 2012 11:58:16 -0800 Subject: [Freeswitch-users] Can't get Paging to work on SPA962 Message-ID: I have not been able to get paging to work on SPA962s for a year now, I ran traces searched the web, etc. No luck. I found a post from 2009 that said it works but no description on how. Here is what happens: If I connect my old SPA9000 PBX up and power it on, then turn it off, FreeSwitch paging works! But, if a phone is powered off, FreeSwitch paging no longer works. Power on/off SPA9000 and all is well. Anyone out there have it working that could tell me the secret? I am using *11 to page to trigger the rtp_multicast_page extension. Thanks in advance! MarioG -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/527e419d/attachment.html From kevinw at centrallogic.com Wed Feb 1 23:04:08 2012 From: kevinw at centrallogic.com (Kevin White) Date: Wed, 1 Feb 2012 14:04:08 -0600 Subject: [Freeswitch-users] FreeSwitch compilation error... Message-ID: Hi all, Trying to get freeswitch to compile here. I took the following steps: 1. used the instructions to download the source from git 2. ./bootstrap.sh 3. Edit modules.conf to uncomment the flite and freetdm modules 4. ./configure 5. Make I get the following during the 'make' process: making all mod_dialplan_xml Compiling /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c... mkdir .libs Compiling /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c ... /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c: In function 'parse_exten': /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c:147: error: expected ';' before '}' token make[5]: *** [mod_dialplan_xml.lo] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_dialplan_xml-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Any tips? Thanks, Kevin ________________________________ Kevin White Senior Software Engineer (801) 727-2362 office CentralLogic www.centrallogic.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/dde5a816/attachment-0001.html From mario_fs at mgtech.com Wed Feb 1 23:07:03 2012 From: mario_fs at mgtech.com (Mario G) Date: Wed, 1 Feb 2012 12:07:03 -0800 Subject: [Freeswitch-users] Community Alert: Our Friend Brian West Is In The Hospital In-Reply-To: <4F27C668.2090808@telefaks.de> References: <03C6C35D-FECD-4973-9A95-E18F71923320@freeswitch.org> <4F27C668.2090808@telefaks.de> Message-ID: <91010650-1994-4565-B9F3-2C8F6039E500@mgtech.com> Don't worry, get well, and do what they tell you after you leave the hospital. Best wishes for your quick recovery! MarioG On Jan 31, 2012, at 2:46 AM, Peter Steinbach wrote: > Get well soon Brian > > Peter > > > Am 31.01.2012 10:34, schrieb Brian West: >> >> Thanks everyone for your kind words. I'm hanging in there trying to keep a level head about everything and have a positive outlook on everything. >> >> Thanks, >> Brian >> >> On Jan 31, 2012, at 1:22 AM, Zohair Raza wrote: >> >>> Get well soon Brian, >>> >>> Regards, >>> Zohair Raza >>> >>> On Mon, Jan 30, 2012 at 9:45 PM, Massimiliano Ravelli < >>> massimiliano.ravelli at gmail.com> wrote: >>> >>>> 2012/1/30 Michael Collins >>>> >>>> For those who are interested in helping in more practical ways, such as >>>>> helping to defray the considerable medical costs that no doubt will be >>>>> incurred, we will have more information shortly. >>>>> >>>> >>>> Let us know ! >>>> >>>> Massimiliano >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> 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 >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >> >> -- >> Brian West >> FreeSWITCH Solutions, LLC >> Phone: +1 (918) 420-9266 >> Fax: +1 (918) 420-9267 >> brian at freeswitch.org >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -- > With kind regards > Peter Steinbach > > Telefaks Services GmbH > mailto:lists (att) telefaks.de > Internet: www.telefaks.de > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/8b8c1f1f/attachment.html From gerald.weber at besharp.at Wed Feb 1 23:10:23 2012 From: gerald.weber at besharp.at (Gerald Weber) Date: Wed, 1 Feb 2012 20:10:23 +0000 Subject: [Freeswitch-users] Using Oracle as a database Message-ID: FS "pings " the db to keep the connection open. The statement is "select 1" by default, which is invalid for ORACLE. There is no config file to set this, you have to edit the file src/switch_odbc.c (iirc) and recompile. Sent from my Android phone using TouchDown (www.nitrodesk.com) -----Original Message----- From: Ira Tessler [ira at connectmevoice.com] Received: Mittwoch, 01 Feb. 2012, 20:44 To: FreeSWITCH-users at lists.freeswitch.org [FreeSWITCH-users at lists.freeswitch.org] Subject: [Freeswitch-users] Using Oracle as a database I have odbc set up for Oracle on my FS Linux box. I can connect to Oracle using isql and run queies. I am trying to configure Freeswitch to using the database but when I start up Freeswitch I get the following error message: 2012-02-01 14:29:07.056793 [CRIT] switch_odbc.c:280 The sql server is not responding for DSN dbnj [STATE: HY000 CODE 923 ERROR: [Oracle][ODBC][Ora]ORA-00923: FROM keyword not found where expected Anyone have any ideas ? Thanks, Ira Tessler ConnectMe (732) 490-9007 x2 www.connectmevoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/598a1e67/attachment.html From dflawal at hotmail.com Wed Feb 1 23:25:55 2012 From: dflawal at hotmail.com (David Lawal) Date: Wed, 1 Feb 2012 13:25:55 -0700 Subject: [Freeswitch-users] Community Alert: Our Friend Brian West Is In The Hospital In-Reply-To: References: , , , <03C6C35D-FECD-4973-9A95-E18F71923320@freeswitch.org>, , Message-ID: Wishing you a speedy recovery Brian. Best wishes. David Date: Tue, 31 Jan 2012 18:08:51 +0800 From: gcd at i.ph To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] Community Alert: Our Friend Brian West Is In The Hospital that's right Brian. get well soon. with our prayers. -Nandy On Tue, Jan 31, 2012 at 5:39 PM, Gabriel Gunderson wrote: On Tue, Jan 31, 2012 at 2:34 AM, Brian West wrote: > Thanks everyone for your kind words. I'm hanging in there trying to keep a > level head about everything and have a positive outlook on everything. Just know that there are 100s of people hoping for your speedy recovery. Be well. 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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/df023beb/attachment.html From msc at freeswitch.org Thu Feb 2 00:08:23 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Feb 2012 13:08:23 -0800 Subject: [Freeswitch-users] FreeSwitch compilation error... In-Reply-To: References: Message-ID: You might be missing some dependencies. Double-check by using this handy install script that the FS devs use for the sake of convenience: http://www.freeswitch.org/eg/Makefile -MC On Wed, Feb 1, 2012 at 12:04 PM, Kevin White wrote: > Hi all,**** > > ** ** > > Trying to get freeswitch to compile here. I took the following steps:**** > > ** ** > > **1. **used the instructions to download the source from git**** > > **2. **./bootstrap.sh**** > > **3. **Edit modules.conf to uncomment the flite and freetdm modules* > *** > > **4. **./configure**** > > **5. **Make**** > > ** ** > > I get the following during the ?make? process:**** > > ** ** > > making all mod_dialplan_xml**** > > Compiling > /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c... > **** > > mkdir .libs**** > > Compiling > /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c > ...**** > > /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c: > In function 'parse_exten':**** > > /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c:147: > error: expected ';' before '}' token**** > > make[5]: *** [mod_dialplan_xml.lo] Error 1**** > > make[4]: *** [all] Error 1**** > > make[3]: *** [mod_dialplan_xml-all] Error 1**** > > make[2]: *** [all-recursive] Error 1**** > > make[1]: *** [all-recursive] Error 1**** > > make: *** [all] Error 2**** > > ** ** > > Any tips?**** > > ** ** > > Thanks,**** > > ** ** > > Kevin**** > > ** ** > > ** ** > ------------------------------ > > Kevin White**** > > Senior Software Engineer**** > > ** ** > > (801) 727-2362 office**** > > CentralLogic**** > > ** ** > > www.centrallogic.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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/753299bd/attachment-0001.html From msc at freeswitch.org Thu Feb 2 00:12:08 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Feb 2012 13:12:08 -0800 Subject: [Freeswitch-users] exit lua script in hangup hook In-Reply-To: References: <1328040393457-7241003.post@n2.nabble.com> Message-ID: On Wed, Feb 1, 2012 at 12:48 AM, Gerald Weber wrote: > Hello,**** > > ** ** > > Just tried all 4 cases (error(), return ?exit?, return ?die?, > session:destroy(?42?))**** > > All of them are working as expected. Thanks for implementing !**** > > ** ** > > I just checked the wiki and the example for the setHangupHook Call ( > http://wiki.freeswitch.org/wiki/Mod_lua#session:setHangupHook)**** > > In the code, error() is used to exit the script ? which is now working.*** > * > > So I guess the possibility to exit the script was already there, but not > working until the patch from Anthony ?**** > > ** ** > > Anyways, thanks !**** > > ** ** > > Now I have to figure out how to handle outgoing calls from lua. And how to > control them.**** > > (seems like ESL is the only choice for fully asynchronous call > handling ?) > Yes, fully async call control is best handled with ESL or the new mod_httapi. If you do a Lua dialplan script then you are kind of in your own little world (or thread, as it were). External call control has many advantages and very few disadvantages. > **** > > ** ** > > PS:**** > > I?ve updated the wiki page. > Thanks! -MC > **** > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/c8504b2f/attachment.html From kevinw at centrallogic.com Thu Feb 2 00:15:04 2012 From: kevinw at centrallogic.com (Kevin White) Date: Wed, 1 Feb 2012 15:15:04 -0600 Subject: [Freeswitch-users] FreeSwitch compilation error... In-Reply-To: References: Message-ID: So, I temporarily fixed the error with adding a semicolon to my source file, to get it to compile and install. When that was done, I deleted that file did a "make sure" which pulled from git, overriding that file, and giving me a successful build. (I had a hunch that someone might fix it in git, and it looks like that's the case based on the git log). Anyway, all's good now. Latest git source has it fixed. Thanks for the help, Kevin From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Wednesday, February 01, 2012 2:08 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSwitch compilation error... You might be missing some dependencies. Double-check by using this handy install script that the FS devs use for the sake of convenience: http://www.freeswitch.org/eg/Makefile -MC On Wed, Feb 1, 2012 at 12:04 PM, Kevin White > wrote: Hi all, Trying to get freeswitch to compile here. I took the following steps: 1. used the instructions to download the source from git 2. ./bootstrap.sh 3. Edit modules.conf to uncomment the flite and freetdm modules 4. ./configure 5. Make I get the following during the 'make' process: making all mod_dialplan_xml Compiling /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c... mkdir .libs Compiling /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c ... /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c: In function 'parse_exten': /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c:147: error: expected ';' before '}' token make[5]: *** [mod_dialplan_xml.lo] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_dialplan_xml-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 Any tips? Thanks, Kevin ________________________________ Kevin White Senior Software Engineer (801) 727-2362 office CentralLogic www.centrallogic.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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/097d82e4/attachment.html From freeswitch at earthspike.net Thu Feb 2 00:40:43 2012 From: freeswitch at earthspike.net (John) Date: Wed, 01 Feb 2012 21:40:43 +0000 Subject: [Freeswitch-users] FreeSwitch compilation error... In-Reply-To: References: Message-ID: <4F29B15B.2090409@earthspike.net> You're right. I looked a few minutes ago and it was about one or 2 commits ago. John On 01/02/12 21:15, Kevin White wrote: > > So, I temporarily fixed the error with adding a semicolon to my source > file, to get it to compile and install. When that was done, I deleted > that file did a "make sure" which pulled from git, overriding that > file, and giving me a successful build. (I had a hunch that someone > might fix it in git, and it looks like that's the case based on the > git log). > > Anyway, all's good now. Latest git source has it fixed. > > Thanks for the help, > > Kevin > > *From:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of > *Michael Collins > *Sent:* Wednesday, February 01, 2012 2:08 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] FreeSwitch compilation error... > > You might be missing some dependencies. Double-check by using this > handy install script that the FS devs use for the sake of convenience: > > http://www.freeswitch.org/eg/Makefile > > -MC > > On Wed, Feb 1, 2012 at 12:04 PM, Kevin White > wrote: > > Hi all, > > Trying to get freeswitch to compile here. I took the following steps: > > 1.used the instructions to download the source from git > > 2../bootstrap.sh > > 3.Edit modules.conf to uncomment the flite and freetdm modules > > 4../configure > > 5.Make > > I get the following during the 'make' process: > > making all mod_dialplan_xml > > Compiling > /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c... > > mkdir .libs > > Compiling > /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c > ... > > /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c: > In function 'parse_exten': > > /usr/local/src/freeswitch/src/mod/dialplans/mod_dialplan_xml/mod_dialplan_xml.c:147: > error: expected ';' before '}' token > > make[5]: *** [mod_dialplan_xml.lo] Error 1 > > make[4]: *** [all] Error 1 > > make[3]: *** [mod_dialplan_xml-all] Error 1 > > make[2]: *** [all-recursive] Error 1 > > make[1]: *** [all-recursive] Error 1 > > make: *** [all] Error 2 > > Any tips? > > Thanks, > > Kevin > > ------------------------------------------------------------------------ > > Kevin White > > Senior Software Engineer > > (801) 727-2362 office > > CentralLogic > > www.centrallogic.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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/07eacea9/attachment-0001.html From lliu at multinet.net.au Thu Feb 2 01:08:25 2012 From: lliu at multinet.net.au (Louie Liu) Date: Thu, 2 Feb 2012 09:08:25 +1100 Subject: [Freeswitch-users] unable to make outbound fax Message-ID: <005801cce12e$05358300$0fa08900$@net.au> Hi, I've got problem with outbound fax, the fax machine is connected to FS server via ATA device in a pass through mode, inbound fax works fine, but could get outbound fax to work, here is the dialplan for it, do I miss any parameters? What's wrong with the dialplan? Fax_A (4919)->ATA_A->SIP->FreeSWITCH->SIP provider->PSTN->Fax B Cheers, Louie -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120202/f6bda4a5/attachment.html From msc at freeswitch.org Thu Feb 2 01:23:14 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 1 Feb 2012 14:23:14 -0800 Subject: [Freeswitch-users] unable to make outbound fax In-Reply-To: <005801cce12e$05358300$0fa08900$@net.au> References: <005801cce12e$05358300$0fa08900$@net.au> Message-ID: You might want to pastebin the console debug log of the failed call. -MC On Wed, Feb 1, 2012 at 2:08 PM, Louie Liu wrote: > ** ** > > Hi,**** > > ** ** > > I?ve got problem with outbound fax, the fax machine is connected to FS > server via ATA device in a pass through mode, inbound fax works fine, but > could get outbound fax to work, here is the dialplan for it, do I miss any > parameters? What?s wrong with the dialplan?**** > > ** ** > > Fax_A (4919)->ATA_A->SIP->FreeSWITCH->SIP provider->PSTN->Fax B**** > > ** ** > > ** ** > > **** > > **** > > **** > > data="{sip_invite_domain=$${domain}}user/${dialed_extension}@ > ${domain_name}"/>**** > > **** > > **** > > ** ** > > ** ** > > Cheers,**** > > Louie**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/42ba9301/attachment.html From admin at blindi.net Thu Feb 2 06:04:07 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Thu, 2 Feb 2012 04:04:07 +0100 (CET) Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: References: <005801cce12e$05358300$0fa08900$@net.au> Message-ID: Hi all, I have setup a useraccount. But Fs don.t send my Voicemails to the emialadress. my useraccount settings is: Fs ignore the maildelivery Can you help please? --------------- 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 djbinter at gmail.com Thu Feb 2 07:06:02 2012 From: djbinter at gmail.com (DJB International) Date: Wed, 1 Feb 2012 20:06:02 -0800 Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: References: <005801cce12e$05358300$0fa08900$@net.au> Message-ID: Did you check your maillog? Try to see whether it reached your mail server. -djbinter On Wed, Feb 1, 2012 at 7:04 PM, Thomas Hoellriegel wrote: > Hi all, > I have setup a useraccount. But Fs don.t send my Voicemails to the > emialadress. > my useraccount settings is: > > > > > > > > > > > > > > > > > > > > > > > > > > > > value="$${outbound_caller_id}"**/> > > > > > > Fs ignore the maildelivery > Can you help please? > > > --------------- > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120201/a74e5ee6/attachment.html From sharad at coraltele.com Thu Feb 2 07:54:33 2012 From: sharad at coraltele.com (Sharad Garg) Date: Thu, 2 Feb 2012 10:24:33 +0530 Subject: [Freeswitch-users] Brian...get well soon. References: <03C6C35D-FECD-4973-9A95-E18F71923320@freeswitch.org><4F27C668.2090808@telefaks.de> <91010650-1994-4565-B9F3-2C8F6039E500@mgtech.com> Message-ID: <1F108F4482994B829DA831A4FD761C1F@sharad> Get well soon Brian... God Bless you.. Regards Sharad ----- Original Message ----- From: Mario G To: FreeSWITCH Users Help Sent: Thursday, February 02, 2012 1:37 AM Subject: Re: [Freeswitch-users] Community Alert: Our Friend Brian West IsIn The Hospital Don't worry, get well, and do what they tell you after you leave the hospital. Best wishes for your quick recovery! MarioG On Jan 31, 2012, at 2:46 AM, Peter Steinbach wrote: Get well soon Brian Peter Am 31.01.2012 10:34, schrieb Brian West: Thanks everyone for your kind words. I'm hanging in there trying to keep a level head about everything and have a positive outlook on everything. Thanks, Brian On Jan 31, 2012, at 1:22 AM, Zohair Raza wrote: Get well soon Brian, Regards, Zohair Raza On Mon, Jan 30, 2012 at 9:45 PM, Massimiliano Ravelli < massimiliano.ravelli at gmail.com> wrote: 2012/1/30 Michael Collins For those who are interested in helping in more practical ways, such as helping to defray the considerable medical costs that no doubt will be incurred, we will have more information shortly. Let us know ! Massimiliano _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- With kind regards Peter Steinbach Telefaks Services GmbH mailto:lists (att) telefaks.de Internet: www.telefaks.de _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120202/2938c801/attachment-0001.html From joe.jflemmings at gmail.com Thu Feb 2 11:07:30 2012 From: joe.jflemmings at gmail.com (Joe Flemmings) Date: Thu, 2 Feb 2012 00:07:30 -0800 Subject: [Freeswitch-users] FreeSwtich DB Message-ID: Is their a reason why freeswitch still creates DB files in /usr/local/freeswitch/db/ even when using ODBC at the core as decsribed here http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120202/046e7ec4/attachment.html From brian at systemix.com Thu Feb 2 02:50:25 2012 From: brian at systemix.com (Brian Cuthie) Date: Wed, 1 Feb 2012 18:50:25 -0500 Subject: [Freeswitch-users] FreeTDM problems Message-ID: Hi, When I call an FXS port it keeps generating ringing for quite some time after the caller hangs up. I don't know a whole lot about FreeTDM at this point, but I did notice that in channel_on_hangup() (mod_freetdm.c:591) the call to ftdm_channel_call_hangup() is being skipped. Is that normal? Thanks -brian From mytemike72 at gmail.com Thu Feb 2 12:04:49 2012 From: mytemike72 at gmail.com (Michael Lutz) Date: Thu, 2 Feb 2012 10:04:49 +0100 Subject: [Freeswitch-users] Method of checking if media is being played at a session In-Reply-To: References: Message-ID: Hi Hynek, Hmm, ok I get that, but I wonder if it is realy async? because in my tests I can send a uuid_broadcast to the api, code continues, but play don't start!, I can place for example 5 plays inside the 'play queue', but they won't start untill I do a getDigits(), Then all my plays will start, but are still blocking inside the 'getDigits()' even if I set the timeout to 10 miliseconds ( getDigits(1, "", 10) ) allthough this of course would be a nasty workaround anyway. And then still it will wait for all the plays to finish before code continues (so not after the 10 miliseconds), this is the first moment I can execute my pop() to get the play/stop events, and then they all will arive at once... In what way do you actually start the playbacks, or is this a bug? Regards, Michael Lutz. 2012/2/1 Hynek Cihlar : > I do it a little different, I wait for the EXECUTE event of the playback > application and search for the correlation id. The reason for the > correlation ID is to be able to react even in cases there are multiple > unbridged calls and some IVR logic going on on both of them. > > Hynek > > > > > On Mon, Jan 30, 2012 at 10:54 PM, Michael Lutz wrote: >> >> Hi Hynek (or anyone else ;-) >> >> Thank you for your response. I see you also use the ESL to invoke the >> playback? Is there a particulair for that other than I am using the >> UUID_BROADCAST using the api? >> I am trying to make it work through the ESL but it seems the initial >> events are not received directly after the the broadcast, but it takes >> a while. I will paste some example code of what I have done/am trying >> to do. >> >> Any help apreciated! >> >> In this example I have created a couple of functions (explained in >> code) and try t odo an async playback of an audio file, wait for the >> ESL to confirm the PLAYBACK_START and then call another function which >> waits for all PLAYBACK_STOP's (sum of plays) to return. Then I do >> session:hangup. But it never gets the ESL event. >> >> In theory this should work. But in some way the events do not come in >> right away. This code is testable. (only specify a valid filename in >> the streamFile) >> >> Code: http://pastebin.com/ri0L7wmx >> >> Regards, >> Michael Lutz. >> >> >> 2012/1/30 Hynek Cihlar : >> > Although I do it using ESL it may be helpful for your case. When issuing >> > the >> > broadcast request I create a 'correlation ID' variable with a generated >> > unique ID (can be any suitable string) for each file to be broadcast. >> > The ID >> > is passed back to the ESL app after each particular file stops playing >> > back. >> > >> > In the ESl language I issue the command "execute >> > playback?file_string://myfile.mp3{CorrelationID=myuniqueID}" and then >> > wait >> > for an event with the correct correlation ID. >> > >> > Hynek >> > >> > >> > >> > On Mon, Jan 30, 2012 at 5:40 PM, Michael Lutz >> > wrote: >> >> >> >> Hi All, >> >> >> >> I am dealing with a little problem... >> >> >> >> To play asynchronous audio in my Lua script I use the api function >> >> "uuid_broadcast {uuid} {fileName} aleg". This all works well and audio >> >> is being buffered if I call my function multiple times. >> >> When I want to abort all plays and flush the buffer I use "uuid_break >> >> {uuid} all". This all works excellent! >> >> >> >> My problem is I at certain points need to know if audio is being >> >> played (or awaits in the buffer) and need to pause execution untill >> >> all audio is beging played (or flushed). >> >> >> >> Is there a way of doing this in either Lua or apiFunctions? or a >> >> channel variable perhaps? I have looked and tryed but cannot seem to >> >> find a way doing this. >> >> >> >> >> >> Any hints (or solution ;-) appreciated! >> >> >> >> >> >> Regards, >> >> Michael Lutz >> >> >> >> >> >> _________________________________________________________________________ >> >> Professional FreeSWITCH Consulting Services: >> >> 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 >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > 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 >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From rmorin at blie-ent.com Thu Feb 2 14:58:20 2012 From: rmorin at blie-ent.com (Rob Morin) Date: Thu, 2 Feb 2012 06:58:20 -0500 Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: References: <005801cce12e$05358300$0fa08900$@net.au> Message-ID: <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> I think you're missing the email address: The isn't going to send any voicemails, just send a notification that there are voicemails. You might also need to look in your system log - there're problems with certain platforms sending emails. They'll show up in the system log as a segfault. Good luck, Rob From: DJB International [mailto:djbinter at gmail.com] Sent: Wednesday, February 01, 2012 11:06 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Fs ignoring voicemail to email Did you check your maillog? Try to see whether it reached your mail server. -djbinter On Wed, Feb 1, 2012 at 7:04 PM, Thomas Hoellriegel wrote: Hi all, I have setup a useraccount. But Fs don.t send my Voicemails to the emialadress. my useraccount settings is: Fs ignore the maildelivery Can you help please? --------------- 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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120202/3fd31d4a/attachment.html From henrikaagaardsorensen at gmail.com Thu Feb 2 15:13:40 2012 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Thu, 2 Feb 2012 13:13:40 +0100 Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> References: <005801cce12e$05358300$0fa08900$@net.au> <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> Message-ID: Perhaps stupid answer, but have you installed sendmail or similar? On Thu, Feb 2, 2012 at 12:58 PM, Rob Morin wrote: > I think you're missing the email address: > > ??????????????? > > > > The isn't going to > send any voicemails, just send a notification that there are voicemails. > > > > You might also need to look in your system log - there're problems with > certain platforms sending emails. They'll show up in the system log as a > segfault. > > > > Good luck, > > Rob > > > > > > > > From: DJB International [mailto:djbinter at gmail.com] > Sent: Wednesday, February 01, 2012 11:06 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Fs ignoring voicemail to email > > > > Did you check your maillog?? Try to see whether it reached your mail server. > > -djbinter > > On Wed, Feb 1, 2012 at 7:04 PM, Thomas Hoellriegel wrote: > > Hi all, > I have setup a useraccount. But Fs don.t send my Voicemails to the > emialadress. > my useraccount settings is: > > > ? > ? ? > > ? ? ? > > > > > > > > > > > > > > ? ? > ? > ? ? ? > > > ? > ? > ? > ? > ? > ? ? > ? > > > Fs ignore the maildelivery > Can you help please? > > > --------------- > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kerem.erciyes at gmail.com Thu Feb 2 16:27:34 2012 From: kerem.erciyes at gmail.com (Kerem Erciyes) Date: Thu, 2 Feb 2012 15:27:34 +0200 Subject: [Freeswitch-users] SIP Multiplexed Trunk as IAX2 or TDM In-Reply-To: References: <6180011.valLcmJJfL@sos> <4A81A8E2C3972444A68BB9D0093D783834096E@telisimo-mail-mx.tlisimo.dom> Message-ID: I used IAX2 trunks between Asterisk boxes back when I did VoIP with Asterisk and even though it was bandwidth efficient on paper, whenever the MPLS backbone (with Voice networks prioritised mind you) was a little congested we experienced stutter in all cals made on the trunk and only fix was to hang-up and dial again. When we switched to SIP stutters and problems still happened sometimes, but not all calls failed at the same time, which made users believe the PBX was not broken after all. Just another of my horrible experiences debugging VoIP problems and running on minimal infrastructure. Regards, Kerem On Tue, Jan 31, 2012 at 7:36 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > I have pondered this topic for years and have never come to a > conclusion I was happy with to move on it. > > Basically, on one hand, as Ken mentioned if you increase the ptime of > every call, you get a lot of the same benefits of trunking at the cost > of more audio lost if a packet disappears. This is not a big risk > even at 60ms its only a minor loss of audio. The trunking approach > does offer much less latency but its a matter of deciding if a complex > implementation of trunking is worth the few ms of latency. > > If you look at trunking as a whole, its the idea of muxing in as many > calls bound to the same destination into one stream to avoid overhead. > One problem on the internet is that many devices very tightly obey a > small MTU and even drop packets that exceed it. There are some ideas > floating around but determining the acceptable MTU all the way across > the internet is somewhat tricky. > > Assuming you can choose any MTU you want, trunking looks more > attractive, the max allowed size of a UDP packet of 64K can contain > several hundred calls even at PCMU. But this is unrealistic. We are > most likely limited to the standard MTU in the neighborhood of 1500 > and most guidelines suggest you only use a max percentage and you end > up with 1200 bytes per packet for payload data. This only allows room > for trunking 7 PCMU calls. > > Based on this conclusion it's obvious that only codecs that can > compress the audio better are even practical in trunking. G.729 for > instance, can hold dozens of calls since its very compressed. That > makes me feel to even bother making trunking, it should probably > revolve around some specific low bitrate codec. > > Then there is a matter of implementation. There are a few drafts on > how to do SIP/RTP trunking but none are formally adopted and new sip > drafts tend to be over engineered. I've had some ideas on it but the > more I think of it, it pushes me towards making a dedicated protocol > for it, and if I bother with that, I may as well make a full blown > protocol that does everything else I always wanted from VoIP. > > So every time I think about this issue i go in an endless circle and > end up just suggesting with Ken did and say use bigger ptimes between > the boxes in question. > > > > > > > On Mon, Jan 30, 2012 at 6:53 PM, Nowlin, Win wrote: > > Josue, > > > >> Is there any way to have a SIP Trunk. I mean to have for example 32 > > channels merged in one? or something like this? > >>When i try to find SIP trunk on internet i just see options for TDM > > gateway or similar but not really a multiplexed trunk. > > > > Let's define a SIP trunk for this discussion as "virtual > > internet connection" between your switch or gateway's IP address and > > your SIP provider's IP address. Over this connection your SIP provider > > can send you any number of simultaneous conversations (basically > > equivalent to "channels" or "time slots" in the TDM world). A SIP > > "trunk" can have as many simultaneous conversations ("Channels") as you > > wish, limited only by your internet bandwidth, how many "channels" you > > wish to pay for, and any limits set by your SIP provider. Also the SIP > > trunk can have as many different DID numbers as you wish or as limited > > by your provider. In this scenario, your provider is multiplexing the > > TDM sources into your SIP trunk. In its most basic form, a SIP provider > > can "point" the SIP traffic directly at your Freeswitch's external IP > > address and there you are!! Set up Freeswitch to be compatible with > > your Provider's requirements and you are there. > > > > As far as TDM gateways are concerned, if you are using legacy > > equipment that requires TDM service, there are several SIP-TDM gateways > > available that can receive the SIP traffic from your Provider and > > convert it back-and-forth between SIP and TDM. As far as the > > suitability and hardware requirements for Freeswitch to perform that > > function, since I am newly acquainted with Freeswitch, I leave that > > discussion to those who are experienced. > > > > Win N. > > > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > Sergey Okhapkin > > Sent: Saturday, January 28, 2012 12:14 > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] SIP Multiplexed Trunk as IAX2 or TDM > > > > SIP (and RTP) have no concept of trunking/audio frames multiplexing > > unlike > > IAX2 and TDM. > > > > On Saturday 28 January 2012 21:06:50 Josue Diaz Cruz wrote: > >> Is there any way to have a SIP Trunk. I mean to have for example 32 > > channels > >> merged in one? or something like this? When i try to find SIP trunk on > >> internet i just see options for TDM gateway or similar but not really > > a > >> multiplexed trunk. > >> > >> Can we do something with freeswitch? > >> > >> Josue Diaz Cruz > >> > >> Departamento Tecnico y Soporte > >> > >> jdiaz at coinfru.com > >> > >> > >> > >> C/ Balsicas 3 > >> > >> Alquerias | 30580 | Murcia > >> > >> www.coinfru.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 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > 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 > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kerem Erciyes - Sistem Danismani http://keremerciyes.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120202/46e6aefc/attachment.html From freeswitch at earthspike.net Thu Feb 2 16:49:54 2012 From: freeswitch at earthspike.net (John) Date: Thu, 02 Feb 2012 13:49:54 +0000 Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: References: <005801cce12e$05358300$0fa08900$@net.au> <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> Message-ID: <4F2A9482.1020100@earthspike.net> You do need to check that you can send an email outside of FS as FS does very little error reporting on emails. Basically, once the email is accepted for queueing, FS has lost interest. To test outside FS, try first a minimal email in a text file: --email.txt---8<------------------------- From: To: Date: Thu, 2 Feb 2012 13:13:40 +0100 Subject: Test 131340 (I always put a timestamp in the subject line when testing email due to the queueing and routing and stuff) Minimal test email 131340 ---------------8<--------------------------- Then send it from the command line with (on one line): # cat email.txt | sendmail -f freeswitch at mydomain.com testuser at mydomain.com They nearly all have a 'sendmail' alias even if they are not sendmail itself. Depending on the local mail server you are using, you may have to include the -t switch which extracts the To address(es) from the message itself. If you don't have a sendmail installed, you can choose from a wide variety, but I would recommend nullmailer or ssmtp if you only need to originate email. ssmtp has advantages over nullmailer if you need to connect to a MTA that requires authentication and/or SSL/TLS. Once that works, transfer your settings into conf/autoload_configs/switch.conf.xml eg (I am using ssmtp which barfs with -t but needs -f.) Note that if you need to run sendmail with no args, explicitly state them as as the default if not specified is to use "-t". Hope that helps. Maybe I should put this on the wiki... John On 02/02/12 12:13, Henrik Aagaard S?rensen wrote: > Perhaps stupid answer, but have you installed sendmail or similar? > > On Thu, Feb 2, 2012 at 12:58 PM, Rob Morin wrote: >> I think you're missing the email address: >> >> >> >> >> >> The isn't going to >> send any voicemails, just send a notification that there are voicemails. >> >> >> >> You might also need to look in your system log - there're problems with >> certain platforms sending emails. They'll show up in the system log as a >> segfault. >> >> >> >> Good luck, >> >> Rob >> >> >> >> >> >> >> >> From: DJB International [mailto:djbinter at gmail.com] >> Sent: Wednesday, February 01, 2012 11:06 PM >> To: FreeSWITCH Users Help >> Subject: Re: [Freeswitch-users] Fs ignoring voicemail to email >> >> >> >> Did you check your maillog? Try to see whether it reached your mail server. >> >> -djbinter >> >> On Wed, Feb 1, 2012 at 7:04 PM, Thomas Hoellriegel wrote: >> >> Hi all, >> I have setup a useraccount. But Fs don.t send my Voicemails to the >> emialadress. >> my useraccount settings is: >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> Fs ignore the maildelivery >> Can you help please? >> >> >> --------------- >> 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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > 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 Feb 2 19:46:57 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 2 Feb 2012 08:46:57 -0800 Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: <4F2A9482.1020100@earthspike.net> References: <005801cce12e$05358300$0fa08900$@net.au> <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> <4F2A9482.1020100@earthspike.net> Message-ID: > > Hope that helps. Maybe I should put this on the wiki... > No maybe about - definitely throw that up on the wiki! :) -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120202/81e45c2a/attachment.html From krice at freeswitch.org Thu Feb 2 20:37:26 2012 From: krice at freeswitch.org (Ken Rice) Date: Thu, 02 Feb 2012 11:37:26 -0600 Subject: [Freeswitch-users] Good News Everybody - As Professor Hubert J Farnsworth would say! Message-ID: Just a quick Update... BKW aka Brian West has made it out of the hospital in 1 piece... (except for the pieces the vampires^H^H^H^H^H^H^H errrrr doctors wanted to keep. He?s currently recuperating from the comforts of his own home. Prognosis is good at this time, but we do know there are going to be follow up visits on this... If you havent already done so please consider tossing in a few bucks to help out with his hospital stay. As most of you in the US knows, insurance is not a total life saver for a week long stay with the Drs and Nurses... Send any donations to brian at bkw.org and please mark them personal and drop him a note to get well so paypal doesn?t pull what paypal is known for... If you want to send something other then paypal I?m sure he would appreciate suite cases of cash or precious metals. Additional contact information for such items can be found on http://bkw.org And to Brian, Thanks for all the years of helping people in the OpenSource VoIP community. I for one wouldn?t be here today if it were not for your help all those years ago when I first discovered this type of software. Rest and Get Well my friend! K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120202/746a9fd0/attachment.html From sat at calgaryit.com Thu Feb 2 20:01:19 2012 From: sat at calgaryit.com (George Sapak) Date: Thu, 02 Feb 2012 10:01:19 -0700 Subject: [Freeswitch-users] Inbound Calls Caller ID Message-ID: <1328202079.2557.1.camel@george-desktop> On my inbound calls I get incoming call ID in this format , is there any way of stripping this out so that I end up with just numbers before the system starts routing the call? I have a few work arounds setup so that the call routs properly and I get the correct caller ID on my handsets, it would be nice not to have to use these. Thank You, George From msc at freeswitch.org Thu Feb 2 21:23:15 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 2 Feb 2012 10:23:15 -0800 Subject: [Freeswitch-users] Good News Everybody - As Professor Hubert J Farnsworth would say! In-Reply-To: References: Message-ID: Well said, Mr. Rice! -MC On Thu, Feb 2, 2012 at 9:37 AM, Ken Rice wrote: > Just a quick Update... > > BKW aka Brian West has made it out of the hospital in 1 piece... (except > for the pieces the vampires^H^H^H^H^H^H^H errrrr doctors wanted to keep. > > He?s currently recuperating from the comforts of his own home. > > Prognosis is good at this time, but we do know there are going to be > follow up visits on this... > > If you havent already done so please consider tossing in a few bucks to > help out with his hospital stay. As most of you in the US knows, insurance > is not a total life saver for a week long stay with the Drs and Nurses... > Send any donations to brian at bkw.org and please mark them personal and > drop him a note to get well so paypal doesn?t pull what paypal is known > for... > > If you want to send something other then paypal I?m sure he would > appreciate suite cases of cash or precious metals. Additional contact > information for such items can be found on http://bkw.org > > And to Brian, Thanks for all the years of helping people in the OpenSource > VoIP community. I for one wouldn?t be here today if it were not for your > help all those years ago when I first discovered this type of software. > Rest and Get Well my friend! > > K > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120202/4f07dcb7/attachment.html From freeswitch at earthspike.net Thu Feb 2 21:32:10 2012 From: freeswitch at earthspike.net (John) Date: Thu, 02 Feb 2012 18:32:10 +0000 Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: References: <005801cce12e$05358300$0fa08900$@net.au> <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> <4F2A9482.1020100@earthspike.net> Message-ID: <4F2AD6AA.1060003@earthspike.net> On 02/02/12 16:46, Michael Collins wrote: > > > Hope that helps. Maybe I should put this on the wiki... > > > No maybe about - definitely throw that up on the wiki! :) > -MC New sections in mod_voicemail wiki page on sSMTP and debugging external mailers. Open to constructive criticism... John -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120202/01cd5085/attachment.html From djbinter at gmail.com Thu Feb 2 22:08:57 2012 From: djbinter at gmail.com (DJB International) Date: Thu, 2 Feb 2012 11:08:57 -0800 Subject: [Freeswitch-users] Inbound Calls Caller ID In-Reply-To: <1328202079.2557.1.camel@george-desktop> References: <1328202079.2557.1.camel@george-desktop> Message-ID: Check on "sip_invite_from_params" -djbinter On Thu, Feb 2, 2012 at 9:01 AM, George Sapak wrote: > On my inbound calls I get incoming call ID in this format > , is there any way of stripping this > out so that I end up with just numbers before the system starts routing > the call? I have a few work arounds setup so that the call routs > properly and I get the correct caller ID on my handsets, it would be > nice not to have to use these. > > Thank You, George > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120202/ba18c3ce/attachment.html From elliott at zoogmedia.com Thu Feb 2 23:43:43 2012 From: elliott at zoogmedia.com (Elliott Vogel) Date: Thu, 2 Feb 2012 20:43:43 +0000 Subject: [Freeswitch-users] Sending message Message-ID: <461821C20115054F9056ED0E52EAC4DAE51A8F@BY2PRD0710MB378.namprd07.prod.outlook.com> Is there a way to send an im to a sip user using the prompt? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120202/af2c0528/attachment-0001.html From brian at freeswitch.org Fri Feb 3 00:05:18 2012 From: brian at freeswitch.org (Brian West) Date: Thu, 2 Feb 2012 15:05:18 -0600 Subject: [Freeswitch-users] Good News Everybody - As Professor Hubert J Farnsworth would say! In-Reply-To: References: Message-ID: Ken, Thank you. I would like to thank Tayeb Meftah, Bogos Dan-Cristian, Seven Du, Yehavi Bourvine, Michael Kennedy, YGNetworking ,Carlo Dimaggio, Russ Herrold, Norman Tomlins, Kevin Berry, Jimmy Godbout, Internet Infinity Inc and Massimiliano Ravelli for their donations to help with the expenses associated with an extended hospital visit like this. Hey at least my heart can take the sticker shock now! :P I'll be back to work Monday. Thank you all. /b On Feb 2, 2012, at 11:37 AM, Ken Rice wrote: > > BKW aka Brian West has made it out of the hospital in 1 piece... (except for > the pieces the vampires^H^H^H^H^H^H^H errrrr doctors wanted to keep. > > He?s currently recuperating from the comforts of his own home. > > Prognosis is good at this time, but we do know there are going to be follow > up visits on this... > > If you havent already done so please consider tossing in a few bucks to help > out with his hospital stay. As most of you in the US knows, insurance is not > a total life saver for a week long stay with the Drs and Nurses... Send any > donations to brian at bkw.org and please mark them personal and drop him a note > to get well so paypal doesn?t pull what paypal is known for... > > If you want to send something other then paypal I?m sure he would appreciate > suite cases of cash or precious metals. Additional contact information for > such items can be found on http://bkw.org > > And to Brian, Thanks for all the years of helping people in the OpenSource > VoIP community. I for one wouldn?t be here today if it were not for your > help all those years ago when I first discovered this type of software. Rest > and Get Well my friend! > > K -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org From freeswitch at earthspike.net Fri Feb 3 00:20:31 2012 From: freeswitch at earthspike.net (John) Date: Thu, 02 Feb 2012 21:20:31 +0000 Subject: [Freeswitch-users] Sending message In-Reply-To: <461821C20115054F9056ED0E52EAC4DAE51A8F@BY2PRD0710MB378.namprd07.prod.outlook.com> References: <461821C20115054F9056ED0E52EAC4DAE51A8F@BY2PRD0710MB378.namprd07.prod.outlook.com> Message-ID: <4F2AFE1F.9000100@earthspike.net> chat sip|||message eg chat sip|1001 at earthspike.net|1002 at earthspike.net|Can you hear me? http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_chat John On 02/02/12 20:43, Elliott Vogel wrote: > Is there a way to send an im to a sip user using the prompt? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120202/2192d679/attachment.html From b2m at a-cti.com Thu Feb 2 22:06:21 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Fri, 3 Feb 2012 00:36:21 +0530 Subject: [Freeswitch-users] help : CDR variables not reachable to dialplan/Lua script In-Reply-To: References: Message-ID: Hi team, I am trying to get these values from the lua script for custom CDR report, I am able to get few values but many variables are missing or getting null value. Any help would be appreciated. local aa=session:getVariable("context"); local bb=session:getVariable("destination_number"); local cc=session:getVariable("caller_id_name"); local dd=session:getVariable("caller_id_number"); local ee=session:getVariable("network_addr"); local ff=session:getVariable("ani"); local ii=session:getVariable("source"); local jj=session:getVariable("chan_name"); local kk=session:getVariable("uuid"); local ll=session:getVariable("created_time"); local aa1=session:getVariable("direction"); local bb1=session:getVariable("username"); local cc1=session:getVariable("dialplan"); local dd1=session:getVariable("caller_id_number"); local ee1=session:getVariable("unique_id"); session:execute("bridge","user/601"); session:hangup(); local ff1=session:getVariable("answered_time"); local hh1=session:getVariable("created_time"); local gg1=session:getVariable("hangup_time"); local gg2=session:getVariable("end_stamp"); I am getting values for all the above variables except end_stamp, hangup_time, duration, billsec,hangup_cause. Thanks for your time. Thanks, Bala -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120203/6d247357/attachment.html From msc at freeswitch.org Fri Feb 3 00:30:08 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 2 Feb 2012 13:30:08 -0800 Subject: [Freeswitch-users] help : CDR variables not reachable to dialplan/Lua script In-Reply-To: References: Message-ID: Okay, let's try again: http://wiki.freeswitch.org/wiki/Lua#Special_Case:_env_object -MC On Thu, Feb 2, 2012 at 11:06 AM, Balamurugan Mahendran wrote: > Hi team, > > I am trying to get these values from the lua script for custom CDR report, > I am able to get few values but many variables are missing or getting null > value. Any help would be appreciated. > > local aa=session:getVariable("context"); > local bb=session:getVariable("destination_number"); > local cc=session:getVariable("caller_id_name"); > local dd=session:getVariable("caller_id_number"); > local ee=session:getVariable("network_addr"); > local ff=session:getVariable("ani"); > local ii=session:getVariable("source"); > local jj=session:getVariable("chan_name"); > local kk=session:getVariable("uuid"); > local ll=session:getVariable("created_time"); > local aa1=session:getVariable("direction"); > local bb1=session:getVariable("username"); > local cc1=session:getVariable("dialplan"); > local dd1=session:getVariable("caller_id_number"); > local ee1=session:getVariable("unique_id"); > > session:execute("bridge","user/601"); > session:hangup(); > > local ff1=session:getVariable("answered_time"); > local hh1=session:getVariable("created_time"); > local gg1=session:getVariable("hangup_time"); > local gg2=session:getVariable("end_stamp"); > > I am getting values for all the above variables except end_stamp, > hangup_time, duration, billsec,hangup_cause. > > > Thanks for your time. > > Thanks, > Bala > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120202/f10980a5/attachment.html From trever.adams at gmail.com Fri Feb 3 03:16:36 2012 From: trever.adams at gmail.com (Trever L. Adams) Date: Thu, 02 Feb 2012 17:16:36 -0700 Subject: [Freeswitch-users] FreeTDM problems In-Reply-To: References: Message-ID: <4F2B2764.2010900@gmail.com> On 02/01/2012 04:50 PM, Brian Cuthie wrote: > Hi, > > When I call an FXS port it keeps generating ringing for quite some time after the caller hangs up. I don't know a whole lot about FreeTDM at this point, but I did notice that in channel_on_hangup() (mod_freetdm.c:591) the call to ftdm_channel_call_hangup() is being skipped. Is that normal? > > Thanks > > -brian This is a bug (http://jira.freeswitch.org/browse/OPENZAP-173) which has kept me from deploying FreeSWITCH. Please add anything you think is relevant to the bug. Thank you for any info you add, Trever -------------- next part -------------- A non-text attachment was scrubbed... Name: signature.asc Type: application/pgp-signature Size: 262 bytes Desc: OpenPGP digital signature Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120202/47af1bcb/attachment.bin From sat at calgaryit.com Fri Feb 3 04:29:49 2012 From: sat at calgaryit.com (George Sapak) Date: Thu, 2 Feb 2012 18:29:49 -0700 (MST) Subject: [Freeswitch-users] Inbound Calls Caller ID In-Reply-To: Message-ID: <1547590980.1285.1328232589312.JavaMail.root@server3> This -> "sip_invite_from_params" is pretty under documented, here is the initial SIP conversation: INVITE sip:403yyyyyyy;phone-context=national at 10.185.16.170;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.185.16.169:5060;branch=z9hG4bKti89c3kqffttnvqrom790lopv7 From: ;tag=SDhcnk501-80466 To: Call-ID: SDhcnk501-1fd6435299a4405392f0b06a40d518fa-o0t3g30 CSeq: 906 INVITE Content-Type: application/sdp Contact: User-Agent: Nortel SESM 14.0.6.0 Max-Forwards: 19 Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,com.nortelnetworks.im.encryption,replaces,100rel,tdialog Allow: UPDATE,REFER x-nt-corr-id: 9b0cf0d0-29e6-1b21-a632-000e0cb7d3a0 x-nt-location: -1 Content-Length: 201 Route: just need this removed -> ;phone-context=national Thank You, George. ----- Original Message ----- From: "DJB International" To: "FreeSWITCH Users Help" Sent: Thursday, February 2, 2012 12:08:57 PM Subject: Re: [Freeswitch-users] Inbound Calls Caller ID Check on "sip_invite_from_params" -djbinter On Thu, Feb 2, 2012 at 9:01 AM, George Sapak < sat at calgaryit.com > wrote: On my inbound calls I get incoming call ID in this format , is there any way of stripping this out so that I end up with just numbers before the system starts routing the call? I have a few work arounds setup so that the call routs properly and I get the correct caller ID on my handsets, it would be nice not to have to use these. Thank You, George _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 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 Fri Feb 3 05:30:02 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Fri, 3 Feb 2012 03:30:02 +0100 (CET) Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: References: <005801cce12e$05358300$0fa08900$@net.au> <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> Message-ID: Am 02.02.12 um 13:13 schrieb Henrik Aagaard S?rensen: > Perhaps stupid answer, but have you installed sendmail or similar? yes, i install postfix. Postfix is working, but i have not action in my logfile "/var/log/mail.log" I using debian squeeze And my Mailadress is defined in the directoryuse: > From admin at blindi.net Fri Feb 3 06:13:34 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Fri, 3 Feb 2012 04:13:34 +0100 (CET) Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: <4F2A9482.1020100@earthspike.net> References: <005801cce12e$05358300$0fa08900$@net.au> <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> <4F2A9482.1020100@earthspike.net> Message-ID: Am 02.02.12 um 13:49 schrieb John: > You do need to check that you can send an email outside of FS as FS does > very little error reporting on emails. Basically, once the email is Thanks, yes i can send mails outside from fs. Postfix log this acton under: /var/log/mail.log. > Once that works, transfer your settings into > conf/autoload_configs/switch.conf.xml Maildelivery don.t works. I don.t understand this error. Postfix give no actionlog, the The voicemail did.nt send to my adress. why is the problem please? Thanks. > --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From lists at redbonez.net Fri Feb 3 03:34:08 2012 From: lists at redbonez.net (Adam Ford) Date: Thu, 2 Feb 2012 17:34:08 -0700 Subject: [Freeswitch-users] FreeTDM [MANDATORY_IE_MISSING] In-Reply-To: <4F10567B.4060601@freeswitch.org> References: <1026601ccce6e$a1b61190$e52234b0$@redbonez.net> <017001ccd195$f1d9b210$d58d1630$@redbonez.net> <4F10567B.4060601@freeswitch.org> Message-ID: <02e301cce20b$8cfb9f80$a6f2de80$@redbonez.net> Thank you so much for your response. I have confirmed this patch has fixed the issue, and FreeTDM is now working flawlessly with libpri + DAHDI + Redfone foneBridge2. -Adam -----Original Message----- From: Stefan Knoblich [mailto:stkn at freeswitch.org] Sent: Friday, January 13, 2012 9:06 AM To: FreeSWITCH Users Help Cc: Adam Ford Subject: Re: [Freeswitch-users] FreeTDM [MANDATORY_IE_MISSING] On 13.01.2012 02:51, Adam Ford wrote: > I am still having issues with MANDATORY_IE_MISSING on incoming calls > when using FreeTDM + libpri + DAHDI + foneBridge2. Can anyone help me > figure out if this is a configuration issue or a bug/incompatibility? > > As stated before, I am running the latest git trunk with all > default/stock settings with the exception of FreeTDM configuration and > minor modification to the dialplan to pass my DID to the default > extension 1001. Outgoing calls are working great. > Try the attached patch, that should fix it (and hopefully doesn't have any side effects). -- ---------------------------------------------------------------------------- --- Stefan Knoblich | Web: http://www.axsentis.de/ axsentis GmbH | http://oss.axsentis.de/ Eupener Str. 74, 50933 Koeln, Germany | Amtsgericht Koeln: HR B 56238 | Email: s.knoblich at axsentis.de UST-ID: DE244977565 | JID: s.knoblich at jabber.axsentis.de ---------------------------------------------------------------------------- --- Web: http://stkn.techmage.de/ Email: stkn at freeswitch.org IRC: #freeswitch-de @ irc.freenode.net From freeswitch at earthspike.net Fri Feb 3 10:22:57 2012 From: freeswitch at earthspike.net (John) Date: Fri, 03 Feb 2012 07:22:57 +0000 Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: References: <005801cce12e$05358300$0fa08900$@net.au> <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> <4F2A9482.1020100@earthspike.net> Message-ID: <4F2B8B51.2080506@earthspike.net> Are you using the latest git? There was a bug a few weeks ago for a short while where temporary files were being misnamed and email from FS failed for those FS installations where bin/freeswitch was running as a non-root user. John On 03/02/12 03:13, Thomas Hoellriegel wrote: > Am 02.02.12 um 13:49 schrieb John: > >> You do need to check that you can send an email outside of FS as FS does >> very little error reporting on emails. Basically, once the email is > > Thanks, yes i can send mails outside from fs. > > Postfix log this acton under: /var/log/mail.log. > >> Once that works, transfer your settings into >> conf/autoload_configs/switch.conf.xml > > Maildelivery don.t works. > I don.t understand this error. > Postfix give no actionlog, the The voicemail did.nt send to my adress. > why is the problem 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120203/dc3832b2/attachment.html From djbinter at gmail.com Fri Feb 3 10:40:32 2012 From: djbinter at gmail.com (DJB International) Date: Thu, 2 Feb 2012 23:40:32 -0800 Subject: [Freeswitch-users] Inbound Calls Caller ID In-Reply-To: <1547590980.1285.1328232589312.JavaMail.root@server3> References: <1547590980.1285.1328232589312.JavaMail.root@server3> Message-ID: Try to unset the variable. -djbinter On Thu, Feb 2, 2012 at 5:29 PM, George Sapak wrote: > This -> "sip_invite_from_params" is pretty under documented, here is the > initial SIP conversation: > > INVITE sip:403yyyyyyy;phone-context=national at 10.185.16.170;user=phone > SIP/2.0 > Via: SIP/2.0/UDP 10.185.16.169:5060 > ;branch=z9hG4bKti89c3kqffttnvqrom790lopv7 > From: ;user=phone;isup-oli=00>;tag=SDhcnk501-80466 > To: > Call-ID: SDhcnk501-1fd6435299a4405392f0b06a40d518fa-o0t3g30 > CSeq: 906 INVITE > Content-Type: application/sdp > Contact: ;transport=udp> > User-Agent: Nortel SESM 14.0.6.0 > Max-Forwards: 19 > Supported: > com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,com.nortelnetworks.im.encryption,replaces,100rel,tdialog > Allow: UPDATE,REFER > x-nt-corr-id: 9b0cf0d0-29e6-1b21-a632-000e0cb7d3a0 > x-nt-location: -1 > Content-Length: 201 > Route: ;user=phone;lr> > > just need this removed -> ;phone-context=national > > Thank You, George. > > ----- Original Message ----- > From: "DJB International" > To: "FreeSWITCH Users Help" > Sent: Thursday, February 2, 2012 12:08:57 PM > Subject: Re: [Freeswitch-users] Inbound Calls Caller ID > > > Check on "sip_invite_from_params" > > -djbinter > > > > On Thu, Feb 2, 2012 at 9:01 AM, George Sapak < sat at calgaryit.com > wrote: > > > On my inbound calls I get incoming call ID in this format > , is there any way of stripping this > out so that I end up with just numbers before the system starts routing > the call? I have a few work arounds setup so that the call routs > properly and I get the correct caller ID on my handsets, it would be > nice not to have to use these. > > Thank You, George > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120202/44630abe/attachment-0001.html From spencer at 5ninesolutions.com Fri Feb 3 13:01:18 2012 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Fri, 3 Feb 2012 02:01:18 -0800 Subject: [Freeswitch-users] Cannot Send Display Update Warning Message-ID: <91EC5E74-6ED1-43EB-A62C-FFF65FD163FE@5ninesolutions.com> Hello, On our SBCs which consist of FreeSWITCH behind Kamailio I'm seeing the following log entry on every channel. What could be the cause and is it something to worry about? This occurs when the call receives a 183: 2012-02-03 09:53:51.966214 [WARNING] mod_sofia.c:2102 Cannot send display update to sofia/outbound/@x.x.x.x Did not receive reply to last update or channel has not been answered yet. I've looked at traces and it seems that every message send is actually passed. Both Kamailio and FreeSWITCH use a single server with a public ip, just different ports. Thanks, Spencer From rmorin at blie-ent.com Fri Feb 3 15:22:55 2012 From: rmorin at blie-ent.com (Rob Morin) Date: Fri, 3 Feb 2012 07:22:55 -0500 Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: References: <005801cce12e$05358300$0fa08900$@net.au> <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> <4F2A9482.1020100@earthspike.net> Message-ID: <04b101cce26e$8f7a5220$ae6ef660$@blie-ent.com> As I suggested, the voicemail itself won't be sent to the address provided by . You have to use . That said, you *should* be receiving text messages because of the "vm-notify-mailto" parameter, which apparently you aren't. In your conf/autoload_configs/switch.conf.xml file, what do you have for You do need to check that you can send an email outside of FS as FS > does very little error reporting on emails. Basically, once the email > is Thanks, yes i can send mails outside from fs. Postfix log this acton under: /var/log/mail.log. > Once that works, transfer your settings into > conf/autoload_configs/switch.conf.xml Maildelivery don.t works. I don.t understand this error. Postfix give no actionlog, the The voicemail did.nt send to my adress. why is the problem 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 georg at riseup.net Fri Feb 3 18:52:11 2012 From: georg at riseup.net (georg at riseup.net) Date: Fri, 3 Feb 2012 16:52:11 +0100 Subject: [Freeswitch-users] Questions regarding mod_fifo and mod_callcenter In-Reply-To: References: Message-ID: > I would like to implement a quite simple queue, but I don't know if > mod_fifo is able to handle this. Would be great if someone could clarify > this: > > - One phone number, two phones > - If both or one is idle, call should be routed directly to the phone(s), > no moh should be played (just ringtones) > - if both in use, call should be routed into a queue, caller should hear > moh > - if after this a phone gets idle, call should be routed to the phone, moh > should stopped and ringtones be played to the caller > - if a call comes from the queue, and both phones are idle, both phones > should ring at the same time (I tried this with mod_fifo, is this > possible?) Isn't this possible? Thanks, Georg From msc at freeswitch.org Fri Feb 3 19:15:49 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 3 Feb 2012 08:15:49 -0800 Subject: [Freeswitch-users] Questions regarding mod_fifo and mod_callcenter In-Reply-To: References: Message-ID: On Fri, Feb 3, 2012 at 7:52 AM, wrote: > > I would like to implement a quite simple queue, but I don't know if > > mod_fifo is able to handle this. Would be great if someone could clarify > > this: > > > > - One phone number, two phones > > - If both or one is idle, call should be routed directly to the phone(s), > > no moh should be played (just ringtones) > > - if both in use, call should be routed into a queue, caller should hear > > moh > > - if after this a phone gets idle, call should be routed to the phone, > moh > > should stopped and ringtones be played to the caller > > - if a call comes from the queue, and both phones are idle, both phones > > should ring at the same time (I tried this with mod_fifo, is this > > possible?) > > Isn't this possible? > I'm pretty sure that this is how mod_callcenter works - like a traditional ACD. In mod_fifo it generally doesn't play ringing tones to the caller while it's looking for an agent. If anyone knows how to have mod_fifo mimic ACD by letting the caller hear ringback while dialing the agent's phone please let me know and I will add it to the wiki. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120203/e9656a19/attachment.html From Stefan.Weigel at allianz-warranty.com Fri Feb 3 19:58:42 2012 From: Stefan.Weigel at allianz-warranty.com (Weigel, Stefan) Date: Fri, 3 Feb 2012 17:58:42 +0100 Subject: [Freeswitch-users] Antw.: Questions regarding mod_fifo and mod_callcenter Message-ID: <2F9C2AB4-7468-4837-99C6-226CC872E515@allianz-warranty.com> Hi, I have the same request regarding the ringtones. With mod_callcenter it's not possible to play ringing tones when acd tries to ring an agent. Am I wrong? Stefan ----- Reply message ----- Von: "Michael Collins" An: "FreeSWITCH Users Help" Betreff: [Freeswitch-users] Questions regarding mod_fifo and mod_callcenter Datum: Fr., Feb. 3, 2012 17:17 On Fri, Feb 3, 2012 at 7:52 AM, > wrote: > I would like to implement a quite simple queue, but I don't know if > mod_fifo is able to handle this. Would be great if someone could clarify > this: > > - One phone number, two phones > - If both or one is idle, call should be routed directly to the phone(s), > no moh should be played (just ringtones) > - if both in use, call should be routed into a queue, caller should hear > moh > - if after this a phone gets idle, call should be routed to the phone, moh > should stopped and ringtones be played to the caller > - if a call comes from the queue, and both phones are idle, both phones > should ring at the same time (I tried this with mod_fifo, is this > possible?) Isn't this possible? I'm pretty sure that this is how mod_callcenter works - like a traditional ACD. In mod_fifo it generally doesn't play ringing tones to the caller while it's looking for an agent. If anyone knows how to have mod_fifo mimic ACD by letting the caller hear ringback while dialing the agent's phone please let me know and I will add it to the wiki. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120203/3fbc8f1e/attachment.html From anthony.minessale at gmail.com Fri Feb 3 20:50:11 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 3 Feb 2012 11:50:11 -0600 Subject: [Freeswitch-users] Cannot Send Display Update Warning In-Reply-To: <91EC5E74-6ED1-43EB-A62C-FFF65FD163FE@5ninesolutions.com> References: <91EC5E74-6ED1-43EB-A62C-FFF65FD163FE@5ninesolutions.com> Message-ID: That's a harmless error, i'm taking it out. On Fri, Feb 3, 2012 at 4:01 AM, Spencer Thomason wrote: > Hello, > On our SBCs which consist of FreeSWITCH behind Kamailio I'm seeing the following log entry on every channel. ?What could be the cause and is it something to worry about? > > This occurs when the call receives a 183: > 2012-02-03 09:53:51.966214 [WARNING] mod_sofia.c:2102 Cannot send display update to sofia/outbound/@x.x.x.x Did not receive reply to last update or channel has not been answered yet. > > I've looked at traces and it seems that every message send is actually passed. ?Both Kamailio and FreeSWITCH use a single server with a public ip, just different ports. > > > Thanks, > Spencer > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 admin at blindi.net Fri Feb 3 21:00:39 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Fri, 3 Feb 2012 19:00:39 +0100 (CET) Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: <4F2B8B51.2080506@earthspike.net> References: <005801cce12e$05358300$0fa08900$@net.au> <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> <4F2A9482.1020100@earthspike.net> <4F2B8B51.2080506@earthspike.net> Message-ID: Hi John, thanks for your nice help. I have update the gitversion. And i have all the user re-edited. The uservariables are now: my settings in: /usr/local/freeswitch/conf/autoload_configs/switch.conf.xml Its 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 Fri Feb 3 21:10:22 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Fri, 3 Feb 2012 19:10:22 +0100 (CET) Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: <04b101cce26e$8f7a5220$ae6ef660$@blie-ent.com> References: <005801cce12e$05358300$0fa08900$@net.au> <035101cc <04b101cce26e$8f7a5220$ae6ef660$@blie-ent.com> Message-ID: Hi rob, thanks for your nice Message Unfortunately I did not know the difference. --------------- 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 gerry at pstn2.net Fri Feb 3 21:15:48 2012 From: gerry at pstn2.net (Gerry Hull) Date: Fri, 3 Feb 2012 13:15:48 -0500 Subject: [Freeswitch-users] extension-in-contact setting does not seem to work Message-ID: Hello, We have extension-in-contact is set to true in our gateway, but our SIP register still shows contact using the gateway name. ( sip:gw+nortelcs1k at 10.98.0.50:5080) Any help is appreciated... stuck on this one. -Gerry 1. 2. 3. 4. 5. 6. 7. 8. 9. 10. 11. 12. 13. 14. 15. 16. 17. 18. 23. 24. 25. 26. 27. 28. 29. 30. SIP PACKET 31. REGISTER sip:sands.com;transport=udp SIP/2.0 32. 33. Via: SIP/2.0/UDP 10.98.0.50:5080;rport;branch=z9hG4bKBKK5SyQ8y5yeQ 34. 35. Max-Forwards: 70 36. 37. From: ;tag=BKKN0U8BNX50a 38. 39. To: 40. 41. Call-ID: 5915f7b9-df9d-4216-a4c6-bec38807c437 42. 43. CSeq: 23809179 REGISTER 44. 45. Contact: < sip:gw+nortelcs1k at 10.98.0.50:5080;tport=udp;transport=udp;gw=nortelcs1k> 46. 47. Expires: 3600 48. 49. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-c2ac8f5 2012-01-27 16-04-54 -0600 50. 51. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY 52. 53. Supported: timer, precondition, path, replaces 54. 55. Content-Length: 0 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120203/70ce04e8/attachment.html From brad at tech21.com Fri Feb 3 21:36:16 2012 From: brad at tech21.com (Brad Mina) Date: Fri, 3 Feb 2012 10:36:16 -0800 Subject: [Freeswitch-users] FreeSwtich DB In-Reply-To: References: Message-ID: Check your odbc database and make sure FS has created (and uses) that database. If it's creating db files, it's probably not using your odbc db. On Thu, Feb 2, 2012 at 12:07 AM, Joe Flemmings wrote: > Is their a reason why freeswitch still creates DB files in > /usr/local/freeswitch/db/ even when using ODBC at the core as decsribed > here http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120203/196c3b75/attachment.html From darcy at primrose.ws Fri Feb 3 19:28:19 2012 From: darcy at primrose.ws (Darcy) Date: Fri, 3 Feb 2012 11:28:19 -0500 Subject: [Freeswitch-users] using public IP and Port versus sdp port Message-ID: <5418E766588F49C7958EE4207AB74A83@DWP> I have a server sitting on a public IP and all endpoints are behind routers. In some cases I get rtp where the sdp port differs from the received port. The freeswitch picks up the public ip to route to but uses the sdp port to route the rtp back to versus the public port, anyone have any suggestions. I believe I corrected this on another server awhile back but cannot remember the setting. Thanks Darcy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120203/a8fbafb0/attachment.html From msc at freeswitch.org Fri Feb 3 21:48:07 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 3 Feb 2012 10:48:07 -0800 Subject: [Freeswitch-users] extension-in-contact setting does not seem to work In-Reply-To: References: Message-ID: Gerry, I'm no expert, but it looks like your "extension" param in your gateway is commented out, so the sip profile doesn't know what value to put in the Contact header and just falls back to default behavior. I would uncomment the extension param line (see line #21 of your listing) and restart the gateway. -MC On Fri, Feb 3, 2012 at 10:15 AM, Gerry Hull wrote: > Hello, > > > We have extension-in-contact is set to true in our gateway, but our SIP > register still shows contact using the gateway name. ( > sip:gw+nortelcs1k at 10.98.0.50:5080) > > > Any help is appreciated... stuck on this one. > > > -Gerry > > 1. > 2. > 3. > 4. > 5. > 6. > 7. > 8. > 9. > 10. > 11. > 12. > 13. > 14. > 15. > 16. > 17. > 18. > 23. > 24. > 25. > 26. > 27. > 28. > 29. > 30. SIP PACKET > 31. REGISTER sip:sands.com;transport=udp SIP/2.0 > 32. > 33. Via: SIP/2.0/UDP 10.98.0.50:5080;rport;branch=z9hG4bKBKK5SyQ8y5yeQ > 34. > 35. Max-Forwards: 70 > 36. > 37. From: ;tag=BKKN0U8BNX50a > 38. > 39. To: > 40. > 41. Call-ID: 5915f7b9-df9d-4216-a4c6-bec38807c437 > 42. > 43. CSeq: 23809179 REGISTER > 44. > 45. Contact: < > sip:gw+nortelcs1k at 10.98.0.50:5080;tport=udp;transport=udp;gw=nortelcs1k > > > 46. > 47. Expires: 3600 > 48. > 49. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-c2ac8f5 2012-01-27 > 16-04-54 -0600 > 50. > 51. Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > 52. > 53. Supported: timer, precondition, path, replaces > 54. > 55. 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120203/bc3dd4cb/attachment-0001.html From msc at freeswitch.org Fri Feb 3 21:51:41 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 3 Feb 2012 10:51:41 -0800 Subject: [Freeswitch-users] using public IP and Port versus sdp port In-Reply-To: <5418E766588F49C7958EE4207AB74A83@DWP> References: <5418E766588F49C7958EE4207AB74A83@DWP> Message-ID: Have made sure that you disabled any SIP ALG's in the routers? -MC On Fri, Feb 3, 2012 at 8:28 AM, Darcy wrote: > I have a server sitting on a public IP and all endpoints are behind > routers. In some cases I get rtp where the sdp port differs from the > received port. The freeswitch picks up the public ip to route to but uses > the sdp port to route the rtp back to versus the public port, anyone have > any suggestions. I believe I corrected this on another server awhile back > but cannot remember the setting. > > Thanks > Darcy > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120203/81d47997/attachment.html From freeswitch at earthspike.net Fri Feb 3 22:01:14 2012 From: freeswitch at earthspike.net (John) Date: Fri, 03 Feb 2012 19:01:14 +0000 Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: References: <005801cce12e$05358300$0fa08900$@net.au> <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> <4F2A9482.1020100@earthspike.net> <4F2B8B51.2080506@earthspike.net> Message-ID: <4F2C2EFA.9030105@earthspike.net> On 03/02/12 18:00, Thomas Hoellriegel wrote: > > Its works fine!! Good news! From curriegrad2004 at gmail.com Fri Feb 3 23:19:16 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 3 Feb 2012 12:19:16 -0800 Subject: [Freeswitch-users] FreeSwtich DB In-Reply-To: References: Message-ID: There are some components in FreeSWITCH that doesn't use the ODBC stuff yet. On Fri, Feb 3, 2012 at 10:36 AM, Brad Mina wrote: > Check your odbc database and make sure FS has created (and uses) that > database. If it's creating db files, it's probably not using your odbc db. > > On Thu, Feb 2, 2012 at 12:07 AM, Joe Flemmings > wrote: >> >> Is their a reason why freeswitch still creates DB files in >> /usr/local/freeswitch/db/ even when using ODBC at the core as decsribed here >> http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From darcy at primrose.ws Fri Feb 3 23:55:05 2012 From: darcy at primrose.ws (Darcy) Date: Fri, 3 Feb 2012 15:55:05 -0500 Subject: [Freeswitch-users] using public IP and Port versus sdp port In-Reply-To: References: <5418E766588F49C7958EE4207AB74A83@DWP> Message-ID: <444E8FB96B7145849BF57F614609D129@DWP> The router is a sonic wall, I always run into this problem with them for some reason, On another freeswitch, I have this working, something to do with ndlb I just can?t seem to grasp it. Darcy From: Michael Collins Sent: Friday, February 03, 2012 1:51 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] using public IP and Port versus sdp port Have made sure that you disabled any SIP ALG's in the routers? -MC On Fri, Feb 3, 2012 at 8:28 AM, Darcy wrote: I have a server sitting on a public IP and all endpoints are behind routers. In some cases I get rtp where the sdp port differs from the received port. The freeswitch picks up the public ip to route to but uses the sdp port to route the rtp back to versus the public port, anyone have any suggestions. I believe I corrected this on another server awhile back but cannot remember the setting. Thanks Darcy _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120203/204af36c/attachment.html From krice at freeswitch.org Sat Feb 4 00:02:09 2012 From: krice at freeswitch.org (Ken Rice) Date: Fri, 03 Feb 2012 15:02:09 -0600 Subject: [Freeswitch-users] using public IP and Port versus sdp port In-Reply-To: <444E8FB96B7145849BF57F614609D129@DWP> Message-ID: Probably NDLB-force-rport also... You already identified your problem its the sonic wall... Those things cause more problems with it comes to people running VoIP.... K On 2/3/12 2:55 PM, "Darcy" wrote: > The router is a sonic wall, I always run into this problem with them for some > reason, On another freeswitch, I have this working, something to do with ndlb > I just can?t seem to grasp it. > > Darcy > > From: Michael Collins > Sent: Friday, February 03, 2012 1:51 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] using public IP and Port versus sdp port > > Have made sure that you disabled any SIP ALG's in the routers? > -MC > > On Fri, Feb 3, 2012 at 8:28 AM, Darcy wrote: >> >> >> >> >> I have a server sitting on a public IP and all endpoints are behind routers. >> In some cases I get rtp where the sdp port differs from the received port. >> The freeswitch picks up the public ip to route to but uses the sdp port to >> route the rtp back to versus the public port, anyone have any suggestions. >> I believe I corrected this on another server awhile back but cannot remember >> the setting. >> >> >> >> Thanks >> >> Darcy >> >> _________________________________________________________________________ >> 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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120203/ff97cf88/attachment-0001.html From darcy at primrose.ws Sat Feb 4 00:17:18 2012 From: darcy at primrose.ws (Darcy) Date: Fri, 3 Feb 2012 16:17:18 -0500 Subject: [Freeswitch-users] using public IP and Port versus sdp port In-Reply-To: References: Message-ID: <7D3AE1D542C149CCB7F52DB4DB000C2C@DWP> Re: [Freeswitch-users] using public IP and Port versus sdp portI agree with you, I am a big fan of dd-wrt software versus sonic wall, usually push people in that direction, it is not the NDLB-force-rport however, I already set that up, I am thinking it is something different in their sonic wall settings versus the other sites, my issue is they are using msfax with the voip plug in, so I can?t move them to my dsl, I am going to get the sonic wall guys to play with it for a bit, thanks for the assistance. From: Ken Rice Sent: Friday, February 03, 2012 4:02 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] using public IP and Port versus sdp port Probably NDLB-force-rport also... You already identified your problem its the sonic wall... Those things cause more problems with it comes to people running VoIP.... K On 2/3/12 2:55 PM, "Darcy" wrote: The router is a sonic wall, I always run into this problem with them for some reason, On another freeswitch, I have this working, something to do with ndlb I just can?t seem to grasp it. Darcy From: Michael Collins Sent: Friday, February 03, 2012 1:51 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] using public IP and Port versus sdp port Have made sure that you disabled any SIP ALG's in the routers? -MC On Fri, Feb 3, 2012 at 8:28 AM, Darcy wrote: I have a server sitting on a public IP and all endpoints are behind routers. In some cases I get rtp where the sdp port differs from the received port. The freeswitch picks up the public ip to route to but uses the sdp port to route the rtp back to versus the public port, anyone have any suggestions. I believe I corrected this on another server awhile back but cannot remember the setting. Thanks Darcy _________________________________________________________________________ Professional FreeSWITCH Consulting Services: wlmailhtml: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 FreeSWITCH-users mailing list wlmailhtml:FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: wlmailhtml: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 wlmailhtml:FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ------------------------------------------------------------------------------ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: wlmailhtml: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 wlmailhtml:FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120203/b2c85706/attachment.html From msc at freeswitch.org Sat Feb 4 02:55:04 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 3 Feb 2012 15:55:04 -0800 Subject: [Freeswitch-users] It's that time of year: Buy the devs dinner! Message-ID: Hello all! Each February the FreeSWITCH team converges on a small Milwaukee suburb to plot and plan how we will continue to conquer the world of open source telephony. From personal experience I can attest to the fact that it's really difficult to conquer the world on an empty stomach. :) We'd like to invite everyone to throw a few bucks into the hat to buy dinner for the FreeSWITCH core development team. The easiest way to donate is to hit the gold "donate" button on the right-hand side of the main FreeSWITCH page. Be sure to include a memo line, something like "bon appetit," so that we can keep track. Thanks again for supporting the FreeSWITCH team! We have an awesome community and you continue to show it on a daily basis. -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120203/ed83f971/attachment.html From b2m at a-cti.com Sat Feb 4 10:14:39 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Sat, 4 Feb 2012 12:44:39 +0530 Subject: [Freeswitch-users] Is OPenVox Gsm Card compatible for FS? In-Reply-To: References: Message-ID: Hi team, Can some one please let me know whether gsm card(OpenVox G400P4 QuadBand GSM Card) will work on FS, I am trying to route calls to gsm network. I saw a driver but they said its compatible only for asterisk. Do we need separate driver for this card? is there any other brand/make where I can route gsm traffic? Thanks, Bala -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120204/5843ab7d/attachment.html From jmesquita at freeswitch.org Sat Feb 4 16:59:55 2012 From: jmesquita at freeswitch.org (Jmesquita@freeswitch.org) Date: Sat, 4 Feb 2012 10:59:55 -0300 Subject: [Freeswitch-users] Is OPenVox Gsm Card compatible for FS? In-Reply-To: References: Message-ID: <2E9955B9-21E6-4A6A-A099-B31DFAFD4400@freeswitch.org> Khomp makes GSM cards compatible with freeswitch. www.khomp.com.br Regards, Jo?o Mesquita On 04/02/2012, at 04:14 a.m., Balamurugan Mahendran wrote: > Hi team, > > Can some one please let me know whether gsm card(OpenVox G400P4 QuadBand GSM Card) will work on FS, I am trying to route calls to gsm network. I saw a driver but they said its compatible only for asterisk. Do we need separate driver for this card? is there any other brand/make where I can route gsm traffic? > > Thanks, > Bala > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120204/b0fcb356/attachment.html From asrivas at gmail.com Sat Feb 4 09:23:28 2012 From: asrivas at gmail.com (Anurag Srivastava) Date: Fri, 3 Feb 2012 22:23:28 -0800 Subject: [Freeswitch-users] external_sip_ip not refreshed after ip address change. In-Reply-To: References: Message-ID: I have a question about nat behavior in freeswitch. Basically my external calls to freeswitch are getting disconnected after 30 seconds of two way audio when my external ip address changes. I have dhcp from my ISP and am using external_sip_ip and external_rtp_ip as stun:. When my IP changes I see that external_sip_ip does not get refreshed but external_rtp_ip does. I am not allowed to enable upnp/nat-pmp on my router. Apparently it is a known issue that external_sip_ip is read just at load time and not refreshed even if it is specified in stun format Is there a fix to this problem? There is always the option of restarting profile when ddclient notes an ip change. Is there something inbuilt into FS. It is already finding that ip address has changed as reflected in external_rtp_ip which does use stun and gets the right ip address. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120203/3872e726/attachment-0001.html From Ryan at ocens.com Sun Feb 5 06:21:03 2012 From: Ryan at ocens.com (Ryan Watkins) Date: Sun, 5 Feb 2012 03:21:03 +0000 Subject: [Freeswitch-users] Conference Auto Outcall to External Numbers In-Reply-To: <44E5C0A9D48A3246966A4AE04692014D132E155E@CH1PRD0610MB355.namprd06.prod.outlook.com> References: <44E5C0A9D48A3246966A4AE04692014D132E145D@CH1PRD0610MB355.namprd06.prod.outlook.com> <44E5C0A9D48A3246966A4AE04692014D132E155E@CH1PRD0610MB355.namprd06.prod.outlook.com> Message-ID: <44E5C0A9D48A3246966A4AE04692014D132E3941@CH1PRD0610MB355.namprd06.prod.outlook.com> Sorry for resending this, but need some assistance on this: Is there a way to set a Conference Auto Outcall to numbers external of the PBX? Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ryan Watkins Sent: Tuesday, January 31, 2012 2:28 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference Auto Outcall to External Numbers The call forwarding function itself works, if I set call forwarding for a specific extension and then call that extension it will ring at the forwarded external phone. Also, with call forwarding disabled on the extensions the conference auto outcall does call the internal extensions. There just seems to be a disconnect between the conference outcall to a forwarded extension. But perhaps I should rephrase the question... The conference extension I'm needing to setup should auto outcall to a set of external numbers. I figured the way to do it would be to create internal extensions, and set the call forwarding to the external number. But perhaps there's a better way to approach this need? For example, the outcall user listing in the conference.xml is as follows: Is there a way to change the data syntax to instead direct the outbound auto call to an external number instead of an internal user? Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, January 31, 2012 1:10 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference Auto Outcall to External Numbers What actually happens when you set the call forwarding? Console debug log w/ SIP trace would be helpful, as would any relevant configs you've got in place. -MC On Tue, Jan 31, 2012 at 12:46 PM, Ryan Watkins > wrote: Morning/Afternoon/Evening all... I've been trying to setup a conference call that would use the auto outcall function, and it would call external numbers (like their cell). I've tried to setup extensions for the individuals, and set the call forwarding to their external number. Although the auto outcall function will successfully call the extensions when they are set to an internal SIP phone, when call forwarding is enabled the conference call will not call the external number. Any ideas on how to accomplish this? Thanks a lot! Ryan _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120205/138a6d5c/attachment.html From daggelinckxmichel at gmail.com Sun Feb 5 07:23:38 2012 From: daggelinckxmichel at gmail.com (Michel Daggelinckx) Date: Sun, 05 Feb 2012 05:23:38 +0100 Subject: [Freeswitch-users] Is OPenVox Gsm Card compatible for FS? In-Reply-To: References: Message-ID: <4F2E044A.8030601@gmail.com> Op 4/02/2012 8:14, Balamurugan Mahendran schreef: > Hi team, > > Can some one please let me know whether gsm card(OpenVox G400P4 > QuadBand GSM Card) will work on FS, I am trying to route calls to gsm > network. I saw a driver but they said its compatible only for > asterisk. Do we need separate driver for this card? is there any other > brand/make where I can route gsm traffic? > > Thanks, > Bala > > If the card uses dahdi drivers it will work with fs/freetdm Michel > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120205/59dcf452/attachment.html From darcy at primrose.ws Sun Feb 5 07:54:38 2012 From: darcy at primrose.ws (Darcy) Date: Sat, 4 Feb 2012 23:54:38 -0500 Subject: [Freeswitch-users] vm_delete Message-ID: I have noticed that vm_delete does not actually delete the vm file from the storage directory, it does delete the file as far as vm_list or from the endpoint, but the file stays in the directory, is this a known issue or am I missing something. Darcy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120204/f59aec0a/attachment.html From krice at freeswitch.org Sun Feb 5 09:34:28 2012 From: krice at freeswitch.org (Ken Rice) Date: Sun, 05 Feb 2012 00:34:28 -0600 Subject: [Freeswitch-users] Conference Auto Outcall to External Numbers In-Reply-To: <44E5C0A9D48A3246966A4AE04692014D132E3941@CH1PRD0610MB355.namprd06.prod.outlook.com> Message-ID: Yes... Just specify a proper dialstring just like you would any other bridge command.... On 2/4/12 9:21 PM, "Ryan Watkins" wrote: > Sorry for resending this, but need some assistance on this: > > Is there a way to set a Conference Auto Outcall to numbers external of the > PBX? > > Thanks! > > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ryan > Watkins > Sent: Tuesday, January 31, 2012 2:28 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Conference Auto Outcall to External Numbers > > The call forwarding function itself works, if I set call forwarding for a > specific extension and then call that extension it will ring at the forwarded > external phone. Also, with call forwarding disabled on the extensions the > conference auto outcall does call the internal extensions. There just seems to > be a disconnect between the conference outcall to a forwarded extension. But > perhaps I should rephrase the question? > > The conference extension I?m needing to setup should auto outcall to a set of > external numbers. I figured the way to do it would be to create internal > extensions, and set the call forwarding to the external number. But perhaps > there?s a better way to approach this need? > > For example, the outcall user listing in the conference.xml is as follows: > > > > Is there a way to change the data syntax to instead direct the outbound auto > call to an external number instead of an internal user? > > Thanks! > > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael > Collins > Sent: Tuesday, January 31, 2012 1:10 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Conference Auto Outcall to External Numbers > > What actually happens when you set the call forwarding? Console debug log w/ > SIP trace would be helpful, as would any relevant configs you've got in place. > > -MC > > On Tue, Jan 31, 2012 at 12:46 PM, Ryan Watkins wrote: > > Morning/Afternoon/Evening all? > > I?ve been trying to setup a conference call that would use the auto outcall > function, and it would call external numbers (like their cell). I?ve tried to > setup extensions for the individuals, and set the call forwarding to their > external number. Although the auto outcall function will successfully call the > extensions when they are set to an internal SIP phone, when call forwarding is > enabled the conference call will not call the external number. Any ideas on > how to accomplish this? > > Thanks a lot! > > Ryan > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120205/bcc1517a/attachment-0001.html From avi at avimarcus.net Sun Feb 5 13:12:52 2012 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 5 Feb 2012 12:12:52 +0200 Subject: [Freeswitch-users] sip_gateway dialplan variable In-Reply-To: References: Message-ID: At worst, you can create a variable in the gateway settings so it populates a channel variable. I couldn't find the syntax on the wiki, though.. should make that easy to find. We need a real gateway page.. -Avi On Wed, Feb 1, 2012 at 9:29 AM, Hynek Cihlar wrote: > Hello Mike and thanks for the reply. > > Unfortunately sip_req_host only gives me the sip host. My setup is > probably not quite usual, I have a multihome system with many gateways, > each gateway name assigned to a unique id. The unique id is simply a random > string identifying the gateway in the whole system. The unique id is also > persisted in the DB, based on the id additional call processing takes > place. My initial idea was to get the gateway name in the dialplan and pass > it to an ESL app. But it really looks like it is not possible to retrieve > the gateway name at all, sadly. > > Hynek > > > > On Tue, Jan 24, 2012 at 8:42 AM, Michael Lutz wrote: > >> Hi Hynek, >> >> {sip_req_host} should give your own gateway (local), and {network_addr} >> should give the remote gateway. >> >> Otherwise writing a cdr and look will help you with this. >> >> Regards, >> Mike >> >> Op 24 jan. 2012 om 08:27 heeft Hynek Cihlar het >> volgende geschreven: >> >> In one of the examples on wiki I found ${sip_gateway}, but that doesn't >> seem to work. Also, there is ${sip_gateway_name}, but that works on >> outbound calls only. >> >> Hynek >> >> >> >> On Mon, Jan 23, 2012 at 7:03 PM, Hynek Cihlar wrote: >> >>> Dear, is there a dialplan variable that would hold the gateway name an >>> incoming call came through? Searching through the source code and wiki >>> didn't yield any positive result. >>> >>> Thanks! >>> >>> Hynek >>> >>> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120205/851ef646/attachment.html From moises.silva at gmail.com Sun Feb 5 21:19:59 2012 From: moises.silva at gmail.com (Moises Silva) Date: Sun, 5 Feb 2012 13:19:59 -0500 Subject: [Freeswitch-users] Is OPenVox Gsm Card compatible for FS? In-Reply-To: <4F2E044A.8030601@gmail.com> References: <4F2E044A.8030601@gmail.com> Message-ID: On Sat, Feb 4, 2012 at 11:23 PM, Michel Daggelinckx < daggelinckxmichel at gmail.com> wrote: > Op 4/02/2012 8:14, Balamurugan Mahendran schreef: > > Hi team, > > Can some one please let me know whether gsm card(OpenVox G400P4 QuadBand > GSM Card) will work on FS, I am trying to route calls to gsm network. I > saw a driver but they said its compatible only for asterisk. Do we need > separate driver for this card? is there any other brand/make where I can > route gsm traffic? > > Thanks, > Bala > > > If the card uses dahdi drivers it will work with fs/freetdm > That is most likely not true. Most (if not all) gsm cards will need to send/recv AT commands, this is typically done in user space, meaning you still need some code in FreeTDM to talk to the GSM modules. Sangoma has been working in the past months in a library for call control using AT commands for GSM modules called libwat (Wireless AT) See http://git.sangoma.com/gitweb.cgi?p=wat.git The integration of this library into FreeTDM already started, I estimate it will be in a working state by early March. However, the initial support is only for the "Telit" GSM modules (used in Sangoma W400 GSM card), which I don't think is the same module OpenVox cards use. Getting to work the OpenVox card it is probably a matter of writing the proper AT command plugin into libwat. *Moises Silva **Software Engineer, Development Manager*** 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/20120205/464bd986/attachment.html From cogs66 at gmail.com Sun Feb 5 21:21:33 2012 From: cogs66 at gmail.com (cogs66) Date: Sun, 5 Feb 2012 10:21:33 -0800 (PST) Subject: [Freeswitch-users] vm_delete In-Reply-To: References: Message-ID: <1328466093250-7256341.post@n2.nabble.com> Hi I am experiencing the same but not been able to find a resolve to the issue yet. Thanks Andy -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/vm-delete-tp7255354p7256341.html Sent from the freeswitch-users mailing list archive at Nabble.com. From hynek.cihlar at gmail.com Sun Feb 5 21:41:26 2012 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Sun, 5 Feb 2012 19:41:26 +0100 Subject: [Freeswitch-users] sip_gateway dialplan variable In-Reply-To: References: Message-ID: Yes! Thanks for the tip! Hynek On Sun, Feb 5, 2012 at 11:12 AM, Avi Marcus wrote: > At worst, you can create a variable in the gateway settings so it > populates a channel variable. I couldn't find the syntax on the wiki, > though.. should make that easy to find. We need a real gateway page.. > > -Avi > > > On Wed, Feb 1, 2012 at 9:29 AM, Hynek Cihlar wrote: > >> Hello Mike and thanks for the reply. >> >> Unfortunately sip_req_host only gives me the sip host. My setup is >> probably not quite usual, I have a multihome system with many gateways, >> each gateway name assigned to a unique id. The unique id is simply a random >> string identifying the gateway in the whole system. The unique id is also >> persisted in the DB, based on the id additional call processing takes >> place. My initial idea was to get the gateway name in the dialplan and pass >> it to an ESL app. But it really looks like it is not possible to retrieve >> the gateway name at all, sadly. >> >> Hynek >> >> >> >> On Tue, Jan 24, 2012 at 8:42 AM, Michael Lutz wrote: >> >>> Hi Hynek, >>> >>> {sip_req_host} should give your own gateway (local), and {network_addr} >>> should give the remote gateway. >>> >>> Otherwise writing a cdr and look will help you with this. >>> >>> Regards, >>> Mike >>> >>> Op 24 jan. 2012 om 08:27 heeft Hynek Cihlar >>> het volgende geschreven: >>> >>> In one of the examples on wiki I found ${sip_gateway}, but that doesn't >>> seem to work. Also, there is ${sip_gateway_name}, but that works on >>> outbound calls only. >>> >>> Hynek >>> >>> >>> >>> On Mon, Jan 23, 2012 at 7:03 PM, Hynek Cihlar wrote: >>> >>>> Dear, is there a dialplan variable that would hold the gateway name an >>>> incoming call came through? Searching through the source code and wiki >>>> didn't yield any positive result. >>>> >>>> Thanks! >>>> >>>> Hynek >>>> >>>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> 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 >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> 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 >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120205/e72ccdf4/attachment.html From darcy at primrose.ws Sun Feb 5 22:47:32 2012 From: darcy at primrose.ws (Darcy) Date: Sun, 5 Feb 2012 14:47:32 -0500 Subject: [Freeswitch-users] vm_delete In-Reply-To: <1328466093250-7256341.post@n2.nabble.com> References: <1328466093250-7256341.post@n2.nabble.com> Message-ID: <131B09351F274242AD39E78609870EC0@DWP> I did a work around, because I am running this from a cgi, after I issue the command to the fs to delete the email, I actually delete the file with the cgi, not the best way, but it works until I find a seemless fix from fs. Darcy -----Original Message----- From: cogs66 Sent: Sunday, February 05, 2012 1:21 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] vm_delete Hi I am experiencing the same but not been able to find a resolve to the issue yet. Thanks Andy -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/vm-delete-tp7255354p7256341.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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.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 Sun Feb 5 18:07:01 2012 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Sun, 5 Feb 2012 09:07:01 -0600 Subject: [Freeswitch-users] Server with 2 NICS Message-ID: Helo :) I'm new to FS but not to linux. I've sucessfully install FS into a centos 6.2. But I have an issue. Server has 2 nics: venet0: 10.9.19.5/16 eth0: when starting fs, it listen SIP (5060/udp) using venet0. Phones can register and they talk each other. However, we are using an external GW and that fails. Doing a depth analisys I've realized that the reason is that because venet0 is being used, SIP payload has venet0 within but server IP is eth0's so this is tipical mitchach. I've sucessfully configure in vars.xml the local_ip4 (or whatever it writes) to use public IP. but after that no phones are registering. (before anyone says, yes phones points to public IP of course). Nmap reports that SIP (5060/udp) is listen by eth0 but when I got phone SIP, i only get Registering request without an answer. I really need to make this listen in eth0. What suggestion do you have. Please void telling me to turn of venet0 (not possible for politics). TIA From sherifomran2000 at yahoo.com Mon Feb 6 01:00:10 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sun, 5 Feb 2012 14:00:10 -0800 (PST) Subject: [Freeswitch-users] how to store cdr data to sql dB In-Reply-To: <131B09351F274242AD39E78609870EC0@DWP> Message-ID: <1328479210.46134.YahooMailClassic@web110814.mail.gq1.yahoo.com> Hi guys, any body knows how to store cdr data into mysql dB through xml-curl? could you please help thank you kind regards Sherif -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120205/651a6a84/attachment.html From avi at avimarcus.net Mon Feb 6 01:05:18 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 6 Feb 2012 00:05:18 +0200 Subject: [Freeswitch-users] how to store cdr data to sql dB In-Reply-To: <1328479210.46134.YahooMailClassic@web110814.mail.gq1.yahoo.com> References: <131B09351F274242AD39E78609870EC0@DWP> <1328479210.46134.YahooMailClassic@web110814.mail.gq1.yahoo.com> Message-ID: You must mean http://wiki.freeswitch.org/wiki/Mod_xml_cdr or http://wiki.freeswitch.org/wiki/Xmlcdrd xml_cdr "merely" posts the CDR data in XML form to a web page. You can then have you python/perl/ruby/php/node.js/lua script pick out the columns you want and save it to the mysql db. If you want a php sample, fusionpbx has the open source code to save to mysql, sqlite, or pg, via PDO. -Avi On Mon, Feb 6, 2012 at 12:00 AM, Sherif Omran wrote: > Hi guys, > > any body knows how to store cdr data into mysql dB through xml-curl? > > could you please help > > thank you > kind regards > > Sherif > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/60170e87/attachment-0001.html From brian at freeswitch.org Mon Feb 6 01:08:02 2012 From: brian at freeswitch.org (Brian West) Date: Sun, 5 Feb 2012 16:08:02 -0600 Subject: [Freeswitch-users] vm_delete In-Reply-To: <131B09351F274242AD39E78609870EC0@DWP> References: <1328466093250-7256341.post@n2.nabble.com> <131B09351F274242AD39E78609870EC0@DWP> Message-ID: <182CED90-BB0A-428C-AA75-ECD4E6C70485@freeswitch.org> How about you open a jira on this issue? http://jira.freeswitch.org On Feb 5, 2012, at 1:47 PM, Darcy wrote: > I did a work around, because I am running this from a cgi, after I issue > the command to the fs to delete the email, I actually delete the file with > the cgi, not the best way, but it works until I find a seemless fix from fs. > > Darcy -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120205/acde22a2/attachment.html From jmesquita at freeswitch.org Mon Feb 6 01:33:39 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Sun, 5 Feb 2012 19:33:39 -0300 Subject: [Freeswitch-users] Server with 2 NICS In-Reply-To: References: Message-ID: Look in the vars.xml and replace the $${local_ipv4} by your configured eth0 ip address. FreeSWITCH resolves $${local_ipv4} to the IP of the interface who is connected to you default gateway. Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Sunday, February 5, 2012 at 12:07 PM, Luis Daniel Lucio Quiroz wrote: > Helo :) > > I'm new to FS but not to linux. I've sucessfully install FS into a > centos 6.2. But I have an issue. > > Server has 2 nics: > venet0: 10.9.19.5/16 > eth0: > > when starting fs, it listen SIP (5060/udp) using venet0. Phones can > register and they talk each other. However, we are using an external > GW and that fails. Doing a depth analisys I've realized that the > reason is that because venet0 is being used, SIP payload has venet0 > within but server IP is eth0's so this is tipical mitchach. > > I've sucessfully configure in vars.xml the local_ip4 (or whatever it > writes) to use public IP. but after that no phones are registering. > (before anyone says, yes phones points to public IP of course). Nmap > reports that SIP (5060/udp) is listen by eth0 but when I got phone > SIP, i only get Registering request without an answer. > > I really need to make this listen in eth0. What suggestion do you > have. Please void telling me to turn of venet0 (not possible for > politics). > > TIA > > _________________________________________________________________________ > 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/20120205/cd0bb47c/attachment.html From sherifomran2000 at yahoo.com Mon Feb 6 03:51:33 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sun, 5 Feb 2012 16:51:33 -0800 (PST) Subject: [Freeswitch-users] a2billing for freeswitch In-Reply-To: Message-ID: <1328489493.67899.YahooMailClassic@web110806.mail.gq1.yahoo.com> Hi all, any body knows a solution already made to use a2biling for freeswitch? So that not to reinvent the wheel? thank you S.O -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120205/5ddc43ce/attachment.html From krice at freeswitch.org Mon Feb 6 04:12:21 2012 From: krice at freeswitch.org (Ken Rice) Date: Sun, 05 Feb 2012 19:12:21 -0600 Subject: [Freeswitch-users] a2billing for freeswitch In-Reply-To: <1328489493.67899.YahooMailClassic@web110806.mail.gq1.yahoo.com> Message-ID: The a2billing guys at one point were doing this already... Email them to see what they ended up with On 2/5/12 6:51 PM, "Sherif Omran" wrote: > Hi all, > > any body knows a solution already made to use a2biling for freeswitch? So that > not to reinvent the wheel? > > > thank you > > S.O > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120205/05377740/attachment.html From piyush.sharma at coraltele.com Mon Feb 6 08:15:43 2012 From: piyush.sharma at coraltele.com (Piyush Sharma) Date: Mon, 06 Feb 2012 05:15:43 -0000 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 68, Issue 22 In-Reply-To: References: Message-ID: <1336366836.3502.7.camel@localhost.localdomain> I am new to FreeSWITCH, I am telling you how I did it, In autoload config there is a file cdr_csv.conf.xml there are few queries written, I made my own according to my cdr requirement, After that, I wrote a function to insert into sql in mod_cdr_csv.c and called it from my_on_reporting (written function in mod_cdr_cdv.c), and it made my cdr to mysql. On Mon, 2012-02-06 at 01:05 +0300, 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: sip_gateway dialplan variable (Hynek Cihlar) > 2. Re: vm_delete (Darcy) > 3. Server with 2 NICS (Luis Daniel Lucio Quiroz) > 4. how to store cdr data to sql dB (Sherif Omran) > 5. Re: how to store cdr data to sql dB (Avi Marcus) > MHTML Document attachment > > -------- Forwarded Message -------- > > From: Hynek Cihlar > > Reply-To: FreeSWITCH Users Help > > > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] sip_gateway dialplan variable > > Date: Sun, 5 Feb 2012 19:41:26 +0100 > > > > Yes! Thanks for the tip! > > > > Hynek > > > > > > > > On Sun, Feb 5, 2012 at 11:12 AM, Avi Marcus > > wrote: > > At worst, you can create a variable in the gateway settings > > so it populates a channel variable. I couldn't find the > > syntax on the wiki, though.. should make that easy to find. > > We need a real gateway page.. > > > > -Avi > > > > > > On Wed, Feb 1, 2012 at 9:29 AM, Hynek Cihlar > > wrote: > > Hello Mike and thanks for the reply. > > > > > > Unfortunately sip_req_host only gives me the sip > > host. My setup is probably not quite usual, I have a > > multihome system with many gateways, each gateway > > name assigned to a unique id. The unique id is > > simply a random string identifying the gateway in > > the whole system. The unique id is also persisted in > > the DB, based on the id additional call processing > > takes place. My initial idea was to get the gateway > > name in the dialplan and pass it to an ESL app. But > > it really looks like it is not possible to retrieve > > the gateway name at all, sadly. > > > > Hynek > > > > > > > > On Tue, Jan 24, 2012 at 8:42 AM, Michael Lutz > > wrote: > > Hi Hynek, > > > > > > {sip_req_host} should give your own gateway > > (local), and {network_addr} should give the > > remote gateway. > > > > > > Otherwise writing a cdr and look will help > > you with this. > > > > > > Regards, > > Mike > > > > Op 24 jan. 2012 om 08:27 heeft Hynek Cihlar > > het volgende > > geschreven: > > > > > > > > > In one of the examples on wiki I found > > > ${sip_gateway}, but that doesn't seem to > > > work. Also, there is ${sip_gateway_name}, > > > but that works on outbound calls only. > > > > > > Hynek > > > > > > > > > > > > On Mon, Jan 23, 2012 at 7:03 PM, Hynek > > > Cihlar wrote: > > > Dear, is there a dialplan variable > > > that would hold the gateway name > > > an incoming call came through? > > > Searching through the source code > > > and wiki didn't yield any positive > > > result. > > > > > > > > > Thanks! > > > > > > > > > Hynek > > > > > > > > > > > > _________________________________________________________________________ > > > 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 > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE: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 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: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 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > 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 > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > MHTML Document attachment > > -------- Forwarded Message -------- > > From: Darcy > > Reply-To: FreeSWITCH Users Help > > > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] vm_delete > > Date: Sun, 5 Feb 2012 14:47:32 -0500 > > > > I did a work around, because I am running this from a cgi, after I issue > > the command to the fs to delete the email, I actually delete the file with > > the cgi, not the best way, but it works until I find a seemless fix from fs. > > > > Darcy > > > > -----Original Message----- > > From: cogs66 > > Sent: Sunday, February 05, 2012 1:21 PM > > To: freeswitch-users at lists.freeswitch.org > > Subject: Re: [Freeswitch-users] vm_delete > > > > Hi > > > > I am experiencing the same but not been able to find a resolve to the issue > > yet. > > > > Thanks > > > > Andy > > > > -- > > View this message in context: > > http://freeswitch-users.2379917.n2.nabble.com/vm-delete-tp7255354p7256341.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 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > > MHTML Document attachment > > -------- Forwarded Message -------- > > From: Luis Daniel Lucio Quiroz > > Reply-To: FreeSWITCH Users Help > > > > To: freeswitch-users at lists.freeswitch.org > > Subject: [Freeswitch-users] Server with 2 NICS > > Date: Sun, 5 Feb 2012 09:07:01 -0600 > > > > Helo :) > > > > I'm new to FS but not to linux. I've sucessfully install FS into a > > centos 6.2. But I have an issue. > > > > Server has 2 nics: > > venet0: 10.9.19.5/16 > > eth0: > > > > when starting fs, it listen SIP (5060/udp) using venet0. Phones can > > register and they talk each other. However, we are using an external > > GW and that fails. Doing a depth analisys I've realized that the > > reason is that because venet0 is being used, SIP payload has venet0 > > within but server IP is eth0's so this is tipical mitchach. > > > > I've sucessfully configure in vars.xml the local_ip4 (or whatever it > > writes) to use public IP. but after that no phones are registering. > > (before anyone says, yes phones points to public IP of course). Nmap > > reports that SIP (5060/udp) is listen by eth0 but when I got phone > > SIP, i only get Registering request without an answer. > > > > I really need to make this listen in eth0. What suggestion do you > > have. Please void telling me to turn of venet0 (not possible for > > politics). > > > > TIA > > > > > MHTML Document attachment > > -------- Forwarded Message -------- > > From: Sherif Omran > > Reply-To: FreeSWITCH Users Help > > > > To: FreeSWITCH Users Help > > Subject: [Freeswitch-users] how to store cdr data to sql dB > > Date: Sun, 5 Feb 2012 14:00:10 -0800 (PST) > > > > Hi guys, > > > > any body knows how to store cdr data into mysql dB through xml-curl? > > > > could you please help > > > > thank you > > kind regards > > > > Sherif > > > > > MHTML Document attachment > > -------- Forwarded Message -------- > > From: Avi Marcus > > Reply-To: FreeSWITCH Users Help > > > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] how to store cdr data to sql dB > > Date: Mon, 6 Feb 2012 00:05:18 +0200 > > > > You must > > mean http://wiki.freeswitch.org/wiki/Mod_xml_cdr or http://wiki.freeswitch.org/wiki/Xmlcdrd > > xml_cdr "merely" posts the CDR data in XML form to a web page. You > > can then have you python/perl/ruby/php/node.js/lua script pick out > > the columns you want and save it to the mysql db. > > If you want a php sample, fusionpbx has the open source code to save > > to mysql, sqlite, or pg, via PDO. > > > > > > -Avi > > > > > > On Mon, Feb 6, 2012 at 12:00 AM, Sherif Omran > > wrote: > > Hi guys, > > > > any body knows how to store cdr data into mysql dB through > > xml-curl? > > > > could you please help > > > > thank you > > kind regards > > > > Sherif > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > 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 > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Piyush Sharma From miha at softnet.si Mon Feb 6 10:44:06 2012 From: miha at softnet.si (Miha Zoubek) Date: Mon, 06 Feb 2012 08:44:06 +0100 Subject: [Freeswitch-users] Striping destination number In-Reply-To: References: <4F2903AB.3030905@softnet.si> Message-ID: <4F2F84C6.6060203@softnet.si> Hi @Michael, I set in dialplan like this: sip_redirect_contact_user_0 is set to 051325856, so 0 should be trimed. But in log I get this: EXECUTE sofia/internal/0038618100100 at xxx.xxx.xxx.xxx bridge(sofia/external/386051357952 at xxx.xxx.xxx.xxx) You can see that 0 is not trimed. Can you pls help me why? Thanks! BR, Miha I have one problem. On 2/1/2012 7:25 PM, Michael Collins wrote: > If you know for a fact that it's only a leading zero then you can use > the variable manipulation features: > > http://wiki.freeswitch.org/wiki/Manipulating_Channel_Variables > > -MC > > On Wed, Feb 1, 2012 at 1:19 AM, Miha Zoubek > wrote: > > Hi, > > how can I strip destination number in my dialplan if the destination > number is not in condition option. > > data="sofia/external/386${sip_redirect_contact_user_0}@xxx.xxx.xxx.xxx" > /> > I need to stip leading zero from ${sip_redirect_contact_user_0} > variable. > > Thanks! > Miha > > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/feccc625/attachment.html From avi at avimarcus.net Mon Feb 6 10:50:01 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 6 Feb 2012 09:50:01 +0200 Subject: [Freeswitch-users] Striping destination number In-Reply-To: <4F2F84C6.6060203@softnet.si> References: <4F2903AB.3030905@softnet.si> <4F2F84C6.6060203@softnet.si> Message-ID: Try ${sip_redirect_contact_user_0:1} to tell it to skip character 0... -Avi On Mon, Feb 6, 2012 at 9:44 AM, Miha Zoubek wrote: > Hi @Michael, > > I set in dialplan like this: > > "sofia/external/386${sip_redirect_contact_user_0:0}@xxx.xxx.xxx.xxx"/> > > sip_redirect_contact_user_0 is set to 051325856, so 0 should be trimed. > > But in log I get this: > > > EXECUTE sofia/internal/0038618100100 at xxx.xxx.xxx.xxx bridge( > sofia/external/386051357952 at xxx.xxx.xxx.xxx) > > You can see that 0 is not trimed. Can you pls help me why? > > Thanks! > > BR, > Miha > > I have one problem. > > On 2/1/2012 7:25 PM, Michael Collins wrote: > > If you know for a fact that it's only a leading zero then you can use the > variable manipulation features: > > http://wiki.freeswitch.org/wiki/Manipulating_Channel_Variables > > -MC > > On Wed, Feb 1, 2012 at 1:19 AM, Miha Zoubek wrote: > >> Hi, >> >> how can I strip destination number in my dialplan if the destination >> number is not in condition option. >> >> > data="sofia/external/386${sip_redirect_contact_user_0}@xxx.xxx.xxx.xxx"/> >> I need to stip leading zero from ${sip_redirect_contact_user_0} variable. >> >> Thanks! >> Miha >> >> >> -- >> Best regards / Lep Pozdrav >> Miha Zoubek >> Softnet d.o.o. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/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 > > 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 > > > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/b8d06297/attachment.html From miha at softnet.si Mon Feb 6 11:11:03 2012 From: miha at softnet.si (Miha Zoubek) Date: Mon, 06 Feb 2012 09:11:03 +0100 Subject: [Freeswitch-users] Striping destination number In-Reply-To: References: <4F2903AB.3030905@softnet.si> <4F2F84C6.6060203@softnet.si> Message-ID: <4F2F8B17.9000802@softnet.si> HI @Avi, I do not know how did I missed that. Not it works:) Thanks! BR, Miha On 2/6/2012 8:50 AM, Avi Marcus wrote: > Try ${sip_redirect_contact_user_0:1} to tell it to skip character 0... > -Avi > > On Mon, Feb 6, 2012 at 9:44 AM, Miha Zoubek > wrote: > > Hi @Michael, > > I set in dialplan like this: > > data="sofia/external/386${sip_redirect_contact_user_0:0}@xxx.xxx.xxx.xxx" > > /> > > sip_redirect_contact_user_0 is set to 051325856, so 0 should be > trimed. > > But in log I get this: > > > EXECUTE sofia/internal/0038618100100 at xxx.xxx.xxx.xxx > > bridge(sofia/external/386051357952 at xxx.xxx.xxx.xxx > ) > > You can see that 0 is not trimed. Can you pls help me why? > > Thanks! > > BR, > Miha > > I have one problem. > > On 2/1/2012 7:25 PM, Michael Collins wrote: >> If you know for a fact that it's only a leading zero then you can >> use the variable manipulation features: >> >> http://wiki.freeswitch.org/wiki/Manipulating_Channel_Variables >> >> -MC >> >> On Wed, Feb 1, 2012 at 1:19 AM, Miha Zoubek > > wrote: >> >> Hi, >> >> how can I strip destination number in my dialplan if the >> destination >> number is not in condition option. >> >> > data="sofia/external/386${sip_redirect_contact_user_0}@xxx.xxx.xxx.xxx" >> >> /> >> I need to stip leading zero from >> ${sip_redirect_contact_user_0} variable. >> >> Thanks! >> Miha >> >> >> -- >> Best regards / Lep Pozdrav >> Miha Zoubek >> Softnet d.o.o. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/7e9af931/attachment-0001.html From miha at softnet.si Mon Feb 6 12:36:58 2012 From: miha at softnet.si (Miha Zoubek) Date: Mon, 06 Feb 2012 10:36:58 +0100 Subject: [Freeswitch-users] event_socket.conf Message-ID: <4F2F9F3A.1080306@softnet.si> Hi, I have changed default password for event_socket.conf (clueclone). Now I am unable to connect to FS via fs_cli (I can connect remotely). I have also set ACL which will also do it for localhost. Where can I find configuration file for FS_cli that I can change default password (If I set password to clueclone in event_socket.conf it works). I have try to change it in: /usr/local/src/freeswitch/libs/esl/fs_cli, but I guess this is not the right file. My ACL: Thanks! Regards, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/284786a8/attachment.html From peter.olsson at visionutveckling.se Mon Feb 6 12:52:56 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 6 Feb 2012 09:52:56 +0000 Subject: [Freeswitch-users] event_socket.conf In-Reply-To: <4F2F9F3A.1080306@softnet.si> References: <4F2F9F3A.1080306@softnet.si> Message-ID: <1FFF97C269757C458224B7C895F35F15038B40@cantor.std.visionutv.se> Just start with param -p "fs_cli -p password". I don't think fs_cli has a config file anywhere. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Miha Zoubek Skickat: den 6 februari 2012 10:37 Till: FreeSWITCH Users Help ?mne: [Freeswitch-users] event_socket.conf Hi, I have changed default password for event_socket.conf (clueclone). Now I am unable to connect to FS via fs_cli (I can connect remotely). I have also set ACL which will also do it for localhost. Where can I find configuration file for FS_cli that I can change default password (If I set password to clueclone in event_socket.conf it works). I have try to change it in: /usr/local/src/freeswitch/libs/esl/fs_cli, but I guess this is not the right file. My ACL: Thanks! Regards, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. !DSPAM:4f2f9ef132761401295748! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/2c1a31e1/attachment.html From avi at avimarcus.net Mon Feb 6 12:53:48 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 6 Feb 2012 11:53:48 +0200 Subject: [Freeswitch-users] event_socket.conf In-Reply-To: <4F2F9F3A.1080306@softnet.si> References: <4F2F9F3A.1080306@softnet.si> Message-ID: As per: http://wiki.freeswitch.org/wiki/Fs_cli You can use fs_cli -p clueclone Or.. "An optional configuration file can be set up in the user's home directory. The file name is *.fs_cli_conf*. " see the syntax on the wiki. -Avi On Mon, Feb 6, 2012 at 11:36 AM, Miha Zoubek wrote: > Hi, > > I have changed default password for event_socket.conf (clueclone). Now I > am unable to connect to FS via fs_cli (I can connect remotely). > > I have also set ACL which will also do it for localhost. Where can I find > configuration file for FS_cli that I can change default password (If I set > password to clueclone in event_socket.conf it works). > > I have try to change it in: /usr/local/src/freeswitch/libs/esl/fs_cli, but > I guess this is not the right file. > > My ACL: > > > > > > > > > Thanks! > > Regards, > Miha > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/4e2f4433/attachment.html From miha at softnet.si Mon Feb 6 13:27:06 2012 From: miha at softnet.si (Miha Zoubek) Date: Mon, 06 Feb 2012 11:27:06 +0100 Subject: [Freeswitch-users] event_socket.conf In-Reply-To: References: <4F2F9F3A.1080306@softnet.si> Message-ID: <4F2FAAFA.10406@softnet.si> Hi, just to let you know, because this is very wird. My password in event_socket.conf was reload52166XML. From remote I could connection, from localhost fs_cli -p my_pass I was unable to make a connection. Now I have set password with just a words and it works from remote and locally. Thanks! Miha On 2/6/2012 10:53 AM, Avi Marcus wrote: > As per: http://wiki.freeswitch.org/wiki/Fs_cli > You can use fs_cli -p clueclone > > Or.. > > "An optional configuration file can be set up in the user's home > directory. The file name is *.fs_cli_conf*. " see the syntax on the wiki. > > -Avi > > > On Mon, Feb 6, 2012 at 11:36 AM, Miha Zoubek > wrote: > > Hi, > > I have changed default password for event_socket.conf (clueclone). > Now I am unable to connect to FS via fs_cli (I can connect remotely). > > I have also set ACL which will also do it for localhost. Where can > I find configuration file for FS_cli that I can change default > password (If I set password to clueclone in event_socket.conf it > works). > > I have try to change it in: > /usr/local/src/freeswitch/libs/esl/fs_cli, but I guess this is not > the right file. > > My ACL: > > > > > > > > > Thanks! > > Regards, > Miha > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/d9077f8b/attachment-0001.html From miha at softnet.si Mon Feb 6 13:32:07 2012 From: miha at softnet.si (Miha Zoubek) Date: Mon, 06 Feb 2012 11:32:07 +0100 Subject: [Freeswitch-users] Striping destination number In-Reply-To: <4F2F8B17.9000802@softnet.si> References: <4F2903AB.3030905@softnet.si> <4F2F84C6.6060203@softnet.si> <4F2F8B17.9000802@softnet.si> Message-ID: <4F2FAC27.7050007@softnet.si> Just one questione. In my info I can see that variable is set: variable_sip_redirect_contact_user_0: [051357952] After this variable is set I a condition in my dialplan: but in my log I can see that this varible is not set in condition : Regex (PASS) [IZVEDI_PREUSMERITEV] destination_number(IZVEDI_PREUSMERITEV) =~ e Dialplan: sofia/internal/0038618100100 at xxx.xxx.xxx.xxx Regex (PASS) [IZVEDI_PREUSMERITEV] sip_redirect_contact_user_0() =~ /^$/ break I am asking this because I can not make appropriate condition if this variable is not in my condition. Thanks! miha On 2/6/2012 9:11 AM, Miha Zoubek wrote: > HI @Avi, > > I do not know how did I missed that. > > Not it works:) > > Thanks! > > BR, > Miha > > On 2/6/2012 8:50 AM, Avi Marcus wrote: >> Try ${sip_redirect_contact_user_0:1} to tell it to skip character 0... >> -Avi >> >> On Mon, Feb 6, 2012 at 9:44 AM, Miha Zoubek > > wrote: >> >> Hi @Michael, >> >> I set in dialplan like this: >> >> > data="sofia/external/386${sip_redirect_contact_user_0:0}@xxx.xxx.xxx.xxx" >> >> /> >> >> sip_redirect_contact_user_0 is set to 051325856, so 0 should be >> trimed. >> >> But in log I get this: >> >> >> EXECUTE sofia/internal/0038618100100 at xxx.xxx.xxx.xxx >> >> bridge(sofia/external/386051357952 at xxx.xxx.xxx.xxx >> ) >> >> You can see that 0 is not trimed. Can you pls help me why? >> >> Thanks! >> >> BR, >> Miha >> >> I have one problem. >> >> On 2/1/2012 7:25 PM, Michael Collins wrote: >>> If you know for a fact that it's only a leading zero then you >>> can use the variable manipulation features: >>> >>> http://wiki.freeswitch.org/wiki/Manipulating_Channel_Variables >>> >>> -MC >>> >>> On Wed, Feb 1, 2012 at 1:19 AM, Miha Zoubek >> > wrote: >>> >>> Hi, >>> >>> how can I strip destination number in my dialplan if the >>> destination >>> number is not in condition option. >>> >>> >> data="sofia/external/386${sip_redirect_contact_user_0}@xxx.xxx.xxx.xxx" >>> >>> /> >>> I need to stip leading zero from >>> ${sip_redirect_contact_user_0} variable. >>> >>> Thanks! >>> Miha >>> >>> >>> -- >>> Best regards / Lep Pozdrav >>> Miha Zoubek >>> Softnet d.o.o. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> 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 >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> 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 >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> -- >> Best regards / Lep Pozdrav >> Miha Zoubek >> Softnet d.o.o. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/ced7c9c3/attachment.html From avi at avimarcus.net Mon Feb 6 13:34:48 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 6 Feb 2012 12:34:48 +0200 Subject: [Freeswitch-users] event_socket.conf In-Reply-To: <4F2FAAFA.10406@softnet.si> References: <4F2F9F3A.1080306@softnet.si> <4F2FAAFA.10406@softnet.si> Message-ID: Please file a http://jira.freeswitch.org with exactly what worked and what didn't. -Avi On Mon, Feb 6, 2012 at 12:27 PM, Miha Zoubek wrote: > Hi, > > just to let you know, because this is very wird. > > My password in event_socket.conf was reload52166XML. > From remote I could connection, from localhost fs_cli -p my_pass I was > unable to make a connection. > > Now I have set password with just a words and it works from remote and > locally. > > > > Thanks! > Miha > > > On 2/6/2012 10:53 AM, Avi Marcus wrote: > > As per: http://wiki.freeswitch.org/wiki/Fs_cli > You can use fs_cli -p clueclone > > Or.. > > "An optional configuration file can be set up in the user's home > directory. The file name is *.fs_cli_conf*. " see the syntax on the wiki. > > -Avi > > > On Mon, Feb 6, 2012 at 11:36 AM, Miha Zoubek wrote: > >> Hi, >> >> I have changed default password for event_socket.conf (clueclone). Now I >> am unable to connect to FS via fs_cli (I can connect remotely). >> >> I have also set ACL which will also do it for localhost. Where can I find >> configuration file for FS_cli that I can change default password (If I set >> password to clueclone in event_socket.conf it works). >> >> I have try to change it in: /usr/local/src/freeswitch/libs/esl/fs_cli, >> but I guess this is not the right file. >> >> My ACL: >> >> >> >> >> >> >> >> >> Thanks! >> >> Regards, >> Miha >> >> -- >> Best regards / Lep Pozdrav >> Miha Zoubek >> Softnet d.o.o. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/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 > > 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 > > > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/d536c2a8/attachment-0001.html From avi at avimarcus.net Mon Feb 6 13:40:06 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 6 Feb 2012 12:40:06 +0200 Subject: [Freeswitch-users] Striping destination number In-Reply-To: <4F2FAC27.7050007@softnet.si> References: <4F2903AB.3030905@softnet.si> <4F2F84C6.6060203@softnet.si> <4F2F8B17.9000802@softnet.si> <4F2FAC27.7050007@softnet.si> Message-ID: I think you need to use ${sip_redirect_contact_user_0} - the "raw" form only works for thing in the caller_profile, e.g. destination_number. -Avi On Mon, Feb 6, 2012 at 12:32 PM, Miha Zoubek wrote: > Just one questione. > > In my info I can see that variable is set: > > variable_sip_redirect_contact_user_0: [051357952] > > After this variable is set I a condition in my dialplan: > > > > but in my log I can see that this varible is not set in condition : > > Regex (PASS) [IZVEDI_PREUSMERITEV] > destination_number(IZVEDI_PREUSMERITEV) =~ e > Dialplan: sofia/internal/0038618100100 at xxx.xxx.xxx.xxx Regex (PASS) > [IZVEDI_PREUSMERITEV] sip_redirect_contact_user_0() =~ /^$/ break > > I am asking this because I can not make appropriate condition if this > variable is not in my condition. > > Thanks! > > miha > > > On 2/6/2012 9:11 AM, Miha Zoubek wrote: > > HI @Avi, > > I do not know how did I missed that. > > Not it works:) > > Thanks! > > BR, > Miha > > On 2/6/2012 8:50 AM, Avi Marcus wrote: > > Try ${sip_redirect_contact_user_0:1} to tell it to skip character 0... > -Avi > > On Mon, Feb 6, 2012 at 9:44 AM, Miha Zoubek wrote: > >> Hi @Michael, >> >> I set in dialplan like this: >> >> > "sofia/external/386${sip_redirect_contact_user_0:0}@xxx.xxx.xxx.xxx"/> >> >> sip_redirect_contact_user_0 is set to 051325856, so 0 should be trimed. >> >> But in log I get this: >> >> >> EXECUTE sofia/internal/0038618100100 at xxx.xxx.xxx.xxx bridge( >> sofia/external/386051357952 at xxx.xxx.xxx.xxx) >> >> You can see that 0 is not trimed. Can you pls help me why? >> >> Thanks! >> >> BR, >> Miha >> >> I have one problem. >> >> On 2/1/2012 7:25 PM, Michael Collins wrote: >> >> If you know for a fact that it's only a leading zero then you can use the >> variable manipulation features: >> >> http://wiki.freeswitch.org/wiki/Manipulating_Channel_Variables >> >> -MC >> >> On Wed, Feb 1, 2012 at 1:19 AM, Miha Zoubek wrote: >> >>> Hi, >>> >>> how can I strip destination number in my dialplan if the destination >>> number is not in condition option. >>> >>> >> data="sofia/external/386${sip_redirect_contact_user_0}@xxx.xxx.xxx.xxx"/> >>> I need to stip leading zero from ${sip_redirect_contact_user_0} variable. >>> >>> Thanks! >>> Miha >>> >>> >>> -- >>> Best regards / Lep Pozdrav >>> Miha Zoubek >>> Softnet d.o.o. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> 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 >>> http://lists.freeswitch.org/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 >> >> 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 >> >> >> >> -- >> Best regards / Lep Pozdrav >> Miha Zoubek >> Softnet d.o.o. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/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 > > 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 > > > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > 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 > > > > -- > Best regards / Lep Pozdrav > Miha Zoubek > Softnet d.o.o. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/faeaacec/attachment.html From gb10hkzo-freeswitch at yahoo.co.uk Mon Feb 6 15:06:52 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Mon, 6 Feb 2012 12:06:52 +0000 (GMT) Subject: [Freeswitch-users] Getting USER_NOT_REGISTERED for no reason ? Message-ID: <1328530012.19186.YahooMailNeo@web29402.mail.ird.yahoo.com> As you can see from http://pastebin.freeswitch.org/18299, I have a Snom handset successfully registered with my FreeSwitch instance over an OpenVPN connection. As per the trace, I can make outbound calls without issue.? But inbound, FreeSwithc seems to think I am not registered when I am ? To make absolutley sure I am registered, I have issued a " sofia profile flush_inbound_reg reboot" and sure enough, the Snom unit reboots and re-registeres without issue. Help ! Thanks Bob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/06d7832c/attachment-0001.html From jaasmailing at gmail.com Mon Feb 6 15:09:58 2012 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Mon, 06 Feb 2012 13:09:58 +0100 Subject: [Freeswitch-users] Custom filename - XML CDR Message-ID: <4F2FC316.9030002@gmail.com> Hi Guys, I need a prefix in the filename created my mod_xml_cdr (ex: -.cdr.xml). Is there any way to set this custom filename in the dialplan? A reason could be performance in the billing procedure in order to process only that particular account and not the entire directory. Best regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/102571e7/attachment.html From avi at avimarcus.net Mon Feb 6 15:17:33 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 6 Feb 2012 14:17:33 +0200 Subject: [Freeswitch-users] Custom filename - XML CDR In-Reply-To: <4F2FC316.9030002@gmail.com> References: <4F2FC316.9030002@gmail.com> Message-ID: Without hacking the module to add that, cdr_csv and posting the xml_cdr are two options you can use. http://wiki.freeswitch.org/wiki/Mod_cdr_csv lets you have different templates for different accountcodes to you can customize what it saves. Or, have mod_xml_cdr post to a script that does different things depending on the accountcode. You could patch mod_xml_cdr - the filename is chosen for normal saving in this line: freeswitch/src/mod/xml_int/mod_xml_cdr/mod_xml_cdr.c: path = switch_mprintf("%s%s%s%s.cdr.xml", logdir, SWITCH_PATH_SEPARATOR, a_prefix, switch_core_session_get_uuid(session)); -Avi On Mon, Feb 6, 2012 at 2:09 PM, Carlo Dimaggio wrote: > Hi Guys, > > I need a prefix in the filename created my mod_xml_cdr (ex: or other var>-.cdr.xml). > Is there any way to set this custom filename in the dialplan? > > A reason could be performance in the billing procedure in order to process > only that particular account and not the entire directory. > > > Best 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/bcb0b186/attachment.html From bdfoster at endigotech.com Mon Feb 6 15:33:02 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 6 Feb 2012 07:33:02 -0500 Subject: [Freeswitch-users] Getting USER_NOT_REGISTERED for no reason ? In-Reply-To: <1328530012.19186.YahooMailNeo@web29402.mail.ird.yahoo.com> References: <1328530012.19186.YahooMailNeo@web29402.mail.ird.yahoo.com> Message-ID: You are using the internal profile when your phone is clearly registered on the openvpn profile. On Feb 6, 2012 7:08 AM, "Bob Smith" wrote: > As you can see from http://pastebin.freeswitch.org/18299, I have a Snom > handset successfully registered with my FreeSwitch instance over an OpenVPN > connection. > > As per the trace, I can make outbound calls without issue. But inbound, > FreeSwithc seems to think I am not registered when I am ? > > To make absolutley sure I am registered, I have issued a " sofia profile > flush_inbound_reg reboot" and sure enough, the Snom unit > reboots and re-registeres without issue. > > Help ! > > Thanks > > Bob > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/6f95a0e3/attachment.html From bdfoster at endigotech.com Mon Feb 6 15:36:10 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 6 Feb 2012 07:36:10 -0500 Subject: [Freeswitch-users] event_socket.conf In-Reply-To: References: <4F2F9F3A.1080306@softnet.si> <4F2FAAFA.10406@softnet.si> Message-ID: I cannot reproduce on latest git head. On Feb 6, 2012 5:36 AM, "Avi Marcus" wrote: > Please file a http://jira.freeswitch.org with exactly what worked and > what didn't. > > -Avi > > > On Mon, Feb 6, 2012 at 12:27 PM, Miha Zoubek wrote: > >> Hi, >> >> just to let you know, because this is very wird. >> >> My password in event_socket.conf was reload52166XML. >> From remote I could connection, from localhost fs_cli -p my_pass I was >> unable to make a connection. >> >> Now I have set password with just a words and it works from remote and >> locally. >> >> >> >> Thanks! >> Miha >> >> >> On 2/6/2012 10:53 AM, Avi Marcus wrote: >> >> As per: http://wiki.freeswitch.org/wiki/Fs_cli >> You can use fs_cli -p clueclone >> >> Or.. >> >> "An optional configuration file can be set up in the user's home >> directory. The file name is *.fs_cli_conf*. " see the syntax on the wiki. >> >> -Avi >> >> >> On Mon, Feb 6, 2012 at 11:36 AM, Miha Zoubek wrote: >> >>> Hi, >>> >>> I have changed default password for event_socket.conf (clueclone). Now I >>> am unable to connect to FS via fs_cli (I can connect remotely). >>> >>> I have also set ACL which will also do it for localhost. Where can I >>> find configuration file for FS_cli that I can change default password (If I >>> set password to clueclone in event_socket.conf it works). >>> >>> I have try to change it in: /usr/local/src/freeswitch/libs/esl/fs_cli, >>> but I guess this is not the right file. >>> >>> My ACL: >>> >>> >>> >>> >>> >>> >>> >>> >>> Thanks! >>> >>> Regards, >>> Miha >>> >>> -- >>> Best regards / Lep Pozdrav >>> Miha Zoubek >>> Softnet d.o.o. >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> 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 >>> http://lists.freeswitch.org/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 >> >> 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 >> >> >> >> -- >> Best regards / Lep Pozdrav >> Miha Zoubek >> Softnet d.o.o. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/61893db4/attachment-0001.html From gb10hkzo-freeswitch at yahoo.co.uk Mon Feb 6 15:43:01 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Mon, 6 Feb 2012 12:43:01 +0000 (GMT) Subject: [Freeswitch-users] Getting USER_NOT_REGISTERED for no reason ? Message-ID: <1328532181.6461.YahooMailNeo@web29405.mail.ird.yahoo.com> I am confused.... ;-( Basically, the only reason the openvpn profile exists is to enable FreeSwitch to listen on the local OpenVPN IP.?? Other than that, I want OpenVPN profile users to be treated the same way as Internal profile users ? In order to achieve that goal, do I need to do more than just copy internal.xml to a new file and change the profile name and IP addresses ?? > You are using the internal profile when your phone is clearly registered on the openvpn profile. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/8c4183f4/attachment.html From jaasmailing at gmail.com Mon Feb 6 16:50:51 2012 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Mon, 06 Feb 2012 14:50:51 +0100 Subject: [Freeswitch-users] Custom filename - XML CDR In-Reply-To: References: <4F2FC316.9030002@gmail.com> Message-ID: <4F2FDABB.9090000@gmail.com> Ok, thanks. I think is better posting the xml_cdr. How I can configure mod_xml_cdr to insert in the file only the variabiles I need? The file contains a lot of information that are not useful for the billing process. Regards Il 06/02/12 13.17, Avi Marcus ha scritto: > Without hacking the module to add that, cdr_csv and posting the > xml_cdr are two options you can use. > http://wiki.freeswitch.org/wiki/Mod_cdr_csv lets you have different > templates for different accountcodes to you can customize what it saves. > > Or, have mod_xml_cdr post to a script that does different things > depending on the accountcode. > > You could patch mod_xml_cdr - the filename is chosen for normal saving > in this line: > freeswitch/src/mod/xml_int/mod_xml_cdr/mod_xml_cdr.c: > path = switch_mprintf("%s%s%s%s.cdr.xml", logdir, > SWITCH_PATH_SEPARATOR, a_prefix, switch_core_session_get_uuid(session)); > > -Avi > > > On Mon, Feb 6, 2012 at 2:09 PM, Carlo Dimaggio > wrote: > > Hi Guys, > > I need a prefix in the filename created my mod_xml_cdr (ex: > -.cdr.xml). > Is there any way to set this custom filename in the dialplan? > > A reason could be performance in the billing procedure in order to > process only that particular account and not the entire directory. > > > Best 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/9faaedf3/attachment.html From avi at avimarcus.net Mon Feb 6 16:56:52 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 6 Feb 2012 15:56:52 +0200 Subject: [Freeswitch-users] Custom filename - XML CDR In-Reply-To: <4F2FDABB.9090000@gmail.com> References: <4F2FC316.9030002@gmail.com> <4F2FDABB.9090000@gmail.com> Message-ID: You can't. xml_cdr is designed to show you everything, then you use your own thing to only save what you want.. or all of it. If you only want a few channel variables, give cdr_csv a try.. or cdr_sqlite maybe. Both of those let you specify which channel variables to save. See: http://wiki.freeswitch.org/wiki/Cdr -Avi On Mon, Feb 6, 2012 at 3:50 PM, Carlo Dimaggio wrote: > Ok, thanks. > I think is better posting the xml_cdr. > How I can configure mod_xml_cdr to insert in the file only the variabiles > I need? The file contains a lot of information that are not useful for the > billing process. > > Regards > > > > Il 06/02/12 13.17, Avi Marcus ha scritto: > > Without hacking the module to add that, cdr_csv and posting the xml_cdr > are two options you can use. > http://wiki.freeswitch.org/wiki/Mod_cdr_csv lets you have different > templates for different accountcodes to you can customize what it saves. > > Or, have mod_xml_cdr post to a script that does different things > depending on the accountcode. > > You could patch mod_xml_cdr - the filename is chosen for normal saving > in this line: > freeswitch/src/mod/xml_int/mod_xml_cdr/mod_xml_cdr.c: > path = switch_mprintf("%s%s%s%s.cdr.xml", logdir, > SWITCH_PATH_SEPARATOR, a_prefix, switch_core_session_get_uuid(session)); > > -Avi > > > On Mon, Feb 6, 2012 at 2:09 PM, Carlo Dimaggio wrote: > >> Hi Guys, >> >> I need a prefix in the filename created my mod_xml_cdr (ex: > or other var>-.cdr.xml). >> Is there any way to set this custom filename in the dialplan? >> >> A reason could be performance in the billing procedure in order to >> process only that particular account and not the entire directory. >> >> >> Best 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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/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 > > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/5d94db82/attachment-0001.html From jaasmailing at gmail.com Mon Feb 6 17:03:38 2012 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Mon, 06 Feb 2012 15:03:38 +0100 Subject: [Freeswitch-users] Custom filename - XML CDR In-Reply-To: References: <4F2FC316.9030002@gmail.com> <4F2FDABB.9090000@gmail.com> Message-ID: <4F2FDDBA.6020708@gmail.com> Cdr_csv and cdr_sqlite don't create a single file per call. And no posting feature. I need these features for real time billing (I though to use xml_cdr and log err files in case of unavailability of http server). Real time is important to calculate quota (postpaid) and credit (prepaid). Il 06/02/12 14.56, Avi Marcus ha scritto: > You can't. xml_cdr is designed to show you everything, then you use > your own thing to only save what you want.. or all of it. > > If you only want a few channel variables, give cdr_csv a try.. or > cdr_sqlite maybe. Both of those let you specify which channel > variables to save. See: http://wiki.freeswitch.org/wiki/Cdr > > -Avi > > > On Mon, Feb 6, 2012 at 3:50 PM, Carlo Dimaggio > wrote: > > Ok, thanks. > I think is better posting the xml_cdr. > How I can configure mod_xml_cdr to insert in the file only the > variabiles I need? The file contains a lot of information that are > not useful for the billing process. > > Regards > > > > Il 06/02/12 13.17, Avi Marcus ha scritto: >> Without hacking the module to add that, cdr_csv and posting the >> xml_cdr are two options you can use. >> http://wiki.freeswitch.org/wiki/Mod_cdr_csv lets you have >> different templates for different accountcodes to you can >> customize what it saves. >> >> Or, have mod_xml_cdr post to a script that does different things >> depending on the accountcode. >> >> You could patch mod_xml_cdr - the filename is chosen for normal >> saving in this line: >> freeswitch/src/mod/xml_int/mod_xml_cdr/mod_xml_cdr.c: >> path = switch_mprintf("%s%s%s%s.cdr.xml", logdir, >> SWITCH_PATH_SEPARATOR, a_prefix, >> switch_core_session_get_uuid(session)); >> >> -Avi >> >> >> On Mon, Feb 6, 2012 at 2:09 PM, Carlo Dimaggio >> > wrote: >> >> Hi Guys, >> >> I need a prefix in the filename created my mod_xml_cdr (ex: >> -.cdr.xml). >> Is there any way to set this custom filename in the dialplan? >> >> A reason could be performance in the billing procedure in >> order to process only that particular account and not the >> entire directory. >> >> >> Best 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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/129e9730/attachment.html From avi at avimarcus.net Mon Feb 6 17:29:36 2012 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 6 Feb 2012 16:29:36 +0200 Subject: [Freeswitch-users] Custom filename - XML CDR In-Reply-To: <4F2FDDBA.6020708@gmail.com> References: <4F2FC316.9030002@gmail.com> <4F2FDABB.9090000@gmail.com> <4F2FDDBA.6020708@gmail.com> Message-ID: You don't need a single file per call to do billing... I personally POST and have my import script do the billing management. You could use a lua hangup-hook to do your billing. Or a perl script that tails the cdr_csv file and keeps track of where it's up to. Or several other options.. "One file per call" is not necessary for realtime.. -Avi On Mon, Feb 6, 2012 at 4:03 PM, Carlo Dimaggio wrote: > Cdr_csv and cdr_sqlite don't create a single file per call. And no > posting feature. > I need these features for real time billing (I though to use xml_cdr and > log err files in case of unavailability of http server). Real time is > important to calculate quota (postpaid) and credit (prepaid). > > > > > Il 06/02/12 14.56, Avi Marcus ha scritto: > > You can't. xml_cdr is designed to show you everything, then you use your > own thing to only save what you want.. or all of it. > > If you only want a few channel variables, give cdr_csv a try.. or > cdr_sqlite maybe. Both of those let you specify which channel variables to > save. See: http://wiki.freeswitch.org/wiki/Cdr > > -Avi > > > On Mon, Feb 6, 2012 at 3:50 PM, Carlo Dimaggio wrote: > >> Ok, thanks. >> I think is better posting the xml_cdr. >> How I can configure mod_xml_cdr to insert in the file only the variabiles >> I need? The file contains a lot of information that are not useful for the >> billing process. >> >> Regards >> >> >> >> Il 06/02/12 13.17, Avi Marcus ha scritto: >> >> Without hacking the module to add that, cdr_csv and posting the xml_cdr >> are two options you can use. >> http://wiki.freeswitch.org/wiki/Mod_cdr_csv lets you have different >> templates for different accountcodes to you can customize what it saves. >> >> Or, have mod_xml_cdr post to a script that does different things >> depending on the accountcode. >> >> You could patch mod_xml_cdr - the filename is chosen for normal saving >> in this line: >> freeswitch/src/mod/xml_int/mod_xml_cdr/mod_xml_cdr.c: >> path = switch_mprintf("%s%s%s%s.cdr.xml", logdir, >> SWITCH_PATH_SEPARATOR, a_prefix, switch_core_session_get_uuid(session)); >> >> -Avi >> >> >> On Mon, Feb 6, 2012 at 2:09 PM, Carlo Dimaggio wrote: >> >>> Hi Guys, >>> >>> I need a prefix in the filename created my mod_xml_cdr (ex: >> or other var>-.cdr.xml). >>> Is there any way to set this custom filename in the dialplan? >>> >>> A reason could be performance in the billing procedure in order to >>> process only that particular account and not the entire directory. >>> >>> >>> Best 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 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/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 >> >> 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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/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 > > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/b1848a15/attachment-0001.html From jaasmailing at gmail.com Mon Feb 6 17:54:33 2012 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Mon, 06 Feb 2012 15:54:33 +0100 Subject: [Freeswitch-users] Custom filename - XML CDR In-Reply-To: References: <4F2FC316.9030002@gmail.com> <4F2FDABB.9090000@gmail.com> <4F2FDDBA.6020708@gmail.com> Message-ID: <4F2FE9A9.8020403@gmail.com> Il 06/02/12 15.29, Avi Marcus ha scritto: > You don't need a single file per call to do billing... > I personally POST and have my import script do the billing management. what about network problems? For example, if the web server is unreachable the call must be stored on the filesystem. Suppose that arrives a new call, in this case the state of the billing system is inconsistent and must be resynchronized before sending quota or credit info. One file per call allows me to process just the single (or multiple) call without the need to parse the entire cdr file (after this process, I could move the file to another directory). perl + cdr_csv could be interesting but no posting. What do you think about? From brian at freeswitch.org Mon Feb 6 17:56:45 2012 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Feb 2012 08:56:45 -0600 Subject: [Freeswitch-users] Getting USER_NOT_REGISTERED for no reason ? In-Reply-To: <1328532181.6461.YahooMailNeo@web29405.mail.ird.yahoo.com> References: <1328532181.6461.YahooMailNeo@web29405.mail.ird.yahoo.com> Message-ID: <34321EE0-CF74-42E1-8988-699D1A814ACB@freeswitch.org> You're getting it because it can't find the USER to call. How are you calling the users on this interface in your dialplan? /b On Feb 6, 2012, at 6:43 AM, Bob Smith wrote: > I am confused.... ;-( > > Basically, the only reason the openvpn profile exists is to enable FreeSwitch to > listen on the local OpenVPN IP. Other than that, I want OpenVPN > profile users to be treated the same way as Internal profile users ? > > In order to achieve that goal, do I need to do more than just copy > internal.xml to a new file and change the profile name and IP addresses > ? -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/f9806d1f/attachment.html From krice at freeswitch.org Mon Feb 6 18:03:10 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 06 Feb 2012 09:03:10 -0600 Subject: [Freeswitch-users] Custom filename - XML CDR In-Reply-To: <4F2FE9A9.8020403@gmail.com> Message-ID: XML_CDR allows this now... And you don't have to use the posting... Also, it allows for single call exceptions Drop the CDRs to the File System Use favorite scripting language to scrape and process, if exception move to exception directory and fire alert, if good move to done directory, If no DB connection, time out and wait a minute before even processing them in the first place Even you use the post method, you still have to do the above steps to make sure a post didn't fail... Also you, this way you don't have to SIGHUP FS constantly to rotate the CSV CDRs... On 2/6/12 8:54 AM, "Carlo Dimaggio" wrote: > Il 06/02/12 15.29, Avi Marcus ha scritto: >> You don't need a single file per call to do billing... >> I personally POST and have my import script do the billing management. > what about network problems? For example, if the web server is > unreachable the call must be stored on the filesystem. Suppose that > arrives a new call, in this case the state of the billing system is > inconsistent and must be resynchronized before sending quota or credit info. > One file per call allows me to process just the single (or multiple) > call without the need to parse the entire cdr file (after this process, > I could move the file to another directory). > > perl + cdr_csv could be interesting but no posting. > > What do you think about? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From jaasmailing at gmail.com Mon Feb 6 18:33:51 2012 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Mon, 06 Feb 2012 16:33:51 +0100 Subject: [Freeswitch-users] Custom filename - XML CDR In-Reply-To: References: Message-ID: <4F2FF2DF.2020705@gmail.com> Il 06/02/12 16.03, Ken Rice ha scritto: > XML_CDR allows this now... And you don't have to use the posting... Also, it > allows for single call exceptions > > Drop the CDRs to the File System > Use favorite scripting language to scrape and process, if exception move to > exception directory and fire alert, if good move to done directory, > If no DB connection, time out and wait a minute before even processing them > in the first place this is the same process that I can handle with mod_xml_cdr and post configuration (log-dir, err-log-dir) plus a custom "retriever" script, isn't it? The only difference is who sends the call cdr... XML_CDR Post allows me to use failover and https features out of the box. However in a fs load balancing scenario, after the import process, I need another routine that sort calls and bill the account. From gb10hkzo-freeswitch at yahoo.co.uk Mon Feb 6 18:35:53 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Mon, 6 Feb 2012 15:35:53 +0000 (GMT) Subject: [Freeswitch-users] Getting USER_NOT_REGISTERED for no reason ? Message-ID: <1328542553.92989.YahooMailNeo@web29406.mail.ird.yahoo.com> ?In the public dial plan?? ============================== ? ? ? ?? ? ? "1000" being a user under directory/default/ =========================================== ? ??? ??? ? ????? ?????? ??? ??? ????? ????? ????? ????? ??? --> ????? ????? ????? ??? ? >You're getting it because it can't find the USER to call. How are you calling the users on this interface in your dialplan? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/0a3faa72/attachment.html From krice at freeswitch.org Mon Feb 6 18:46:50 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 06 Feb 2012 09:46:50 -0600 Subject: [Freeswitch-users] Custom filename - XML CDR In-Reply-To: <4F2FF2DF.2020705@gmail.com> Message-ID: The thing is you don't have to depend on yet another piece of software... In this case apache or nginx and their respective CGI modules... Not to mention you now don't need a custom retrieve script... The processor can run right on the FS instance in parallel with FreeSWITCH... This also scales to crazy level (I have personally processed 500million CDRs in a single month this way and still had headroom to process more... The real limitation with this method then becomes your DB system and most likely its underlying disk subsystem. The biggest bottle necks I have seen are on Insert IO... K On 2/6/12 9:33 AM, "Carlo Dimaggio" wrote: > Il 06/02/12 16.03, Ken Rice ha scritto: >> XML_CDR allows this now... And you don't have to use the posting... Also, it >> allows for single call exceptions >> >> Drop the CDRs to the File System >> Use favorite scripting language to scrape and process, if exception move to >> exception directory and fire alert, if good move to done directory, >> If no DB connection, time out and wait a minute before even processing them >> in the first place > > this is the same process that I can handle with mod_xml_cdr and post > configuration (log-dir, err-log-dir) plus a custom "retriever" script, > isn't it? The only difference is who sends the call cdr... > XML_CDR Post allows me to use failover and https features out of the box. > However in a fs load balancing scenario, after the import process, I > need another routine that sort calls and bill the account. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > 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 Mon Feb 6 18:52:09 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Mon, 6 Feb 2012 12:52:09 -0300 Subject: [Freeswitch-users] Custom filename - XML CDR In-Reply-To: References: Message-ID: <72A9CD6627414891A02B158D548535D1@freeswitch.org> That's the big problem with CDR processing. If you do it right, no matter how correct, you will always hit the wall with IO. Then you have to go and tell the customer that he needs to spend money on storage. Some of them just won't understand? LOL -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Monday, February 6, 2012 at 12:46 PM, Ken Rice wrote: > The thing is you don't have to depend on yet another piece of software... In > this case apache or nginx and their respective CGI modules... > > Not to mention you now don't need a custom retrieve script... The processor > can run right on the FS instance in parallel with FreeSWITCH... > > This also scales to crazy level (I have personally processed 500million CDRs > in a single month this way and still had headroom to process more... The > real limitation with this method then becomes your DB system and most likely > its underlying disk subsystem. The biggest bottle necks I have seen are on > Insert IO... > > K > > > On 2/6/12 9:33 AM, "Carlo Dimaggio" wrote: > > > Il 06/02/12 16.03, Ken Rice ha scritto: > > > XML_CDR allows this now... And you don't have to use the posting... Also, it > > > allows for single call exceptions > > > > > > Drop the CDRs to the File System > > > Use favorite scripting language to scrape and process, if exception move to > > > exception directory and fire alert, if good move to done directory, > > > If no DB connection, time out and wait a minute before even processing them > > > in the first place > > > > > > > > > this is the same process that I can handle with mod_xml_cdr and post > > configuration (log-dir, err-log-dir) plus a custom "retriever" script, > > isn't it? The only difference is who sends the call cdr... > > XML_CDR Post allows me to use failover and https features out of the box. > > However in a fs load balancing scenario, after the import process, I > > need another routine that sort calls and bill the account. > > > > > > _________________________________________________________________________ > > 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 > > > > > > > _________________________________________________________________________ > 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/20120206/254757e3/attachment-0001.html From brian at freeswitch.org Mon Feb 6 20:16:35 2012 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Feb 2012 11:16:35 -0600 Subject: [Freeswitch-users] Getting USER_NOT_REGISTERED for no reason ? In-Reply-To: <1328542553.92989.YahooMailNeo@web29406.mail.ird.yahoo.com> References: <1328542553.92989.YahooMailNeo@web29406.mail.ird.yahoo.com> Message-ID: You'll have to modify the dial-string for the user 1000 to make sure it searches this new profile ... /b On Feb 6, 2012, at 9:35 AM, Bob Smith wrote: > In the public dial plan?? > ============================== > > > > > > > > > "1000" being a user under directory/default/ > =========================================== > > > > > > > > > > > > > --> > > > > > > > > > > >> You're getting it because it can't find the USER to call. How are you calling the users on this interface in your dialplan? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/7128658a/attachment.html From brian at freeswitch.org Mon Feb 6 20:25:01 2012 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Feb 2012 11:25:01 -0600 Subject: [Freeswitch-users] I'm Alive! Message-ID: I would like to thank everyone that has donated money to help me pay for all the expense of having a stay in the hospital. It was a pretty scary week for me last week. If it were not for the morphine last week would have sucked even more! I have missed out on the annual Developer meeting in Milwaukee this year due to this event. So please make sure you guys donate some money for the dinner for those guys. Again Thank you all for your support. -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/4b38be49/attachment.html From basit.engg at gmail.com Mon Feb 6 20:44:05 2012 From: basit.engg at gmail.com (Abdul Basit) Date: Mon, 6 Feb 2012 22:44:05 +0500 Subject: [Freeswitch-users] I'm Alive! In-Reply-To: References: Message-ID: Good news. Welcome back. -- Regards, Abdul Basit On Mon, Feb 6, 2012 at 10:25 PM, Brian West wrote: > I would like to thank everyone that has donated money to help me pay for > all the expense of having a stay in the hospital. It was a pretty scary > week for me last week. If it were not for the morphine last week would > have sucked even more! I have missed out on the annual Developer meeting > in Milwaukee this year due to this event. So please make sure you guys > donate some money for the dinner for those guys. > > Again Thank you all for your support. > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/de5b054d/attachment.html From gb10hkzo-freeswitch at yahoo.co.uk Mon Feb 6 21:05:43 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Mon, 6 Feb 2012 18:05:43 +0000 (GMT) Subject: [Freeswitch-users] Getting USER_NOT_REGISTERED for no reason ? Message-ID: <1328551543.31598.YahooMailNeo@web29404.mail.ird.yahoo.com> Welcome back by the way Brian... ;-) If only it were that easy...... in conf/directory/default.xml I changed.... to read Then the OpenVPN phone rings, but the internally registered phone does not ?? > You'll have to modify the dial-string for the user 1000 to make sure it searches this new profile ... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/6c6ff107/attachment.html From tm at supertec.com Mon Feb 6 10:50:28 2012 From: tm at supertec.com (Tariq Mahmood) Date: Mon, 6 Feb 2012 12:50:28 +0500 Subject: [Freeswitch-users] One Way Audio, from one side. Message-ID: Hi, I hope you guys are doing great. I am having an issue with my number, When i dial the number with my asterisk server, the callee can hear me but i cant hear him. But if the same guy call me, the conversation goes great. Any Help??? -- Best Regards Tareq Khan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/8255930a/attachment-0001.html From tm at supertec.com Mon Feb 6 13:00:09 2012 From: tm at supertec.com (Tariq Mahmood) Date: Mon, 6 Feb 2012 15:00:09 +0500 Subject: [Freeswitch-users] One way Audio when dialing from one end. Message-ID: Hi, I hope you guys are doing great. I am having an issue with my number, When i dial the number with my asterisk server, the callee can hear me but i cant hear him. But if the same guy call me, the conversation goes great. Any Help??? -- Best Regards Tareq Khan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/e877538f/attachment-0001.html From asrivas at gmail.com Mon Feb 6 21:48:34 2012 From: asrivas at gmail.com (Anurag Srivastava) Date: Mon, 6 Feb 2012 10:48:34 -0800 Subject: [Freeswitch-users] external_sip_ip not refreshed after ip address change. In-Reply-To: References: Message-ID: Can somebody help me with this? I have a question about nat behavior in freeswitch. Basically my external > calls to freeswitch are getting disconnected after 30 seconds of two way > audio when my external ip address changes. I have dhcp from my ISP and am > using external_sip_ip and external_rtp_ip as stun:. When my IP > changes I see that external_sip_ip does not get refreshed but > external_rtp_ip does. I am not allowed to enable upnp/nat-pmp on my router. > Apparently it is a known issue that external_sip_ip is read just > at load time and not refreshed even if it is specified in stun format > Is there a fix to this problem? > > There is always the option of restarting profile when ddclient notes an ip > change. Is there something inbuilt into FS. It is already finding that ip > address has changed as reflected in external_rtp_ip which does use stun and > gets the right ip address. > -- Regards Anurag -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/513d3daf/attachment.html From brian at freeswitch.org Mon Feb 6 22:04:07 2012 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Feb 2012 13:04:07 -0600 Subject: [Freeswitch-users] One Way Audio, from one side. In-Reply-To: References: Message-ID: <20E91FA1-F4C7-493C-8D0B-46A08E9AF230@freeswitch.org> Check the sip trace. /b On Feb 6, 2012, at 1:50 AM, Tariq Mahmood wrote: > Any Help??? -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/1aa8410c/attachment.html From brian at freeswitch.org Mon Feb 6 22:05:02 2012 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Feb 2012 13:05:02 -0600 Subject: [Freeswitch-users] external_sip_ip not refreshed after ip address change. In-Reply-To: References: Message-ID: <-- try this in the main sofia.conf.xml /b On Feb 6, 2012, at 12:48 PM, Anurag Srivastava wrote: > Can somebody help me with this? > > I have a question about nat behavior in freeswitch. Basically my external >> calls to freeswitch are getting disconnected after 30 seconds of two way >> audio when my external ip address changes. I have dhcp from my ISP and am >> using external_sip_ip and external_rtp_ip as stun:. When my IP >> changes I see that external_sip_ip does not get refreshed but >> external_rtp_ip does. I am not allowed to enable upnp/nat-pmp on my router. >> Apparently it is a known issue that external_sip_ip is read just >> at load time and not refreshed even if it is specified in stun format >> Is there a fix to this problem? >> >> There is always the option of restarting profile when ddclient notes an ip >> change. Is there something inbuilt into FS. It is already finding that ip >> address has changed as reflected in external_rtp_ip which does use stun and >> gets the right ip address. >> > > > > -- > Regards > Anurag > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org From brian at freeswitch.org Mon Feb 6 22:07:02 2012 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Feb 2012 13:07:02 -0600 Subject: [Freeswitch-users] Getting USER_NOT_REGISTERED for no reason ? In-Reply-To: <1328551543.31598.YahooMailNeo@web29404.mail.ird.yahoo.com> References: <1328551543.31598.YahooMailNeo@web29404.mail.ird.yahoo.com> Message-ID: <8E5654DC-822C-4D8F-A9E2-C5BCB6153112@freeswitch.org> OK just put a dial-string param on that ONE user: (You'll have to make sure sofia_contact profile/user at domain resolves at fs_cli first... ) The starts just aren't lined up yet and this is why you MUST understand how the parts of freeswitch interact and cause this to either work or not work. /b On Feb 6, 2012, at 12:05 PM, Bob Smith wrote: > Welcome back by the way Brian... ;-) > > If only it were that easy...... in conf/directory/default.xml I changed.... > > > > to read > > > > > > > Then the OpenVPN phone rings, but the internally registered phone does not ? > > > >> You'll have to modify the dial-string for the user 1000 to make sure it searches this new profile ... > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org From brian at freeswitch.org Mon Feb 6 22:09:56 2012 From: brian at freeswitch.org (Brian West) Date: Mon, 6 Feb 2012 13:09:56 -0600 Subject: [Freeswitch-users] Getting USER_NOT_REGISTERED for no reason ? In-Reply-To: <1328551543.31598.YahooMailNeo@web29404.mail.ird.yahoo.com> References: <1328551543.31598.YahooMailNeo@web29404.mail.ird.yahoo.com> Message-ID: btw LOOK at the logs... bet it says something about only dialing one element... use a :_: instead of a , might work also. uri:_:uri:_:uri vs uri,uri,uri /b On Feb 6, 2012, at 12:05 PM, Bob Smith wrote: > > Then the OpenVPN phone rings, but the internally registered phone does not ? -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/2489aa64/attachment.html From joe.jflemmings at gmail.com Mon Feb 6 23:05:30 2012 From: joe.jflemmings at gmail.com (Joe Flemmings) Date: Mon, 6 Feb 2012 12:05:30 -0800 Subject: [Freeswitch-users] FreeSwtich DB In-Reply-To: References: Message-ID: In my email, you can see that i said FreeSwitch creates the database files. It creates both the ODBC(MySQL) and DB files. Questions is WHY? On Fri, Feb 3, 2012 at 10:36 AM, Brad Mina wrote: > Check your odbc database and make sure FS has created (and uses) that > database. If it's creating db files, it's probably not using your odbc db. > > On Thu, Feb 2, 2012 at 12:07 AM, Joe Flemmings wrote: > >> Is their a reason why freeswitch still creates DB files in >> /usr/local/freeswitch/db/ even when using ODBC at the core as decsribed >> here http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/14f92305/attachment-0001.html From dgarcia at anew.com.ve Mon Feb 6 23:15:19 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Mon, 06 Feb 2012 15:45:19 -0430 Subject: [Freeswitch-users] Mod_dingaling and openfire issue In-Reply-To: References: <1328551543.31598.YahooMailNeo@web29404.mail.ird.yahoo.com> Message-ID: <4F3034D7.4060906@anew.com.ve> Hi, I am testing Openfire and FS (mod_dingaling) Openfire and FS are installed on same server. Before install Openfire, FS worked fine After install Openfire, FS just exit when openfire send messages to FS or when a just execute the command: chat jingle....... The message I receive is "Socket interrupted ...." This is what a got in fs_cli before exit 2012-02-06 15:22:00.155318 [NOTICE] libdingaling.c:1373 SEND: ------------------------------------------------------------------------------- 2012-02-06 15:22:00.175334 [INFO] libdingaling.c:1371 RECV: ------------------------------------------------------------------------------- 2012-02-06 15:22:00.175334 [NOTICE] libdingaling.c:1373 SEND: ------------------------------------------------------------------------------- b33672a70d17d0d5020b141b1dff686493def49c 2012-02-06 15:22:00.175334 [INFO] libdingaling.c:1371 RECV: ------------------------------------------------------------------------------- 2012-02-06 15:22:00.175334 [DEBUG] libdingaling.c:1194 XMPP authenticated 2012-02-06 15:22:00.215322 [INFO] libdingaling.c:1371 RECV: ------------------------------------------------------------------------------- 2012-02-06 15:22:00.215322 [NOTICE] libdingaling.c:1373 SEND: ------------------------------------------------------------------------------- ----------- my dingle xml file: Thanks in advance From ira at connectmevoice.com Mon Feb 6 23:30:35 2012 From: ira at connectmevoice.com (Ira Tessler) Date: Mon, 6 Feb 2012 15:30:35 -0500 Subject: [Freeswitch-users] No outbound audio on new fs build Message-ID: The following are two pcap files that have one outbound call in them. Everything is identical except the version of Freeswitch. With the later version of Freeswitch, I am getting one way audio on the outbound call. On the later version of Freeswitch, there is an Update sip message that is not in the older version. Can anyone help me and let me know how I can fix the one way audio? I only get the one way audio when using Yealink phones behind an Edgemarc router. Old version of Freeswitch pcap: http://dl.dropbox.com/u/47191639/lan_side_yealink_t28_good_audio_edgemarc_old_fs_version.pcap New Version of Freeswitch pcap: http://dl.dropbox.com/u/47191639/lan_side_yealink_t28_no_audio_edgemarc.pcap Thanks, Ira Tessler ConnectMe (732) 490-9007 x2 www.connectmevoice.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/cf11a61f/attachment.html From nasida at live.ru Mon Feb 6 23:32:07 2012 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 7 Feb 2012 00:32:07 +0400 Subject: [Freeswitch-users] voicemail_say_phone_number Message-ID: Hello list, I would like to have one simple ability to listen the phone number of caller when I check my voicemail. At present I listen date of message only. I have found the macro "voicemail_say_phone_number" in conf/lang/en/vm/sounds.xml. But... when FS uses it ? Can I get it without modifying of source code of voicemail module ? Please advise.Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/48f84ee4/attachment.html From Kyle.Haefner at colostate.edu Mon Feb 6 22:58:47 2012 From: Kyle.Haefner at colostate.edu (Kyle Haefner) Date: Mon, 6 Feb 2012 12:58:47 -0700 Subject: [Freeswitch-users] xml_curl dialplan and static files Message-ID: Hi All, I have not been able to glean enough from the wiki and Google searches to figure this out. I have freeswitch xml_curl fetching a dial plan. What I would *like* to do is have one context fetched dynamically while another context is from a static file (or fail-back to a static file). I would appreciate any help. Thanks! -- Kyle Haefner, M.S. Communication Systems Programmer Colorado State University Fort Collins, CO Phone: 970-491-1012 Email: ?kyle.haefner at colostate.edu From nasida at live.ru Mon Feb 6 23:43:26 2012 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 7 Feb 2012 00:43:26 +0400 Subject: [Freeswitch-users] xml_curl dialplan and static files In-Reply-To: References: Message-ID: You can use them both simultaneously (context from static xml file and curl ) but curl will have priority as far as I remember. Thanks. > Date: Mon, 6 Feb 2012 12:58:47 -0700 > From: Kyle.Haefner at colostate.edu > To: freeswitch-users at lists.freeswitch.org > Subject: [Freeswitch-users] xml_curl dialplan and static files > > Hi All, > > I have not been able to glean enough from the wiki and Google searches > to figure this out. > > I have freeswitch xml_curl fetching a dial plan. What I would *like* > to do is have one context fetched > dynamically while another context is from a static file (or fail-back > to a static file). > > I would appreciate any help. > > Thanks! > > > > > -- > Kyle Haefner, M.S. > Communication Systems Programmer > Colorado State University > Fort Collins, CO > Phone: 970-491-1012 > Email: kyle.haefner at colostate.edu > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/44640bd7/attachment.html From bdfoster at endigotech.com Tue Feb 7 00:16:50 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Mon, 6 Feb 2012 16:16:50 -0500 Subject: [Freeswitch-users] FreeSwtich DB In-Reply-To: References: Message-ID: Can you tell us which DB has the recent activity? I'm almost certain that the sqlite isn't acutally being used, or you don't have odbc configured properly. On Feb 6, 2012 3:06 PM, "Joe Flemmings" wrote: > In my email, you can see that i said FreeSwitch creates the database > files. It creates both the ODBC(MySQL) and DB files. > > Questions is WHY? > > On Fri, Feb 3, 2012 at 10:36 AM, Brad Mina wrote: > >> Check your odbc database and make sure FS has created (and uses) that >> database. If it's creating db files, it's probably not using your odbc db. >> >> On Thu, Feb 2, 2012 at 12:07 AM, Joe Flemmings wrote: >> >>> Is their a reason why freeswitch still creates DB files in >>> /usr/local/freeswitch/db/ even when using ODBC at the core as decsribed >>> here http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> 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 >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/8645b496/attachment-0001.html From asrivas at gmail.com Tue Feb 7 00:17:35 2012 From: asrivas at gmail.com (Anurag Srivastava) Date: Mon, 6 Feb 2012 13:17:35 -0800 Subject: [Freeswitch-users] external_sip_ip not refreshed after ip address change. In-Reply-To: References: Message-ID: Hi Brian, Thanks for your help. From what I see the parameter auto-restart=true will only restart profiles if the local interface ip address on the machine changes. Will it also restart if my public ip address changes? My machine is behind a NAT with port forwarding. I am trying to restart profiles if my public ip address changes. Thanks Can somebody help me with this? > > > I have a question about nat behavior in freeswitch. Basically my external >> calls to freeswitch are getting disconnected after 30 seconds of two way >> audio when my external ip address changes. I have dhcp from my ISP and am >> using external_sip_ip and external_rtp_ip as stun:. When my IP >> changes I see that external_sip_ip does not get refreshed but >> external_rtp_ip does. I am not allowed to enable upnp/nat-pmp on my router. >> Apparently it is a known issue that external_sip_ip is read just >> at load time and not refreshed even if it is specified in stun format >> Is there a fix to this problem? >> >> There is always the option of restarting profile when ddclient notes an >> ip change. Is there something inbuilt into FS. It is already finding that >> ip address has changed as reflected in external_rtp_ip which does use stun >> and gets the right ip address. >> > > > > -- > Regards > Anurag > -- Regards Anurag -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/74fe562c/attachment.html From wayne at hamilton.net Tue Feb 7 00:22:52 2012 From: wayne at hamilton.net (Wayne) Date: Mon, 6 Feb 2012 15:22:52 -0600 Subject: [Freeswitch-users] Freeswitch sharing same local host with sip server Message-ID: <6159C33D94034776ABAEC713D6A64098@ccs.local> Hello all, I am using Freeswitch for a Voicemail server as a part of a larger project. It is sharing the same local host with a different SIP server. The other server is setting on port 5070. Freeswitch is sending calls from 5080 to 5070 the call gets connected and works fine. Currently I am testing with sip phones registered on both systems. When I hang up on the phone registered with freeswitch the call does not get torn down on the other sip server. A whireshark trace shows the hang up going to port 5060 not 5070. So Freeswitch is sending it's self a disconnect message on port 5060 and replies with a 481 call does not exist. What am I missing? Wayne From joe.jflemmings at gmail.com Tue Feb 7 00:28:59 2012 From: joe.jflemmings at gmail.com (Joe Flemmings) Date: Mon, 6 Feb 2012 13:28:59 -0800 Subject: [Freeswitch-users] FreeSwtich DB In-Reply-To: References: Message-ID: It seams to be both, Like when i make call, i see them on ODBC sip_dialogs table. I see sip registrations on both DB and ODBC tables. When i delete /usr/local/freeswitch/db/core.db and restart FreeSwitch, the database is recreated On Thu, Feb 2, 2012 at 12:07 AM, Joe Flemmings wrote: > Is their a reason why freeswitch still creates DB files in > /usr/local/freeswitch/db/ even when using ODBC at the core as decsribed > here http://wiki.freeswitch.org/wiki/Using_ODBC_in_the_core -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/8eb1d5f4/attachment.html From msc at freeswitch.org Tue Feb 7 00:36:01 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 6 Feb 2012 13:36:01 -0800 Subject: [Freeswitch-users] xml_curl dialplan and static files In-Reply-To: References: Message-ID: Kyle, Welcome to FreeSWITCH! We'll be happy to help answer your question. In return we would just like to know what you guys are up to over there at CSU. :) Are you using FreeSWITCH for anything in particular? As far as your question goes, you'll most likely need to employ the "dynamic config w/ static fallback" setup. The way it works is that if your xml curl program returns a special "not found" config then FreeSWITCH will go looking in the static XML for the relevant information. The trick for you will be to return the "not found" message any time that you want the static XML to effect. The "special" config is this:
So, you'll need your xml curl script to recognize which dialplan contexts to handle and which to send back w/ the "not found" message. Let us know how it goes. -MC On Mon, Feb 6, 2012 at 11:58 AM, Kyle Haefner wrote: > Hi All, > > I have not been able to glean enough from the wiki and Google searches > to figure this out. > > I have freeswitch xml_curl fetching a dial plan. What I would *like* > to do is have one context fetched > dynamically while another context is from a static file (or fail-back > to a static file). > > I would appreciate any help. > > Thanks! > > > > > -- > Kyle Haefner, M.S. > Communication Systems Programmer > Colorado State University > Fort Collins, CO > Phone: 970-491-1012 > Email: kyle.haefner at colostate.edu > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/5b85d7b2/attachment.html From msc at freeswitch.org Tue Feb 7 01:00:52 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 6 Feb 2012 14:00:52 -0800 Subject: [Freeswitch-users] voicemail_say_phone_number In-Reply-To: References: Message-ID: Yuriy, I looked on mod_voicemail.c and it does not appear that this macro is used anywhere. However, I stumbled across what I believe is the answer to your question: the vm_announce_cid channel variable. And guess what? It's *not* on the wiki, so you are totally forgiven for not reading the documentation. :) Please set vm_announce_cid to true prior to checking voicemail and see if it works. If it does, please let me know. If you can add it to the wiki then do so, otherwise one of our intrepid community members will do it. -MC 2012/2/6 Yuriy Nasida > Hello list, > > I would like to have one simple ability to listen the phone number of > caller when I check my voicemail. At present I listen date of message only. > I have found the macro "voicemail_say_phone_number" in > conf/lang/en/vm/sounds.xml. But... when FS uses it ? Can I get it without > modifying of source code of voicemail module ? > > Please advise. > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/298b80cc/attachment.html From Kyle.Haefner at colostate.edu Tue Feb 7 01:01:45 2012 From: Kyle.Haefner at colostate.edu (Kyle Haefner) Date: Mon, 6 Feb 2012 15:01:45 -0700 Subject: [Freeswitch-users] xml_curl dialplan and static files In-Reply-To: <6a1d6a530b9649a794f1e9ad038ba2ee@EXCH1.ColoState.EDU> References: <6a1d6a530b9649a794f1e9ad038ba2ee@EXCH1.ColoState.EDU> Message-ID: Hi Michael, Awesome.. I was actually very close, I was trying the "not found" route but had the section name as "dialplan", changing it to "result" made it work. thanks! At CSU we use Freeswitch as part of sipXecs and BigBlueButton production deployments. Really, though, Freeswitch is our go-to for anything that needs to be customized or scripted. The most recent example of this is using the xml_curl function to build a pre-set conference for emergency situations. Freeswitch dials out to a group of high-level administrators and once they answer and prove they are human by pressing a number key they are bridged into a conference. This a good way to get groups of people like our Public Safety Team together. Thanks! Kyle On Mon, Feb 6, 2012 at 2:36 PM, Michael Collins wrote: > Kyle, > > Welcome to FreeSWITCH! We'll be happy to help answer your question. In return we would just like to know what you guys are up to over there at CSU. :) Are you using FreeSWITCH for anything in particular? > > As far as your question goes, you'll most likely need to employ the "dynamic config w/ static fallback" setup. The way it works is that if your xml curl program returns a special "not found" config then FreeSWITCH will go looking in the static XML for the relevant information. The trick for you will be to return the "not found" message any time that you want the static XML to effect. The "special" config is this: > > > > ?
> ? ? > ?
> > > So, you'll need your xml curl script to recognize which dialplan contexts to handle and which to send back w/ the "not found" message. > > Let us know how it goes. > -MC > > On Mon, Feb 6, 2012 at 11:58 AM, Kyle Haefner > wrote: > Hi All, > > I have not been able to glean enough from the wiki and Google searches > to figure this out. > > I have freeswitch xml_curl fetching a dial plan. What I would *like* > to do is have one context fetched > dynamically while another context is from a static file (or fail-back > to a static file). > > I would appreciate any help. > > Thanks! > > > > > -- > Kyle Haefner, M.S. > Communication Systems Programmer > Colorado State University > Fort Collins, CO > Phone: 970-491-1012 > Email: ?kyle.haefner at colostate.edu > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kyle Haefner, M.S. Communication Systems Programmer Colorado State University Fort Collins, CO Phone: 970-491-1012 Email: ?kyle.haefner at colostate.edu From msc at freeswitch.org Tue Feb 7 01:05:19 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 6 Feb 2012 14:05:19 -0800 Subject: [Freeswitch-users] external_sip_ip not refreshed after ip address change. In-Reply-To: References: Message-ID: On Mon, Feb 6, 2012 at 1:17 PM, Anurag Srivastava wrote: > Hi Brian, > Thanks for your help. From what I see the parameter auto-restart=true > will only restart profiles if the local interface ip address on the machine > changes. Will it also restart if my public ip address changes? My machine > is behind a NAT with port forwarding. I am trying to restart profiles if my > public ip address changes. > Thanks > > Are you starting fs with the -nonat flag? -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/5e9543d9/attachment.html From freeswitch at earthspike.net Tue Feb 7 02:22:58 2012 From: freeswitch at earthspike.net (John) Date: Mon, 06 Feb 2012 23:22:58 +0000 Subject: [Freeswitch-users] voicemail_say_phone_number In-Reply-To: References: Message-ID: <4F3060D2.7050505@earthspike.net> Michael, It does work, but it's a bit 'rough': all it does it speak the number just before the date. So I can understand why it was not documented... I'm just opining here, but I think it needs a 'Call from {caller-id} at {date/time} {message}' rather than just '{caller-id}{date/time} {message}'. It also ought to trap for '00000000' and say 'unknown caller' or such like; not sure if it does that or not. But for me it's not important enough to spend the time working out the patch to do all that. As I said, I'm just opining... :) John PS. For those who want to know where to insert this, put the line marked with + into your conf/dialplan/default.xml file: + On 06/02/12 22:00, Michael Collins wrote: > Yuriy, > > I looked on mod_voicemail.c and it does not appear that this macro is > used anywhere. However, I stumbled across what I believe is the answer > to your question: the vm_announce_cid channel variable. And guess > what? It's *not* on the wiki, so you are totally forgiven for not > reading the documentation. :) > > Please set vm_announce_cid to true prior to checking voicemail and see > if it works. If it does, please let me know. If you can add it to the > wiki then do so, otherwise one of our intrepid community members will > do it. > > -MC > > 2012/2/6 Yuriy Nasida > > > Hello list, > > I would like to have one simple ability to listen the phone number > of caller when I check my voicemail. At present I listen date of > message only. I have found the macro "voicemail_say_phone_number" > in conf/lang/en/vm/sounds.xml. But... when FS uses it ? Can I get > it without modifying of source code of voicemail module ? > > Please advise. > Thanks. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/1ce73e91/attachment.html From msc at freeswitch.org Tue Feb 7 04:03:19 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 6 Feb 2012 17:03:19 -0800 Subject: [Freeswitch-users] voicemail_say_phone_number In-Reply-To: <4F3060D2.7050505@earthspike.net> References: <4F3060D2.7050505@earthspike.net> Message-ID: John, This actually makes sense, so I'm adding a new prompt to my 'to be ordered' list. We can make a new phrase macro that handles all the 000/anon handling. Anyway, when the new prompts arrive in a week or two I will test out a new phrase macro and then have you try it for your installation. BTW, if you are feeling ambitious you can modify your say number phrase macro to include a little pause and an existing sound file. I was thinking you could use ivr-this_is_a_call_from.wav. Try this macro in place of the existing one: Keep in mind that I just made that up and didn't test it, so please do some diagnostics if it doesn't work the first time. Thanks, MC On Mon, Feb 6, 2012 at 3:22 PM, John wrote: > Michael, > > It does work, but it's a bit 'rough': all it does it speak the number just > before the date. So I can understand why it was not documented... > > I'm just opining here, but I think it needs a 'Call from {caller-id} at > {date/time} {message}' rather than just '{caller-id}{date/time} > {message}'. It also ought to trap for '00000000' and say 'unknown caller' > or such like; not sure if it does that or not. But for me it's not > important enough to spend the time working out the patch to do all that. > As I said, I'm just opining... :) > > John > > > PS. For those who want to know where to insert this, put the line marked > with + into your conf/dialplan/default.xml file: > > > > expression="^vmain$|^4000$|^\*98$"> > > > + > > > > > > On 06/02/12 22:00, Michael Collins wrote: > > Yuriy, > > I looked on mod_voicemail.c and it does not appear that this macro is used > anywhere. However, I stumbled across what I believe is the answer to your > question: the vm_announce_cid channel variable. And guess what? It's *not* > on the wiki, so you are totally forgiven for not reading the documentation. > :) > > Please set vm_announce_cid to true prior to checking voicemail and see if > it works. If it does, please let me know. If you can add it to the wiki > then do so, otherwise one of our intrepid community members will do it. > > -MC > > 2012/2/6 Yuriy Nasida > >> Hello list, >> >> I would like to have one simple ability to listen the phone number of >> caller when I check my voicemail. At present I listen date of message only. >> I have found the macro "voicemail_say_phone_number" in >> conf/lang/en/vm/sounds.xml. But... when FS uses it ? Can I get it without >> modifying of source code of voicemail module ? >> >> Please advise. >> 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/40e722f4/attachment.html From msc at freeswitch.org Tue Feb 7 04:17:13 2012 From: msc at freeswitch.org (Michael Collins) Date: Mon, 6 Feb 2012 17:17:13 -0800 Subject: [Freeswitch-users] Freeswitch sharing same local host with sip server In-Reply-To: <6159C33D94034776ABAEC713D6A64098@ccs.local> References: <6159C33D94034776ABAEC713D6A64098@ccs.local> Message-ID: You're missing a SIP trace and debug logs from both machines. :P Collect debug log and SIP trace on either FS box and put into pastebin.freeswitch.org. Don't forget to use "FreeSWITCH Log" as the syntax highlighting type. -MC On Mon, Feb 6, 2012 at 1:22 PM, Wayne wrote: > Hello all, > > I am using Freeswitch for a Voicemail server as a part of a larger project. > It is sharing the same local host with a different SIP server. The other > server is setting on port 5070. Freeswitch is sending calls from 5080 to > 5070 the call gets connected and works fine. Currently I am testing with > sip > phones registered on both systems. When I hang up on the phone registered > with freeswitch the call does not get torn down on the other sip server. A > whireshark trace shows the hang up going to port 5060 not 5070. So > Freeswitch is sending it's self a disconnect message on port 5060 and > replies with a 481 call does not exist. > > > What am I missing? > Wayne > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/188280ab/attachment-0001.html From darcy at primrose.ws Tue Feb 7 06:01:56 2012 From: darcy at primrose.ws (Darcy) Date: Mon, 6 Feb 2012 22:01:56 -0500 Subject: [Freeswitch-users] voicemail_say_phone_number In-Reply-To: References: <4F3060D2.7050505@earthspike.net> Message-ID: <6574CFF3DDE54C118E0E9250AC19EDD7@DWP> The message, this_is_a_call_from actually has to be set in the dialplan it appears, the fs plays the file set in vm_announce_cid, a simple dial plan below reflects one way of doing this, tested and it works. Needs more time to suit the total requirements, but this makes it a little more professional by adding the message in front of the number. Darcy From: Michael Collins Sent: Monday, February 06, 2012 8:03 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] voicemail_say_phone_number John, This actually makes sense, so I'm adding a new prompt to my 'to be ordered' list. We can make a new phrase macro that handles all the 000/anon handling. Anyway, when the new prompts arrive in a week or two I will test out a new phrase macro and then have you try it for your installation. BTW, if you are feeling ambitious you can modify your say number phrase macro to include a little pause and an existing sound file. I was thinking you could use ivr-this_is_a_call_from.wav. Try this macro in place of the existing one: Keep in mind that I just made that up and didn't test it, so please do some diagnostics if it doesn't work the first time. Thanks, MC On Mon, Feb 6, 2012 at 3:22 PM, John wrote: Michael, It does work, but it's a bit 'rough': all it does it speak the number just before the date. So I can understand why it was not documented... I'm just opining here, but I think it needs a 'Call from {caller-id} at {date/time} {message}' rather than just '{caller-id}{date/time} {message}'. It also ought to trap for '00000000' and say 'unknown caller' or such like; not sure if it does that or not. But for me it's not important enough to spend the time working out the patch to do all that. As I said, I'm just opining... :) John PS. For those who want to know where to insert this, put the line marked with + into your conf/dialplan/default.xml file: + On 06/02/12 22:00, Michael Collins wrote: Yuriy, I looked on mod_voicemail.c and it does not appear that this macro is used anywhere. However, I stumbled across what I believe is the answer to your question: the vm_announce_cid channel variable. And guess what? It's *not* on the wiki, so you are totally forgiven for not reading the documentation. :) Please set vm_announce_cid to true prior to checking voicemail and see if it works. If it does, please let me know. If you can add it to the wiki then do so, otherwise one of our intrepid community members will do it. -MC 2012/2/6 Yuriy Nasida Hello list, I would like to have one simple ability to listen the phone number of caller when I check my voicemail. At present I listen date of message only. I have found the macro "voicemail_say_phone_number" in conf/lang/en/vm/sounds.xml. But... when FS uses it ? Can I get it without modifying of source code of voicemail module ? Please advise. 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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/c12d7e71/attachment.html From admin at blindi.net Tue Feb 7 07:08:32 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Tue, 7 Feb 2012 05:08:32 +0100 (CET) Subject: [Freeswitch-users] Astconf2fsconf a software or joke? In-Reply-To: <1336366836.3502.7.camel@localhost.localdomain> References: <1336366836.3502.7.camel@localhost.localdomain> Message-ID: Hi guys, I searched the web with a sofware "Astconf2fsconf". I found a description on: http://wiki.freeswitch.org/wiki/Astconf2fsconf Why this site exists, over all? No download link, no installationhowto. What sense does it report about a software which does not appear there? Very funny-). 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 bote_radio at botecomm.com Tue Feb 7 07:14:19 2012 From: bote_radio at botecomm.com (Bote Man) Date: Mon, 6 Feb 2012 23:14:19 -0500 Subject: [Freeswitch-users] voicemail_say_phone_number In-Reply-To: <6574CFF3DDE54C118E0E9250AC19EDD7@DWP> References: <4F3060D2.7050505@earthspike.net> <6574CFF3DDE54C118E0E9250AC19EDD7@DWP> Message-ID: <01b601cce54e$f865c250$e93146f0$@com> Well, now I'm cornfused. The original example by John used vm_announce_cid as a Boolean switch. But you've tested it as a string that points to the desired sound file? I have no means to test this nor access to sources right now and I just added John's example to the wiki. I should back it out or correct it based on your results. Please advise. John Boteler Bote Communications in rainy Fort Lauderdale, FL From: Darcy Sent: Monday, 06 February, 2012 22:02 The message, this_is_a_call_from actually has to be set in the dialplan it appears, the fs plays the file set in vm_announce_cid, a simple dial plan below reflects one way of doing this, tested and it works. Needs more time to suit the total requirements, but this makes it a little more professional by adding the message in front of the number. Darcy ? On Mon, Feb 6, 2012 at 3:22 PM, John wrote: Michael, It does work, but it's a bit 'rough': all it does it speak the number just before the date. So I can understand why it was not documented... ? John PS. For those who want to know where to insert this, put the line marked with + into your conf/dialplan/default.xml file: + On 06/02/12 22:00, Michael Collins wrote: Yuriy, ?Please set vm_announce_cid to true prior to checking voicemail and see if it works. If it does, please let me know. If you can add it to the wiki then do so, otherwise one of our intrepid community members will do it. -MC 2012/2/6 Yuriy Nasida Hello list, I would like to have one simple ability to listen the phone number of caller when I check my voicemail. At present I listen date of message only. I have found the macro "voicemail_say_phone_number" in conf/lang/en/vm/sounds.xml. But... when FS uses it ? Can I get it without modifying of source code of voicemail module ? Please advise. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/3784983a/attachment-0001.html From darcy at primrose.ws Tue Feb 7 07:27:23 2012 From: darcy at primrose.ws (Darcy) Date: Mon, 6 Feb 2012 23:27:23 -0500 Subject: [Freeswitch-users] voicemail_say_phone_number In-Reply-To: <01b601cce54e$f865c250$e93146f0$@com> References: <4F3060D2.7050505@earthspike.net> <6574CFF3DDE54C118E0E9250AC19EDD7@DWP> <01b601cce54e$f865c250$e93146f0$@com> Message-ID: <26ADC64FC37D4049A68787B3A0DCF4FA@DWP> If John?s works, use it, I could not make it work but the example I show below played the greeting before the clid, I have not added anything to play an annonymous greeting yet as I have not had time. in mod_voicemail.c you have the following code: if (!zstr(cbt->cid_number) && (vm_announce_cid = switch_channel_get_variable(channel, "vm_announce_cid"))) { switch_ivr_play_file(session, NULL, vm_announce_cid, NULL); switch_ivr_sleep(session, 500, SWITCH_TRUE, NULL); switch_ivr_say(session, cbt->cid_number, NULL, "name_spelled", "pronounced", NULL, NULL); } Which indicates you will play the file in variable ?vm_announce_cid?. Darcy In Sunny Ottawa Canada From: Bote Man Sent: Monday, February 06, 2012 11:14 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] voicemail_say_phone_number Well, now I'm cornfused. The original example by John used vm_announce_cid as a Boolean switch. But you've tested it as a string that points to the desired sound file? I have no means to test this nor access to sources right now and I just added John's example to the wiki. I should back it out or correct it based on your results. Please advise. John Boteler Bote Communications in rainy Fort Lauderdale, FL From: Darcy Sent: Monday, 06 February, 2012 22:02 The message, this_is_a_call_from actually has to be set in the dialplan it appears, the fs plays the file set in vm_announce_cid, a simple dial plan below reflects one way of doing this, tested and it works. Needs more time to suit the total requirements, but this makes it a little more professional by adding the message in front of the number. Darcy ? On Mon, Feb 6, 2012 at 3:22 PM, John wrote: Michael, It does work, but it's a bit 'rough': all it does it speak the number just before the date. So I can understand why it was not documented... ? John PS. For those who want to know where to insert this, put the line marked with + into your conf/dialplan/default.xml file: + On 06/02/12 22:00, Michael Collins wrote: Yuriy, ?Please set vm_announce_cid to true prior to checking voicemail and see if it works. If it does, please let me know. If you can add it to the wiki then do so, otherwise one of our intrepid community members will do it. -MC 2012/2/6 Yuriy Nasida Hello list, I would like to have one simple ability to listen the phone number of caller when I check my voicemail. At present I listen date of message only. I have found the macro "voicemail_say_phone_number" in conf/lang/en/vm/sounds.xml. But... when FS uses it ? Can I get it without modifying of source code of voicemail module ? Please advise. 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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/9c9abe88/attachment.html From manavid at gmail.com Tue Feb 7 07:42:04 2012 From: manavid at gmail.com (Mohammad Amin Navid) Date: Mon, 6 Feb 2012 20:42:04 -0800 Subject: [Freeswitch-users] I'm Alive! In-Reply-To: References: Message-ID: <85C28B5F-B088-4BFA-ABF9-2BC071174658@gmail.com> Best news! Welcome back... On Feb 6, 2012, at 9:25 AM, Brian West wrote: > I would like to thank everyone that has donated money to help me pay for all the expense of having a stay in the hospital. It was a pretty scary week for me last week. If it were not for the morphine last week would have sucked even more! I have missed out on the annual Developer meeting in Milwaukee this year due to this event. So please make sure you guys donate some money for the dinner for those guys. > > Again Thank you all for your support. > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120206/2d7a644f/attachment-0001.html From curriegrad2004 at gmail.com Tue Feb 7 09:11:12 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Mon, 6 Feb 2012 22:11:12 -0800 Subject: [Freeswitch-users] I'm Alive! In-Reply-To: <85C28B5F-B088-4BFA-ABF9-2BC071174658@gmail.com> References: <85C28B5F-B088-4BFA-ABF9-2BC071174658@gmail.com> Message-ID: I knew you'd survive this one Brian. But great to hear that you're getting better On Mon, Feb 6, 2012 at 8:42 PM, Mohammad Amin Navid wrote: > Best news! > > Welcome back... > > On Feb 6, 2012, at 9:25 AM, Brian West wrote: > > I would like to thank everyone that has donated money to help me pay for all > the expense of having a stay in the hospital. ?It was a pretty scary week > for me last week. ?If it were not for the morphine last week would have > sucked even more! ?I have missed out on the annual Developer meeting in > Milwaukee this year due to this event. ?So please make sure you guys donate > some money for the dinner for those guys. > > Again Thank you all for your support. > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: ? +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From engineerzuhairraza at gmail.com Tue Feb 7 09:51:34 2012 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Tue, 7 Feb 2012 10:51:34 +0400 Subject: [Freeswitch-users] I'm Alive! In-Reply-To: References: Message-ID: Glad to hear that Best of luck for the rest Brian Regards, Zohair Raza On Mon, Feb 6, 2012 at 9:25 PM, Brian West wrote: > I would like to thank everyone that has donated money to help me pay for > all the expense of having a stay in the hospital. It was a pretty scary > week for me last week. If it were not for the morphine last week would > have sucked even more! I have missed out on the annual Developer meeting > in Milwaukee this year due to this event. So please make sure you guys > donate some money for the dinner for those guys. > > Again Thank you all for your support. > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/9d3bac2c/attachment.html From engineerzuhairraza at gmail.com Tue Feb 7 10:02:23 2012 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Tue, 7 Feb 2012 11:02:23 +0400 Subject: [Freeswitch-users] Astconf2fsconf a software or joke? In-Reply-To: References: <1336366836.3502.7.camel@localhost.localdomain> Message-ID: Hi Thomas, :) Good one! but it has a link that points to http://wiki.freeswitch.org/wiki/Rosetta_Stone which has a lot of information on equivalents of Asterisk in Freeswitch. Regards, Zohair Raza On Tue, Feb 7, 2012 at 8:08 AM, Thomas Hoellriegel wrote: > Hi guys, > I searched the web with a sofware "Astconf2fsconf". > I found a description on: > http://wiki.freeswitch.org/**wiki/Astconf2fsconf > Why this site exists, over all? > No download link, no installationhowto. > What sense does it report about a software which does not appear there? > Very funny-). > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/18074001/attachment.html From shaheryarkh at googlemail.com Tue Feb 7 11:55:29 2012 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Tue, 7 Feb 2012 13:55:29 +0500 Subject: [Freeswitch-users] I'm Alive! In-Reply-To: References: Message-ID: Wow! you are alive, that's great. Thank you. On Tue, Feb 7, 2012 at 11:51 AM, Zohair Raza wrote: > Glad to hear that > > Best of luck for the rest Brian > > Regards, > Zohair Raza > > On Mon, Feb 6, 2012 at 9:25 PM, Brian West wrote: > >> I would like to thank everyone that has donated money to help me pay for >> all the expense of having a stay in the hospital. It was a pretty scary >> week for me last week. If it were not for the morphine last week would >> have sucked even more! I have missed out on the annual Developer meeting >> in Milwaukee this year due to this event. So please make sure you guys >> donate some money for the dinner for those guys. >> >> Again Thank you all for your support. >> >> -- >> Brian West >> FreeSWITCH Solutions, LLC >> Phone: +1 (918) 420-9266 >> Fax: +1 (918) 420-9267 >> brian at freeswitch.org >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > 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/20120207/fbdb6509/attachment.html From nasida at live.ru Tue Feb 7 13:22:38 2012 From: nasida at live.ru (Yuriy Nasida) Date: Tue, 7 Feb 2012 14:22:38 +0400 Subject: [Freeswitch-users] voicemail_say_phone_number In-Reply-To: <26ADC64FC37D4049A68787B3A0DCF4FA@DWP> References: <4F3060D2.7050505@earthspike.net> <6574CFF3DDE54C118E0E9250AC19EDD7@DWP>, <01b601cce54e$f865c250$e93146f0$@com>, <26ADC64FC37D4049A68787B3A0DCF4FA@DWP> Message-ID: Thank you guys! It works. The example with 'vm_announce_cid=ivr/ivr-this_is_a_call_from.wav' is more correct. Otherwise FS tries to play message true.wav :)So, voicemail module doesn't use macro "voicemail_say_phone_number" and as far as I see the source code of mod_voicemail.c explains this behaviour. Yes, ability for playing of some wav file if I will have vm from annonymous would be very useful. Probably I can add it independently but I believe that your modifying will more correctly :)Darcy please let me know if you plan to add this feature in the near future. Anyway thanks again! From: darcy at primrose.ws To: freeswitch-users at lists.freeswitch.org Date: Mon, 6 Feb 2012 23:27:23 -0500 Subject: Re: [Freeswitch-users] voicemail_say_phone_number If John?s works, use it, I could not make it work but the example I show below played the greeting before the clid, I have not added anything to play an annonymous greeting yet as I have not had time. in mod_voicemail.c you have the following code: if (!zstr(cbt->cid_number) && (vm_announce_cid = switch_channel_get_variable(channel, "vm_announce_cid"))) { switch_ivr_play_file(session, NULL, vm_announce_cid, NULL); switch_ivr_sleep(session, 500, SWITCH_TRUE, NULL); switch_ivr_say(session, cbt->cid_number, NULL, "name_spelled", "pronounced", NULL, NULL); } Which indicates you will play the file in variable ?vm_announce_cid?. Darcy In Sunny Ottawa Canada From: Bote Man Sent: Monday, February 06, 2012 11:14 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] voicemail_say_phone_number Well, now I'm cornfused. The original example by John used vm_announce_cid as a Boolean switch. But you've tested it as a string that points to the desired sound file? I have no means to test this nor access to sources right now and I just added John's example to the wiki. I should back it out or correct it based on your results. Please advise. John Boteler Bote Communications in rainy Fort Lauderdale, FL From: Darcy Sent: Monday, 06 February, 2012 22:02 The message, this_is_a_call_from actually has to be set in the dialplan it appears, the fs plays the file set in vm_announce_cid, a simple dial plan below reflects one way of doing this, tested and it works. Needs more time to suit the total requirements, but this makes it a little more professional by adding the message in front of the number. Darcy ? On Mon, Feb 6, 2012 at 3:22 PM, John wrote: Michael, It does work, but it's a bit 'rough': all it does it speak the number just before the date. So I can understand why it was not documented... ? John PS. For those who want to know where to insert this, put the line marked with + into your conf/dialplan/default.xml file: + On 06/02/12 22:00, Michael Collins wrote: Yuriy, ?Please set vm_announce_cid to true prior to checking voicemail and see if it works. If it does, please let me know. If you can add it to the wiki then do so, otherwise one of our intrepid community members will do it. -MC 2012/2/6 Yuriy Nasida Hello list, I would like to have one simple ability to listen the phone number of caller when I check my voicemail. At present I listen date of message only. I have found the macro "voicemail_say_phone_number" in conf/lang/en/vm/sounds.xml. But... when FS uses it ? Can I get it without modifying of source code of voicemail module ? Please advise. Thanks. _________________________________________________________________________ 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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/8c28c601/attachment-0001.html From gb10hkzo-freeswitch at yahoo.co.uk Tue Feb 7 15:12:24 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Tue, 7 Feb 2012 12:12:24 +0000 (GMT) Subject: [Freeswitch-users] Getting USER_NOT_REGISTERED for no reason ? Message-ID: <1328616744.81316.YahooMailNeo@web29403.mail.ird.yahoo.com> Hi Brian, Have tried your various suggestions without luck. Have also tried replacing ? with? ? in the dial plan and that doesn't work either Everything resolves fine though...... freeswitch at internal> expand echo ${sofia_contact(openvpn_udp/1000)} sofia/openvpn_udp/sip:1000 at 10.82.1.6:3072;line=2sf2mz02 freeswitch at internal> expand echo ${sofia_contact(internal/1000)} sofia/internal/sip:1000 at 10.14.2.2:12592;rinstance=5e131f94a204160f And yes I have looked through the logs, but there's nothing as blatantly obvious in there as you seem to think there is ? Bob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/d82e1ac5/attachment.html From gb10hkzo-freeswitch at yahoo.co.uk Tue Feb 7 15:15:53 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Tue, 7 Feb 2012 12:15:53 +0000 (GMT) Subject: [Freeswitch-users] Mod_dingaling and openfire issue Message-ID: <1328616953.19943.YahooMailNeo@web29404.mail.ird.yahoo.com> Hi, What are you looking to do with Openfire and FreeSwitch once you have it up and running ? Speaking as someone who's tried to integrate FreeSwitch and Openfire in relation to conferencing, I've found it to be a bit of a waste of time, as mod_dingaling isn't ready for prime time as far as XMPP integration goes. Others may disagree with me here, but I've given up and abandoned the idea until the codebase is improved sometime in the distant future. However good luck with your project and keep us updated Bob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/dc2e3162/attachment.html From jaasmailing at gmail.com Tue Feb 7 15:19:52 2012 From: jaasmailing at gmail.com (Carlo Dimaggio) Date: Tue, 07 Feb 2012 13:19:52 +0100 Subject: [Freeswitch-users] timeout param in mod_xml_cdr Message-ID: <4F3116E8.7000200@gmail.com> Hi all, I have discovered a param (timeout) implemented in mod_xml_cdr but not documented. The param sets a timeout of the posting action to the web server (in the example wait 5 seconds before set the web post as "failed"): It is interested when you want to be sure that there are no unhandled open sessions (in my case I want to be sure that all cdr are submitted to the billing engine). Regards, -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/a638c6f5/attachment.html From shouldbeq931 at gmail.com Tue Feb 7 15:52:46 2012 From: shouldbeq931 at gmail.com (shouldbe q931) Date: Tue, 7 Feb 2012 12:52:46 +0000 Subject: [Freeswitch-users] Astconf2fsconf a software or joke? In-Reply-To: References: <1336366836.3502.7.camel@localhost.localdomain> Message-ID: On Tue, Feb 7, 2012 at 4:08 AM, Thomas Hoellriegel wrote: > Hi guys, > I searched the web with a sofware "Astconf2fsconf". > I found a description on: > http://wiki.freeswitch.org/wiki/Astconf2fsconf > Why this site exists, over all? > No download link, no installationhowto. > What sense does it report about a software which does not appear there? > Very funny-). > thanks. > > That's unfortunately because you found the notes page for what astconf2fsconf should do, this might make more sense, http://wiki.freeswitch.org/wiki/Bounty_challenged From govoiper at gmail.com Tue Feb 7 16:21:59 2012 From: govoiper at gmail.com (Sammy Govind) Date: Tue, 7 Feb 2012 18:21:59 +0500 Subject: [Freeswitch-users] One way Audio when dialing from one end. In-Reply-To: References: Message-ID: Hey hey hey, Thats really great !! Woww.!! wwohhh...lucky you. You've discovered the restricted call mode. Do tell me which options you configured your ASTERISK with, that'd be really nice of you. LOL,Ok fine, the thing is I'm sure you misspelled FreeSWITCH here. The most famous and obvious of all one-way audio issue has struck again here. Please see your Network configurations (specially related to NAT). Also paste the relevant SIP traces for the one-way audio call here. Regards, Sammy On Mon, Feb 6, 2012 at 3:00 PM, Tariq Mahmood wrote: > Hi, > > I hope you guys are doing great. > > I am having an issue with my number, When i dial the number with my *asterisk > *server, the callee can hear me but i cant hear him. But if the same guy > call me, the conversation goes great. > > Any Help??? > > -- > Best Regards > Tareq Khan > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/54c9cf8d/attachment.html From dgarcia at anew.com.ve Tue Feb 7 16:38:47 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Tue, 07 Feb 2012 09:08:47 -0430 Subject: [Freeswitch-users] Mod_dingaling and openfire issue In-Reply-To: <1328616953.19943.YahooMailNeo@web29404.mail.ird.yahoo.com> References: <1328616953.19943.YahooMailNeo@web29404.mail.ird.yahoo.com> Message-ID: <4F312967.5050807@anew.com.ve> Thanks for your comment, well, I trying to use openfire as screen pop-up. My project was to build an IVR script in lua, collect some data, consume some webservice, then populate a variable. Then transfer the call to a queue, then when the call is delivered to an agent, use chat command to send a pop-up to the agent wih the collected data. So far, Openfire seems to be working fine, I have two clients registered in Openfire: one using spark, the second using SJPhone (also, it is registered in FS with an extension X). I can send message between both clients, thats why I think openfire is working fine. When I activate server-server integration in FS to connect to Openfire , as soon I use chat app or FS receive Openfire messaging FS stop/exit, I mean, FS is not longer running. Why? FS should not end if mod fail or other end send wrong or unexpected messages When mod_dingaling was developed which jabber server used? On 2/7/2012 7:45 AM, Bob Smith wrote: > Hi, > > What are you looking to do with Openfire and FreeSwitch once you have > it up and running ? > > Speaking as someone who's tried to integrate FreeSwitch and Openfire > in relation to conferencing, I've found it to be a bit of a waste of > time, as mod_dingaling isn't ready for prime time as far as XMPP > integration goes. > > Others may disagree with me here, but I've given up and abandoned the > idea until the codebase is improved sometime in the distant future. > > However good luck with your project and keep us updated > > Bob > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > 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: 2012.0.1913 / Virus Database: 2112/4794 - Release Date: 02/07/12 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/428c6fa4/attachment-0001.html From b2m at a-cti.com Tue Feb 7 16:47:14 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Tue, 7 Feb 2012 19:17:14 +0530 Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: <4F2C2EFA.9030105@earthspike.net> References: <005801cce12e$05358300$0fa08900$@net.au> <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> <4F2A9482.1020100@earthspike.net> <4F2B8B51.2080506@earthspike.net> <4F2C2EFA.9030105@earthspike.net> Message-ID: All, I am having the same issue, its not sending email(extension --> lua script) getting *"Segmentation fault"* xml : dialplan : Lua : caller=503; freeswitch.consoleLog("info","From :"..caller); session:set_tts_parms("flite", "slt"); session:speak("Welcome To Voice Mail !. You Can Leave Your Message Here."); path="/usr/local/freeswitch/recordings/"; prompt=caller..".mp3"; recpath=path..prompt; freeswitch.consoleLog("info","record path="..recpath); session:recordFile(recpath,30,10,10); session:speak("Thank you."); freeswitch.consoleLog("info","testing"); freeswitch.email("b2m at a-cti.com", "saraswathi.devaraj at a-cti.com", "subject: Voicemail from 801\n", "Hello,\n\nYou've got a voicemail, click the attachment to listen to it.", "/usr/local/freeswitch/recordings/503.mp3", "", ""); freeswitch.consoleLog("info","hai"); switchconf : Thanks for your help!! Thanks, Bala On Sat, Feb 4, 2012 at 12:31 AM, John wrote: > On 03/02/12 18:00, Thomas Hoellriegel wrote: > > > > Its works fine!! > Good news! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/ed8c7f61/attachment.html From peter.olsson at visionutveckling.se Tue Feb 7 16:51:20 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Tue, 7 Feb 2012 13:51:20 +0000 Subject: [Freeswitch-users] Mod_dingaling and openfire issue In-Reply-To: <4F312967.5050807@anew.com.ve> References: <1328616953.19943.YahooMailNeo@web29404.mail.ird.yahoo.com> <4F312967.5050807@anew.com.ve> Message-ID: <1FFF97C269757C458224B7C895F35F15039F7C@cantor.std.visionutv.se> If FS crashes, it?s a bug, and should be reported to Jira. Please follow the instructions here to file a complete bug report. http://wiki.freeswitch.org/wiki/Reporting_Bugs /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Saugort Dario Garcia Tovar Skickat: den 7 februari 2012 14:39 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Mod_dingaling and openfire issue Thanks for your comment, well, I trying to use openfire as screen pop-up. My project was to build an IVR script in lua, collect some data, consume some webservice, then populate a variable. Then transfer the call to a queue, then when the call is delivered to an agent, use chat command to send a pop-up to the agent wih the collected data. So far, Openfire seems to be working fine, I have two clients registered in Openfire: one using spark, the second using SJPhone (also, it is registered in FS with an extension X). I can send message between both clients, thats why I think openfire is working fine. When I activate server-server integration in FS to connect to Openfire , as soon I use chat app or FS receive Openfire messaging FS stop/exit, I mean, FS is not longer running. Why? FS should not end if mod fail or other end send wrong or unexpected messages When mod_dingaling was developed which jabber server used? On 2/7/2012 7:45 AM, Bob Smith wrote: Hi, What are you looking to do with Openfire and FreeSwitch once you have it up and running ? Speaking as someone who's tried to integrate FreeSwitch and Openfire in relation to conferencing, I've found it to be a bit of a waste of time, as mod_dingaling isn't ready for prime time as far as XMPP integration goes. Others may disagree with me here, but I've given up and abandoned the idea until the codebase is improved sometime in the distant future. However good luck with your project and keep us updated Bob _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 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: 2012.0.1913 / Virus Database: 2112/4794 - Release Date: 02/07/12 -- Atentamente, Dario Garc?a Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve !DSPAM:4f3128b332769590918158! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/59dc0b78/attachment-0001.html From shaon.khan at gmail.com Tue Feb 7 17:30:43 2012 From: shaon.khan at gmail.com (Arafath-uz-zaman khan) Date: Tue, 7 Feb 2012 20:30:43 +0600 Subject: [Freeswitch-users] Setting ptime in inbound and outbound call Message-ID: Hello Is that possible the given diagram in freeswitch. G729 40 ptime -----> G729 -----> 20 ptime User Agent ------------------------------------ Fresswitch ------------------------------------ PSTN GW G729 60 ptime <----- G729 <----- 20 ptime If it is possible to do that can any one guide me how can i do that. With Regards -- Arafath-uz-zaman khan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/dba26ead/attachment.html From freeswitch at earthspike.net Tue Feb 7 17:39:35 2012 From: freeswitch at earthspike.net (John) Date: Tue, 07 Feb 2012 14:39:35 +0000 Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: References: <005801cce12e$05358300$0fa08900$@net.au> <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> <4F2A9482.1020100@earthspike.net> <4F2B8B51.2080506@earthspike.net> <4F2C2EFA.9030105@earthspike.net> Message-ID: <4F3137A7.2040908@earthspike.net> I updated the wiki mod_voicemail page a few days ago with some instructions for debugging email from freeswitch. Have you tried those? On 07/02/12 13:47, Balamurugan Mahendran wrote: > All, > > I am having the same issue, its not sending email(extension --> lua > script) getting *"Segmentation fault"* > > xml : > > > > > > > > > > > > > > > > > > > > > > > > > value="$${outbound_caller_name}"/> > value="$${outbound_caller_id}"/> > > > > > > dialplan : > > > > > > > > > > > Lua : > > caller=503; > freeswitch.consoleLog("info","From :"..caller); > session:set_tts_parms("flite", "slt"); > session:speak("Welcome To Voice Mail !. You Can Leave Your Message > Here."); > path="/usr/local/freeswitch/recordings/"; > prompt=caller..".mp3"; > recpath=path..prompt; > freeswitch.consoleLog("info","record path="..recpath); > session:recordFile(recpath,30,10,10); > session:speak("Thank you."); > > freeswitch.consoleLog("info","testing"); > freeswitch.email("b2m at a-cti.com ", > "saraswathi.devaraj at a-cti.com > ", > "subject: Voicemail from 801\n", > "Hello,\n\nYou've got a voicemail, click the > attachment to listen to it.", > "/usr/local/freeswitch/recordings/503.mp3", > "", > ""); > freeswitch.consoleLog("info","hai"); > > > > switchconf : > > > > > > > Thanks for your help!! > > Thanks, > Bala > > > > On Sat, Feb 4, 2012 at 12:31 AM, John > wrote: > > On 03/02/12 18:00, Thomas Hoellriegel wrote: > > > > Its works fine!! > Good news! > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/28b62aa0/attachment.html From philippe at ppmt.org Tue Feb 7 17:42:18 2012 From: philippe at ppmt.org (Philippe Le Toquin) Date: Tue, 7 Feb 2012 09:42:18 -0500 Subject: [Freeswitch-users] needs some advice to secure my system Message-ID: Hello, Sorry to ask like that but could someone points me to some site that explains exactly what I need to open towards the internet so that my FS server is working while limiting its visibility? since 1st of February I have an IP that continually sends me SIP Register request at a rate of 70KB/s. I have complained to my internet provider but they refuse to help saying that the problem is on my side. I also logged a complain to the provider on that IP and am waiting on that. At the moment on my firewall I opened port 5060 and 5080 (well now I blocked as well that IP) but I want to know if both are really needed or if I could block one of them or may be limit the port to some IP. Any help/links will be gladly received thanks /Philippe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/d612319c/attachment.html From freeswitch at earthspike.net Tue Feb 7 17:42:51 2012 From: freeswitch at earthspike.net (John) Date: Tue, 07 Feb 2012 14:42:51 +0000 Subject: [Freeswitch-users] Mod_dingaling and openfire issue In-Reply-To: <1FFF97C269757C458224B7C895F35F15039F7C@cantor.std.visionutv.se> References: <1328616953.19943.YahooMailNeo@web29404.mail.ird.yahoo.com> <4F312967.5050807@anew.com.ve> <1FFF97C269757C458224B7C895F35F15039F7C@cantor.std.visionutv.se> Message-ID: <4F31386B.70604@earthspike.net> Notwithstanding FS crashing, for which a JIRA bug should be raised, there are issues with s2s in Openfire 3.7.0 (specifically IP dialback authentication) so you should use 3.7.1. That said, there are some fairly ugly thread pool issues on Openfire that make me favour ejabberd for production environments where I have a choice in the matter. John On 07/02/12 13:51, Peter Olsson wrote: > > If FS crashes, it's a bug, and should be reported to Jira. Please > follow the instructions here to file a complete bug report. > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > /Peter > > *Fr?n:*freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *F?r *Saugort > Dario Garcia Tovar > *Skickat:* den 7 februari 2012 14:39 > *Till:* freeswitch-users at lists.freeswitch.org > *?mne:* Re: [Freeswitch-users] Mod_dingaling and openfire issue > > Thanks for your comment, > > well, I trying to use openfire as screen pop-up. My project was to > build an IVR script in lua, collect some data, consume some > webservice, then populate a variable. Then transfer the call to a > queue, then when the call is delivered to an agent, use chat command > to send a pop-up to the agent wih the collected data. > > So far, Openfire seems to be working fine, I have two clients > registered in Openfire: one using spark, the second using SJPhone > (also, it is registered in FS with an extension X). I can send message > between both clients, thats why I think openfire is working fine. > > When I activate server-server integration in FS to connect to Openfire > , as soon I use chat app or FS receive Openfire messaging FS > stop/exit, I mean, FS is not longer running. Why? FS should not end if > mod fail or other end send wrong or unexpected messages > > When mod_dingaling was developed which jabber server used? > > > > > On 2/7/2012 7:45 AM, Bob Smith wrote: > > Hi, > > What are you looking to do with Openfire and FreeSwitch once you have > it up and running ? > > Speaking as someone who's tried to integrate FreeSwitch and Openfire > in relation to conferencing, I've found it to be a bit of a waste of > time, as mod_dingaling isn't ready for prime time as far as XMPP > integration goes. > > Others may disagree with me here, but I've given up and abandoned the > idea until the codebase is improved sometime in the distant future. > > However good luck with your project and keep us updated > > Bob > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/4b7ba579/attachment-0001.html From gb10hkzo-freeswitch at yahoo.co.uk Tue Feb 7 17:47:33 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Tue, 7 Feb 2012 14:47:33 +0000 (GMT) Subject: [Freeswitch-users] needs some advice to secure my system Message-ID: <1328626053.1329.YahooMailNeo@web29404.mail.ird.yahoo.com> Hello Philippe, The idea I am currently working towards implementing is : - SIP Origination / Inbound SIP? =? IP range ACL with carrier - User Origination / Devlivery = OpenVPN + SNOM Handsets (they have a built-in OpenVPN client, quite cool !) You can lock down OpenVPN quite tight so it hardly reponds at all to unauthorised requests. I have only just started my testing, but other than some issues with inbound calls and multiple profiles? that I'm trying to iron out at the moment, everything seems to be working ok. Bob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/359b686f/attachment.html From philippe at ppmt.org Tue Feb 7 17:54:22 2012 From: philippe at ppmt.org (Philippe Le Toquin) Date: Tue, 7 Feb 2012 09:54:22 -0500 Subject: [Freeswitch-users] needs some advice to secure my system In-Reply-To: <1328626053.1329.YahooMailNeo@web29404.mail.ird.yahoo.com> References: <1328626053.1329.YahooMailNeo@web29404.mail.ird.yahoo.com> Message-ID: wow! Not sure I understood all I am afraid. - SIP Origination / Inbound SIP = IP range ACL with carrier Does the above mean to find out the ip of the SIP provider I use and only authorising these ones? /Philippe On 7 February 2012 09:47, Bob Smith wrote: > Hello Philippe, > > The idea I am currently working towards implementing is : > > - SIP Origination / Inbound SIP = IP range ACL with carrier > - User Origination / Devlivery = OpenVPN + SNOM Handsets (they have a > built-in OpenVPN client, quite cool !) > > You can lock down OpenVPN quite tight so it hardly reponds at all to > unauthorised requests. > > I have only just started my testing, but other than some issues with > inbound calls and multiple profiles that I'm trying to iron out at the > moment, everything seems to be working ok. > > Bob > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/043cf849/attachment.html From krice at freeswitch.org Tue Feb 7 17:58:27 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 07 Feb 2012 08:58:27 -0600 Subject: [Freeswitch-users] needs some advice to secure my system In-Reply-To: Message-ID: Just check out the FS wiki for Fail2Ban... What many of you are probably seeing is a SipVicious brute force attack... Fail2Ban will greatly reduce those problems K On 2/7/12 8:54 AM, "Philippe Le Toquin" wrote: > wow! > > Not sure I understood all I am afraid. > > - SIP Origination / Inbound SIP? =? IP range ACL with carrier > > Does the above mean to find out the ip of the SIP provider I use and only > authorising these ones? > > /Philippe > > On 7 February 2012 09:47, Bob Smith wrote: >> Hello Philippe, >> >> The idea I am currently working towards implementing is : >> >> - SIP Origination / Inbound SIP? =? IP range ACL with carrier >> - User Origination / Devlivery = OpenVPN + SNOM Handsets (they have a >> built-in OpenVPN client, quite cool !) >> >> You can lock down OpenVPN quite tight so it hardly reponds at all to >> unauthorised requests. >> >> I have only just started my testing, but other than some issues with inbound >> calls and multiple profiles? that I'm trying to iron out at the moment, >> everything seems to be working ok. >> >> Bob >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/6e4eb37f/attachment.html From avi at avimarcus.net Tue Feb 7 18:00:10 2012 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 7 Feb 2012 17:00:10 +0200 Subject: [Freeswitch-users] needs some advice to secure my system In-Reply-To: References: <1328626053.1329.YahooMailNeo@web29404.mail.ird.yahoo.com> Message-ID: 1) Only open ports that are needed, see a list here: http://wiki.freeswitch.org/wiki/Firewall 2) For linux, fail2ban is.. necessary? http://wiki.freeswitch.org/wiki/Fail2ban The DOS filter would have banned those registrations in just a few seconds... If you're on *bsd, you can certainly manually block that IP with whatever firewall is there. If you're on windows.. there isn't anything like fail2ban as far as I know.. -Avi On Tue, Feb 7, 2012 at 4:54 PM, Philippe Le Toquin wrote: > wow! > > Not sure I understood all I am afraid. > > > - SIP Origination / Inbound SIP = IP range ACL with carrier > > Does the above mean to find out the ip of the SIP provider I use and only > authorising these ones? > > /Philippe > > On 7 February 2012 09:47, Bob Smith wrote: > >> Hello Philippe, >> >> The idea I am currently working towards implementing is : >> >> - SIP Origination / Inbound SIP = IP range ACL with carrier >> - User Origination / Devlivery = OpenVPN + SNOM Handsets (they have a >> built-in OpenVPN client, quite cool !) >> >> You can lock down OpenVPN quite tight so it hardly reponds at all to >> unauthorised requests. >> >> I have only just started my testing, but other than some issues with >> inbound calls and multiple profiles that I'm trying to iron out at the >> moment, everything seems to be working ok. >> >> Bob >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/354d1b10/attachment-0001.html From b2m at a-cti.com Tue Feb 7 18:04:39 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Tue, 7 Feb 2012 20:34:39 +0530 Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: References: <005801cce12e$05358300$0fa08900$@net.au> <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> <4F2A9482.1020100@earthspike.net> <4F2B8B51.2080506@earthspike.net> <4F2C2EFA.9030105@earthspike.net> <4F3137A7.2040908@earthspike.net> Message-ID: I did, also I have no issue sending email outside FS. Thanks, Bala On Tue, Feb 7, 2012 at 8:31 PM, Balamurugan Mahendran < balamurugan at adaptavant.com> wrote: > I did, also I have no issue sending email outside FS. > > Thanks, > Bala > > On Tue, Feb 7, 2012 at 8:09 PM, John wrote: > >> I updated the wiki mod_voicemail page a few days ago with some >> instructions for debugging email from freeswitch. Have you tried those? >> >> >> On 07/02/12 13:47, Balamurugan Mahendran wrote: >> >> All, >> >> I am having the same issue, its not sending email(extension --> lua >> script) getting *"Segmentation fault"* >> >> xml : >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> > value="$${outbound_caller_name}"/> >> > value="$${outbound_caller_id}"/> >> >> >> >> >> >> dialplan : >> >> >> >> >> >> >> >> >> >> >> Lua : >> >> caller=503; >> freeswitch.consoleLog("info","From :"..caller); >> session:set_tts_parms("flite", "slt"); >> session:speak("Welcome To Voice Mail !. You Can Leave Your Message >> Here."); >> path="/usr/local/freeswitch/recordings/"; >> prompt=caller..".mp3"; >> recpath=path..prompt; >> freeswitch.consoleLog("info","record path="..recpath); >> session:recordFile(recpath,30,10,10); >> session:speak("Thank you."); >> >> freeswitch.consoleLog("info","testing"); >> freeswitch.email("b2m at a-cti.com", >> "saraswathi.devaraj at a-cti.com", >> "subject: Voicemail from 801\n", >> "Hello,\n\nYou've got a voicemail, click the attachment >> to listen to it.", >> "/usr/local/freeswitch/recordings/503.mp3", >> "", >> ""); >> freeswitch.consoleLog("info","hai"); >> >> >> >> switchconf : >> >> >> >> >> >> >> Thanks for your help!! >> >> Thanks, >> Bala >> >> >> >> On Sat, Feb 4, 2012 at 12:31 AM, John wrote: >> >>> On 03/02/12 18:00, Thomas Hoellriegel wrote: >>> > >>> > Its works fine!! >>> Good news! >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/7d45def1/attachment.html From gb10hkzo-freeswitch at yahoo.co.uk Tue Feb 7 18:11:27 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Tue, 7 Feb 2012 15:11:27 +0000 (GMT) Subject: [Freeswitch-users] needs some advice to secure my system Message-ID: <1328627487.82720.YahooMailNeo@web29405.mail.ird.yahoo.com> Yes, that is exactly what I mean. My VoIP carrier provides details of a /24 IP address range that they use on their side.? Therefore all I need to do is filter those out? at firewall level. >Does the above mean to find out the ip of the SIP provider I use and only authorising these ones? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/f18c6b2d/attachment.html From dgarcia at anew.com.ve Tue Feb 7 18:20:45 2012 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Tue, 07 Feb 2012 10:50:45 -0430 Subject: [Freeswitch-users] Mod_dingaling and openfire issue In-Reply-To: <4F31386B.70604@earthspike.net> References: <1328616953.19943.YahooMailNeo@web29404.mail.ird.yahoo.com> <4F312967.5050807@anew.com.ve> <1FFF97C269757C458224B7C895F35F15039F7C@cantor.std.visionutv.se> <4F31386B.70604@earthspike.net> Message-ID: <4F31414D.8060402@anew.com.ve> Hi John, I have opened a ticket in Jira. I am doing a "make current" to re-check if the issue still. I use Openfire openfire-3.7.1-1.i386.rpm Also, I will test other jabber server to check if the issue rise, i will thinking in use tigase or ejabberd. Thanks for your comments On 2/7/2012 10:12 AM, John wrote: > Notwithstanding FS crashing, for which a JIRA bug should be raised, > there are issues with s2s in Openfire 3.7.0 (specifically IP dialback > authentication) so you should use 3.7.1. That said, there are some > fairly ugly thread pool issues on Openfire that make me favour > ejabberd for production environments where I have a choice in the matter. > > John > > On 07/02/12 13:51, Peter Olsson wrote: >> >> If FS crashes, it's a bug, and should be reported to Jira. Please >> follow the instructions here to file a complete bug report. >> >> http://wiki.freeswitch.org/wiki/Reporting_Bugs >> >> /Peter >> >> *Fr?n:*freeswitch-users-bounces at lists.freeswitch.org >> [mailto:freeswitch-users-bounces at lists.freeswitch.org] *F?r *Saugort >> Dario Garcia Tovar >> *Skickat:* den 7 februari 2012 14:39 >> *Till:* freeswitch-users at lists.freeswitch.org >> *?mne:* Re: [Freeswitch-users] Mod_dingaling and openfire issue >> >> Thanks for your comment, >> >> well, I trying to use openfire as screen pop-up. My project was to >> build an IVR script in lua, collect some data, consume some >> webservice, then populate a variable. Then transfer the call to a >> queue, then when the call is delivered to an agent, use chat command >> to send a pop-up to the agent wih the collected data. >> >> So far, Openfire seems to be working fine, I have two clients >> registered in Openfire: one using spark, the second using SJPhone >> (also, it is registered in FS with an extension X). I can send >> message between both clients, thats why I think openfire is working fine. >> >> When I activate server-server integration in FS to connect to >> Openfire , as soon I use chat app or FS receive Openfire messaging FS >> stop/exit, I mean, FS is not longer running. Why? FS should not end >> if mod fail or other end send wrong or unexpected messages >> >> When mod_dingaling was developed which jabber server used? >> >> >> >> >> On 2/7/2012 7:45 AM, Bob Smith wrote: >> >> Hi, >> >> What are you looking to do with Openfire and FreeSwitch once you have >> it up and running ? >> >> Speaking as someone who's tried to integrate FreeSwitch and Openfire >> in relation to conferencing, I've found it to be a bit of a waste of >> time, as mod_dingaling isn't ready for prime time as far as XMPP >> integration goes. >> >> Others may disagree with me here, but I've given up and abandoned the >> idea until the codebase is improved sometime in the distant future. >> >> However good luck with your project and keep us updated >> >> Bob >> >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > 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: 2012.0.1913 / Virus Database: 2112/4794 - Release Date: 02/07/12 > -- Atentamente, *Dario Garc?a* Consultor. CCCT, Nivel C2, Sector Yarey, Mz, Ofc. MZ03a. Caracas-Venezuela. Tel?fono: +58 212 9081842 Cel: +58 412 2221515 dgarcia at anew.com.ve http://www.anew.com.ve -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/ffc39492/attachment-0001.html From sat at calgaryit.com Tue Feb 7 18:27:05 2012 From: sat at calgaryit.com (George Sapak) Date: Tue, 7 Feb 2012 08:27:05 -0700 (MST) Subject: [Freeswitch-users] sip_invite_from_params In-Reply-To: <739396024.1793.1328628304395.JavaMail.root@server3> Message-ID: <596381508.1795.1328628424989.JavaMail.root@server3> can someone give me an example in use, I have looked at the wiki and its pretty unclear, I am trying to remove a string in an invite -> ;phone-context=national INVITE sip:403XXXXXXX;phone-context=national at 10.185.16.170;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.185.16.169:5060;branch=z9hG4bKght6ecqg4eh7d5sluaj9cl2rr4 From: "Caller";tag=SDj6q1a01-98964 To: Call-ID: SDj6q1a01-2afa0c6426482b0cfa2885f32878c86c-o0t3g30 CSeq: 62218 INVITE Content-Type: application/sdp Contact: User-Agent: Nortel SESM 14.0.6.0 Max-Forwards: 19 Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,com.nortelnetworks.im.encryption,r eplaces,100rel,tdialog Allow: UPDATE,REFER x-nt-corr-id: 871f3514-29e7-1b21-a633-000e0cb7d3a0 x-nt-location: -1 Content-Length: 206 Route: Thank You, George From chris.chen2004 at gmail.com Tue Feb 7 18:29:20 2012 From: chris.chen2004 at gmail.com (Chris Chen) Date: Tue, 7 Feb 2012 10:29:20 -0500 Subject: [Freeswitch-users] Mod_dingaling and openfire issue In-Reply-To: <4F31414D.8060402@anew.com.ve> References: <1328616953.19943.YahooMailNeo@web29404.mail.ird.yahoo.com> <4F312967.5050807@anew.com.ve> <1FFF97C269757C458224B7C895F35F15039F7C@cantor.std.visionutv.se> <4F31386B.70604@earthspike.net> <4F31414D.8060402@anew.com.ve> Message-ID: Actually my FS has been working with ejabberd on the same server for a couple of years without the issue you described. Just go with ejabberd. Thanks, Chris On Tue, Feb 7, 2012 at 10:20 AM, Saugort Dario Garcia Tovar wrote: > Hi John, > > I have opened a ticket in Jira. > > I am doing a "make current" to re-check if the issue still. I use Openfire > openfire-3.7.1-1.i386.rpm > > Also, I will test other jabber server to check if the issue rise, i will > thinking in use tigase or ejabberd. > > Thanks for your comments > > > On 2/7/2012 10:12 AM, John wrote: > > Notwithstanding FS crashing, for which a JIRA bug should be raised, there > are issues with s2s in Openfire 3.7.0 (specifically IP dialback > authentication) so you should use 3.7.1.? That said, there are some fairly > ugly thread pool issues on Openfire that make me favour ejabberd for > production environments where I have a choice in the matter. > > John > > On 07/02/12 13:51, Peter Olsson wrote: > > If FS crashes, it?s a bug, and should be reported to Jira. Please follow the > instructions here to file a complete bug report. > > > > http://wiki.freeswitch.org/wiki/Reporting_Bugs > > > > /Peter > > > > Fr?n: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Saugort Dario > Garcia Tovar > Skickat: den 7 februari 2012 14:39 > Till: freeswitch-users at lists.freeswitch.org > ?mne: Re: [Freeswitch-users] Mod_dingaling and openfire issue > > > > Thanks for your comment, > > well, I trying to use openfire as screen pop-up. My project was to build an > IVR script in lua, collect some data, consume some webservice, then populate > a variable. Then transfer the call to a queue, then when the call is > delivered to an agent, use chat command to send a pop-up to the agent wih > the collected data. > > So far, Openfire seems to be working fine, I have two clients registered in > Openfire:? one using spark, the second using SJPhone (also, it is registered > in FS with an extension X). I can send message between both clients, thats > why I think openfire is working fine. > > When I activate server-server integration in FS to connect to Openfire , as > soon I use chat app or FS receive Openfire messaging FS stop/exit, I mean, > FS is not longer running. Why? FS should not end if mod fail or other end > send wrong or unexpected messages > > When mod_dingaling was developed which jabber server used? > > > > > On 2/7/2012 7:45 AM, Bob Smith wrote: > > Hi, > > What are you looking to do with Openfire and FreeSwitch once you have it up > and running ? > > Speaking as someone who's tried to integrate FreeSwitch and Openfire in > relation to conferencing, I've found it to be a bit of a waste of time, as > mod_dingaling isn't ready for prime time as far as XMPP integration goes. > > Others may disagree with me here, but I've given up and abandoned the idea > until the codebase is improved sometime in the distant future. > > However good luck with your project and keep us updated > > Bob > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > 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: 2012.0.1913 / Virus Database: 2112/4794 - Release Date: 02/07/12 > > > > -- > Atentamente, > Dario Garc?a > Consultor. > > CCCT, Nivel C2, Sector Yarey, Mz, > Ofc. MZ03a. > Caracas-Venezuela. > Tel?fono: +58 212 9081842 > Cel: +58 412 2221515 > dgarcia at anew.com.ve > http://www.anew.com.ve > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > 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 Tue Feb 7 18:31:31 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Tue, 7 Feb 2012 12:31:31 -0300 Subject: [Freeswitch-users] timeout param in mod_xml_cdr In-Reply-To: <4F3116E8.7000200@gmail.com> References: <4F3116E8.7000200@gmail.com> Message-ID: Can you wikify that? Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Tuesday, February 7, 2012 at 9:19 AM, Carlo Dimaggio wrote: > Hi all, > > I have discovered a param (timeout) implemented in mod_xml_cdr but not documented. > The param sets a timeout of the posting action to the web server (in the example wait 5 seconds before set the web post as "failed"): > > > > It is interested when you want to be sure that there are no unhandled open sessions (in my case I want to be sure that all cdr are submitted to the billing engine). > > > 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 > > 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/20120207/96f5443b/attachment.html From philippe at ppmt.org Tue Feb 7 18:40:03 2012 From: philippe at ppmt.org (Philippe Le Toquin) Date: Tue, 7 Feb 2012 10:40:03 -0500 Subject: [Freeswitch-users] needs some advice to secure my system In-Reply-To: References: Message-ID: Hi Actually I already have fail2ban installed and it does block the IP correctly but my main issue is that in Canada we don't have unlimited internet and this is eating 6Gig per day out of my 50Gig monthly :( I am going to change the IP of my router so that the request will no longer reach me but eventually he will find me back unless I can make myself more invisible. On 7 February 2012 09:58, Ken Rice wrote: > Just check out the FS wiki for Fail2Ban... > > What many of you are probably seeing is a SipVicious brute force attack... > Fail2Ban will greatly reduce those problems > > K > > > > On 2/7/12 8:54 AM, "Philippe Le Toquin" wrote: > > wow! > > Not sure I understood all I am afraid. > > - SIP Origination / Inbound SIP = IP range ACL with carrier > > Does the above mean to find out the ip of the SIP provider I use and only > authorising these ones? > > /Philippe > > On 7 February 2012 09:47, Bob Smith > wrote: > > Hello Philippe, > > The idea I am currently working towards implementing is : > > - SIP Origination / Inbound SIP = IP range ACL with carrier > - User Origination / Devlivery = OpenVPN + SNOM Handsets (they have a > built-in OpenVPN client, quite cool !) > > You can lock down OpenVPN quite tight so it hardly reponds at all to > unauthorised requests. > > I have only just started my testing, but other than some issues with > inbound calls and multiple profiles that I'm trying to iron out at the > moment, everything seems to be working ok. > > Bob > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/164a242d/attachment.html From kris at kriskinc.com Tue Feb 7 18:40:05 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 7 Feb 2012 10:40:05 -0500 Subject: [Freeswitch-users] needs some advice to secure my system In-Reply-To: <1328627487.82720.YahooMailNeo@web29405.mail.ird.yahoo.com> References: <1328627487.82720.YahooMailNeo@web29405.mail.ird.yahoo.com> Message-ID: Bob, Set inbound-acl on your provider/internet facing profile to the providers range: x.x.x.x/24. If you want "firewall" functionality you'll need to use another firewall or iptables. On Tue, Feb 7, 2012 at 10:11 AM, Bob Smith wrote: > > Yes, that is exactly what I mean. > > My VoIP carrier provides details of a /24 IP address range that they use on > their side.? Therefore all I need to do is filter those out? at firewall > level. > >>Does the above mean to find out the ip of the SIP provider I use and only >> authorising these ones? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From brian at freeswitch.org Tue Feb 7 18:43:54 2012 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Feb 2012 09:43:54 -0600 Subject: [Freeswitch-users] sip_invite_from_params In-Reply-To: <596381508.1795.1328628424989.JavaMail.root@server3> References: <596381508.1795.1328628424989.JavaMail.root@server3> Message-ID: <296E4016-6F0A-4E8E-84B7-2FE60CB29FE6@freeswitch.org> Why? What is your goal? This should be a simple dialplan rewrite really. /b On Feb 7, 2012, at 9:27 AM, George Sapak wrote: > can someone give me an example in use, I have looked at the wiki and its pretty unclear, I am trying to remove a string in an invite -> ;phone-context=national > > INVITE sip:403XXXXXXX;phone-context=national at 10.185.16.170;user=phone SIP/2.0 > Via: SIP/2.0/UDP 10.185.16.169:5060;branch=z9hG4bKght6ecqg4eh7d5sluaj9cl2rr4 > From: "Caller";tag=SDj6q1a01-98964 > To: > Call-ID: SDj6q1a01-2afa0c6426482b0cfa2885f32878c86c-o0t3g30 > CSeq: 62218 INVITE > Content-Type: application/sdp > Contact: > User-Agent: Nortel SESM 14.0.6.0 > Max-Forwards: 19 > Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,com.nortelnetworks.im.encryption,r eplaces,100rel,tdialog > Allow: UPDATE,REFER > x-nt-corr-id: 871f3514-29e7-1b21-a633-000e0cb7d3a0 > x-nt-location: -1 > Content-Length: 206 > Route: > > > Thank You, > George -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/eba30040/attachment.html From philippe at ppmt.org Tue Feb 7 18:47:05 2012 From: philippe at ppmt.org (Philippe Le Toquin) Date: Tue, 7 Feb 2012 10:47:05 -0500 Subject: [Freeswitch-users] needs some advice to secure my system In-Reply-To: References: <1328626053.1329.YahooMailNeo@web29404.mail.ird.yahoo.com> Message-ID: Thanks Marcus I will have a look at all these links I really appreciate everyone trying to help and not being dismissive of my amateur status! /Philippe On 7 February 2012 10:00, Avi Marcus wrote: > 1) Only open ports that are needed, see a list here: > http://wiki.freeswitch.org/wiki/Firewall > 2) For linux, fail2ban is.. necessary? > http://wiki.freeswitch.org/wiki/Fail2ban > The DOS filter would have banned those registrations in just a few > seconds... > > If you're on *bsd, you can certainly manually block that IP with whatever > firewall is there. > > If you're on windows.. there isn't anything like fail2ban as far as I > know.. > > -Avi > > > On Tue, Feb 7, 2012 at 4:54 PM, Philippe Le Toquin wrote: > >> wow! >> >> Not sure I understood all I am afraid. >> >> >> - SIP Origination / Inbound SIP = IP range ACL with carrier >> >> Does the above mean to find out the ip of the SIP provider I use and only >> authorising these ones? >> >> /Philippe >> >> On 7 February 2012 09:47, Bob Smith wrote: >> >>> Hello Philippe, >>> >>> The idea I am currently working towards implementing is : >>> >>> - SIP Origination / Inbound SIP = IP range ACL with carrier >>> - User Origination / Devlivery = OpenVPN + SNOM Handsets (they have a >>> built-in OpenVPN client, quite cool !) >>> >>> You can lock down OpenVPN quite tight so it hardly reponds at all to >>> unauthorised requests. >>> >>> I have only just started my testing, but other than some issues with >>> inbound calls and multiple profiles that I'm trying to iron out at the >>> moment, everything seems to be working ok. >>> >>> Bob >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> 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 >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/38c18336/attachment.html From gb10hkzo-freeswitch at yahoo.co.uk Tue Feb 7 19:06:17 2012 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob Smith) Date: Tue, 7 Feb 2012 16:06:17 +0000 (GMT) Subject: [Freeswitch-users] needs some advice to secure my system Message-ID: <1328630777.82360.YahooMailNeo@web29403.mail.ird.yahoo.com> Kristian, Huh ? I was the one who suggested the firewall filtering and OpenVPN in the first place, no need to teach granny how to suck eggs.? ;-) Bob From David at jajah.com Tue Feb 7 12:05:13 2012 From: David at jajah.com (=?iso-8859-1?Q?David_Artu=F1edo_Guill=E9n?=) Date: Tue, 7 Feb 2012 11:05:13 +0200 Subject: [Freeswitch-users] FreeSwitch state machines for CHANNEL_STATE and CHANNEL_CALLSTATE Message-ID: <569384504C492C4580E88B5D54DFEAEA1AD955D1E1@jjex01.jajah.dublin> Hi, I'm new to FreeSwitch world. I'm looking at using the mod_event_socket interface to write some simple call control app. The Events I'm interested at are CHANNEL_STATE and CHANNEL_CALLSTATE to recognize the status of the calls and how they progress. I have not found any docs describing the state machines for channels and Calls. I look up in the Source code as well and couldn't find the place where the state machine is implemented. Does anybody know where can I find this info? EG, when setting up some call through FS these are the events I am getting with the uuids associated: bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_INIT bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_CALLSTATE RINGING bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_ROUTING bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_CONSUME_MEDIA 9a66eaf9-e11e-4f9c-9623-19deac73fc2e CHANNEL_CALLSTATE ACTIVE bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_CALLSTATE EARLY bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_CALLSTATE ACTIVE bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_EXECUTE bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_HIBERNATE 9a66eaf9-e11e-4f9c-9623-19deac73fc2e CHANNEL_STATE CS_HIBERNATE bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_RESET 9a66eaf9-e11e-4f9c-9623-19deac73fc2e CHANNEL_STATE CS_RESET 9a66eaf9-e11e-4f9c-9623-19deac73fc2e CHANNEL_STATE CS_SOFT_EXECUTE bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_SOFT_EXECUTE bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_CONSUME_MEDIA bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_EXCHANGE_MEDIA bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_CALLSTATE HANGUP bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_HANGUP 9a66eaf9-e11e-4f9c-9623-19deac73fc2e CHANNEL_STATE CS_PARK bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_REPORTING bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_CALLSTATE DOWN bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_DESTROY Thanks in advance. David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/faa8a8b7/attachment-0001.html From bligthart at btlnet.co.uk Tue Feb 7 13:42:15 2012 From: bligthart at btlnet.co.uk (Boudewyn Ligthart) Date: Tue, 7 Feb 2012 02:42:15 -0800 Subject: [Freeswitch-users] Unexpected BYE's after 100 calls Message-ID: Hi, We have been testing with Freeswitch (hacked-20120106T151802Z). There are two profiles, allowing calls through using ACL, the core is running with ODBC on MySQL, all running on ubuntu, virtualized in XEN. When testing with SIPP and standard uac towards Freeswitch there are no issue's with the SIP messaging, SIPP on another machine acts as uas. So call flows SIPP --> Freeswitch --> SIPP. When testing with SIPP (using same routing) streaming audio after about 100 calls (5 seconds in) there are unexpected BYE's coming from Freeswitch (I have even lowered the CPS count). I do not see any error's in the logs, just the BYE's. What could be causing this issue? Thanks, Bou -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/f0fcff69/attachment.html From vishal.kakkar at gmail.com Tue Feb 7 14:37:38 2012 From: vishal.kakkar at gmail.com (Vishal Kakkar) Date: Tue, 7 Feb 2012 17:07:38 +0530 Subject: [Freeswitch-users] Invalid UTF-8 character Message-ID: Hi All, Anybody have idea, what could cause this WARNING.. I am getting this one since i upgraded to latest git last week, dont know if its related to that though. *switch_xml.c:2329 Invalid UTF-8 character to ampersand, skip it* There are 100s of such lines getting printed on each call. So want to fix the same before i make it live. Thanks a lot. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/72c800a0/attachment.html From brian at freeswitch.org Tue Feb 7 19:10:55 2012 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Feb 2012 10:10:55 -0600 Subject: [Freeswitch-users] Invalid UTF-8 character In-Reply-To: References: Message-ID: <451E0C1C-7F3E-47D8-8D09-1884654F3E26@freeswitch.org> fix it in the lang_de stuff or remove the german stuff. /b On Feb 7, 2012, at 5:37 AM, Vishal Kakkar wrote: > Hi All, > > Anybody have idea, what could cause this WARNING.. I am getting this one > since i upgraded to latest git last week, dont know if its related to that > though. > > *switch_xml.c:2329 Invalid UTF-8 character to ampersand, skip it* > > There are 100s of such lines getting printed on each call. So want to fix > the same before i make it live. > > Thanks a lot. -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/bfc20c04/attachment.html From msc at freeswitch.org Tue Feb 7 19:11:45 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Feb 2012 08:11:45 -0800 Subject: [Freeswitch-users] voicemail_say_phone_number In-Reply-To: References: <4F3060D2.7050505@earthspike.net> <6574CFF3DDE54C118E0E9250AC19EDD7@DWP> <01b601cce54e$f865c250$e93146f0$@com> <26ADC64FC37D4049A68787B3A0DCF4FA@DWP> Message-ID: Yeah, this was my bad. I started on this and got interrupted, then came back like an hour later and totally forgot about the fact that of mod_voicemail.c not using the macro. I think today I can do a quick test on that and if it works I'll commit it to git master and then you all can test it. Thanks, MC On Tue, Feb 7, 2012 at 2:22 AM, Yuriy Nasida wrote: > Thank you guys! > > It works. The example with > 'vm_announce_cid=ivr/ivr-this_is_a_call_from.wav' is more correct. > Otherwise FS tries to play message true.wav :) > So, voicemail module doesn't use macro "voicemail_say_phone_number" and as > far as I see the source code of mod_voicemail.c explains this behaviour. > Yes, ability for playing of some wav file if I will have vm from > annonymous would be very useful. > Probably I can add it independently but I believe that your modifying > will more correctly :) > Darcy please let me know if you plan to add this feature in the near > future. > > Anyway thanks again! > > ------------------------------ > From: darcy at primrose.ws > To: freeswitch-users at lists.freeswitch.org > Date: Mon, 6 Feb 2012 23:27:23 -0500 > Subject: Re: [Freeswitch-users] voicemail_say_phone_number > > > If John?s works, use it, I could not make it work but the example I show > below played the greeting before the clid, I have not added anything to > play an annonymous greeting yet as I have not had time. > > in mod_voicemail.c you have the following code: > if (!zstr(cbt->cid_number) && (vm_announce_cid = > switch_channel_get_variable(channel, "vm_announce_cid"))) { > switch_ivr_play_file(session, NULL, vm_announce_cid, NULL); > switch_ivr_sleep(session, 500, SWITCH_TRUE, NULL); > switch_ivr_say(session, cbt->cid_number, NULL, "name_spelled", > "pronounced", NULL, NULL); > } > Which indicates you will play the file in variable ?vm_announce_cid?. > > Darcy > In Sunny Ottawa Canada > > *From:* Bote Man > *Sent:* Monday, February 06, 2012 11:14 PM > *To:* 'FreeSWITCH Users Help' > *Subject:* Re: [Freeswitch-users] voicemail_say_phone_number > > > Well, now I'm cornfused. The original example by John used vm_announce_cid > as a Boolean switch. But you've tested it as a string that points to the > desired sound file? > > > > I have no means to test this nor access to sources right now and I just > added John's example to the wiki. I should back it out or correct it based > on your results. > > > > Please advise. > > > > > > John Boteler > > Bote Communications > > in rainy Fort Lauderdale, FL > > > > > > > > *From:* Darcy > *Sent:* Monday, 06 February, 2012 22:02 > > The message, this_is_a_call_from actually has to be set in the > dialplan it appears, the fs plays the file set in vm_announce_cid, a simple > dial plan below reflects one way of doing this, tested and it works. > Needs more time to suit the total requirements, but this makes it a little > more professional by adding the message in front of the number. > > > > > > > > > > data="vm_announce_cid=ivr/ivr-this_is_a_call_from.wav"/> > > > > > > > > > > > > Darcy > > > > *?* > > On Mon, Feb 6, 2012 at 3:22 PM, John wrote: > > Michael, > > It does work, but it's a bit 'rough': all it does it speak the number just > before the date. So I can understand why it was not documented... > ? > > John > > PS. For those who want to know where to insert this, put the line marked > with + into your conf/dialplan/default.xml file: > > > > expression="^vmain$|^4000$|^\*98$"> > > > + > > > > > > > On 06/02/12 22:00, Michael Collins wrote: > > Yuriy, > > ?Please set vm_announce_cid to true prior to checking voicemail and see > if it works. If it does, please let me know. If you can add it to the wiki > then do so, otherwise one of our intrepid community members will do it. > > -MC > > 2012/2/6 Yuriy Nasida > > Hello list, > > > > I would like to have one simple ability to listen the phone number of > caller when I check my voicemail. At present I listen date of message only. > I have found the macro "voicemail_say_phone_number" in > conf/lang/en/vm/sounds.xml. But... when FS uses it ? Can I get it without > modifying of source code of voicemail module ? > > > > Please advise. > > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: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 FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/bbd016d0/attachment-0001.html From sat at calgaryit.com Tue Feb 7 19:29:49 2012 From: sat at calgaryit.com (George Sapak) Date: Tue, 7 Feb 2012 09:29:49 -0700 (MST) Subject: [Freeswitch-users] sip_invite_from_params In-Reply-To: <2048568642.1819.1328632138522.JavaMail.root@server3> Message-ID: <1907034309.1821.1328632189470.JavaMail.root@server3> i already have it setup, my calls go through to the right extensions and I have done some rewitting in public.xml file, but I have having some wierdness on the handsets with call ID, when the phone rings the ID show correctly but when its picked up it shows it with the ;national... sting included. Here is how I have done the routing and rewritte: public.xml Thank You, George ----- Original Message ----- From: "Brian West" To: "FreeSWITCH Users Help" Sent: Tuesday, February 7, 2012 8:43:54 AM Subject: Re: [Freeswitch-users] sip_invite_from_params Why? What is your goal? This should be a simple dialplan rewrite really. /b On Feb 7, 2012, at 9:27 AM, George Sapak wrote: can someone give me an example in use, I have looked at the wiki and its pretty unclear, I am trying to remove a string in an invite -> ;phone-context=national INVITE sip:403XXXXXXX;phone-context=national at 10.185.16.170;user=phone SIP/2.0 Via: SIP/2.0/UDP 10.185.16.169:5060;branch=z9hG4bKght6ecqg4eh7d5sluaj9cl2rr4 From: "Caller"< sip:403YYYYYYY;phone-context=national at 10.185.16.169;user=phone;isup-oli=00 >;tag=SDj6q1a01-98964 To: < sip:403XXXXXXX;phone-context=national at 10.185.16.170;user=phone > Call-ID: SDj6q1a01-2afa0c6426482b0cfa2885f32878c86c-o0t3g30 CSeq: 62218 INVITE Content-Type: application/sdp Contact: < sip:403YYYYYYY;phone-context=national at 10.185.16.169:5060;transport=udp > User-Agent: Nortel SESM 14.0.6.0 Max-Forwards: 19 Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x-nortel-sipvc,com.nortelnetworks.im.encryption,r eplaces,100rel,tdialog Allow: UPDATE,REFER x-nt-corr-id: 871f3514-29e7-1b21-a633-000e0cb7d3a0 x-nt-location: -1 Content-Length: 206 Route: < sip:403XXXXXXX;phone-context=national at 10.185.16.170:5060;user=phone;lr > Thank You, George -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 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 Tue Feb 7 19:47:32 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Tue, 7 Feb 2012 13:47:32 -0300 Subject: [Freeswitch-users] FreeSwitch state machines for CHANNEL_STATE and CHANNEL_CALLSTATE In-Reply-To: <569384504C492C4580E88B5D54DFEAEA1AD955D1E1@jjex01.jajah.dublin> References: <569384504C492C4580E88B5D54DFEAEA1AD955D1E1@jjex01.jajah.dublin> Message-ID: The documentation of this state machine is something that I would love to have documented somewhere. Unfortunately, very few people really know how it works and most of them are busy actually coding the core of freeswitch. I wonder if someday someone will accept this challenge and document it, it would help a lot of other not-so-core developers to understand how things work inside FS. Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Tuesday, February 7, 2012 at 6:05 AM, David Artu?edo Guill?n wrote: > > Hi, > > > > > > I?m new to FreeSwitch world. I?m looking at using the mod_event_socket interface to write some simple call control app. The Events I?m interested at are CHANNEL_STATE and CHANNEL_CALLSTATE to recognize the status of the calls and how they progress. I have not found any docs describing the state machines for channels and Calls. I look up in the Source code as well and couldn?t find the place where the state machine is implemented. > > > > > > Does anybody know where can I find this info? > > > > > > EG, when setting up some call through FS these are the events I am getting with the uuids associated: > > > bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_INIT > > > bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_CALLSTATE RINGING > > > bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_ROUTING > > > bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_CONSUME_MEDIA > > > 9a66eaf9-e11e-4f9c-9623-19deac73fc2e CHANNEL_CALLSTATE ACTIVE > > > bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_CALLSTATE EARLY > > > bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_CALLSTATE ACTIVE > > > bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_EXECUTE > > > bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_HIBERNATE > > > 9a66eaf9-e11e-4f9c-9623-19deac73fc2e CHANNEL_STATE CS_HIBERNATE > > > bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_RESET > > > 9a66eaf9-e11e-4f9c-9623-19deac73fc2e CHANNEL_STATE CS_RESET > > > 9a66eaf9-e11e-4f9c-9623-19deac73fc2e CHANNEL_STATE CS_SOFT_EXECUTE > > > bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_SOFT_EXECUTE > > > bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_CONSUME_MEDIA > > > bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_EXCHANGE_MEDIA > > > bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_CALLSTATE HANGUP > > > bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_HANGUP > > > 9a66eaf9-e11e-4f9c-9623-19deac73fc2e CHANNEL_STATE CS_PARK > > > bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_REPORTING > > > bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_CALLSTATE DOWN > > > bbac39e0-9025-4969-a6ba-b49c89b12237 CHANNEL_STATE CS_DESTROY > > > > > > Thanks in advance. > > > > > > David > > > > > > > _________________________________________________________________________ > 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/20120207/b6c09a3a/attachment.html From b2m at a-cti.com Tue Feb 7 19:53:03 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Tue, 7 Feb 2012 22:23:03 +0530 Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: References: <005801cce12e$05358300$0fa08900$@net.au> <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> <4F2A9482.1020100@earthspike.net> <4F2B8B51.2080506@earthspike.net> <4F2C2EFA.9030105@earthspike.net> <4F3137A7.2040908@earthspike.net> Message-ID: Is there anyway for more logs? Thanks for your help!! --Bala On Tue, Feb 7, 2012 at 8:34 PM, Balamurugan Mahendran wrote: > I did, also I have no issue sending email outside FS. > > Thanks, > Bala > > On Tue, Feb 7, 2012 at 8:31 PM, Balamurugan Mahendran < > balamurugan at adaptavant.com> wrote: > >> I did, also I have no issue sending email outside FS. >> >> Thanks, >> Bala >> >> On Tue, Feb 7, 2012 at 8:09 PM, John wrote: >> >>> I updated the wiki mod_voicemail page a few days ago with some >>> instructions for debugging email from freeswitch. Have you tried those? >>> >>> >>> On 07/02/12 13:47, Balamurugan Mahendran wrote: >>> >>> All, >>> >>> I am having the same issue, its not sending email(extension --> lua >>> script) getting *"Segmentation fault"* >>> >>> xml : >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >> value="$${outbound_caller_name}"/> >>> >> value="$${outbound_caller_id}"/> >>> >>> >>> >>> >>> >>> dialplan : >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> >>> Lua : >>> >>> caller=503; >>> freeswitch.consoleLog("info","From :"..caller); >>> session:set_tts_parms("flite", "slt"); >>> session:speak("Welcome To Voice Mail !. You Can Leave Your Message >>> Here."); >>> path="/usr/local/freeswitch/recordings/"; >>> prompt=caller..".mp3"; >>> recpath=path..prompt; >>> freeswitch.consoleLog("info","record path="..recpath); >>> session:recordFile(recpath,30,10,10); >>> session:speak("Thank you."); >>> >>> freeswitch.consoleLog("info","testing"); >>> freeswitch.email("b2m at a-cti.com", >>> "saraswathi.devaraj at a-cti.com", >>> "subject: Voicemail from 801\n", >>> "Hello,\n\nYou've got a voicemail, click the attachment >>> to listen to it.", >>> "/usr/local/freeswitch/recordings/503.mp3", >>> "", >>> ""); >>> freeswitch.consoleLog("info","hai"); >>> >>> >>> >>> switchconf : >>> >>> >>> >>> >>> >>> >>> Thanks for your help!! >>> >>> Thanks, >>> Bala >>> >>> >>> >>> On Sat, Feb 4, 2012 at 12:31 AM, John wrote: >>> >>>> On 03/02/12 18:00, Thomas Hoellriegel wrote: >>>> > >>>> > Its works fine!! >>>> Good news! >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> 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 >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/c578e258/attachment-0001.html From freeswitch-list at puzzled.xs4all.nl Tue Feb 7 21:13:21 2012 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Tue, 07 Feb 2012 19:13:21 +0100 Subject: [Freeswitch-users] FreeSwitch state machines for CHANNEL_STATE and CHANNEL_CALLSTATE In-Reply-To: References: <569384504C492C4580E88B5D54DFEAEA1AD955D1E1@jjex01.jajah.dublin> Message-ID: <4F3169C1.8040708@puzzled.xs4all.nl> On 07-02-12 17:47, Jo?o Mesquita wrote: > The documentation of this state machine is something that I would love > to have documented somewhere. Unfortunately, very few people really know > how it works and most of them are busy actually coding the core of > freeswitch. > > I wonder if someday someone will accept this challenge and document it, > it would help a lot of other not-so-core developers to understand how > things work inside FS. Isn't there a description in the The Book? It's been a while since I last read it but I recall something like that. I might be wrong though. Regards, Patrick From jmesquita at freeswitch.org Tue Feb 7 21:17:00 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Tue, 7 Feb 2012 15:17:00 -0300 Subject: [Freeswitch-users] FreeSwitch state machines for CHANNEL_STATE and CHANNEL_CALLSTATE In-Reply-To: <4F3169C1.8040708@puzzled.xs4all.nl> References: <569384504C492C4580E88B5D54DFEAEA1AD955D1E1@jjex01.jajah.dublin> <4F3169C1.8040708@puzzled.xs4all.nl> Message-ID: <4FAF459084934BD8A735B92DE6BA3533@freeswitch.org> Which section? I just opened up the book and I can't really find anything related. Besides, I honestly didn't think it was one of the purposes of the book. Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Tuesday, February 7, 2012 at 3:13 PM, Patrick Lists wrote: > On 07-02-12 17:47, Jo?o Mesquita wrote: > > The documentation of this state machine is something that I would love > > to have documented somewhere. Unfortunately, very few people really know > > how it works and most of them are busy actually coding the core of > > freeswitch. > > > > I wonder if someday someone will accept this challenge and document it, > > it would help a lot of other not-so-core developers to understand how > > things work inside FS. > > > > > Isn't there a description in the The Book? It's been a while since I > last read it but I recall something like that. I might be wrong though. > > Regards, > Patrick > > _________________________________________________________________________ > 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/20120207/45d460db/attachment.html From msc at freeswitch.org Tue Feb 7 21:28:22 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Feb 2012 10:28:22 -0800 Subject: [Freeswitch-users] voicemail_say_phone_number In-Reply-To: References: <4F3060D2.7050505@earthspike.net> <6574CFF3DDE54C118E0E9250AC19EDD7@DWP> <01b601cce54e$f865c250$e93146f0$@com> <26ADC64FC37D4049A68787B3A0DCF4FA@DWP> Message-ID: Okay, try latest git. I changed mod_voicemail to use the macro and I set vm_announce_cid to be true/false instead of using a file name. Try it with various caller ID number values and let us know how it goes. -MC On Tue, Feb 7, 2012 at 8:11 AM, Michael Collins wrote: > Yeah, this was my bad. I started on this and got interrupted, then came > back like an hour later and totally forgot about the fact that of > mod_voicemail.c not using the macro. I think today I can do a quick test on > that and if it works I'll commit it to git master and then you all can test > it. > > Thanks, > MC > > > On Tue, Feb 7, 2012 at 2:22 AM, Yuriy Nasida wrote: > >> Thank you guys! >> >> It works. The example with >> 'vm_announce_cid=ivr/ivr-this_is_a_call_from.wav' is more correct. >> Otherwise FS tries to play message true.wav :) >> So, voicemail module doesn't use macro "voicemail_say_phone_number" and >> as far as I see the source code of mod_voicemail.c explains this behaviour. >> Yes, ability for playing of some wav file if I will have vm from >> annonymous would be very useful. >> Probably I can add it independently but I believe that your modifying >> will more correctly :) >> Darcy please let me know if you plan to add this feature in the near >> future. >> >> Anyway thanks again! >> >> ------------------------------ >> From: darcy at primrose.ws >> To: freeswitch-users at lists.freeswitch.org >> Date: Mon, 6 Feb 2012 23:27:23 -0500 >> Subject: Re: [Freeswitch-users] voicemail_say_phone_number >> >> >> If John?s works, use it, I could not make it work but the example I >> show below played the greeting before the clid, I have not added anything >> to play an annonymous greeting yet as I have not had time. >> >> in mod_voicemail.c you have the following code: >> if (!zstr(cbt->cid_number) && (vm_announce_cid = >> switch_channel_get_variable(channel, "vm_announce_cid"))) { >> switch_ivr_play_file(session, NULL, vm_announce_cid, NULL); >> switch_ivr_sleep(session, 500, SWITCH_TRUE, NULL); >> switch_ivr_say(session, cbt->cid_number, NULL, >> "name_spelled", "pronounced", NULL, NULL); >> } >> Which indicates you will play the file in variable ?vm_announce_cid?. >> >> Darcy >> In Sunny Ottawa Canada >> >> *From:* Bote Man >> *Sent:* Monday, February 06, 2012 11:14 PM >> *To:* 'FreeSWITCH Users Help' >> *Subject:* Re: [Freeswitch-users] voicemail_say_phone_number >> >> >> Well, now I'm cornfused. The original example by John used >> vm_announce_cid as a Boolean switch. But you've tested it as a string that >> points to the desired sound file? >> >> >> >> I have no means to test this nor access to sources right now and I just >> added John's example to the wiki. I should back it out or correct it based >> on your results. >> >> >> >> Please advise. >> >> >> >> >> >> John Boteler >> >> Bote Communications >> >> in rainy Fort Lauderdale, FL >> >> >> >> >> >> >> >> *From:* Darcy >> *Sent:* Monday, 06 February, 2012 22:02 >> >> The message, this_is_a_call_from actually has to be set in the >> dialplan it appears, the fs plays the file set in vm_announce_cid, a simple >> dial plan below reflects one way of doing this, tested and it works. >> Needs more time to suit the total requirements, but this makes it a little >> more professional by adding the message in front of the number. >> >> >> >> >> >> >> >> >> >> > data="vm_announce_cid=ivr/ivr-this_is_a_call_from.wav"/> >> >> >> >> >> >> >> >> >> >> >> >> Darcy >> >> >> >> *?* >> >> On Mon, Feb 6, 2012 at 3:22 PM, John wrote: >> >> Michael, >> >> It does work, but it's a bit 'rough': all it does it speak the number >> just before the date. So I can understand why it was not documented... >> ? >> >> John >> >> PS. For those who want to know where to insert this, put the line marked >> with + into your conf/dialplan/default.xml file: >> >> >> >> > expression="^vmain$|^4000$|^\*98$"> >> >> >> + >> >> >> >> >> >> >> On 06/02/12 22:00, Michael Collins wrote: >> >> Yuriy, >> >> ?Please set vm_announce_cid to true prior to checking voicemail and see >> if it works. If it does, please let me know. If you can add it to the wiki >> then do so, otherwise one of our intrepid community members will do it. >> >> -MC >> >> 2012/2/6 Yuriy Nasida >> >> Hello list, >> >> >> >> I would like to have one simple ability to listen the phone number of >> caller when I check my voicemail. At present I listen date of message only. >> I have found the macro "voicemail_say_phone_number" in >> conf/lang/en/vm/sounds.xml. But... when FS uses it ? Can I get it without >> modifying of source code of voicemail module ? >> >> >> >> Please advise. >> >> 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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE: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 FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-usersUNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/baf8adb1/attachment-0001.html From rico-freeswitch at ricozome.net Tue Feb 7 19:40:14 2012 From: rico-freeswitch at ricozome.net (rico-freeswitch at ricozome.net) Date: Tue, 07 Feb 2012 17:40:14 +0100 Subject: [Freeswitch-users] Wrong system variables assignations on FreeSwitch system with two NICs Message-ID: <4F3153EE.7000805@ricozome.net> Hi guys, I'm facing a stupid issue on a fresh installed Debian Squeeze (amd64) server with two NICs, and FreeSwitch bottstraped from Git tree : - eth0 is connected to the local LAN (172.16.96.3/24) - eth1 is connected to Internet and have a public IP address. To be exact, eth0 is on a DMZ, but anyway, it's a RFC1918 address. FS is : FreeSWITCH Version 1.0.head (git-d2c9fb5 2012-02-06 14-12-22 -0600) The problem is when I start the server with its default config, mod_sofia automatically binds into the public IP address, and nothing listens on the local IP address ! thougth the CLI, I get this : freeswitch at internal> sofia status Name Type Data State ================================================================================================= 194.79.xxx.xxx alias internal ALIASED internal profile sip:mod_sofia at 194.79.xxx.xxx:5060 RUNNING (0) internal-ipv6 profile sip:mod_sofia@[::1]:5060 RUNNING (0) external profile sip:mod_sofia at 194.79.xxx.xxx:5080 RUNNING (0) external::example.com gateway sip:joeuser at example.com NOREG ================================================================================================= 3 profiles 1 alias freeswitch at internal> eval ${local_ip_v4} 194.79.xxx.xxx I have to precise it is my first attemps to install FS, and I'm discovering the software, so I may forget something... But I didn't found a way to force FS to link "internal" profile with the local IP address. I also could modify sip_profiles/internal.xml and hardcode the correct IP address, but my feeling is it's not the right way, and I guess it would probably have some side effects on the setup ? Last, I tried both old 1.0.6 branch and lastest Git tree, both with compiled packages "ala" Debian way, and bootstraped, and I always got the same behavior... So my question is simple : is this normal, ? and how should I handle FS setup with mutiple NICs ? Thanks for your help, -- Eric Belhomme From Ryan at ocens.com Tue Feb 7 21:32:59 2012 From: Ryan at ocens.com (Ryan Watkins) Date: Tue, 7 Feb 2012 18:32:59 +0000 Subject: [Freeswitch-users] Conference Auto Outcall to External Numbers In-Reply-To: References: <44E5C0A9D48A3246966A4AE04692014D132E3941@CH1PRD0610MB355.namprd06.prod.outlook.com>, Message-ID: <44E5C0A9D48A3246966A4AE04692014D132E9634@CH1PRD0610MB355.namprd06.prod.outlook.com> Thanks Ken! so, I've changed the action application line to the following syntax our SIP provider is flowroute, so I've also tried @sip.flowroute.com and @flowroute.com looking at the logs, I'm getting the following errors when it attempts to bridge the conference to the outbound numbers: 2012-02-07 10:16:23.214806 [ERR] sofia_reg.c:2170 Cannot locate any authentication credentials to complete an authentication request for realm '"sip.flowroute.com"' 2012-02-07 10:16:23.214806 [DEBUG] switch_channel.c:2852 (sofia/external/13605217334 at flowroute.com) Callstate Change RINGING -> HANGUP 2012-02-07 10:16:23.214806 [NOTICE] sofia_reg.c:2190 Hangup sofia/external/13605217334 at flowroute.com [CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING] I am, however, able to make calls from a registered extension to an external number... what does [MANDATORY_IE_MISSING] mean? and any suggestions on how to get this to work? Thanks again! Ryan ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Ken Rice [krice at freeswitch.org] Sent: Saturday, February 04, 2012 10:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference Auto Outcall to External Numbers Yes... Just specify a proper dialstring just like you would any other bridge command.... On 2/4/12 9:21 PM, "Ryan Watkins" > wrote: Sorry for resending this, but need some assistance on this: Is there a way to set a Conference Auto Outcall to numbers external of the PBX? Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ryan Watkins Sent: Tuesday, January 31, 2012 2:28 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference Auto Outcall to External Numbers The call forwarding function itself works, if I set call forwarding for a specific extension and then call that extension it will ring at the forwarded external phone. Also, with call forwarding disabled on the extensions the conference auto outcall does call the internal extensions. There just seems to be a disconnect between the conference outcall to a forwarded extension. But perhaps I should rephrase the question? The conference extension I?m needing to setup should auto outcall to a set of external numbers. I figured the way to do it would be to create internal extensions, and set the call forwarding to the external number. But perhaps there?s a better way to approach this need? For example, the outcall user listing in the conference.xml is as follows: Is there a way to change the data syntax to instead direct the outbound auto call to an external number instead of an internal user? Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, January 31, 2012 1:10 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference Auto Outcall to External Numbers What actually happens when you set the call forwarding? Console debug log w/ SIP trace would be helpful, as would any relevant configs you've got in place. -MC On Tue, Jan 31, 2012 at 12:46 PM, Ryan Watkins > wrote: Morning/Afternoon/Evening all? I?ve been trying to setup a conference call that would use the auto outcall function, and it would call external numbers (like their cell). I?ve tried to setup extensions for the individuals, and set the call forwarding to their external number. Although the auto outcall function will successfully call the extensions when they are set to an internal SIP phone, when call forwarding is enabled the conference call will not call the external number. Any ideas on how to accomplish this? Thanks a lot! Ryan _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/41fb4df1/attachment.html From brian at freeswitch.org Tue Feb 7 21:34:12 2012 From: brian at freeswitch.org (Brian West) Date: Tue, 7 Feb 2012 12:34:12 -0600 Subject: [Freeswitch-users] Wrong system variables assignations on FreeSwitch system with two NICs In-Reply-To: <4F3153EE.7000805@ricozome.net> References: <4F3153EE.7000805@ricozome.net> Message-ID: You'll have to do the manual assignments of your IP's in each profile. You can't depend on the automatic stuff we have in our default configs. hint sip-ip and rtp-ip need to be set for each profile to the proper values. /b On Feb 7, 2012, at 10:40 AM, rico-freeswitch at ricozome.net wrote: > Hi guys, > > I'm facing a stupid issue on a fresh installed Debian Squeeze (amd64) > server with two NICs, and FreeSwitch bottstraped from Git tree : > > - eth0 is connected to the local LAN (172.16.96.3/24) > - eth1 is connected to Internet and have a public IP address. > To be exact, eth0 is on a DMZ, but anyway, it's a RFC1918 address. > > FS is : > FreeSWITCH Version 1.0.head (git-d2c9fb5 2012-02-06 14-12-22 -0600) > > The problem is when I start the server with its default config, > mod_sofia automatically binds into the public IP address, and nothing > listens on the local IP address ! > > thougth the CLI, I get this : > freeswitch at internal> sofia status > Name Type > Data State > ================================================================================================= > 194.79.xxx.xxx alias > internal ALIASED > internal profile > sip:mod_sofia at 194.79.xxx.xxx:5060 RUNNING (0) > internal-ipv6 profile > sip:mod_sofia@[::1]:5060 RUNNING (0) > external profile > sip:mod_sofia at 194.79.xxx.xxx:5080 RUNNING (0) > external::example.com gateway > sip:joeuser at example.com NOREG > ================================================================================================= > 3 profiles 1 alias > freeswitch at internal> eval ${local_ip_v4} > 194.79.xxx.xxx > > I have to precise it is my first attemps to install FS, and I'm > discovering the software, so I may forget something... But I didn't > found a way to force FS to link "internal" profile with the local IP > address. > I also could modify sip_profiles/internal.xml and hardcode the correct > IP address, but my feeling is it's not the right way, and I guess it > would probably have some side effects on the setup ? > > Last, I tried both old 1.0.6 branch and lastest Git tree, both with > compiled packages "ala" Debian way, and bootstraped, and I always got > the same behavior... > > So my question is simple : is this normal, ? and how should I handle FS > setup with mutiple NICs ? > > Thanks for your help, > > -- > Eric Belhomme > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/44818ebc/attachment-0001.html From msc at freeswitch.org Tue Feb 7 21:39:19 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Feb 2012 10:39:19 -0800 Subject: [Freeswitch-users] Astconf2fsconf a software or joke? In-Reply-To: References: <1336366836.3502.7.camel@localhost.localdomain> Message-ID: That page was written 4.5 years ago by someone who rarely comes around any more. I have no idea what the original purpose of the document was. In any case, the information is basically correct. -MC On Mon, Feb 6, 2012 at 8:08 PM, Thomas Hoellriegel wrote: > Hi guys, > I searched the web with a sofware "Astconf2fsconf". > I found a description on: > http://wiki.freeswitch.org/**wiki/Astconf2fsconf > Why this site exists, over all? > No download link, no installationhowto. > What sense does it report about a software which does not appear there? > Very funny-). > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/57a2d1e2/attachment.html From msc at freeswitch.org Tue Feb 7 21:41:31 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Feb 2012 10:41:31 -0800 Subject: [Freeswitch-users] Getting USER_NOT_REGISTERED for no reason ? In-Reply-To: <1328616744.81316.YahooMailNeo@web29403.mail.ird.yahoo.com> References: <1328616744.81316.YahooMailNeo@web29403.mail.ird.yahoo.com> Message-ID: post the logs and a siptrace to pastebin.freeswitch.org so that we can take a look. Without seeing exactly what's happening it's difficult to suggestion anything else. -MC On Tue, Feb 7, 2012 at 4:12 AM, Bob Smith wrote: > Hi Brian, > > Have tried your various suggestions without luck. > > Have also tried replacing > > > > with > > data="sofia/sip/1000%openvpn_udp:_:sofia/sip/1000%internal"/> > > in the dial plan and that doesn't work either > > > Everything resolves fine though...... > > freeswitch at internal> expand echo ${sofia_contact(openvpn_udp/1000)} > sofia/openvpn_udp/sip:1000 at 10.82.1.6:3072;line=2sf2mz02 > freeswitch at internal> expand echo ${sofia_contact(internal/1000)} > sofia/internal/sip:1000 at 10.14.2.2:12592;rinstance=5e131f94a204160f > > > > And yes I have looked through the logs, but there's nothing as blatantly > obvious in there as you seem to think there is ? > > Bob > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/8fe2f507/attachment.html From msc at freeswitch.org Tue Feb 7 21:45:02 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Feb 2012 10:45:02 -0800 Subject: [Freeswitch-users] Conference Auto Outcall to External Numbers In-Reply-To: <44E5C0A9D48A3246966A4AE04692014D132E9634@CH1PRD0610MB355.namprd06.prod.outlook.com> References: <44E5C0A9D48A3246966A4AE04692014D132E3941@CH1PRD0610MB355.namprd06.prod.outlook.com> <44E5C0A9D48A3246966A4AE04692014D132E9634@CH1PRD0610MB355.namprd06.prod.outlook.com> Message-ID: Do you have a gateway set up for the flowroute connection? If not, you'll need to configure one because it looks like flowroute is sending back an auth challenge. If you set up a gateway then you'll be able to respond to the auth challenge and supply the proper username/password. -MC On Tue, Feb 7, 2012 at 10:32 AM, Ryan Watkins wrote: > Thanks Ken! > > so, I've changed the action application line to the following syntax > > data="sofia/external/externalnumber@$${domain}" /> > > our SIP provider is flowroute, so I've also tried @sip.flowroute.com and @ > flowroute.com > > looking at the logs, I'm getting the following errors when it attempts to > bridge the conference to the outbound numbers: > > 2012-02-07 10:16:23.214806 [ERR] sofia_reg.c:2170 Cannot locate any > authentication credentials to complete an authentication request for realm > '"sip.flowroute.com"' > 2012-02-07 10:16:23.214806 [DEBUG] switch_channel.c:2852 (sofia/external/ > 13605217334 at flowroute.com) Callstate Change RINGING -> HANGUP > 2012-02-07 10:16:23.214806 [NOTICE] sofia_reg.c:2190 Hangup sofia/external/ > 13605217334 at flowroute.com [CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING] > > I am, however, able to make calls from a registered extension to an > external number... > > what does [MANDATORY_IE_MISSING] mean? and any suggestions on how to get > this to work? > > Thanks again! > > Ryan > ------------------------------ > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] on behalf of Ken Rice [ > krice at freeswitch.org] > *Sent:* Saturday, February 04, 2012 10:34 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Conference Auto Outcall to External > Numbers > > Yes... Just specify a proper dialstring just like you would any other > bridge command.... > > > On 2/4/12 9:21 PM, "Ryan Watkins" > > wrote: > > Sorry for resending this, but need some assistance on this: > > Is there a way to set a Conference Auto Outcall to numbers external of the > PBX? > > Thanks! > > > *From:* freeswitch-users-bounces at lists.freeswitch.org[ > mailto:freeswitch-users-bounces at lists.freeswitch.org] > *On Behalf Of *Ryan Watkins > *Sent:* Tuesday, January 31, 2012 2:28 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Conference Auto Outcall to External > Numbers > > The call forwarding function itself works, if I set call forwarding for a > specific extension and then call that extension it will ring at the > forwarded external phone. Also, with call forwarding disabled on the > extensions the conference auto outcall does call the internal extensions. > There just seems to be a disconnect between the conference outcall to a > forwarded extension. But perhaps I should rephrase the question? > > The conference extension I?m needing to setup should auto outcall to a set > of external numbers. I figured the way to do it would be to create internal > extensions, and set the call forwarding to the external number. But perhaps > there?s a better way to approach this need? > > For example, the outcall user listing in the conference.xml is as follows: > > $${domain}?/> > > Is there a way to change the data syntax to instead direct the outbound > auto call to an external number instead of an internal user? > > Thanks! > > *From:* freeswitch-users-bounces at lists.freeswitch.org[ > mailto:freeswitch-users-bounces at lists.freeswitch.org] > *On Behalf Of *Michael Collins > *Sent:* Tuesday, January 31, 2012 1:10 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Conference Auto Outcall to External > Numbers > > What actually happens when you set the call forwarding? Console debug log > w/ SIP trace would be helpful, as would any relevant configs you've got in > place. > > -MC > > On Tue, Jan 31, 2012 at 12:46 PM, Ryan Watkins > > wrote: > > Morning/Afternoon/Evening all? > > I?ve been trying to setup a conference call that would use the auto > outcall function, and it would call external numbers (like their cell). > I?ve tried to setup extensions for the individuals, and set the call > forwarding to their external number. Although the auto outcall function > will successfully call the extensions when they are set to an internal SIP > phone, when call forwarding is enabled the conference call will not call > the external number. Any ideas on how to accomplish this? > > Thanks a lot! > > *Ryan > * > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/45363477/attachment-0001.html From Ryan at ocens.com Tue Feb 7 22:02:37 2012 From: Ryan at ocens.com (Ryan Watkins) Date: Tue, 7 Feb 2012 19:02:37 +0000 Subject: [Freeswitch-users] Conference Auto Outcall to External Numbers In-Reply-To: References: <44E5C0A9D48A3246966A4AE04692014D132E3941@CH1PRD0610MB355.namprd06.prod.outlook.com> <44E5C0A9D48A3246966A4AE04692014D132E9634@CH1PRD0610MB355.namprd06.prod.outlook.com>, Message-ID: <44E5C0A9D48A3246966A4AE04692014D132E9AA0@CH1PRD0610MB355.namprd06.prod.outlook.com> Thanks for the reply Michael I do have a gateway setup: and the SIP Status shows that the gateway is registered... Would there need to be something in place on the xml for the conference extension to use the gateway... more so then the action application lines specifying the outbound calls? Thanks again ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Michael Collins [msc at freeswitch.org] Sent: Tuesday, February 07, 2012 10:45 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference Auto Outcall to External Numbers Do you have a gateway set up for the flowroute connection? If not, you'll need to configure one because it looks like flowroute is sending back an auth challenge. If you set up a gateway then you'll be able to respond to the auth challenge and supply the proper username/password. -MC On Tue, Feb 7, 2012 at 10:32 AM, Ryan Watkins > wrote: Thanks Ken! so, I've changed the action application line to the following syntax our SIP provider is flowroute, so I've also tried @sip.flowroute.com and @flowroute.com looking at the logs, I'm getting the following errors when it attempts to bridge the conference to the outbound numbers: 2012-02-07 10:16:23.214806 [ERR] sofia_reg.c:2170 Cannot locate any authentication credentials to complete an authentication request for realm '"sip.flowroute.com"' 2012-02-07 10:16:23.214806 [DEBUG] switch_channel.c:2852 (sofia/external/13605217334@flowroute.com) Callstate Change RINGING -> HANGUP 2012-02-07 10:16:23.214806 [NOTICE] sofia_reg.c:2190 Hangup sofia/external/13605217334@flowroute.com [CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING] I am, however, able to make calls from a registered extension to an external number... what does [MANDATORY_IE_MISSING] mean? and any suggestions on how to get this to work? Thanks again! Ryan ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Ken Rice [krice at freeswitch.org] Sent: Saturday, February 04, 2012 10:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference Auto Outcall to External Numbers Yes... Just specify a proper dialstring just like you would any other bridge command.... On 2/4/12 9:21 PM, "Ryan Watkins" > wrote: Sorry for resending this, but need some assistance on this: Is there a way to set a Conference Auto Outcall to numbers external of the PBX? Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ryan Watkins Sent: Tuesday, January 31, 2012 2:28 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference Auto Outcall to External Numbers The call forwarding function itself works, if I set call forwarding for a specific extension and then call that extension it will ring at the forwarded external phone. Also, with call forwarding disabled on the extensions the conference auto outcall does call the internal extensions. There just seems to be a disconnect between the conference outcall to a forwarded extension. But perhaps I should rephrase the question? The conference extension I?m needing to setup should auto outcall to a set of external numbers. I figured the way to do it would be to create internal extensions, and set the call forwarding to the external number. But perhaps there?s a better way to approach this need? For example, the outcall user listing in the conference.xml is as follows: Is there a way to change the data syntax to instead direct the outbound auto call to an external number instead of an internal user? Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, January 31, 2012 1:10 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference Auto Outcall to External Numbers What actually happens when you set the call forwarding? Console debug log w/ SIP trace would be helpful, as would any relevant configs you've got in place. -MC On Tue, Jan 31, 2012 at 12:46 PM, Ryan Watkins > wrote: Morning/Afternoon/Evening all? I?ve been trying to setup a conference call that would use the auto outcall function, and it would call external numbers (like their cell). I?ve tried to setup extensions for the individuals, and set the call forwarding to their external number. Although the auto outcall function will successfully call the extensions when they are set to an internal SIP phone, when call forwarding is enabled the conference call will not call the external number. Any ideas on how to accomplish this? Thanks a lot! Ryan _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/785d3fea/attachment.html From sherifomran2000 at yahoo.com Tue Feb 7 22:03:17 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Tue, 7 Feb 2012 11:03:17 -0800 (PST) Subject: [Freeswitch-users] rtmp help In-Reply-To: Message-ID: <1328641397.66169.YahooMailClassic@web110803.mail.gq1.yahoo.com> Hello all, Any body knows how to adjust the rtmp? When i try to place a call and give 1000 at serverip, I can see the call in fs_cli. However, it says freeswitch at internal> 2012-02-07 19:01:32.972040 [DEBUG] rtmp.c:869 [chunk_stream=3 type=0x14 ts=12247591 stream_id=0x0] len=58 2012-02-07 19:01:32.972040 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE for makeCall 2012-02-07 19:01:32.972040 [WARNING] rtmp_sig.c:296 Unauthorized call to sip:1000 at serverip.com, client is not logged in. How can i make the client log in? I have the following in the autoconfig/rtmp.conf.xml. ? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ??? ? any help is appreciated thank you regards, Sherif Omran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/fb86746c/attachment-0001.html From msc at freeswitch.org Tue Feb 7 22:24:51 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Feb 2012 11:24:51 -0800 Subject: [Freeswitch-users] Conference Auto Outcall to External Numbers In-Reply-To: <44E5C0A9D48A3246966A4AE04692014D132E9AA0@CH1PRD0610MB355.namprd06.prod.outlook.com> References: <44E5C0A9D48A3246966A4AE04692014D132E3941@CH1PRD0610MB355.namprd06.prod.outlook.com> <44E5C0A9D48A3246966A4AE04692014D132E9634@CH1PRD0610MB355.namprd06.prod.outlook.com> <44E5C0A9D48A3246966A4AE04692014D132E9AA0@CH1PRD0610MB355.namprd06.prod.outlook.com> Message-ID: On Tue, Feb 7, 2012 at 11:02 AM, Ryan Watkins wrote: > Thanks for the reply Michael > > I do have a gateway setup: > > > > > > > > > > > > > > > and the SIP Status shows that the gateway is registered... > > Would there need to be something in place on the xml for the conference > extension to use the gateway... more so then the action application lines > specifying the outbound calls? > > Thanks again > ------------------------------ > *From:* freeswitch-users-bounces at lists.freeswitch.org [ > freeswitch-users-bounces at lists.freeswitch.org] on behalf of Michael > Collins [msc at freeswitch.org] > *Sent:* Tuesday, February 07, 2012 10:45 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Conference Auto Outcall to External > Numbers > > Do you have a gateway set up for the flowroute connection? If not, > you'll need to configure one because it looks like flowroute is sending > back an auth challenge. If you set up a gateway then you'll be able to > respond to the auth challenge and supply the proper username/password. > > -MC > > On Tue, Feb 7, 2012 at 10:32 AM, Ryan Watkins wrote: > >> Thanks Ken! >> >> so, I've changed the action application line to the following syntax >> >> > data="sofia/external/externalnumber@$${domain}" /> >> >> our SIP provider is flowroute, so I've also tried @sip.flowroute.com and >> @flowroute.com >> >> looking at the logs, I'm getting the following errors when it attempts to >> bridge the conference to the outbound numbers: >> >> 2012-02-07 10:16:23.214806 [ERR] sofia_reg.c:2170 Cannot locate any >> authentication credentials to complete an authentication request for realm >> '"sip.flowroute.com"' >> 2012-02-07 10:16:23.214806 [DEBUG] switch_channel.c:2852 (sofia/external/ >> 13605217334 at flowroute.com) Callstate Change RINGING -> HANGUP >> 2012-02-07 10:16:23.214806 [NOTICE] sofia_reg.c:2190 Hangup >> sofia/external/13605217334 at flowroute.com [CS_CONSUME_MEDIA] >> [MANDATORY_IE_MISSING] >> >> I am, however, able to make calls from a registered extension to an >> external number... >> >> what does [MANDATORY_IE_MISSING] mean? and any suggestions on how to get >> this to work? >> >> Thanks again! >> >> Ryan >> ------------------------------ >> *From:* freeswitch-users-bounces at lists.freeswitch.org [ >> freeswitch-users-bounces at lists.freeswitch.org] on behalf of Ken Rice [ >> krice at freeswitch.org] >> *Sent:* Saturday, February 04, 2012 10:34 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Conference Auto Outcall to External >> Numbers >> >> Yes... Just specify a proper dialstring just like you would any other >> bridge command.... >> >> >> On 2/4/12 9:21 PM, "Ryan Watkins" > >> wrote: >> >> Sorry for resending this, but need some assistance on this: >> >> Is there a way to set a Conference Auto Outcall to numbers external of >> the PBX? >> >> Thanks! >> >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org[ >> mailto:freeswitch-users-bounces at lists.freeswitch.org] >> *On Behalf Of *Ryan Watkins >> *Sent:* Tuesday, January 31, 2012 2:28 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Conference Auto Outcall to External >> Numbers >> >> The call forwarding function itself works, if I set call forwarding for a >> specific extension and then call that extension it will ring at the >> forwarded external phone. Also, with call forwarding disabled on the >> extensions the conference auto outcall does call the internal extensions. >> There just seems to be a disconnect between the conference outcall to a >> forwarded extension. But perhaps I should rephrase the question? >> >> The conference extension I?m needing to setup should auto outcall to a >> set of external numbers. I figured the way to do it would be to create >> internal extensions, and set the call forwarding to the external number. >> But perhaps there?s a better way to approach this need? >> >> For example, the outcall user listing in the conference.xml is as follows: >> >> > $${domain}?/> >> >> Is there a way to change the data syntax to instead direct the outbound >> auto call to an external number instead of an internal user? >> >> Thanks! >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org[ >> mailto:freeswitch-users-bounces at lists.freeswitch.org] >> *On Behalf Of *Michael Collins >> *Sent:* Tuesday, January 31, 2012 1:10 PM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] Conference Auto Outcall to External >> Numbers >> >> What actually happens when you set the call forwarding? Console debug log >> w/ SIP trace would be helpful, as would any relevant configs you've got in >> place. >> >> -MC >> >> On Tue, Jan 31, 2012 at 12:46 PM, Ryan Watkins > >> wrote: >> >> Morning/Afternoon/Evening all? >> >> I?ve been trying to setup a conference call that would use the auto >> outcall function, and it would call external numbers (like their cell). >> I?ve tried to setup extensions for the individuals, and set the call >> forwarding to their external number. Although the auto outcall function >> will successfully call the extensions when they are set to an internal SIP >> phone, when call forwarding is enabled the conference call will not call >> the external number. Any ideas on how to accomplish this? >> >> Thanks a lot! >> >> *Ryan >> * >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------ >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/9b9a3154/attachment.html From bote_radio at botecomm.com Tue Feb 7 22:46:36 2012 From: bote_radio at botecomm.com (Bote Man) Date: Tue, 7 Feb 2012 14:46:36 -0500 Subject: [Freeswitch-users] sip_invite_from_params In-Reply-To: <1907034309.1821.1328632189470.JavaMail.root@server3> References: <2048568642.1819.1328632138522.JavaMail.root@server3> <1907034309.1821.1328632189470.JavaMail.root@server3> Message-ID: <037301cce5d1$34bd5290$9e37f7b0$@com> I am an absolute newbie to FS, but from reading about dialplan regular expressions this morning I would guess that this line: is missing the required trailing "$" to end the regular expression. It should be: no? Bote > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > George Sapak > Sent: Tuesday, 07 February, 2012 11:30 > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] sip_invite_from_params > > i already have it setup, my calls go through to the right extensions > and I have done some rewitting in public.xml file, but I have having > some wierdness on the handsets with call ID, when the phone rings the > ID show correctly but when its picked up it shows it with the > ;national... sting included. > > Here is how I have done the routing and rewritte: > > > > > expression="^403XXXXXXX(.+)$"> > > > > > public.xml > > > > > > > > > Thank You, > George > > ----- Original Message ----- > From: "Brian West" > To: "FreeSWITCH Users Help" > Sent: Tuesday, February 7, 2012 8:43:54 AM > Subject: Re: [Freeswitch-users] sip_invite_from_params > > > Why? What is your goal? This should be a simple dialplan rewrite > really. > > > /b > > > > On Feb 7, 2012, at 9:27 AM, George Sapak wrote: > > > can someone give me an example in use, I have looked at the wiki and > its pretty unclear, I am trying to remove a string in an invite -> > ;phone-context=national > > INVITE sip:403XXXXXXX;phone-context=national at 10.185.16.170;user=phone > SIP/2.0 > Via: SIP/2.0/UDP > 10.185.16.169:5060;branch=z9hG4bKght6ecqg4eh7d5sluaj9cl2rr4 > From: "Caller"< sip:403YYYYYYY;phone- > context=national at 10.185.16.169;user=phone;isup-oli=00 >;tag=SDj6q1a01- > 98964 > To: < sip:403XXXXXXX;phone-context=national at 10.185.16.170;user=phone > > Call-ID: SDj6q1a01-2afa0c6426482b0cfa2885f32878c86c-o0t3g30 > CSeq: 62218 INVITE > Content-Type: application/sdp > Contact: < sip:403YYYYYYY;phone- > context=national at 10.185.16.169:5060;transport=udp > > User-Agent: Nortel SESM 14.0.6.0 > Max-Forwards: 19 > Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x- > nortel-sipvc,com.nortelnetworks.im.encryption,r eplaces,100rel,tdialog > Allow: UPDATE,REFER > x-nt-corr-id: 871f3514-29e7-1b21-a633-000e0cb7d3a0 > x-nt-location: -1 > Content-Length: 206 > Route: < sip:403XXXXXXX;phone- > context=national at 10.185.16.170:5060;user=phone;lr > > > > Thank You, > George > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > ______________________________________________________________________ > ___ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org > > ______________________________________________________________________ > ___ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From Ryan at ocens.com Tue Feb 7 22:52:00 2012 From: Ryan at ocens.com (Ryan Watkins) Date: Tue, 7 Feb 2012 19:52:00 +0000 Subject: [Freeswitch-users] Conference Auto Outcall to External Numbers In-Reply-To: References: <44E5C0A9D48A3246966A4AE04692014D132E3941@CH1PRD0610MB355.namprd06.prod.outlook.com> <44E5C0A9D48A3246966A4AE04692014D132E9634@CH1PRD0610MB355.namprd06.prod.outlook.com> <44E5C0A9D48A3246966A4AE04692014D132E9AA0@CH1PRD0610MB355.namprd06.prod.outlook.com>, Message-ID: <44E5C0A9D48A3246966A4AE04692014D132E9B0E@CH1PRD0610MB355.namprd06.prod.outlook.com> Awesome Michael, that worked! Thanks!! ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Michael Collins [msc at freeswitch.org] Sent: Tuesday, February 07, 2012 11:24 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference Auto Outcall to External Numbers On Tue, Feb 7, 2012 at 11:02 AM, Ryan Watkins > wrote: Thanks for the reply Michael I do have a gateway setup: and the SIP Status shows that the gateway is registered... Would there need to be something in place on the xml for the conference extension to use the gateway... more so then the action application lines specifying the outbound calls? Thanks again ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Michael Collins [msc at freeswitch.org] Sent: Tuesday, February 07, 2012 10:45 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference Auto Outcall to External Numbers Do you have a gateway set up for the flowroute connection? If not, you'll need to configure one because it looks like flowroute is sending back an auth challenge. If you set up a gateway then you'll be able to respond to the auth challenge and supply the proper username/password. -MC On Tue, Feb 7, 2012 at 10:32 AM, Ryan Watkins > wrote: Thanks Ken! so, I've changed the action application line to the following syntax our SIP provider is flowroute, so I've also tried @sip.flowroute.com and @flowroute.com looking at the logs, I'm getting the following errors when it attempts to bridge the conference to the outbound numbers: 2012-02-07 10:16:23.214806 [ERR] sofia_reg.c:2170 Cannot locate any authentication credentials to complete an authentication request for realm '"sip.flowroute.com"' 2012-02-07 10:16:23.214806 [DEBUG] switch_channel.c:2852 (sofia/external/13605217334@flowroute.com) Callstate Change RINGING -> HANGUP 2012-02-07 10:16:23.214806 [NOTICE] sofia_reg.c:2190 Hangup sofia/external/13605217334@flowroute.com [CS_CONSUME_MEDIA] [MANDATORY_IE_MISSING] I am, however, able to make calls from a registered extension to an external number... what does [MANDATORY_IE_MISSING] mean? and any suggestions on how to get this to work? Thanks again! Ryan ________________________________ From: freeswitch-users-bounces at lists.freeswitch.org [freeswitch-users-bounces at lists.freeswitch.org] on behalf of Ken Rice [krice at freeswitch.org] Sent: Saturday, February 04, 2012 10:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference Auto Outcall to External Numbers Yes... Just specify a proper dialstring just like you would any other bridge command.... On 2/4/12 9:21 PM, "Ryan Watkins" > wrote: Sorry for resending this, but need some assistance on this: Is there a way to set a Conference Auto Outcall to numbers external of the PBX? Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Ryan Watkins Sent: Tuesday, January 31, 2012 2:28 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference Auto Outcall to External Numbers The call forwarding function itself works, if I set call forwarding for a specific extension and then call that extension it will ring at the forwarded external phone. Also, with call forwarding disabled on the extensions the conference auto outcall does call the internal extensions. There just seems to be a disconnect between the conference outcall to a forwarded extension. But perhaps I should rephrase the question? The conference extension I?m needing to setup should auto outcall to a set of external numbers. I figured the way to do it would be to create internal extensions, and set the call forwarding to the external number. But perhaps there?s a better way to approach this need? For example, the outcall user listing in the conference.xml is as follows: Is there a way to change the data syntax to instead direct the outbound auto call to an external number instead of an internal user? Thanks! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Tuesday, January 31, 2012 1:10 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Conference Auto Outcall to External Numbers What actually happens when you set the call forwarding? Console debug log w/ SIP trace would be helpful, as would any relevant configs you've got in place. -MC On Tue, Jan 31, 2012 at 12:46 PM, Ryan Watkins > wrote: Morning/Afternoon/Evening all? I?ve been trying to setup a conference call that would use the auto outcall function, and it would call external numbers (like their cell). I?ve tried to setup extensions for the individuals, and set the call forwarding to their external number. Although the auto outcall function will successfully call the extensions when they are set to an internal SIP phone, when call forwarding is enabled the conference call will not call the external number. Any ideas on how to accomplish this? Thanks a lot! Ryan _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/ba2d2ef4/attachment-0001.html From kris at kriskinc.com Tue Feb 7 23:01:33 2012 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 7 Feb 2012 15:01:33 -0500 Subject: [Freeswitch-users] needs some advice to secure my system In-Reply-To: <1328630777.82360.YahooMailNeo@web29403.mail.ird.yahoo.com> References: <1328630777.82360.YahooMailNeo@web29403.mail.ird.yahoo.com> Message-ID: Bob, I suggest you reply to existing e-mails rather than always start new messages (and threads). It has been virtually impossible to follow the various threads you've been participating in (including this one). If I had seen you already made the same suggestion I certainly wouldn't have replied with the same thing (although I also included information about inbound-acl). Welcome! On Tue, Feb 7, 2012 at 11:06 AM, Bob Smith wrote: > > > Kristian, > > Huh ? I was the one who suggested the firewall filtering and OpenVPN in the first place, no need to teach granny how to suck eggs.? ;-) > > Bob > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Kristian Kielhofner From grsingh750 at gmail.com Tue Feb 7 23:02:44 2012 From: grsingh750 at gmail.com (guru singh) Date: Wed, 8 Feb 2012 01:32:44 +0530 Subject: [Freeswitch-users] needs some advice to secure my system In-Reply-To: References: Message-ID: Hi Philippe, Have a look at http://wiki.freeswitch.org/wiki/Fail2ban On Tue, Feb 7, 2012 at 8:12 PM, Philippe Le Toquin wrote: > Hello, > > Sorry to ask like that but could someone points me to some site that > explains exactly what I need to open towards the internet so that > my FS server is working while limiting its visibility? > > since 1st of February I have an IP that continually sends me SIP Register > request at a rate of 70KB/s. I have complained to my internet > provider but they refuse to help saying that the problem is on my side. I > also logged a complain to the provider on that IP and am waiting on that. > > At the moment on my firewall I opened port 5060 and 5080 (well now I blocked > as well that IP) but I want to know if both are really needed or if I could > block one of them > ?or may be limit the port to some IP. > > Any help/links will be gladly received > > thanks > > /Philippe > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > 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 Feb 7 23:04:38 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Feb 2012 12:04:38 -0800 Subject: [Freeswitch-users] sip_invite_from_params In-Reply-To: <037301cce5d1$34bd5290$9e37f7b0$@com> References: <2048568642.1819.1328632138522.JavaMail.root@server3> <1907034309.1821.1328632189470.JavaMail.root@server3> <037301cce5d1$34bd5290$9e37f7b0$@com> Message-ID: Technically that is not a big issue, however I will explain it for the sake of discussion. The ^ and $ are "anchors" for the beginning and end of the string. Consider these patterns: \d (match a single digit) \d+ (match one or more digits) ^\d+ (match a string that begins with one or more digits) \d+$ (match a string that ends with one or more digits) ^\d+$ (match a string that contains only one or more digits) So, the pattern ^(\d+) literally means: "Match a string that begins with one or more digits and capture those digits into the variable $1." The only issue with not having a $ at the end of the pattern is that there could be other stuff in the string after the digits. For example, these strings would all match the pattern "^(\d+)": 1x 43abc 12345abcd 998877LA_LA_LA_WHATEVER 18005551212!hoohah If you added the $ at the end of the pattern then none of the above strings would match. Hope that makes sense.... -MC On Tue, Feb 7, 2012 at 11:46 AM, Bote Man wrote: > I am an absolute newbie to FS, but from reading about dialplan regular > expressions this morning I would guess that this line: > > > > is missing the required trailing "$" to end the regular expression. > > It should be: > > > no? > > Bote > > > > -----Original Message----- > > From: freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of > > George Sapak > > Sent: Tuesday, 07 February, 2012 11:30 > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] sip_invite_from_params > > > > i already have it setup, my calls go through to the right extensions > > and I have done some rewitting in public.xml file, but I have having > > some wierdness on the handsets with call ID, when the phone rings the > > ID show correctly but when its picked up it shows it with the > > ;national... sting included. > > > > Here is how I have done the routing and rewritte: > > > > > > > > > > > expression="^403XXXXXXX(.+)$"> > > > > > > > > > > public.xml > > > > > > > > > > > > > > > > > > Thank You, > > George > > > > ----- Original Message ----- > > From: "Brian West" > > To: "FreeSWITCH Users Help" > > Sent: Tuesday, February 7, 2012 8:43:54 AM > > Subject: Re: [Freeswitch-users] sip_invite_from_params > > > > > > Why? What is your goal? This should be a simple dialplan rewrite > > really. > > > > > > /b > > > > > > > > On Feb 7, 2012, at 9:27 AM, George Sapak wrote: > > > > > > can someone give me an example in use, I have looked at the wiki and > > its pretty unclear, I am trying to remove a string in an invite -> > > ;phone-context=national > > > > INVITE sip:403XXXXXXX;phone-context=national at 10.185.16.170;user=phone > > SIP/2.0 > > Via: SIP/2.0/UDP > > 10.185.16.169:5060;branch=z9hG4bKght6ecqg4eh7d5sluaj9cl2rr4 > > From: "Caller"< sip:403YYYYYYY;phone- > > context=national at 10.185.16.169;user=phone;isup-oli=00 >;tag=SDj6q1a01- > > 98964 > > To: < sip:403XXXXXXX;phone-context=national at 10.185.16.170;user=phone > > > Call-ID: SDj6q1a01-2afa0c6426482b0cfa2885f32878c86c-o0t3g30 > > CSeq: 62218 INVITE > > Content-Type: application/sdp > > Contact: < sip:403YYYYYYY;phone- > > context=national at 10.185.16.169:5060;transport=udp > > > User-Agent: Nortel SESM 14.0.6.0 > > Max-Forwards: 19 > > Supported: com.nortelnetworks.firewall,p-3rdpartycontrol,nosec,join,x- > > nortel-sipvc,com.nortelnetworks.im.encryption,r eplaces,100rel,tdialog > > Allow: UPDATE,REFER > > x-nt-corr-id: 871f3514-29e7-1b21-a633-000e0cb7d3a0 > > x-nt-location: -1 > > Content-Length: 206 > > Route: < sip:403XXXXXXX;phone- > > context=national at 10.185.16.170:5060;user=phone;lr > > > > > > > Thank You, > > George > > > > -- > > Brian West > > FreeSWITCH Solutions, LLC > > Phone: +1 (918) 420-9266 > > Fax: +1 (918) 420-9267 > > brian at freeswitch.org > > http://www.freeswitch.org > > > > ______________________________________________________________________ > > ___ > > Professional FreeSWITCH Consulting Services: > > 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 > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > > > ______________________________________________________________________ > > ___ > > Professional FreeSWITCH Consulting Services: > > 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 > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/f7c7d496/attachment.html From msc at freeswitch.org Tue Feb 7 23:17:25 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Feb 2012 12:17:25 -0800 Subject: [Freeswitch-users] FreeSWITCH Conference Call Tomorrow Message-ID: Hello all! Just a reminder that we are having Travis Cross come back and talk to us about git again. I think we've all had some time to digest the information from his first presentation and hopefully we have some intelligent questions for him. I'm wondering if he can show us an example of git bisect. :) Here's the agenda page: http://wiki.freeswitch.org/wiki/FS_weekly_2012_02_08 FYI, I am trying to get caught up on the release notes so I've found a few new FreeSWITCH things that I'd like to have you all help me get documented. :) Thanks, Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/1b8d44d9/attachment.html From rsaavedra at ecogizmos.com Tue Feb 7 23:18:32 2012 From: rsaavedra at ecogizmos.com (rsaavedra at ecogizmos.com) Date: Tue, 7 Feb 2012 15:18:32 -0500 Subject: [Freeswitch-users] rtmp help In-Reply-To: <1328641397.66169.YahooMailClassic@web110803.mail.gq1.yahoo.com> References: <1328641397.66169.YahooMailClassic@web110803.mail.gq1.yahoo.com> Message-ID: <6c3a924280e0b9a7ddf9483deb200e91.squirrel@emailmg.globat.com> Hello, you have to permit anonymous calls on the external profile. Bye, Ricardo Saavedra > Hello all, > > Any body knows how to adjust the rtmp? > > When i try to place a call and give 1000 at serverip, I can see the call in > fs_cli. However, it says > > freeswitch at internal> 2012-02-07 19:01:32.972040 [DEBUG] rtmp.c:869 > [chunk_stream=3 type=0x14 ts=12247591 stream_id=0x0] len=58 > 2012-02-07 19:01:32.972040 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE for > makeCall > 2012-02-07 19:01:32.972040 [WARNING] rtmp_sig.c:296 Unauthorized call to > sip:1000 at serverip.com, client is not logged in. > > How can i make the client log in? > I have the following in the autoconfig/rtmp.conf.xml. > > > > ? > ??? > ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? > ??? > ? > > > any help is appreciated > > thank you > > regards, > Sherif Omran > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From sherifomran2000 at yahoo.com Wed Feb 8 00:28:14 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Tue, 7 Feb 2012 13:28:14 -0800 (PST) Subject: [Freeswitch-users] rtmp help In-Reply-To: <6c3a924280e0b9a7ddf9483deb200e91.squirrel@emailmg.globat.com> Message-ID: <1328650094.35674.YahooMailClassic@web110805.mail.gq1.yahoo.com> What do you mean by external profile? I have bluebox installed and have 3 profiles (1 authenticated, 1 not authenticated, 1 for nat behind devices and authenticated): sofia status sipinterface_3??? profile??? ????????? port 5080??? RUNNING (0) sipinterface_1??? profile??? ???????????????? 5060??? RUNNING (0) sipinterface_2??? profile??? ???????????????? 5070??? RUNNING (0) Probably you mean sipinterface_3 but it uses port 5080. how should i act? thank you --- On Tue, 2/7/12, rsaavedra at ecogizmos.com wrote: From: rsaavedra at ecogizmos.com Subject: Re: [Freeswitch-users] rtmp help To: "FreeSWITCH Users Help" Date: Tuesday, February 7, 2012, 10:18 PM Hello, you have to permit anonymous calls on the external profile. Bye, Ricardo Saavedra > Hello all, > > Any body knows how to adjust the rtmp? > > When i try to place a call and give 1000 at serverip, I can see the call in > fs_cli. However, it says > > freeswitch at internal> 2012-02-07 19:01:32.972040 [DEBUG] rtmp.c:869 > [chunk_stream=3 type=0x14 ts=12247591 stream_id=0x0] len=58 > 2012-02-07 19:01:32.972040 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE for > makeCall > 2012-02-07 19:01:32.972040 [WARNING] rtmp_sig.c:296 Unauthorized call to > sip:1000 at serverip.com, client is not logged in. > > How can i make the client log in? > I have the following in the autoconfig/rtmp.conf.xml. > > > > ? > ??? > ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? > ??? > ? > > > any help is appreciated > > thank you > > regards, > Sherif Omran > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/2a21fa06/attachment.html From sherifomran2000 at yahoo.com Wed Feb 8 00:47:48 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Tue, 7 Feb 2012 13:47:48 -0800 (PST) Subject: [Freeswitch-users] rtmp help In-Reply-To: <6c3a924280e0b9a7ddf9483deb200e91.squirrel@emailmg.globat.com> Message-ID: <1328651268.85219.YahooMailClassic@web110816.mail.gq1.yahoo.com> Ok, now i get the following ?[NO_ROUTE_DESTINATION] could you please help? thank you --- On Tue, 2/7/12, rsaavedra at ecogizmos.com wrote: From: rsaavedra at ecogizmos.com Subject: Re: [Freeswitch-users] rtmp help To: "FreeSWITCH Users Help" Date: Tuesday, February 7, 2012, 10:18 PM Hello, you have to permit anonymous calls on the external profile. Bye, Ricardo Saavedra > Hello all, > > Any body knows how to adjust the rtmp? > > When i try to place a call and give 1000 at serverip, I can see the call in > fs_cli. However, it says > > freeswitch at internal> 2012-02-07 19:01:32.972040 [DEBUG] rtmp.c:869 > [chunk_stream=3 type=0x14 ts=12247591 stream_id=0x0] len=58 > 2012-02-07 19:01:32.972040 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE for > makeCall > 2012-02-07 19:01:32.972040 [WARNING] rtmp_sig.c:296 Unauthorized call to > sip:1000 at serverip.com, client is not logged in. > > How can i make the client log in? > I have the following in the autoconfig/rtmp.conf.xml. > > > > ? > ??? > ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? ??? > ??? ??? > ??? > ? > > > any help is appreciated > > thank you > > regards, > Sherif Omran > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/3fd782cd/attachment.html From msc at freeswitch.org Wed Feb 8 01:55:50 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Feb 2012 14:55:50 -0800 Subject: [Freeswitch-users] rtmp help In-Reply-To: <1328651268.85219.YahooMailClassic@web110816.mail.gq1.yahoo.com> References: <6c3a924280e0b9a7ddf9483deb200e91.squirrel@emailmg.globat.com> <1328651268.85219.YahooMailClassic@web110816.mail.gq1.yahoo.com> Message-ID: This could be a dialplan issue. You will need to supply a console debug log. Use pastebin.freeswitch.org -MC On Tue, Feb 7, 2012 at 1:47 PM, Sherif Omran wrote: > Ok, now i get the following > > [NO_ROUTE_DESTINATION] > > > could you please help? > > > thank you > > > > --- On *Tue, 2/7/12, rsaavedra at ecogizmos.com *wrote: > > > From: rsaavedra at ecogizmos.com > Subject: Re: [Freeswitch-users] rtmp help > To: "FreeSWITCH Users Help" > Date: Tuesday, February 7, 2012, 10:18 PM > > Hello, > you have to permit anonymous calls on the external profile. > > Bye, > > Ricardo Saavedra > > > > Hello all, > > > > Any body knows how to adjust the rtmp? > > > > When i try to place a call and give 1000 at serverip, I can see the call in > > fs_cli. However, it says > > > > freeswitch at internal> 2012-02-07 19:01:32.972040 [DEBUG] rtmp.c:869 > > [chunk_stream=3 type=0x14 ts=12247591 stream_id=0x0] len=58 > > 2012-02-07 19:01:32.972040 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE > for > > makeCall > > 2012-02-07 19:01:32.972040 [WARNING] rtmp_sig.c:296 Unauthorized call to > > sip:1000 at serverip.com , client > is not logged in. > > > > How can i make the client log in? > > I have the following in the autoconfig/rtmp.conf.xml. > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > any help is appreciated > > > > thank you > > > > regards, > > Sherif Omran > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > 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 > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/8c90f21c/attachment-0001.html From cjbujold at accra.ca Wed Feb 8 02:46:07 2012 From: cjbujold at accra.ca (Charles Bujold) Date: Tue, 7 Feb 2012 19:46:07 -0400 Subject: [Freeswitch-users] error on outgoing calls Message-ID: <000601cce5f2$aa040be0$fe0c23a0$@accra.ca> Freeswitch was working properly and today our outgoing calls started to get rejected. Here is the log of an outgoing call. Can somebody point to the error and how to fix it. Thanks CJB 2012-02-07 19:33:13.052955 [DEBUG] switch_core_session.c:875 Send signal sofia/external/150XXXXXXX [BREAK] 2012-02-07 19:33:13.052955 [DEBUG] switch_core_session.c:875 Send signal sofia/external/150XXXXXXX [BREAK] 2012-02-07 19:33:13.052955 [DEBUG] sofia.c:5508 Channel sofia/external/1506XXXXXXX entering state [terminated][403] 2012-02-07 19:33:13.052955 [DEBUG] switch_channel.c:2852 (sofia/external/1506XXXXXX) Callstate Change RINGING -> HANGUP 2012-02-07 19:33:13.052955 [NOTICE] sofia.c:6272 Hangup sofia/external/1506XXXXXXX [CS_CONSUME_MEDIA] [CALL_REJECTED] 2012-02-07 19:33:13.052955 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 21 [CALL_REJECTED] 2012-02-07 19:33:13.052955 [DEBUG] switch_channel.c:2875 Send signal sofia/external/150XXXXXXXX [KILL] 2012-02-07 19:33:13.052955 [INFO] mod_dptools.c:2916 Originate Failed. Cause: CALL_REJECTED -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/2e3b7f68/attachment.html From asrivas at gmail.com Wed Feb 8 02:53:12 2012 From: asrivas at gmail.com (Anurag Srivastava) Date: Tue, 7 Feb 2012 15:53:12 -0800 Subject: [Freeswitch-users] external_sip_ip not refreshed after ip address change. In-Reply-To: References: Message-ID: I am not running freeswitch with nonat. Just to reiterate my original problem, external_sip_ip is reloaded only once at startup even though stun is specified which is the cause of hte problem. external_rtp_ip is updated correctly with stun. Hi Brian, > Thanks for your help. From what I see the parameter auto-restart=true > will only restart profiles if the local interface ip address on the machine > changes. Will it also restart if my public ip address changes? My machine > is behind a NAT with port forwarding. I am trying to restart profiles if my > public ip address changes. > Thanks > > Can somebody help me with this? >> >> >> I have a question about nat behavior in freeswitch. Basically my external >>> calls to freeswitch are getting disconnected after 30 seconds of two way >>> audio when my external ip address changes. I have dhcp from my ISP and am >>> using external_sip_ip and external_rtp_ip as stun:. When my IP >>> changes I see that external_sip_ip does not get refreshed but >>> external_rtp_ip does. I am not allowed to enable upnp/nat-pmp on my router. >>> Apparently it is a known issue that external_sip_ip is read just >>> at load time and not refreshed even if it is specified in stun format >>> Is there a fix to this problem? >>> >>> There is always the option of restarting profile when ddclient notes an >>> ip change. Is there something inbuilt into FS. It is already finding that >>> ip address has changed as reflected in external_rtp_ip which does use stun >>> and gets the right ip address. >>> >> >> >> >> -- >> Regards >> Anurag >> > > > > -- > Regards > Anurag > -- Regards Anurag -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/d22c644e/attachment.html From msc at freeswitch.org Wed Feb 8 03:23:21 2012 From: msc at freeswitch.org (Michael Collins) Date: Tue, 7 Feb 2012 16:23:21 -0800 Subject: [Freeswitch-users] error on outgoing calls In-Reply-To: <000601cce5f2$aa040be0$fe0c23a0$@accra.ca> References: <000601cce5f2$aa040be0$fe0c23a0$@accra.ca> Message-ID: The server on the far end is rejecting the call but you don't know why. I would get a sip trace and see if the reject message has any information, like "bad number" or something like that. You may also need to call the provider and find out what they don't like about what you're sending. -MC On Tue, Feb 7, 2012 at 3:46 PM, Charles Bujold wrote: > Freeswitch was working properly and today our outgoing calls started to > get rejected. Here is the log of an outgoing call. Can somebody point to > the error and how to fix it.**** > > ** ** > > Thanks**** > > ** ** > > CJB**** > > ** ** > > ** ** > > 2012-02-07 19:33:13.052955 [DEBUG] switch_core_session.c:875 Send signal > sofia/external/150XXXXXXX [BREAK] > 2012-02-07 19:33:13.052955 [DEBUG] switch_core_session.c:875 Send signal > sofia/external/150XXXXXXX [BREAK] > 2012-02-07 19:33:13.052955 [DEBUG] sofia.c:5508 Channel > sofia/external/1506XXXXXXX entering state [terminated][403] > 2012-02-07 19:33:13.052955 [DEBUG] switch_channel.c:2852 > (sofia/external/1506XXXXXX) Callstate Change RINGING -> HANGUP > 2012-02-07 19:33:13.052955 [NOTICE] sofia.c:6272 Hangup > sofia/external/1506XXXXXXX [CS_CONSUME_MEDIA] [CALL_REJECTED] > 2012-02-07 19:33:13.052955 [DEBUG] switch_ivr_originate.c:3364 Originate > Resulted in Error Cause: 21 [CALL_REJECTED] > 2012-02-07 19:33:13.052955 [DEBUG] switch_channel.c:2875 Send signal > sofia/external/150XXXXXXXX [KILL] > 2012-02-07 19:33:13.052955 [INFO] mod_dptools.c:2916 Originate Failed. > Cause: CALL_REJECTED**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/6fcff571/attachment.html From mayamatakeshi at gmail.com Wed Feb 8 05:17:49 2012 From: mayamatakeshi at gmail.com (mayamatakeshi) Date: Wed, 8 Feb 2012 11:17:49 +0900 Subject: [Freeswitch-users] Unexpected BYE's after 100 calls In-Reply-To: References: Message-ID: On Tue, Feb 7, 2012 at 7:42 PM, Boudewyn Ligthart wrote: > ** ** > > Hi,**** > > ** ** > > We have been testing with Freeswitch (hacked-20120106T151802Z). There are > two profiles, allowing calls > through using ACL, the core is running with ODBC on MySQL, all running on > ubuntu, virtualized in XEN. **** > > ** ** > > When testing with SIPP and standard uac towards Freeswitch there are no > issue?s with the SIP messaging,**** > > SIPP on another machine acts as uas. So call flows SIPP ? Freeswitch ?SIPP. > **** > > ** ** > > When testing with SIPP (using same routing) streaming audio after about > 100 calls (5 seconds in) there are unexpected **** > > BYE?s coming from Freeswitch (I have even lowered the CPS count). **** > > ** ** > > I do not see any error?s in the logs, just the BYE?s. What could be > causing this issue? > FS might tell you something in the header Reason in the BYE. On a normally terminated call it would be like this: Reason: Q.850;cause=16;text="NORMAL_CLEARING" Other than that, set loglevel to "debug" to check what FS is deciding to do. regards, Takeshi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/5c79789d/attachment.html From admin at blindi.net Wed Feb 8 05:39:22 2012 From: admin at blindi.net (Thomas Hoellriegel) Date: Wed, 8 Feb 2012 03:39:22 +0100 (CET) Subject: [Freeswitch-users] Astconf2fsconf a software or joke? In-Reply-To: References: <1336366836.3502.7.camel@localhost.localdomain> Message-ID: Hi shouldbe I don.t find a downloadlink from: http://wiki.freeswitch.org/wiki/Bounty_challenged http://wiki.freeswitch.org/wiki/Astconf2fsconf http://wiki.freeswitch.org/wiki/Rosetta_Stone I see a description, but no link to download these software. why? Do you have a downloadlink for me please? thnaks. --------------- 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 krice at freeswitch.org Wed Feb 8 06:23:25 2012 From: krice at freeswitch.org (Ken Rice) Date: Tue, 07 Feb 2012 21:23:25 -0600 Subject: [Freeswitch-users] Astconf2fsconf a software or joke? In-Reply-To: Message-ID: It doesn't exist... There is no such thing... Someone probably created this page ages ago intending to add something... On 2/7/12 8:39 PM, "Thomas Hoellriegel" wrote: > > Hi shouldbe > I don.t find a downloadlink from: > http://wiki.freeswitch.org/wiki/Bounty_challenged > http://wiki.freeswitch.org/wiki/Astconf2fsconf > http://wiki.freeswitch.org/wiki/Rosetta_Stone > I see a description, but no link to download these software. why? > Do you have a downloadlink for me please? > > thnaks. > > --------------- > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From b2m at a-cti.com Wed Feb 8 09:37:54 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Wed, 8 Feb 2012 12:07:54 +0530 Subject: [Freeswitch-users] How to make calls from Browser? Message-ID: Hi all, I am trying to make calls from browser(chrome, safari, FireFox, Opera, IE) is there any known open source tool available already? Thanks, Bala -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/62f5a253/attachment.html From davidwaf at gmail.com Wed Feb 8 10:47:10 2012 From: davidwaf at gmail.com (David Wafula) Date: Wed, 8 Feb 2012 09:47:10 +0200 Subject: [Freeswitch-users] How to make calls from Browser? In-Reply-To: References: Message-ID: The Flex client: http://wiki.freeswitch.org/wiki/Mod_rtmp or build a java applet with http://peers.sourceforge.net/ On Wed, Feb 8, 2012 at 8:37 AM, Balamurugan Mahendran wrote: > Hi all, > > I am trying to make calls from?browser(chrome, safari, FireFox, Opera, IE) > is there any known open source tool available already? > > Thanks, > Bala > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- David Wafula From spautz2 at telefaks.biz Wed Feb 8 11:20:18 2012 From: spautz2 at telefaks.biz (David) Date: Wed, 08 Feb 2012 09:20:18 +0100 Subject: [Freeswitch-users] Freeswitch opened multiple outbound channels for one call with different uuid Message-ID: <4F323042.8080603@telefaks.biz> Hi, i wonder because when i do a normal call e.g. from directory 220 to directory 240 (all internal), then freeswitch opened 5 channels. One for that inbound from 220, second, a channel for outbound to 240. Thats ok, but there was 3 addition channels (all with direction=outbound and dest=240 and all the same data (shown in database table core_channels)). Only the uuid is different. The problem is that, if the callee accept this invite, three hangup-events get fired by freeswitch. How can i avoid this creation of three additional channels? What are the reason for this channels? Thanks From b2m at a-cti.com Wed Feb 8 14:52:42 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Wed, 8 Feb 2012 17:22:42 +0530 Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: References: <005801cce12e$05358300$0fa08900$@net.au> <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> <4F2A9482.1020100@earthspike.net> <4F2B8B51.2080506@earthspike.net> <4F2C2EFA.9030105@earthspike.net> <4F3137A7.2040908@earthspike.net> Message-ID: I think found the log, not sure how to proceed. Feb 8 11:51:03 ip-10-252-39-229 kernel: [7140198.319410] sendmail[15835]: segfault at 7fff89aa2ed0 ip 00007f75670e9a8a sp 00007fff89aa2ea0 error 6 in sendmail[7f7567056000+cd000] Thanks, Bala On Tue, Feb 7, 2012 at 10:23 PM, Balamurugan Mahendran wrote: > Is there anyway for more logs? Thanks for your help!! > > --Bala > > > On Tue, Feb 7, 2012 at 8:34 PM, Balamurugan Mahendran wrote: > >> I did, also I have no issue sending email outside FS. >> >> Thanks, >> Bala >> >> On Tue, Feb 7, 2012 at 8:31 PM, Balamurugan Mahendran < >> balamurugan at adaptavant.com> wrote: >> >>> I did, also I have no issue sending email outside FS. >>> >>> Thanks, >>> Bala >>> >>> On Tue, Feb 7, 2012 at 8:09 PM, John wrote: >>> >>>> I updated the wiki mod_voicemail page a few days ago with some >>>> instructions for debugging email from freeswitch. Have you tried those? >>>> >>>> >>>> On 07/02/12 13:47, Balamurugan Mahendran wrote: >>>> >>>> All, >>>> >>>> I am having the same issue, its not sending email(extension --> lua >>>> script) getting *"Segmentation fault"* >>>> >>>> xml : >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>> value="$${outbound_caller_name}"/> >>>> >>> value="$${outbound_caller_id}"/> >>>> >>>> >>>> >>>> >>>> >>>> dialplan : >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> Lua : >>>> >>>> caller=503; >>>> freeswitch.consoleLog("info","From :"..caller); >>>> session:set_tts_parms("flite", "slt"); >>>> session:speak("Welcome To Voice Mail !. You Can Leave Your Message >>>> Here."); >>>> path="/usr/local/freeswitch/recordings/"; >>>> prompt=caller..".mp3"; >>>> recpath=path..prompt; >>>> freeswitch.consoleLog("info","record path="..recpath); >>>> session:recordFile(recpath,30,10,10); >>>> session:speak("Thank you."); >>>> >>>> freeswitch.consoleLog("info","testing"); >>>> freeswitch.email("b2m at a-cti.com", >>>> "saraswathi.devaraj at a-cti.com", >>>> "subject: Voicemail from 801\n", >>>> "Hello,\n\nYou've got a voicemail, click the >>>> attachment to listen to it.", >>>> "/usr/local/freeswitch/recordings/503.mp3", >>>> "", >>>> ""); >>>> freeswitch.consoleLog("info","hai"); >>>> >>>> >>>> >>>> switchconf : >>>> >>>> >>>> >>>> >>>> >>>> >>>> Thanks for your help!! >>>> >>>> Thanks, >>>> Bala >>>> >>>> >>>> >>>> On Sat, Feb 4, 2012 at 12:31 AM, John wrote: >>>> >>>>> On 03/02/12 18:00, Thomas Hoellriegel wrote: >>>>> > >>>>> > Its works fine!! >>>>> Good news! >>>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> 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 >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/0ecc68d1/attachment-0001.html From b2m at a-cti.com Wed Feb 8 15:04:09 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Wed, 8 Feb 2012 17:34:09 +0530 Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: References: <005801cce12e$05358300$0fa08900$@net.au> <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> <4F2A9482.1020100@earthspike.net> <4F2B8B51.2080506@earthspike.net> <4F2C2EFA.9030105@earthspike.net> <4F3137A7.2040908@earthspike.net> Message-ID: Also looks like FS creates file in /tmp location, but I cannot find this file in this location. fs_cli: 2012-02-08 12:01:26.506096 [DEBUG] switch_utils.c:767 Emailed file [/tmp/mail.13287024861013] to [b2m at a-cti.com] syslog : Feb 8 12:01:26 ip-10-252-39-229 kernel: [7140816.707307] show_signal_msg: 3 callbacks suppressed Feb 8 12:01:26 ip-10-252-39-229 kernel: [7140816.707313] sendmail[1335]: segfault at 7fff4a735100 ip 00007fb706a67a8a sp 00007fff4a7350d0 error 6 in sendmail[7fb7069d4000+cd000] Thanks, Bala On Wed, Feb 8, 2012 at 5:22 PM, Balamurugan Mahendran wrote: > I think found the log, not sure how to proceed. > > Feb 8 11:51:03 ip-10-252-39-229 kernel: [7140198.319410] sendmail[15835]: > segfault at 7fff89aa2ed0 ip 00007f75670e9a8a sp 00007fff89aa2ea0 error 6 in > sendmail[7f7567056000+cd000] > > Thanks, > Bala > > > On Tue, Feb 7, 2012 at 10:23 PM, Balamurugan Mahendran wrote: > >> Is there anyway for more logs? Thanks for your help!! >> >> --Bala >> >> >> On Tue, Feb 7, 2012 at 8:34 PM, Balamurugan Mahendran wrote: >> >>> I did, also I have no issue sending email outside FS. >>> >>> Thanks, >>> Bala >>> >>> On Tue, Feb 7, 2012 at 8:31 PM, Balamurugan Mahendran < >>> balamurugan at adaptavant.com> wrote: >>> >>>> I did, also I have no issue sending email outside FS. >>>> >>>> Thanks, >>>> Bala >>>> >>>> On Tue, Feb 7, 2012 at 8:09 PM, John wrote: >>>> >>>>> I updated the wiki mod_voicemail page a few days ago with some >>>>> instructions for debugging email from freeswitch. Have you tried those? >>>>> >>>>> >>>>> On 07/02/12 13:47, Balamurugan Mahendran wrote: >>>>> >>>>> All, >>>>> >>>>> I am having the same issue, its not sending email(extension --> lua >>>>> script) getting *"Segmentation fault"* >>>>> >>>>> xml : >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> value="domestic,international,local"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> value="$${outbound_caller_name}"/> >>>>> >>>> value="$${outbound_caller_id}"/> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> dialplan : >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Lua : >>>>> >>>>> caller=503; >>>>> freeswitch.consoleLog("info","From :"..caller); >>>>> session:set_tts_parms("flite", "slt"); >>>>> session:speak("Welcome To Voice Mail !. You Can Leave Your Message >>>>> Here."); >>>>> path="/usr/local/freeswitch/recordings/"; >>>>> prompt=caller..".mp3"; >>>>> recpath=path..prompt; >>>>> freeswitch.consoleLog("info","record path="..recpath); >>>>> session:recordFile(recpath,30,10,10); >>>>> session:speak("Thank you."); >>>>> >>>>> freeswitch.consoleLog("info","testing"); >>>>> freeswitch.email("b2m at a-cti.com", >>>>> "saraswathi.devaraj at a-cti.com", >>>>> "subject: Voicemail from 801\n", >>>>> "Hello,\n\nYou've got a voicemail, click the >>>>> attachment to listen to it.", >>>>> "/usr/local/freeswitch/recordings/503.mp3", >>>>> "", >>>>> ""); >>>>> freeswitch.consoleLog("info","hai"); >>>>> >>>>> >>>>> >>>>> switchconf : >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> Thanks for your help!! >>>>> >>>>> Thanks, >>>>> Bala >>>>> >>>>> >>>>> >>>>> On Sat, Feb 4, 2012 at 12:31 AM, John wrote: >>>>> >>>>>> On 03/02/12 18:00, Thomas Hoellriegel wrote: >>>>>> > >>>>>> > Its works fine!! >>>>>> Good news! >>>>>> >>>>> >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> 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 >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE: >>>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>>> http://www.freeswitch.org >>>>> >>>>> >>>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/4287e31f/attachment.html From gerald.weber at besharp.at Wed Feb 8 15:08:11 2012 From: gerald.weber at besharp.at (Gerald Weber) Date: Wed, 8 Feb 2012 12:08:11 +0000 Subject: [Freeswitch-users] Hangup Cause in LUA Hangup Hook Message-ID: Hello, is there a way to get the HangupCause in a HangupHook in LUA ? following commands return nil: s:getVariable("hangup_cause") s:getVariable("hangup_cause_q850") s:hangupCause() returns "NONE" I know I can set api_hangup_hook and use the env object to get the channel vars, but i think that's a litte overkill to use 2 scripts for 1 Call. Thx gw -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/46f45ec7/attachment.html From spautz2 at telefaks.biz Wed Feb 8 15:45:56 2012 From: spautz2 at telefaks.biz (David) Date: Wed, 08 Feb 2012 13:45:56 +0100 Subject: [Freeswitch-users] Freeswitch opened multiple outbound channels for one call with different uuid In-Reply-To: <4F323042.8080603@telefaks.biz> References: <4F323042.8080603@telefaks.biz> Message-ID: <4F326E84.6030302@telefaks.biz> Am 08.02.2012 09:20, schrieb David: > Hi, > > i wonder because when i do a normal call e.g. from directory 220 to > directory 240 (all internal), then freeswitch opened 5 channels. > > One for that inbound from 220, second, a channel for outbound to 240. > Thats ok, but there was 3 addition channels (all with direction=outbound > and dest=240 and all the same data (shown in database table > core_channels)). Only the uuid is different. > > The problem is that, if the callee accept this invite, three > hangup-events get fired by freeswitch. > > How can i avoid this creation of three additional channels? What are the > reason for this channels? > > > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org It was found that freeswitch performs multiple INVITES. But why? 2012-02-08 13:42:22.464450 [NOTICE] switch_channel.c:926 New Channel sofia/internal/220 at entwick1 [ac3dbe28-12df-42ff-ba63-7939d3fe5c77] 2012-02-08 13:42:22.464450 [INFO] mod_dialplan_xml.c:485 Processing User 220 <220>->240 in context default nta: timer J fired, terminate 200 response nta_incoming_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/3 free nta: timer K fired, terminate OPTIONS (24015567) nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/5 term, 1/5 free nta: timer K fired, terminate OPTIONS (24015565) nta: timer K fired, terminate OPTIONS (24015566) nta_outgoing_timer: 0/0 resent, 0/0 tout, 2/4 term, 2/4 free nta: timer K fired, terminate OPTIONS (24015564) nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/2 term, 1/2 free nta: timer K fired, terminate OPTIONS (24015568) nta_outgoing_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/1 free 2012-02-08 13:42:24.224610 [NOTICE] mod_sofia.c:2506 Ring-Ready sofia/internal/220 at entwick1! nta: sent 180 Ringing for INVITE (2) nua(0xb6ea4088): call state changed: received -> early 2012-02-08 13:42:24.224610 [NOTICE] mod_dptools.c:891 Ring Ready sofia/internal/220 at entwick1! 2012-02-08 13:42:25.364453 [NOTICE] switch_channel.c:926 New Channel sofia/internal/sip:240 at 192.168.1.11:5060 [cddc3d8d-e30d-4985-be48-5702bc7f32b0] 2012-02-08 13:42:25.364453 [NOTICE] switch_channel.c:926 New Channel sofia/internal/sip:240 at 192.168.1.11:5060 [368c5156-f353-4141-9def-8ccd1df64579] 2012-02-08 13:42:25.364453 [NOTICE] switch_channel.c:926 New Channel sofia/internal/sip:240 at 192.168.1.11:5060 [1f955ed3-62a7-4c37-8016-39655a992017] nua(0xb6427168): adding session usage 2012-02-08 13:42:25.364453 [NOTICE] switch_channel.c:926 New Channel sofia/internal/sip:240 at 192.168.1.11:5060 [b6f6c2a8-9098-4528-a38e-8a463e38b12c] nta: sent INVITE (24016408) to */192.168.1.11:5060 nua(0xb6427168): call state changed: init -> calling, sent offer nua(0xb689fe28): adding session usage nta: sent INVITE (24016408) to */192.168.1.11:5060 nua(0xb689fe28): call state changed: init -> calling, sent offer nua(0xb68018d8): adding session usage nta: sent INVITE (24016408) to */192.168.1.11:5060 nua(0xb68018d8): call state changed: init -> calling, sent offer nua(0x8ed3d50): adding session usage nta: sent INVITE (24016408) to */192.168.1.11:5060 nua(0x8ed3d50): call state changed: init -> calling, sent offer nta: received 180 Ringing for INVITE (24016408) nta: 180 Ringing is going to a transaction nua(0xb6427168): call state changed: calling -> proceeding 2012-02-08 13:42:25.504438 [NOTICE] sofia.c:5604 Ring-Ready sofia/internal/sip:240 at 192.168.1.11:5060! 2012-02-08 13:42:25.504438 [NOTICE] switch_ivr_originate.c:483 Ring Ready sofia/internal/220 at entwick1! nta: received 486 Busy Here for INVITE (24016408) nta: 486 Busy Here is going to a transaction nta: sent ACK (24016408) to UDP/192.168.1.11:5060 nua(0xb689fe28): call state changed: calling -> init nua(0xb689fe28): removing session usage 2012-02-08 13:42:25.604444 [NOTICE] sofia.c:6276 Hangup sofia/internal/sip:240 at 192.168.1.11:5060 [CS_CONSUME_MEDIA] [USER_BUSY] nta: received 486 Busy Here for INVITE (24016408) nta: 486 Busy Here is going to a transaction nta: sent ACK (24016408) to UDP/192.168.1.11:5060 nua(0xb68018d8): call state changed: calling -> init nua(0xb68018d8): removing session usage 2012-02-08 13:42:25.704439 [NOTICE] sofia.c:6276 Hangup sofia/internal/sip:240 at 192.168.1.11:5060 [CS_CONSUME_MEDIA] [USER_BUSY] nta: timer I fired, terminate 407 response nta_incoming_timer: 0/0 resent, 0/0 tout, 1/1 term, 1/2 free nta: timer A fired, retransmit INVITE (24016408) nta: resent INVITE (24016408) to */192.168.1.11:5060 nta_outgoing_timer: 1/1 resent, 0/1 tout, 0/2 term, 0/4 free nta: received 486 Busy Here for INVITE (24016408) nta: 486 Busy Here is going to a transaction nta: sent ACK (24016408) to UDP/192.168.1.11:5060 nua(0x8ed3d50): call state changed: calling -> init nua(0x8ed3d50): removing session usage 2012-02-08 13:42:26.464439 [NOTICE] sofia.c:6276 Hangup sofia/internal/sip:240 at 192.168.1.11:5060 [CS_CONSUME_MEDIA] [USER_BUSY] nta: received 486 Busy Here for INVITE (24016408) nta: 486 Busy Here is going to a transaction nta: sent ACK (24016408) to UDP/192.168.1.11:5060 nta: 486 Busy Here is duplicate response to 24016408 INVITE Via: SIP/2.0/UDP 192.168.1.4 ;branch=z9hG4bKva9aUQeyeN81p nta: received 486 Busy Here for INVITE (24016408) nta: 486 Busy Here is going to a transaction nta: sent ACK (24016408) to UDP/192.168.1.11:5060 nta: 486 Busy Here is duplicate response to 24016408 INVITE Via: SIP/2.0/UDP 192.168.1.4 ;branch=z9hG4bKva9aUQeyeN81p nta: received 486 Busy Here for INVITE (24016408) nta: 486 Busy Here is going to a transaction nta: sent ACK (24016408) to UDP/192.168.1.11:5060 nta: 486 Busy Here is duplicate response to 24016408 INVITE Via: SIP/2.0/UDP 192.168.1.4 ;branch=z9hG4bKva9aUQeyeN81p nta: received CANCEL sip:240 at entwick1;user=phone SIP/2.0 (CSeq 2) nta: CANCEL (2) is going to INVITE (2) nta: sent 487 Request Terminated for INVITE (2) nua(0xb6ea4088): removing session usage nua(0xb6ea4088): call state changed: early -> terminated From jdiaz at coinfru.com Wed Feb 8 15:50:59 2012 From: jdiaz at coinfru.com (Josue Diaz Cruz) Date: Wed, 8 Feb 2012 13:50:59 +0100 Subject: [Freeswitch-users] I have 4 different ethernet ports with different ip's Message-ID: I have a server with 4 ETH. Each one have a different IP In 2 cases i fixed the ip address, In the other 2 cases i have 2 cable connections in bridge mode so i receive the public ip in dhcp mode. eth0 local address 192.168.1.100 eth1 local address 192.168.2.100 eth2 public address dhcp (cable) eth3 public address dhcp (cable) I was thinking to build 4 different profiles for each eth port, but i came to know that if we change the place of colocation it will not work properly because the dhcp ip. It is any way to attach a profile to the eth port, not the ip? Or if you know a better way to manage this case. I would like to receive traffic from the 2 public ip's and route it both like if i have just one profile. Josue Diaz Cruz Departamento Tecnico y Soporte jdiaz at coinfru.com C/ Balsicas 3 Alquerias | 30580 | Murcia www.coinfru.com -------------- next part -------------- An HTML attachment was scrubbed... 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Name: not available Type: image/jpeg Size: 4705 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/a9068384/attachment.jpe From rmorin at blie-ent.com Wed Feb 8 15:50:50 2012 From: rmorin at blie-ent.com (Rob Morin) Date: Wed, 8 Feb 2012 07:50:50 -0500 Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: References: <005801cce12e$05358300$0fa08900$@net.au> <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> <4F2A9482.1020100@earthspike.net> <4F2B8B51.2080506@earthspike.net> <4F2C2EFA.9030105@earthspike.net> <4F3137A7.2040908@earthspike.net> Message-ID: <117001cce660$49c7edf0$dd57c9d0$@blie-ent.com> What you're seeing is sendmail is having a memory tracking problem - it's attempting to write to memory that's already allocated to something else. This forum is a Freeswitch forum. You'll need to troubleshoot this problem with the sendmail team, on their own forum. (The reason you won't get any help here is because Freeswitch has already delivered the email to sendmail. Sendmail acknowledged receiving it and is now showing the segfault (or memory problem). It isn't a Freeswitch issue, though, and this team is not familiar with the sendmail code in such a way as to troubleshoot it or fix it.) I had the same issue. I ended up installing SSMTP which simply routes the mail off of my PBX to an SMTP server. Unless you actually want your Freeswitch server to also be a mail server, that is the path that I recommend. If you're running CentOS, yum install ssmtp and put the following in your /usr/local/freeswitch/conf/autoload_configs/switch.conf.xml Good luck, Rob From: Balamurugan Mahendran [mailto:b2m at a-cti.com] Sent: Wednesday, February 08, 2012 6:53 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Fs ignoring voicemail to email I think found the log, not sure how to proceed. Feb 8 11:51:03 ip-10-252-39-229 kernel: [7140198.319410] sendmail[15835]: segfault at 7fff89aa2ed0 ip 00007f75670e9a8a sp 00007fff89aa2ea0 error 6 in sendmail[7f7567056000+cd000] Thanks, Bala On Tue, Feb 7, 2012 at 10:23 PM, Balamurugan Mahendran wrote: Is there anyway for more logs? Thanks for your help!! --Bala On Tue, Feb 7, 2012 at 8:34 PM, Balamurugan Mahendran wrote: I did, also I have no issue sending email outside FS. Thanks, Bala On Tue, Feb 7, 2012 at 8:31 PM, Balamurugan Mahendran wrote: I did, also I have no issue sending email outside FS. Thanks, Bala On Tue, Feb 7, 2012 at 8:09 PM, John wrote: I updated the wiki mod_voicemail page a few days ago with some instructions for debugging email from freeswitch. Have you tried those? On 07/02/12 13:47, Balamurugan Mahendran wrote: All, I am having the same issue, its not sending email(extension --> lua script) getting "Segmentation fault" xml : dialplan : Lua : caller=503; freeswitch.consoleLog("info","From :"..caller); session:set_tts_parms("flite", "slt"); session:speak("Welcome To Voice Mail !. You Can Leave Your Message Here."); path="/usr/local/freeswitch/recordings/"; prompt=caller..".mp3"; recpath=path..prompt; freeswitch.consoleLog("info","record path="..recpath); session:recordFile(recpath,30,10,10); session:speak("Thank you."); freeswitch.consoleLog("info","testing"); freeswitch.email("b2m at a-cti.com", "saraswathi.devaraj at a-cti.com", "subject: Voicemail from 801\n", "Hello,\n\nYou've got a voicemail, click the attachment to listen to it.", "/usr/local/freeswitch/recordings/503.mp3", "", ""); freeswitch.consoleLog("info","hai"); switchconf : Thanks for your help!! Thanks, Bala On Sat, Feb 4, 2012 at 12:31 AM, John wrote: On 03/02/12 18:00, Thomas Hoellriegel wrote: > > Its works fine!! Good news! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/6fb32887/attachment-0001.html From binni at itanet.nu Wed Feb 8 10:01:06 2012 From: binni at itanet.nu (Brynjolfur Thorvardsson) Date: Wed, 8 Feb 2012 08:01:06 +0100 Subject: [Freeswitch-users] How to make calls from Browser? In-Reply-To: References: Message-ID: Hi, ZoIPer has this functionality coupled with Asterisk. Fra: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] P? vegne af Balamurugan Mahendran Sendt: 8. februar 2012 07:38 Til: FreeSWITCH Users Help Emne: [Freeswitch-users] How to make calls from Browser? Hi all, I am trying to make calls from browser(chrome, safari, FireFox, Opera, IE) is there any known open source tool available already? Thanks, Bala -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/c62c4a3c/attachment-0001.html From brett at launch3.net Wed Feb 8 02:24:40 2012 From: brett at launch3.net (Brett Wilson) Date: Tue, 7 Feb 2012 18:24:40 -0500 Subject: [Freeswitch-users] Help with attended transfer / att_xfer Message-ID: <029e01cce5ef$aaadc0c0$00094240$@launch3.net> Hey guys, I need a pointer. I'm trying to get some kind of functionality where after a attended transfer is completed, ie. The ouside caller is connected to the 2nd phone after the 1st phone has hung up or chosen to continue the transfer. I would like a beep to sound for the 2nd phone or for both legs, just to let the recipient of the transfer know that it has gone through and they can start talking. Any ideas? Thanks Brett Wilson 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: Blogger-logo Description: Description: FaceBook-Logo Description: Description: Twitter-Logo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/f6937045/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 1815 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/f6937045/attachment-0003.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 1680 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/f6937045/attachment-0004.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 1715 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120207/f6937045/attachment-0005.jpe From virbhati at gmail.com Wed Feb 8 15:50:02 2012 From: virbhati at gmail.com (virendra bhati) Date: Wed, 8 Feb 2012 18:20:02 +0530 Subject: [Freeswitch-users] I am not able to reg with 5060 via softphone Message-ID: Hi list, I am new for FreeSwitch. I was working with asterisk since 4 years. Now i want to learn FreeSwitch. I have installed at my server. But when I try to reg with server .I am not able to register with it with port 5060. but if I will 5080 then reg with server but can't call to local extension like 1001 - 1019. Please guide me what is the issue ? *fs_CLI log:-* EXECUTE sofia/external/1006 at 78.129.163.44 bridge(user/1005 at 78.129.163.44) 2012-02-08 12:59:37.493777 [DEBUG] switch_ivr_originate.c:1885 variable string 0 = [presence_id=1005 at 78.129.163.44] 2012-02-08 12:59:37.493777 [ERR] switch_ivr_originate.c:2430 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2012-02-08 12:59:37.493777 [DEBUG] switch_ivr_originate.c:3228 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2012-02-08 12:59:37.493777 [ERR] switch_ivr_originate.c:2430 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2012-02-08 12:59:37.493777 [DEBUG] switch_ivr_originate.c:3228 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2012-02-08 12:59:37.493777 [INFO] mod_dptools.c:2355 Originate Failed. Cause: USER_NOT_REGISTERED EXECUTE sofia/external/1006 at 78.129.163.44 answer() -- Thanks and regards Virendra Bhati +91-8885268942 Software Engineer E-mail-: virbhati at gmail.com Skype id:- virbhati2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/6712ec3f/attachment-0001.html From brett at launch3.net Wed Feb 8 17:46:42 2012 From: brett at launch3.net (Brett Wilson) Date: Wed, 8 Feb 2012 09:46:42 -0500 Subject: [Freeswitch-users] Help with attended transfer / att_xfer Message-ID: <032301cce670$78c0ab00$6a420100$@launch3.net> Sorry if I double posted this. I did not see the message in the digest so I am reposting. Hey guys, I need a pointer. I'm trying to get some kind of functionality where after a attended transfer is completed, ie. The ouside caller is connected to the 2nd phone after the 1st phone has hung up or chosen to continue the transfer. I would like a beep to sound for the 2nd phone or for both legs, just to let the recipient of the transfer know that it has gone through and they can start talking. Any ideas? Thanks Brett Wilson 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: Blogger-logo Description: Description: FaceBook-Logo Description: Description: Twitter-Logo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/a91eff23/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... 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Name: not available Type: image/jpeg Size: 1715 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/a91eff23/attachment-0005.jpe From mario_fs at mgtech.com Wed Feb 8 19:56:26 2012 From: mario_fs at mgtech.com (Mario G) Date: Wed, 8 Feb 2012 08:56:26 -0800 Subject: [Freeswitch-users] Can't get Paging to work on SPA962 In-Reply-To: References: Message-ID: Anyone out there that can help with a simple page all phones on Linksys SPA phones? On Feb 1, 2012, at 11:58 AM, Mario G wrote: > I have not been able to get paging to work on SPA962s for a year now, I ran traces searched the web, etc. No luck. I found a post from 2009 that said it works but no description on how. Here is what happens: If I connect my old SPA9000 PBX up and power it on, then turn it off, FreeSwitch paging works! But, if a phone is powered off, FreeSwitch paging no longer works. Power on/off SPA9000 and all is well. Anyone out there have it working that could tell me the secret? I am using *11 to page to trigger the rtp_multicast_page extension. Thanks in advance! > MarioG > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/b65de142/attachment.html From djbinter at gmail.com Wed Feb 8 20:26:03 2012 From: djbinter at gmail.com (DJB International) Date: Wed, 8 Feb 2012 09:26:03 -0800 Subject: [Freeswitch-users] Polycom BLF Message-ID: Does anyone ever experience with BLF for DND on polycom expansion module? The BLF is working fine when the extension is busy, but when the extension has DND on, it did not change the LED light. Any sugestion? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/c36c6cdb/attachment.html From jdiaz at coinfru.com Wed Feb 8 20:28:51 2012 From: jdiaz at coinfru.com (Josue Diaz Cruz) Date: Wed, 8 Feb 2012 18:28:51 +0100 Subject: [Freeswitch-users] Can someone help me with a complicated configuration? Message-ID: I have a server with 4 ETH. Each one have a different IP In 2 cases i fixed the ip address, In the other 2 cases i have 2 cable connections in bridge mode so i receive the public ip in dhcp mode. eth0 local address 192.168.1.100 eth1 local address 192.168.2.100 eth2 public address dhcp (cable) eth3 public address dhcp (cable) I was thinking to build 4 different profiles for each eth port, but i came to know that if we change the place of colocation it will not work properly because the dhcp ip. It is any way to attach a profile to the eth port, not the ip? Or if you know a better way to manage this case. I would like to receive traffic from the 2 public ip's and route it both like if i have just one profile. Josue Diaz Cruz Departamento Tecnico y Soporte jdiaz at coinfru.com C/ Balsicas 3 Alquerias | 30580 | Murcia www.coinfru.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/5edda5ed/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4705 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/5edda5ed/attachment.jpe From msc at freeswitch.org Wed Feb 8 20:36:38 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 8 Feb 2012 09:36:38 -0800 Subject: [Freeswitch-users] Freeswitch opened multiple outbound channels for one call with different uuid In-Reply-To: <4F326E84.6030302@telefaks.biz> References: <4F323042.8080603@telefaks.biz> <4F326E84.6030302@telefaks.biz> Message-ID: > It was found that freeswitch performs multiple INVITES. But why? > > One reason why multiple invites are sent out is if FreeSWITCH doesn't receive anything back. This could mean the far end is down for some reason or that there is a network issue between FS and the other endpoint. NAT is a common issue in this sort of configuration. I recommend that you look at SIP traces on each end of the SIP dialog. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/518bb963/attachment.html From msc at freeswitch.org Wed Feb 8 20:40:55 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 8 Feb 2012 09:40:55 -0800 Subject: [Freeswitch-users] Help with attended transfer / att_xfer In-Reply-To: <029e01cce5ef$aaadc0c0$00094240$@launch3.net> References: <029e01cce5ef$aaadc0c0$00094240$@launch3.net> Message-ID: Brett, No worries on the double-post - we have a first-time sender moderation filter so I had to allow your messages through. You should be go to go from here on out. As to your question, are you using the default configuration or are you doing something different? Also, how are you executing the transfer - using the transfer button on the telephone or the *1 feature code? Thanks, MC On Tue, Feb 7, 2012 at 3:24 PM, Brett Wilson wrote: > Hey guys,**** > > I need a pointer. I?m trying to get some kind of functionality where after > a attended transfer is completed, ie. The ouside caller is connected to the > 2nd phone after the 1st phone has hung up or chosen to continue the > transfer. I would like a beep to sound for the 2nd phone or for both > legs, just to let the recipient of the transfer know that it has gone > through and they can start talking.**** > > Any ideas?**** > > ** ** > > Thanks**** > > ** ** > > ** ** > > *Brett Wilson* > > *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: Blogger-logo][image: > Description: Description: FaceBook-Logo][image: > Description: Description: Twitter-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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/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/jpeg Size: 1815 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/e1b59e27/attachment-0005.jpe From asrivas at gmail.com Wed Feb 8 21:35:38 2012 From: asrivas at gmail.com (Anurag Srivastava) Date: Wed, 8 Feb 2012 10:35:38 -0800 Subject: [Freeswitch-users] external_sip_ip not refreshed after ip address change. In-Reply-To: References: Message-ID: Hi, Could anybody help with this. I am not running freeswitch with nonat. Just to reiterate my original > problem, external_sip_ip is reloaded only once at startup even though stun > is specified which is the cause of hte problem. external_rtp_ip is updated > correctly with stun. > > Hi Brian, >> Thanks for your help. From what I see the parameter auto-restart=true >> will only restart profiles if the local interface ip address on the machine >> changes. Will it also restart if my public ip address changes? My machine >> is behind a NAT with port forwarding. I am trying to restart profiles if my >> public ip address changes. >> Thanks >> >> Can somebody help me with this? >>> >>> >>> I have a question about nat behavior in freeswitch. Basically my >>>> external calls to freeswitch are getting disconnected after 30 seconds of >>>> two way audio when my external ip address changes. I have dhcp from my ISP >>>> and am using external_sip_ip and external_rtp_ip as stun:. >>>> When my IP changes I see that external_sip_ip does not get refreshed but >>>> external_rtp_ip does. I am not allowed to enable upnp/nat-pmp on my router. >>>> Apparently it is a known issue that external_sip_ip is read just >>>> at load time and not refreshed even if it is specified in stun format >>>> Is there a fix to this problem? >>>> >>>> There is always the option of restarting profile when ddclient notes an >>>> ip change. Is there something inbuilt into FS. It is already finding that >>>> ip address has changed as reflected in external_rtp_ip which does use stun >>>> and gets the right ip address. >>>> >>> >>> >>> >>> -- >>> Regards >>> Anurag >>> >> >> >> >> -- >> Regards >> Anurag >> > > > > -- > Regards > Anurag > -- Regards Anurag -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/bd5bae0a/attachment.html From brian at freeswitch.org Wed Feb 8 22:12:31 2012 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Feb 2012 13:12:31 -0600 Subject: [Freeswitch-users] external_sip_ip not refreshed after ip address change. In-Reply-To: References: Message-ID: You'll need to understand how FreeSWITCH works. that external_sip_ip variable is probably fake and in vars.xml and is setup to stun: something... You can't really depend on that in your case so you'll need to get creative to set your variables properly that get used down in the sofia configs to bind to the proper IP's. This exercise is sadly left up to you to do. I use a script to write out some XML that gets included but if your IP changes you need to restart the sofia profiles. /b On Feb 8, 2012, at 12:35 PM, Anurag Srivastava wrote: > Hi, > Could anybody help with this. -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/91148c67/attachment.html From freeswitch at earthspike.net Wed Feb 8 22:18:58 2012 From: freeswitch at earthspike.net (John) Date: Wed, 08 Feb 2012 19:18:58 +0000 Subject: [Freeswitch-users] Fs ignoring voicemail to email In-Reply-To: References: <005801cce12e$05358300$0fa08900$@net.au> <035101cce1a1$f574f990$e05eecb0$@blie-ent.com> <4F2A9482.1020100@earthspike.net> <4F2B8B51.2080506@earthspike.net> <4F2C2EFA.9030105@earthspike.net> <4F3137A7.2040908@earthspike.net> Message-ID: <4F32CAA2.4060801@earthspike.net> It gets deleted after the message is sent. By then, it's an attachment to the email and no longer required. John On 08/02/12 12:04, Balamurugan Mahendran wrote: > Also looks like FS creates file in /tmp location, but I cannot find > this file in this location. > > fs_cli: > > 2012-02-08 12:01:26.506096 [DEBUG] switch_utils.c:767 Emailed file > [/tmp/mail.13287024861013] to [b2m at a-cti.com ] > > syslog : > > Feb 8 12:01:26 ip-10-252-39-229 kernel: [7140816.707307] > show_signal_msg: 3 callbacks suppressed > Feb 8 12:01:26 ip-10-252-39-229 kernel: [7140816.707313] > sendmail[1335]: segfault at 7fff4a735100 ip 00007fb706a67a8a sp > 00007fff4a7350d0 error 6 in sendmail[7fb7069d4000+cd000] > > > Thanks, > Bala > > On Wed, Feb 8, 2012 at 5:22 PM, Balamurugan Mahendran > wrote: > > I think found the log, not sure how to proceed. > > Feb 8 11:51:03 ip-10-252-39-229 kernel: [7140198.319410] > sendmail[15835]: segfault at 7fff89aa2ed0 ip 00007f75670e9a8a sp > 00007fff89aa2ea0 error 6 in sendmail[7f7567056000+cd000] > > Thanks, > Bala > > > On Tue, Feb 7, 2012 at 10:23 PM, Balamurugan Mahendran > > wrote: > > Is there anyway for more logs? Thanks for your help!! > > --Bala > > > On Tue, Feb 7, 2012 at 8:34 PM, Balamurugan Mahendran > > wrote: > > I did, also I have no issue sending email outside FS. > > Thanks, > Bala > > On Tue, Feb 7, 2012 at 8:31 PM, Balamurugan Mahendran > > wrote: > > I did, also I have no issue sending email outside FS. > > Thanks, > Bala > > On Tue, Feb 7, 2012 at 8:09 PM, John > > wrote: > > I updated the wiki mod_voicemail page a few days > ago with some instructions for debugging email > from freeswitch. Have you tried those? > > > On 07/02/12 13:47, Balamurugan Mahendran wrote: >> All, >> >> I am having the same issue, its not sending >> email(extension --> lua script) getting >> *"Segmentation fault"* >> >> xml : >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> > value="domestic,international,local"/> >> >> >> > value="Extension 503"/> >> > value="503"/> >> > value="$${outbound_caller_name}"/> >> > value="$${outbound_caller_id}"/> >> >> >> >> >> >> dialplan : >> >> >> >> > expression="^(503)$"> >> >> >> >> >> >> >> Lua : >> >> caller=503; >> freeswitch.consoleLog("info","From :"..caller); >> session:set_tts_parms("flite", "slt"); >> session:speak("Welcome To Voice Mail !. You Can >> Leave Your Message Here."); >> path="/usr/local/freeswitch/recordings/"; >> prompt=caller..".mp3"; >> recpath=path..prompt; >> freeswitch.consoleLog("info","record >> path="..recpath); >> session:recordFile(recpath,30,10,10); >> session:speak("Thank you."); >> >> freeswitch.consoleLog("info","testing"); >> freeswitch.email("b2m at a-cti.com >> ", >> "saraswathi.devaraj at a-cti.com >> ", >> "subject: Voicemail from 801\n", >> "Hello,\n\nYou've got a >> voicemail, click the attachment to listen to it.", >> >> "/usr/local/freeswitch/recordings/503.mp3", >> "", >> ""); >> freeswitch.consoleLog("info","hai"); >> >> >> >> switchconf : >> >> >> >> >> >> >> Thanks for your help!! >> >> Thanks, >> Bala >> >> >> >> On Sat, Feb 4, 2012 at 12:31 AM, John >> > > wrote: >> >> On 03/02/12 18:00, Thomas Hoellriegel wrote: >> > >> > Its works fine!! >> Good news! >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/85236223/attachment-0001.html From aakashviswam at gmail.com Wed Feb 8 22:49:51 2012 From: aakashviswam at gmail.com (Aakash) Date: Wed, 8 Feb 2012 11:49:51 -0800 (PST) Subject: [Freeswitch-users] skipping vm and plays enter the pwd Message-ID: <1328730591696-7267135.post@n2.nabble.com> Hi All, I have configured has any inbound calls enter to my inbound number play an ivr and transfer to user ext (eg 1080)pressed by customer,if the user dint pick the call enter to their voicemail automatically.But sometimes i am getting please enter password followed by # by skipping voicemail message.I have attached my logs below http://pastebin.freeswitch.org/18321 Regards, Aakash.V -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/skipping-vm-and-plays-enter-the-pwd-tp7267135p7267135.html Sent from the freeswitch-users mailing list archive at Nabble.com. From brian at freeswitch.org Wed Feb 8 23:21:14 2012 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Feb 2012 14:21:14 -0600 Subject: [Freeswitch-users] Freeswitch opened multiple outbound channels for one call with different uuid In-Reply-To: <4F326E84.6030302@telefaks.biz> References: <4F323042.8080603@telefaks.biz> <4F326E84.6030302@telefaks.biz> Message-ID: <6CCD037B-A934-4541-8553-9170FB7182C5@freeswitch.org> select count(*) from sip_registrations where sip_user='XXX'; /b On Feb 8, 2012, at 6:45 AM, David wrote: > It was found that freeswitch performs multiple INVITES. But why? -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/a17ea2ed/attachment.html From brian at freeswitch.org Wed Feb 8 23:30:52 2012 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Feb 2012 14:30:52 -0600 Subject: [Freeswitch-users] external_sip_ip not refreshed after ip address change. In-Reply-To: References: Message-ID: <02D95CE1-FC86-4601-8457-61F05071D5F3@freeswitch.org> #!/usr/bin/perl $ip = `/usr/bin/snmpget -c m00se -v1 192.168.1.1 IP-MIB::ipAdEntAddr.5 | /usr/bin/awk '{print \$4}'`; $ip =~ s/\n//g; chomp($ip); print "\n\t\n\n"; I do something like this in mine before. /b On Feb 8, 2012, at 1:12 PM, Brian West wrote: > You'll need to understand how FreeSWITCH works. that external_sip_ip variable is probably fake and in vars.xml and is setup to stun: something... You can't really depend on that in your case so you'll need to get creative to set your variables properly that get used down in the sofia configs to bind to the proper IP's. This exercise is sadly left up to you to do. > > I use a script to write out some XML that gets included but if your IP changes you need to restart the sofia profiles. > > /b > > On Feb 8, 2012, at 12:35 PM, Anurag Srivastava wrote: > >> Hi, >> Could anybody help with this. > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org From vipkilla at gmail.com Thu Feb 9 00:41:11 2012 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 8 Feb 2012 16:41:11 -0500 Subject: [Freeswitch-users] presence on latest GIT Message-ID: I've noticed the past week that BLF was not working, I pulled latest GIT and tried to figure out what was going wrong. After trying many different things on the Polycom, i enabled presence debug in sofia. right awaythere were a lot of errors thrown. Is sofia presence broken in latest GIT? If so, any ETA on when it will be fixed? Thanks. From anthony.minessale at gmail.com Thu Feb 9 00:48:55 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 8 Feb 2012 15:48:55 -0600 Subject: [Freeswitch-users] presence on latest GIT In-Reply-To: References: Message-ID: The errors are really the debug info at error levels for color effects. What do you mean by not working? And why report it here after many requests to report issues to Jira? On Feb 8, 2012 3:42 PM, "Vik Killa" wrote: > I've noticed the past week that BLF was not working, I pulled latest > GIT and tried to figure out what was going wrong. After trying many > different things on the Polycom, i enabled presence debug in sofia. > right awaythere were a lot of errors thrown. Is sofia presence broken > in latest GIT? If so, any ETA on when it will be fixed? 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/656405ec/attachment.html From vipkilla at gmail.com Thu Feb 9 00:53:29 2012 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 8 Feb 2012 16:53:29 -0500 Subject: [Freeswitch-users] presence on latest GIT In-Reply-To: References: Message-ID: Well if that is the case perhaps it is not broken, I just assumed it was because when debug is enabled i'm seeing: [ERR] sofia_presence.c:1127 EVENT DUMP: everytime presence is called. It is really confusing because BLF is not working anymore and I see that error. On Wed, Feb 8, 2012 at 4:48 PM, Anthony Minessale wrote: > The errors are really the debug info at error levels for color effects. > > What do you mean by not working?? And why report it here after many requests > to report issues to Jira? > > On Feb 8, 2012 3:42 PM, "Vik Killa" wrote: >> >> I've noticed the past week that BLF was not working, I pulled latest >> GIT and tried to figure out what was going wrong. After trying many >> different things on the Polycom, i enabled presence debug in sofia. >> right awaythere were a lot of errors thrown. Is sofia presence broken >> in latest GIT? If so, any ETA on when it will be fixed? 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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From anthony.minessale at gmail.com Thu Feb 9 00:56:33 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 8 Feb 2012 15:56:33 -0600 Subject: [Freeswitch-users] presence on latest GIT In-Reply-To: References: Message-ID: It should be working but you can use that data to see why. If you enable sip trace too and capture the logs, it can be used to check for problems On Wed, Feb 8, 2012 at 3:53 PM, Vik Killa wrote: > Well if that is the case perhaps it is not broken, I just assumed it > was because when debug is enabled i'm seeing: > [ERR] sofia_presence.c:1127 EVENT DUMP: > > everytime presence is called. It is really confusing because BLF is > not working anymore and I see that error. > > On Wed, Feb 8, 2012 at 4:48 PM, Anthony Minessale > wrote: >> The errors are really the debug info at error levels for color effects. >> >> What do you mean by not working?? And why report it here after many requests >> to report issues to Jira? >> >> On Feb 8, 2012 3:42 PM, "Vik Killa" wrote: >>> >>> I've noticed the past week that BLF was not working, I pulled latest >>> GIT and tried to figure out what was going wrong. After trying many >>> different things on the Polycom, i enabled presence debug in sofia. >>> right awaythere were a lot of errors thrown. Is sofia presence broken >>> in latest GIT? If so, any ETA on when it will be fixed? 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 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 Thu Feb 9 01:08:24 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Wed, 8 Feb 2012 19:08:24 -0300 Subject: [Freeswitch-users] Help with attended transfer / att_xfer In-Reply-To: References: <029e01cce5ef$aaadc0c0$00094240$@launch3.net> Message-ID: <399BEF93D8F8443BA14E1A90B8443AC0@freeswitch.org> I believe he is using the feature code, therefore att_xfer so this would require a patch of some kind. I am this on my todo list which is quite large nowadays and I don't even know if it is feasible yet. I can take a look at the code later on. Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Wednesday, February 8, 2012 at 2:40 PM, Michael Collins wrote: > Brett, > > No worries on the double-post - we have a first-time sender moderation filter so I had to allow your messages through. You should be go to go from here on out. > > As to your question, are you using the default configuration or are you doing something different? Also, how are you executing the transfer - using the transfer button on the telephone or the *1 feature code? > > Thanks, > MC > > On Tue, Feb 7, 2012 at 3:24 PM, Brett Wilson wrote: > > > > Hey guys, > > > > > > I need a pointer. I?m trying to get some kind of functionality where after a attended transfer is completed, ie. The ouside caller is connected to the 2nd phone after the 1st phone has hung up or chosen to continue the transfer. I would like a beep to sound for the 2nd phone or for both legs, just to let the recipient of the transfer know that it has gone through and they can start talking. > > > > > > Any ideas? > > > > > > > > > > > > Thanks > > > > > > > > > > > > > > > > > > Brett Wilson > > > > > > Launch 3 Ventures, LLC > > > > > > 134 Myer Street > > > > > > Hackensack, NJ 07601 > > > > > > Phone: 877.878.9134 (tel:877.878.9134) > > Fax: 646.536.3866 (tel:646.536.3866) > > > > > > Email: Brett.Wilson at launch3.net (mailto:Brett.Wilson at launch3.net) > > > > > > AOL IM: Brett.Wilson at launch3.net (mailto:Brett.Wilson at launch3.net) > > > > > > www.Launch3.net > > > > > > www.Launch3telecom.com (http://www.launch3telecom.com/) > > > > > > *************************** > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > 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 > > > > _________________________________________________________________________ > 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... 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Name: image001.jpg Type: image/jpeg Size: 1815 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/09a2c688/attachment-0005.jpg From kemen04 at gmail.com Thu Feb 9 01:55:53 2012 From: kemen04 at gmail.com (Travis Kemen) Date: Wed, 8 Feb 2012 16:55:53 -0600 Subject: [Freeswitch-users] Polycom BLF In-Reply-To: References: Message-ID: Freeswitch does not support this, see http://jira.freeswitch.org/browse/FS-2731 Travis On Wed, Feb 8, 2012 at 11:26 AM, DJB International wrote: > Does anyone ever experience with BLF for DND on polycom expansion module? > The BLF is working fine when the extension is busy, but when the extension > has DND on, it did not change the LED light. Any sugestion? > > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/7d8f48e2/attachment.html From djbinter at gmail.com Thu Feb 9 02:08:53 2012 From: djbinter at gmail.com (DJB International) Date: Wed, 8 Feb 2012 15:08:53 -0800 Subject: [Freeswitch-users] Polycom BLF In-Reply-To: References: Message-ID: Thank you, Travis. On Wed, Feb 8, 2012 at 2:55 PM, Travis Kemen wrote: > Freeswitch does not support this, see > http://jira.freeswitch.org/browse/FS-2731 > > Travis > > On Wed, Feb 8, 2012 at 11:26 AM, DJB International wrote: > >> Does anyone ever experience with BLF for DND on polycom expansion >> module? The BLF is working fine when the extension is busy, but when the >> extension has DND on, it did not change the LED light. Any sugestion? >> >> 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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/f40661e0/attachment.html From asrivas at gmail.com Thu Feb 9 02:16:36 2012 From: asrivas at gmail.com (Anurag Srivastava) Date: Wed, 8 Feb 2012 15:16:36 -0800 Subject: [Freeswitch-users] external_sip_ip not refreshed after ip address change. In-Reply-To: <02D95CE1-FC86-4601-8457-61F05071D5F3@freeswitch.org> References: <02D95CE1-FC86-4601-8457-61F05071D5F3@freeswitch.org> Message-ID: I do something similar like below. Shouldn't this be done by freeswitch itself? After an ip change, i do notice that external_rtp_ip is correctly got from stun, but external_sip_ip is still set to its old value at load time. while [ 1 ]; do if ip=`wget -q -O - myip.dnsomatic.com`; then if [ "$ip" == "`cat $CACHE_DIR/last_ip_freeswitch`" ]; then : # do nothing else $log_info "ip address has changed" $log_info "executing freeswitch profile restarts" # reload freeswitch external profiles /usr/local/freeswitch/bin/fs_cli -x 'sofia profile remote_extensions rescan' 2>&1 | $log_info /usr/local/freeswitch/bin/fs_cli -x 'sofia profile remote_extensions restart' 2>&1 | $log_info /usr/local/freeswitch/bin/fs_cli -x 'sofia profile external rescan' 2>&1 | $log_info /usr/local/freeswitch/bin/fs_cli -x 'sofia profile external restart' 2>&1 | $log_info echo $ip > $CACHE_DIR/last_ip_freeswitch fi On Wed, Feb 8, 2012 at 12:30 PM, Brian West wrote: > #!/usr/bin/perl > > $ip = `/usr/bin/snmpget -c m00se -v1 192.168.1.1 IP-MIB::ipAdEntAddr.5 | > /usr/bin/awk '{print \$4}'`; > $ip =~ s/\n//g; > chomp($ip); > > print "\n\t data=\"external_ip=$ip\"/>\n\n"; > > > I do something like this in mine before. > > /b > > On Feb 8, 2012, at 1:12 PM, Brian West wrote: > > > You'll need to understand how FreeSWITCH works. that external_sip_ip > variable is probably fake and in vars.xml and is setup to stun: > something... You can't really depend on that in your case so you'll need to > get creative to set your variables properly that get used down in the sofia > configs to bind to the proper IP's. This exercise is sadly left up to you > to do. > > > > I use a script to write out some XML that gets included but if your IP > changes you need to restart the sofia profiles. > > > > /b > > > > On Feb 8, 2012, at 12:35 PM, Anurag Srivastava wrote: > > > >> Hi, > >> Could anybody help with this. > > > > -- > > Brian West > > FreeSWITCH Solutions, LLC > > Phone: +1 (918) 420-9266 > > Fax: +1 (918) 420-9267 > > brian at freeswitch.org > > http://www.freeswitch.org > > > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Regards Anurag -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/0b36e541/attachment.html From brett at launch3.net Thu Feb 9 03:14:06 2012 From: brett at launch3.net (Brett Wilson) Date: Wed, 8 Feb 2012 19:14:06 -0500 Subject: [Freeswitch-users] Help with attended transfer / att_xfer In-Reply-To: <399BEF93D8F8443BA14E1A90B8443AC0@freeswitch.org> References: <029e01cce5ef$aaadc0c0$00094240$@launch3.net> <399BEF93D8F8443BA14E1A90B8443AC0@freeswitch.org> Message-ID: <03d901cce6bf$bcc682d0$36538870$@launch3.net> Thanks guys I am using the FS att_xfer along with the meta bind. I tried attended transfer on my gxp2100 phones and it works but I am under the impression that the phone is actually handling the sip switching itself for that functionality, and our MOH was not being played while the two internal parties were speaking, before putting through the transfer. It was just dead silence which did not sit well with our customers. So I switched to the FS feature, and it works great. Only problem is that we don?t know when to start speaking after the transfer! Also I have another question. I have FS installed via the FusionPBX linux disk image. What is the best way to keep FS updated? Do I need to pull from git and build from source or what? Brett Wilson 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: Blogger-logo Description: Description: FaceBook-Logo Description: Description: Twitter-Logo From: Jo?o Mesquita [mailto:jmesquita at freeswitch.org] Sent: Wednesday, February 08, 2012 5:08 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help with attended transfer / att_xfer I believe he is using the feature code, therefore att_xfer so this would require a patch of some kind. I am this on my todo list which is quite large nowadays and I don't even know if it is feasible yet. I can take a look at the code later on. Regards, -- Jo?o Mesquita Sent with Sparrow On Wednesday, February 8, 2012 at 2:40 PM, Michael Collins wrote: Brett, No worries on the double-post - we have a first-time sender moderation filter so I had to allow your messages through. You should be go to go from here on out. As to your question, are you using the default configuration or are you doing something different? Also, how are you executing the transfer - using the transfer button on the telephone or the *1 feature code? Thanks, MC On Tue, Feb 7, 2012 at 3:24 PM, Brett Wilson wrote: Hey guys, I need a pointer. I?m trying to get some kind of functionality where after a attended transfer is completed, ie. The ouside caller is connected to the 2nd phone after the 1st phone has hung up or chosen to continue the transfer. I would like a beep to sound for the 2nd phone or for both legs, just to let the recipient of the transfer know that it has gone through and they can start talking. Any ideas? Thanks Brett Wilson 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: Blogger-logo Description: Description: FaceBook-Logo Description: Description: Twitter-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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/b6e8c9e7/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 1815 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/b6e8c9e7/attachment-0003.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 1680 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/b6e8c9e7/attachment-0004.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 1715 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/b6e8c9e7/attachment-0005.jpe From vipkilla at gmail.com Thu Feb 9 03:17:25 2012 From: vipkilla at gmail.com (Vik Killa) Date: Wed, 8 Feb 2012 19:17:25 -0500 Subject: [Freeswitch-users] presence on latest GIT In-Reply-To: References: Message-ID: Would anyone be willing to share their working BLF polycom configs? On Wed, Feb 8, 2012 at 4:56 PM, Anthony Minessale wrote: > It should be working but you can use that data to see why. > If you enable sip trace too and capture the logs, it can be used to > check for problems From daniel at lumicall.org Thu Feb 9 01:11:18 2012 From: daniel at lumicall.org (Daniel - Lumicall.org) Date: Wed, 08 Feb 2012 23:11:18 +0100 Subject: [Freeswitch-users] finally, an open alternative to Viber Message-ID: <4F32F306.9050309@lumicall.org> The rise of Viber has, for some people, been a case of `skype, not again!' until now... Lumicall is now in the Android market - and it fully interacts with other SIP products using ENUM and SRV records. Any feedback on this is welcome. Interconnect (FreeSwitch calling to/from Lumicall users) is based on the idea of federated SIP, it is explained at http://www.lumicall.org for those who want to connect up to it. Please bear in mind: Lumicall supports ICE (RFC 5245) for NAT traversal, it is using the ice4j implementation from the Jitsi community. This makes the SIP packets bigger and often they are too big for the MTU of a UDP packet. When using ICE, it seems essential to use TLS, to avoid the MTU problems and also to avoid routers mangling the SIP headers (ICE doesn't need help from routers, they only confuse the algorithm) Various other issues are addressed in the release notes and FAQ http://www.lumicall.org/faq If there are specific issues with FreeSwitch compatibility (either for the Lumicall app, or for interconnect using ENUM/SRV/TLS) then I'm happy to start a dedicated page or wiki to collect the solutions in one place for FreeSwitch users. From freeswitch-users.scott at comsetic.net Thu Feb 9 02:05:05 2012 From: freeswitch-users.scott at comsetic.net (Scott) Date: Thu, 09 Feb 2012 10:05:05 +1100 Subject: [Freeswitch-users] Blind Transfer Message-ID: <20120209100505.620465yayx3v3g4c@mail.horsley.id.au> Hi list, Trying to resolve an issue with blind transfers. When transferring a call, the A leg is terminated if a blind transfer is started, an attended transfer doesn't create this scenario. Here are the transcript logs from FS relating to the event. I have log level set to 6 but can increase this if I need more information. Call flow is as follows.. ext18: Calls ext51 ext18: transfers call to ext15 ext15: rings for a split second ext51: lands in voicemail of 15 ext18: clears as normal OS: CentOS release 5.7 (Final) freeswitch at internal> version FreeSWITCH Version 1.0.head (git-9de1e1a 2012-02-07 14-36-22 -0500) +OK log level [7] freeswitch at internal> 2012-02-08 16:08:10.291633 [NOTICE] switch_channel.c:920 New Channel sofia/internal/18 at 10.100.0.9 [6fd7a0b0-c518-42ff-bb24-e2acf8121b9e] 2012-02-08 16:08:10.291633 [INFO] mod_dialplan_xml.c:481 Processing Scott <18>->51 in context default 2012-02-08 16:08:10.311819 [NOTICE] switch_channel.c:920 New Channel sofia/internal/sip:51 at 10.100.0.16:1024 [f8f27c7d-9198-4225-a94d-2eadb9bb58a8] 2012-02-08 16:08:10.391998 [NOTICE] sofia.c:5458 Ring-Ready sofia/internal/sip:51 at 10.100.0.16:1024! 2012-02-08 16:08:10.391998 [INFO] switch_ivr_originate.c:1115 Sending early media 2012-02-08 16:08:10.411314 [NOTICE] sofia_glue.c:3899 Pre-Answer sofia/internal/18 at 10.100.0.9! 2012-02-08 16:08:10.431704 [INFO] mod_com_g729.c:119 ENCODER CREATE - 0xb6ed5160 0x8b23a08 2012-02-08 16:08:10.731162 [INFO] mod_com_g729.c:148 DECODER CREATE - 0xb6ed5108 0x8c094c0 2012-02-08 16:08:11.977555 [NOTICE] sofia.c:6070 Channel [sofia/internal/sip:51 at 10.100.0.16:1024] has been answered 2012-02-08 16:08:11.996869 [NOTICE] switch_ivr_originate.c:3209 Channel [sofia/internal/18 at 10.100.0.9] has been answered 2012-02-08 16:08:12.098595 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0xb4d025b0 (nil) 2012-02-08 16:08:12.098595 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0xb4d025b0 (nil) 2012-02-08 16:08:12.177926 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0xb4d02558 (nil) 2012-02-08 16:08:12.177926 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0xb4d02558 (nil) 2012-02-08 16:08:14.183102 [INFO] mod_com_g729.c:119 ENCODER CREATE - 0xb4d029e0 0xb6e6ac00 2012-02-08 16:08:15.750004 [NOTICE] switch_ivr.c:1711 Transfer sofia/internal/sip:51 at 10.100.0.16:1024 to XML[15 at default] 2012-02-08 16:08:15.769362 [NOTICE] switch_ivr_bridge.c:1372 Hangup sofia/internal/18 at 10.100.0.9 [CS_EXECUTE] [NORMAL_CLEARING] 2012-02-08 16:08:15.789438 [INFO] mod_dialplan_xml.c:481 Processing Scott <18>->15 in context default 2012-02-08 16:08:15.789438 [INFO] switch_channel.c:2695 sofia/internal/sip:51 at 10.100.0.16:1024 Flipping CID from "Scott" <18> to "Outbound Call" <51> 2012-02-08 16:08:15.789438 [NOTICE] switch_channel.c:920 New Channel sofia/internal/sip:15 at 10.100.0.15:5064 [acf45601-79ad-4518-907f-7a58155748ab] 2012-02-08 16:08:16.231456 [NOTICE] sofia.c:5458 Ring-Ready sofia/internal/sip:15 at 10.100.0.15:5064! 2012-02-08 16:08:16.291519 [INFO] mod_com_g729.c:148 DECODER CREATE - 0xb4d029f8 0xb6c41798 2012-02-08 16:08:16.291519 [NOTICE] switch_ivr_originate.c:3182 Hangup sofia/internal/sip:15 at 10.100.0.15:5064 [CS_CONSUME_MEDIA] [NO_ANSWER] 2012-02-08 16:08:16.291519 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [NO_ANSWER] 2012-02-08 16:08:16.291519 [INFO] mod_dptools.c:2884 Originate Failed. Cause: NO_ANSWER 2012-02-08 16:08:16.291519 [NOTICE] switch_core_session.c:1395 Session 991 (sofia/internal/sip:15 at 10.100.0.15:5064) Ended 2012-02-08 16:08:16.291519 [NOTICE] switch_core_session.c:1397 Close Channel sofia/internal/sip:15 at 10.100.0.15:5064 [CS_DESTROY] 2012-02-08 16:08:16.371076 [NOTICE] switch_core_session.c:1395 Session 989 (sofia/internal/18 at 10.100.0.9) Ended 2012-02-08 16:08:16.371076 [NOTICE] switch_core_session.c:1397 Close Channel sofia/internal/18 at 10.100.0.9 [CS_DESTROY] 2012-02-08 16:08:16.371076 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0xb6ed5108 (nil) 2012-02-08 16:08:16.371076 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0xb6ed5108 0x8c094c0 2012-02-08 16:08:16.371076 [INFO] mod_com_g729.c:89 DECODER DESTROY - 0xb6ed5108 0x8c094c0 2012-02-08 16:08:16.371076 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - 0xb6ed5160 0x8b23a08 2012-02-08 16:08:16.371076 [INFO] mod_com_g729.c:79 DECODER DESTROYX - 0xb6ed5160 (nil) 2012-02-08 16:08:16.371076 [INFO] mod_com_g729.c:83 ENCODER DESTROY - 0xb6ed5160 0x8b23a08 2012-02-08 16:08:17.292845 [NOTICE] switch_channel.c:920 New Channel loopback/app=voicemail:default 10.100.0.9 15-a [7b76ae61-3465-40bb-9d65-41f6cba2c3d2] 2012-02-08 16:08:17.292845 [NOTICE] switch_channel.c:918 Rename Channel loopback/app=voicemail:default 10.100.0.9 15-a->loopback/voicemail-a [7b76ae61-3465-40bb-9d65-41f6cba2c3d2] 2012-02-08 16:08:17.292845 [NOTICE] switch_channel.c:920 New Channel loopback/voicemail-b [e02cc153-8651-422c-bb98-f6f7a3244418] 2012-02-08 16:08:17.292845 [NOTICE] mod_loopback.c:760 Pre-Answer loopback/voicemail-a! 2012-02-08 16:08:17.292845 [NOTICE] mod_dptools.c:1143 Pre-Answer loopback/voicemail-b! 2012-02-08 16:08:17.414279 [INFO] mod_com_g729.c:119 ENCODER CREATE - 0x8b6e120 0xb6ec4118 From mojo1736 at privatedemail.net Thu Feb 9 02:50:09 2012 From: mojo1736 at privatedemail.net (Josh) Date: Wed, 08 Feb 2012 23:50:09 +0000 Subject: [Freeswitch-users] [newbie] questions Message-ID: <4F330A31.4060804@privatedemail.net> I am considering using FreeSWITCH as a replacement of our (rather crappy, outdated and buggy) sip proxy system, and have the following system setup on the server: * eth0 faces the external Internet interface, *but* it does not have a public IP address (it has a private one given to it by my ISP's DHCP server); * eth1 faces our internal network (say 10.1.1.0/24); * tun0 serves all mobile smartphones and connects to the internal network (it has a different ip range, say 10.1.2.0/24) - they are all connected via the Internet using OpenVPN; I would like to configure FreeSWITCH as a local registrar and ask it to handle internal calls between ourselves (eth1<->tun0). I would also like to use one external VOIP provider to which FreeSWITCH registers on startup. This also needs to have all NAT-related issues dealt with as well (hopefully by FreeSWITCH, if possible). I would also need to emphasise that FreeSWITCH *cannot* just listen on any (public-facing) interface port in order to receive calls from the external provider, simply because my ISP won't allow it (i.e. I can't just "open" that port and hope that someone will contact me - the connection needs to be initiated by our server for incoming call to get routed through). The purpose of registering this external account is so that both the smart phones (tun0) and the internal net (eth1) users could use this account to make external calls (starting with "0"). Obviously, I need these calls to be routed properly via the external VOIP account. In addition to that, I would also need to receive calls from that external account to a nominated internal one (say on extension 20). Could this be done relatively easily in FreeSWITCH? A question about binding: in order to be able to use both tun0 and eth1 interfaces so that FreeSWITCH serves the calls from both eth1 and tun0 devices/networks, is it possible to instruct it to bind on different IP addresses (on tun0 and eth1 respectively) in order to serve calls to/from these networks? I do *not* wish to bind to 0.0.0.0 - just use the interfaces I am interested in - tun0 and eth1 in my case. If all of the above is possible, do I have to mess about and configure SNAT/DNAT separately and use sip kernel module "helpers" etc (I *do* have the routing set up properly though - that I am sure works 100%) or is FreeSWITCH perfectly capable of setting it all up for me? One final query: is there a decent distribution source package of FreeSWITCH, which I could take and then use it to build a binary package (I use Fedora on all of our machines). I need this, because all of our images are built using the rpm packaging system and I cannot just "fetch the latest git", compile and install FreeSWITCH - I need it packaged, so that the image builder picks it up and distributes it to the machine concerned. I checked the SUSE repository (as instructed on the FreeSWITCH wiki), but found 2 packages, both of which cannot be build due to errors! Many thanks in advance! From msc at freeswitch.org Thu Feb 9 03:39:26 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 8 Feb 2012 16:39:26 -0800 Subject: [Freeswitch-users] Help with attended transfer / att_xfer In-Reply-To: <03d901cce6bf$bcc682d0$36538870$@launch3.net> References: <029e01cce5ef$aaadc0c0$00094240$@launch3.net> <399BEF93D8F8443BA14E1A90B8443AC0@freeswitch.org> <03d901cce6bf$bcc682d0$36538870$@launch3.net> Message-ID: I agree w/ jmesquita - this probably needs a patch, although I'd need to dig further to see if there's a non-patchy workaround. As far as FusionPBX goes, you should be able to do a git pull and make install to keep yourself updated. Just confirm with Mark Crane (IRC: mcrane) or one of the other fusionpbx guys about any caveats with things like the target install directory, etc. -MC On Wed, Feb 8, 2012 at 4:14 PM, Brett Wilson wrote: > Thanks guys I am using the FS att_xfer along with the meta bind. I tried > attended transfer on my gxp2100 phones and it works but I am under the > impression that the phone is actually handling the sip switching itself for > that functionality, and our MOH was not being played while the two internal > parties were speaking, before putting through the transfer. It was just > dead silence which did not sit well with our customers. So I switched to > the FS feature, and it works great. Only problem is that we don?t know when > to start speaking after the transfer!**** > > ** ** > > Also I have another question. I have FS installed via the FusionPBX linux > disk image. What is the best way to keep FS updated? Do I need to pull from > git and build from source or what?**** > > ** ** > > *Brett Wilson* > > *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: Blogger-logo][image: > Description: Description: FaceBook-Logo][image: > Description: Description: Twitter-Logo] > **** > > ** ** > > *From:* Jo?o Mesquita [mailto:jmesquita at freeswitch.org] > *Sent:* Wednesday, February 08, 2012 5:08 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help with attended transfer / att_xfer** > ** > > ** ** > > I believe he is using the feature code, therefore att_xfer so this would > require a patch of some kind. I am this on my todo list which is quite > large nowadays and I don't even know if it is feasible yet. I can take a > look at the code later on.**** > > ** ** > > Regards,**** > > ** ** > > -- **** > > Jo?o Mesquita**** > > Sent with Sparrow **** > > ** ** > > On Wednesday, February 8, 2012 at 2:40 PM, Michael Collins wrote:**** > > Brett, > > No worries on the double-post - we have a first-time sender moderation > filter so I had to allow your messages through. You should be go to go from > here on out. > > As to your question, are you using the default configuration or are you > doing something different? Also, how are you executing the transfer - using > the transfer button on the telephone or the *1 feature code? > > Thanks, > MC**** > > On Tue, Feb 7, 2012 at 3:24 PM, Brett Wilson wrote: > > **** > > Hey guys,**** > > I need a pointer. I?m trying to get some kind of functionality where after > a attended transfer is completed, ie. The ouside caller is connected to the > 2nd phone after the 1st phone has hung up or chosen to continue the > transfer. I would like a beep to sound for the 2nd phone or for both > legs, just to let the recipient of the transfer know that it has gone > through and they can start talking.**** > > Any ideas?**** > > **** > > Thanks**** > > **** > > **** > > *Brett Wilson***** > > *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: Blogger-logo][image: > Description: Description: FaceBook-Logo][image: > Description: Description: Twitter-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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________* > *** > > Professional FreeSWITCH Consulting Services:**** > > 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**** > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users**** > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users** > ** > > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/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/jpeg Size: 1715 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/e4259107/attachment-0005.jpe From msc at freeswitch.org Thu Feb 9 03:59:55 2012 From: msc at freeswitch.org (Michael Collins) Date: Wed, 8 Feb 2012 16:59:55 -0800 Subject: [Freeswitch-users] [newbie] questions In-Reply-To: <4F330A31.4060804@privatedemail.net> References: <4F330A31.4060804@privatedemail.net> Message-ID: Welcome to FreeSWITCH! Comments inline... -MC On Wed, Feb 8, 2012 at 3:50 PM, Josh wrote: > I am considering using FreeSWITCH as a replacement of our (rather > crappy, outdated and buggy) sip proxy system, and have the following > system setup on the server: > > * eth0 faces the external Internet interface, *but* it does not have a > public IP address (it has a private one given to it by my ISP's DHCP > server); > * eth1 faces our internal network (say 10.1.1.0/24); > * tun0 serves all mobile smartphones and connects to the internal > network (it has a different ip range, say 10.1.2.0/24) - they are all > connected via the Internet using OpenVPN; > > I would like to configure FreeSWITCH as a local registrar and ask it to > handle internal calls between ourselves (eth1<->tun0). I would also like > to use one external VOIP provider to which FreeSWITCH registers on > startup. This also needs to have all NAT-related issues dealt with as > well (hopefully by FreeSWITCH, if possible). I would also need to > emphasise that FreeSWITCH *cannot* just listen on any (public-facing) > interface port in order to receive calls from the external provider, > simply because my ISP won't allow it (i.e. I can't just "open" that port > and hope that someone will contact me - the connection needs to be > initiated by our server for incoming call to get routed through). > But calls will be sent to FreeSWITCH by some device, correct? If it's good old-fashioned SIP then FreeSWITCH will handle it just fine. > > The purpose of registering this external account is so that both the > smart phones (tun0) and the internal net (eth1) users could use this > account to make external calls (starting with "0"). Obviously, I need > these calls to be routed properly via the external VOIP account. In > addition to that, I would also need to receive calls from that external > account to a nominated internal one (say on extension 20). > > Could this be done relatively easily in FreeSWITCH? > "Relatively?" Of course! It's relatively easy for someone with some experience. I highly recommend that you ask consulting at freeswitch.org for professional assistance if you are not comfortable doing this all by yourself. > > A question about binding: in order to be able to use both tun0 and eth1 > interfaces so that FreeSWITCH serves the calls from both eth1 and tun0 > devices/networks, is it possible to instruct it to bind on different IP > addresses (on tun0 and eth1 respectively) in order to serve calls > to/from these networks? I do *not* wish to bind to 0.0.0.0 - just use > the interfaces I am interested in - tun0 and eth1 in my case. > Yes, FreeSWITCH can bind to multiple interfaces. In FreeSWITCH lingo that would mean that you set up a separate SIP profile for each interface. (In fact, you can have more than one SIP profile on a given interface since the profile is a unique combo of IP addr and port number.) > > If all of the above is possible, do I have to mess about and configure > SNAT/DNAT separately and use sip kernel module "helpers" etc (I *do* > have the routing set up properly though - that I am sure works 100%) or > is FreeSWITCH perfectly capable of setting it all up for me? > "Some assembly required." :D FreeSWITCH can do some stuff for you, but you definitely need to make sure that your NAT is not behaving badly, like having a SIP ALG. > > One final query: is there a decent distribution source package of > FreeSWITCH, which I could take and then use it to build a binary package > (I use Fedora on all of our machines). I need this, because all of our > images are built using the rpm packaging system and I cannot just "fetch > the latest git", compile and install FreeSWITCH - I need it packaged, so > that the image builder picks it up and distributes it to the machine > concerned. I checked the SUSE repository (as instructed on the > FreeSWITCH wiki), but found 2 packages, both of which cannot be build > due to errors! > I'll have to defer to Ken Rice on this one. I know he's working on RPMs for FreeSWITCH but I think it's all RedHat right now. > > Many thanks in advance! > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/f3fd94ae/attachment.html From spautz2 at telefaks.biz Thu Feb 9 04:37:57 2012 From: spautz2 at telefaks.biz (David) Date: Thu, 09 Feb 2012 02:37:57 +0100 Subject: [Freeswitch-users] Freeswitch opened multiple outbound channels for one call with different uuid In-Reply-To: <6CCD037B-A934-4541-8553-9170FB7182C5@freeswitch.org> References: <4F323042.8080603@telefaks.biz> <4F326E84.6030302@telefaks.biz> <6CCD037B-A934-4541-8553-9170FB7182C5@freeswitch.org> Message-ID: <4F332375.7020404@telefaks.biz> There is only one sip_registration for every sip_user Am 08.02.2012 21:21, schrieb Brian West: > select count(*) from sip_registrations where sip_user='XXX'; > > /b > > On Feb 8, 2012, at 6:45 AM, David wrote: > >> It was found that freeswitch performs multiple INVITES. But why? > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/51829132/attachment.html From spautz2 at telefaks.biz Thu Feb 9 04:50:37 2012 From: spautz2 at telefaks.biz (David) Date: Thu, 09 Feb 2012 02:50:37 +0100 Subject: [Freeswitch-users] Freeswitch opened multiple outbound channels for one call with different uuid In-Reply-To: References: <4F323042.8080603@telefaks.biz> <4F326E84.6030302@telefaks.biz> Message-ID: <4F33266D.2050300@telefaks.biz> I cannot find any network problems. interface: any filter: (ip or ip6) and ( net 192.168.1.12 or net 192.168.1.11 ) # U 192.168.1.12:2048 -> 192.168.1.4:5060 INVITE sip:240 at entwick1;user=phone SIP/2.0. Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-v4ybiqjk8t4e;rport. From: "User 220" ;tag=vx1czcrfys. To: . Call-ID: 3c28bad818ad-q0ms60zsfpst. CSeq: 1 INVITE. Max-Forwards: 70. Contact: ;reg-id=1. P-Key-Flags: keys="3". User-Agent: snom300/7.3.30. Accept: application/sdp. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO. Allow-Events: talk, hold, refer, call-info. Supported: timer, 100rel, replaces, from-change. Session-Expires: 3600;refresher=uas. Min-SE: 90. Content-Type: application/sdp. Content-Length: 218. . v=0. o=root 930692018 930692018 IN IP4 192.168.1.12. s=call. c=IN IP4 192.168.1.12. t=0 0. m=audio 49908 RTP/AVP 8 101. a=rtpmap:8 pcma/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. # U 192.168.1.4:5060 -> 192.168.1.12:2048 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-v4ybiqjk8t4e;rport=2048. From: "User 220" ;tag=vx1czcrfys. To: . Call-ID: 3c28bad818ad-q0ms60zsfpst. CSeq: 1 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 -0600. Content-Length: 0. . # U 192.168.1.4:5060 -> 192.168.1.12:2048 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-v4ybiqjk8t4e;rport=2048. From: "User 220" ;tag=vx1czcrfys. To: ;tag=rFZa6489Z27rK. Call-ID: 3c28bad818ad-q0ms60zsfpst. CSeq: 1 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 -0600. Accept: application/sdp. 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. Proxy-Authenticate: Digest realm="entwick1", nonce="b755765a-a341-4070-a981-dff43e8f7d08", algorithm=MD5, qop="auth". Content-Length: 0. . # U 192.168.1.12:2048 -> 192.168.1.4:5060 ACK sip:240 at entwick1;user=phone SIP/2.0. Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-v4ybiqjk8t4e;rport. From: "User 220" ;tag=vx1czcrfys. To: ;tag=rFZa6489Z27rK. Call-ID: 3c28bad818ad-q0ms60zsfpst. CSeq: 1 ACK. Max-Forwards: 70. Contact: ;reg-id=1. Content-Length: 0. . # U 192.168.1.12:2048 -> 192.168.1.4:5060 INVITE sip:240 at entwick1;user=phone SIP/2.0. Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-eyrl6ghugshz;rport. From: "User 220" ;tag=vx1czcrfys. To: . Call-ID: 3c28bad818ad-q0ms60zsfpst. CSeq: 2 INVITE. Max-Forwards: 70. Contact: ;reg-id=1. P-Key-Flags: keys="3". User-Agent: snom300/7.3.30. Accept: application/sdp. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO. Allow-Events: talk, hold, refer, call-info. Supported: timer, 100rel, replaces, from-change. Session-Expires: 3600;refresher=uas. Min-SE: 90. Proxy-Authorization: Digest username="220",realm="entwick1",nonce="b755765a-a341-4070-a981-dff43e8f7d08",uri="sip:240 at entwick1;user=phone",qop=auth,nc=00000001,cnonce="408b0a1e",response="3261384c8f62187a6486c1fee434d06a",algorithm=MD5. Content-Type: application/sdp. Content-Length: 218. . v=0. o=root 930692018 930692018 IN IP4 192.168.1.12. s=call. c=IN IP4 192.168.1.12. t=0 0. m=audio 49908 RTP/AVP 8 101. a=rtpmap:8 pcma/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. a=sendrecv. # U 192.168.1.4:5060 -> 192.168.1.12:2048 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-eyrl6ghugshz;rport=2048. From: "User 220" ;tag=vx1czcrfys. To: . Call-ID: 3c28bad818ad-q0ms60zsfpst. CSeq: 2 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 -0600. Content-Length: 0. . # U 192.168.1.4:5060 -> 192.168.1.12:2048 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-eyrl6ghugshz;rport=2048. From: "User 220" ;tag=vx1czcrfys. To: ;tag=Srr37ZSDXByBF. Call-ID: 3c28bad818ad-q0ms60zsfpst. CSeq: 2 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 -0600. Accept: application/sdp. 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-Length: 0. Remote-Party-ID: "240" ;party=calling;privacy=off;screen=no. . # U 192.168.1.4:5060 -> 192.168.1.11:5060 INVITE sip:240 at 192.168.1.11:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK4DN4eH402KHmF. Max-Forwards: 69. From: "User 220" ;tag=t1Hv9taHtmmya. To: . Call-ID: ad561287-cd61-122f-8d91-001966eeb846. CSeq: 24039705 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 -0600. 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. Session-Expires: 120;refresher=uac. Min-SE: 120. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 233. P-Key-Flags: keys="3". X-FS-Support: update_display,send_info. Remote-Party-ID: "User 220" ;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1328729538 1328729539 IN IP4 192.168.1.4. s=FreeSWITCH. c=IN IP4 192.168.1.4. t=0 0. m=audio 22000 RTP/AVP 8 0 98 3 18 101 13. a=rtpmap:98 SPEEX/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. # U 192.168.1.4:5060 -> 192.168.1.11:5060 INVITE sip:240 at 192.168.1.11:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. Max-Forwards: 69. From: "User 220" ;tag=UaBNBpUmQXaHp. To: . Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. CSeq: 24039705 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 -0600. 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. Session-Expires: 120;refresher=uac. Min-SE: 120. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 233. P-Key-Flags: keys="3". X-FS-Support: update_display,send_info. Remote-Party-ID: "User 220" ;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1328731870 1328731871 IN IP4 192.168.1.4. s=FreeSWITCH. c=IN IP4 192.168.1.4. t=0 0. m=audio 19668 RTP/AVP 8 0 98 3 18 101 13. a=rtpmap:98 SPEEX/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. # U 192.168.1.4:5060 -> 192.168.1.11:5060 INVITE sip:240 at 192.168.1.11:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK6Z7Nj757v5XSp. Max-Forwards: 69. From: "User 220" ;tag=vK4DDHcrm603H. To: . Call-ID: ad5660f4-cd61-122f-8d91-001966eeb846. CSeq: 24039705 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 -0600. 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. Session-Expires: 120;refresher=uac. Min-SE: 120. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 233. P-Key-Flags: keys="3". X-FS-Support: update_display,send_info. Remote-Party-ID: "User 220" ;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1328721426 1328721427 IN IP4 192.168.1.4. s=FreeSWITCH. c=IN IP4 192.168.1.4. t=0 0. m=audio 30112 RTP/AVP 8 0 98 3 18 101 13. a=rtpmap:98 SPEEX/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. # U 192.168.1.4:5060 -> 192.168.1.11:5060 INVITE sip:240 at 192.168.1.11:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK780em2pBtemcj. Max-Forwards: 69. From: "User 220" ;tag=XvX6ecXUHFQpD. To: . Call-ID: ad566b4a-cd61-122f-8d91-001966eeb846. CSeq: 24039705 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 -0600. 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. Session-Expires: 120;refresher=uac. Min-SE: 120. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 233. P-Key-Flags: keys="3". X-FS-Support: update_display,send_info. Remote-Party-ID: "User 220" ;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1328730428 1328730429 IN IP4 192.168.1.4. s=FreeSWITCH. c=IN IP4 192.168.1.4. t=0 0. m=audio 21110 RTP/AVP 8 0 98 3 18 101 13. a=rtpmap:98 SPEEX/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. # U 192.168.1.11:5060 -> 192.168.1.4:5060 SIP/2.0 180 Ringing. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK4DN4eH402KHmF. Call-ID: ad561287-cd61-122f-8d91-001966eeb846. CSeq: 24039705 INVITE. From: "User 220" ;tag=t1Hv9taHtmmya. To: ;tag=iN29e6PanzlAC2JA. Contact: . Content-Length: 0. . # U 192.168.1.11:5060 -> 192.168.1.4:5060 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. CSeq: 24039705 INVITE. From: "User 220" ;tag=UaBNBpUmQXaHp. To: ;tag=vrAASGKzlwRxb5pm. Contact: . Content-Length: 0. . # U 192.168.1.4:5060 -> 192.168.1.11:5060 ACK sip:240 at 192.168.1.11:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. Max-Forwards: 69. From: "User 220" ;tag=UaBNBpUmQXaHp. To: ;tag=vrAASGKzlwRxb5pm. Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. CSeq: 24039705 ACK. Content-Length: 0. . # U 192.168.1.11:5060 -> 192.168.1.4:5060 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK6Z7Nj757v5XSp. Call-ID: ad5660f4-cd61-122f-8d91-001966eeb846. CSeq: 24039705 INVITE. From: "User 220" ;tag=vK4DDHcrm603H. To: ;tag=2Urps2fOoZQSDiid. Contact: . Content-Length: 0. . # U 192.168.1.4:5060 -> 192.168.1.11:5060 ACK sip:240 at 192.168.1.11:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK6Z7Nj757v5XSp. Max-Forwards: 69. From: "User 220" ;tag=vK4DDHcrm603H. To: ;tag=2Urps2fOoZQSDiid. Call-ID: ad5660f4-cd61-122f-8d91-001966eeb846. CSeq: 24039705 ACK. Content-Length: 0. . # U 192.168.1.4:5060 -> 192.168.1.11:5060 INVITE sip:240 at 192.168.1.11:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK780em2pBtemcj. Max-Forwards: 69. From: "User 220" ;tag=XvX6ecXUHFQpD. To: . Call-ID: ad566b4a-cd61-122f-8d91-001966eeb846. CSeq: 24039705 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 -0600. 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. Session-Expires: 120;refresher=uac. Min-SE: 120. Content-Type: application/sdp. Content-Disposition: session. Content-Length: 233. P-Key-Flags: keys="3". X-FS-Support: update_display,send_info. Remote-Party-ID: "User 220" ;party=calling;screen=yes;privacy=off. . v=0. o=FreeSWITCH 1328730428 1328730429 IN IP4 192.168.1.4. s=FreeSWITCH. c=IN IP4 192.168.1.4. t=0 0. m=audio 21110 RTP/AVP 8 0 98 3 18 101 13. a=rtpmap:98 SPEEX/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=ptime:20. # U 192.168.1.11:5060 -> 192.168.1.4:5060 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK780em2pBtemcj. Call-ID: ad566b4a-cd61-122f-8d91-001966eeb846. CSeq: 24039705 INVITE. From: "User 220" ;tag=XvX6ecXUHFQpD. To: ;tag=qh5PsDcfdulD0LlQ. Contact: . Content-Length: 0. . # U 192.168.1.4:5060 -> 192.168.1.11:5060 ACK sip:240 at 192.168.1.11:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK780em2pBtemcj. Max-Forwards: 69. From: "User 220" ;tag=XvX6ecXUHFQpD. To: ;tag=qh5PsDcfdulD0LlQ. Call-ID: ad566b4a-cd61-122f-8d91-001966eeb846. CSeq: 24039705 ACK. Content-Length: 0. . # U 192.168.1.11:5060 -> 192.168.1.4:5060 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. CSeq: 24039705 INVITE. From: "User 220" ;tag=UaBNBpUmQXaHp. To: ;tag=vrAASGKzlwRxb5pm. Contact: . Content-Length: 0. . # U 192.168.1.4:5060 -> 192.168.1.11:5060 ACK sip:240 at 192.168.1.11:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. Max-Forwards: 69. From: "User 220" ;tag=UaBNBpUmQXaHp. To: ;tag=vrAASGKzlwRxb5pm. Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. CSeq: 24039705 ACK. Content-Length: 0. . # U 192.168.1.11:5060 -> 192.168.1.4:5060 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. CSeq: 24039705 INVITE. From: "User 220" ;tag=UaBNBpUmQXaHp. To: ;tag=vrAASGKzlwRxb5pm. Contact: . Content-Length: 0. . # U 192.168.1.4:5060 -> 192.168.1.11:5060 ACK sip:240 at 192.168.1.11:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. Max-Forwards: 69. From: "User 220" ;tag=UaBNBpUmQXaHp. To: ;tag=vrAASGKzlwRxb5pm. Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. CSeq: 24039705 ACK. Content-Length: 0. . # U 192.168.1.4:5060 -> 192.168.1.11:5060 OPTIONS sip:240 at 192.168.1.11:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK8Ht7NX7eQQaZD. Max-Forwards: 70. From: ;tag=y5pZg7DZerD9r. To: . Call-ID: b0fa8901-cd61-122f-8d91-001966eeb846. CSeq: 24029141 OPTIONS. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 -0600. 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-Length: 0. . # U 192.168.1.4:5060 -> 192.168.1.12:2048 OPTIONS sip:220 at 192.168.1.12:2048;line=9yhon28z SIP/2.0. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK9tK0Qrrjm00HS. Max-Forwards: 70. From: ;tag=Zegrj2y2B13Um. To: . Call-ID: b0fa9032-cd61-122f-8d91-001966eeb846. CSeq: 24029142 OPTIONS. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 -0600. 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-Length: 0. . # U 192.168.1.12:2048 -> 192.168.1.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.4;rport=5060;branch=z9hG4bK9tK0Qrrjm00HS. From: ;tag=Zegrj2y2B13Um. To: . Call-ID: b0fa9032-cd61-122f-8d91-001966eeb846. CSeq: 24029142 OPTIONS. Contact: ;reg-id=1. User-Agent: snom300/7.3.30. Accept-Language: en. Accept: application/sdp. Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO. Allow-Events: talk, hold, refer, call-info. Supported: timer, 100rel, replaces, from-change. Content-Length: 0. . # U 192.168.1.11:5060 -> 192.168.1.4:5060 SIP/2.0 486 Busy Here. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK8Ht7NX7eQQaZD. Call-ID: b0fa8901-cd61-122f-8d91-001966eeb846. CSeq: 24029141 OPTIONS. From: ;tag=y5pZg7DZerD9r. To: ;tag=4rjgSs2dsNvcVyjW. Contact: . Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, NOTIFY, INFO, PRACK, UPDATE. Accept: application/sdp. Accept-Encoding: identity. Accept-Language: en. Supported: 100rel, replaces. Content-Type: application/sdp. Content-Length: 267. . v=0. o=240 32101693 45791413 IN IP4 192.168.1.11. s=SIP CALL. c=IN IP4 192.168.1.11. t=0 0. m=audio 6000 RTP/AVP 8 0 3 18 101. a=rtpmap:8 PCMA/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:3 GSM/8000. a=rtpmap:18 G729/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 192.168.1.12:2048 -> 192.168.1.4:5060 CANCEL sip:240 at entwick1;user=phone SIP/2.0. Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-eyrl6ghugshz;rport. From: "User 220" ;tag=vx1czcrfys. To: . Call-ID: 3c28bad818ad-q0ms60zsfpst. CSeq: 2 CANCEL. Max-Forwards: 70. Reason: SIP;cause=487;text="Request terminated by user". Content-Length: 0. . # U 192.168.1.4:5060 -> 192.168.1.12:2048 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-eyrl6ghugshz;rport=2048. From: "User 220" ;tag=vx1czcrfys. To: ;tag=Srr37ZSDXByBF. Call-ID: 3c28bad818ad-q0ms60zsfpst. CSeq: 2 CANCEL. Content-Length: 0. . # U 192.168.1.4:5060 -> 192.168.1.12:2048 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-eyrl6ghugshz;rport=2048. From: "User 220" ;tag=vx1czcrfys. To: ;tag=Srr37ZSDXByBF. Call-ID: 3c28bad818ad-q0ms60zsfpst. CSeq: 2 INVITE. User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 -0600. 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-Length: 0. . # U 192.168.1.4:5060 -> 192.168.1.11:5060 CANCEL sip:240 at 192.168.1.11:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK4DN4eH402KHmF. Max-Forwards: 69. From: "User 220" ;tag=t1Hv9taHtmmya. To: . Call-ID: ad561287-cd61-122f-8d91-001966eeb846. CSeq: 24039705 CANCEL. Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL". Content-Length: 0. . # U 192.168.1.11:5060 -> 192.168.1.4:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK4DN4eH402KHmF. Call-ID: ad561287-cd61-122f-8d91-001966eeb846. CSeq: 24039705 CANCEL. From: "User 220" ;tag=t1Hv9taHtmmya. To: ;tag=iN29e6PanzlAC2JA. Contact: . Content-Length: 0. . # U 192.168.1.11:5060 -> 192.168.1.4:5060 SIP/2.0 487 Request Terminated. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK4DN4eH402KHmF. Call-ID: ad561287-cd61-122f-8d91-001966eeb846. CSeq: 24039705 INVITE. From: "User 220" ;tag=t1Hv9taHtmmya. To: ;tag=iN29e6PanzlAC2JA. Contact: . Content-Length: 0. . # U 192.168.1.4:5060 -> 192.168.1.11:5060 ACK sip:240 at 192.168.1.11:5060 SIP/2.0. Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK4DN4eH402KHmF. Max-Forwards: 69. From: "User 220" ;tag=t1Hv9taHtmmya. To: ;tag=iN29e6PanzlAC2JA. Call-ID: ad561287-cd61-122f-8d91-001966eeb846. CSeq: 24039705 ACK. Content-Length: 0. . # Am 08.02.2012 18:36, schrieb Michael Collins: > > It was found that freeswitch performs multiple INVITES. But why? > > > One reason why multiple invites are sent out is if FreeSWITCH doesn't > receive anything back. This could mean the far end is down for some > reason or that there is a network issue between FS and the other > endpoint. NAT is a common issue in this sort of configuration. I > recommend that you look at SIP traces on each end of the SIP dialog. > > -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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/6b369e81/attachment-0001.html From jmesquita at freeswitch.org Thu Feb 9 05:30:20 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Wed, 8 Feb 2012 23:30:20 -0300 Subject: [Freeswitch-users] Help with attended transfer / att_xfer In-Reply-To: References: <029e01cce5ef$aaadc0c0$00094240$@launch3.net> <399BEF93D8F8443BA14E1A90B8443AC0@freeswitch.org> <03d901cce6bf$bcc682d0$36538870$@launch3.net> Message-ID: <52A04E7E1B414CD0AF0D33A279B68C4F@freeswitch.org> If I am reading code correctly, you should be able to insert the required play code around line #2105 of mod_dptools.c I might add this patch tomorrow because I believe it is a pretty cool feature to have with a configurable tone to be played. Regards, -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Wednesday, February 8, 2012 at 9:39 PM, Michael Collins wrote: > I agree w/ jmesquita - this probably needs a patch, although I'd need to dig further to see if there's a non-patchy workaround. > > As far as FusionPBX goes, you should be able to do a git pull and make install to keep yourself updated. Just confirm with Mark Crane (IRC: mcrane) or one of the other fusionpbx guys about any caveats with things like the target install directory, etc. > > -MC > > On Wed, Feb 8, 2012 at 4:14 PM, Brett Wilson wrote: > > > > Thanks guys I am using the FS att_xfer along with the meta bind. I tried attended transfer on my gxp2100 phones and it works but I am under the impression that the phone is actually handling the sip switching itself for that functionality, and our MOH was not being played while the two internal parties were speaking, before putting through the transfer. It was just dead silence which did not sit well with our customers. So I switched to the FS feature, and it works great. Only problem is that we don?t know when to start speaking after the transfer! > > > > > > > > > > > > Also I have another question. I have FS installed via the FusionPBX linux disk image. What is the best way to keep FS updated? Do I need to pull from git and build from source or what? > > > > > > > > > > > > Brett Wilson > > > > > > Launch 3 Ventures, LLC > > > > > > 134 Myer Street > > > > > > Hackensack, NJ 07601 > > > > > > Phone: 877.878.9134 (tel:877.878.9134) > > Fax: 646.536.3866 (tel:646.536.3866) > > > > > > Email: Brett.Wilson at launch3.net (mailto:Brett.Wilson at launch3.net) > > > > > > AOL IM: Brett.Wilson at launch3.net (mailto:Brett.Wilson at launch3.net) > > > > > > www.Launch3.net > > > > > > www.Launch3telecom.com (http://www.launch3telecom.com/) > > > > > > *************************** > > > > > > > > > > > > > > > > > > > > From: Jo?o Mesquita [mailto:jmesquita at freeswitch.org] > > Sent: Wednesday, February 08, 2012 5:08 PM > > To: FreeSWITCH Users Help > > Subject: Re: [Freeswitch-users] Help with attended transfer / att_xfer > > > > > > > > > > > > I believe he is using the feature code, therefore att_xfer so this would require a patch of some kind. I am this on my todo list which is quite large nowadays and I don't even know if it is feasible yet. I can take a look at the code later on. > > > > > > > > > > > > > > Regards, > > > > > > > > > > > > > > > > > > -- > > > > > > > > Jo?o Mesquita > > > > > > > > Sent with Sparrow (http://www.sparrowmailapp.com/?sig) > > > > > > > > > > > > > > > > On Wednesday, February 8, 2012 at 2:40 PM, Michael Collins wrote: > > > > > > Brett, > > > > > > No worries on the double-post - we have a first-time sender moderation filter so I had to allow your messages through. You should be go to go from here on out. > > > > > > As to your question, are you using the default configuration or are you doing something different? Also, how are you executing the transfer - using the transfer button on the telephone or the *1 feature code? > > > > > > Thanks, > > > MC > > > > > > > > > On Tue, Feb 7, 2012 at 3:24 PM, Brett Wilson wrote: > > > > > > > > > > > > > > > Hey guys, > > > > > > > > > I need a pointer. I?m trying to get some kind of functionality where after a attended transfer is completed, ie. The ouside caller is connected to the 2nd phone after the 1st phone has hung up or chosen to continue the transfer. I would like a beep to sound for the 2nd phone or for both legs, just to let the recipient of the transfer know that it has gone through and they can start talking. > > > > > > > > > Any ideas? > > > > > > > > > > > > > > > > > > Thanks > > > > > > > > > > > > > > > > > > > > > > > > > > > Brett Wilson > > > > > > > > > Launch 3 Ventures, LLC > > > > > > > > > 134 Myer Street > > > > > > > > > Hackensack, NJ 07601 > > > > > > > > > Phone: 877.878.9134 (tel:877.878.9134) > > > Fax: 646.536.3866 (tel:646.536.3866) > > > > > > > > > Email: Brett.Wilson at launch3.net (mailto:Brett.Wilson at launch3.net) > > > > > > > > > AOL IM: Brett.Wilson at launch3.net (mailto:Brett.Wilson at launch3.net) > > > > > > > > > www.Launch3.net (http://www.Launch3.net) > > > > > > > > > www.Launch3telecom.com (http://www.launch3telecom.com/) > > > > > > > > > *************************** > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > 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 > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > > > > > > > > > > > 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 > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > 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 > > > > _________________________________________________________________________ > 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... 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Name: image003.jpg Type: image/jpeg Size: 1715 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/dc5def90/attachment-0005.jpg From aromberg at gmail.com Thu Feb 9 05:30:46 2012 From: aromberg at gmail.com (aromberg at gmail.com) Date: Wed, 8 Feb 2012 20:30:46 -0600 Subject: [Freeswitch-users] Choppy Audio Recordings Message-ID: Hello, I'm having a persistent problem across multiple installations. All are running GIT Head. Windows 2008 R2 x64 Dual Xeon 2.8, 4G of ram, Hyper-V instance, IIS running a few sites, nothing major. Debian 6, dual core 2.13GHz Xeon, 512mb dedicated 1gb burst, OpenVZ instance, freeswitch + small apache instance CentOS 5, 8 core 2.27 Xeon, 512MB RAM, Xen instance, freeswitch + apache (doing nothing) Centos6, single core 3.33Ghz 512MB ram, dedicated machine (no instance), nohz setting enabled per the wiki, only freeswitch running Any recordings come out choppy and slow, no matter what I do. Doesn't matter if it was WAV or MP3 using mod_shout. The endpoint I'm using is a Linksys SPA942. Any pointers would be helpful at this point. Thanks, Adam From brian at freeswitch.org Thu Feb 9 06:44:27 2012 From: brian at freeswitch.org (Brian West) Date: Wed, 8 Feb 2012 21:44:27 -0600 Subject: [Freeswitch-users] Choppy Audio Recordings In-Reply-To: References: Message-ID: <9D59C823-7B8A-451F-8CDD-D5F46335903E@freeswitch.org> do you have the rtp time set to 0.020 or is it default 0.030? /b On Feb 8, 2012, at 8:30 PM, aromberg at gmail.com wrote: > > The endpoint I'm using is a Linksys SPA942. -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120208/1e587a55/attachment.html From mantung at commverge.com Thu Feb 9 07:03:02 2012 From: mantung at commverge.com (Man) Date: Thu, 09 Feb 2012 12:03:02 +0800 Subject: [Freeswitch-users] System shown no route but connected with false caller ID Message-ID: <4F334576.4030404@commverge.com> User registered to the system and try to make call to PSTN network but system show "No Route, Aborting" at first and connect the call right after it. The destination number ring wrong caller ID displayed. 40% calls happened with this issue. Any possible reason? FreeSWITCH Version 1.0.head (git-7f5b8fb 2012-02-02 20-41-15 -0600) Checked the CDR both normal and fault call are no different. My dialplan is simply bridge the call to the voice gateway: Thanks a lot. Armand -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/37378330/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: Error.JPG Type: image/jpeg Size: 90229 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/37378330/attachment-0001.jpe From yehavi.bourvine at gmail.com Thu Feb 9 08:28:49 2012 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Thu, 9 Feb 2012 07:28:49 +0200 Subject: [Freeswitch-users] presence on latest GIT In-Reply-To: References: Message-ID: Hello Vik, BLF behaviour is somewhat erratic since major changes done after 15-Oct-2011. I am working with Anthony to track it down (see FS-3794). It happens mostly on our production system rather than the test system, so I am quite limited in testing the fixes and it takes time... Here is a sample config of phone 11111 doing BLF on 22222: Note that *ThirdPartyName* is *not* used in the REG section. Regards, __Yehavi: 2012/2/9 Vik Killa > Would anyone be willing to share their working BLF polycom configs? > > On Wed, Feb 8, 2012 at 4:56 PM, Anthony Minessale > wrote: > > It should be working but you can use that data to see why. > > If you enable sip trace too and capture the logs, it can be used to > > check for problems > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/db5e3737/attachment.html From aromberg at gmail.com Thu Feb 9 08:33:34 2012 From: aromberg at gmail.com (aromberg at gmail.com) Date: Wed, 8 Feb 2012 23:33:34 -0600 Subject: [Freeswitch-users] Choppy Audio Recordings In-Reply-To: <9D59C823-7B8A-451F-8CDD-D5F46335903E@freeswitch.org> References: <9D59C823-7B8A-451F-8CDD-D5F46335903E@freeswitch.org> Message-ID: It was set to the default of 0.030, now it's changed to 0.020 and the problems are getting better. Now, I can get perfect (or close to it) audio on the Hyper-V host, but only thru google voice. Anything else still produces choppy audio. Tested with FlowRoute & TFGW. Suggestions? Thanks, Adam On Wed, Feb 8, 2012 at 9:44 PM, Brian West wrote: > do you have the rtp time set to 0.020 or is it default 0.030? > > /b > > On Feb 8, 2012, at 8:30 PM, aromberg at gmail.com wrote: > > > The endpoint I'm using is a Linksys SPA942. > > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: ? +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > 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 Thu Feb 9 08:51:02 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Thu, 09 Feb 2012 00:51:02 -0500 Subject: [Freeswitch-users] Registration problem after attempting to install mod_opal and updating to latest git pull Message-ID: <00ba01cce6ee$d1422640$73c672c0$@com> After a recent unsuccessful attempt to install mod_opal and updating to a recent git, I now get some odd reports of gateway registration failures, although everything appears to be working ok call-wise for the moment. Doing a 'make uninstall' from /root/opal and updating to the latest git tonight yielded no improvement. Here are some excerpts of failures reported in fs_cli: 2012-02-09 00:12:59.992099 [WARNING] sofia.c:4717 Unregister QS8002 2012-02-09 00:12:59.992099 [WARNING] sofia.c:4730 Ping failed QS8002 with code 900 - count -1/-1/1, state UP 2012-02-09 00:12:59.992099 [WARNING] sofia.c:4717 Unregister HL7612 2012-02-09 00:12:59.992099 [WARNING] sofia.c:4730 Ping failed HL7612 with code 900 - count -1/-1/1, state UP 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4717 Unregister HL7611 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4730 Ping failed HL7611 with code 900 - count -1/-1/1, state UP 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4717 Unregister HL7519 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4730 Ping failed HL7519 with code 900 - count -1/-1/1, state UP 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4717 Unregister CallCentric_T.38 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4730 Ping failed CallCentric_T.38 with code 900 - count -1/-1/1, state UP 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 QS8002 Failed Registration [0], setting retry to 60 seconds. 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 HL7612 Failed Registration [0], setting retry to 90 seconds. 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 HL7611 Failed Registration [0], setting retry to 90 seconds. 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 HL7519 Failed Registration [0], setting retry to 60 seconds. 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 CallCentric_T.38 Failed Registration [0], setting retry to 60 seconds. To give a sense of the timing, the three groups of warnings below are contiguous, from a single fs_cli session: 2012-02-09 00:21:18.984384 [WARNING] sofia.c:4717 Unregister HL7611 2012-02-09 00:21:18.984384 [WARNING] sofia.c:4730 Ping failed HL7611 with code 900 - count -1/-1/1, state UP 2012-02-09 00:21:20.988379 [WARNING] sofia_reg.c:474 HL7611 Failed Registration [0], setting retry to 90 seconds. 2012-02-09 00:21:22.988382 [WARNING] sofia.c:4717 Unregister HL7519 2012-02-09 00:21:22.988382 [WARNING] sofia.c:4730 Ping failed HL7519 with code 900 - count -1/-1/1, state UP 2012-02-09 00:21:24.988318 [WARNING] sofia_reg.c:474 HL7519 Failed Registration [0], setting retry to 60 seconds. 2012-02-09 00:22:00.447778 [WARNING] sofia_reg.c:1422 SIP auth challenge (REGISTER) on sofia profile 'internal' for [226 at qsystemseng.no-ip.org] from ip 74.93.222.182 2012-02-09 00:22:04.147688 [WARNING] sofia.c:4717 Unregister HL7612 2012-02-09 00:22:04.147688 [WARNING] sofia.c:4730 Ping failed HL7612 with code 900 - count -1/-1/1, state UP 2012-02-09 00:22:04.167699 [WARNING] sofia.c:4717 Unregister HL7611 2012-02-09 00:22:04.167699 [WARNING] sofia.c:4730 Ping failed HL7611 with code 900 - count -1/-1/1, state UP 2012-02-09 00:22:04.167699 [WARNING] sofia.c:4717 Unregister CallCentric_T.38 2012-02-09 00:22:04.167699 [WARNING] sofia.c:4730 Ping failed CallCentric_T.38 with code 900 - count -1/-1/1, state UP 2012-02-09 00:22:05.167744 [WARNING] sofia_reg.c:474 HL7611 Failed Registration [0], setting retry to 90 seconds. 2012-02-09 00:22:05.167744 [WARNING] sofia_reg.c:474 CallCentric_T.38 Failed Registration [0], setting retry to 60 seconds. 2012-02-09 00:22:05.167744 [WARNING] sofia.c:4717 Unregister QS8002 2012-02-09 00:22:05.167744 [WARNING] sofia.c:4730 Ping failed QS8002 with code 900 - count -1/-1/1, state UP 2012-02-09 00:22:07.171680 [WARNING] sofia_reg.c:474 QS8002 Failed Registration [0], setting retry to 60 seconds. 2012-02-09 00:22:07.171680 [WARNING] sofia.c:4717 Unregister HL7519 2012-02-09 00:22:07.171680 [WARNING] sofia.c:4730 Ping failed HL7519 with code 900 - count -1/-1/1, state UP One of the gateways is SIPBroker, and calls through that gate starting failing after the update. I realized that the gateway was configured to ping SIPBroker unnecessarily and restarting the gateway would restore service until the ping. I disabled the ping for that gateway and calls kept working after that. Any ideas as to what might be going on? There was no such problem before the FS update/attempt to use mod_opal. I can provide additional/more detailed info if needed. Thanks! Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com From dujinfang at gmail.com Thu Feb 9 09:00:45 2012 From: dujinfang at gmail.com (Seven Du) Date: Thu, 9 Feb 2012 14:00:45 +0800 Subject: [Freeswitch-users] cluecon 2012 date confirm Message-ID: Hi, Had a wonderful time in Cluecon 2011 and would like to confirm the date as it looks like no schedule for 2012 on cluecon.com and I only found a count down on freeswitch.org. So would like to confirm will it be Aug. 6th this year? 179 Days10 Hours2 Minutes37 (http://www.cluecon.com/?fs1) select now() + interval '179 days'; ?column? ------------------------------- 2012-08-06 13:47:33.990137+08 (1 row) 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/20120209/90a666e8/attachment.html From avi at avimarcus.net Thu Feb 9 10:59:06 2012 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 9 Feb 2012 09:59:06 +0200 Subject: [Freeswitch-users] System shown no route but connected with false caller ID In-Reply-To: <4F334576.4030404@commverge.com> References: <4F334576.4030404@commverge.com> Message-ID: Can we see the call on /log 7? You don't have a failover bridge destination? -Avi On Thu, Feb 9, 2012 at 6:03 AM, Man wrote: > User registered to the system and try to make call to PSTN network but > system show "No Route, Aborting" at first and connect the call right after > it. The destination number ring wrong caller ID displayed. 40% calls > happened with this issue. > > Any possible reason? > > FreeSWITCH Version 1.0.head (git-7f5b8fb 2012-02-02 20-41-15 -0600) > > Checked the CDR both normal and fault call are no different. > > My dialplan is simply bridge the call to the voice gateway: > > > > /> > > > > Thanks a lot. > > Armand > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/a7f251b1/attachment-0001.html From rush0621 at yahoo.com.hk Thu Feb 9 08:04:14 2012 From: rush0621 at yahoo.com.hk (liu rush) Date: Thu, 9 Feb 2012 13:04:14 +0800 (SGT) Subject: [Freeswitch-users] Extend feature for HuaWei Softswitch Message-ID: <1328763854.46840.YahooMailNeo@web190804.mail.sg3.yahoo.com> Dear, I am currently working on a project that the situation is?similar?to Hybrid SIPS encryption in below link as I think. http://wiki.freeswitch.org/wiki/SIP_TLS#Hybrid_Encryption? My actual situation is like this.? Since, the HuaWei Softswitch isn't support for SRTP, i suppose Freeswitch can be setup as an intermediary to do it which it will be look like this. IP-phone <-SRTP-> FS <-RTP-> HuaWei Softswitch. But I have no idea how to set it up with Freeswitch (I'm newbie in FS, I jumped into FS for less than a month). Could anyone give me some advice or direction if FS is capable to do it? Regard, Rush -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/a2ef3451/attachment.html From tknchris at gmail.com Thu Feb 9 10:01:12 2012 From: tknchris at gmail.com (chris) Date: Thu, 9 Feb 2012 02:01:12 -0500 Subject: [Freeswitch-users] New to FS Message-ID: Hello, I am new to FS but I'm impressed already with my first experiences with it. I was just wondering about somethins that I've always wanted on asterisk that was possible but always been a pain. Is it possible to do source based routing and route certain SIP extensions/devices to certain outbound trunks? If so how can I do it easily? Thanks for any advice / info, chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/4a861ec6/attachment.html From peter.olsson at visionutveckling.se Thu Feb 9 11:29:44 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 9 Feb 2012 08:29:44 +0000 Subject: [Freeswitch-users] Choppy Audio Recordings In-Reply-To: References: <9D59C823-7B8A-451F-8CDD-D5F46335903E@freeswitch.org> Message-ID: <1FFF97C269757C458224B7C895F35F1503BA1D@cantor.std.visionutv.se> I'm not 100% sure how your setup is. Are you running FS in the actual Host OS, or in a virtual (Hyper-V) machine? If running FS virtual you will get these kinds of troubles, beacuse of not so accurate timing in those setups. If you want things to work without problems, use real hardware. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r aromberg at gmail.com Skickat: den 9 februari 2012 06:34 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Choppy Audio Recordings It was set to the default of 0.030, now it's changed to 0.020 and the problems are getting better. Now, I can get perfect (or close to it) audio on the Hyper-V host, but only thru google voice. Anything else still produces choppy audio. Tested with FlowRoute & TFGW. Suggestions? Thanks, Adam On Wed, Feb 8, 2012 at 9:44 PM, Brian West wrote: > do you have the rtp time set to 0.020 or is it default 0.030? > > /b > > On Feb 8, 2012, at 8:30 PM, aromberg at gmail.com wrote: > > > The endpoint I'm using is a Linksys SPA942. > > > -- > Brian West > FreeSWITCH Solutions, LLC > Phone: +1 (918) 420-9266 > Fax: +1 (918) 420-9267 > brian at freeswitch.org > http://www.freeswitch.org > > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > 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 > 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 FreeSWITCH-users mailing list FreeSWITCH-users at 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:4f335a6732769129577974! From mantung at commverge.com Thu Feb 9 11:31:50 2012 From: mantung at commverge.com (Man) Date: Thu, 09 Feb 2012 16:31:50 +0800 Subject: [Freeswitch-users] System shown no route but connected with false caller ID In-Reply-To: References: <4F334576.4030404@commverge.com> Message-ID: <4F338476.3070700@commverge.com> No failover bridge is configured and I cannot find any similar command in the dialplan directory. The log is a bit long here: Dialplan: sofia/external/3175xxxx at 203.x.x.80 parsing [public->DID_31759999] continue=false Dialplan: sofia/external/3175xxxx at 203.x.x.80 Regex (FAIL) [DID_31759999] destination_number(6578xxxx) =~ /31759999/ break=on-false Dialplan: sofia/external/3175xxxx at 203.x.x.80 parsing [public->IVR2] continue=false Dialplan: sofia/external/3175xxxx at 203.x.x.80 Regex (PASS) [IVR] context(public) =~ /public/ break=on-false Dialplan: sofia/external/3175xxxx at 203.x.x.80 Regex (FAIL) [IVR] destination_number(6578xxxx) =~ /023128002/ break=on-false 2012-02-09 17:09:17.199902 [INFO] switch_core_state_machine.c:177 No Route, Aborting 2012-02-09 17:09:17.199902 [DEBUG] switch_channel.c:2848 (sofia/external/3175xxxx at 203.x.x.80) Callstate Change RINGING -> HANGUP 2012-02-09 17:09:17.199902 [NOTICE] switch_core_state_machine.c:178 Hangup sofia/external/3175xxxx at 203.x.x.80 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2012-02-09 17:09:17.199902 [DEBUG] switch_channel.c:2871 Send signal sofia/external/3175xxxx at 203.x.x.80 [KILL] 2012-02-09 17:09:17.199902 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/3175xxxx at 203.x.x.80 [BREAK] 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:410 (sofia/external/3175xxxx at 203.x.x.80) State ROUTING going to sleep 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:362 (sofia/external/3175xxxx at 203.x.x.80) Running State Change CS_HANGUP 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:602 (sofia/external/3175xxxx at 203.x.x.80) State HANGUP 2012-02-09 17:09:17.199902 [DEBUG] mod_sofia.c:469 Channel sofia/external/3175xxxx at 203.x.x.80 hanging up, cause: NO_ROUTE_DESTINATION 2012-02-09 17:09:17.199902 [DEBUG] mod_sofia.c:534 Responding to INVITE with: 404 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:47 sofia/external/3175xxxx at 203.x.x.80 Standard HANGUP, cause: NO_ROUTE_DESTINATION 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:602 (sofia/external/3175xxxx at 203.x.x.80) State HANGUP going to sleep 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:393 (sofia/external/3175xxxx at 203.x.x.80) State Change CS_HANGUP -> CS_REPORTING 2012-02-09 17:09:17.199902 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/3175xxxx at 203.x.x.80 [BREAK] 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:362 (sofia/external/3175xxxx at 203.x.x.80) Running State Change CS_REPORTING 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:662 (sofia/external/3175xxxx at 203.x.x.80) State REPORTING 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:79 sofia/external/3175xxxx at 203.x.x.80 Standard REPORTING, cause: NO_ROUTE_DESTINATION 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:662 (sofia/external/3175xxxx at 203.x.x.80) State REPORTING going to sleep 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:387 (sofia/external/3175xxxx at 203.x.x.80) State Change CS_REPORTING -> CS_DESTROY 2012-02-09 17:09:17.199902 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/3175xxxx at 203.x.x.80 [BREAK] 2012-02-09 17:09:17.199902 [DEBUG] switch_core_session.c:1380 Session 23374 (sofia/external/3175xxxx at 203.x.x.80) Locked, Waiting on external entities 2012-02-09 17:09:17.199902 [NOTICE] switch_core_session.c:1398 Session 23374 (sofia/external/3175xxxx at 203.x.x.80) Ended 2012-02-09 17:09:17.199902 [NOTICE] switch_core_session.c:1400 Close Channel sofia/external/3175xxxx at 203.x.x.80 [CS_DESTROY] 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:491 (sofia/external/3175xxxx at 203.x.x.80) Callstate Change HANGUP -> DOWN 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:494 (sofia/external/3175xxxx at 203.x.x.80) Running State Change CS_DESTROY 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:504 (sofia/external/3175xxxx at 203.x.x.80) State DESTROY 2012-02-09 17:09:17.199902 [DEBUG] mod_sofia.c:374 sofia/external/3175xxxx at 203.x.x.80 SOFIA DESTROY 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:86 sofia/external/3175xxxx at 203.x.x.80 Standard DESTROY 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:504 (sofia/external/3175xxxx at 203.x.x.80) State DESTROY going to sleep 2012-02-09 17:09:17.327945 [DEBUG] switch_core_io.c:340 Setting BUG Codec PCMU:0 2012-02-09 17:09:17.519934 [INFO] switch_rtp.c:3188 Auto Changing port from 10.0.166.104:16384 to 118.x.x.115:2590 2012-02-09 17:09:20.136096 [DEBUG] switch_core_session.c:875 Send signal sofia/external/6578xxxx at 203.x.x.80 [BREAK] 2012-02-09 17:09:20.136096 [DEBUG] switch_core_session.c:875 Send signal sofia/external/6578xxxx at 203.x.x.80 [BREAK] 2012-02-09 17:09:20.136096 [DEBUG] sofia.c:5512 Channel sofia/external/6578xxxx at 203.x.x.80 entering state [proceeding][183] 2012-02-09 17:09:20.136096 [DEBUG] sofia.c:5523 Remote SDP: v=0 o=CiscoSystemsSIP-GW-UserAgent 1393 6748 IN IP4 203.x.x.80 s=SIP Call c=IN IP4 203.x.x.80 t=0 0 m=audio 17294 RTP/AVP 0 101 c=IN IP4 203.x.x.80 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:4374 Activate Buggy RFC2833 Mode! 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:2919 Set Codec sofia/external/6578xxxx at 203.x.x.80 PCMU/8000 20 ms 160 samples 64000 bits 2012-02-09 17:09:20.136096 [DEBUG] switch_core_codec.c:111 sofia/external/6578xxxx at 203.x.x.80 Original read codec set to PCMU:0 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send payload to 101 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:3171 AUDIO RTP [sofia/external/6578xxxx at 203.x.x.80] 203.x.x.91 port 28814 -> 203.x.x.80 port 17294 codec: 0 ms: 20 2012-02-09 17:09:20.136096 [DEBUG] switch_rtp.c:1669 Not using a timer 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:3435 Set 2833 dtmf send payload to 101 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:3441 Set 2833 dtmf receive payload to 101 2012-02-09 17:09:20.136096 [NOTICE] sofia_glue.c:3945 Pre-Answer sofia/external/6578xxxx at 203.x.x.80! Seem the system treat this outbound call as inbound. It loaded the inbound dialplan. As following entries are only in ../freeswitch/conf/dialplan/public: Dialplan: sofia/external/3175xxxx at 203.x.x.80 parsing [public->DID_31759999] continue=false Dialplan: sofia/external/3175xxxx at 203.x.x.80 Regex (FAIL) [DID_31759999] destination_number(6578xxxx) =~ /31759999/ break=on-false Dialplan: sofia/external/3175xxxx at 203.x.x.80 parsing [public->IVR2] continue=false Dialplan: sofia/external/3175xxxx at 203.x.x.80 Regex (PASS) [IVR] context(public) =~ /public/ break=on-false Dialplan: sofia/external/3175xxxx at 203.x.x.80 Regex (FAIL) [IVR] destination_number(6578xxxx) =~ /023128002/ break=on-false Thank you. Armand On 2012-02-09 3:59 PM, Avi Marcus wrote: > Can we see the call on /log 7? You don't have a failover bridge > destination? > -Avi > > On Thu, Feb 9, 2012 at 6:03 AM, Man > wrote: > > User registered to the system and try to make call to PSTN network > but system show "No Route, Aborting" at first and connect the call > right after it. The destination number ring wrong caller ID > displayed. 40% calls happened with this issue. > > Any possible reason? > > FreeSWITCH Version 1.0.head (git-7f5b8fb 2012-02-02 20-41-15 -0600) > > Checked the CDR both normal and fault call are no different. > > My dialplan is simply bridge the call to the voice gateway: > > > /> > > > > Thanks a lot. > > Armand > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/e1d67be3/attachment-0001.html From avi at avimarcus.net Thu Feb 9 11:51:45 2012 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 9 Feb 2012 10:51:45 +0200 Subject: [Freeswitch-users] System shown no route but connected with false caller ID In-Reply-To: <4F338476.3070700@commverge.com> References: <4F334576.4030404@commverge.com> <4F338476.3070700@commverge.com> Message-ID: If you're seeing unexpected dialplan stuff, then can you show us the entire dialplan parsing log, from start to finish? I'm guessing you have an extra bridge getting pulled in there... -Avi On Thu, Feb 9, 2012 at 10:31 AM, Man wrote: > No failover bridge is configured and I cannot find any similar command in > the dialplan directory. > > The log is a bit long here: > > Dialplan: sofia/external/3175xxxx at 203.x.x.80 parsing > [public->DID_31759999] continue=false > Dialplan: sofia/external/3175xxxx at 203.x.x.80 Regex (FAIL) [DID_31759999] > destination_number(6578xxxx) =~ /31759999/ break=on-false > Dialplan: sofia/external/3175xxxx at 203.x.x.80 parsing [public->IVR2] > continue=false > Dialplan: sofia/external/3175xxxx at 203.x.x.80 Regex (PASS) [IVR] > context(public) =~ /public/ break=on-false > Dialplan: sofia/external/3175xxxx at 203.x.x.80 Regex (FAIL) [IVR] > destination_number(6578xxxx) =~ /023128002/ break=on-false > 2012-02-09 17:09:17.199902 [INFO] switch_core_state_machine.c:177 No > Route, Aborting > 2012-02-09 17:09:17.199902 [DEBUG] switch_channel.c:2848 ( > sofia/external/3175xxxx at 203.x.x.80) Callstate Change RINGING -> HANGUP > 2012-02-09 17:09:17.199902 [NOTICE] switch_core_state_machine.c:178 Hangup > sofia/external/3175xxxx at 203.x.x.80 [CS_ROUTING] [NO_ROUTE_DESTINATION] > 2012-02-09 17:09:17.199902 [DEBUG] switch_channel.c:2871 Send signal > sofia/external/3175xxxx at 203.x.x.80 [KILL] > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_session.c:1180 Send signal > sofia/external/3175xxxx at 203.x.x.80 [BREAK] > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:410 ( > sofia/external/3175xxxx at 203.x.x.80) State ROUTING going to sleep > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:362 ( > sofia/external/3175xxxx at 203.x.x.80) Running State Change CS_HANGUP > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:602 ( > sofia/external/3175xxxx at 203.x.x.80) State HANGUP > 2012-02-09 17:09:17.199902 [DEBUG] mod_sofia.c:469 Channel > sofia/external/3175xxxx at 203.x.x.80 hanging up, cause: NO_ROUTE_DESTINATION > 2012-02-09 17:09:17.199902 [DEBUG] mod_sofia.c:534 Responding to INVITE > with: 404 > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:47 > sofia/external/3175xxxx at 203.x.x.80 Standard HANGUP, cause: > NO_ROUTE_DESTINATION > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:602 ( > sofia/external/3175xxxx at 203.x.x.80) State HANGUP going to sleep > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:393 ( > sofia/external/3175xxxx at 203.x.x.80) State Change CS_HANGUP -> CS_REPORTING > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_session.c:1180 Send signal > sofia/external/3175xxxx at 203.x.x.80 [BREAK] > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:362 ( > sofia/external/3175xxxx at 203.x.x.80) Running State Change CS_REPORTING > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:662 ( > sofia/external/3175xxxx at 203.x.x.80) State REPORTING > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:79 > sofia/external/3175xxxx at 203.x.x.80 Standard REPORTING, cause: > NO_ROUTE_DESTINATION > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:662 ( > sofia/external/3175xxxx at 203.x.x.80) State REPORTING going to sleep > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:387 ( > sofia/external/3175xxxx at 203.x.x.80) State Change CS_REPORTING -> > CS_DESTROY > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_session.c:1180 Send signal > sofia/external/3175xxxx at 203.x.x.80 [BREAK] > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_session.c:1380 Session > 23374 (sofia/external/3175xxxx at 203.x.x.80) Locked, Waiting on external > entities > 2012-02-09 17:09:17.199902 [NOTICE] switch_core_session.c:1398 Session > 23374 (sofia/external/3175xxxx at 203.x.x.80) Ended > 2012-02-09 17:09:17.199902 [NOTICE] switch_core_session.c:1400 Close > Channel sofia/external/3175xxxx at 203.x.x.80 [CS_DESTROY] > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:491 ( > sofia/external/3175xxxx at 203.x.x.80) Callstate Change HANGUP -> DOWN > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:494 ( > sofia/external/3175xxxx at 203.x.x.80) Running State Change CS_DESTROY > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:504 ( > sofia/external/3175xxxx at 203.x.x.80) State DESTROY > 2012-02-09 17:09:17.199902 [DEBUG] mod_sofia.c:374 > sofia/external/3175xxxx at 203.x.x.80 SOFIA DESTROY > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:86 > sofia/external/3175xxxx at 203.x.x.80 Standard DESTROY > 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:504 ( > sofia/external/3175xxxx at 203.x.x.80) State DESTROY going to sleep > 2012-02-09 17:09:17.327945 [DEBUG] switch_core_io.c:340 Setting BUG Codec > PCMU:0 > 2012-02-09 17:09:17.519934 [INFO] switch_rtp.c:3188 Auto Changing port > from 10.0.166.104:16384 to 118.x.x.115:2590 > 2012-02-09 17:09:20.136096 [DEBUG] switch_core_session.c:875 Send signal > sofia/external/6578xxxx at 203.x.x.80 [BREAK] > 2012-02-09 17:09:20.136096 [DEBUG] switch_core_session.c:875 Send signal > sofia/external/6578xxxx at 203.x.x.80 [BREAK] > 2012-02-09 17:09:20.136096 [DEBUG] sofia.c:5512 Channel > sofia/external/6578xxxx at 203.x.x.80 entering state [proceeding][183] > 2012-02-09 17:09:20.136096 [DEBUG] sofia.c:5523 Remote SDP: > v=0 > o=CiscoSystemsSIP-GW-UserAgent 1393 6748 IN IP4 203.x.x.80 > s=SIP Call > c=IN IP4 203.x.x.80 > t=0 0 > m=audio 17294 RTP/AVP 0 101 > c=IN IP4 203.x.x.80 > a=rtpmap:0 PCMU/8000 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > > 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:4374 Activate Buggy > RFC2833 Mode! > 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:4798 Audio Codec Compare > [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] > 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:2919 Set Codec > sofia/external/6578xxxx at 203.x.x.80 PCMU/8000 20 ms 160 samples 64000 bits > 2012-02-09 17:09:20.136096 [DEBUG] switch_core_codec.c:111 > sofia/external/6578xxxx at 203.x.x.80 Original read codec set to PCMU:0 > 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send > payload to 101 > 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:3171 AUDIO RTP [ > sofia/external/6578xxxx at 203.x.x.80] 203.x.x.91 port 28814 -> 203.x.x.80 > port 17294 codec: 0 ms: 20 > 2012-02-09 17:09:20.136096 [DEBUG] switch_rtp.c:1669 Not using a timer > 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:3435 Set 2833 dtmf send > payload to 101 > 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:3441 Set 2833 dtmf receive > payload to 101 > 2012-02-09 17:09:20.136096 [NOTICE] sofia_glue.c:3945 Pre-Answer > sofia/external/6578xxxx at 203.x.x.80! > > Seem the system treat this outbound call as inbound. It loaded the inbound > dialplan. As following entries are only in > ../freeswitch/conf/dialplan/public: > Dialplan: sofia/external/3175xxxx at 203.x.x.80 parsing > [public->DID_31759999] continue=false > Dialplan: sofia/external/3175xxxx at 203.x.x.80 Regex (FAIL) [DID_31759999] > destination_number(6578xxxx) =~ /31759999/ break=on-false > Dialplan: sofia/external/3175xxxx at 203.x.x.80 parsing [public->IVR2] > continue=false > Dialplan: sofia/external/3175xxxx at 203.x.x.80 Regex (PASS) [IVR] > context(public) =~ /public/ break=on-false > Dialplan: sofia/external/3175xxxx at 203.x.x.80 Regex (FAIL) [IVR] > destination_number(6578xxxx) =~ /023128002/ break=on-false > > Thank you. > > Armand > > On 2012-02-09 3:59 PM, Avi Marcus wrote: > > Can we see the call on /log 7? You don't have a failover bridge > destination? > -Avi > > On Thu, Feb 9, 2012 at 6:03 AM, Man wrote: > >> User registered to the system and try to make call to PSTN network but >> system show "No Route, Aborting" at first and connect the call right after >> it. The destination number ring wrong caller ID displayed. 40% calls >> happened with this issue. >> >> Any possible reason? >> >> FreeSWITCH Version 1.0.head (git-7f5b8fb 2012-02-02 20-41-15 -0600) >> >> Checked the CDR both normal and fault call are no different. >> >> My dialplan is simply bridge the call to the voice gateway: >> >> >> >> /> >> >> >> >> Thanks a lot. >> >> Armand >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/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 > > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/b76a1752/attachment-0001.html From gabe at gundy.org Thu Feb 9 12:37:19 2012 From: gabe at gundy.org (Gabriel Gunderson) Date: Thu, 9 Feb 2012 02:37:19 -0700 Subject: [Freeswitch-users] New to FS In-Reply-To: References: Message-ID: On Thu, Feb 9, 2012 at 12:01 AM, chris wrote: > I am new to FS but I'm impressed already with my first experiences with it. > I was just wondering about somethins that I've always wanted on asterisk > that was possible but always been a pain.?Is it possible to do source based > routing and route certain SIP extensions/devices to certain outbound trunks? > If so how can I do it easily? You can put the GW info in the directory: http://wiki.freeswitch.org/wiki/Clarification:gateways#conf.2Fdirectory.2Fdefault.2Fexample.com.xml Hope that helps. Welcome to FreeSWITCH! Gabe From miha at softnet.si Thu Feb 9 13:08:48 2012 From: miha at softnet.si (Miha Zoubek) Date: Thu, 09 Feb 2012 11:08:48 +0100 Subject: [Freeswitch-users] 302 redirect Message-ID: <4F339B30.8090806@softnet.si> Hi, I have set 302 redicet as is described on wiki (everything works ok). Problem that I am dealing with is regarding radius CDR issue. On FS I have set radius for AAA. As I do not need radius AAA for incoming calls, so I have set variable disable_radius_stop and disable_radius_start on true in public dialplan. After I set on phone 302 redirect the call should be seen in my sql (radius cdr). beacuse the variable is fist set (disable_radius_) on true I am not getting radius start and stop packet. After my dialplan recognized 302 redirect I set disable_radius_ on fals but it do not work because of the previous action (disable_radius_, I disable radius for start and stop). Please help as I do not know how to get this working. I also can not make a condition of the 302 redirected number as this variable is set further in dialplan. p.s.: in vars.xml I have set to radius diasble as I do need it only b leg. Thanks! BR, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. From wstephen80 at gmail.com Thu Feb 9 14:39:30 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Thu, 9 Feb 2012 12:39:30 +0100 Subject: [Freeswitch-users] T.38 SDP not forwarded by FS in t38-passthru mode Message-ID: I have trouble with fax because the INVITE generated by FS doesn't contain the image part. In my profile I have set: and the dialplan (simplified) is: but when is received an INVITE with SDP: v=0 o=- 2598299616 0 IN IP4 213.204.31.103 s=session t=0 0 m=audio 4640 RTP/AVP 18 8 96 c=IN IP4 213.204.31.32 a=rtpmap:18 G729/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:96 telephone-event/8000 m=image 4642 udptl t38 c=IN IP4 213.204.31.32 a=T38FaxVersion:0 a=T38MaxBitRate:14400 a=T38FaxUdpEC:t38UDPRedundancy a=T38FaxRateManagement:transferredTCF is generated in the outbound leg an SDP: v=0 o=FreeSWITCH 1328751960 1328751961 IN IP4 217.19.146.173 s=FreeSWITCH c=IN IP4 217.19.146.173 t=0 0 m=audio 31196 RTP/AVP 18 8 0 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 without image part. Here the log: http://pastebin.freeswitch.org/18333 Any advice? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/9c98dc24/attachment.html From neilp at cs.stanford.edu Thu Feb 9 15:22:16 2012 From: neilp at cs.stanford.edu (Neil Patel) Date: Thu, 9 Feb 2012 17:52:16 +0530 Subject: [Freeswitch-users] unable to compile mod_shout on latest git Message-ID: I'm on Ubuntu LTS and just did a pull from git. I've installed all the dependencies, and am getting this error: making all mod_shout Compiling /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c... quiet_libtool: compile: gcc -I/usr/src/freeswitch/libs/libshout-2.2.2/include -I/usr/src/freeswitch/libs/lame-3.98.4/include -I/usr/src/freeswitch/libs/mpg123-1.13.2/src -I/usr/src/freeswitch/libs/mpg123-1.13.2/src -I/usr/src/freeswitch/libs/libshout-2.2.2/include -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 -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c -fPIC -DPIC -o .libs/mod_shout.o quiet_libtool: compile: gcc -I/usr/src/freeswitch/libs/libshout-2.2.2/include -I/usr/src/freeswitch/libs/lame-3.98.4/include -I/usr/src/freeswitch/libs/mpg123-1.13.2/src -I/usr/src/freeswitch/libs/mpg123-1.13.2/src -I/usr/src/freeswitch/libs/libshout-2.2.2/include -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 -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c -o mod_shout.o >/dev/null 2>&1 Creating mod_shout.la... quiet_libtool: link: cannot find the library `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la' or unhandled argument `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib// libvorbis.la' make[5]: *** [mod_shout.la] Error 1 make[4]: *** [all] Error 1 make[3]: *** [mod_shout-all] Error 1 make[2]: *** [all-recursive] Error 1 make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/2652088c/attachment.html From bdfoster at endigotech.com Thu Feb 9 15:29:51 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Thu, 9 Feb 2012 07:29:51 -0500 Subject: [Freeswitch-users] unable to compile mod_shout on latest git In-Reply-To: References: Message-ID: cannot find the library `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib// libvorbis.la' or unhandled argument `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la Probably should look at filling that dependency.... On Feb 9, 2012 7:24 AM, "Neil Patel" wrote: > I'm on Ubuntu LTS and just did a pull from git. I've installed all the > dependencies, > and am getting this error: > > making all mod_shout > Compiling /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c... > quiet_libtool: compile: gcc > -I/usr/src/freeswitch/libs/libshout-2.2.2/include > -I/usr/src/freeswitch/libs/lame-3.98.4/include > -I/usr/src/freeswitch/libs/mpg123-1.13.2/src > -I/usr/src/freeswitch/libs/mpg123-1.13.2/src > -I/usr/src/freeswitch/libs/libshout-2.2.2/include > -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 > -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE > -DHAVE_CONFIG_H -c > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c -fPIC -DPIC -o > .libs/mod_shout.o > quiet_libtool: compile: gcc > -I/usr/src/freeswitch/libs/libshout-2.2.2/include > -I/usr/src/freeswitch/libs/lame-3.98.4/include > -I/usr/src/freeswitch/libs/mpg123-1.13.2/src > -I/usr/src/freeswitch/libs/mpg123-1.13.2/src > -I/usr/src/freeswitch/libs/libshout-2.2.2/include > -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 > -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE > -DHAVE_CONFIG_H -c > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c -o mod_shout.o > >/dev/null 2>&1 > Creating mod_shout.la... > quiet_libtool: link: cannot find the library > `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la' or > unhandled argument `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib// > libvorbis.la' > make[5]: *** [mod_shout.la] Error 1 > make[4]: *** [all] Error 1 > make[3]: *** [mod_shout-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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/f80bb919/attachment.html From neilp at cs.stanford.edu Thu Feb 9 16:11:56 2012 From: neilp at cs.stanford.edu (Neil Patel) Date: Thu, 9 Feb 2012 18:41:56 +0530 Subject: [Freeswitch-users] unable to compile mod_shout on latest git In-Reply-To: References: Message-ID: # apt-get install libvorbis0a libogg0 libogg-dev libvorbis-dev Reading package lists... Done Building dependency tree Reading state information... Done libvorbis0a is already the newest version. libogg0 is already the newest version. libogg-dev is already the newest version. libvorbis-dev is already the newest version. The following packages were automatically installed and are no longer required: firefox-3.5-branding sdparm Use 'apt-get autoremove' to remove them. 0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded. On Thu, Feb 9, 2012 at 5:59 PM, Brian Foster wrote: > cannot find the library > `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la' or > unhandled argument `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib// > libvorbis.la > > Probably should look at filling that dependency.... > On Feb 9, 2012 7:24 AM, "Neil Patel" wrote: > >> I'm on Ubuntu LTS and just did a pull from git. I've installed all the >> dependencies, >> and am getting this error: >> >> making all mod_shout >> Compiling /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c... >> quiet_libtool: compile: gcc >> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >> -I/usr/src/freeswitch/libs/lame-3.98.4/include >> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >> -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 >> -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE >> -DHAVE_CONFIG_H -c >> /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c -fPIC -DPIC -o >> .libs/mod_shout.o >> quiet_libtool: compile: gcc >> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >> -I/usr/src/freeswitch/libs/lame-3.98.4/include >> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >> -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 >> -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE >> -DHAVE_CONFIG_H -c >> /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c -o mod_shout.o >> >/dev/null 2>&1 >> Creating mod_shout.la... >> quiet_libtool: link: cannot find the library >> `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la' or >> unhandled argument `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib// >> libvorbis.la' >> make[5]: *** [mod_shout.la] Error 1 >> make[4]: *** [all] Error 1 >> make[3]: *** [mod_shout-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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/b9367660/attachment-0001.html From mojo1736 at privatedemail.net Thu Feb 9 17:09:33 2012 From: mojo1736 at privatedemail.net (Josh) Date: Thu, 09 Feb 2012 14:09:33 +0000 Subject: [Freeswitch-users] [newbie] questions In-Reply-To: References: <4F330A31.4060804@privatedemail.net> Message-ID: <4F33D39D.5070807@privatedemail.net> > Welcome to FreeSWITCH! Pleasure. > But calls will be sent to FreeSWITCH by some device, correct? If it's > good old-fashioned SIP then FreeSWITCH will handle it just fine. Yes, there are all softphones (PC machines with old-fashioned headset & mic and smartphones running a sip client). > > Could this be done relatively easily in FreeSWITCH? > > "Relatively?" Of course! It's relatively easy for someone with some > experience. I highly recommend that you ask consulting at freeswitch.org > for professional assistance if you > are not comfortable doing this all by yourself. I'd rather do it by myself. For now though, I'd like to see whether what I want to achieve is possible in FreeSWITCH. I haven't yet made a decision what to use though - Asterisk or FreeSWITCH, it will all depend on whether I could set it up properly and "relatively" easy. Having said all that, as I developer with quite a bit of experience behind me, I am not afraid to delve in and get my hands dirty, if needed. I just want to make sure that what I want in terms of set up and functionality is possible. > Yes, FreeSWITCH can bind to multiple interfaces. In FreeSWITCH lingo > that would mean that you set up a separate SIP profile for each > interface. (In fact, you can have more than one SIP profile on a given > interface since the profile is a unique combo of IP addr and port number.) I presume different profiles can "talk" to each other, right? In other words calls/media can be routed/transferred from one interface to another (eth1<->tun0 for example)? > "Some assembly required." :D > FreeSWITCH can do some stuff for you, but you definitely need to make > sure that your NAT is not behaving badly, like having a SIP ALG. This is what I am trying to figure out - do I rely entirely on FreeSWITCH (if not, what is expected of me to set up so that FreeSWITCH can do its job?), or do I have to do it all by myself with the kernel module helpers (sip, h323 etc) and ip/iptables? > I'll have to defer to Ken Rice on this one. I know he's working on > RPMs for FreeSWITCH but I think it's all RedHat right now. RedHat is good, all I need is a decent .spec file - I'll do the rest myself, no problem. One thing I seem to have forgotten from my newbie list of questions - I take it FreeSWITCH can do call-recording (in both directions) right? If so, how is this stored/implemented (I hope both ends are stored as a single sound file)? I would like all calls to be recorded as a matter of policy, so I do need this implemented if I am going to use FreeSWITCH. Many thanks again. From mojo1736 at privatedemail.net Thu Feb 9 17:20:58 2012 From: mojo1736 at privatedemail.net (Josh) Date: Thu, 09 Feb 2012 14:20:58 +0000 Subject: [Freeswitch-users] [newbie] questions In-Reply-To: <4F33D39D.5070807@privatedemail.net> References: <4F330A31.4060804@privatedemail.net> <4F33D39D.5070807@privatedemail.net> Message-ID: <4F33D64A.4010607@privatedemail.net> >> But calls will be sent to FreeSWITCH by some device, correct? If it's >> good old-fashioned SIP then FreeSWITCH will handle it just fine. > Yes, there are all softphones (PC machines with old-fashioned headset > & mic and smartphones running a sip client). Forgot to add - external calls (via the external VOIP provider) cannot be received via the FreeSWITCH listening port because of the fact that my ISP blocks connections initiated from outside. When a connection request (SYN) is initiated from my side, my ISP allows it, no problem (I believe the applicable term in such scenarios would be "fascist firewall"). What usually happens is that the control channel to my external VOIP provider on port 5060 is initiated by my end and when a call needs to be received from outside, this is communicated through the control channel (which is kept open at all times) and then my client opens both rtp data stream connections to receive the call. Does that make sense? From avi at avimarcus.net Thu Feb 9 17:27:33 2012 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 9 Feb 2012 16:27:33 +0200 Subject: [Freeswitch-users] [newbie] questions In-Reply-To: <4F33D39D.5070807@privatedemail.net> References: <4F330A31.4060804@privatedemail.net> <4F33D39D.5070807@privatedemail.net> Message-ID: > > One thing I seem to have forgotten from my newbie list of questions - I > take it FreeSWITCH can do call-recording (in both directions) right? If > so, how is this stored/implemented (I hope both ends are stored as a > single sound file)? I would like all calls to be recorded as a matter of > policy, so I do need this implemented if I am going to use FreeSWITCH. > Yes, FS can record as long as it's in the media path. You can save as any format that FS can handle, including .mp3 if mod_shout is loaded - see http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session It saves both channels to one file, but you have the option to record only caller/callee or to swap the channels. There doesn't seem to be a "flatten to mono" option. See the RECORD_* links at the bottom of that page. -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/31fad2ab/attachment.html From curriegrad2004 at gmail.com Thu Feb 9 18:10:10 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Thu, 9 Feb 2012 07:10:10 -0800 Subject: [Freeswitch-users] unable to compile mod_shout on latest git In-Reply-To: References: Message-ID: In this case, what you could do is download the libogg and libvorbis libraries, configure them to link each other seperately and install them in a location other than /usr. After you've done that you'll have to edit the mod_shout makefile to relflect this change. I remember doing this on another platform, but I can't remember the specific details on going about that. On Thu, Feb 9, 2012 at 5:11 AM, Neil Patel wrote: > # apt-get install libvorbis0a libogg0 libogg-dev libvorbis-dev > Reading package lists... Done > Building dependency tree > Reading state information... Done > libvorbis0a is already the newest version. > libogg0 is already the newest version. > libogg-dev is already the newest version. > libvorbis-dev is already the newest version. > The following packages were automatically installed and are no longer > required: > ? firefox-3.5-branding sdparm > Use 'apt-get autoremove' to remove them. > 0 upgraded, 0 newly installed, 0 to remove and 0 not upgraded. > > > On Thu, Feb 9, 2012 at 5:59 PM, Brian Foster > wrote: >> >> cannot find the library >> `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la' or >> unhandled argument >> `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la >> >> Probably should look at filling that dependency.... >> >> On Feb 9, 2012 7:24 AM, "Neil Patel" wrote: >>> >>> I'm on Ubuntu LTS and just did a pull from git. I've installed all the >>> dependencies, and am getting this error: >>> >>> making all mod_shout >>> Compiling /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c... >>> quiet_libtool: compile: ?gcc >>> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >>> -I/usr/src/freeswitch/libs/lame-3.98.4/include >>> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >>> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >>> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >>> -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 -g -O2 -Wall -std=c99 -pedantic >>> -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c >>> /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c ?-fPIC -DPIC -o >>> .libs/mod_shout.o >>> quiet_libtool: compile: ?gcc >>> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >>> -I/usr/src/freeswitch/libs/lame-3.98.4/include >>> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >>> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >>> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >>> -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 -g -O2 -Wall -std=c99 -pedantic >>> -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c >>> /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c -o mod_shout.o >>> >/dev/null 2>&1 >>> Creating mod_shout.la... >>> quiet_libtool: link: cannot find the library >>> `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la' or >>> unhandled argument >>> `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la' >>> make[5]: *** [mod_shout.la] Error 1 >>> make[4]: *** [all] Error 1 >>> make[3]: *** [mod_shout-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 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From mojo1736 at privatedemail.net Thu Feb 9 18:24:21 2012 From: mojo1736 at privatedemail.net (Josh) Date: Thu, 09 Feb 2012 15:24:21 +0000 Subject: [Freeswitch-users] [newbie] questions In-Reply-To: References: <4F330A31.4060804@privatedemail.net> <4F33D39D.5070807@privatedemail.net> Message-ID: <4F33E525.4070305@privatedemail.net> > Yes, FS can record as long as it's in the media path. > You can save as any format that FS can handle, including .mp3 if > mod_shout is loaded - > see http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_record_session Thanks, that's a good find! I also found solution to another common feature I was planning to add: call screening (screening of calls with unknown/missing CID or from numbers I don't know or have placed in a special group) - in Asterisk this is a bit limited to about 2, 3 options which I have no control over, but in FreeSWITCH, it seems, I have a complete freedom of how/what to do when call screening is initiated - http://wiki.freeswitch.org/wiki/Dialplan_XML (Example 13) is exactly what I needed, though I take it I could customise it a bit so that: I (as the called party) have the option (by pressing a specific key after receiving the caller's name) to: 1) accept the call; 2) send a message back to the caller saying this number is unavailable and then terminate his/her call; 3) redirect him/her to voicemail; 4) drop the call entirely without giving any explanation at all (or, alternatively, torture them with some moh if they appear to be telemarketing group for example). By reading that example, it looks as though this can be achieved. One other thing: in Asterisk, there is a vast sound library for various messages/music/media which could be used. Is there a similar thing in FreeSWITCH or should I use Asterisk's media pack for now? > It saves both channels to one file, but you have the option to record > only caller/callee or to swap the channels. There doesn't seem to be a > "flatten to mono" option. See the RECORD_* links at the bottom of that > page. If I use would that be sufficient to record mono instead? From brian at freeswitch.org Thu Feb 9 18:24:52 2012 From: brian at freeswitch.org (Brian West) Date: Thu, 9 Feb 2012 09:24:52 -0600 Subject: [Freeswitch-users] T.38 SDP not forwarded by FS in t38-passthru mode In-Reply-To: References: Message-ID: <27902030-E48C-4E49-865A-EF81480C7035@freeswitch.org> I'm going to guess its because SOA has rejected the image part on the first leg. Please provide a complete SIP trace and open a jira. Thanks, Brian On Feb 9, 2012, at 5:39 AM, Stephen Wilde wrote: > I have trouble with fax because the INVITE generated by FS doesn't contain > the image part. > > In my profile I have set: > > > > and the dialplan (simplified) is: > > > > > but when is received an INVITE with SDP: > > v=0 > o=- 2598299616 0 IN IP4 213.204.31.103 > s=session > t=0 0 > m=audio 4640 RTP/AVP 18 8 96 > c=IN IP4 213.204.31.32 > a=rtpmap:18 G729/8000 > a=rtpmap:8 PCMA/8000 > a=rtpmap:96 telephone-event/8000 > m=image 4642 udptl t38 > c=IN IP4 213.204.31.32 > a=T38FaxVersion:0 > a=T38MaxBitRate:14400 > a=T38FaxUdpEC:t38UDPRedundancy > a=T38FaxRateManagement:transferredTCF > > is generated in the outbound leg an SDP: > > v=0 > o=FreeSWITCH 1328751960 1328751961 IN IP4 217.19.146.173 > s=FreeSWITCH > c=IN IP4 217.19.146.173 > t=0 0 > m=audio 31196 RTP/AVP 18 8 0 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > > without image part. > > Here the log: > > http://pastebin.freeswitch.org/18333 > > Any advice? > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/a9a7e1e0/attachment-0001.html From avi at avimarcus.net Thu Feb 9 18:45:11 2012 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 9 Feb 2012 17:45:11 +0200 Subject: [Freeswitch-users] [newbie] questions In-Reply-To: <4F33E525.4070305@privatedemail.net> References: <4F330A31.4060804@privatedemail.net> <4F33D39D.5070807@privatedemail.net> <4F33E525.4070305@privatedemail.net> Message-ID: > > I also found solution to another common feature I was planning to add: > call screening (screening of calls with unknown/missing CID or from > numbers I don't know or have placed in a special group) - in Asterisk > this is a bit limited to about 2, 3 options which I have no control > over, but in FreeSWITCH, it seems, I have a complete freedom of how/what > to do when call screening is initiated - > http://wiki.freeswitch.org/wiki/Dialplan_XML (Example 13) is exactly > what I needed, though I take it I could customise it a bit so that: > > I (as the called party) have the option (by pressing a specific key > after receiving the caller's name) to: 1) accept the call; 2) send a > message back to the caller saying this number is unavailable and then > terminate his/her call; 3) redirect him/her to voicemail; 4) drop the > call entirely without giving any explanation at all (or, alternatively, > torture them with some moh if they appear to be telemarketing group for > example). By reading that example, it looks as though this can be achieved. > > If you only need "answer or reject to continue with the dialplan" then simply use the pre-built Answer Confirmation: http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation Anything else will be a bit more complicated. For telemarketers, you could bridge them to http://itslenny.com/ > One other thing: in Asterisk, there is a vast sound library for various > messages/music/media which could be used. Is there a similar thing in > FreeSWITCH or should I use Asterisk's media pack for now? > > When installing if you did "make sounds-install" you got the basic sound package which is quite extensive. I recommend looking at the docs/phrase/phrase_en.xml and see the list of the phrases / pre-recorded sound files under the name "callie". -Avi -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/573db6c1/attachment.html From brad at tech21.com Thu Feb 9 20:29:25 2012 From: brad at tech21.com (Brad Mina) Date: Thu, 9 Feb 2012 09:29:25 -0800 Subject: [Freeswitch-users] New to FS In-Reply-To: References: Message-ID: > Hope that helps. Welcome to FreeSWITCH! *The future of telephony!* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/24d907ee/attachment.html From mantung at commverge.com Thu Feb 9 12:28:45 2012 From: mantung at commverge.com (Man) Date: Thu, 09 Feb 2012 17:28:45 +0800 Subject: [Freeswitch-users] System shown no route but connected with false caller ID In-Reply-To: References: <4F334576.4030404@commverge.com> <4F338476.3070700@commverge.com> Message-ID: <4F3391CD.6040305@commverge.com> Full log here: 3175x102 is the caller and 6578xxxx is my mobile. It is strange that the call should be already bridged to the voice gateway 203.x.x.80. Why the call loop back to load the "public" dialplan? 2012-02-09 17:09:17.164122 [DEBUG] sofia.c:7491 IP 203.x.x.81 Approved by acl "domains[]". Access Granted. 2012-02-09 17:09:17.164122 [NOTICE] switch_channel.c:926 New Channel sofia/internal/3175x102 at 203.x.x.81:5060 [19a60d67-29ea-4d02-8b46-fe0b6091cbe3] 2012-02-09 17:09:17.164122 [DEBUG] sofia.c:5512 Channel sofia/internal/3175x102 at 203.x.x.81:5060 entering state [received][100] 2012-02-09 17:09:17.164122 [DEBUG] sofia.c:5523 Remote SDP: v=0 o=DBLE 1328775427 1328775427 IN IP4 10.0.166.104 s=DBLE c=IN IP4 10.0.166.104 t=0 0 m=audio 16384 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2012-02-09 17:09:17.164122 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [G729:18:8000:20:8000]/[PCMU:0:8000:20:64000] 2012-02-09 17:09:17.164122 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2012-02-09 17:09:17.164122 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-02-09 17:09:17.164122 [DEBUG] sofia_glue.c:2919 Set Codec sofia/internal/3175x102 at 203.x.x.81:5060 PCMU/8000 20 ms 160 samples 64000 bits 2012-02-09 17:09:17.164122 [DEBUG] switch_core_codec.c:111 sofia/internal/3175x102 at 203.x.x.81:5060 Original read codec set to PCMU:0 2012-02-09 17:09:17.164122 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf send/recv payload to 101 2012-02-09 17:09:17.164122 [DEBUG] sofia.c:5735 (sofia/internal/3175x102 at 203.x.x.81:5060) State Change CS_NEW -> CS_INIT 2012-02-09 17:09:17.164122 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/3175x102 at 203.x.x.81:5060 [BREAK] 2012-02-09 17:09:17.164122 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/3175x102 at 203.x.x.81:5060) Running State Change CS_INIT 2012-02-09 17:09:17.164122 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/3175x102 at 203.x.x.81:5060) State INIT 2012-02-09 17:09:17.164122 [DEBUG] mod_sofia.c:85 sofia/internal/3175x102 at 203.x.x.81:5060 SOFIA INIT 2012-02-09 17:09:17.164122 [DEBUG] mod_sofia.c:125 (sofia/internal/3175x102 at 203.x.x.81:5060) State Change CS_INIT -> CS_ROUTING 2012-02-09 17:09:17.164122 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/3175x102 at 203.x.x.81:5060 [BREAK] 2012-02-09 17:09:17.164122 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/3175x102 at 203.x.x.81:5060) State INIT going to sleep 2012-02-09 17:09:17.164122 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/3175x102 at 203.x.x.81:5060) Running State Change CS_ROUTING 2012-02-09 17:09:17.164122 [DEBUG] switch_channel.c:1886 (sofia/internal/3175x102 at 203.x.x.81:5060) Callstate Change DOWN -> RINGING 2012-02-09 17:09:17.164122 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/3175x102 at 203.x.x.81:5060) State ROUTING 2012-02-09 17:09:17.164122 [DEBUG] mod_sofia.c:148 sofia/internal/3175x102 at 203.x.x.81:5060 SOFIA ROUTING 2012-02-09 17:09:17.164122 [DEBUG] switch_core_state_machine.c:104 sofia/internal/3175x102 at 203.x.x.81:5060 Standard ROUTING 2012-02-09 17:09:17.164122 [INFO] mod_dialplan_xml.c:485 Processing 3175x102 <3175x102>->6578xxxx in context default Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->unloop] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->suspension_0c] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [suspension_0c] caller_id_number(3175x102) =~ /3175x323/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->suspension_0d] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [suspension_0d] caller_id_number(3175x102) =~ /3175x119/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->suspension_0g] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [suspension_0g] caller_id_number(3175x102) =~ /3175x208/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->suspension_0h] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [suspension_0h] caller_id_number(3175x102) =~ /31780526/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->suspension_0h] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [suspension_0h] caller_id_number(3175x102) =~ /3175x241/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->suspension_0j] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [suspension_0j] caller_id_number(3175x102) =~ /31780566/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->suspension_0k] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [suspension_0k] caller_id_number(3175x102) =~ /31780646/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->suspension_0l] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [suspension_0l] caller_id_number(3175x102) =~ /31780626/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->suspension_0m] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [suspension_0m] caller_id_number(3175x102) =~ /3175x156/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_outbound_3175x129] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_outbound_3175x129] caller_id_number(3175x102) =~ /3175x129/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x252] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x252] caller_id_number(3175x102) =~ /3175x252/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x246] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x246] caller_id_number(3175x102) =~ /3175x246/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x247] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x247] caller_id_number(3175x102) =~ /3175x247/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x248] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x248] caller_id_number(3175x102) =~ /3175x248/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x249] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x249] caller_id_number(3175x102) =~ /3175x249/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x253] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x253] caller_id_number(3175x102) =~ /3175x253/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x254] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x254] caller_id_number(3175x102) =~ /3175x254/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x255] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x255] caller_id_number(3175x102) =~ /3175x255/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x256] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x256] caller_id_number(3175x102) =~ /3175x256/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x257] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x257] caller_id_number(3175x102) =~ /3175x257/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x258] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x258] caller_id_number(3175x102) =~ /3175x258/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x259] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x259] caller_id_number(3175x102) =~ /3175x259/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x260] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x260] caller_id_number(3175x102) =~ /3175x260/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x261] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x261] caller_id_number(3175x102) =~ /3175x261/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x263] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x263] caller_id_number(3175x102) =~ /3175x263/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x264] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x264] caller_id_number(3175x102) =~ /3175x264/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x298] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x298] caller_id_number(3175x102) =~ /3175x298/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x299] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x299] caller_id_number(3175x102) =~ /3175x299/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x330] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x330] caller_id_number(3175x102) =~ /3175x330/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x331] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x331] caller_id_number(3175x102) =~ /3175x331/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x332] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x332] caller_id_number(3175x102) =~ /3175x332/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_3175x333] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_3175x333] caller_id_number(3175x102) =~ /3175x333/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_outbound_133] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_outbound_133] destination_number(6578xxxx) =~ /^133(\d{8})$/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_outbound_hk_3digits] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_outbound_hk_3digits] destination_number(6578xxxx) =~ /^(\d{3})$/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_outbound_hk_4digits] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_outbound_hk_4digits] destination_number(6578xxxx) =~ /^(\d{4})$/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_outbound_hk_5dights] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_outbound_hk_5dights] destination_number(6578xxxx) =~ /^(\d{5})$/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_outbound_hk_7dights] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_outbound_hk_7dights] destination_number(6578xxxx) =~ /^(\d{7})$/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_outbound_test72992] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_outbound_test72992] destination_number(6578xxxx) =~ /^72992?(\d+)$/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_outbound_idd_0852] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_outbound_idd_0852] destination_number(6578xxxx) =~ /^0852?(\d+)$/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_outbound_idd_00852] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (FAIL) [DID_outbound_idd_00852] destination_number(6578xxxx) =~ /^00852?(\d+)$/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 parsing [default->DID_outbound] continue=false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Regex (PASS) [DID_outbound] destination_number(6578xxxx) =~ /^(\d{8})$/ break=on-false Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Action set(effective_caller_id_number=${caller_id_number}) Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Action set(ringback=${hk-ring}) Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Action set_audio_level(read 1) Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Action set_audio_level(write 1) _Dialplan: sofia/internal/3175x102 at 203.x.x.81:5060 Action bridge(sofia/external/6578xxxx at 203.x.x.80) _ 2012-02-09 17:09:17.164122 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/3175x102 at 203.x.x.81:5060) State Change CS_ROUTING -> CS_EXECUTE 2012-02-09 17:09:17.164122 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/3175x102 at 203.x.x.81:5060 [BREAK] 2012-02-09 17:09:17.164122 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/3175x102 at 203.x.x.81:5060) State ROUTING going to sleep 2012-02-09 17:09:17.164122 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/3175x102 at 203.x.x.81:5060) Running State Change CS_EXECUTE 2012-02-09 17:09:17.164122 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/3175x102 at 203.x.x.81:5060) State EXECUTE 2012-02-09 17:09:17.164122 [DEBUG] mod_sofia.c:241 sofia/internal/3175x102 at 203.x.x.81:5060 SOFIA EXECUTE 2012-02-09 17:09:17.164122 [DEBUG] switch_core_state_machine.c:192 sofia/internal/3175x102 at 203.x.x.81:5060 Standard EXECUTE EXECUTE sofia/internal/3175x102 at 203.x.x.81:5060 set(effective_caller_id_number=3175x102) 2012-02-09 17:09:17.164122 [DEBUG] mod_dptools.c:1281 sofia/internal/3175x102 at 203.x.x.81:5060 SET [effective_caller_id_number]=[3175x102] EXECUTE sofia/internal/3175x102 at 203.x.x.81:5060 set(ringback=%(400,200,440,480);%(400,3000,440,480)) 2012-02-09 17:09:17.164122 [DEBUG] mod_dptools.c:1281 sofia/internal/3175x102 at 203.x.x.81:5060 SET [ringback]=[%(400,200,440,480);%(400,3000,440,480)] 2012-02-09 17:09:17.175916 [DEBUG] switch_core_session.c:2133 Application set_audio_level Requires media! pre_answering channel sofia/internal/3175x102 at 203.x.x.81:5060 2012-02-09 17:09:17.175916 [INFO] switch_core_session.c:2135 Sending early media 2012-02-09 17:09:17.175916 [DEBUG] sofia_glue.c:3171 AUDIO RTP [sofia/internal/3175x102 at 203.x.x.81:5060] 203.x.x.81 port 17908 -> 10.0.166.104 port 16384 codec: 0 ms: 20 2012-02-09 17:09:17.175916 [DEBUG] switch_rtp.c:1661 Starting timer [soft] 160 bytes per 20ms 2012-02-09 17:09:17.175916 [DEBUG] sofia_glue.c:3435 Set 2833 dtmf send payload to 101 2012-02-09 17:09:17.175916 [DEBUG] sofia_glue.c:3441 Set 2833 dtmf receive payload to 101 2012-02-09 17:09:17.175916 [DEBUG] mod_sofia.c:2573 Ring SDP: v=0 o=FreeSWITCH 1328760649 1328760650 IN IP4 203.x.x.81 s=FreeSWITCH c=IN IP4 203.x.x.81 t=0 0 m=audio 17908 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 a=sendrecv 2012-02-09 17:09:17.175916 [NOTICE] mod_sofia.c:2576 Pre-Answer sofia/internal/3175x102 at 203.x.x.81:5060! 2012-02-09 17:09:17.175916 [DEBUG] switch_channel.c:2932 (sofia/internal/3175x102 at 203.x.x.81:5060) Callstate Change RINGING -> EARLY 2012-02-09 17:09:17.175916 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/3175x102 at 203.x.x.81:5060 [BREAK] EXECUTE sofia/internal/3175x102 at 203.x.x.81:5060 set_audio_level(read 1) 2012-02-09 17:09:17.175916 [DEBUG] switch_core_media_bug.c:456 Attaching BUG to sofia/internal/3175x102 at 203.x.x.81:5060 2012-02-09 17:09:17.175916 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/3175x102 at 203.x.x.81:5060 [BREAK] 2012-02-09 17:09:17.175916 [DEBUG] sofia.c:5505 Channel sofia/internal/3175x102 at 203.x.x.81:5060 skipping state [early][183] EXECUTE sofia/internal/3175x102 at 203.x.x.81:5060 set_audio_level(write 1) EXECUTE sofia/internal/3175x102 at 203.x.x.81:5060 bridge(sofia/external/6578xxxx at 203.x.x.80) 2012-02-09 17:09:17.175916 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-02-09 17:09:17.175916 [NOTICE] switch_channel.c:926 New Channel sofia/external/6578xxxx at 203.x.x.80 [456102df-897a-4dd5-bb46-92c45072e8e9] 2012-02-09 17:09:17.175916 [DEBUG] mod_sofia.c:4670 (sofia/external/6578xxxx at 203.x.x.80) State Change CS_NEW -> CS_INIT 2012-02-09 17:09:17.175916 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/6578xxxx at 203.x.x.80 [BREAK] 2012-02-09 17:09:17.175916 [DEBUG] switch_core_state_machine.c:362 (sofia/external/6578xxxx at 203.x.x.80) Running State Change CS_INIT 2012-02-09 17:09:17.175916 [DEBUG] switch_core_state_machine.c:401 (sofia/external/6578xxxx at 203.x.x.80) State INIT 2012-02-09 17:09:17.175916 [DEBUG] mod_sofia.c:85 sofia/external/6578xxxx at 203.x.x.80 SOFIA INIT 2012-02-09 17:09:17.175916 [DEBUG] mod_sofia.c:125 (sofia/external/6578xxxx at 203.x.x.80) State Change CS_INIT -> CS_ROUTING 2012-02-09 17:09:17.175916 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/6578xxxx at 203.x.x.80 [BREAK] 2012-02-09 17:09:17.175916 [DEBUG] switch_core_state_machine.c:401 (sofia/external/6578xxxx at 203.x.x.80) State INIT going to sleep 2012-02-09 17:09:17.175916 [DEBUG] switch_core_state_machine.c:362 (sofia/external/6578xxxx at 203.x.x.80) Running State Change CS_ROUTING 2012-02-09 17:09:17.175916 [DEBUG] switch_channel.c:1886 (sofia/external/6578xxxx at 203.x.x.80) Callstate Change DOWN -> RINGING 2012-02-09 17:09:17.175916 [DEBUG] switch_core_state_machine.c:410 (sofia/external/6578xxxx at 203.x.x.80) State ROUTING 2012-02-09 17:09:17.175916 [DEBUG] mod_sofia.c:148 sofia/external/6578xxxx at 203.x.x.80 SOFIA ROUTING 2012-02-09 17:09:17.175916 [DEBUG] switch_ivr_originate.c:66 (sofia/external/6578xxxx at 203.x.x.80) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-02-09 17:09:17.175916 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/6578xxxx at 203.x.x.80 [BREAK] 2012-02-09 17:09:17.175916 [DEBUG] switch_core_state_machine.c:410 (sofia/external/6578xxxx at 203.x.x.80) State ROUTING going to sleep 2012-02-09 17:09:17.175916 [DEBUG] switch_core_state_machine.c:362 (sofia/external/6578xxxx at 203.x.x.80) Running State Change CS_CONSUME_MEDIA 2012-02-09 17:09:17.175916 [DEBUG] switch_core_state_machine.c:429 (sofia/external/6578xxxx at 203.x.x.80) State CONSUME_MEDIA 2012-02-09 17:09:17.175916 [DEBUG] switch_core_state_machine.c:429 (sofia/external/6578xxxx at 203.x.x.80) State CONSUME_MEDIA going to sleep 2012-02-09 17:09:17.175916 [DEBUG] switch_core_session.c:875 Send signal sofia/external/6578xxxx at 203.x.x.80 [BREAK] 2012-02-09 17:09:17.175916 [DEBUG] sofia.c:5512 Channel sofia/external/6578xxxx at 203.x.x.80 entering state [calling][0] 2012-02-09 17:09:17.199902 [NOTICE] switch_channel.c:926 New Channel sofia/external/3175x102 at 203.x.x.80 [2856d320-da64-4c4f-aa83-642a63bcf07d] 2012-02-09 17:09:17.199902 [DEBUG] sofia.c:8451 Setting NAT mode based on via port 2012-02-09 17:09:17.199902 [DEBUG] sofia.c:5512 Channel sofia/external/3175x102 at 203.x.x.80 entering state [received][100] 2012-02-09 17:09:17.199902 [DEBUG] sofia.c:5523 Remote SDP: v=0 o=CiscoSystemsSIP-GW-UserAgent 206 122 IN IP4 203.x.x.80 s=SIP Call c=IN IP4 203.x.x.80 t=0 0 m=audio 18928 RTP/AVP 0 8 101 101 c=IN IP4 203.x.x.80 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 2012-02-09 17:09:17.199902 [DEBUG] sofia_glue.c:4374 Activate Buggy RFC2833 Mode! 2012-02-09 17:09:17.199902 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-02-09 17:09:17.199902 [DEBUG] sofia_glue.c:2919 Set Codec sofia/external/3175x102 at 203.x.x.80 PCMU/8000 20 ms 160 samples 64000 bits 2012-02-09 17:09:17.199902 [DEBUG] switch_core_codec.c:111 sofia/external/3175x102 at 203.x.x.80 Original read codec set to PCMU:0 2012-02-09 17:09:17.199902 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf send/recv payload to 101 2012-02-09 17:09:17.199902 [DEBUG] sofia.c:5735 (sofia/external/3175x102 at 203.x.x.80) State Change CS_NEW -> CS_INIT 2012-02-09 17:09:17.199902 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/3175x102 at 203.x.x.80 [BREAK] 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:362 (sofia/external/3175x102 at 203.x.x.80) Running State Change CS_INIT 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:401 (sofia/external/3175x102 at 203.x.x.80) State INIT 2012-02-09 17:09:17.199902 [DEBUG] mod_sofia.c:85 sofia/external/3175x102 at 203.x.x.80 SOFIA INIT 2012-02-09 17:09:17.199902 [DEBUG] mod_sofia.c:125 (sofia/external/3175x102 at 203.x.x.80) State Change CS_INIT -> CS_ROUTING 2012-02-09 17:09:17.199902 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/3175x102 at 203.x.x.80 [BREAK] 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:401 (sofia/external/3175x102 at 203.x.x.80) State INIT going to sleep 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:362 (sofia/external/3175x102 at 203.x.x.80) Running State Change CS_ROUTING 2012-02-09 17:09:17.199902 [DEBUG] switch_channel.c:1886 (sofia/external/3175x102 at 203.x.x.80) Callstate Change DOWN -> RINGING 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:410 (sofia/external/3175x102 at 203.x.x.80) State ROUTING 2012-02-09 17:09:17.199902 [DEBUG] mod_sofia.c:148 sofia/external/3175x102 at 203.x.x.80 SOFIA ROUTING 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:104 sofia/external/3175x102 at 203.x.x.80 Standard ROUTING 2012-02-09 17:09:17.199902 [INFO] mod_dialplan_xml.c:485 Processing 3175x102 <3175x102>->6578xxxx in context public Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->unloop] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780500] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780500] destination_number(6578xxxx) =~ /31780500/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780501] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780501] destination_number(6578xxxx) =~ /31780501/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780502] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780502] destination_number(6578xxxx) =~ /31780502/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780503] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780503] destination_number(6578xxxx) =~ /31780503/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780504] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780504] destination_number(6578xxxx) =~ /31780504/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780505] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780505] destination_number(6578xxxx) =~ /31780505/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780506] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780506] destination_number(6578xxxx) =~ /31780506/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780507] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780507] destination_number(6578xxxx) =~ /31780507/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780508] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780508] destination_number(6578xxxx) =~ /31780508/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780509] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780509] destination_number(6578xxxx) =~ /31780509/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780510] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780510] destination_number(6578xxxx) =~ /31780510/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780511] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780511] destination_number(6578xxxx) =~ /31780511/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780512] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780512] destination_number(6578xxxx) =~ /31780512/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780513] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780513] destination_number(6578xxxx) =~ /31780513/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780514] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780514] destination_number(6578xxxx) =~ /31780514/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780515] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780515] destination_number(6578xxxx) =~ /31780515/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780516] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780516] destination_number(6578xxxx) =~ /31780516/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780517] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780517] destination_number(6578xxxx) =~ /31780517/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780518] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780518] destination_number(6578xxxx) =~ /31780518/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780519] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780519] destination_number(6578xxxx) =~ /31780519/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780520] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780520] destination_number(6578xxxx) =~ /31780520/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780521] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780521] destination_number(6578xxxx) =~ /31780521/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780522] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780522] destination_number(6578xxxx) =~ /31780522/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780523] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780523] destination_number(6578xxxx) =~ /31780523/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780524] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780524] destination_number(6578xxxx) =~ /31780524/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780525] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780525] destination_number(6578xxxx) =~ /31780525/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780526] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780526] destination_number(6578xxxx) =~ /31780526/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780527] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780527] destination_number(6578xxxx) =~ /31780527/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780528] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780528] destination_number(6578xxxx) =~ /31780528/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780529] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780529] destination_number(6578xxxx) =~ /31780529/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780530] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780530] destination_number(6578xxxx) =~ /31780530/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780531] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780531] destination_number(6578xxxx) =~ /31780531/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780532] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780532] destination_number(6578xxxx) =~ /31780532/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780533] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780533] destination_number(6578xxxx) =~ /31780533/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780534] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780534] destination_number(6578xxxx) =~ /31780534/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780535] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780535] destination_number(6578xxxx) =~ /31780535/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780536] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780536] destination_number(6578xxxx) =~ /31780536/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780537] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780537] destination_number(6578xxxx) =~ /31780537/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780538] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780538] destination_number(6578xxxx) =~ /31780538/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780539] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780539] destination_number(6578xxxx) =~ /31780539/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780540] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780540] destination_number(6578xxxx) =~ /31780540/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780541] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780541] destination_number(6578xxxx) =~ /31780541/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780542] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780542] destination_number(6578xxxx) =~ /31780542/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780543] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780543] destination_number(6578xxxx) =~ /31780543/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780544] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780544] destination_number(6578xxxx) =~ /31780544/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780545] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780545] destination_number(6578xxxx) =~ /31780545/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780546] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780546] destination_number(6578xxxx) =~ /31780546/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780547] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780547] destination_number(6578xxxx) =~ /31780547/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780548] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780548] destination_number(6578xxxx) =~ /31780548/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780549] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780549] destination_number(6578xxxx) =~ /31780549/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780550] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780550] destination_number(6578xxxx) =~ /31780550/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780551] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780551] destination_number(6578xxxx) =~ /31780551/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780552] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780552] destination_number(6578xxxx) =~ /31780552/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780553] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780553] destination_number(6578xxxx) =~ /31780553/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780554] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780554] destination_number(6578xxxx) =~ /31780554/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780555] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780555] destination_number(6578xxxx) =~ /31780555/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780556] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780556] destination_number(6578xxxx) =~ /31780556/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780557] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780557] destination_number(6578xxxx) =~ /31780557/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780558] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780558] destination_number(6578xxxx) =~ /31780558/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780559] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780559] destination_number(6578xxxx) =~ /31780559/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780560] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780560] destination_number(6578xxxx) =~ /31780560/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780561] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780561] destination_number(6578xxxx) =~ /31780561/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780562] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780562] destination_number(6578xxxx) =~ /31780562/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780563] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780563] destination_number(6578xxxx) =~ /31780563/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780564] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780564] destination_number(6578xxxx) =~ /31780564/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780565] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780565] destination_number(6578xxxx) =~ /31780565/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780566] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780566] destination_number(6578xxxx) =~ /31780566/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780567] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780567] destination_number(6578xxxx) =~ /31780567/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780568] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780568] destination_number(6578xxxx) =~ /31780568/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780569] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780569] destination_number(6578xxxx) =~ /31780569/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780570] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780570] destination_number(6578xxxx) =~ /31780570/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780571] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780571] destination_number(6578xxxx) =~ /31780571/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780572] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780572] destination_number(6578xxxx) =~ /31780572/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780573] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780573] destination_number(6578xxxx) =~ /31780573/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780574] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780574] destination_number(6578xxxx) =~ /31780574/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780575] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780575] destination_number(6578xxxx) =~ /31780575/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780576] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780576] destination_number(6578xxxx) =~ /31780576/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780577] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780577] destination_number(6578xxxx) =~ /31780577/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780578] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780578] destination_number(6578xxxx) =~ /31780578/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780579] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780579] destination_number(6578xxxx) =~ /31780579/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780580] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780580] destination_number(6578xxxx) =~ /31780580/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780581] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780581] destination_number(6578xxxx) =~ /31780581/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780582] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780582] destination_number(6578xxxx) =~ /31780582/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780583] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780583] destination_number(6578xxxx) =~ /31780583/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780584] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780584] destination_number(6578xxxx) =~ /31780584/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780585] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780585] destination_number(6578xxxx) =~ /31780585/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780586] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780586] destination_number(6578xxxx) =~ /31780586/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780587] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780587] destination_number(6578xxxx) =~ /31780587/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780588] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780588] destination_number(6578xxxx) =~ /31780588/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780589] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780589] destination_number(6578xxxx) =~ /31780589/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780590] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780590] destination_number(6578xxxx) =~ /31780590/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780591] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780591] destination_number(6578xxxx) =~ /31780591/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780592] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780592] destination_number(6578xxxx) =~ /31780592/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780593] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780593] destination_number(6578xxxx) =~ /31780593/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780594] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780594] destination_number(6578xxxx) =~ /31780594/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780595] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780595] destination_number(6578xxxx) =~ /31780595/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780596] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780596] destination_number(6578xxxx) =~ /31780596/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780597] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780597] destination_number(6578xxxx) =~ /31780597/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780598] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780598] destination_number(6578xxxx) =~ /31780598/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780599] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780599] destination_number(6578xxxx) =~ /31780599/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780600] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780600] destination_number(6578xxxx) =~ /31780600/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780601] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780601] destination_number(6578xxxx) =~ /31780601/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780602] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780602] destination_number(6578xxxx) =~ /31780602/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780603] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780603] destination_number(6578xxxx) =~ /31780603/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780604] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780604] destination_number(6578xxxx) =~ /31780604/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780605] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780605] destination_number(6578xxxx) =~ /31780605/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780606] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780606] destination_number(6578xxxx) =~ /31780606/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780607] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780607] destination_number(6578xxxx) =~ /31780607/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780608] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780608] destination_number(6578xxxx) =~ /31780608/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780609] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780609] destination_number(6578xxxx) =~ /31780609/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780610] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780610] destination_number(6578xxxx) =~ /31780610/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780611] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780611] destination_number(6578xxxx) =~ /31780611/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780612] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780612] destination_number(6578xxxx) =~ /31780612/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780613] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780613] destination_number(6578xxxx) =~ /31780613/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780614] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780614] destination_number(6578xxxx) =~ /31780614/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780615] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780615] destination_number(6578xxxx) =~ /31780615/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780616] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780616] destination_number(6578xxxx) =~ /31780616/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780617] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780617] destination_number(6578xxxx) =~ /31780617/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780618] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780618] destination_number(6578xxxx) =~ /31780618/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780619] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780619] destination_number(6578xxxx) =~ /31780619/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780620] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780620] destination_number(6578xxxx) =~ /31780620/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780621] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780621] destination_number(6578xxxx) =~ /31780621/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780622] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780622] destination_number(6578xxxx) =~ /31780622/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780623] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780623] destination_number(6578xxxx) =~ /31780623/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780624] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780624] destination_number(6578xxxx) =~ /31780624/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780625] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780625] destination_number(6578xxxx) =~ /31780625/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780626] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780626] destination_number(6578xxxx) =~ /31780626/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780627] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780627] destination_number(6578xxxx) =~ /31780627/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780628] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780628] destination_number(6578xxxx) =~ /31780628/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780629] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780629] destination_number(6578xxxx) =~ /31780629/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780630] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780630] destination_number(6578xxxx) =~ /31780630/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780631] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780631] destination_number(6578xxxx) =~ /31780631/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780632] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780632] destination_number(6578xxxx) =~ /31780632/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780633] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780633] destination_number(6578xxxx) =~ /31780633/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780634] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780634] destination_number(6578xxxx) =~ /31780634/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780635] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780635] destination_number(6578xxxx) =~ /31780635/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780636] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780636] destination_number(6578xxxx) =~ /31780636/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780637] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780637] destination_number(6578xxxx) =~ /31780637/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780639] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (PASS) [DID_31780639] context(public) =~ /public/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780639] destination_number(6578xxxx) =~ /31780639/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780640] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780640] destination_number(6578xxxx) =~ /31780640/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780641] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780641] destination_number(6578xxxx) =~ /31780641/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780642] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780642] destination_number(6578xxxx) =~ /31780642/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780643] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780643] destination_number(6578xxxx) =~ /31780643/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780644] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780644] destination_number(6578xxxx) =~ /31780644/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780645] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780645] destination_number(6578xxxx) =~ /31780645/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780646] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780646] destination_number(6578xxxx) =~ /31780646/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780647] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780647] destination_number(6578xxxx) =~ /31780647/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780648] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780648] destination_number(6578xxxx) =~ /31780648/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780649] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780649] destination_number(6578xxxx) =~ /31780649/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780650] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780650] destination_number(6578xxxx) =~ /31780650/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780651] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780651] destination_number(6578xxxx) =~ /31780651/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780652] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780652] destination_number(6578xxxx) =~ /31780652/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780653] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780653] destination_number(6578xxxx) =~ /31780653/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780654] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780654] destination_number(6578xxxx) =~ /31780654/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780655] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780655] destination_number(6578xxxx) =~ /31780655/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780656] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780656] destination_number(6578xxxx) =~ /31780656/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780657] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780657] destination_number(6578xxxx) =~ /31780657/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780658] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780658] destination_number(6578xxxx) =~ /31780658/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780659] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780659] destination_number(6578xxxx) =~ /31780659/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780660] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780660] destination_number(6578xxxx) =~ /31780660/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780661] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780661] destination_number(6578xxxx) =~ /31780661/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780662] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780662] destination_number(6578xxxx) =~ /31780662/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780663] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780663] destination_number(6578xxxx) =~ /31780663/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780664] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780664] destination_number(6578xxxx) =~ /31780664/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780665] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780665] destination_number(6578xxxx) =~ /31780665/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780666] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780666] destination_number(6578xxxx) =~ /31780666/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780667] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780667] destination_number(6578xxxx) =~ /31780667/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780668] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780668] destination_number(6578xxxx) =~ /31780668/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780669] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780669] destination_number(6578xxxx) =~ /31780669/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780670] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780670] destination_number(6578xxxx) =~ /31780670/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780671] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780671] destination_number(6578xxxx) =~ /31780671/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780672] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780672] destination_number(6578xxxx) =~ /31780672/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780673] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780673] destination_number(6578xxxx) =~ /31780673/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780674] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780674] destination_number(6578xxxx) =~ /31780674/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780675] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780675] destination_number(6578xxxx) =~ /31780675/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780676] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780676] destination_number(6578xxxx) =~ /31780676/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780677] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780677] destination_number(6578xxxx) =~ /31780677/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780678] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780678] destination_number(6578xxxx) =~ /31780678/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780679] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780679] destination_number(6578xxxx) =~ /31780679/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780680] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780680] destination_number(6578xxxx) =~ /31780680/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780681] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780681] destination_number(6578xxxx) =~ /31780681/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780682] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780682] destination_number(6578xxxx) =~ /31780682/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780683] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780683] destination_number(6578xxxx) =~ /31780683/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780684] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780684] destination_number(6578xxxx) =~ /31780684/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780685] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780685] destination_number(6578xxxx) =~ /31780685/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780686] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780686] destination_number(6578xxxx) =~ /31780686/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780687] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780687] destination_number(6578xxxx) =~ /31780687/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780688] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780688] destination_number(6578xxxx) =~ /31780688/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780689] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780689] destination_number(6578xxxx) =~ /31780689/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780690] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780690] destination_number(6578xxxx) =~ /31780690/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780691] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780691] destination_number(6578xxxx) =~ /31780691/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780692] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780692] destination_number(6578xxxx) =~ /31780692/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780693] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780693] destination_number(6578xxxx) =~ /31780693/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780694] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780694] destination_number(6578xxxx) =~ /31780694/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780695] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780695] destination_number(6578xxxx) =~ /31780695/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780696] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780696] destination_number(6578xxxx) =~ /31780696/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780697] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780697] destination_number(6578xxxx) =~ /31780697/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780698] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780698] destination_number(6578xxxx) =~ /31780698/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_31780699] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_31780699] destination_number(6578xxxx) =~ /31780699/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->suspension_03] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [suspension_03] destination_number(6578xxxx) =~ /4323/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->suspension_04] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [suspension_04] destination_number(6578xxxx) =~ /4119/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->suspension_07] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [suspension_07] destination_number(6578xxxx) =~ /3175x208/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->suspension_08] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [suspension_08] destination_number(6578xxxx) =~ /31780526/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->suspension_09] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [suspension_09] destination_number(6578xxxx) =~ /4241/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->suspension_10] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [suspension_10] destination_number(6578xxxx) =~ /31780566/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->suspension_11] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [suspension_11] destination_number(6578xxxx) =~ /31780646/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->suspension_12] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [suspension_12] destination_number(6578xxxx) =~ /31780626/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->suspension_13] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [suspension_13] destination_number(6578xxxx) =~ /4156/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x100] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x100] destination_number(6578xxxx) =~ /3175x100/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x101] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x101] destination_number(6578xxxx) =~ /3175x101/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x102] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x102] destination_number(6578xxxx) =~ /3175x102/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x103] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x103] destination_number(6578xxxx) =~ /3175x103/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x104] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x104] destination_number(6578xxxx) =~ /3175x104/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x105] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x105] destination_number(6578xxxx) =~ /3175x105/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x106] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x106] destination_number(6578xxxx) =~ /3175x106/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x107] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x107] destination_number(6578xxxx) =~ /3175x107/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x108] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x108] destination_number(6578xxxx) =~ /3175x108/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x109] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x109] destination_number(6578xxxx) =~ /3175x109/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x110] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x110] destination_number(6578xxxx) =~ /3175x110/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x111] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x111] destination_number(6578xxxx) =~ /3175x111/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x112] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x112] destination_number(6578xxxx) =~ /3175x112/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x113] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x113] destination_number(6578xxxx) =~ /3175x113/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x114] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x114] destination_number(6578xxxx) =~ /3175x114/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x115] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x115] destination_number(6578xxxx) =~ /3175x115/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x116] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x116] destination_number(6578xxxx) =~ /3175x116/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x117] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x117] destination_number(6578xxxx) =~ /3175x117/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x118] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x118] destination_number(6578xxxx) =~ /3175x118/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x119] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x119] destination_number(6578xxxx) =~ /3175x119/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x120] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x120] destination_number(6578xxxx) =~ /3175x120/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x121] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x121] destination_number(6578xxxx) =~ /3175x121/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x122] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x122] destination_number(6578xxxx) =~ /3175x122/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x123] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x123] destination_number(6578xxxx) =~ /3175x123/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x124] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x124] destination_number(6578xxxx) =~ /3175x124/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x125] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x125] destination_number(6578xxxx) =~ /3175x125/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x126] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x126] destination_number(6578xxxx) =~ /3175x126/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x127] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x127] destination_number(6578xxxx) =~ /3175x127/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x128] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x128] destination_number(6578xxxx) =~ /3175x128/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x129] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x129] destination_number(6578xxxx) =~ /3175x129/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x130] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x130] destination_number(6578xxxx) =~ /3175x130/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x131] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x131] destination_number(6578xxxx) =~ /3175x131/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x132] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x132] destination_number(6578xxxx) =~ /3175x132/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->fax_receive] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [fax_receive] destination_number(6578xxxx) =~ /3175x133/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x134] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x134] destination_number(6578xxxx) =~ /3175x134/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x135] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x135] destination_number(6578xxxx) =~ /3175x135/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x136] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x136] destination_number(6578xxxx) =~ /3175x136/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x137] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x137] destination_number(6578xxxx) =~ /3175x137/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x138] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x138] destination_number(6578xxxx) =~ /3175x138/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x139] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x139] destination_number(6578xxxx) =~ /3175x139/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x140] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x140] destination_number(6578xxxx) =~ /3175x140/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x141] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x141] destination_number(6578xxxx) =~ /3175x141/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x142] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x142] destination_number(6578xxxx) =~ /3175x142/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x143] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x143] destination_number(6578xxxx) =~ /3175x143/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x144] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x144] destination_number(6578xxxx) =~ /3175x144/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x145] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x145] destination_number(6578xxxx) =~ /3175x145/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x146] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x146] destination_number(6578xxxx) =~ /3175x146/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x147] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x147] destination_number(6578xxxx) =~ /3175x147/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x148] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x148] destination_number(6578xxxx) =~ /3175x148/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x149] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x149] destination_number(6578xxxx) =~ /3175x149/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x150] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x150] destination_number(6578xxxx) =~ /3175x150/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x151] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x151] destination_number(6578xxxx) =~ /3175x151/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x152] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x152] destination_number(6578xxxx) =~ /3175x152/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x153] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x153] destination_number(6578xxxx) =~ /3175x153/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x154] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x154] destination_number(6578xxxx) =~ /3175x154/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x155] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x155] destination_number(6578xxxx) =~ /3175x155/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x156] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x156] destination_number(6578xxxx) =~ /3175x156/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x157] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x157] destination_number(6578xxxx) =~ /3175x157/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x158] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x158] destination_number(6578xxxx) =~ /3175x158/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x159] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x159] destination_number(6578xxxx) =~ /3175x159/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x160] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x160] destination_number(6578xxxx) =~ /3175x160/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x161] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x161] destination_number(6578xxxx) =~ /3175x161/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x162] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x162] destination_number(6578xxxx) =~ /3175x162/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x163] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x163] destination_number(6578xxxx) =~ /3175x163/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x164] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x164] destination_number(6578xxxx) =~ /3175x164/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x165] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x165] destination_number(6578xxxx) =~ /3175x165/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x166] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x166] destination_number(6578xxxx) =~ /3175x166/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x167] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x167] destination_number(6578xxxx) =~ /3175x167/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x168] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x168] destination_number(6578xxxx) =~ /3175x168/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x169] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x169] destination_number(6578xxxx) =~ /3175x169/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x170] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x170] destination_number(6578xxxx) =~ /3175x170/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x171] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x171] destination_number(6578xxxx) =~ /3175x171/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x172] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x172] destination_number(6578xxxx) =~ /3175x172/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x173] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x173] destination_number(6578xxxx) =~ /3175x173/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x174] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x174] destination_number(6578xxxx) =~ /3175x174/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x175] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x175] destination_number(6578xxxx) =~ /3175x175/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x176] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x176] destination_number(6578xxxx) =~ /3175x176/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x177] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x177] destination_number(6578xxxx) =~ /3175x177/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x178] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x178] destination_number(6578xxxx) =~ /3175x178/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x179] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x179] destination_number(6578xxxx) =~ /3175x179/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x180] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x180] destination_number(6578xxxx) =~ /3175x180/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x181] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x181] destination_number(6578xxxx) =~ /3175x181/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x182] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x182] destination_number(6578xxxx) =~ /3175x182/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x183] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x183] destination_number(6578xxxx) =~ /3175x183/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x184] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x184] destination_number(6578xxxx) =~ /3175x184/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x185] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x185] destination_number(6578xxxx) =~ /3175x185/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x186] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x186] destination_number(6578xxxx) =~ /3175x186/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x187] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x187] destination_number(6578xxxx) =~ /3175x187/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x188] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x188] destination_number(6578xxxx) =~ /3175x188/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x189] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x189] destination_number(6578xxxx) =~ /3175x189/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x190] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x190] destination_number(6578xxxx) =~ /3175x190/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x191] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x191] destination_number(6578xxxx) =~ /3175x191/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x192] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x192] destination_number(6578xxxx) =~ /3175x192/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x193] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x193] destination_number(6578xxxx) =~ /3175x193/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x194] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x194] destination_number(6578xxxx) =~ /3175x194/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x195] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x195] destination_number(6578xxxx) =~ /3175x195/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x196] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x196] destination_number(6578xxxx) =~ /3175x196/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x197] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x197] destination_number(6578xxxx) =~ /3175x197/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x198] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x198] destination_number(6578xxxx) =~ /3175x198/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x199] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x199] destination_number(6578xxxx) =~ /3175x199/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x200] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x200] destination_number(6578xxxx) =~ /3175x200/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x201] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x201] destination_number(6578xxxx) =~ /3175x201/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x202] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x202] destination_number(6578xxxx) =~ /3175x202/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x203] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x203] destination_number(6578xxxx) =~ /3175x203/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x204] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x204] destination_number(6578xxxx) =~ /3175x204/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x205] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x205] destination_number(6578xxxx) =~ /3175x205/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x206] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x206] destination_number(6578xxxx) =~ /3175x206/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x207] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x207] destination_number(6578xxxx) =~ /3175x207/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x208] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x208] destination_number(6578xxxx) =~ /3175x208/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x209] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x209] destination_number(6578xxxx) =~ /3175x209/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x210] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x210] destination_number(6578xxxx) =~ /3175x210/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x211] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x211] destination_number(6578xxxx) =~ /3175x211/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x212] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x212] destination_number(6578xxxx) =~ /3175x212/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x213] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x213] destination_number(6578xxxx) =~ /3175x213/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x214] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x214] destination_number(6578xxxx) =~ /3175x214/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x215] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x215] destination_number(6578xxxx) =~ /3175x215/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x216] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x216] destination_number(6578xxxx) =~ /3175x216/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x217] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x217] destination_number(6578xxxx) =~ /3175x217/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x218] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x218] destination_number(6578xxxx) =~ /3175x218/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x219] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x219] destination_number(6578xxxx) =~ /3175x219/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x220] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x220] destination_number(6578xxxx) =~ /3175x220/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x221] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x221] destination_number(6578xxxx) =~ /3175x221/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x222] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x222] destination_number(6578xxxx) =~ /3175x222/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x223] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x223] destination_number(6578xxxx) =~ /3175x223/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x224] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x224] destination_number(6578xxxx) =~ /3175x224/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x225] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x225] destination_number(6578xxxx) =~ /3175x225/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x226] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x226] destination_number(6578xxxx) =~ /3175x226/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x227] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x227] destination_number(6578xxxx) =~ /3175x227/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x228] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x228] destination_number(6578xxxx) =~ /3175x228/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x229] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x229] destination_number(6578xxxx) =~ /3175x229/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x230] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x230] destination_number(6578xxxx) =~ /3175x230/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x231] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x231] destination_number(6578xxxx) =~ /3175x231/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x232] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x232] destination_number(6578xxxx) =~ /3175x232/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x233] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x233] destination_number(6578xxxx) =~ /3175x233/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x234] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x234] destination_number(6578xxxx) =~ /3175x234/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x235] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x235] destination_number(6578xxxx) =~ /3175x235/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x236] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x236] destination_number(6578xxxx) =~ /3175x236/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x237] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x237] destination_number(6578xxxx) =~ /3175x237/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x238] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x238] destination_number(6578xxxx) =~ /3175x238/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x239] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x239] destination_number(6578xxxx) =~ /3175x239/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x240] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x240] destination_number(6578xxxx) =~ /3175x240/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x241] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x241] destination_number(6578xxxx) =~ /3175x241/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x242] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x242] destination_number(6578xxxx) =~ /3175x242/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x243] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x243] destination_number(6578xxxx) =~ /3175x243/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x244] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x244] destination_number(6578xxxx) =~ /3175x244/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x245] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x245] destination_number(6578xxxx) =~ /3175x245/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x246] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x246] destination_number(6578xxxx) =~ /3175x246/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x247] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x247] destination_number(6578xxxx) =~ /3175x247/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x248] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x248] destination_number(6578xxxx) =~ /3175x248/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x249] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x249] destination_number(6578xxxx) =~ /3175x249/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x250] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x250] destination_number(6578xxxx) =~ /3175x250/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x251] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x251] destination_number(6578xxxx) =~ /3175x251/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x252] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x252] destination_number(6578xxxx) =~ /3175x252/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x253] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x253] destination_number(6578xxxx) =~ /3175x253/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x254] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x254] destination_number(6578xxxx) =~ /3175x254/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x255] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x255] destination_number(6578xxxx) =~ /3175x255/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x256] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x256] destination_number(6578xxxx) =~ /3175x256/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x257] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x257] destination_number(6578xxxx) =~ /3175x257/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->Fax_3175x257] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [Fax_3175x257] destination_number(6578xxxx) =~ /^fax$/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x258] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x258] destination_number(6578xxxx) =~ /3175x258/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x259] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x259] destination_number(6578xxxx) =~ /3175x259/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x260] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x260] destination_number(6578xxxx) =~ /3175x260/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x261] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x261] destination_number(6578xxxx) =~ /3175x261/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x262] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x262] destination_number(6578xxxx) =~ /3175x262/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x263] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x263] destination_number(6578xxxx) =~ /3175x263/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x264] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x264] destination_number(6578xxxx) =~ /3175x264/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x265] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x265] destination_number(6578xxxx) =~ /3175x265/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x266] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x266] destination_number(6578xxxx) =~ /3175x266/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x267] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x267] destination_number(6578xxxx) =~ /3175x267/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x268] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x268] destination_number(6578xxxx) =~ /3175x268/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x269] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x269] destination_number(6578xxxx) =~ /3175x269/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x270] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x270] destination_number(6578xxxx) =~ /3175x270/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x271] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x271] destination_number(6578xxxx) =~ /3175x271/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x272] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x272] destination_number(6578xxxx) =~ /3175x272/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x273] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x273] destination_number(6578xxxx) =~ /3175x273/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x274] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x274] destination_number(6578xxxx) =~ /3175x274/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x275] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x275] destination_number(6578xxxx) =~ /3175x275/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x276] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x276] destination_number(6578xxxx) =~ /3175x276/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x277] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x277] destination_number(6578xxxx) =~ /3175x277/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x278] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x278] destination_number(6578xxxx) =~ /3175x278/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x279] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x279] destination_number(6578xxxx) =~ /3175x279/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x280] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x280] destination_number(6578xxxx) =~ /3175x280/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x281] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x281] destination_number(6578xxxx) =~ /3175x281/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x282] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x282] destination_number(6578xxxx) =~ /3175x282/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x283] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x283] destination_number(6578xxxx) =~ /3175x283/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x284] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x284] destination_number(6578xxxx) =~ /3175x284/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x285] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x285] destination_number(6578xxxx) =~ /3175x285/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x286] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x286] destination_number(6578xxxx) =~ /3175x286/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x287] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x287] destination_number(6578xxxx) =~ /3175x287/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x288] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x288] destination_number(6578xxxx) =~ /3175x288/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x289] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x289] destination_number(6578xxxx) =~ /3175x289/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x290] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x290] destination_number(6578xxxx) =~ /3175x290/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x291] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x291] destination_number(6578xxxx) =~ /3175x291/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x292] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x292] destination_number(6578xxxx) =~ /3175x292/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x293] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x293] destination_number(6578xxxx) =~ /3175x293/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x294] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x294] destination_number(6578xxxx) =~ /3175x294/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x295] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x295] destination_number(6578xxxx) =~ /3175x295/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x296] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x296] destination_number(6578xxxx) =~ /3175x296/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x297] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x297] destination_number(6578xxxx) =~ /3175x297/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x298] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x298] destination_number(6578xxxx) =~ /3175x298/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x299] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x299] destination_number(6578xxxx) =~ /3175x299/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x300] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x300] destination_number(6578xxxx) =~ /3175x300/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x301] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x301] destination_number(6578xxxx) =~ /3175x301/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x302] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x302] destination_number(6578xxxx) =~ /3175x302/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x303] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x303] destination_number(6578xxxx) =~ /3175x303/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x304] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x304] destination_number(6578xxxx) =~ /3175x304/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x305] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x305] destination_number(6578xxxx) =~ /3175x305/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x306] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x306] destination_number(6578xxxx) =~ /3175x306/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x307] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x307] destination_number(6578xxxx) =~ /3175x307/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x308] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x308] destination_number(6578xxxx) =~ /3175x308/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x309] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x309] destination_number(6578xxxx) =~ /3175x309/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x310] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x310] destination_number(6578xxxx) =~ /3175x310/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x311] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x311] destination_number(6578xxxx) =~ /3175x311/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x312] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x312] destination_number(6578xxxx) =~ /3175x312/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x313] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x313] destination_number(6578xxxx) =~ /3175x313/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x314] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x314] destination_number(6578xxxx) =~ /3175x314/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x315] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x315] destination_number(6578xxxx) =~ /3175x315/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x316] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x316] destination_number(6578xxxx) =~ /3175x316/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x317] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x317] destination_number(6578xxxx) =~ /3175x317/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x318] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x318] destination_number(6578xxxx) =~ /3175x318/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x319] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x319] destination_number(6578xxxx) =~ /3175x319/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x320] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x320] destination_number(6578xxxx) =~ /3175x320/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x321] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x321] destination_number(6578xxxx) =~ /3175x321/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x322] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x322] destination_number(6578xxxx) =~ /3175x322/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x323] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x323] destination_number(6578xxxx) =~ /3175x323/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x324] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x324] destination_number(6578xxxx) =~ /3175x324/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x325] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x325] destination_number(6578xxxx) =~ /3175x325/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x326] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x326] destination_number(6578xxxx) =~ /3175x326/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x327] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x327] destination_number(6578xxxx) =~ /3175x327/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x328] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x328] destination_number(6578xxxx) =~ /3175x328/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x329] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x329] destination_number(6578xxxx) =~ /3175x329/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x330] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x330] destination_number(6578xxxx) =~ /3175x330/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x331] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x331] destination_number(6578xxxx) =~ /3175x331/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x332] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x332] destination_number(6578xxxx) =~ /3175x332/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x333] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x333] destination_number(6578xxxx) =~ /3175x333/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x334] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x334] destination_number(6578xxxx) =~ /3175x334/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x335] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x335] destination_number(6578xxxx) =~ /3175x335/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x336] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x336] destination_number(6578xxxx) =~ /3175x336/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x337] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x337] destination_number(6578xxxx) =~ /3175x337/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x338] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x338] destination_number(6578xxxx) =~ /3175x338/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x339] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x339] destination_number(6578xxxx) =~ /3175x339/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x340] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x340] destination_number(6578xxxx) =~ /3175x340/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x341] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x341] destination_number(6578xxxx) =~ /3175x341/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x342] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x342] destination_number(6578xxxx) =~ /3175x342/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x343] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x343] destination_number(6578xxxx) =~ /3175x343/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x344] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x344] destination_number(6578xxxx) =~ /3175x344/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x345] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x345] destination_number(6578xxxx) =~ /3175x345/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x346] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x346] destination_number(6578xxxx) =~ /3175x346/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x347] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x347] destination_number(6578xxxx) =~ /3175x347/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x348] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x348] destination_number(6578xxxx) =~ /3175x348/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x349] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x349] destination_number(6578xxxx) =~ /3175x349/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x350] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x350] destination_number(6578xxxx) =~ /3175x350/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x351] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x351] destination_number(6578xxxx) =~ /3175x351/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x352] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x352] destination_number(6578xxxx) =~ /3175x352/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x353] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x353] destination_number(6578xxxx) =~ /3175x353/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x354] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x354] destination_number(6578xxxx) =~ /3175x354/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x355] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x355] destination_number(6578xxxx) =~ /3175x355/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x356] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x356] destination_number(6578xxxx) =~ /3175x356/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x357] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x357] destination_number(6578xxxx) =~ /3175x357/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x358] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x358] destination_number(6578xxxx) =~ /3175x358/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x359] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x359] destination_number(6578xxxx) =~ /3175x359/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x360] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x360] destination_number(6578xxxx) =~ /3175x360/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x361] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x361] destination_number(6578xxxx) =~ /3175x361/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x362] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x362] destination_number(6578xxxx) =~ /3175x362/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x363] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x363] destination_number(6578xxxx) =~ /3175x363/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x364] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x364] destination_number(6578xxxx) =~ /3175x364/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x365] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x365] destination_number(6578xxxx) =~ /3175x365/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x366] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x366] destination_number(6578xxxx) =~ /3175x366/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x367] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x367] destination_number(6578xxxx) =~ /3175x367/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_=3175x368] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_=3175x368] destination_number(6578xxxx) =~ /3175x368/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x369] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x369] destination_number(6578xxxx) =~ /3175x369/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x370] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x370] destination_number(6578xxxx) =~ /3175x370/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x371] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x371] destination_number(6578xxxx) =~ /3175x371/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x372] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x372] destination_number(6578xxxx) =~ /3175x372/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x373] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x373] destination_number(6578xxxx) =~ /3175x373/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x374] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x374] destination_number(6578xxxx) =~ /3175x374/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x375] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x375] destination_number(6578xxxx) =~ /3175x375/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x376] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x376] destination_number(6578xxxx) =~ /3175x376/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x377] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x377] destination_number(6578xxxx) =~ /3175x377/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x378] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x378] destination_number(6578xxxx) =~ /3175x378/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x379] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x379] destination_number(6578xxxx) =~ /3175x379/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x380] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x380] destination_number(6578xxxx) =~ /3175x380/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x381] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x381] destination_number(6578xxxx) =~ /3175x381/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x382] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x382] destination_number(6578xxxx) =~ /3175x382/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x383] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x383] destination_number(6578xxxx) =~ /3175x383/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x384] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x384] destination_number(6578xxxx) =~ /3175x384/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x385] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x385] destination_number(6578xxxx) =~ /3175x385/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x386] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x386] destination_number(6578xxxx) =~ /3175x386/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x387] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x387] destination_number(6578xxxx) =~ /3175x387/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x388] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x388] destination_number(6578xxxx) =~ /3175x388/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x389] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x389] destination_number(6578xxxx) =~ /3175x389/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x390] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x390] destination_number(6578xxxx) =~ /3175x390/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x391] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x391] destination_number(6578xxxx) =~ /3175x391/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x392] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x392] destination_number(6578xxxx) =~ /3175x392/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x393] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x393] destination_number(6578xxxx) =~ /3175x393/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x394] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x394] destination_number(6578xxxx) =~ /3175x394/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x395] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x395] destination_number(6578xxxx) =~ /3175x395/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x396] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x396] destination_number(6578xxxx) =~ /3175x396/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x397] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x397] destination_number(6578xxxx) =~ /3175x397/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x398] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x398] destination_number(6578xxxx) =~ /3175x398/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->DID_3175x399] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [DID_3175x399] destination_number(6578xxxx) =~ /3175x399/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->IVR_demo_inbound] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (PASS) [IVR_demo_inbound] context(public) =~ /public/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [IVR_demo_inbound] destination_number(6578xxxx) =~ /85212345678/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->m800_IVR_4001200638] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (PASS) [m800_IVR_4001200638] context(public) =~ /public/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [m800_IVR_4001200638] destination_number(6578xxxx) =~ /4001200638/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->m800_IVR_0800638] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (PASS) [m800_IVR_0800638] context(public) =~ /public/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [m800_IVR_0800638] destination_number(6578xxxx) =~ /0800638/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->m800_IVR_31780638] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (PASS) [m800_IVR_31780638] context(public) =~ /public/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [m800_IVR_31780638] destination_number(6578xxxx) =~ /31780638/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->IVR_demo] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (PASS) [IVR_demo] context(public) =~ /public/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [IVR_demo] destination_number(6578xxxx) =~ /023128001/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->IVR2] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (PASS) [IVR2] context(public) =~ /public/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [IVR2] destination_number(6578xxxx) =~ /023128002/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->IVR_demo_inbound] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (PASS) [IVR_demo_inbound] context(public) =~ /public/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [IVR_demo_inbound] destination_number(6578xxxx) =~ /29022400/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->IVR_demo_inbound] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (PASS) [IVR_demo_inbound] context(public) =~ /public/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [IVR_demo_inbound] destination_number(6578xxxx) =~ /85212345678/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->m800_IVR_4001200638] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (PASS) [m800_IVR_4001200638] context(public) =~ /public/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [m800_IVR_4001200638] destination_number(6578xxxx) =~ /4001200638/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->m800_IVR_0800638] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (PASS) [m800_IVR_0800638] context(public) =~ /public/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [m800_IVR_0800638] destination_number(6578xxxx) =~ /0800638/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->m800_IVR_31780638] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (PASS) [m800_IVR_31780638] context(public) =~ /public/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [m800_IVR_31780638] destination_number(6578xxxx) =~ /31780638/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->IVR_demo] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (PASS) [IVR_demo] context(public) =~ /public/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [IVR_demo] destination_number(6578xxxx) =~ /023128001/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 parsing [public->IVR2] continue=false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (PASS) [IVR2] context(public) =~ /public/ break=on-false Dialplan: sofia/external/3175x102 at 203.x.x.80 Regex (FAIL) [IVR2] destination_number(6578xxxx) =~ /023128002/ break=on-false 2012-02-09 17:09:17.199902 [INFO] switch_core_state_machine.c:177 No Route, Aborting 2012-02-09 17:09:17.199902 [DEBUG] switch_channel.c:2848 (sofia/external/3175x102 at 203.x.x.80) Callstate Change RINGING -> HANGUP 2012-02-09 17:09:17.199902 [NOTICE] switch_core_state_machine.c:178 Hangup sofia/external/3175x102 at 203.x.x.80 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2012-02-09 17:09:17.199902 [DEBUG] switch_channel.c:2871 Send signal sofia/external/3175x102 at 203.x.x.80 [KILL] 2012-02-09 17:09:17.199902 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/3175x102 at 203.x.x.80 [BREAK] 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:410 (sofia/external/3175x102 at 203.x.x.80) State ROUTING going to sleep 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:362 (sofia/external/3175x102 at 203.x.x.80) Running State Change CS_HANGUP 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:602 (sofia/external/3175x102 at 203.x.x.80) State HANGUP 2012-02-09 17:09:17.199902 [DEBUG] mod_sofia.c:469 Channel sofia/external/3175x102 at 203.x.x.80 hanging up, cause: NO_ROUTE_DESTINATION 2012-02-09 17:09:17.199902 [DEBUG] mod_sofia.c:534 Responding to INVITE with: 404 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:47 sofia/external/3175x102 at 203.x.x.80 Standard HANGUP, cause: NO_ROUTE_DESTINATION 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:602 (sofia/external/3175x102 at 203.x.x.80) State HANGUP going to sleep 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:393 (sofia/external/3175x102 at 203.x.x.80) State Change CS_HANGUP -> CS_REPORTING 2012-02-09 17:09:17.199902 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/3175x102 at 203.x.x.80 [BREAK] 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:362 (sofia/external/3175x102 at 203.x.x.80) Running State Change CS_REPORTING 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:662 (sofia/external/3175x102 at 203.x.x.80) State REPORTING 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:79 sofia/external/3175x102 at 203.x.x.80 Standard REPORTING, cause: NO_ROUTE_DESTINATION 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:662 (sofia/external/3175x102 at 203.x.x.80) State REPORTING going to sleep 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:387 (sofia/external/3175x102 at 203.x.x.80) State Change CS_REPORTING -> CS_DESTROY 2012-02-09 17:09:17.199902 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/3175x102 at 203.x.x.80 [BREAK] 2012-02-09 17:09:17.199902 [DEBUG] switch_core_session.c:1380 Session 23374 (sofia/external/3175x102 at 203.x.x.80) Locked, Waiting on external entities 2012-02-09 17:09:17.199902 [NOTICE] switch_core_session.c:1398 Session 23374 (sofia/external/3175x102 at 203.x.x.80) Ended 2012-02-09 17:09:17.199902 [NOTICE] switch_core_session.c:1400 Close Channel sofia/external/3175x102 at 203.x.x.80 [CS_DESTROY] 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:491 (sofia/external/3175x102 at 203.x.x.80) Callstate Change HANGUP -> DOWN 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:494 (sofia/external/3175x102 at 203.x.x.80) Running State Change CS_DESTROY 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:504 (sofia/external/3175x102 at 203.x.x.80) State DESTROY 2012-02-09 17:09:17.199902 [DEBUG] mod_sofia.c:374 sofia/external/3175x102 at 203.x.x.80 SOFIA DESTROY 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:86 sofia/external/3175x102 at 203.x.x.80 Standard DESTROY 2012-02-09 17:09:17.199902 [DEBUG] switch_core_state_machine.c:504 (sofia/external/3175x102 at 203.x.x.80) State DESTROY going to sleep 2012-02-09 17:09:17.327945 [DEBUG] switch_core_io.c:340 Setting BUG Codec PCMU:0 2012-02-09 17:09:17.519934 [INFO] switch_rtp.c:3188 Auto Changing port from 10.0.166.104:16384 to 118.x.x.115:2590 2012-02-09 17:09:20.136096 [DEBUG] switch_core_session.c:875 Send signal sofia/external/6578xxxx at 203.x.x.80 [BREAK] 2012-02-09 17:09:20.136096 [DEBUG] switch_core_session.c:875 Send signal sofia/external/6578xxxx at 203.x.x.80 [BREAK] 2012-02-09 17:09:20.136096 [DEBUG] sofia.c:5512 Channel sofia/external/6578xxxx at 203.x.x.80 entering state [proceeding][183] 2012-02-09 17:09:20.136096 [DEBUG] sofia.c:5523 Remote SDP: v=0 o=CiscoSystemsSIP-GW-UserAgent 1393 6748 IN IP4 203.x.x.80 s=SIP Call c=IN IP4 203.x.x.80 t=0 0 m=audio 17294 RTP/AVP 0 101 c=IN IP4 203.x.x.80 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:4374 Activate Buggy RFC2833 Mode! 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:2919 Set Codec sofia/external/6578xxxx at 203.x.x.80 PCMU/8000 20 ms 160 samples 64000 bits 2012-02-09 17:09:20.136096 [DEBUG] switch_core_codec.c:111 sofia/external/6578xxxx at 203.x.x.80 Original read codec set to PCMU:0 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:4912 Set 2833 dtmf send payload to 101 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:3171 AUDIO RTP [sofia/external/6578xxxx at 203.x.x.80] 203.x.x.91 port 28814 -> 203.x.x.80 port 17294 codec: 0 ms: 20 2012-02-09 17:09:20.136096 [DEBUG] switch_rtp.c:1669 Not using a timer 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:3435 Set 2833 dtmf send payload to 101 2012-02-09 17:09:20.136096 [DEBUG] sofia_glue.c:3441 Set 2833 dtmf receive payload to 101 2012-02-09 17:09:20.136096 [NOTICE] sofia_glue.c:3945 Pre-Answer sofia/external/6578xxxx at 203.x.x.80! 2012-02-09 17:09:20.136096 [DEBUG] switch_channel.c:2932 (sofia/external/6578xxxx at 203.x.x.80) Callstate Change RINGING -> EARLY 2012-02-09 17:09:20.136096 [DEBUG] switch_channel.c:2974 Send signal sofia/internal/3175x102 at 203.x.x.81:5060 [BREAK] 2012-02-09 17:09:20.148158 [DEBUG] switch_ivr_originate.c:3266 Originate Resulted in Success: [sofia/external/6578xxxx at 203.x.x.80] 2012-02-09 17:09:20.148158 [DEBUG] switch_core_session.c:729 Send signal sofia/external/6578xxxx at 203.x.x.80 [BREAK] 2012-02-09 17:09:20.148158 [DEBUG] switch_core_session.c:729 Send signal sofia/internal/3175x102 at 203.x.x.81:5060 [BREAK] 2012-02-09 17:09:20.148158 [DEBUG] switch_ivr_bridge.c:1327 (sofia/external/6578xxxx at 203.x.x.80) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2012-02-09 17:09:20.148158 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/6578xxxx at 203.x.x.80 [BREAK] 2012-02-09 17:09:20.148158 [DEBUG] switch_core_state_machine.c:362 (sofia/external/6578xxxx at 203.x.x.80) Running State Change CS_EXCHANGE_MEDIA 2012-02-09 17:09:20.148158 [DEBUG] switch_core_state_machine.c:420 (sofia/external/6578xxxx at 203.x.x.80) State EXCHANGE_MEDIA 2012-02-09 17:09:20.148158 [DEBUG] mod_sofia.c:578 SOFIA EXCHANGE_MEDIA 2012-02-09 17:09:20.184179 [DEBUG] switch_rtp.c:3205 Correct ip/port confirmed. 2012-02-09 17:09:22.244227 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/3175x102 at 203.x.x.81:5060 [BREAK] 2012-02-09 17:09:22.244227 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/3175x102 at 203.x.x.81:5060 [BREAK] 2012-02-09 17:09:22.244227 [DEBUG] switch_core_session.c:875 Send signal sofia/internal/3175x102 at 203.x.x.81:5060 [BREAK] 2012-02-09 17:09:22.268284 [DEBUG] sofia.c:5512 Channel sofia/internal/3175x102 at 203.x.x.81:5060 entering state [terminated][487] 2012-02-09 17:09:22.268284 [DEBUG] switch_channel.c:2848 (sofia/internal/3175x102 at 203.x.x.81:5060) Callstate Change EARLY -> HANGUP 2012-02-09 17:09:22.268284 [NOTICE] sofia.c:6276 Hangup sofia/internal/3175x102 at 203.x.x.81:5060 [CS_EXECUTE] [ORIGINATOR_CANCEL] 2012-02-09 17:09:22.268284 [DEBUG] switch_channel.c:2871 Send signal sofia/internal/3175x102 at 203.x.x.81:5060 [KILL] 2012-02-09 17:09:22.268284 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/3175x102 at 203.x.x.81:5060 [BREAK] 2012-02-09 17:09:22.268284 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/3175x102 at 203.x.x.81:5060] 2012-02-09 17:09:22.268284 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/external/6578xxxx at 203.x.x.80 [BREAK] 2012-02-09 17:09:22.268284 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/external/6578xxxx at 203.x.x.80] 2012-02-09 17:09:22.268284 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/internal/3175x102 at 203.x.x.81:5060 [BREAK] 2012-02-09 17:09:22.268284 [DEBUG] switch_channel.c:2848 (sofia/external/6578xxxx at 203.x.x.80) Callstate Change EARLY -> HANGUP 2012-02-09 17:09:22.268284 [NOTICE] switch_ivr_bridge.c:667 Hangup sofia/external/6578xxxx at 203.x.x.80 [CS_EXCHANGE_MEDIA] [ORIGINATOR_CANCEL] 2012-02-09 17:09:22.268284 [DEBUG] switch_channel.c:2871 Send signal sofia/external/6578xxxx at 203.x.x.80 [KILL] 2012-02-09 17:09:22.268284 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/6578xxxx at 203.x.x.80 [BREAK] 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:420 (sofia/external/6578xxxx at 203.x.x.80) State EXCHANGE_MEDIA going to sleep 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:362 (sofia/external/6578xxxx at 203.x.x.80) Running State Change CS_HANGUP 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:602 (sofia/external/6578xxxx at 203.x.x.80) State HANGUP 2012-02-09 17:09:22.268284 [DEBUG] mod_sofia.c:463 sofia/external/6578xxxx at 203.x.x.80 Overriding SIP cause 487 with 487 from the other leg 2012-02-09 17:09:22.268284 [DEBUG] mod_sofia.c:469 Channel sofia/external/6578xxxx at 203.x.x.80 hanging up, cause: ORIGINATOR_CANCEL 2012-02-09 17:09:22.268284 [DEBUG] mod_sofia.c:523 Sending CANCEL to sofia/external/6578xxxx at 203.x.x.80 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:47 sofia/external/6578xxxx at 203.x.x.80 Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:602 (sofia/external/6578xxxx at 203.x.x.80) State HANGUP going to sleep 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:393 (sofia/external/6578xxxx at 203.x.x.80) State Change CS_HANGUP -> CS_REPORTING 2012-02-09 17:09:22.268284 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/6578xxxx at 203.x.x.80 [BREAK] 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:362 (sofia/external/6578xxxx at 203.x.x.80) Running State Change CS_REPORTING 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:662 (sofia/external/6578xxxx at 203.x.x.80) State REPORTING 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:79 sofia/external/6578xxxx at 203.x.x.80 Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:662 (sofia/external/6578xxxx at 203.x.x.80) State REPORTING going to sleep 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:387 (sofia/external/6578xxxx at 203.x.x.80) State Change CS_REPORTING -> CS_DESTROY 2012-02-09 17:09:22.268284 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/6578xxxx at 203.x.x.80 [BREAK] 2012-02-09 17:09:22.268284 [DEBUG] switch_core_session.c:1380 Session 23373 (sofia/external/6578xxxx at 203.x.x.80) Locked, Waiting on external entities 2012-02-09 17:09:22.268284 [DEBUG] switch_ivr_bridge.c:1405 sofia/internal/3175x102 at 203.x.x.81:5060 skip receive message [UNBRIDGE] (channel is hungup already) 2012-02-09 17:09:22.268284 [DEBUG] switch_core_session.c:2285 sofia/internal/3175x102 at 203.x.x.81:5060 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/3175x102 at 203.x.x.81:5060) State EXECUTE going to sleep 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/3175x102 at 203.x.x.81:5060) Running State Change CS_HANGUP 2012-02-09 17:09:22.268284 [DEBUG] switch_core_media_bug.c:576 Removing BUG from sofia/internal/3175x102 at 203.x.x.81:5060 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/3175x102 at 203.x.x.81:5060) State HANGUP 2012-02-09 17:09:22.268284 [DEBUG] mod_sofia.c:469 Channel sofia/internal/3175x102 at 203.x.x.81:5060 hanging up, cause: ORIGINATOR_CANCEL 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:47 sofia/internal/3175x102 at 203.x.x.81:5060 Standard HANGUP, cause: ORIGINATOR_CANCEL 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/3175x102 at 203.x.x.81:5060) State HANGUP going to sleep 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/3175x102 at 203.x.x.81:5060) State Change CS_HANGUP -> CS_REPORTING 2012-02-09 17:09:22.268284 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/3175x102 at 203.x.x.81:5060 [BREAK] 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/3175x102 at 203.x.x.81:5060) Running State Change CS_REPORTING 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/3175x102 at 203.x.x.81:5060) State REPORTING 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:79 sofia/internal/3175x102 at 203.x.x.81:5060 Standard REPORTING, cause: ORIGINATOR_CANCEL 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/3175x102 at 203.x.x.81:5060) State REPORTING going to sleep 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/3175x102 at 203.x.x.81:5060) State Change CS_REPORTING -> CS_DESTROY 2012-02-09 17:09:22.268284 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/3175x102 at 203.x.x.81:5060 [BREAK] 2012-02-09 17:09:22.268284 [DEBUG] switch_core_session.c:1380 Session 23372 (sofia/internal/3175x102 at 203.x.x.81:5060) Locked, Waiting on external entities 2012-02-09 17:09:22.268284 [NOTICE] switch_core_session.c:1398 Session 23372 (sofia/internal/3175x102 at 203.x.x.81:5060) Ended 2012-02-09 17:09:22.268284 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/3175x102 at 203.x.x.81:5060 [CS_DESTROY] 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/3175x102 at 203.x.x.81:5060) Callstate Change HANGUP -> DOWN 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/3175x102 at 203.x.x.81:5060) Running State Change CS_DESTROY 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/3175x102 at 203.x.x.81:5060) State DESTROY 2012-02-09 17:09:22.268284 [DEBUG] mod_sofia.c:374 sofia/internal/3175x102 at 203.x.x.81:5060 SOFIA DESTROY 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:86 sofia/internal/3175x102 at 203.x.x.81:5060 Standard DESTROY 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/3175x102 at 203.x.x.81:5060) State DESTROY going to sleep 2012-02-09 17:09:22.268284 [NOTICE] switch_core_session.c:1398 Session 23373 (sofia/external/6578xxxx at 203.x.x.80) Ended 2012-02-09 17:09:22.268284 [NOTICE] switch_core_session.c:1400 Close Channel sofia/external/6578xxxx at 203.x.x.80 [CS_DESTROY] 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:491 (sofia/external/6578xxxx at 203.x.x.80) Callstate Change HANGUP -> DOWN 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:494 (sofia/external/6578xxxx at 203.x.x.80) Running State Change CS_DESTROY 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:504 (sofia/external/6578xxxx at 203.x.x.80) State DESTROY 2012-02-09 17:09:22.268284 [DEBUG] mod_sofia.c:374 sofia/external/6578xxxx at 203.x.x.80 SOFIA DESTROY 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:86 sofia/external/6578xxxx at 203.x.x.80 Standard DESTROY 2012-02-09 17:09:22.268284 [DEBUG] switch_core_state_machine.c:504 (sofia/external/6578xxxx at 203.x.x.80) State DESTROY going to sleep Thank you. Armand On 2012-02-09 4:51 PM, Avi Marcus wrote: > If you're seeing unexpected dialplan stuff, then can you show us the > entire dialplan parsing log, from start to finish? I'm guessing you > have an extra bridge getting pulled in there... > -Avi > > > On Thu, Feb 9, 2012 at 10:31 AM, Man > wrote: > > No failover bridge is configured and I cannot find any similar > command in the dialplan directory. > > The log is a bit long here: > > -log- > > Thank you. > > Armand > > On 2012-02-09 3:59 PM, Avi Marcus wrote: >> Can we see the call on /log 7? You don't have a failover bridge >> destination? >> -Avi >> >> On Thu, Feb 9, 2012 at 6:03 AM, Man > > wrote: >> >> User registered to the system and try to make call to PSTN >> network but system show "No Route, Aborting" at first and >> connect the call right after it. The destination number ring >> wrong caller ID displayed. 40% calls happened with this issue. >> >> Any possible reason? >> >> FreeSWITCH Version 1.0.head (git-7f5b8fb 2012-02-02 20-41-15 >> -0600) >> >> Checked the CDR both normal and fault call are no different. >> >> My dialplan is simply bridge the call to the voice gateway: >> >> >> > data="sofia/external/$1 at 203.194.x.x" >> /> >> >> >> >> Thanks a lot. >> >> Armand >> >> _________________________________________________________________________ >> >> -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/399d240f/attachment-0001.html From auslands-kv at gmx.de Thu Feb 9 15:08:08 2012 From: auslands-kv at gmx.de (Michael) Date: Thu, 09 Feb 2012 13:08:08 +0100 Subject: [Freeswitch-users] Howto setup BLF event lists? Message-ID: Hi I am pretty new to all this, so bear with me. I have Grandstream phones running with my freeswitch setup. I would like to use the BLF which works via event lists. It seems that freeswitch supports this. However, I do not find any documentation how to setup such an event list. The only thing I know is where I need to enter the URL (whatever it may look like...) into the Grandstream. Can anybody hint me to some documentation? Thanks a lot Michael From msc at freeswitch.org Thu Feb 9 21:52:32 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Feb 2012 10:52:32 -0800 Subject: [Freeswitch-users] [newbie] questions In-Reply-To: <4F33D39D.5070807@privatedemail.net> References: <4F330A31.4060804@privatedemail.net> <4F33D39D.5070807@privatedemail.net> Message-ID: If you're willing to get your hands dirty then that's great. I recommend the following resources: * Our "bridge" book - it covers a lot of the basics of FreeSWITCH. There are links to Amazon and Packt Publishing from our main wiki page * The forthcoming FreeSWITCH Cookbook. You can pre-order now. I'll throw some links up on the main page and wiki page, but if you go to PacktPub.com or Amazon.com just do a book search for FreeSWITCH and you'll find it. There's only two titles. :P * You're already on the mailing list, so karma++ for you * The IRC channel: #freeswitch on irc.freenode.net. This is, hands down, the best place to talk in real time about FreeSWITCH stuff * The wiki - it's got TONS of information. Yes, I'll admit you may have to dig and you may find confusing or contradictory information but we do our best to keep it updated. Just let us know if you find that something is wrong/confusing/missing on the wiki * The Wednesday conference calls - We do a weekly conference call for members of the community. You can interact with power users and occasionally even the developers themselves. We usually have a "feature presentation" after which we let folks do an open Q&A. As for the choice between Asterisk and FreeSWITCH: we respect your decision, and I promise we won't cry if you choose Asterisk or PIAF or whatever. However, many members of the FS community are "Asterisk refugees" who've been burned by mysterious deadlocks, segfaults, scaling issues, etc. If you ask us about our Asterisk experiences be prepared for horror stories. :D That being said, lots of people use Asterisk with no apparent issues. It's up to you. No matter which way you go, be prepared for aggravating problems and pulling your hair out. VoIP and telephony are always frustrating for the new user. Most of our community members remember their newbie pain and are quite willing to offer their knowledge and experience to help you keep at least some of your hair. ;) Let us know what you decide! -MC (IRC: mercutioviz) On Thu, Feb 9, 2012 at 6:09 AM, Josh wrote: > > > Welcome to FreeSWITCH! > Pleasure. > > > But calls will be sent to FreeSWITCH by some device, correct? If it's > > good old-fashioned SIP then FreeSWITCH will handle it just fine. > Yes, there are all softphones (PC machines with old-fashioned headset & > mic and smartphones running a sip client). > > > > > Could this be done relatively easily in FreeSWITCH? > > > > "Relatively?" Of course! It's relatively easy for someone with some > > experience. I highly recommend that you ask consulting at freeswitch.org > > for professional assistance if you > > are not comfortable doing this all by yourself. > I'd rather do it by myself. For now though, I'd like to see whether what > I want to achieve is possible in FreeSWITCH. I haven't yet made a > decision what to use though - Asterisk or FreeSWITCH, it will all depend > on whether I could set it up properly and "relatively" easy. Having said > all that, as I developer with quite a bit of experience behind me, I am > not afraid to delve in and get my hands dirty, if needed. I just want to > make sure that what I want in terms of set up and functionality is > possible. > > > Yes, FreeSWITCH can bind to multiple interfaces. In FreeSWITCH lingo > > that would mean that you set up a separate SIP profile for each > > interface. (In fact, you can have more than one SIP profile on a given > > interface since the profile is a unique combo of IP addr and port > number.) > I presume different profiles can "talk" to each other, right? In other > words calls/media can be routed/transferred from one interface to > another (eth1<->tun0 for example)? > > > "Some assembly required." :D > > FreeSWITCH can do some stuff for you, but you definitely need to make > > sure that your NAT is not behaving badly, like having a SIP ALG. > This is what I am trying to figure out - do I rely entirely on > FreeSWITCH (if not, what is expected of me to set up so that FreeSWITCH > can do its job?), or do I have to do it all by myself with the kernel > module helpers (sip, h323 etc) and ip/iptables? > > > I'll have to defer to Ken Rice on this one. I know he's working on > > RPMs for FreeSWITCH but I think it's all RedHat right now. > RedHat is good, all I need is a decent .spec file - I'll do the rest > myself, no problem. > > One thing I seem to have forgotten from my newbie list of questions - I > take it FreeSWITCH can do call-recording (in both directions) right? If > so, how is this stored/implemented (I hope both ends are stored as a > single sound file)? I would like all calls to be recorded as a matter of > policy, so I do need this implemented if I am going to use FreeSWITCH. > > Many thanks again. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/ad10c936/attachment.html From msc at freeswitch.org Thu Feb 9 21:55:21 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Feb 2012 10:55:21 -0800 Subject: [Freeswitch-users] New to FS In-Reply-To: References: Message-ID: Chris, Welcome to the FS community. Glad to have you. It might be easier for you to hop on the IRC channel and talk to some of the guys there. Go to #freeswitch on irc.freenode.net. We usually have a few hundred people in there and one of them is bound to be able to discuss your questions. -MC (IRC: mercutioviz) On Wed, Feb 8, 2012 at 11:01 PM, chris wrote: > Hello, > > I am new to FS but I'm impressed already with my first experiences with > it. I was just wondering about somethins that I've always wanted on > asterisk that was possible but always been a pain. Is it possible to do > source based routing and route certain SIP extensions/devices to certain > outbound trunks? If so how can I do it easily? > > Thanks for any advice / info, > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/491681ba/attachment.html From msc at freeswitch.org Thu Feb 9 21:56:42 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Feb 2012 10:56:42 -0800 Subject: [Freeswitch-users] Hangup Cause in LUA Hangup Hook In-Reply-To: References: Message-ID: Sorry for the delayed response. I'm 99% sure that you want this: http://wiki.freeswitch.org/wiki/Lua#Special_Case:_env_object -MC On Wed, Feb 8, 2012 at 4:08 AM, Gerald Weber wrote: > Hello,**** > > ** ** > > is there a way to get the HangupCause in a HangupHook in LUA ?**** > > following commands return nil:**** > > s:getVariable("hangup_cause")**** > > s:getVariable("hangup_cause_q850")**** > > ** ** > > s:hangupCause() returns ?NONE?**** > > ** ** > > I know I can set api_hangup_hook and use the env object to get the channel > vars, but i think that?s a litte overkill to use 2 scripts for 1 Call.**** > > ** ** > > Thx**** > > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/13b3dc53/attachment-0001.html From msc at freeswitch.org Thu Feb 9 22:02:02 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Feb 2012 11:02:02 -0800 Subject: [Freeswitch-users] I am not able to reg with 5060 via softphone In-Reply-To: References: Message-ID: You need to find out why you can't reg on port 5060. I see that you have an external IP address listed there. Is that the only interface on your server? One trick you can do is to swap the internal.xml and external.xml sip profiles. Basically just rename the files and restart freeswitch and see if you can register on 5060. If not you'll need to supply console logs w/ siptraces and put them on pastebin.freeswitch.org so the gang here can take a look. -MC On Wed, Feb 8, 2012 at 4:50 AM, virendra bhati wrote: > > Hi list, > > I am new for FreeSwitch. I was working with asterisk since 4 years. Now i > want to learn FreeSwitch. I have installed at my server. But when I try to > reg with server .I am not able to register with it with port 5060. but if I > will 5080 then reg with server but can't call to local extension like 1001 > - 1019. > > Please guide me what is the issue ? > > *fs_CLI log:-* > > EXECUTE sofia/external/1006 at 78.129.163.44 bridge(user/1005 at 78.129.163.44) > 2012-02-08 12:59:37.493777 [DEBUG] switch_ivr_originate.c:1885 variable > string 0 = [presence_id=1005 at 78.129.163.44] > 2012-02-08 12:59:37.493777 [ERR] switch_ivr_originate.c:2430 Cannot create > outgoing channel of type [error] cause: [USER_NOT_REGISTERED] > 2012-02-08 12:59:37.493777 [DEBUG] switch_ivr_originate.c:3228 Originate > Resulted in Error Cause: 606 [USER_NOT_REGISTERED] > 2012-02-08 12:59:37.493777 [ERR] switch_ivr_originate.c:2430 Cannot create > outgoing channel of type [user] cause: [USER_NOT_REGISTERED] > 2012-02-08 12:59:37.493777 [DEBUG] switch_ivr_originate.c:3228 Originate > Resulted in Error Cause: 606 [USER_NOT_REGISTERED] > 2012-02-08 12:59:37.493777 [INFO] mod_dptools.c:2355 Originate Failed. > Cause: USER_NOT_REGISTERED > EXECUTE sofia/external/1006 at 78.129.163.44 answer() > > -- > > Thanks and regards > > Virendra Bhati > +91-8885268942 > Software Engineer > E-mail-: virbhati at gmail.com > Skype id:- virbhati2 > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/ade65e8b/attachment.html From msc at freeswitch.org Thu Feb 9 22:03:18 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Feb 2012 11:03:18 -0800 Subject: [Freeswitch-users] Can someone help me with a complicated configuration? In-Reply-To: References: Message-ID: What kind of devices will connect to the public IP addresses? How will *they* handle a colo change that results in a new IP address? -MC On Wed, Feb 8, 2012 at 9:28 AM, Josue Diaz Cruz wrote: > ** > I have a server with 4 ETH. Each one have a different IP > In 2 cases i fixed the ip address, In the other 2 cases i have 2 cable > connections in bridge mode so i receive the public ip in dhcp mode. > > eth0 local address 192.168.1.100 > eth1 local address 192.168.2.100 > eth2 public address dhcp (cable) > eth3 public address dhcp (cable) > > I was thinking to build 4 different profiles for each eth port, but i came > to know that if we change the place of colocation it will not work properly > because the dhcp ip. It is any way to attach a profile to the eth port, not > the ip? > > Or if you know a better way to manage this case. I would like to receive > traffic from the 2 public ip's and route it both like if i have just one > profile. > > * > > Josue Diaz Cruz > > Departamento Tecnico y Soporte > > jdiaz at coinfru.com > > > > C/ Balsicas 3 > > Alquerias | 30580 | Murcia > > www.coinfru.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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/9caecd92/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4705 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/9caecd92/attachment.jpe From msc at freeswitch.org Thu Feb 9 22:09:20 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Feb 2012 11:09:20 -0800 Subject: [Freeswitch-users] skipping vm and plays enter the pwd In-Reply-To: <1328730591696-7267135.post@n2.nabble.com> References: <1328730591696-7267135.post@n2.nabble.com> Message-ID: This isn't enough information to diagnose. Please supply a log of the call from start to finish. Also, you probably should include the dialplan extension that is handling this process. -MC On Wed, Feb 8, 2012 at 11:49 AM, Aakash wrote: > Hi All, > > I have configured has any inbound calls enter to my inbound number play an > ivr and transfer to user ext (eg 1080)pressed by customer,if the user dint > pick the call enter to their voicemail automatically.But sometimes i am > getting please enter password followed by # by skipping voicemail message.I > have attached my logs below > > http://pastebin.freeswitch.org/18321 > > > Regards, > Aakash.V > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/skipping-vm-and-plays-enter-the-pwd-tp7267135p7267135.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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/64eb351c/attachment-0001.html From msc at freeswitch.org Thu Feb 9 22:11:00 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Feb 2012 11:11:00 -0800 Subject: [Freeswitch-users] Blind Transfer In-Reply-To: <20120209100505.620465yayx3v3g4c@mail.horsley.id.au> References: <20120209100505.620465yayx3v3g4c@mail.horsley.id.au> Message-ID: Definitely get a loglevel 7 on this and put it into pastebin.freeswitch.org. Also, set the syntax highlighting to "FreeSWITCH Log" so that we can see the pretty colors. -MC On Wed, Feb 8, 2012 at 3:05 PM, Scott wrote: > Hi list, > > Trying to resolve an issue with blind transfers. > > When transferring a call, the A leg is terminated if a blind transfer > is started, an attended transfer doesn't create this scenario. > > Here are the transcript logs from FS relating to the event. I have log > level set to 6 but can increase this if I need more information. > > Call flow is as follows.. > > ext18: Calls ext51 > ext18: transfers call to ext15 > ext15: rings for a split second > ext51: lands in voicemail of 15 > ext18: clears as normal > > OS: CentOS release 5.7 (Final) > > freeswitch at internal> version > FreeSWITCH Version 1.0.head (git-9de1e1a 2012-02-07 14-36-22 -0500) > > +OK log level [7] > freeswitch at internal> > > > 2012-02-08 16:08:10.291633 [NOTICE] switch_channel.c:920 New Channel > sofia/internal/18 at 10.100.0.9 [6fd7a0b0-c518-42ff-bb24-e2acf8121b9e] > 2012-02-08 16:08:10.291633 [INFO] mod_dialplan_xml.c:481 Processing > Scott <18>->51 in context default > 2012-02-08 16:08:10.311819 [NOTICE] switch_channel.c:920 New Channel > sofia/internal/sip:51 at 10.100.0.16:1024 > [f8f27c7d-9198-4225-a94d-2eadb9bb58a8] > 2012-02-08 16:08:10.391998 [NOTICE] sofia.c:5458 Ring-Ready > sofia/internal/sip:51 at 10.100.0.16:1024! > 2012-02-08 16:08:10.391998 [INFO] switch_ivr_originate.c:1115 Sending > early media > 2012-02-08 16:08:10.411314 [NOTICE] sofia_glue.c:3899 Pre-Answer > sofia/internal/18 at 10.100.0.9! > 2012-02-08 16:08:10.431704 [INFO] mod_com_g729.c:119 ENCODER CREATE - > 0xb6ed5160 0x8b23a08 > 2012-02-08 16:08:10.731162 [INFO] mod_com_g729.c:148 DECODER CREATE - > 0xb6ed5108 0x8c094c0 > 2012-02-08 16:08:11.977555 [NOTICE] sofia.c:6070 Channel > [sofia/internal/sip:51 at 10.100.0.16:1024] has been answered > 2012-02-08 16:08:11.996869 [NOTICE] switch_ivr_originate.c:3209 > Channel [sofia/internal/18 at 10.100.0.9] has been answered > 2012-02-08 16:08:12.098595 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - > 0xb4d025b0 (nil) > 2012-02-08 16:08:12.098595 [INFO] mod_com_g729.c:79 DECODER DESTROYX - > 0xb4d025b0 (nil) > 2012-02-08 16:08:12.177926 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - > 0xb4d02558 (nil) > 2012-02-08 16:08:12.177926 [INFO] mod_com_g729.c:79 DECODER DESTROYX - > 0xb4d02558 (nil) > 2012-02-08 16:08:14.183102 [INFO] mod_com_g729.c:119 ENCODER CREATE - > 0xb4d029e0 0xb6e6ac00 > 2012-02-08 16:08:15.750004 [NOTICE] switch_ivr.c:1711 Transfer > sofia/internal/sip:51 at 10.100.0.16:1024 to XML[15 at default] > 2012-02-08 16:08:15.769362 [NOTICE] switch_ivr_bridge.c:1372 Hangup > sofia/internal/18 at 10.100.0.9 [CS_EXECUTE] [NORMAL_CLEARING] > 2012-02-08 16:08:15.789438 [INFO] mod_dialplan_xml.c:481 Processing > Scott <18>->15 in context default > 2012-02-08 16:08:15.789438 [INFO] switch_channel.c:2695 > sofia/internal/sip:51 at 10.100.0.16:1024 Flipping CID from "Scott" <18> > to "Outbound Call" <51> > 2012-02-08 16:08:15.789438 [NOTICE] switch_channel.c:920 New Channel > sofia/internal/sip:15 at 10.100.0.15:5064 > [acf45601-79ad-4518-907f-7a58155748ab] > 2012-02-08 16:08:16.231456 [NOTICE] sofia.c:5458 Ring-Ready > sofia/internal/sip:15 at 10.100.0.15:5064! > 2012-02-08 16:08:16.291519 [INFO] mod_com_g729.c:148 DECODER CREATE - > 0xb4d029f8 0xb6c41798 > 2012-02-08 16:08:16.291519 [NOTICE] switch_ivr_originate.c:3182 Hangup > sofia/internal/sip:15 at 10.100.0.15:5064 [CS_CONSUME_MEDIA] [NO_ANSWER] > 2012-02-08 16:08:16.291519 [NOTICE] switch_ivr_originate.c:2459 Cannot > create outgoing channel of type [user] cause: [NO_ANSWER] > 2012-02-08 16:08:16.291519 [INFO] mod_dptools.c:2884 Originate Failed. > Cause: NO_ANSWER > 2012-02-08 16:08:16.291519 [NOTICE] switch_core_session.c:1395 Session > 991 (sofia/internal/sip:15 at 10.100.0.15:5064) Ended > 2012-02-08 16:08:16.291519 [NOTICE] switch_core_session.c:1397 Close > Channel sofia/internal/sip:15 at 10.100.0.15:5064 [CS_DESTROY] > 2012-02-08 16:08:16.371076 [NOTICE] switch_core_session.c:1395 Session > 989 (sofia/internal/18 at 10.100.0.9) Ended > 2012-02-08 16:08:16.371076 [NOTICE] switch_core_session.c:1397 Close > Channel sofia/internal/18 at 10.100.0.9 [CS_DESTROY] > 2012-02-08 16:08:16.371076 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - > 0xb6ed5108 (nil) > 2012-02-08 16:08:16.371076 [INFO] mod_com_g729.c:79 DECODER DESTROYX - > 0xb6ed5108 0x8c094c0 > 2012-02-08 16:08:16.371076 [INFO] mod_com_g729.c:89 DECODER DESTROY - > 0xb6ed5108 0x8c094c0 > 2012-02-08 16:08:16.371076 [INFO] mod_com_g729.c:78 ENCODER DESTROYX - > 0xb6ed5160 0x8b23a08 > 2012-02-08 16:08:16.371076 [INFO] mod_com_g729.c:79 DECODER DESTROYX - > 0xb6ed5160 (nil) > 2012-02-08 16:08:16.371076 [INFO] mod_com_g729.c:83 ENCODER DESTROY - > 0xb6ed5160 0x8b23a08 > 2012-02-08 16:08:17.292845 [NOTICE] switch_channel.c:920 New Channel > loopback/app=voicemail:default 10.100.0.9 15-a > [7b76ae61-3465-40bb-9d65-41f6cba2c3d2] > 2012-02-08 16:08:17.292845 [NOTICE] switch_channel.c:918 Rename > Channel loopback/app=voicemail:default 10.100.0.9 > 15-a->loopback/voicemail-a [7b76ae61-3465-40bb-9d65-41f6cba2c3d2] > 2012-02-08 16:08:17.292845 [NOTICE] switch_channel.c:920 New Channel > loopback/voicemail-b [e02cc153-8651-422c-bb98-f6f7a3244418] > 2012-02-08 16:08:17.292845 [NOTICE] mod_loopback.c:760 Pre-Answer > loopback/voicemail-a! > 2012-02-08 16:08:17.292845 [NOTICE] mod_dptools.c:1143 Pre-Answer > loopback/voicemail-b! > 2012-02-08 16:08:17.414279 [INFO] mod_com_g729.c:119 ENCODER CREATE - > 0x8b6e120 0xb6ec4118 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/20ef76cb/attachment.html From msc at freeswitch.org Thu Feb 9 22:14:14 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Feb 2012 11:14:14 -0800 Subject: [Freeswitch-users] Freeswitch opened multiple outbound channels for one call with different uuid In-Reply-To: <4F33266D.2050300@telefaks.biz> References: <4F323042.8080603@telefaks.biz> <4F326E84.6030302@telefaks.biz> <4F33266D.2050300@telefaks.biz> Message-ID: I would clear out your registrations for this user and start over. Watch to see what happens when the phone registers w/ FreeSWITCH. In fact, you may want to update to latest git, then zap all the *.db files except for the voicemail.db one. Is this a production system? If so then tread carefully or do some testing on your sandbox/test system before implementing changes on the production system. -MC On Wed, Feb 8, 2012 at 5:50 PM, David wrote: > ** > I cannot find any network problems. > > interface: any > filter: (ip or ip6) and ( net 192.168.1.12 or net 192.168.1.11 ) > > # > U 192.168.1.12:2048 -> 192.168.1.4:5060 > INVITE sip:240 at entwick1;user=phone SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-v4ybiqjk8t4e;rport. > From: "User 220" ;tag=vx1czcrfys. > To: . > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 1 INVITE. > Max-Forwards: 70. > Contact: ;reg-id=1. > P-Key-Flags: keys="3". > User-Agent: snom300/7.3.30. > Accept: application/sdp. > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, > MESSAGE, INFO. > Allow-Events: talk, hold, refer, call-info. > Supported: timer, 100rel, replaces, from-change. > Session-Expires: 3600;refresher=uas. > Min-SE: 90. > Content-Type: application/sdp. > Content-Length: 218. > . > v=0. > o=root 930692018 930692018 IN IP4 192.168.1.12. > s=call. > c=IN IP4 192.168.1.12. > t=0 0. > m=audio 49908 RTP/AVP 8 101. > a=rtpmap:8 pcma/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > a=sendrecv. > > # > U 192.168.1.4:5060 -> 192.168.1.12:2048 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-v4ybiqjk8t4e;rport=2048. > From: "User 220" ;tag=vx1czcrfys. > To: . > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 1 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 > -0600. > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.12:2048 > SIP/2.0 407 Proxy Authentication Required. > Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-v4ybiqjk8t4e;rport=2048. > From: "User 220" ;tag=vx1czcrfys. > To: ;tag=rFZa6489Z27rK. > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 1 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 > -0600. > Accept: application/sdp. > 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. > Proxy-Authenticate: Digest realm="entwick1", > nonce="b755765a-a341-4070-a981-dff43e8f7d08", algorithm=MD5, qop="auth". > Content-Length: 0. > . > > # > U 192.168.1.12:2048 -> 192.168.1.4:5060 > ACK sip:240 at entwick1;user=phone SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-v4ybiqjk8t4e;rport. > From: "User 220" ;tag=vx1czcrfys. > To: ;tag=rFZa6489Z27rK. > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 1 ACK. > Max-Forwards: 70. > Contact: ;reg-id=1. > Content-Length: 0. > . > > # > U 192.168.1.12:2048 -> 192.168.1.4:5060 > INVITE sip:240 at entwick1;user=phone SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-eyrl6ghugshz;rport. > From: "User 220" ;tag=vx1czcrfys. > To: . > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 2 INVITE. > Max-Forwards: 70. > Contact: ;reg-id=1. > P-Key-Flags: keys="3". > User-Agent: snom300/7.3.30. > Accept: application/sdp. > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, > MESSAGE, INFO. > Allow-Events: talk, hold, refer, call-info. > Supported: timer, 100rel, replaces, from-change. > Session-Expires: 3600;refresher=uas. > Min-SE: 90. > Proxy-Authorization: Digest > username="220",realm="entwick1",nonce="b755765a-a341-4070-a981-dff43e8f7d08",uri= > "sip:240 at entwick1;user=phone" > ,qop=auth,nc=00000001,cnonce="408b0a1e",response="3261384c8f62187a6486c1fee434d06a",algorithm=MD5. > Content-Type: application/sdp. > Content-Length: 218. > . > v=0. > o=root 930692018 930692018 IN IP4 192.168.1.12. > s=call. > c=IN IP4 192.168.1.12. > t=0 0. > m=audio 49908 RTP/AVP 8 101. > a=rtpmap:8 pcma/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > a=sendrecv. > > # > U 192.168.1.4:5060 -> 192.168.1.12:2048 > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-eyrl6ghugshz;rport=2048. > From: "User 220" ;tag=vx1czcrfys. > To: . > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 2 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 > -0600. > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.12:2048 > SIP/2.0 180 Ringing. > Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-eyrl6ghugshz;rport=2048. > From: "User 220" ;tag=vx1czcrfys. > To: ;tag=Srr37ZSDXByBF. > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 2 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 > -0600. > Accept: application/sdp. > 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-Length: 0. > Remote-Party-ID: "240" > ;party=calling;privacy=off;screen=no. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > INVITE sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK4DN4eH402KHmF. > Max-Forwards: 69. > From: "User 220" ;tag=t1Hv9taHtmmya. > To: . > Call-ID: ad561287-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 > -0600. > 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. > Session-Expires: 120;refresher=uac. > Min-SE: 120. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 233. > P-Key-Flags: keys="3". > X-FS-Support: update_display,send_info. > Remote-Party-ID: "User 220" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1328729538 1328729539 IN IP4 192.168.1.4. > s=FreeSWITCH. > c=IN IP4 192.168.1.4. > t=0 0. > m=audio 22000 RTP/AVP 8 0 98 3 18 101 13. > a=rtpmap:98 SPEEX/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > INVITE sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. > Max-Forwards: 69. > From: "User 220" ;tag=UaBNBpUmQXaHp. > To: . > Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 > -0600. > 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. > Session-Expires: 120;refresher=uac. > Min-SE: 120. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 233. > P-Key-Flags: keys="3". > X-FS-Support: update_display,send_info. > Remote-Party-ID: "User 220" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1328731870 1328731871 IN IP4 192.168.1.4. > s=FreeSWITCH. > c=IN IP4 192.168.1.4. > t=0 0. > m=audio 19668 RTP/AVP 8 0 98 3 18 101 13. > a=rtpmap:98 SPEEX/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > INVITE sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK6Z7Nj757v5XSp. > Max-Forwards: 69. > From: "User 220" ;tag=vK4DDHcrm603H. > To: . > Call-ID: ad5660f4-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 > -0600. > 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. > Session-Expires: 120;refresher=uac. > Min-SE: 120. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 233. > P-Key-Flags: keys="3". > X-FS-Support: update_display,send_info. > Remote-Party-ID: "User 220" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1328721426 1328721427 IN IP4 192.168.1.4. > s=FreeSWITCH. > c=IN IP4 192.168.1.4. > t=0 0. > m=audio 30112 RTP/AVP 8 0 98 3 18 101 13. > a=rtpmap:98 SPEEX/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > INVITE sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK780em2pBtemcj. > Max-Forwards: 69. > From: "User 220" ;tag=XvX6ecXUHFQpD. > To: . > Call-ID: ad566b4a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 > -0600. > 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. > Session-Expires: 120;refresher=uac. > Min-SE: 120. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 233. > P-Key-Flags: keys="3". > X-FS-Support: update_display,send_info. > Remote-Party-ID: "User 220" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1328730428 1328730429 IN IP4 192.168.1.4. > s=FreeSWITCH. > c=IN IP4 192.168.1.4. > t=0 0. > m=audio 21110 RTP/AVP 8 0 98 3 18 101 13. > a=rtpmap:98 SPEEX/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > > # > U 192.168.1.11:5060 -> 192.168.1.4:5060 > SIP/2.0 180 Ringing. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK4DN4eH402KHmF. > Call-ID: ad561287-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > From: "User 220" ;tag=t1Hv9taHtmmya. > To: ;tag=iN29e6PanzlAC2JA. > Contact: . > Content-Length: 0. > . > > # > U 192.168.1.11:5060 -> 192.168.1.4:5060 > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. > Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > From: "User 220" ;tag=UaBNBpUmQXaHp. > To: ;tag=vrAASGKzlwRxb5pm. > Contact: . > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > ACK sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. > Max-Forwards: 69. > From: "User 220" ;tag=UaBNBpUmQXaHp. > To: ;tag=vrAASGKzlwRxb5pm. > Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 ACK. > Content-Length: 0. > . > > # > U 192.168.1.11:5060 -> 192.168.1.4:5060 > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK6Z7Nj757v5XSp. > Call-ID: ad5660f4-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > From: "User 220" ;tag=vK4DDHcrm603H. > To: ;tag=2Urps2fOoZQSDiid. > Contact: . > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > ACK sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK6Z7Nj757v5XSp. > Max-Forwards: 69. > From: "User 220" ;tag=vK4DDHcrm603H. > To: ;tag=2Urps2fOoZQSDiid. > Call-ID: ad5660f4-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 ACK. > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > INVITE sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK780em2pBtemcj. > Max-Forwards: 69. > From: "User 220" ;tag=XvX6ecXUHFQpD. > To: . > Call-ID: ad566b4a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 > -0600. > 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. > Session-Expires: 120;refresher=uac. > Min-SE: 120. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 233. > P-Key-Flags: keys="3". > X-FS-Support: update_display,send_info. > Remote-Party-ID: "User 220" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1328730428 1328730429 IN IP4 192.168.1.4. > s=FreeSWITCH. > c=IN IP4 192.168.1.4. > t=0 0. > m=audio 21110 RTP/AVP 8 0 98 3 18 101 13. > a=rtpmap:98 SPEEX/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > > # > U 192.168.1.11:5060 -> 192.168.1.4:5060 > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK780em2pBtemcj. > Call-ID: ad566b4a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > From: "User 220" ;tag=XvX6ecXUHFQpD. > To: ;tag=qh5PsDcfdulD0LlQ. > Contact: . > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > ACK sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK780em2pBtemcj. > Max-Forwards: 69. > From: "User 220" ;tag=XvX6ecXUHFQpD. > To: ;tag=qh5PsDcfdulD0LlQ. > Call-ID: ad566b4a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 ACK. > Content-Length: 0. > . > > # > U 192.168.1.11:5060 -> 192.168.1.4:5060 > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. > Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > From: "User 220" ;tag=UaBNBpUmQXaHp. > To: ;tag=vrAASGKzlwRxb5pm. > Contact: . > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > ACK sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. > Max-Forwards: 69. > From: "User 220" ;tag=UaBNBpUmQXaHp. > To: ;tag=vrAASGKzlwRxb5pm. > Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 ACK. > Content-Length: 0. > . > > # > U 192.168.1.11:5060 -> 192.168.1.4:5060 > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. > Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > From: "User 220" ;tag=UaBNBpUmQXaHp. > To: ;tag=vrAASGKzlwRxb5pm. > Contact: . > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > ACK sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. > Max-Forwards: 69. > From: "User 220" ;tag=UaBNBpUmQXaHp. > To: ;tag=vrAASGKzlwRxb5pm. > Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 ACK. > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > OPTIONS sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK8Ht7NX7eQQaZD. > Max-Forwards: 70. > From: ;tag=y5pZg7DZerD9r. > To: . > Call-ID: b0fa8901-cd61-122f-8d91-001966eeb846. > CSeq: 24029141 OPTIONS. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 > -0600. > 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-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.12:2048 > OPTIONS sip:220 at 192.168.1.12:2048;line=9yhon28z SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK9tK0Qrrjm00HS. > Max-Forwards: 70. > From: ;tag=Zegrj2y2B13Um. > To: . > Call-ID: b0fa9032-cd61-122f-8d91-001966eeb846. > CSeq: 24029142 OPTIONS. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 > -0600. > 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-Length: 0. > . > > # > U 192.168.1.12:2048 -> 192.168.1.4:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 192.168.1.4;rport=5060;branch=z9hG4bK9tK0Qrrjm00HS. > From: ;tag=Zegrj2y2B13Um. > To: . > Call-ID: b0fa9032-cd61-122f-8d91-001966eeb846. > CSeq: 24029142 OPTIONS. > Contact: ;reg-id=1. > User-Agent: snom300/7.3.30. > Accept-Language: en. > Accept: application/sdp. > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, > MESSAGE, INFO. > Allow-Events: talk, hold, refer, call-info. > Supported: timer, 100rel, replaces, from-change. > Content-Length: 0. > . > > # > U 192.168.1.11:5060 -> 192.168.1.4:5060 > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK8Ht7NX7eQQaZD. > Call-ID: b0fa8901-cd61-122f-8d91-001966eeb846. > CSeq: 24029141 OPTIONS. > From: ;tag=y5pZg7DZerD9r. > To: ;tag=4rjgSs2dsNvcVyjW. > Contact: . > Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, NOTIFY, INFO, > PRACK, UPDATE. > Accept: application/sdp. > Accept-Encoding: identity. > Accept-Language: en. > Supported: 100rel, replaces. > Content-Type: application/sdp. > Content-Length: 267. > . > v=0. > o=240 32101693 45791413 IN IP4 192.168.1.11. > s=SIP CALL. > c=IN IP4 192.168.1.11. > t=0 0. > m=audio 6000 RTP/AVP 8 0 3 18 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:18 G729/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-15. > > # > U 192.168.1.12:2048 -> 192.168.1.4:5060 > CANCEL sip:240 at entwick1;user=phone SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-eyrl6ghugshz;rport. > From: "User 220" ;tag=vx1czcrfys. > To: . > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 2 CANCEL. > Max-Forwards: 70. > Reason: SIP;cause=487;text="Request terminated by user". > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.12:2048 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-eyrl6ghugshz;rport=2048. > From: "User 220" ;tag=vx1czcrfys. > To: ;tag=Srr37ZSDXByBF. > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 2 CANCEL. > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.12:2048 > SIP/2.0 487 Request Terminated. > Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-eyrl6ghugshz;rport=2048. > From: "User 220" ;tag=vx1czcrfys. > To: ;tag=Srr37ZSDXByBF. > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 2 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 14-12-22 > -0600. > 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-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > CANCEL sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK4DN4eH402KHmF. > Max-Forwards: 69. > From: "User 220" ;tag=t1Hv9taHtmmya. > To: . > Call-ID: ad561287-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 CANCEL. > Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL". > Content-Length: 0. > . > > # > U 192.168.1.11:5060 -> 192.168.1.4:5060 > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK4DN4eH402KHmF. > Call-ID: ad561287-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 CANCEL. > From: "User 220" ;tag=t1Hv9taHtmmya. > To: ;tag=iN29e6PanzlAC2JA. > Contact: . > Content-Length: 0. > . > > # > U 192.168.1.11:5060 -> 192.168.1.4:5060 > SIP/2.0 487 Request Terminated. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK4DN4eH402KHmF. > Call-ID: ad561287-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > From: "User 220" ;tag=t1Hv9taHtmmya. > To: ;tag=iN29e6PanzlAC2JA. > Contact: . > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > ACK sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK4DN4eH402KHmF. > Max-Forwards: 69. > From: "User 220" ;tag=t1Hv9taHtmmya. > To: ;tag=iN29e6PanzlAC2JA. > Call-ID: ad561287-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 ACK. > Content-Length: 0. > . > > # > > Am 08.02.2012 18:36, schrieb Michael Collins: > > > It was found that freeswitch performs multiple INVITES. But why? >> >> > One reason why multiple invites are sent out is if FreeSWITCH doesn't > receive anything back. This could mean the far end is down for some reason > or that there is a network issue between FS and the other endpoint. NAT is > a common issue in this sort of configuration. I recommend that you look at > SIP traces on each end of the SIP dialog. > > -MC > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.orghttp://www.freeswitchsolutions.com > > > > Official FreeSWITCH Siteshttp://www.freeswitch.orghttp://wiki.freeswitch.orghttp://www.cluecon.com > > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/78d49874/attachment-0001.html From anthony.minessale at gmail.com Thu Feb 9 22:15:01 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Feb 2012 13:15:01 -0600 Subject: [Freeswitch-users] cluecon 2012 date confirm In-Reply-To: References: Message-ID: Aug 7-9th but you can come on the 6th too and enjoy the ramp up day. On Thu, Feb 9, 2012 at 12:00 AM, Seven Du wrote: > Hi, > > Had a wonderful time in Cluecon 2011 and would like to confirm the date as > it looks like no schedule for 2012 on cluecon.com and I only found a count > down on freeswitch.org. > > So would like to confirm will it be Aug. 6th this year? > > > 179 > Days10 > Hours2 > Minutes37 > > > > select now() + interval '179 days'; > ? ? ? ? ? ??column? > ------------------------------- > ?2012-08-06 13:47:33.990137+08 > (1 row) > > > Thanks. > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: ?http://www.freeswitch.org.cn > > Sent with Sparrow > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 Thu Feb 9 22:16:46 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Feb 2012 11:16:46 -0800 Subject: [Freeswitch-users] cluecon 2012 date confirm In-Reply-To: References: Message-ID: Nice work. Yes, it is Aug 7-9, but Aug 6 is the Monday before. We have a new cluecon website you can check out: www-test.cluecon.com. Stay tuned for updates - I promise they are coming! -MC On Wed, Feb 8, 2012 at 10:00 PM, Seven Du wrote: > Hi, > > Had a wonderful time in Cluecon 2011 and would like to confirm the date as > it looks like no schedule for 2012 on cluecon.com and I only found a > count down on freeswitch.org. > > So would like to confirm will it be Aug. 6th this year? > > > 179 > Days10 > Hours2 > Minutes37 > > > > > select now() + interval '179 days'; > ?column? > ------------------------------- > 2012-08-06 13:47:33.990137+08 > (1 row) > > > Thanks. > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > Sent with Sparrow > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/ebd29bab/attachment.html From krice at freeswitch.org Thu Feb 9 22:18:54 2012 From: krice at freeswitch.org (Ken Rice) Date: Thu, 09 Feb 2012 13:18:54 -0600 Subject: [Freeswitch-users] http://www.uceprotect.net the craptastic RBL Message-ID: So... It appears the FreeSwitch list server has been listed as a Spam source at http://www.uceprotect.net UCEProtects solution wait atleast a week then maybe they will drop the list server from their spam listing or pay them $115 and they?ll delist it Oh! BTW we can prevent the server from being listed again for $115 for 2 years of white listing.... 1. Why would we want to do that? 2. Why don?t they provide a way for obvious legit projects such as FreeSWITCH to get immediately Delisted? 3. Why do they allow anyone willing to pay them a few bucks to get themselves delisted? Sounds like they are just in it for the money... If you are using this RBL drop it, its not really blocking your spam K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/48cca240/attachment.html From msc at freeswitch.org Thu Feb 9 22:20:30 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Feb 2012 11:20:30 -0800 Subject: [Freeswitch-users] unable to compile mod_shout on latest git In-Reply-To: References: Message-ID: Well, I believe Tony recently added Ogg and Flac support for mod_shout so yeah, libvorbis would make sense. Try: apt-get install libvorbis-dev Hopefully that will take care of it. -MC On Thu, Feb 9, 2012 at 4:29 AM, Brian Foster wrote: > cannot find the library > `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la' or > unhandled argument `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib// > libvorbis.la > > Probably should look at filling that dependency.... > On Feb 9, 2012 7:24 AM, "Neil Patel" wrote: > >> I'm on Ubuntu LTS and just did a pull from git. I've installed all the >> dependencies, >> and am getting this error: >> >> making all mod_shout >> Compiling /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c... >> quiet_libtool: compile: gcc >> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >> -I/usr/src/freeswitch/libs/lame-3.98.4/include >> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >> -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 >> -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE >> -DHAVE_CONFIG_H -c >> /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c -fPIC -DPIC -o >> .libs/mod_shout.o >> quiet_libtool: compile: gcc >> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >> -I/usr/src/freeswitch/libs/lame-3.98.4/include >> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >> -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 >> -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE >> -DHAVE_CONFIG_H -c >> /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c -o mod_shout.o >> >/dev/null 2>&1 >> Creating mod_shout.la... >> quiet_libtool: link: cannot find the library >> `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la' or >> unhandled argument `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib// >> libvorbis.la' >> make[5]: *** [mod_shout.la] Error 1 >> make[4]: *** [all] Error 1 >> make[3]: *** [mod_shout-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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/828d2ae5/attachment.html From jmesquita at freeswitch.org Thu Feb 9 22:20:53 2012 From: jmesquita at freeswitch.org (=?utf-8?Q?Jo=C3=A3o_Mesquita?=) Date: Thu, 9 Feb 2012 16:20:53 -0300 Subject: [Freeswitch-users] http://www.uceprotect.net the craptastic RBL In-Reply-To: References: Message-ID: <83B983042422405AA570F618E2D90769@freeswitch.org> What can we do about it? We can maybe really spam them? LOL -- Jo?o Mesquita Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Thursday, February 9, 2012 at 4:18 PM, Ken Rice wrote: > http://www.uceprotect.net the craptastic RBL So... It appears the FreeSwitch list server has been listed as a Spam source at http://www.uceprotect.net > > UCEProtects solution wait atleast a week then maybe they will drop the list server from their spam listing or pay them $115 and they?ll delist it > > Oh! BTW we can prevent the server from being listed again for $115 for 2 years of white listing.... > > Why would we want to do that? > Why don?t they provide a way for obvious legit projects such as FreeSWITCH to get immediately Delisted? > Why do they allow anyone willing to pay them a few bucks to get themselves delisted? > > Sounds like they are just in it for the money... If you are using this RBL drop it, its not really blocking your spam > > K > _________________________________________________________________________ > 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/20120209/f94b64a9/attachment-0001.html From msc at freeswitch.org Thu Feb 9 22:21:52 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Feb 2012 11:21:52 -0800 Subject: [Freeswitch-users] unable to compile mod_shout on latest git In-Reply-To: References: Message-ID: Doh! I didn't see your other email. Evidently it's not finding your libvorbis for some reason. Can someone who is familiar with these kinds of issues on Debian-based systems chime in? -MC On Thu, Feb 9, 2012 at 4:22 AM, Neil Patel wrote: > I'm on Ubuntu LTS and just did a pull from git. I've installed all the > dependencies, > and am getting this error: > > making all mod_shout > Compiling /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c... > quiet_libtool: compile: gcc > -I/usr/src/freeswitch/libs/libshout-2.2.2/include > -I/usr/src/freeswitch/libs/lame-3.98.4/include > -I/usr/src/freeswitch/libs/mpg123-1.13.2/src > -I/usr/src/freeswitch/libs/mpg123-1.13.2/src > -I/usr/src/freeswitch/libs/libshout-2.2.2/include > -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 > -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE > -DHAVE_CONFIG_H -c > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c -fPIC -DPIC -o > .libs/mod_shout.o > quiet_libtool: compile: gcc > -I/usr/src/freeswitch/libs/libshout-2.2.2/include > -I/usr/src/freeswitch/libs/lame-3.98.4/include > -I/usr/src/freeswitch/libs/mpg123-1.13.2/src > -I/usr/src/freeswitch/libs/mpg123-1.13.2/src > -I/usr/src/freeswitch/libs/libshout-2.2.2/include > -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 > -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE > -DHAVE_CONFIG_H -c > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c -o mod_shout.o > >/dev/null 2>&1 > Creating mod_shout.la... > quiet_libtool: link: cannot find the library > `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la' or > unhandled argument `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib// > libvorbis.la' > make[5]: *** [mod_shout.la] Error 1 > make[4]: *** [all] Error 1 > make[3]: *** [mod_shout-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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/f7e4383e/attachment.html From msc at freeswitch.org Thu Feb 9 22:25:26 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Feb 2012 11:25:26 -0800 Subject: [Freeswitch-users] http://www.uceprotect.net the craptastic RBL In-Reply-To: References: Message-ID: Smells like a scam to me. I would stay away from them, not give them a dime, etc. Most likely someone doofus signed up for our mailing list and then, instead of just unsubscribing, he kept reporting our list emails as spam to these guys. Anyway, anyone who uses this service should either drop them or whitelist our list server. -MC On Thu, Feb 9, 2012 at 11:18 AM, Ken Rice wrote: > So... It appears the FreeSwitch list server has been listed as a Spam > source at http://www.uceprotect.net > > UCEProtects solution wait atleast a week then maybe they will drop the > list server from their spam listing or pay them $115 and they?ll delist it > > Oh! BTW we can prevent the server from being listed again for $115 for 2 > years of white listing.... > > > 1. Why would we want to do that? > 2. Why don?t they provide a way for obvious legit projects such as > FreeSWITCH to get immediately Delisted? > 3. Why do they allow anyone willing to pay them a few bucks to get > themselves delisted? > > > Sounds like they are just in it for the money... If you are using this RBL > drop it, its not really blocking your spam > > K > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/66974e72/attachment.html From brian at freeswitch.org Thu Feb 9 22:25:44 2012 From: brian at freeswitch.org (Brian West) Date: Thu, 9 Feb 2012 13:25:44 -0600 Subject: [Freeswitch-users] http://www.uceprotect.net the craptastic RBL In-Reply-To: References: Message-ID: This usually means that some idiot^H^H^H^H^Hperson has subscribed to our list... gotten mad because of their own inability to unsubscribe, then they report it. This just means that if you use UCEProtect.net you don't get our list emails thats just how we are going to roll on this one. /b On Feb 9, 2012, at 1:18 PM, Ken Rice wrote: > So... It appears the FreeSwitch list server has been listed as a Spam source > at http://www.uceprotect.net > > UCEProtects solution wait atleast a week then maybe they will drop the list > server from their spam listing or pay them $115 and they?ll delist it > > Oh! BTW we can prevent the server from being listed again for $115 for 2 > years of white listing.... > > 1. Why would we want to do that? > 2. Why don?t they provide a way for obvious legit projects such as > FreeSWITCH to get immediately Delisted? > 3. Why do they allow anyone willing to pay them a few bucks to get > themselves delisted? > > Sounds like they are just in it for the money... If you are using this RBL > drop it, its not really blocking your spam > > K -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/b4240c27/attachment.html From brian at freeswitch.org Thu Feb 9 22:26:28 2012 From: brian at freeswitch.org (Brian West) Date: Thu, 9 Feb 2012 13:26:28 -0600 Subject: [Freeswitch-users] http://www.uceprotect.net the craptastic RBL In-Reply-To: <83B983042422405AA570F618E2D90769@freeswitch.org> References: <83B983042422405AA570F618E2D90769@freeswitch.org> Message-ID: <7AE6624D-1F23-42CA-8E59-AF7419FC2088@freeswitch.org> This is where a black market for GPS guided smart bombs would really thrive! On Feb 9, 2012, at 1:20 PM, Jo?o Mesquita wrote: > What can we do about it? We can maybe really spam them? LOL > > -- > Jo?o Mesquita > Sent with Sparrow (http://www.sparrowmailapp.com/?sig) -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/59241914/attachment.html From msc at freeswitch.org Thu Feb 9 22:33:12 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Feb 2012 11:33:12 -0800 Subject: [Freeswitch-users] finally, an open alternative to Viber In-Reply-To: <4F32F306.9050309@lumicall.org> References: <4F32F306.9050309@lumicall.org> Message-ID: Daniel, This looks quite interesting. I think we'll have some of our users test it out. I hope this kind of service catches on. Question: Would you mind doing a minor update on your site? Could you change "FreeSwitch" to "FreeSWITCH"? :D Thanks, Michael On Wed, Feb 8, 2012 at 2:11 PM, Daniel - Lumicall.org wrote: > > > The rise of Viber has, for some people, been a case of `skype, not again!' > > until now... > > Lumicall is now in the Android market - and it fully interacts with > other SIP products using ENUM and SRV records. Any feedback on this is > welcome. > > Interconnect (FreeSwitch calling to/from Lumicall users) is based on the > idea of federated SIP, it is explained at http://www.lumicall.org for > those who want to connect up to it. > > Please bear in mind: Lumicall supports ICE (RFC 5245) for NAT traversal, > it is using the ice4j implementation from the Jitsi community. This > makes the SIP packets bigger and often they are too big for the MTU of a > UDP packet. When using ICE, it seems essential to use TLS, to avoid the > MTU problems and also to avoid routers mangling the SIP headers (ICE > doesn't need help from routers, they only confuse the algorithm) > > Various other issues are addressed in the release notes and FAQ > http://www.lumicall.org/faq > > If there are specific issues with FreeSwitch compatibility (either for > the Lumicall app, or for interconnect using ENUM/SRV/TLS) then I'm happy > to start a dedicated page or wiki to collect the solutions in one place > for FreeSwitch users. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/1c7f5ac9/attachment-0001.html From oseslija at gmail.com Thu Feb 9 23:43:39 2012 From: oseslija at gmail.com (Ognjen Seslija) Date: Thu, 9 Feb 2012 21:43:39 +0100 Subject: [Freeswitch-users] presence on latest GIT In-Reply-To: References: Message-ID: I'm running in production a version of 30-Oct-2011 and it's working all fine. On Thu, Feb 9, 2012 at 6:28 AM, Yehavi Bourvine wrote: > Hello Vik, > > BLF behaviour is somewhat erratic since major changes done after > 15-Oct-2011. I am working with Anthony to track it down (see FS-3794). It > happens mostly on our production system rather than the test system, so I > am quite limited in testing the fixes and it takes time... > > Here is a sample config of phone 11111 doing BLF on 22222: > > > reg.1.label="11111" reg.1.type="shared" reg.1.auth.userId="11111" > reg.1.auth.password="PPPPPP" reg.1.ringType="2" /> > > attendant.resourceList.1.label="22222" /> > > Note that *ThirdPartyName* is *not* used in the REG section. > > Regards, __Yehavi: > > > 2012/2/9 Vik Killa > >> Would anyone be willing to share their working BLF polycom configs? >> >> On Wed, Feb 8, 2012 at 4:56 PM, Anthony Minessale >> wrote: >> > It should be working but you can use that data to see why. >> > If you enable sip trace too and capture the logs, it can be used to >> > check for problems >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/fc4d4aca/attachment.html From anthony.minessale at gmail.com Fri Feb 10 00:31:09 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Feb 2012 15:31:09 -0600 Subject: [Freeswitch-users] presence on latest GIT In-Reply-To: References: Message-ID: Don't encourage october, its gone... never coming back..... Find something actually wrong, check multi homed params, we found many bugs even going back to then that were impacting using many domains and things may have to change for good. On Thu, Feb 9, 2012 at 2:43 PM, Ognjen Seslija wrote: > I'm running in production a version of 30-Oct-2011 and it's working all > fine. > > > On Thu, Feb 9, 2012 at 6:28 AM, Yehavi Bourvine > wrote: >> >> Hello Vik, >> >> ? BLF behaviour is somewhat erratic since major changes done after >> 15-Oct-2011. I am working with Anthony to track it down (see FS-3794). It >> happens mostly on our production system rather than the test system, so I am >> quite limited in testing the fixes and it takes time... >> >> Here is a sample config of phone 11111 doing BLF on 22222: >> >> ? >> ??? > reg.1.label="11111" reg.1.type="shared" reg.1.auth.userId="11111" >> reg.1.auth.password="PPPPPP" reg.1.ringType="2" /> >> >> > attendant.resourceList.1.label="22222"?/> >> >> Note that ThirdPartyName is not used in the REG section. >> >> ???????????????????????? Regards, __Yehavi: >> >> >> 2012/2/9 Vik Killa >>> >>> Would anyone be willing to share their working BLF polycom configs? >>> >>> On Wed, Feb 8, 2012 at 4:56 PM, Anthony Minessale >>> wrote: >>> > It should be working but you can use that data to see why. >>> > If you enable sip trace too and capture the logs, it can be used to >>> > check for problems >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> 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 >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 oseslija at gmail.com Fri Feb 10 00:38:30 2012 From: oseslija at gmail.com (Ognjen Seslija) Date: Thu, 9 Feb 2012 22:38:30 +0100 Subject: [Freeswitch-users] presence on latest GIT In-Reply-To: References: Message-ID: Of course. Just wanted to point out that 15-Oct isn't the last known good-BLF day. And yes, I tested with Tony lately new code (Tony was on my machine coding and myself dialing etc.) and it was all fine. I will test more and respond. On Thu, Feb 9, 2012 at 10:31 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Don't encourage october, its gone... never coming back..... > Find something actually wrong, check multi homed params, we found many > bugs even going back to then that were impacting using many domains > and things may have to change for good. > > > > On Thu, Feb 9, 2012 at 2:43 PM, Ognjen Seslija wrote: > > I'm running in production a version of 30-Oct-2011 and it's working all > > fine. > > > > > > On Thu, Feb 9, 2012 at 6:28 AM, Yehavi Bourvine < > yehavi.bourvine at gmail.com> > > wrote: > >> > >> Hello Vik, > >> > >> BLF behaviour is somewhat erratic since major changes done after > >> 15-Oct-2011. I am working with Anthony to track it down (see FS-3794). > It > >> happens mostly on our production system rather than the test system, so > I am > >> quite limited in testing the fixes and it takes time... > >> > >> Here is a sample config of phone 11111 doing BLF on 22222: > >> > >> > >> >> reg.1.label="11111" reg.1.type="shared" reg.1.auth.userId="11111" > >> reg.1.auth.password="PPPPPP" reg.1.ringType="2" /> > >> > >> >> attendant.resourceList.1.label="22222" /> > >> > >> Note that ThirdPartyName is not used in the REG section. > >> > >> Regards, __Yehavi: > >> > >> > >> 2012/2/9 Vik Killa > >>> > >>> Would anyone be willing to share their working BLF polycom configs? > >>> > >>> On Wed, Feb 8, 2012 at 4:56 PM, Anthony Minessale > >>> wrote: > >>> > It should be working but you can use that data to see why. > >>> > If you enable sip trace too and capture the logs, it can be used to > >>> > check for problems > >>> > >>> > >>> > >>> > _________________________________________________________________________ > >>> Professional FreeSWITCH Consulting Services: > >>> 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 > >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >>> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >>> http://www.freeswitch.org > >> > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> 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 > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > 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 > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/96ed29a6/attachment-0001.html From daniel at lumicall.org Fri Feb 10 01:02:57 2012 From: daniel at lumicall.org (Daniel - Lumicall.org) Date: Thu, 09 Feb 2012 23:02:57 +0100 Subject: [Freeswitch-users] finally, an open alternative to Viber In-Reply-To: References: <4F32F306.9050309@lumicall.org> Message-ID: <4F344291.2000009@lumicall.org> > This looks quite interesting. I think we'll have some of our users test > it out. I hope this kind of service catches on. > Viber had 1 million users in their first month - it would be really good to see open source/open standards making similar progress > Question: Would you mind doing a minor update on your site? Could you > change "FreeSwitch" to "FreeSWITCH"? :D Done From holger at freyther.de Thu Feb 9 23:52:15 2012 From: holger at freyther.de (Holger Freyther) Date: Thu, 9 Feb 2012 20:52:15 +0000 (UTC) Subject: [Freeswitch-users] How to inspect the jitterbuffer? Message-ID: Hi all, I have a setup with a FreeTDM/wanpipe trunk and most of the time the audio skips on the path to this trunk (e.g. after a period of silence, RTP marker bit set by the sender). It is not an issue with FreeTDM/wanpipe but more the way the RTP input is handled by FreeSWITCH. My theory is the audio skips because the RTP 'queue' has an under-run (while it shouldn't, there is no packet loss). So how can I see if the jitterbuffer is enabled for a given call (or globally)? Can I somehow see under-runs? The recovered clock/ clock skew? I couldn't find any of this in the wiki[1]. thanks holger [1] http://wiki.freeswitch.org/wiki/Jitterbuffer From msc at freeswitch.org Fri Feb 10 01:33:16 2012 From: msc at freeswitch.org (Michael Collins) Date: Thu, 9 Feb 2012 14:33:16 -0800 Subject: [Freeswitch-users] finally, an open alternative to Viber In-Reply-To: <4F344291.2000009@lumicall.org> References: <4F32F306.9050309@lumicall.org> <4F344291.2000009@lumicall.org> Message-ID: On Thu, Feb 9, 2012 at 2:02 PM, Daniel - Lumicall.org wrote: > > > This looks quite interesting. I think we'll have some of our users test > > it out. I hope this kind of service catches on. > > > > Viber had 1 million users in their first month - it would be really good > to see open source/open standards making similar progress > Agreed! > > > > Question: Would you mind doing a minor update on your site? Could you > > change "FreeSwitch" to "FreeSWITCH"? :D > > Done > Thanks for indulging my OCD. :P -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/015e5b57/attachment.html From mojo1736 at privatedemail.net Fri Feb 10 02:14:39 2012 From: mojo1736 at privatedemail.net (Josh) Date: Thu, 09 Feb 2012 23:14:39 +0000 Subject: [Freeswitch-users] [newbie] questions In-Reply-To: References: <4F330A31.4060804@privatedemail.net> <4F33D39D.5070807@privatedemail.net> <4F33E525.4070305@privatedemail.net> Message-ID: <4F34535F.6050703@privatedemail.net> > I (as the called party) have the option (by pressing a specific key > after receiving the caller's name) to: 1) accept the call; 2) send a > message back to the caller saying this number is unavailable and then > terminate his/her call; 3) redirect him/her to voicemail; 4) drop the > call entirely without giving any explanation at all (or, > alternatively, > torture them with some moh if they appear to be telemarketing > group for > example). By reading that example, it looks as though this can be > achieved. > > If you only need "answer or reject to continue with the dialplan" then > simply use the pre-built Answer Confirmation: > http://wiki.freeswitch.org/wiki/Freeswitch_IVR_Originate#Answer_confirmation > Anything else will be a bit more complicated. According to the example I posted earlier (Example 13) it all seemed configurable as all of the steps taken were separate actions with parameters, which were completely configurable, so, naturally, I presumed that I could tweak this a bit to suit my own needs, isn't that so? > For telemarketers, you could bridge them to http://itslenny.com/ :-) Thanks, I'll keep that in mind! > When installing if you did "make sounds-install" you got the basic > sound package which is quite extensive. > I recommend looking at the docs/phrase/phrase_en.xml and see the list > of the phrases / pre-recorded sound files under the name "callie". I will - thanks for the pointer. I haven't reached the "installation" stage yet - I am yet to decide whether to use FreeSWITCH, but will certainly look at the sound/media resources provided with it. Thanks again for your input. From mojo1736 at privatedemail.net Fri Feb 10 02:14:56 2012 From: mojo1736 at privatedemail.net (Josh) Date: Thu, 09 Feb 2012 23:14:56 +0000 Subject: [Freeswitch-users] [newbie] questions In-Reply-To: References: <4F330A31.4060804@privatedemail.net> <4F33D39D.5070807@privatedemail.net> Message-ID: <4F345370.5090905@privatedemail.net> > If you're willing to get your hands dirty then that's great. I > recommend the following resources: All noted, except irc - not really an option for me at the moment. Thanks! > No matter which way you go, be prepared for aggravating problems and > pulling your hair out. VoIP and telephony are always frustrating for > the new user. Most of our community members remember their newbie pain > and are quite willing to offer their knowledge and experience to help > you keep at least some of your hair. ;) Hope so - the main reason I joined this ML. > Let us know what you decide! I haven't made a decision yet one way or another - "the devil is in the detail" as they say. I would like to identify potential "issues" or "stumbling blocks" as early as possible, so that I don't regret it later on (and start pulling my hair out :-) ) when I truly delve in - I do not wish to spend long hours researching, as well as part with my hard-earned, just to find out midway through that something I'd like to do won't be possible for one reason or another. That would be a complete waste, literally. It would be nice if someone could address my questions, if possible (highlighted again below - with one or two additions ;-) ) - some of these issues I know are not as straight forward in Asterisk, so I'd like to find out how these are solved/implemented in FreeSWITCH? > > Yes, FreeSWITCH can bind to multiple interfaces. In FreeSWITCH lingo > > that would mean that you set up a separate SIP profile for each > > interface. (In fact, you can have more than one SIP profile on a > given > > interface since the profile is a unique combo of IP addr and > port number.) > I presume different profiles can "talk" to each other, right? In other > words calls/media can be routed/transferred from one interface to > another (eth1<->tun0 for example)? > Is this relatively straight-forward to configure? > > "Some assembly required." :D > > FreeSWITCH can do some stuff for you, but you definitely need to > make > > sure that your NAT is not behaving badly, like having a SIP ALG. > This is what I am trying to figure out - do I rely entirely on > FreeSWITCH (if not, what is expected of me to set up so that > FreeSWITCH > can do its job?), or do I have to do it all by myself with the kernel > module helpers (sip, h323 etc) and ip/iptables? > The reason I asked this, is simply because I need to know whether I am required to set it all up (nothing wrong with that, just that I don't want to be put in a position where I stumble across a problem and don't know who is to blame - my sip helper module, iptables, or FreeSWITCH itself) or do I leave it all to FreeSWITCH. Another issue which I did not mention up until now - I *assume* FreeSWITCH operates in a similar way as Asterisk does with regards to separation of channels/streams - in other words, if someone calls via the "public" interface/my external VOIP provider, then FreeSWITCH goes through the configured dialplan and then opens up a separate channel (on the internal interface) to ring and connect the call and then "bridges" the two streams/channels together. Is that the case? If so, would it be possible to have different levels of encryption/security set on the two channels - say TLS/SRTP on the public part and no encryption on the internal system? Would that be possible or do I have to have the same settings on all interfaces so that FreeSWITCH could handle it properly? > > I'll have to defer to Ken Rice on this one. I know he's working on > > RPMs for FreeSWITCH but I think it's all RedHat right now. > RedHat is good, all I need is a decent .spec file - I'll do the rest > myself, no problem. > That, really, is important to me as this is how I'd like to start and experiment. Once I have the .spec file I would be able to package it and distribute it across a couple of testing machine I have here so that I could experiment a bit. The distro this .spec file comes from doesn't matter too much - whether it is Fedora/RedHat, SUSE, etc I would be able to change the .spec file and adapt it to my system - I have done this many times before, so it won't be too much of a problem for me, I think, provided the .spec file is decent. Many thanks yet again! From cmrienzo at gmail.com Fri Feb 10 02:26:58 2012 From: cmrienzo at gmail.com (cmrienzo at gmail.com) Date: Thu, 9 Feb 2012 18:26:58 -0500 Subject: [Freeswitch-users] How to inspect the jitterbuffer? In-Reply-To: References: Message-ID: <7044FA15-75AD-404F-9AA8-AF3B95308E48@gmail.com> The jitter buffer is not enabled by default. On Feb 9, 2012, at 15:52, Holger Freyther wrote: > Hi all, > > I have a setup with a FreeTDM/wanpipe trunk and most of the time > the audio skips on the path to this trunk (e.g. after a period > of silence, RTP marker bit set by the sender). It is not an issue > with FreeTDM/wanpipe but more the way the RTP input is handled by > FreeSWITCH. My theory is the audio skips because the RTP 'queue' > has an under-run (while it shouldn't, there is no packet loss). > > So how can I see if the jitterbuffer is enabled for a given call > (or globally)? Can I somehow see under-runs? The recovered clock/ > clock skew? I couldn't find any of this in the wiki[1]. > > thanks > holger > > > [1] http://wiki.freeswitch.org/wiki/Jitterbuffer > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From aksrini at hotmail.com Fri Feb 10 02:28:26 2012 From: aksrini at hotmail.com (Srini K) Date: Thu, 9 Feb 2012 15:28:26 -0800 Subject: [Freeswitch-users] FreeSWITCH ends the calls after receiving DTMF from the Callee. Message-ID: Hi,I have configured FreeSWITCH to receive DTMF by in both sip_profile internal and external files. Iam using mod managed and I have subscribed to receive DTMF events.I can process DTMF from Caller without any problem. When I receive DTMF from the callee as Sip Info, I do receive the DTMF event from the FreeSWITCH and immediately FreeSWITCH disconnects the call. Snapshot of the log is... [DEBUG] sofia.c:7229 INFO DTMF(1) [DEBUG] sofia.c:7307 dispatched freeswitch event for INFO [DEBUG] switch_core_session.c:875 Send signal sofia/external/yyyyyyyyyy [BREAK] [DEBUG] switch_ivr_bridge.c:383 sofia/external/yyyyyyyyyy ended call via DTMF [DEBUG] switch_ivr_bridge.c:384 Send signal sofia/external/xxxxxxxxxx at 10.10.2.3 [BREAK] [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/external/yyyyyyyyyy] Whether Iam missing anything in the config? Thanks in advance.Srini -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/423e4f96/attachment.html From anthony.minessale at gmail.com Fri Feb 10 02:33:01 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Feb 2012 17:33:01 -0600 Subject: [Freeswitch-users] FreeSWITCH ends the calls after receiving DTMF from the Callee. In-Reply-To: References: Message-ID: bridge with input_callback and did not return success. On Thu, Feb 9, 2012 at 5:28 PM, Srini K wrote: > Hi, > I have configured FreeSWITCH to receive DTMF by > > > > > in both sip_profile internal and external files. > > Iam using mod managed and I have subscribed to receive DTMF events. > I can process DTMF from Caller without any problem. When I receive DTMF from > the callee as Sip Info, I do receive the DTMF event from the FreeSWITCH and > immediately FreeSWITCH disconnects the call. > > Snapshot of the log is... > > [DEBUG] sofia.c:7229 INFO DTMF(1) > [DEBUG] sofia.c:7307 dispatched freeswitch event for INFO > [DEBUG] switch_core_session.c:875 Send signal sofia/external/yyyyyyyyyy > [BREAK] > [DEBUG] switch_ivr_bridge.c:383 sofia/external/yyyyyyyyyy ended call via > DTMF > [DEBUG] switch_ivr_bridge.c:384 Send signal > sofia/external/xxxxxxxxxx at 10.10.2.3 [BREAK] > [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE > [sofia/external/yyyyyyyyyy] > > Whether Iam missing anything in the config? > > Thanks in advance. > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 hynek.cihlar at gmail.com Fri Feb 10 02:38:22 2012 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Fri, 10 Feb 2012 00:38:22 +0100 Subject: [Freeswitch-users] INCOMING busy tone detect Message-ID: The spandsp mod is fantastic! Can it be configured to detect tones originated through a specific sip profile so I would be able to tell whether the tone was originated from the remote side? Hynek -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/17ff1fdd/attachment.html From aksrini at hotmail.com Fri Feb 10 03:06:11 2012 From: aksrini at hotmail.com (Srini K) Date: Thu, 9 Feb 2012 16:06:11 -0800 Subject: [Freeswitch-users] FreeSWITCH ends the calls after receiving DTMF from the Callee. In-Reply-To: References: , Message-ID: Any reason that it didn't return success. Anything Iam missing it out. > Date: Thu, 9 Feb 2012 17:33:01 -0600 > From: anthony.minessale at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after receiving DTMF from the Callee. > > bridge with input_callback and did not return success. > > On Thu, Feb 9, 2012 at 5:28 PM, Srini K wrote: > > Hi, > > I have configured FreeSWITCH to receive DTMF by > > > > > > > > > > in both sip_profile internal and external files. > > > > Iam using mod managed and I have subscribed to receive DTMF events. > > I can process DTMF from Caller without any problem. When I receive DTMF from > > the callee as Sip Info, I do receive the DTMF event from the FreeSWITCH and > > immediately FreeSWITCH disconnects the call. > > > > Snapshot of the log is... > > > > [DEBUG] sofia.c:7229 INFO DTMF(1) > > [DEBUG] sofia.c:7307 dispatched freeswitch event for INFO > > [DEBUG] switch_core_session.c:875 Send signal sofia/external/yyyyyyyyyy > > [BREAK] > > [DEBUG] switch_ivr_bridge.c:383 sofia/external/yyyyyyyyyy ended call via > > DTMF > > [DEBUG] switch_ivr_bridge.c:384 Send signal > > sofia/external/xxxxxxxxxx at 10.10.2.3 [BREAK] > > [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE > > [sofia/external/yyyyyyyyyy] > > > > Whether Iam missing anything in the config? > > > > Thanks in advance. > > 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 > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/235c1ecb/attachment.html From anthony.minessale at gmail.com Fri Feb 10 03:08:55 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Feb 2012 18:08:55 -0600 Subject: [Freeswitch-users] Registration problem after attempting to install mod_opal and updating to latest git pull In-Reply-To: <00ba01cce6ee$d1422640$73c672c0$@com> References: <00ba01cce6ee$d1422640$73c672c0$@com> Message-ID: turn on sip trace sofia global siptrace on And run it again with the ping enabled. and we can see the traffic, we made some changes to options ping that someone requested that were supposed to make it more compliant so maybe your provider does not like this. If you get the trace I can have a look. On Wed, Feb 8, 2012 at 11:51 PM, Phil Quesinberry wrote: > After a recent unsuccessful attempt to install mod_opal and updating to a > recent git, I now get some odd reports of gateway registration failures, > although everything appears to be working ok call-wise for the moment. > Doing a 'make uninstall' from /root/opal and updating to the latest git > tonight yielded no improvement. > > Here are some excerpts of failures reported in fs_cli: > > 2012-02-09 00:12:59.992099 [WARNING] sofia.c:4717 Unregister QS8002 > 2012-02-09 00:12:59.992099 [WARNING] sofia.c:4730 Ping failed QS8002 with > code 900 - count -1/-1/1, state UP > 2012-02-09 00:12:59.992099 [WARNING] sofia.c:4717 Unregister HL7612 > 2012-02-09 00:12:59.992099 [WARNING] sofia.c:4730 Ping failed HL7612 with > code 900 - count -1/-1/1, state UP > 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4717 Unregister HL7611 > 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4730 Ping failed HL7611 with > code 900 - count -1/-1/1, state UP > 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4717 Unregister HL7519 > 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4730 Ping failed HL7519 with > code 900 - count -1/-1/1, state UP > 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4717 Unregister > CallCentric_T.38 > 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4730 Ping failed > CallCentric_T.38 with code 900 - count -1/-1/1, state UP > 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 QS8002 Failed > Registration [0], setting retry to 60 seconds. > 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 HL7612 Failed > Registration [0], setting retry to 90 seconds. > 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 HL7611 Failed > Registration [0], setting retry to 90 seconds. > 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 HL7519 Failed > Registration [0], setting retry to 60 seconds. > 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 CallCentric_T.38 Failed > Registration [0], setting retry to 60 seconds. > > To give a sense of the timing, the three groups of warnings below are > contiguous, from a single fs_cli session: > > 2012-02-09 00:21:18.984384 [WARNING] sofia.c:4717 Unregister HL7611 > 2012-02-09 00:21:18.984384 [WARNING] sofia.c:4730 Ping failed HL7611 with > code 900 - count -1/-1/1, state UP > 2012-02-09 00:21:20.988379 [WARNING] sofia_reg.c:474 HL7611 Failed > Registration [0], setting retry to 90 seconds. > 2012-02-09 00:21:22.988382 [WARNING] sofia.c:4717 Unregister HL7519 > 2012-02-09 00:21:22.988382 [WARNING] sofia.c:4730 Ping failed HL7519 with > code 900 - count -1/-1/1, state UP > 2012-02-09 00:21:24.988318 [WARNING] sofia_reg.c:474 HL7519 Failed > Registration [0], setting retry to 60 seconds. > > 2012-02-09 00:22:00.447778 [WARNING] sofia_reg.c:1422 SIP auth challenge > (REGISTER) on sofia profile 'internal' for [226 at qsystemseng.no-ip.org] from > ip 74.93.222.182 > 2012-02-09 00:22:04.147688 [WARNING] sofia.c:4717 Unregister HL7612 > 2012-02-09 00:22:04.147688 [WARNING] sofia.c:4730 Ping failed HL7612 with > code 900 - count -1/-1/1, state UP > 2012-02-09 00:22:04.167699 [WARNING] sofia.c:4717 Unregister HL7611 > 2012-02-09 00:22:04.167699 [WARNING] sofia.c:4730 Ping failed HL7611 with > code 900 - count -1/-1/1, state UP > 2012-02-09 00:22:04.167699 [WARNING] sofia.c:4717 Unregister > CallCentric_T.38 > 2012-02-09 00:22:04.167699 [WARNING] sofia.c:4730 Ping failed > CallCentric_T.38 with code 900 - count -1/-1/1, state UP > > 2012-02-09 00:22:05.167744 [WARNING] sofia_reg.c:474 HL7611 Failed > Registration [0], setting retry to 90 seconds. > 2012-02-09 00:22:05.167744 [WARNING] sofia_reg.c:474 CallCentric_T.38 Failed > Registration [0], setting retry to 60 seconds. > 2012-02-09 00:22:05.167744 [WARNING] sofia.c:4717 Unregister QS8002 > 2012-02-09 00:22:05.167744 [WARNING] sofia.c:4730 Ping failed QS8002 with > code 900 - count -1/-1/1, state UP > > 2012-02-09 00:22:07.171680 [WARNING] sofia_reg.c:474 QS8002 Failed > Registration [0], setting retry to 60 seconds. > 2012-02-09 00:22:07.171680 [WARNING] sofia.c:4717 Unregister HL7519 > 2012-02-09 00:22:07.171680 [WARNING] sofia.c:4730 Ping failed HL7519 with > code 900 - count -1/-1/1, state UP > > One of the gateways is SIPBroker, and calls through that gate starting > failing after the update. ?I realized that the gateway was configured to > ping SIPBroker unnecessarily and restarting the gateway would restore > service until the ping. ?I disabled the ping for that gateway and calls kept > working after that. > > Any ideas as to what might be going on? ?There was no such problem before > the FS update/attempt to use mod_opal. ?I can provide additional/more > detailed info if needed. > > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 Fri Feb 10 03:17:45 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Feb 2012 18:17:45 -0600 Subject: [Freeswitch-users] FreeSWITCH ends the calls after receiving DTMF from the Callee. In-Reply-To: References: Message-ID: the obvious reason would be setting bridge_terminate_key=1 but I don't know where your bridge is being started based on your description or what version of the code you are using. On Thu, Feb 9, 2012 at 6:06 PM, Srini K wrote: > Any reason that it didn't return success. Anything Iam missing it out. > > >> Date: Thu, 9 Feb 2012 17:33:01 -0600 >> From: anthony.minessale at gmail.com >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after receiving >> DTMF from the Callee. > >> >> bridge with input_callback and did not return success. >> >> On Thu, Feb 9, 2012 at 5:28 PM, Srini K wrote: >> > Hi, >> > I have configured FreeSWITCH to receive DTMF by >> > >> > >> > >> > >> > in both sip_profile internal and external files. >> > >> > Iam using mod managed and I have subscribed to receive DTMF events. >> > I can process DTMF from Caller without any problem. When I receive DTMF >> > from >> > the callee as Sip Info, I do receive the DTMF event from the FreeSWITCH >> > and >> > immediately FreeSWITCH disconnects the call. >> > >> > Snapshot of the log is... >> > >> > [DEBUG] sofia.c:7229 INFO DTMF(1) >> > [DEBUG] sofia.c:7307 dispatched freeswitch event for INFO >> > [DEBUG] switch_core_session.c:875 Send signal sofia/external/yyyyyyyyyy >> > [BREAK] >> > [DEBUG] switch_ivr_bridge.c:383 sofia/external/yyyyyyyyyy ended call via >> > DTMF >> > [DEBUG] switch_ivr_bridge.c:384 Send signal >> > sofia/external/xxxxxxxxxx at 10.10.2.3 [BREAK] >> > [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE >> > [sofia/external/yyyyyyyyyy] >> > >> > Whether Iam missing anything in the config? >> > >> > Thanks in advance. >> > 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 >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 Fri Feb 10 03:27:17 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Feb 2012 18:27:17 -0600 Subject: [Freeswitch-users] How to inspect the jitterbuffer? In-Reply-To: References: Message-ID: what does /etc/wanpipe/wanpipe1.conf look like? and which rev of FS and wanpipe driver are you on. What ptime is the sip traffic? On Thu, Feb 9, 2012 at 2:52 PM, Holger Freyther wrote: > Hi all, > > I have a setup with a FreeTDM/wanpipe trunk and most of the time > the audio skips on the path to this trunk (e.g. after a period > of silence, RTP marker bit set by the sender). It is not an issue > with FreeTDM/wanpipe but more the way the RTP input is handled by > FreeSWITCH. My theory is the audio skips because the RTP 'queue' > has an under-run (while it shouldn't, there is no packet loss). > > So how can I see if the jitterbuffer is enabled for a given call > (or globally)? Can I somehow see under-runs? The recovered clock/ > clock skew? I couldn't find any of this in the wiki[1]. > > thanks > ?holger > > > [1] http://wiki.freeswitch.org/wiki/Jitterbuffer > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 aksrini at hotmail.com Fri Feb 10 03:37:03 2012 From: aksrini at hotmail.com (Srini K) Date: Thu, 9 Feb 2012 16:37:03 -0800 Subject: [Freeswitch-users] FreeSWITCH ends the calls after receiving DTMF from the Callee. In-Reply-To: References: , , , Message-ID: Iam using the latest FreeSWITCh version (updated today morning). FS terminates on receiving any DTMF digits from the callee. I have not set bridge_terminate_key. Iam creation an oubound session and bridging the inbound and oubound session. Code snippet is var session = new ManagedSession("{origination_caller_id_number=" + callerIdNumber + ",originate_timeout=8" + "}sofia/gateway/408xxxxyyyy"); string outBoundUuid = session.GetVariable("uuid"); if (string.IsNullOrEmpty(outBoundUuid)) { // Log error; return; } freeswitch.bridge(inboundSession, session); Also I have tried setting hangup_after_bridge=false. RegardsSrini > Date: Thu, 9 Feb 2012 18:17:45 -0600 > From: anthony.minessale at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after receiving DTMF from the Callee. > > the obvious reason would be setting bridge_terminate_key=1 > but I don't know where your bridge is being started based on your > description or what version of the code you are using. > > On Thu, Feb 9, 2012 at 6:06 PM, Srini K wrote: > > Any reason that it didn't return success. Anything Iam missing it out. > > > > > >> Date: Thu, 9 Feb 2012 17:33:01 -0600 > >> From: anthony.minessale at gmail.com > >> To: freeswitch-users at lists.freeswitch.org > >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after receiving > >> DTMF from the Callee. > > > >> > >> bridge with input_callback and did not return success. > >> > >> On Thu, Feb 9, 2012 at 5:28 PM, Srini K wrote: > >> > Hi, > >> > I have configured FreeSWITCH to receive DTMF by > >> > > >> > > >> > > >> > > >> > in both sip_profile internal and external files. > >> > > >> > Iam using mod managed and I have subscribed to receive DTMF events. > >> > I can process DTMF from Caller without any problem. When I receive DTMF > >> > from > >> > the callee as Sip Info, I do receive the DTMF event from the FreeSWITCH > >> > and > >> > immediately FreeSWITCH disconnects the call. > >> > > >> > Snapshot of the log is... > >> > > >> > [DEBUG] sofia.c:7229 INFO DTMF(1) > >> > [DEBUG] sofia.c:7307 dispatched freeswitch event for INFO > >> > [DEBUG] switch_core_session.c:875 Send signal sofia/external/yyyyyyyyyy > >> > [BREAK] > >> > [DEBUG] switch_ivr_bridge.c:383 sofia/external/yyyyyyyyyy ended call via > >> > DTMF > >> > [DEBUG] switch_ivr_bridge.c:384 Send signal > >> > sofia/external/xxxxxxxxxx at 10.10.2.3 [BREAK] > >> > [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE > >> > [sofia/external/yyyyyyyyyy] > >> > > >> > Whether Iam missing anything in the config? > >> > > >> > Thanks in advance. > >> > 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 > >> > > >> > FreeSWITCH-users mailing list > >> > FreeSWITCH-users at lists.freeswitch.org > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://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 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > 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 > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/abce0682/attachment.html From anthony.minessale at gmail.com Fri Feb 10 04:06:39 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Feb 2012 19:06:39 -0600 Subject: [Freeswitch-users] FreeSWITCH ends the calls after receiving DTMF from the Callee. In-Reply-To: References: Message-ID: do you have input_callback or dtmf_callback or anything defined? On Thu, Feb 9, 2012 at 6:37 PM, Srini K wrote: > Iam using the latest FreeSWITCh version (updated today morning). > > FS terminates on receiving any DTMF digits from the callee. I have not set > bridge_terminate_key. > > Iam creation an oubound session and bridging the inbound and oubound > session. Code snippet is > > var session = new ManagedSession("{origination_caller_id_number=" + > callerIdNumber + ",originate_timeout=8" + "}sofia/gateway/408xxxxyyyy"); > string outBoundUuid = session.GetVariable("uuid"); > if (string.IsNullOrEmpty(outBoundUuid)) > { // Log error; > ??? return; > } > freeswitch.bridge(inboundSession, session); > > Also I have tried setting hangup_after_bridge=false. > > Regards > Srini > > > >> Date: Thu, 9 Feb 2012 18:17:45 -0600 > >> From: anthony.minessale at gmail.com >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after receiving >> DTMF from the Callee. >> >> the obvious reason would be setting bridge_terminate_key=1 >> but I don't know where your bridge is being started based on your >> description or what version of the code you are using. >> >> On Thu, Feb 9, 2012 at 6:06 PM, Srini K wrote: >> > Any reason that it didn't return success. Anything Iam missing it out. >> > >> > >> >> Date: Thu, 9 Feb 2012 17:33:01 -0600 >> >> From: anthony.minessale at gmail.com >> >> To: freeswitch-users at lists.freeswitch.org >> >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after >> >> receiving >> >> DTMF from the Callee. >> > >> >> >> >> bridge with input_callback and did not return success. >> >> >> >> On Thu, Feb 9, 2012 at 5:28 PM, Srini K wrote: >> >> > Hi, >> >> > I have configured FreeSWITCH to receive DTMF by >> >> > >> >> > >> >> > >> >> > >> >> > in both sip_profile internal and external files. >> >> > >> >> > Iam using mod managed and I have subscribed to receive DTMF events. >> >> > I can process DTMF from Caller without any problem. When I receive >> >> > DTMF >> >> > from >> >> > the callee as Sip Info, I do receive the DTMF event from the >> >> > FreeSWITCH >> >> > and >> >> > immediately FreeSWITCH disconnects the call. >> >> > >> >> > Snapshot of the log is... >> >> > >> >> > [DEBUG] sofia.c:7229 INFO DTMF(1) >> >> > [DEBUG] sofia.c:7307 dispatched freeswitch event for INFO >> >> > [DEBUG] switch_core_session.c:875 Send signal >> >> > sofia/external/yyyyyyyyyy >> >> > [BREAK] >> >> > [DEBUG] switch_ivr_bridge.c:383 sofia/external/yyyyyyyyyy ended call >> >> > via >> >> > DTMF >> >> > [DEBUG] switch_ivr_bridge.c:384 Send signal >> >> > sofia/external/xxxxxxxxxx at 10.10.2.3 [BREAK] >> >> > [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE >> >> > [sofia/external/yyyyyyyyyy] >> >> > >> >> > Whether Iam missing anything in the config? >> >> > >> >> > Thanks in advance. >> >> > 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 >> >> > >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > >> >> > UNSUBSCRIBE:http://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 >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > 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 >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 sherifomran2000 at yahoo.com Fri Feb 10 05:00:10 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Thu, 9 Feb 2012 18:00:10 -0800 (PST) Subject: [Freeswitch-users] how to authenticate rtmp connection In-Reply-To: Message-ID: <1328839210.65589.YahooMailClassic@web110801.mail.gq1.yahoo.com> Hi guys, I have setup an rtmp unauthenticated connection, however i need to authenticate it, any body knows how? thank you kind regards Sherif Omran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/58d01ab5/attachment-0001.html From valery.kalinin at gmail.com Fri Feb 10 05:24:46 2012 From: valery.kalinin at gmail.com (Valery Kalinin) Date: Fri, 10 Feb 2012 08:24:46 +0600 Subject: [Freeswitch-users] FreeTDM: cannot set option Message-ID: Hi! Work with FreeTDM (latest git) & wanpipe (3.5.24) freetdm.conf.xml: but in log this options is not set: 2012-02-10 06:09:01.194586 [INFO] ftmod_wanpipe.c:401 [s1c2][1:4] Configured wanpipe device FD: 34, DTMF: softwa 2012-02-10 06:09:01.194646 [INFO] ftdm_io.c:4912 Configured 2 channel(s) 2012-02-10 06:09:01.194702 [DEBUG] mod_freetdm.c:3131 analog_spans var = enable_callerid 2012-02-10 06:09:01.194729 [DEBUG] mod_freetdm.c:3131 analog_spans var = wait_dialtone_timeout 2012-02-10 06:09:01.194748 [DEBUG] mod_freetdm.c:3131 analog_spans var = dialplan 2012-02-10 06:09:01.194757 [DEBUG] mod_freetdm.c:3131 analog_spans var = context 2012-02-10 06:09:01.195109 [INFO] ftdm_io.c:5002 Loading SIG from /usr/local/freeswitch/mod/ftmod_analog.so 2012-02-10 06:09:01.195139 [INFO] ftdm_io.c:5185 auto-loaded 'analog' 2012-02-10 06:09:01.195150 [DEBUG] ftmod_analog.c:195 Configuring span FXO for analog signaling ... 2012-02-10 06:09:01.195165 [DEBUG] ftmod_analog.c:210 Analog config var = tonemap 2012-02-10 06:09:01.195183 [DEBUG] ftmod_analog.c:210 Analog config var = digit_timeout 2012-02-10 06:09:01.195193 [DEBUG] ftmod_analog.c:210 Analog config var = max_dialstr 2012-02-10 06:09:01.195205 [DEBUG] ftmod_analog.c:210 Analog config var = hotline 2012-02-10 06:09:01.195223 [DEBUG] ftmod_analog.c:210 Analog config var = enable_callerid 2012-02-10 06:09:01.195232 [DEBUG] ftmod_analog.c:210 Analog config var = answer_polarity_reverse 2012-02-10 06:09:01.195241 [DEBUG] ftmod_analog.c:210 Analog config var = hangup_polarity_reverse 2012-02-10 06:09:01.195250 [DEBUG] ftmod_analog.c:210 Analog config var = polarity_callerid 2012-02-10 06:09:01.195267 [DEBUG] ftmod_analog.c:210 Analog config var = polarity_delay 2012-02-10 06:09:01.195294 [DEBUG] ftmod_analog.c:210 Analog config var = callwaiting 2012-02-10 06:09:01.195312 [DEBUG] ftmod_analog.c:210 Analog config var = wait_dialtone_timeout 2012-02-10 06:09:01.195322 [DEBUG] ftmod_analog.c:229 Wait dial tone ms = 5000 2012-02-10 06:09:01.195331 [DEBUG] ftmod_analog.c:302 [s1c1][1:3] Enabled call waiting 2012-02-10 06:09:01.195340 [DEBUG] ftmod_analog.c:302 [s1c2][1:4] Enabled call waiting See option: Wait dial tone ms = 5000 But in span I set this parameter to zero! What am I doing wrong? From anthony.minessale at gmail.com Fri Feb 10 07:46:33 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 9 Feb 2012 22:46:33 -0600 Subject: [Freeswitch-users] Help with attended transfer / att_xfer In-Reply-To: <52A04E7E1B414CD0AF0D33A279B68C4F@freeswitch.org> References: <029e01cce5ef$aaadc0c0$00094240$@launch3.net> <399BEF93D8F8443BA14E1A90B8443AC0@freeswitch.org> <03d901cce6bf$bcc682d0$36538870$@launch3.net> <52A04E7E1B414CD0AF0D33A279B68C4F@freeswitch.org> Message-ID: normal sip transfers should support MOH, you just have to set hold_music to the filename of your moh or install the default config and sound sets and swap out the sounds in the moh local_stream 2012/2/8 Jo?o Mesquita > If I am reading code correctly, you should be able to insert the required > play code around line #2105 of mod_dptools.c > > I might add this patch tomorrow because I believe it is a pretty cool > feature to have with a configurable tone to be played. > > Regards, > > -- > Jo?o Mesquita > Sent with Sparrow > > On Wednesday, February 8, 2012 at 9:39 PM, Michael Collins wrote: > > I agree w/ jmesquita - this probably needs a patch, although I'd need to > dig further to see if there's a non-patchy workaround. > > As far as FusionPBX goes, you should be able to do a git pull and make > install to keep yourself updated. Just confirm with Mark Crane (IRC: > mcrane) or one of the other fusionpbx guys about any caveats with things > like the target install directory, etc. > > -MC > > On Wed, Feb 8, 2012 at 4:14 PM, Brett Wilson wrote: > > Thanks guys I am using the FS att_xfer along with the meta bind. I tried > attended transfer on my gxp2100 phones and it works but I am under the > impression that the phone is actually handling the sip switching itself for > that functionality, and our MOH was not being played while the two internal > parties were speaking, before putting through the transfer. It was just > dead silence which did not sit well with our customers. So I switched to > the FS feature, and it works great. Only problem is that we don?t know when > to start speaking after the transfer!**** > > ** ** > > Also I have another question. I have FS installed via the FusionPBX linux > disk image. What is the best way to keep FS updated? Do I need to pull from > git and build from source or what?**** > > ** ** > > *Brett Wilson* > > *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: Blogger-logo][image: > Description: Description: FaceBook-Logo][image: > Description: Description: Twitter-Logo] > **** > > ** ** > > *From:* Jo?o Mesquita [mailto:jmesquita at freeswitch.org] > *Sent:* Wednesday, February 08, 2012 5:08 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help with attended transfer / att_xfer** > ** > > ** ** > > I believe he is using the feature code, therefore att_xfer so this would > require a patch of some kind. I am this on my todo list which is quite > large nowadays and I don't even know if it is feasible yet. I can take a > look at the code later on.**** > > ** ** > > Regards,**** > > ** ** > > -- **** > > Jo?o Mesquita**** > > Sent with Sparrow **** > > ** ** > > On Wednesday, February 8, 2012 at 2:40 PM, Michael Collins wrote:**** > > Brett, > > No worries on the double-post - we have a first-time sender moderation > filter so I had to allow your messages through. You should be go to go from > here on out. > > As to your question, are you using the default configuration or are you > doing something different? Also, how are you executing the transfer - using > the transfer button on the telephone or the *1 feature code? > > Thanks, > MC**** > > On Tue, Feb 7, 2012 at 3:24 PM, Brett Wilson wrote: > > **** > > Hey guys,**** > > I need a pointer. I?m trying to get some kind of functionality where after > a attended transfer is completed, ie. The ouside caller is connected to the > 2nd phone after the 1st phone has hung up or chosen to continue the > transfer. I would like a beep to sound for the 2nd phone or for both > legs, just to let the recipient of the transfer know that it has gone > through and they can start talking.**** > > Any ideas?**** > > **** > > Thanks**** > > **** > > **** > > *Brett Wilson***** > > *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: Blogger-logo][image: > Description: Description: FaceBook-Logo][image: > Description: Description: Twitter-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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________* > *** > > Professional FreeSWITCH Consulting Services:**** > > 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**** > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users**** > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users** > ** > > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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... 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Name: not available Type: image/jpeg Size: 1715 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120209/cb33bdf8/attachment-0005.jpe From aromberg at gmail.com Fri Feb 10 08:05:57 2012 From: aromberg at gmail.com (aromberg at gmail.com) Date: Thu, 9 Feb 2012 23:05:57 -0600 Subject: [Freeswitch-users] Choppy Audio Recordings In-Reply-To: <1FFF97C269757C458224B7C895F35F1503BA1D@cantor.std.visionutv.se> References: <9D59C823-7B8A-451F-8CDD-D5F46335903E@freeswitch.org> <1FFF97C269757C458224B7C895F35F1503BA1D@cantor.std.visionutv.se> Message-ID: > I'm not 100% sure how your setup is. Are you running FS in the actual Host OS, or in a virtual (Hyper-V) machine? If running FS virtual > you will get these kinds of troubles, beacuse of not so accurate timing in those setups. > If you want things to work without problems, use real hardware. It's on a Hyper-V machine. I would have believed your timing reason/excuse if the issue would have persisted on the Hyper-V box and not on the CentOS6 machine (which is a dedicated non-virtualized machine). However, GV works near perfect (~99% with some small blips) and anything routed thru mod_sofia is choppy on the Hyper-V box, and the same is also true on the CentOS6 box. I receive perfect audio on my endpoint, it's just getting that audio written to the disk that's causing me the headache! For reference, here are my setups again: #1) Windows 2008 R2 x64 Dual Xeon 2.8, 4G of ram, Hyper-V instance, IIS running a few sites, nothing major. #2) Debian 6, dual core 2.13GHz Xeon, 512mb dedicated 1gb burst, OpenVZ instance, freeswitch + small apache instance #3) CentOS 5, 8 core 2.27 Xeon, 512MB RAM, Xen instance, freeswitch + apache (doing nothing) #4) Centos6, single core 3.33Ghz 512MB ram, dedicated machine (no instance), nohz setting enabled per the wiki, only freeswitch running Thanks, Adam From valery.kalinin at gmail.com Fri Feb 10 09:24:09 2012 From: valery.kalinin at gmail.com (Valery Kalinin) Date: Fri, 10 Feb 2012 12:24:09 +0600 Subject: [Freeswitch-users] FreeTDM: how to disable callerid? Message-ID: Hi! Work with FreeTDM (latest git) & wanpipe (3.5.24) freetdm.conf.xml: But in log: 2012-02-10 08:23:01.408284 [DEBUG] ftmod_wanpipe.c:1630 [s1c2][1:4] read wanpipe event 7 2012-02-10 08:23:01.408284 [DEBUG] ftmod_analog.c:998 [s1c2][1:4] Received event [RING_START] in state [DOWN] 2012-02-10 08:23:01.408284 [DEBUG] ftmod_analog.c:1026 [s1c2][1:4] Changed state from DOWN to GET_CALLERID 2012-02-10 08:23:01.408284 [DEBUG] ftmod_analog.c:439 [s1c2][1:4] ANALOG CHANNEL thread starting. 2012-02-10 08:23:01.408284 [DEBUG] ftdm_io.c:3133 [s1c2][1:4] Enabled software DTMF detector 2012-02-10 08:23:01.408284 [DEBUG] ftmod_analog.c:459 [s1c2][1:4] Initialized DTMF detection 2012-02-10 08:23:01.408284 [DEBUG] ftmod_analog.c:640 [s1c2][1:4] Completed state change from DOWN to GET_CALLERID 2012-02-10 08:23:01.408284 [DEBUG] ftmod_analog.c:646 [s1c2][1:4] Executing state handler on 1:2 for GET_CALLERID 2012-02-10 08:23:01.428290 [DEBUG] ftmod_wanpipe.c:965 [s1c2][1:4] First packet read stats: Rx queue len: 0, Rx 2012-02-10 08:23:02.168285 [DEBUG] ftmod_wanpipe.c:1630 [s1c2][1:4] read wanpipe event 7 All the same GET_CALLERID is called! From dujinfang at gmail.com Fri Feb 10 09:26:31 2012 From: dujinfang at gmail.com (Seven Du) Date: Fri, 10 Feb 2012 14:26:31 +0800 Subject: [Freeswitch-users] cluecon 2012 date confirm In-Reply-To: References: Message-ID: <56219F9F5CFB491C8E2472204F0822D3@gmail.com> Nice, will buy ticks ahead to take the advantage of low price. Thanks to highlight Wyndham, will cluecon always in Chicago ? ;) On Friday, February 10, 2012 at 3:16 AM, Michael Collins wrote: > Nice work. Yes, it is Aug 7-9, but Aug 6 is the Monday before. We have a new cluecon website you can check out: www-test.cluecon.com (http://www-test.cluecon.com). > > Stay tuned for updates - I promise they are coming! > > -MC > > On Wed, Feb 8, 2012 at 10:00 PM, Seven Du wrote: > > Hi, > > > > Had a wonderful time in Cluecon 2011 and would like to confirm the date as it looks like no schedule for 2012 on cluecon.com (http://cluecon.com) and I only found a count down on freeswitch.org (http://freeswitch.org). > > > > So would like to confirm will it be Aug. 6th this year? > > > > > > 179 > > Days10 > > Hours2 > > Minutes37 (http://www.cluecon.com/?fs1) > > > > > > > > select now() + interval '179 days'; > > ?column? > > ------------------------------- > > 2012-08-06 13:47:33.990137+08 > > (1 row) > > > > > > > > Thanks. > > > > -- > > About: http://about.me/dujinfang > > Blog: http://www.dujinfang.com > > Proj: http://www.freeswitch.org.cn > > > > Sent with Sparrow (http://www.sparrowmailapp.com) > > > > > > _________________________________________________________________________ > > 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 > > > > _________________________________________________________________________ > 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/20120210/11a492e3/attachment.html From luis.daniel.lucio at gmail.com Fri Feb 10 06:10:07 2012 From: luis.daniel.lucio at gmail.com (Luis Daniel Lucio Quiroz) Date: Thu, 9 Feb 2012 21:10:07 -0600 Subject: [Freeswitch-users] Server with 2 NICS In-Reply-To: References: Message-ID: Le 5 f?vrier 2012 16:33, Jo?o Mesquita a ?crit : > Look in the vars.xml and replace the $${local_ipv4} by your configured eth0 > ip address. > > FreeSWITCH resolves $${local_ipv4} to the IP of the interface who is > connected to you default gateway. > > Regards, > > -- > Jo?o Mesquita > Sent with Sparrow > > On Sunday, February 5, 2012 at 12:07 PM, Luis Daniel Lucio Quiroz wrote: > > Helo :) > > I'm new to FS but not to linux. I've sucessfully install FS into a > centos 6.2. But I have an issue. > > Server has 2 nics: > venet0: 10.9.19.5/16 > eth0: > > when starting fs, it listen SIP (5060/udp) using venet0. Phones can > register and they talk each other. However, we are using an external > GW and that fails. Doing a depth analisys I've realized that the > reason is that because venet0 is being used, SIP payload has venet0 > within but server IP is eth0's so this is tipical mitchach. > > I've sucessfully configure in vars.xml the local_ip4 (or whatever it > writes) to use public IP. but after that no phones are registering. > (before anyone says, yes phones points to public IP of course). Nmap > reports that SIP (5060/udp) is listen by eth0 but when I got phone > SIP, i only get Registering request without an answer. > > I really need to make this listen in eth0. What suggestion do you > have. Please void telling me to turn of venet0 (not possible for > politics). > > TIA > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > You are very kind, thanks From anton.jugatsu at gmail.com Fri Feb 10 08:08:26 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Fri, 10 Feb 2012 09:08:26 +0400 Subject: [Freeswitch-users] I am not able to reg with 5060 via softphone In-Reply-To: References: Message-ID: First, lunch ngrep -d eth0 -qt -W byline REGISTER or use ngrep-sip. 9 ??????? 2012 ?. 23:02 ???????????? Michael Collins ???????: > You need to find out why you can't reg on port 5060. I see that you have > an external IP address listed there. Is that the only interface on your > server? One trick you can do is to swap the internal.xml and external.xml > sip profiles. Basically just rename the files and restart freeswitch and > see if you can register on 5060. If not you'll need to supply console logs > w/ siptraces and put them on pastebin.freeswitch.org so the gang here can > take a look. > > -MC > > On Wed, Feb 8, 2012 at 4:50 AM, virendra bhati wrote: > >> >> Hi list, >> >> I am new for FreeSwitch. I was working with asterisk since 4 years. Now i >> want to learn FreeSwitch. I have installed at my server. But when I try to >> reg with server .I am not able to register with it with port 5060. but if I >> will 5080 then reg with server but can't call to local extension like 1001 >> - 1019. >> >> Please guide me what is the issue ? >> >> *fs_CLI log:-* >> >> EXECUTE sofia/external/1006 at 78.129.163.44 bridge(user/1005 at 78.129.163.44) >> 2012-02-08 12:59:37.493777 [DEBUG] switch_ivr_originate.c:1885 variable >> string 0 = [presence_id=1005 at 78.129.163.44] >> 2012-02-08 12:59:37.493777 [ERR] switch_ivr_originate.c:2430 Cannot >> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] >> 2012-02-08 12:59:37.493777 [DEBUG] switch_ivr_originate.c:3228 Originate >> Resulted in Error Cause: 606 [USER_NOT_REGISTERED] >> 2012-02-08 12:59:37.493777 [ERR] switch_ivr_originate.c:2430 Cannot >> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] >> 2012-02-08 12:59:37.493777 [DEBUG] switch_ivr_originate.c:3228 Originate >> Resulted in Error Cause: 606 [USER_NOT_REGISTERED] >> 2012-02-08 12:59:37.493777 [INFO] mod_dptools.c:2355 Originate Failed. >> Cause: USER_NOT_REGISTERED >> EXECUTE sofia/external/1006 at 78.129.163.44 answer() >> >> -- >> >> Thanks and regards >> >> Virendra Bhati >> +91-8885268942 >> Software Engineer >> E-mail-: virbhati at gmail.com >> Skype id:- virbhati2 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/7c75417e/attachment-0001.html From dujinfang at gmail.com Fri Feb 10 11:06:48 2012 From: dujinfang at gmail.com (Seven Du) Date: Fri, 10 Feb 2012 16:06:48 +0800 Subject: [Freeswitch-users] Address already in use on capture Message-ID: <078CABDA6079476FA019BE749E5032F8@gmail.com> Got Address already in use in the following log. FreeSWITCH Version 1.0.head (git-4a649ec 2012-02-01 10-13-28 +0800) Linux CentOS 5.7 I added a nua_log which printed 7060 below and 7060 is supposed to be my capture server and the error is the same even I closed the capture server. Can anyone hightlight where should I look? I can open remote ssh if necessary. btw, I have no problem when test on my Mac. Thanks. freeswitch at internal> sofia profile internal capture on Enabled sip capturing on internal nua: nua_set_params: entering nua((nil)): sent signal r_set_params nua((nil)): recv signal r_set_params nua: nua_stack_set_params: entering soa_set_params(static::0x9314488, ...) called --------------------------------7060------------------------------- capture: su_getaddrinfo(): Address already in use nua((nil)): event r_set_params 200 OK nua: nua_application_event: entering nua: nua_handle_magic: entering freeswitch at internal> freeswitch at internal> sofia profi loglevel all 0 Sofia log level for component [all] has been set to [0] diff --git a/libs/sofia-sip/libsofia-sip-ua/tport/tport_logging.c b/libs/sofia-sip/libsofia-sip-ua/tport/tport_logging.c index 200ca5c..dec2d4a 100644 --- a/libs/sofia-sip/libsofia-sip-ua/tport/tport_logging.c +++ b/libs/sofia-sip/libsofia-sip-ua/tport/tport_logging.c @@ -174,6 +174,7 @@ int tport_open_log(tport_master_t *mr, tagi_t *tags) *p = '\0'; p++; + su_log("--------------------------------%s-------------------------------\n",p); if (atoi(p) <1024 || atoi(p)>65536) { su_log("invalid port number; must be in [1024,65536]\n"); -- 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/20120210/856518c6/attachment.html From hynek.cihlar at gmail.com Fri Feb 10 11:32:25 2012 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Fri, 10 Feb 2012 09:32:25 +0100 Subject: [Freeswitch-users] INCOMING busy tone detect In-Reply-To: References: Message-ID: After playing with the detector a bit, I can confirm it already does what I need. It reports the channel where the tone originated. A related question though. Can I handle an event (or the tone detected event in particular) only in the means of the dialplan or do I need to create a script to sit and wait for the event? A declarative event processing in the dialplan would be cool! Hynek On Fri, Feb 10, 2012 at 12:38 AM, Hynek Cihlar wrote: > The spandsp mod is fantastic! Can it be configured to detect tones > originated through a specific sip profile so I would be able to tell > whether the tone was originated from the remote side? > > Hynek > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/bbfb1f3c/attachment.html From virbhati at gmail.com Fri Feb 10 12:17:37 2012 From: virbhati at gmail.com (virendra bhati) Date: Fri, 10 Feb 2012 14:47:37 +0530 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 68, Issue 77 In-Reply-To: References: Message-ID: Michael, One of my friends from USA is able to loding with server at port 5060. It means it's not server issue at all. In India me at hyderabad and another one from Delhi is not able to register with port 5060. So it's not my ISP issue. Below is the sever logs for port 5060. I have Asterisk and FreeSwitch both thnigs on it. Asterisk is closed and freeSwitch is working now. [root at IS-15502 ~]# netstat -pln | grep 5060 tcp 0 0 78.129.163.44:5060 0.0.0.0:* LISTEN 3933/freeswitch tcp 0 0 ::1:5060 :::* LISTEN 3933/freeswitch udp 0 0 78.129.163.44:5060 0.0.0.0:* 3933/freeswitch udp 0 0 ::1:5060 :::* 3933/freeswitch [root at IS-15502 ~]# iptables -L Chain INPUT (policy ACCEPT) target prot opt source destination Chain FORWARD (policy ACCEPT) target prot opt source destination Chain OUTPUT (policy ACCEPT) target prot opt source destination On Fri, Feb 10, 2012 at 12:39 AM, < 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: I am not able to reg with 5060 via softphone (Michael Collins) > 2. Re: Can someone help me with a complicated configuration? > (Michael Collins) > 3. Re: skipping vm and plays enter the pwd (Michael Collins) > > > ---------- Forwarded message ---------- > From: Michael Collins > To: FreeSWITCH Users Help > Cc: > Date: Thu, 9 Feb 2012 11:02:02 -0800 > Subject: Re: [Freeswitch-users] I am not able to reg with 5060 via > softphone > You need to find out why you can't reg on port 5060. I see that you have > an external IP address listed there. Is that the only interface on your > server? One trick you can do is to swap the internal.xml and external.xml > sip profiles. Basically just rename the files and restart freeswitch and > see if you can register on 5060. If not you'll need to supply console logs > w/ siptraces and put them on pastebin.freeswitch.org so the gang here can > take a look. > > -MC > > On Wed, Feb 8, 2012 at 4:50 AM, virendra bhati wrote: > >> >> Hi list, >> >> I am new for FreeSwitch. I was working with asterisk since 4 years. Now i >> want to learn FreeSwitch. I have installed at my server. But when I try to >> reg with server .I am not able to register with it with port 5060. but if I >> will 5080 then reg with server but can't call to local extension like 1001 >> - 1019. >> >> Please guide me what is the issue ? >> >> *fs_CLI log:-* >> >> EXECUTE sofia/external/1006 at 78.129.163.44 bridge(user/1005 at 78.129.163.44) >> 2012-02-08 12:59:37.493777 [DEBUG] switch_ivr_originate.c:1885 variable >> string 0 = [presence_id=1005 at 78.129.163.44] >> 2012-02-08 12:59:37.493777 [ERR] switch_ivr_originate.c:2430 Cannot >> create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] >> 2012-02-08 12:59:37.493777 [DEBUG] switch_ivr_originate.c:3228 Originate >> Resulted in Error Cause: 606 [USER_NOT_REGISTERED] >> 2012-02-08 12:59:37.493777 [ERR] switch_ivr_originate.c:2430 Cannot >> create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] >> 2012-02-08 12:59:37.493777 [DEBUG] switch_ivr_originate.c:3228 Originate >> Resulted in Error Cause: 606 [USER_NOT_REGISTERED] >> 2012-02-08 12:59:37.493777 [INFO] mod_dptools.c:2355 Originate Failed. >> Cause: USER_NOT_REGISTERED >> EXECUTE sofia/external/1006 at 78.129.163.44 answer() >> >> -- >> >> Thanks and regards >> >> Virendra Bhati >> +91-8885268942 >> Software Engineer >> E-mail-: virbhati at gmail.com >> Skype id:- virbhati2 >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > ---------- Forwarded message ---------- > From: Michael Collins > To: FreeSWITCH Users Help > Cc: > Date: Thu, 9 Feb 2012 11:03:18 -0800 > Subject: Re: [Freeswitch-users] Can someone help me with a complicated > configuration? > What kind of devices will connect to the public IP addresses? How will > *they* handle a colo change that results in a new IP address? > > -MC > > On Wed, Feb 8, 2012 at 9:28 AM, Josue Diaz Cruz wrote: > >> ** >> I have a server with 4 ETH. Each one have a different IP >> In 2 cases i fixed the ip address, In the other 2 cases i have 2 cable >> connections in bridge mode so i receive the public ip in dhcp mode. >> >> eth0 local address 192.168.1.100 >> eth1 local address 192.168.2.100 >> eth2 public address dhcp (cable) >> eth3 public address dhcp (cable) >> >> I was thinking to build 4 different profiles for each eth port, but i >> came to know that if we change the place of colocation it will not work >> properly because the dhcp ip. It is any way to attach a profile to the eth >> port, not the ip? >> >> Or if you know a better way to manage this case. I would like to receive >> traffic from the 2 public ip's and route it both like if i have just one >> profile. >> >> * >> >> Josue Diaz Cruz >> >> Departamento Tecnico y Soporte >> >> jdiaz at coinfru.com >> >> >> >> C/ Balsicas 3 >> >> Alquerias | 30580 | Murcia >> >> www.coinfru.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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: Michael Collins > To: FreeSWITCH Users Help > Cc: > Date: Thu, 9 Feb 2012 11:09:20 -0800 > Subject: Re: [Freeswitch-users] skipping vm and plays enter the pwd > This isn't enough information to diagnose. Please supply a log of the call > from start to finish. Also, you probably should include the dialplan > extension that is handling this process. > > -MC > > On Wed, Feb 8, 2012 at 11:49 AM, Aakash wrote: > >> Hi All, >> >> I have configured has any inbound calls enter to my inbound number play an >> ivr and transfer to user ext (eg 1080)pressed by customer,if the user dint >> pick the call enter to their voicemail automatically.But sometimes i am >> getting please enter password followed by # by skipping voicemail >> message.I >> have attached my logs below >> >> http://pastebin.freeswitch.org/18321 >> >> >> Regards, >> Aakash.V >> >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/skipping-vm-and-plays-enter-the-pwd-tp7267135p7267135.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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 +91-8885268942 Software Engineer E-mail-: virbhati at gmail.com Skype id:- virbhati2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/93a1d5dd/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 4705 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/93a1d5dd/attachment-0001.jpe From spautz2 at telefaks.biz Fri Feb 10 12:21:13 2012 From: spautz2 at telefaks.biz (David) Date: Fri, 10 Feb 2012 10:21:13 +0100 Subject: [Freeswitch-users] Freeswitch opened multiple outbound channels for one call with different uuid In-Reply-To: References: <4F323042.8080603@telefaks.biz> <4F326E84.6030302@telefaks.biz> <4F33266D.2050300@telefaks.biz> Message-ID: <4F34E189.1050008@telefaks.biz> Hi, i found the resolution for this manner. This is not a production system. It is a test-system to develop a multi tenant environment. For this, i have defined multiple domains in /etc/hosts which matched to the same host IP. The default value in vars.xml for the global variable domain=$${local_ip_v4} leads freeswitch send sip-invites for each sofia profile which is defined for the host IP. Now i have replace $${local_ip_v4} with the first domain name. Thanks all David Am 09.02.2012 20:14, schrieb Michael Collins: > I would clear out your registrations for this user and start over. > Watch to see what happens when the phone registers w/ FreeSWITCH. In > fact, you may want to update to latest git, then zap all the *.db > files except for the voicemail.db one. Is this a production system? If > so then tread carefully or do some testing on your sandbox/test system > before implementing changes on the production system. > > -MC > > On Wed, Feb 8, 2012 at 5:50 PM, David > wrote: > > I cannot find any network problems. > > interface: any > filter: (ip or ip6) and ( net 192.168.1.12 or net 192.168.1.11 ) > > # > U 192.168.1.12:2048 -> 192.168.1.4:5060 > > INVITE sip:240 at entwick1;user=phone SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-v4ybiqjk8t4e;rport. > From: "User 220" ;tag=vx1czcrfys. > To: . > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 1 INVITE. > Max-Forwards: 70. > Contact: ;reg-id=1. > P-Key-Flags: keys="3". > User-Agent: snom300/7.3.30. > Accept: application/sdp. > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, > SUBSCRIBE, PRACK, MESSAGE, INFO. > Allow-Events: talk, hold, refer, call-info. > Supported: timer, 100rel, replaces, from-change. > Session-Expires: 3600;refresher=uas. > Min-SE: 90. > Content-Type: application/sdp. > Content-Length: 218. > . > v=0. > o=root 930692018 930692018 IN IP4 192.168.1.12. > s=call. > c=IN IP4 192.168.1.12. > t=0 0. > m=audio 49908 RTP/AVP 8 101. > a=rtpmap:8 pcma/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > a=sendrecv. > > # > U 192.168.1.4:5060 -> 192.168.1.12:2048 > > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP > 192.168.1.12:2048;branch=z9hG4bK-v4ybiqjk8t4e;rport=2048. > From: "User 220" ;tag=vx1czcrfys. > To: . > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 1 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 > 14-12-22 -0600. > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.12:2048 > > SIP/2.0 407 Proxy Authentication Required. > Via: SIP/2.0/UDP > 192.168.1.12:2048;branch=z9hG4bK-v4ybiqjk8t4e;rport=2048. > From: "User 220" ;tag=vx1czcrfys. > To: ;tag=rFZa6489Z27rK. > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 1 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 > 14-12-22 -0600. > Accept: application/sdp. > 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. > Proxy-Authenticate: Digest realm="entwick1", > nonce="b755765a-a341-4070-a981-dff43e8f7d08", algorithm=MD5, > qop="auth". > Content-Length: 0. > . > > # > U 192.168.1.12:2048 -> 192.168.1.4:5060 > > ACK sip:240 at entwick1;user=phone SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-v4ybiqjk8t4e;rport. > From: "User 220" ;tag=vx1czcrfys. > To: ;tag=rFZa6489Z27rK. > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 1 ACK. > Max-Forwards: 70. > Contact: ;reg-id=1. > Content-Length: 0. > . > > # > U 192.168.1.12:2048 -> 192.168.1.4:5060 > > INVITE sip:240 at entwick1;user=phone SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-eyrl6ghugshz;rport. > From: "User 220" ;tag=vx1czcrfys. > To: . > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 2 INVITE. > Max-Forwards: 70. > Contact: ;reg-id=1. > P-Key-Flags: keys="3". > User-Agent: snom300/7.3.30. > Accept: application/sdp. > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, > SUBSCRIBE, PRACK, MESSAGE, INFO. > Allow-Events: talk, hold, refer, call-info. > Supported: timer, 100rel, replaces, from-change. > Session-Expires: 3600;refresher=uas. > Min-SE: 90. > Proxy-Authorization: Digest > username="220",realm="entwick1",nonce="b755765a-a341-4070-a981-dff43e8f7d08",uri="sip:240 at entwick1;user=phone",qop=auth,nc=00000001,cnonce="408b0a1e",response="3261384c8f62187a6486c1fee434d06a",algorithm=MD5. > Content-Type: application/sdp. > Content-Length: 218. > . > v=0. > o=root 930692018 930692018 IN IP4 192.168.1.12. > s=call. > c=IN IP4 192.168.1.12. > t=0 0. > m=audio 49908 RTP/AVP 8 101. > a=rtpmap:8 pcma/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > a=sendrecv. > > # > U 192.168.1.4:5060 -> 192.168.1.12:2048 > > SIP/2.0 100 Trying. > Via: SIP/2.0/UDP > 192.168.1.12:2048;branch=z9hG4bK-eyrl6ghugshz;rport=2048. > From: "User 220" ;tag=vx1czcrfys. > To: . > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 2 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 > 14-12-22 -0600. > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.12:2048 > > SIP/2.0 180 Ringing. > Via: SIP/2.0/UDP > 192.168.1.12:2048;branch=z9hG4bK-eyrl6ghugshz;rport=2048. > From: "User 220" ;tag=vx1czcrfys. > To: ;tag=Srr37ZSDXByBF. > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 2 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 > 14-12-22 -0600. > Accept: application/sdp. > 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-Length: 0. > Remote-Party-ID: "240" > ;party=calling;privacy=off;screen=no. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > > INVITE sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK4DN4eH402KHmF. > Max-Forwards: 69. > From: "User 220" ;tag=t1Hv9taHtmmya. > To: . > Call-ID: ad561287-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 > 14-12-22 -0600. > 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. > Session-Expires: 120;refresher=uac. > Min-SE: 120. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 233. > P-Key-Flags: keys="3". > X-FS-Support: update_display,send_info. > Remote-Party-ID: "User 220" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1328729538 1328729539 IN IP4 192.168.1.4. > s=FreeSWITCH. > c=IN IP4 192.168.1.4. > t=0 0. > m=audio 22000 RTP/AVP 8 0 98 3 18 101 > 13. > a=rtpmap:98 SPEEX/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > > INVITE sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. > Max-Forwards: 69. > From: "User 220" ;tag=UaBNBpUmQXaHp. > To: . > Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 > 14-12-22 -0600. > 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. > Session-Expires: 120;refresher=uac. > Min-SE: 120. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 233. > P-Key-Flags: keys="3". > X-FS-Support: update_display,send_info. > Remote-Party-ID: "User 220" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1328731870 1328731871 IN IP4 192.168.1.4. > s=FreeSWITCH. > c=IN IP4 192.168.1.4. > t=0 0. > m=audio 19668 RTP/AVP 8 0 98 3 18 101 > 13. > a=rtpmap:98 SPEEX/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > > INVITE sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK6Z7Nj757v5XSp. > Max-Forwards: 69. > From: "User 220" ;tag=vK4DDHcrm603H. > To: . > Call-ID: ad5660f4-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 > 14-12-22 -0600. > 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. > Session-Expires: 120;refresher=uac. > Min-SE: 120. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 233. > P-Key-Flags: keys="3". > X-FS-Support: update_display,send_info. > Remote-Party-ID: "User 220" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1328721426 1328721427 IN IP4 192.168.1.4. > s=FreeSWITCH. > c=IN IP4 192.168.1.4. > t=0 0. > m=audio 30112 RTP/AVP 8 0 98 3 18 101 > 13. > a=rtpmap:98 SPEEX/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > > INVITE sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK780em2pBtemcj. > Max-Forwards: 69. > From: "User 220" ;tag=XvX6ecXUHFQpD. > To: . > Call-ID: ad566b4a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 > 14-12-22 -0600. > 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. > Session-Expires: 120;refresher=uac. > Min-SE: 120. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 233. > P-Key-Flags: keys="3". > X-FS-Support: update_display,send_info. > Remote-Party-ID: "User 220" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1328730428 1328730429 IN IP4 192.168.1.4. > s=FreeSWITCH. > c=IN IP4 192.168.1.4. > t=0 0. > m=audio 21110 RTP/AVP 8 0 98 3 18 101 > 13. > a=rtpmap:98 SPEEX/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > > # > U 192.168.1.11:5060 -> 192.168.1.4:5060 > > SIP/2.0 180 Ringing. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK4DN4eH402KHmF. > Call-ID: ad561287-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > From: "User 220" ;tag=t1Hv9taHtmmya. > To: ;tag=iN29e6PanzlAC2JA. > Contact: . > Content-Length: 0. > . > > # > U 192.168.1.11:5060 -> 192.168.1.4:5060 > > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. > Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > From: "User 220" ;tag=UaBNBpUmQXaHp. > To: ;tag=vrAASGKzlwRxb5pm. > Contact: . > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > > ACK sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. > Max-Forwards: 69. > From: "User 220" ;tag=UaBNBpUmQXaHp. > To: ;tag=vrAASGKzlwRxb5pm. > Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 ACK. > Content-Length: 0. > . > > # > U 192.168.1.11:5060 -> 192.168.1.4:5060 > > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK6Z7Nj757v5XSp. > Call-ID: ad5660f4-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > From: "User 220" ;tag=vK4DDHcrm603H. > To: ;tag=2Urps2fOoZQSDiid. > Contact: . > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > > ACK sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK6Z7Nj757v5XSp. > Max-Forwards: 69. > From: "User 220" ;tag=vK4DDHcrm603H. > To: ;tag=2Urps2fOoZQSDiid. > Call-ID: ad5660f4-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 ACK. > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > > INVITE sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK780em2pBtemcj. > Max-Forwards: 69. > From: "User 220" ;tag=XvX6ecXUHFQpD. > To: . > Call-ID: ad566b4a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 > 14-12-22 -0600. > 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. > Session-Expires: 120;refresher=uac. > Min-SE: 120. > Content-Type: application/sdp. > Content-Disposition: session. > Content-Length: 233. > P-Key-Flags: keys="3". > X-FS-Support: update_display,send_info. > Remote-Party-ID: "User 220" > ;party=calling;screen=yes;privacy=off. > . > v=0. > o=FreeSWITCH 1328730428 1328730429 IN IP4 192.168.1.4. > s=FreeSWITCH. > c=IN IP4 192.168.1.4. > t=0 0. > m=audio 21110 RTP/AVP 8 0 98 3 18 101 > 13. > a=rtpmap:98 SPEEX/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-16. > a=ptime:20. > > # > U 192.168.1.11:5060 -> 192.168.1.4:5060 > > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK780em2pBtemcj. > Call-ID: ad566b4a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > From: "User 220" ;tag=XvX6ecXUHFQpD. > To: ;tag=qh5PsDcfdulD0LlQ. > Contact: . > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > > ACK sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK780em2pBtemcj. > Max-Forwards: 69. > From: "User 220" ;tag=XvX6ecXUHFQpD. > To: ;tag=qh5PsDcfdulD0LlQ. > Call-ID: ad566b4a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 ACK. > Content-Length: 0. > . > > # > U 192.168.1.11:5060 -> 192.168.1.4:5060 > > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. > Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > From: "User 220" ;tag=UaBNBpUmQXaHp. > To: ;tag=vrAASGKzlwRxb5pm. > Contact: . > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > > ACK sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. > Max-Forwards: 69. > From: "User 220" ;tag=UaBNBpUmQXaHp. > To: ;tag=vrAASGKzlwRxb5pm. > Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 ACK. > Content-Length: 0. > . > > # > U 192.168.1.11:5060 -> 192.168.1.4:5060 > > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. > Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > From: "User 220" ;tag=UaBNBpUmQXaHp. > To: ;tag=vrAASGKzlwRxb5pm. > Contact: . > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > > ACK sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK5peXgcN4Zv76a. > Max-Forwards: 69. > From: "User 220" ;tag=UaBNBpUmQXaHp. > To: ;tag=vrAASGKzlwRxb5pm. > Call-ID: ad56546a-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 ACK. > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > > OPTIONS sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK8Ht7NX7eQQaZD. > Max-Forwards: 70. > From: ;tag=y5pZg7DZerD9r. > To: . > Call-ID: b0fa8901-cd61-122f-8d91-001966eeb846. > CSeq: 24029141 OPTIONS. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 > 14-12-22 -0600. > 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-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.12:2048 > > OPTIONS sip:220 at 192.168.1.12:2048;line=9yhon28z SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK9tK0Qrrjm00HS. > Max-Forwards: 70. > From: ;tag=Zegrj2y2B13Um. > To: . > Call-ID: b0fa9032-cd61-122f-8d91-001966eeb846. > CSeq: 24029142 OPTIONS. > Contact: . > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 > 14-12-22 -0600. > 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-Length: 0. > . > > # > U 192.168.1.12:2048 -> 192.168.1.4:5060 > > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 192.168.1.4;rport=5060;branch=z9hG4bK9tK0Qrrjm00HS. > From: ;tag=Zegrj2y2B13Um. > To: . > Call-ID: b0fa9032-cd61-122f-8d91-001966eeb846. > CSeq: 24029142 OPTIONS. > Contact: ;reg-id=1. > User-Agent: snom300/7.3.30. > Accept-Language: en. > Accept: application/sdp. > Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, > SUBSCRIBE, PRACK, MESSAGE, INFO. > Allow-Events: talk, hold, refer, call-info. > Supported: timer, 100rel, replaces, from-change. > Content-Length: 0. > . > > # > U 192.168.1.11:5060 -> 192.168.1.4:5060 > > SIP/2.0 486 Busy Here. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK8Ht7NX7eQQaZD. > Call-ID: b0fa8901-cd61-122f-8d91-001966eeb846. > CSeq: 24029141 OPTIONS. > From: ;tag=y5pZg7DZerD9r. > To: ;tag=4rjgSs2dsNvcVyjW. > Contact: . > Allow: INVITE, ACK, OPTIONS, BYE, CANCEL, REGISTER, REFER, NOTIFY, > INFO, PRACK, UPDATE. > Accept: application/sdp. > Accept-Encoding: identity. > Accept-Language: en. > Supported: 100rel, replaces. > Content-Type: application/sdp. > Content-Length: 267. > . > v=0. > o=240 32101693 45791413 IN IP4 192.168.1.11. > s=SIP CALL. > c=IN IP4 192.168.1.11. > t=0 0. > m=audio 6000 RTP/AVP 8 0 3 18 101. > a=rtpmap:8 PCMA/8000. > a=rtpmap:0 PCMU/8000. > a=rtpmap:3 GSM/8000. > a=rtpmap:18 G729/8000. > a=rtpmap:101 telephone-event/8000. > a=fmtp:101 0-15. > > # > U 192.168.1.12:2048 -> 192.168.1.4:5060 > > CANCEL sip:240 at entwick1;user=phone SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.12:2048;branch=z9hG4bK-eyrl6ghugshz;rport. > From: "User 220" ;tag=vx1czcrfys. > To: . > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 2 CANCEL. > Max-Forwards: 70. > Reason: SIP;cause=487;text="Request terminated by user". > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.12:2048 > > SIP/2.0 200 OK. > Via: SIP/2.0/UDP > 192.168.1.12:2048;branch=z9hG4bK-eyrl6ghugshz;rport=2048. > From: "User 220" ;tag=vx1czcrfys. > To: ;tag=Srr37ZSDXByBF. > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 2 CANCEL. > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.12:2048 > > SIP/2.0 487 Request Terminated. > Via: SIP/2.0/UDP > 192.168.1.12:2048;branch=z9hG4bK-eyrl6ghugshz;rport=2048. > From: "User 220" ;tag=vx1czcrfys. > To: ;tag=Srr37ZSDXByBF. > Call-ID: 3c28bad818ad-q0ms60zsfpst. > CSeq: 2 INVITE. > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-d2c9fb5 2012-02-06 > 14-12-22 -0600. > 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-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > > CANCEL sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK4DN4eH402KHmF. > Max-Forwards: 69. > From: "User 220" ;tag=t1Hv9taHtmmya. > To: . > Call-ID: ad561287-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 CANCEL. > Reason: FreeSWITCH;cause=487;text="ORIGINATOR_CANCEL". > Content-Length: 0. > . > > # > U 192.168.1.11:5060 -> 192.168.1.4:5060 > > SIP/2.0 200 OK. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK4DN4eH402KHmF. > Call-ID: ad561287-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 CANCEL. > From: "User 220" ;tag=t1Hv9taHtmmya. > To: ;tag=iN29e6PanzlAC2JA. > Contact: . > Content-Length: 0. > . > > # > U 192.168.1.11:5060 -> 192.168.1.4:5060 > > SIP/2.0 487 Request Terminated. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK4DN4eH402KHmF. > Call-ID: ad561287-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 INVITE. > From: "User 220" ;tag=t1Hv9taHtmmya. > To: ;tag=iN29e6PanzlAC2JA. > Contact: . > Content-Length: 0. > . > > # > U 192.168.1.4:5060 -> 192.168.1.11:5060 > > ACK sip:240 at 192.168.1.11:5060 SIP/2.0. > Via: SIP/2.0/UDP 192.168.1.4;rport;branch=z9hG4bK4DN4eH402KHmF. > Max-Forwards: 69. > From: "User 220" ;tag=t1Hv9taHtmmya. > To: ;tag=iN29e6PanzlAC2JA. > Call-ID: ad561287-cd61-122f-8d91-001966eeb846. > CSeq: 24039705 ACK. > Content-Length: 0. > . > > # > > Am 08.02.2012 18:36, schrieb Michael Collins: >> >> It was found that freeswitch performs multiple INVITES. But why? >> >> >> One reason why multiple invites are sent out is if FreeSWITCH >> doesn't receive anything back. This could mean the far end is >> down for some reason or that there is a network issue between FS >> and the other endpoint. NAT is a common issue in this sort of >> configuration. I recommend that you look at SIP traces on each >> end of the SIP dialog. >> >> -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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/57ea8edd/attachment-0001.html From dujinfang at gmail.com Fri Feb 10 14:27:01 2012 From: dujinfang at gmail.com (Seven Du) Date: Fri, 10 Feb 2012 19:27:01 +0800 Subject: [Freeswitch-users] Address already in use on capture In-Reply-To: <078CABDA6079476FA019BE749E5032F8@gmail.com> References: <078CABDA6079476FA019BE749E5032F8@gmail.com> Message-ID: Update: I made a patch which fixed a string problem http://jira.freeswitch.org/browse/FS-3896 but I still have a problem discribed on jira. Thanks. On Friday, February 10, 2012 at 4:06 PM, Seven Du wrote: > Got Address already in use in the following log. > > FreeSWITCH Version 1.0.head (git-4a649ec 2012-02-01 10-13-28 +0800) Linux CentOS 5.7 > > I added a nua_log which printed 7060 below and 7060 is supposed to be my capture server and the error is the same even I closed the capture server. > > Can anyone hightlight where should I look? I can open remote ssh if necessary. > > btw, I have no problem when test on my Mac. > > Thanks. > > > freeswitch at internal> sofia profile internal capture on > Enabled sip capturing on internal > nua: nua_set_params: entering > nua((nil)): sent signal r_set_params > nua((nil)): recv signal r_set_params > nua: nua_stack_set_params: entering > soa_set_params(static::0x9314488, ...) called > --------------------------------7060------------------------------- > capture: su_getaddrinfo(): Address already in use > nua((nil)): event r_set_params 200 OK > nua: nua_application_event: entering > nua: nua_handle_magic: entering > freeswitch at internal> > freeswitch at internal> sofia profi loglevel all 0 > Sofia log level for component [all] has been set to [0] > > > diff --git a/libs/sofia-sip/libsofia-sip-ua/tport/tport_logging.c b/libs/sofia-sip/libsofia-sip-ua/tport/tport_logging.c > index 200ca5c..dec2d4a 100644 > --- a/libs/sofia-sip/libsofia-sip-ua/tport/tport_logging.c > +++ b/libs/sofia-sip/libsofia-sip-ua/tport/tport_logging.c > @@ -174,6 +174,7 @@ int tport_open_log(tport_master_t *mr, tagi_t *tags) > *p = '\0'; > p++; > > + su_log("--------------------------------%s-------------------------------\n",p); > if (atoi(p) <1024 || atoi(p)>65536) > { > su_log("invalid port number; must be in [1024,65536]\n"); > > > > > > -- > 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/20120210/7903eb9f/attachment.html From brett at launch3.net Fri Feb 10 17:55:53 2012 From: brett at launch3.net (Brett Wilson) Date: Fri, 10 Feb 2012 09:55:53 -0500 Subject: [Freeswitch-users] Help with attended transfer / att_xfer In-Reply-To: References: <029e01cce5ef$aaadc0c0$00094240$@launch3.net> <399BEF93D8F8443BA14E1A90B8443AC0@freeswitch.org> <03d901cce6bf$bcc682d0$36538870$@launch3.net> <52A04E7E1B414CD0AF0D33A279B68C4F@freeswitch.org> Message-ID: <067b01cce804$16741000$435c3000$@launch3.net> I have the MOH setup correctly I think. When I do a blind transfer through the phone, and the other extension is ringing and not picked up yet, MOH does work. Only the phone?s attended transfer exhibits the strange behavior. I?ll have to get around to doing a packet cap one day. Brett Wilson 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: Blogger-logo Description: Description: FaceBook-Logo Description: Description: Twitter-Logo From: Anthony Minessale [mailto:anthony.minessale at gmail.com] Sent: Thursday, February 09, 2012 11:47 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help with attended transfer / att_xfer normal sip transfers should support MOH, you just have to set hold_music to the filename of your moh or install the default config and sound sets and swap out the sounds in the moh local_stream 2012/2/8 Jo?o Mesquita If I am reading code correctly, you should be able to insert the required play code around line #2105 of mod_dptools.c I might add this patch tomorrow because I believe it is a pretty cool feature to have with a configurable tone to be played. Regards, -- Jo?o Mesquita Sent with Sparrow On Wednesday, February 8, 2012 at 9:39 PM, Michael Collins wrote: I agree w/ jmesquita - this probably needs a patch, although I'd need to dig further to see if there's a non-patchy workaround. As far as FusionPBX goes, you should be able to do a git pull and make install to keep yourself updated. Just confirm with Mark Crane (IRC: mcrane) or one of the other fusionpbx guys about any caveats with things like the target install directory, etc. -MC On Wed, Feb 8, 2012 at 4:14 PM, Brett Wilson wrote: Thanks guys I am using the FS att_xfer along with the meta bind. I tried attended transfer on my gxp2100 phones and it works but I am under the impression that the phone is actually handling the sip switching itself for that functionality, and our MOH was not being played while the two internal parties were speaking, before putting through the transfer. It was just dead silence which did not sit well with our customers. So I switched to the FS feature, and it works great. Only problem is that we don?t know when to start speaking after the transfer! Also I have another question. I have FS installed via the FusionPBX linux disk image. What is the best way to keep FS updated? Do I need to pull from git and build from source or what? Brett Wilson 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: Blogger-logo Description: Description: FaceBook-Logo Description: Description: Twitter-Logo From: Jo?o Mesquita [mailto:jmesquita at freeswitch.org] Sent: Wednesday, February 08, 2012 5:08 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Help with attended transfer / att_xfer I believe he is using the feature code, therefore att_xfer so this would require a patch of some kind. I am this on my todo list which is quite large nowadays and I don't even know if it is feasible yet. I can take a look at the code later on. Regards, -- Jo?o Mesquita Sent with Sparrow On Wednesday, February 8, 2012 at 2:40 PM, Michael Collins wrote: Brett, No worries on the double-post - we have a first-time sender moderation filter so I had to allow your messages through. You should be go to go from here on out. As to your question, are you using the default configuration or are you doing something different? Also, how are you executing the transfer - using the transfer button on the telephone or the *1 feature code? Thanks, MC On Tue, Feb 7, 2012 at 3:24 PM, Brett Wilson wrote: Hey guys, I need a pointer. I?m trying to get some kind of functionality where after a attended transfer is completed, ie. The ouside caller is connected to the 2nd phone after the 1st phone has hung up or chosen to continue the transfer. I would like a beep to sound for the 2nd phone or for both legs, just to let the recipient of the transfer know that it has gone through and they can start talking. Any ideas? Thanks Brett Wilson 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: Blogger-logo Description: Description: FaceBook-Logo Description: Description: Twitter-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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://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/20120210/f9dab4f9/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 1815 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/f9dab4f9/attachment-0003.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 1680 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/f9dab4f9/attachment-0004.jpe -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 1715 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/f9dab4f9/attachment-0005.jpe From holger at freyther.de Fri Feb 10 12:41:43 2012 From: holger at freyther.de (Holger Freyther) Date: Fri, 10 Feb 2012 09:41:43 +0000 (UTC) Subject: [Freeswitch-users] How to inspect the jitterbuffer? References: Message-ID: Anthony Minessale writes: > > what does /etc/wanpipe/wanpipe1.conf look like? and which rev of FS > and wanpipe driver are you on. > What ptime is the sip traffic? Hi, the wanpipe1.conf is here[1], I am using wanpipe 3.5.24, I started with a FreeSWITCH from around 20111010, but I retested with the a version from 20120205. What I have done so far to try to understand things: fs_cli> originate freetdm/1/nr &echo(1000) (audio quality is good in both ways, so FreeTDM by itself appears to work) On a 'normal' SIP call (the path to ISDN produces the skips): I used the 'bug' application to record the audio file and it had no skips. Wireshark's RTP Analysis shows no packet loss, little jitter, just a bit of clock drifting. To me it looks like if FreeTDM doesn't get the audio samples in time and then 'nothing' is transmitted. This is why I asked about inspecting the state of the jitterbuffer. regards holger [1] http://paste.lisp.org/display/127633 From shcherbatyuk at belrosbank.by Fri Feb 10 14:53:00 2012 From: shcherbatyuk at belrosbank.by (Eugene Shcherbatyuk) Date: Fri, 10 Feb 2012 14:53:00 +0300 Subject: [Freeswitch-users] FreeSwitch does not start in the background Message-ID: <4F35051C.7050801@belrosbank.by> Hello, The purpose was to evaluate FreeSwitch and decide whether to replace my Asterisks with it or not. I installed FS1.0.6 from git on Ubuntu Server 10.04.3 LTS. Everything went smoothly until FS did not start with the system today. There are no errors in logs. FS starts in console fine. FS does not start in background mode nor from console (with -nc option) nor from init script. Even worse: FS can unexpectedly start (rarely) or do not start (often) with "backgrounding" message. Have a look at the console output below, please. > root at fs:/usr/local/freeswitch/bin# /etc/init.d/freeswitch start > # ******* FS started without a message ********* > root at fs:/usr/local/freeswitch/bin# netstat -ant > Active Internet connections (servers and established) > Proto Recv-Q Send-Q Local Address Foreign Address State > tcp 0 0 172.18.0.130:5060 0.0.0.0:* LISTEN > tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN > tcp 0 0 0.0.0.0:22 0.0.0.0:* LISTEN > tcp 0 0 172.18.0.130:5080 0.0.0.0:* LISTEN > tcp6 0 0 ::1:5060 :::* LISTEN > tcp6 0 0 :::22 :::* LISTEN > root at fs:/usr/local/freeswitch/bin# /etc/init.d/freeswitch stop > Killing: 850 > root at fs:/usr/local/freeswitch/bin# /etc/init.d/freeswitch start > root at fs:/usr/local/freeswitch/bin# 923 Backgrounding. > # ****** Note message above ********* > root at fs:/usr/local/freeswitch/bin# netstat -ant > Active Internet connections (servers and established) > Proto Recv-Q Send-Q Local Address Foreign Address State > tcp 0 0 0.0.0.0:22 0.0.0.0:* LISTEN > tcp6 0 0 :::22 :::* LISTEN Your opinions, please? Thank you in advance, Eugene ========================================================= ?????? ????????? ? ????? ???????? (???????????) ???????? ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? ?????????????. ========================================================= 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. ========================================================= From msc at freeswitch.org Fri Feb 10 19:35:28 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 10 Feb 2012 08:35:28 -0800 Subject: [Freeswitch-users] INCOMING busy tone detect In-Reply-To: References: Message-ID: Off the top of my head I don't recall if the tone_detect app uses mod_spandsp or not. I would check mod_dptools.c and confirm. In any case, if you are detecting tones and wish to react accordingly then look at this app for how to handle it in the dialplan. -MC On Fri, Feb 10, 2012 at 12:32 AM, Hynek Cihlar wrote: > After playing with the detector a bit, I can confirm it already does what > I need. It reports the channel where the tone originated. > > A related question though. Can I handle an event (or the tone detected > event in particular) only in the means of the dialplan or do I need to > create a script to sit and wait for the event? A declarative event > processing in the dialplan would be cool! > > Hynek > > > > > On Fri, Feb 10, 2012 at 12:38 AM, Hynek Cihlar wrote: > >> The spandsp mod is fantastic! Can it be configured to detect tones >> originated through a specific sip profile so I would be able to tell >> whether the tone was originated from the remote side? >> >> Hynek >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/70349728/attachment.html From msc at freeswitch.org Fri Feb 10 19:40:14 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 10 Feb 2012 08:40:14 -0800 Subject: [Freeswitch-users] FreeTDM: how to disable callerid? In-Reply-To: References: Message-ID: You are trying to disable caller ID on an inbound call? May I ask why? -MC On Thu, Feb 9, 2012 at 10:24 PM, Valery Kalinin wrote: > Hi! > Work with FreeTDM (latest git) & wanpipe (3.5.24) > > freetdm.conf.xml: > > > > > > > > > > But in log: > 2012-02-10 08:23:01.408284 [DEBUG] ftmod_wanpipe.c:1630 [s1c2][1:4] > read wanpipe event 7 > 2012-02-10 08:23:01.408284 [DEBUG] ftmod_analog.c:998 [s1c2][1:4] > Received event [RING_START] in state [DOWN] > 2012-02-10 08:23:01.408284 [DEBUG] ftmod_analog.c:1026 [s1c2][1:4] > Changed state from DOWN to GET_CALLERID > 2012-02-10 08:23:01.408284 [DEBUG] ftmod_analog.c:439 [s1c2][1:4] > ANALOG CHANNEL thread starting. > 2012-02-10 08:23:01.408284 [DEBUG] ftdm_io.c:3133 [s1c2][1:4] Enabled > software DTMF detector > 2012-02-10 08:23:01.408284 [DEBUG] ftmod_analog.c:459 [s1c2][1:4] > Initialized DTMF detection > 2012-02-10 08:23:01.408284 [DEBUG] ftmod_analog.c:640 [s1c2][1:4] > Completed state change from DOWN to GET_CALLERID > 2012-02-10 08:23:01.408284 [DEBUG] ftmod_analog.c:646 [s1c2][1:4] > Executing state handler on 1:2 for GET_CALLERID > 2012-02-10 08:23:01.428290 [DEBUG] ftmod_wanpipe.c:965 [s1c2][1:4] > First packet read stats: Rx queue len: 0, Rx > 2012-02-10 08:23:02.168285 [DEBUG] ftmod_wanpipe.c:1630 [s1c2][1:4] > read wanpipe event 7 > > All the same GET_CALLERID is called! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/4b6943be/attachment.html From aakashviswam at gmail.com Fri Feb 10 20:02:27 2012 From: aakashviswam at gmail.com (Aakash) Date: Fri, 10 Feb 2012 09:02:27 -0800 (PST) Subject: [Freeswitch-users] skipping vm and plays enter the pwd In-Reply-To: References: <1328730591696-7267135.post@n2.nabble.com> Message-ID: <1328893347310-7273378.post@n2.nabble.com> Hello, I have pastebin the required files for the mentioned call flow for a better picture. It goes from Public.xml(Inbound)-->Default.xml(Inbound)-->IVR.xml(Get the digits)-->Default.xml(Local extension). In this flow if I send the digits as 1080 and if 1080 is not available, instead of redirecting to voicemail box it's asking for some password. All other extensions are working fine, the problem is only with 1080. Please assist. Public.xml http://pastebin.freeswitch.org/18355 Default.xml http://pastebin.freeswitch.org/18356 demoivr.xml http://pastebin.freeswitch.org/18357 directory/1080.xml http://pastebin.freeswitch.org/18358 Freeswitch log http://pastebin.freeswitch.org/18354 Thanks, Aakash -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/skipping-vm-and-plays-enter-the-pwd-tp7267135p7273378.html Sent from the freeswitch-users mailing list archive at Nabble.com. From curriegrad2004 at gmail.com Fri Feb 10 20:20:37 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Fri, 10 Feb 2012 09:20:37 -0800 Subject: [Freeswitch-users] FreeTDM: how to disable callerid? In-Reply-To: References: Message-ID: Are you trying to disable Caller ID on an outbound call? If so, you can always set the effective caller id to something that's all zeros or what not. On Fri, Feb 10, 2012 at 8:40 AM, Michael Collins wrote: > You are trying to disable caller ID on an inbound call? May I ask why? > -MC > > > On Thu, Feb 9, 2012 at 10:24 PM, Valery Kalinin > wrote: >> >> Hi! >> Work with FreeTDM (latest git) & wanpipe (3.5.24) >> >> freetdm.conf.xml: >> >> ? ? ? ? >> ? ? ? ? ? ? ? ? >> ? ? ? ? ? ? ? ? >> ? ? ? ? ? ? ? ? >> ? ? ? ? ? ? ? ? >> ? ? ? ? >> >> >> But in log: >> 2012-02-10 08:23:01.408284 [DEBUG] ftmod_wanpipe.c:1630 [s1c2][1:4] >> read wanpipe event 7 >> 2012-02-10 08:23:01.408284 [DEBUG] ftmod_analog.c:998 [s1c2][1:4] >> Received event [RING_START] in state [DOWN] >> 2012-02-10 08:23:01.408284 [DEBUG] ftmod_analog.c:1026 [s1c2][1:4] >> Changed state from DOWN to GET_CALLERID >> 2012-02-10 08:23:01.408284 [DEBUG] ftmod_analog.c:439 [s1c2][1:4] >> ANALOG CHANNEL thread starting. >> 2012-02-10 08:23:01.408284 [DEBUG] ftdm_io.c:3133 [s1c2][1:4] Enabled >> software DTMF detector >> 2012-02-10 08:23:01.408284 [DEBUG] ftmod_analog.c:459 [s1c2][1:4] >> Initialized DTMF detection >> 2012-02-10 08:23:01.408284 [DEBUG] ftmod_analog.c:640 [s1c2][1:4] >> Completed state change from DOWN to GET_CALLERID >> 2012-02-10 08:23:01.408284 [DEBUG] ftmod_analog.c:646 [s1c2][1:4] >> Executing state handler on 1:2 for GET_CALLERID >> 2012-02-10 08:23:01.428290 [DEBUG] ftmod_wanpipe.c:965 [s1c2][1:4] >> First packet read stats: Rx queue len: 0, Rx >> 2012-02-10 08:23:02.168285 [DEBUG] ftmod_wanpipe.c:1630 [s1c2][1:4] >> read wanpipe event 7 >> >> All the same GET_CALLERID is called! >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > 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 Feb 10 20:23:55 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 10 Feb 2012 11:23:55 -0600 Subject: [Freeswitch-users] How to inspect the jitterbuffer? In-Reply-To: References: Message-ID: If there is no RTP then the TDM should be generating its own audio. Try MTU = 80 IDLE_FLAG = 0xFF On Fri, Feb 10, 2012 at 3:41 AM, Holger Freyther wrote: > Anthony Minessale writes: > >> >> what does /etc/wanpipe/wanpipe1.conf look like? and which rev of FS >> and wanpipe driver are you on. >> What ptime is the sip traffic? > > Hi, > > the wanpipe1.conf is here[1], I am using wanpipe 3.5.24, I started > with a FreeSWITCH from around 20111010, but I retested with the > a version from 20120205. > > What I have done so far to try to understand things: > fs_cli> originate freetdm/1/nr &echo(1000) > (audio quality is good in both ways, so FreeTDM by itself > appears to work) > > On a 'normal' SIP call (the path to ISDN produces the skips): > I used the 'bug' application to record the audio file and it had > no skips. Wireshark's RTP Analysis shows no packet loss, little > jitter, just a bit of clock drifting. > > To me it looks like if FreeTDM doesn't get the audio samples in > time and then 'nothing' is transmitted. This is why I asked about > inspecting the state of the jitterbuffer. > > > regards > ?holger > > > > > > > [1] http://paste.lisp.org/display/127633 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 Feb 10 20:26:45 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 10 Feb 2012 09:26:45 -0800 Subject: [Freeswitch-users] skipping vm and plays enter the pwd In-Reply-To: <1328893347310-7273378.post@n2.nabble.com> References: <1328730591696-7267135.post@n2.nabble.com> <1328893347310-7273378.post@n2.nabble.com> Message-ID: You still did not put the entire call in here. Capture the entire call from the moment it hits the public dialplan until it is completely done. -MC On Fri, Feb 10, 2012 at 9:02 AM, Aakash wrote: > Hello, > > I have pastebin the required files for the mentioned call flow for a better > picture. It goes from > Public.xml(Inbound)-->Default.xml(Inbound)-->IVR.xml(Get the > digits)-->Default.xml(Local extension). In this flow if I send the digits > as > 1080 and if 1080 is not available, instead of redirecting to voicemail box > it's asking for some password. All other extensions are working fine, the > problem is only with 1080. > > Please assist. > > Public.xml > http://pastebin.freeswitch.org/18355 > > Default.xml > http://pastebin.freeswitch.org/18356 > > demoivr.xml > http://pastebin.freeswitch.org/18357 > > directory/1080.xml > http://pastebin.freeswitch.org/18358 > > Freeswitch log > http://pastebin.freeswitch.org/18354 > > > Thanks, > Aakash > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/skipping-vm-and-plays-enter-the-pwd-tp7267135p7273378.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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/99bd98e3/attachment.html From anton.jugatsu at gmail.com Fri Feb 10 20:38:41 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Fri, 10 Feb 2012 21:38:41 +0400 Subject: [Freeswitch-users] FreeSwitch does not start in the background In-Reply-To: <4F35051C.7050801@belrosbank.by> References: <4F35051C.7050801@belrosbank.by> Message-ID: The first suggestion would be to use latest git and init script from http://wiki.freeswitch.org/wiki/Freeswitch_init. 10 ??????? 2012 ?. 15:53 ???????????? Eugene Shcherbatyuk < shcherbatyuk at belrosbank.by> ???????: > Hello, > > The purpose was to evaluate FreeSwitch and decide whether to replace my > Asterisks with it or not. I installed FS1.0.6 from git on Ubuntu Server > 10.04.3 LTS. Everything went smoothly until FS did not start with the > system today. > > There are no errors in logs. FS starts in console fine. FS does not > start in background mode nor from console (with -nc option) nor from > init script. Even worse: FS can unexpectedly start (rarely) or do not > start (often) with "backgrounding" message. Have a look at the console > output below, please. > > > root at fs:/usr/local/freeswitch/bin# /etc/init.d/freeswitch start > > # ******* FS started without a message ********* > > root at fs:/usr/local/freeswitch/bin# netstat -ant > > Active Internet connections (servers and established) > > Proto Recv-Q Send-Q Local Address Foreign Address State > > tcp 0 0 172.18.0.130:5060 0.0.0.0:* LISTEN > > tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN > > tcp 0 0 0.0.0.0:22 0.0.0.0:* LISTEN > > tcp 0 0 172.18.0.130:5080 0.0.0.0:* LISTEN > > tcp6 0 0 ::1:5060 :::* LISTEN > > tcp6 0 0 :::22 :::* LISTEN > > root at fs:/usr/local/freeswitch/bin# /etc/init.d/freeswitch stop > > Killing: 850 > > root at fs:/usr/local/freeswitch/bin# /etc/init.d/freeswitch start > > root at fs:/usr/local/freeswitch/bin# 923 Backgrounding. > > # ****** Note message above ********* > > root at fs:/usr/local/freeswitch/bin# netstat -ant > > Active Internet connections (servers and established) > > Proto Recv-Q Send-Q Local Address Foreign Address State > > tcp 0 0 0.0.0.0:22 0.0.0.0:* LISTEN > > tcp6 0 0 :::22 :::* LISTEN > > Your opinions, please? > > Thank you in advance, > Eugene > > > > ========================================================= > > ?????? ????????? ? ????? ???????? (<>) ???????? > ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? > ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? > ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. > ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? > "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? > ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? > ?????????????. > > ========================================================= > > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/ae2c7be0/attachment.html From anthony.minessale at gmail.com Fri Feb 10 20:43:29 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 10 Feb 2012 11:43:29 -0600 Subject: [Freeswitch-users] Choppy Audio Recordings In-Reply-To: References: <9D59C823-7B8A-451F-8CDD-D5F46335903E@freeswitch.org> <1FFF97C269757C458224B7C895F35F1503BA1D@cantor.std.visionutv.se> Message-ID: Are you on very latest FS code? We did a large refactoring of recording code recently and it was just complete last week sometime. On Thu, Feb 9, 2012 at 11:05 PM, wrote: >> I'm not 100% sure how your setup is. Are you running FS in the actual Host OS, or in a virtual (Hyper-V) machine? If running FS virtual > you will get these kinds of troubles, beacuse of not so accurate timing in those setups. >> If you want things to work without problems, use real hardware. > > It's on a Hyper-V machine. ?I would have believed your timing > reason/excuse if the issue would have persisted on the Hyper-V box and > not on the CentOS6 machine (which is a dedicated non-virtualized > machine). ?However, GV works near perfect (~99% with some small blips) > and anything routed thru mod_sofia is choppy on the Hyper-V box, and > the same is also true on the CentOS6 box. > > I receive perfect audio on my endpoint, it's just getting that audio > written to the disk that's causing me the headache! > > For reference, here are my setups again: > > #1) Windows 2008 R2 x64 Dual Xeon 2.8, 4G of ram, Hyper-V instance, IIS > running a few sites, nothing major. > #2) Debian 6, dual core 2.13GHz Xeon, 512mb dedicated 1gb burst, OpenVZ > instance, freeswitch + small apache instance > #3) CentOS 5, 8 core 2.27 Xeon, 512MB RAM, Xen instance, freeswitch + > apache (doing nothing) > #4) Centos6, single core 3.33Ghz 512MB ram, dedicated machine (no > instance), nohz setting enabled per the wiki, only freeswitch running > > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 Fri Feb 10 20:48:32 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 10 Feb 2012 11:48:32 -0600 Subject: [Freeswitch-users] cluecon 2012 date confirm In-Reply-To: <56219F9F5CFB491C8E2472204F0822D3@gmail.com> References: <56219F9F5CFB491C8E2472204F0822D3@gmail.com> Message-ID: Yes we like tradition so for the foreseeable future, ClueCon is always in Chicago. On Fri, Feb 10, 2012 at 12:26 AM, Seven Du wrote: > Nice, will buy ticks ahead to take the advantage of low price. > > Thanks to highlight?Wyndham, ?will cluecon always in Chicago ? ?;) > > On Friday, February 10, 2012 at 3:16 AM, Michael Collins wrote: > > Nice work. Yes, it is Aug 7-9, but Aug 6 is the Monday before. We have a new > cluecon website you can check out: www-test.cluecon.com. > > Stay tuned for updates - I promise they are coming! > > -MC > > On Wed, Feb 8, 2012 at 10:00 PM, Seven Du wrote: > > Hi, > > Had a wonderful time in Cluecon 2011 and would like to confirm the date as > it looks like no schedule for 2012 on cluecon.com and I only found a count > down on freeswitch.org. > > So would like to confirm will it be Aug. 6th this year? > > > 179 > Days10 > Hours2 > Minutes37 > > > > select now() + interval '179 days'; > ? ? ? ? ? ??column? > ------------------------------- > ?2012-08-06 13:47:33.990137+08 > (1 row) > > > Thanks. > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: ?http://www.freeswitch.org.cn > > Sent with Sparrow > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 aksrini at hotmail.com Fri Feb 10 21:21:29 2012 From: aksrini at hotmail.com (Srini K) Date: Fri, 10 Feb 2012 10:21:29 -0800 Subject: [Freeswitch-users] FreeSWITCH ends the calls after receiving DTMF from the Callee. In-Reply-To: References: , , , , , Message-ID: Anthony,Thanks for your response.I have not defined any of these input_callback or dtmf_callback. I have attached the log with debug level.http://pastebin.freeswitch.org/18360 Thanks in advance. RegardsSrini > Date: Thu, 9 Feb 2012 19:06:39 -0600 > From: anthony.minessale at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after receiving DTMF from the Callee. > > do you have input_callback or dtmf_callback or anything defined? > > On Thu, Feb 9, 2012 at 6:37 PM, Srini K wrote: > > Iam using the latest FreeSWITCh version (updated today morning). > > > > FS terminates on receiving any DTMF digits from the callee. I have not set > > bridge_terminate_key. > > > > I have created an oubound session and bridging the inbound and oubound > > session. Code snippet is > > > > var session = new ManagedSession("{origination_caller_id_number=" + > > callerIdNumber + ",originate_timeout=8" + "}sofia/gateway/408xxxxyyyy"); > > string outBoundUuid = session.GetVariable("uuid"); > > if (string.IsNullOrEmpty(outBoundUuid)) > > { // Log error; > > return; > > } > > freeswitch.bridge(inboundSession, session); > > > > Also I have tried setting hangup_after_bridge=false. > > > > Regards > > Srini > > > > > > > >> Date: Thu, 9 Feb 2012 18:17:45 -0600 > > > >> From: anthony.minessale at gmail.com > >> To: freeswitch-users at lists.freeswitch.org > >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after receiving > >> DTMF from the Callee. > >> > >> the obvious reason would be setting bridge_terminate_key=1 > >> but I don't know where your bridge is being started based on your > >> description or what version of the code you are using. > >> > >> On Thu, Feb 9, 2012 at 6:06 PM, Srini K wrote: > >> > Any reason that it didn't return success. Anything Iam missing it out. > >> > > >> > > >> >> Date: Thu, 9 Feb 2012 17:33:01 -0600 > >> >> From: anthony.minessale at gmail.com > >> >> To: freeswitch-users at lists.freeswitch.org > >> >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after > >> >> receiving > >> >> DTMF from the Callee. > >> > > >> >> > >> >> bridge with input_callback and did not return success. > >> >> > >> >> On Thu, Feb 9, 2012 at 5:28 PM, Srini K wrote: > >> >> > Hi, > >> >> > I have configured FreeSWITCH to receive DTMF by > >> >> > > >> >> > > >> >> > > >> >> > > >> >> > in both sip_profile internal and external files. > >> >> > > >> >> > Iam using mod managed and I have subscribed to receive DTMF events. > >> >> > I can process DTMF from Caller without any problem. When I receive > >> >> > DTMF > >> >> > from > >> >> > the callee as Sip Info, I do receive the DTMF event from the > >> >> > FreeSWITCH > >> >> > and > >> >> > immediately FreeSWITCH disconnects the call. > >> >> > > >> >> > Snapshot of the log is... > >> >> > > >> >> > [DEBUG] sofia.c:7229 INFO DTMF(1) > >> >> > [DEBUG] sofia.c:7307 dispatched freeswitch event for INFO > >> >> > [DEBUG] switch_core_session.c:875 Send signal > >> >> > sofia/external/yyyyyyyyyy > >> >> > [BREAK] > >> >> > [DEBUG] switch_ivr_bridge.c:383 sofia/external/yyyyyyyyyy ended call > >> >> > via > >> >> > DTMF > >> >> > [DEBUG] switch_ivr_bridge.c:384 Send signal > >> >> > sofia/external/xxxxxxxxxx at 10.10.2.3 [BREAK] > >> >> > [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE > >> >> > [sofia/external/yyyyyyyyyy] > >> >> > > >> >> > Whether Iam missing anything in the config? > >> >> > > >> >> > Thanks in advance. > >> >> > 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 > >> >> > > >> >> > FreeSWITCH-users mailing list > >> >> > FreeSWITCH-users at lists.freeswitch.org > >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > > >> >> > UNSUBSCRIBE:http://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 > >> >> > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> > > >> > > >> > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > 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 > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://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 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > 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 > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/c3afff1a/attachment-0001.html From anthony.minessale at gmail.com Fri Feb 10 21:32:03 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 10 Feb 2012 12:32:03 -0600 Subject: [Freeswitch-users] FreeSWITCH ends the calls after receiving DTMF from the Callee. In-Reply-To: References: Message-ID: I tried this lua script which should be the same thing: b_leg = freeswitch.Session("sofia/internal/foo at bar.com", session); freeswitch.bridge(session, b_leg); See if you get the same problem running this lua code, I was passing dtmf fine. Internally, the code should not even use a callback capable of ending the call as you described unless your program has an input callback of dtmf handler function. Also make sure you are on a recent build of FS. On Fri, Feb 10, 2012 at 12:21 PM, Srini K wrote: > Anthony, > Thanks for your response. > I have not defined any of these input_callback or dtmf_callback. > > I have attached the log with debug level. > http://pastebin.freeswitch.org/18360 > > Thanks in advance. > > Regards > Srini > > > >> Date: Thu, 9 Feb 2012 19:06:39 -0600 > >> From: anthony.minessale at gmail.com >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after receiving >> DTMF from the Callee. >> >> do you have input_callback or dtmf_callback or anything defined? >> >> On Thu, Feb 9, 2012 at 6:37 PM, Srini K wrote: >> > Iam using the latest FreeSWITCh version (updated today morning). >> > >> > FS terminates on receiving any DTMF digits from the callee. I have not >> > set >> > bridge_terminate_key. >> > >> > I have created an oubound session and bridging the inbound and oubound > >> > session. Code snippet is >> > >> > var session = new ManagedSession("{origination_caller_id_number=" + >> > callerIdNumber + ",originate_timeout=8" + "}sofia/gateway/408xxxxyyyy"); >> > string outBoundUuid = session.GetVariable("uuid"); >> > if (string.IsNullOrEmpty(outBoundUuid)) >> > { // Log error; >> > ??? return; >> > } >> > freeswitch.bridge(inboundSession, session); >> > >> > Also I have tried setting hangup_after_bridge=false. >> > >> > Regards >> > Srini >> > >> > >> > >> >> Date: Thu, 9 Feb 2012 18:17:45 -0600 >> > >> >> From: anthony.minessale at gmail.com >> >> To: freeswitch-users at lists.freeswitch.org >> >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after >> >> receiving >> >> DTMF from the Callee. >> >> >> >> the obvious reason would be setting bridge_terminate_key=1 >> >> but I don't know where your bridge is being started based on your >> >> description or what version of the code you are using. >> >> >> >> On Thu, Feb 9, 2012 at 6:06 PM, Srini K wrote: >> >> > Any reason that it didn't return success. Anything Iam missing it >> >> > out. >> >> > >> >> > >> >> >> Date: Thu, 9 Feb 2012 17:33:01 -0600 >> >> >> From: anthony.minessale at gmail.com >> >> >> To: freeswitch-users at lists.freeswitch.org >> >> >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after >> >> >> receiving >> >> >> DTMF from the Callee. >> >> > >> >> >> >> >> >> bridge with input_callback and did not return success. >> >> >> >> >> >> On Thu, Feb 9, 2012 at 5:28 PM, Srini K wrote: >> >> >> > Hi, >> >> >> > I have configured FreeSWITCH to receive DTMF by >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > in both sip_profile internal and external files. >> >> >> > >> >> >> > Iam using mod managed and I have subscribed to receive DTMF >> >> >> > events. >> >> >> > I can process DTMF from Caller without any problem. When I receive >> >> >> > DTMF >> >> >> > from >> >> >> > the callee as Sip Info, I do receive the DTMF event from the >> >> >> > FreeSWITCH >> >> >> > and >> >> >> > immediately FreeSWITCH disconnects the call. >> >> >> > >> >> >> > Snapshot of the log is... >> >> >> > >> >> >> > [DEBUG] sofia.c:7229 INFO DTMF(1) >> >> >> > [DEBUG] sofia.c:7307 dispatched freeswitch event for INFO >> >> >> > [DEBUG] switch_core_session.c:875 Send signal >> >> >> > sofia/external/yyyyyyyyyy >> >> >> > [BREAK] >> >> >> > [DEBUG] switch_ivr_bridge.c:383 sofia/external/yyyyyyyyyy ended >> >> >> > call >> >> >> > via >> >> >> > DTMF >> >> >> > [DEBUG] switch_ivr_bridge.c:384 Send signal >> >> >> > sofia/external/xxxxxxxxxx at 10.10.2.3 [BREAK] >> >> >> > [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE >> >> >> > [sofia/external/yyyyyyyyyy] >> >> >> > >> >> >> > Whether Iam missing anything in the config? >> >> >> > >> >> >> > Thanks in advance. >> >> >> > 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 >> >> >> > >> >> >> > FreeSWITCH-users mailing list >> >> >> > FreeSWITCH-users at lists.freeswitch.org >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> > >> >> >> > >> >> >> > UNSUBSCRIBE:http://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 >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ >> >> > Professional FreeSWITCH Consulting Services: >> >> > 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 >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > >> >> > UNSUBSCRIBE:http://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 >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > 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 >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 lloyd.aloysius at gmail.com Fri Feb 10 21:46:17 2012 From: lloyd.aloysius at gmail.com (Lloyd Aloysius) Date: Fri, 10 Feb 2012 13:46:17 -0500 Subject: [Freeswitch-users] Direct Call Pickup - Issue Message-ID: Hi , FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500) I have a direct call pickup issue / Bug? 1. Extension 201 On the phone 2. Second call Comes to Extension 201 3. Extension 202 Dial **201 4. Extension 201 Lost the Active call and Extension 202 Pickup the Extension 201 Active Call. But Extension 202 should pickup the second call not active call is this a bug? I am using Default Dial plan application Thanks in advanced Lloyd -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/0e1b131a/attachment.html From cjbujold at accra.ca Fri Feb 10 21:48:26 2012 From: cjbujold at accra.ca (Charles Bujold) Date: Fri, 10 Feb 2012 14:48:26 -0400 Subject: [Freeswitch-users] Can't call out more than once- long distance calls Message-ID: <013701cce824$92e79920$b8b6cb60$@accra.ca> The first long distance call going out, works. If I try to make a second call immediately after I get call "Forbidden", I presume it is an incorrect setting but do not know where to start. Here is a log of the call being rejected. The error 403 says the server gets the call, but will not process it, if I interpret the error message properly. Question is what can say to Freeswitch not to process the call? Thanks for the help cb 2012-02-10 14:35:29.082815 [DEBUG] sofia.c:7538 IP 192.168.20.80 Rejected by acl "domains". Falling back to Digest auth. 2012-02-10 14:35:29.082815 [NOTICE] switch_channel.c:930 New Channel sofia/internal/250 at 192.168.250.20 [36c7a641-bb72-403b-99f4-a2783a8898b8] 2012-02-10 14:35:29.082815 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/250 at 192.168.250.20) Running State Change CS_NEW 2012-02-10 14:35:29.082815 [DEBUG] switch_core_state_machine.c:380 (sofia/internal/250 at 192.168.250.20) State NEW 2012-02-10 14:35:29.082815 [DEBUG] sofia.c:5508 Channel sofia/internal/250 at 192.168.250.20 entering state [received][100] 2012-02-10 14:35:29.082815 [DEBUG] sofia.c:5519 Remote SDP: v=0 o=- 20015 20015 IN IP4 192.168.20.80 s=SDP data c=IN IP4 192.168.20.80 t=0 0 m=audio 11796 RTP/AVP 0 8 18 4 9 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:4 G723/8000 a=rtpmap:9 G722/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 2012-02-10 14:35:29.082815 [DEBUG] sofia_glue.c:4798 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2012-02-10 14:35:29.082815 [DEBUG] sofia_glue.c:2919 Set Codec sofia/internal/250 at 192.168.250.20 PCMU/8000 20 ms 160 samples 64000 bits 2012-02-10 14:35:29.082815 [DEBUG] switch_core_codec.c:111 sofia/internal/250 at 192.168.250.20 Original read codec set to PCMU:0 2012-02-10 14:35:29.082815 [DEBUG] sofia_glue.c:4919 Set 2833 dtmf send/recv payload to 101 2012-02-10 14:35:29.082815 [DEBUG] sofia.c:5731 (sofia/internal/250 at 192.168.250.20) State Change CS_NEW -> CS_INIT 2012-02-10 14:35:29.082815 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/250 at 192.168.250.20 [BREAK] 2012-02-10 14:35:29.082815 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/250 at 192.168.250.20) Running State Change CS_INIT 2012-02-10 14:35:29.082815 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/250 at 192.168.250.20) State INIT 2012-02-10 14:35:29.082815 [DEBUG] mod_sofia.c:85 sofia/internal/250 at 192.168.250.20 SOFIA INIT 2012-02-10 14:35:29.082815 [DEBUG] mod_sofia.c:125 (sofia/internal/250 at 192.168.250.20) State Change CS_INIT -> CS_ROUTING 2012-02-10 14:35:29.082815 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/250 at 192.168.250.20 [BREAK] 2012-02-10 14:35:29.082815 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/250 at 192.168.250.20) State INIT going to sleep 2012-02-10 14:35:29.082815 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/250 at 192.168.250.20) Running State Change CS_ROUTING 2012-02-10 14:35:29.082815 [DEBUG] switch_channel.c:1890 (sofia/internal/250 at 192.168.250.20) Callstate Change DOWN -> RINGING 2012-02-10 14:35:29.102799 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/250 at 192.168.250.20) State ROUTING 2012-02-10 14:35:29.102799 [DEBUG] mod_sofia.c:148 sofia/internal/250 at 192.168.250.20 SOFIA ROUTING 2012-02-10 14:35:29.102799 [DEBUG] switch_core_state_machine.c:104 sofia/internal/250 at 192.168.250.20 Standard ROUTING 2012-02-10 14:35:29.102799 [INFO] mod_dialplan_xml.c:481 Processing User1 <250>->18xxxxxxxxx in context default Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->unloop] continue=false Dialplan: sofia/internal/250 at 192.168.250.20 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->call_direction] continue=true Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [call_direction] ${call_direction}() =~ /^(inbound|outbound|local)$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 ANTI-Action set(call_direction=local) Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->tod_example] continue=true Dialplan: sofia/internal/250 at 192.168.250.20 Date/Time Match (PASS) [tod_example] break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 Action set(open=true) Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->holiday_example] continue=true Dialplan: sofia/internal/250 at 192.168.250.20 Date/TimeMatch (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->group-intercept] continue=false Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [group-intercept] destination_number(18xxxxxxxxx) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->redial] continue=false Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [redial] destination_number(18xxxxxxxxx) =~ /^(redial|\*870)$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->global] continue=true Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=never Dialplan: sofia/internal/250 at 192.168.250.20 Absolute Condition [global] Dialplan: sofia/internal/250 at 192.168.250.20 Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/250 at 192.168.250.20 Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_numbe r}) Dialplan: sofia/internal/250 at 192.168.250.20 Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/250 at 192.168.250.20 Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [snom-demo-2] destination_number(18xxxxxxxxx) =~ /^\*9001$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [snom-demo-1] destination_number(18xxxxxxxxx) =~ /^\*9000$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->call_privacy] continue=false Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [call_privacy] destination_number(18xxxxxxxxx) =~ /^\*67(\d+)$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->call_return] continue=false Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [call_return] destination_number(18xxxxxxxxx) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->del-group] continue=false Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [del-group] destination_number(18xxxxxxxxx) =~ /^\*\*80(\d{2})$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->add-group] continue=false Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [add-group] destination_number(18xxxxxxxxx) =~ /^\*\*81(\d{2})$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [call-group-simo] destination_number(18xxxxxxxxx) =~ /^\*\*82(\d{2})$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->call-group-order] continue=false Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [call-group-order] destination_number(18xxxxxxxxx) =~ /^\*83(\d{2})$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [intercept-ext] destination_number(18xxxxxxxxx) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->Local_Extension_Skinny] continue=false Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [Local_Extension_Skinny] destination_number(18xxxxxxxxx) =~ /^(11[01][0-9])$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->send_to_voicemail] continue=false Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [send_to_voicemail] destination_number(18xxxxxxxxx) =~ /^\*99(\d{2,7})$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->VoiceMeUp.911] continue=false Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [VoiceMeUp.911] destination_number(18xxxxxxxxx) =~ /^911$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->VoiceMeUp.7d] continue=false Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [VoiceMeUp.7d] destination_number(18xxxxxxxxx) =~ /^(\d{7})$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->VoiceMeUp.tollfree] continue=false Dialplan: sofia/internal/250 at 192.168.250.20 Regex (FAIL) [VoiceMeUp.tollfree] destination_number(18xxxxxxxxx) =~ /^1?(8(00|55|66|77|88)[2-9]\d{6})$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 parsing [default->VoiceMeUp.11d] continue=false Dialplan: sofia/internal/250 at 192.168.250.20 Regex (PASS) [VoiceMeUp.11d] destination_number(18xxxxxxxxx) =~ /^\+?(\d{11})$/ break=on-false Dialplan: sofia/internal/250 at 192.168.250.20 Action set(sip_h_X-accountcode=${accountcode}) Dialplan: sofia/internal/250 at 192.168.250.20 Action set(call_direction=outbound) Dialplan: sofia/internal/250 at 192.168.250.20 Action set(hangup_after_bridge=true) Dialplan: sofia/internal/250 at 192.168.250.20 Action set(effective_caller_id_name=${outbound_caller_id_name}) Dialplan: sofia/internal/250 at 192.168.250.20 Action set(effective_caller_id_number=${outbound_caller_id_number}) Dialplan: sofia/internal/250 at 192.168.250.20 Action set(inherit_codec=true) Dialplan: sofia/internal/250 at 192.168.250.20 Action bridge(sofia/gateway/VoiceMeUp/18xxxxxxxxx) 2012-02-10 14:35:29.102799 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/250 at 192.168.250.20) State Change CS_ROUTING -> CS_EXECUTE 2012-02-10 14:35:29.102799 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/250 at 192.168.250.20 [BREAK] 2012-02-10 14:35:29.102799 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/250 at 192.168.250.20) State ROUTING going to sleep 2012-02-10 14:35:29.102799 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/250 at 192.168.250.20) Running State Change CS_EXECUTE 2012-02-10 14:35:29.102799 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/250 at 192.168.250.20) State EXECUTE 2012-02-10 14:35:29.102799 [DEBUG] mod_sofia.c:241 sofia/internal/250 at 192.168.250.20 SOFIA EXECUTE 2012-02-10 14:35:29.102799 [DEBUG] switch_core_state_machine.c:192 sofia/internal/250 at 192.168.250.20 Standard EXECUTE EXECUTE sofia/internal/250 at 192.168.250.20 set(call_direction=local) 2012-02-10 14:35:29.102799 [DEBUG] mod_dptools.c:1281 sofia/internal/250 at 192.168.250.20 SET [call_direction]=[local] EXECUTE sofia/internal/250 at 192.168.250.20 set(open=true) 2012-02-10 14:35:29.102799 [DEBUG] mod_dptools.c:1281 sofia/internal/250 at 192.168.250.20 SET [open]=[true] EXECUTE sofia/internal/250 at 192.168.250.20 hash(insert/192.168.250.20-spymap/250/36c7a641-bb72-403b-99f4-a2783a8898b8) EXECUTE sofia/internal/250 at 192.168.250.20 hash(insert/192.168.250.20-last_dial/250/18xxxxxxxxx) EXECUTE sofia/internal/250 at 192.168.250.20 hash(insert/192.168.250.20-last_dial/global/36c7a641-bb72-403b-99f4-a2783a88 98b8) EXECUTE sofia/internal/250 at 192.168.250.20 set(RFC2822_DATE=Fri, 10 Feb 2012 14:35:29 -0400) 2012-02-10 14:35:29.102799 [DEBUG] mod_dptools.c:1281 sofia/internal/250 at 192.168.250.20 SET [RFC2822_DATE]=[Fri, 10 Feb 2012 14:35:29 -0400] EXECUTE sofia/internal/250 at 192.168.250.20 set(sip_h_X-accountcode=1000) 2012-02-10 14:35:29.102799 [DEBUG] mod_dptools.c:1281 sofia/internal/250 at 192.168.250.20 SET [sip_h_X-accountcode]=[1000] EXECUTE sofia/internal/250 at 192.168.250.20 set(call_direction=outbound) 2012-02-10 14:35:29.102799 [DEBUG] mod_dptools.c:1281 sofia/internal/250 at 192.168.250.20 SET [call_direction]=[outbound] EXECUTE sofia/internal/250 at 192.168.250.20 set(hangup_after_bridge=true) 2012-02-10 14:35:29.102799 [DEBUG] mod_dptools.c:1281 sofia/internal/250 at 192.168.250.20 SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/250 at 192.168.250.20 set(effective_caller_id_name=Tel1) 2012-02-10 14:35:29.102799 [DEBUG] mod_dptools.c:1281 sofia/internal/250 at 192.168.250.20 SET [effective_caller_id_name]=[Tel1] EXECUTE sofia/internal/250 at 192.168.250.20 set(effective_caller_id_number=5xxxxxxxxx) 2012-02-10 14:35:29.102799 [DEBUG] mod_dptools.c:1281 sofia/internal/250 at 192.168.250.20 SET [effective_caller_id_number]=[5xxxxxxxxx] EXECUTE sofia/internal/250 at 192.168.250.20 set(inherit_codec=true) 2012-02-10 14:35:29.102799 [DEBUG] mod_dptools.c:1281 sofia/internal/250 at 192.168.250.20 SET [inherit_codec]=[true] EXECUTE sofia/internal/250 at 192.168.250.20 bridge(sofia/gateway/VoiceMeUp/18xxxxxxxxx) 2012-02-10 14:35:29.102799 [DEBUG] switch_channel.c:1051 sofia/internal/250 at 192.168.250.20 EXPORTING[export_vars] [domain_name]=[192.168.250.20] to event 2012-02-10 14:35:29.102799 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2012-02-10 14:35:29.102799 [NOTICE] switch_channel.c:930 New Channel sofia/external/18xxxxxxxxx [5abef3bf-1aa0-4616-89c5-4e134df9328d] 2012-02-10 14:35:29.102799 [DEBUG] mod_sofia.c:4670 (sofia/external/18xxxxxxxxx) State Change CS_NEW -> CS_INIT 2012-02-10 14:35:29.102799 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/18xxxxxxxxx [BREAK] 2012-02-10 14:35:29.102799 [DEBUG] switch_core_state_machine.c:362 (sofia/external/18xxxxxxxxx) Running State Change CS_INIT 2012-02-10 14:35:29.102799 [DEBUG] switch_core_state_machine.c:401 (sofia/external/18xxxxxxxxx) State INIT 2012-02-10 14:35:29.102799 [DEBUG] mod_sofia.c:85 sofia/external/18xxxxxxxxx SOFIA INIT 2012-02-10 14:35:29.102799 [DEBUG] mod_sofia.c:125 (sofia/external/18xxxxxxxxx) State Change CS_INIT -> CS_ROUTING 2012-02-10 14:35:29.102799 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/18xxxxxxxxx [BREAK] 2012-02-10 14:35:29.102799 [DEBUG] switch_core_state_machine.c:401 (sofia/external/18xxxxxxxxx) State INIT going to sleep 2012-02-10 14:35:29.122800 [DEBUG] switch_core_state_machine.c:362 (sofia/external/18xxxxxxxxx) Running State Change CS_ROUTING 2012-02-10 14:35:29.122800 [DEBUG] switch_channel.c:1890 (sofia/external/18xxxxxxxxx) Callstate Change DOWN -> RINGING 2012-02-10 14:35:29.122800 [DEBUG] switch_core_state_machine.c:410 (sofia/external/18xxxxxxxxx) State ROUTING 2012-02-10 14:35:29.122800 [DEBUG] mod_sofia.c:148 sofia/external/18xxxxxxxxx SOFIA ROUTING 2012-02-10 14:35:29.122800 [DEBUG] switch_ivr_originate.c:66 (sofia/external/18xxxxxxxxx) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2012-02-10 14:35:29.122800 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/18xxxxxxxxx [BREAK] 2012-02-10 14:35:29.122800 [DEBUG] switch_core_state_machine.c:410 (sofia/external/18xxxxxxxxx) State ROUTING going to sleep 2012-02-10 14:35:29.122800 [DEBUG] switch_core_state_machine.c:362 (sofia/external/18xxxxxxxxx) Running State Change CS_CONSUME_MEDIA 2012-02-10 14:35:29.122800 [DEBUG] switch_core_state_machine.c:429 (sofia/external/18xxxxxxxxx) State CONSUME_MEDIA 2012-02-10 14:35:29.122800 [DEBUG] switch_core_state_machine.c:429 (sofia/external/18xxxxxxxxx) State CONSUME_MEDIA going to sleep 2012-02-10 14:35:29.122800 [DEBUG] switch_core_session.c:875 Send signal sofia/external/18xxxxxxxxx [BREAK] 2012-02-10 14:35:29.122800 [DEBUG] sofia.c:5508 Channel sofia/external/18xxxxxxxxx entering state [calling][0] 2012-02-10 14:35:29.162820 [DEBUG] switch_core_session.c:875 Send signal sofia/external/18xxxxxxxxx [BREAK] 2012-02-10 14:35:29.162820 [DEBUG] switch_core_session.c:875 Send signal sofia/external/18xxxxxxxxx [BREAK] 2012-02-10 14:35:29.162820 [DEBUG] sofia.c:5508 Channel sofia/external/18xxxxxxxxx entering state [calling][0] 2012-02-10 14:35:29.222819 [DEBUG] switch_core_session.c:875 Send signal sofia/external/18xxxxxxxxx [BREAK] 2012-02-10 14:35:29.222819 [DEBUG] switch_core_session.c:875 Send signal sofia/external/18xxxxxxxxx [BREAK] 2012-02-10 14:35:29.222819 [DEBUG] switch_core_session.c:875 Send signal sofia/external/18xxxxxxxxx [BREAK] 2012-02-10 14:35:29.222819 [DEBUG] sofia.c:5508 Channel sofia/external/18xxxxxxxxx entering state [terminated][403] 2012-02-10 14:35:29.222819 [DEBUG] switch_channel.c:2852 (sofia/external/18xxxxxxxxx) Callstate Change RINGING -> HANGUP 2012-02-10 14:35:29.222819 [NOTICE] sofia.c:6272 Hangup sofia/external/18xxxxxxxxx [CS_CONSUME_MEDIA] [CALL_REJECTED] 2012-02-10 14:35:29.222819 [DEBUG] switch_channel.c:2875 Send signal sofia/external/18xxxxxxxxx [KILL] 2012-02-10 14:35:29.222819 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/18xxxxxxxxx [BREAK] 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:362 (sofia/external/18xxxxxxxxx) Running State Change CS_HANGUP 2012-02-10 14:35:29.222819 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 21 [CALL_REJECTED] 2012-02-10 14:35:29.222819 [INFO] mod_dptools.c:2916 Originate Failed. Cause: CALL_REJECTED 2012-02-10 14:35:29.222819 [DEBUG] switch_channel.c:2852 (sofia/internal/250 at 192.168.250.20) Callstate Change RINGING -> HANGUP 2012-02-10 14:35:29.222819 [NOTICE] mod_dptools.c:3035 Hangup sofia/internal/250 at 192.168.250.20 [CS_EXECUTE] [CALL_REJECTED] 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:602 (sofia/external/18xxxxxxxxx) State HANGUP 2012-02-10 14:35:29.222819 [DEBUG] mod_sofia.c:469 Channel sofia/external/18xxxxxxxxx hanging up, cause: CALL_REJECTED 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:47 sofia/external/18xxxxxxxxx Standard HANGUP, cause: CALL_REJECTED 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:602 (sofia/external/18xxxxxxxxx) State HANGUP going to sleep 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:393 (sofia/external/18xxxxxxxxx) State Change CS_HANGUP -> CS_REPORTING 2012-02-10 14:35:29.222819 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/18xxxxxxxxx [BREAK] 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:362 (sofia/external/18xxxxxxxxx) Running State Change CS_REPORTING 2012-02-10 14:35:29.222819 [DEBUG] switch_channel.c:2875 Send signal sofia/internal/250 at 192.168.250.20 [KILL] 2012-02-10 14:35:29.222819 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/250 at 192.168.250.20 [BREAK] 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:662 (sofia/external/18xxxxxxxxx) State REPORTING 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:79 sofia/external/18xxxxxxxxx Standard REPORTING, cause: CALL_REJECTED 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:662 (sofia/external/18xxxxxxxxx) State REPORTING going to sleep 2012-02-10 14:35:29.222819 [DEBUG] switch_core_session.c:2285 sofia/internal/250 at 192.168.250.20 skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/250 at 192.168.250.20) State EXECUTE going to sleep 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/250 at 192.168.250.20) Running State Change CS_HANGUP 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:387 (sofia/external/18xxxxxxxxx) State Change CS_REPORTING -> CS_DESTROY 2012-02-10 14:35:29.222819 [DEBUG] switch_core_session.c:1180 Send signal sofia/external/18xxxxxxxxx [BREAK] 2012-02-10 14:35:29.222819 [DEBUG] switch_core_session.c:1380 Session 7 (sofia/external/18xxxxxxxxx) Locked, Waiting on external entities 2012-02-10 14:35:29.222819 [NOTICE] switch_core_session.c:1398 Session 7 (sofia/external/18xxxxxxxxx) Ended 2012-02-10 14:35:29.222819 [NOTICE] switch_core_session.c:1400 Close Channel sofia/external/18xxxxxxxxx [CS_DESTROY] 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:491 (sofia/external/18xxxxxxxxx) Callstate Change HANGUP -> DOWN 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:494 (sofia/external/18xxxxxxxxx) Running State Change CS_DESTROY 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:504 (sofia/external/18xxxxxxxxx) State DESTROY 2012-02-10 14:35:29.222819 [DEBUG] mod_sofia.c:374 sofia/external/18xxxxxxxxx SOFIA DESTROY 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/250 at 192.168.250.20) State HANGUP 2012-02-10 14:35:29.222819 [DEBUG] mod_sofia.c:463 sofia/internal/250 at 192.168.250.20 Overriding SIP cause 603 with 403 from the other leg 2012-02-10 14:35:29.222819 [DEBUG] mod_sofia.c:469 Channel sofia/internal/250 at 192.168.250.20 hanging up, cause: CALL_REJECTED 2012-02-10 14:35:29.222819 [DEBUG] mod_sofia.c:534 Responding to INVITE with: 403 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:47 sofia/internal/250 at 192.168.250.20 Standard HANGUP, cause: CALL_REJECTED 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/250 at 192.168.250.20) State HANGUP going to sleep 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/250 at 192.168.250.20) State Change CS_HANGUP -> CS_REPORTING 2012-02-10 14:35:29.222819 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/250 at 192.168.250.20 [BREAK] 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/250 at 192.168.250.20) Running State Change CS_REPORTING 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/250 at 192.168.250.20) State REPORTING 2012-02-10 14:35:29.222819 [DEBUG] switch_nat.c:545 unmapped public port 26608 protocol UDP to localport 26608 2012-02-10 14:35:29.222819 [DEBUG] switch_nat.c:545 unmapped public port 16966 protocol UDP to localport 16966 2012-02-10 14:35:29.222819 [DEBUG] switch_nat.c:545 unmapped public port 16967 protocol UDP to localport 16967 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:86 sofia/external/18xxxxxxxxx Standard DESTROY 2012-02-10 14:35:29.222819 [DEBUG] switch_core_state_machine.c:504 (sofia/external/18xxxxxxxxx) State DESTROY going to sleep 2012-02-10 14:35:29.422846 [DEBUG] switch_core_state_machine.c:79 sofia/internal/250 at 192.168.250.20 Standard REPORTING, cause: CALL_REJECTED 2012-02-10 14:35:29.422846 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/250 at 192.168.250.20) State REPORTING going to sleep 2012-02-10 14:35:29.422846 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/250 at 192.168.250.20) State Change CS_REPORTING -> CS_DESTROY 2012-02-10 14:35:29.422846 [DEBUG] switch_core_session.c:1180 Send signal sofia/internal/250 at 192.168.250.20 [BREAK] 2012-02-10 14:35:29.422846 [DEBUG] switch_core_session.c:1380 Session 6 (sofia/internal/250 at 192.168.250.20) Locked, Waiting on external entities 2012-02-10 14:35:29.422846 [NOTICE] switch_core_session.c:1398 Session 6 (sofia/internal/250 at 192.168.250.20) Ended 2012-02-10 14:35:29.422846 [NOTICE] switch_core_session.c:1400 Close Channel sofia/internal/250 at 192.168.250.20 [CS_DESTROY] 2012-02-10 14:35:29.422846 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/250 at 192.168.250.20) Callstate Change HANGUP -> DOWN 2012-02-10 14:35:29.422846 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/250 at 192.168.250.20) Running State Change CS_DESTROY 2012-02-10 14:35:29.422846 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/250 at 192.168.250.20) State DESTROY 2012-02-10 14:35:29.422846 [DEBUG] mod_sofia.c:374 sofia/internal/250 at 192.168.250.20 SOFIA DESTROY 2012-02-10 14:35:29.422846 [DEBUG] switch_core_state_machine.c:86 sofia/internal/250 at 192.168.250.20 Standard DESTROY 2012-02-10 14:35:29.422846 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/250 at 192.168.250.20) State DESTROY going to sleep -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/1ff1fba6/attachment-0001.html From brian at freeswitch.org Fri Feb 10 22:11:22 2012 From: brian at freeswitch.org (Brian West) Date: Fri, 10 Feb 2012 13:11:22 -0600 Subject: [Freeswitch-users] Direct Call Pickup - Issue In-Reply-To: References: Message-ID: <040950C4-60EC-47AD-9230-D1EF1F4A759B@freeswitch.org> Please use the latest git head. /b On Feb 10, 2012, at 12:46 PM, Lloyd Aloysius wrote: > FreeSWITCH Version 1.0.head (git-0675b59 2011-06-06 21-28-14 -0500) -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/bf02be86/attachment.html From brian at freeswitch.org Fri Feb 10 22:11:53 2012 From: brian at freeswitch.org (Brian West) Date: Fri, 10 Feb 2012 13:11:53 -0600 Subject: [Freeswitch-users] Can't call out more than once- long distance calls In-Reply-To: <013701cce824$92e79920$b8b6cb60$@accra.ca> References: <013701cce824$92e79920$b8b6cb60$@accra.ca> Message-ID: I would ask your provider why they aren't allowing you to make more than once call I suspect they are the ones rejecting it. Sip traces would be helpful too. /b On Feb 10, 2012, at 12:48 PM, Charles Bujold wrote: > > > The first long distance call going out, works. If I try to make a second > call immediately after I get call "Forbidden", I presume it is an incorrect > setting but do not know where to start. Here is a log of the call being > rejected. The error 403 says the server gets the call, but will not > process it, if I interpret the error message properly. Question is what > can say to Freeswitch not to process the call? > > > > Thanks for the help > > > > cb > -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/ed19951b/attachment.html From aksrini at hotmail.com Fri Feb 10 23:13:19 2012 From: aksrini at hotmail.com (Srini K) Date: Fri, 10 Feb 2012 12:13:19 -0800 Subject: [Freeswitch-users] FreeSWITCH ends the calls after receiving DTMF from the Callee. In-Reply-To: References: , , , , , , , Message-ID: The Lua script works fine. No problem in capturing the DTMF from both the call legs. Only in Mod_Managed I have issue. Its simple 3 lines of code nothing fancy... public void Run(AppContext context) { context.Session.Answer(); var newSession = new ManagedSession("sofia/gateway/foo/callenumber"); freeswitch.bridge(newSession, context.Session); } FreeSWITCH version is FreeSWITCH Version 1.0.head (git-f477404 2012-02-09 11-08-52 -0600) ThanksSrini> Date: Fri, 10 Feb 2012 12:32:03 -0600 > From: anthony.minessale at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after receiving DTMF from the Callee. > > I tried this lua script which should be the same thing: > > b_leg = freeswitch.Session("sofia/internal/foo at bar.com", session); > freeswitch.bridge(session, b_leg); > > See if you get the same problem running this lua code, I was passing dtmf fine. > > > Internally, the code should not even use a callback capable of ending > the call as you described unless your program has an input callback of > dtmf handler function. > > > Also make sure you are on a recent build of FS. > > > > On Fri, Feb 10, 2012 at 12:21 PM, Srini K wrote: > > Anthony, > > Thanks for your response. > > I have not defined any of these input_callback or dtmf_callback. > > > > I have attached the log with debug level. > > http://pastebin.freeswitch.org/18360 > > > > Thanks in advance. > > > > Regards > > Srini > > > > > > > >> Date: Thu, 9 Feb 2012 19:06:39 -0600 > > > >> From: anthony.minessale at gmail.com > >> To: freeswitch-users at lists.freeswitch.org > >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after receiving > >> DTMF from the Callee. > >> > >> do you have input_callback or dtmf_callback or anything defined? > >> > >> On Thu, Feb 9, 2012 at 6:37 PM, Srini K wrote: > >> > Iam using the latest FreeSWITCh version (updated today morning). > >> > > >> > FS terminates on receiving any DTMF digits from the callee. I have not > >> > set > >> > bridge_terminate_key. > >> > > >> > I have created an oubound session and bridging the inbound and oubound > > > >> > session. Code snippet is > >> > > >> > var session = new ManagedSession("{origination_caller_id_number=" + > >> > callerIdNumber + ",originate_timeout=8" + "}sofia/gateway/408xxxxyyyy"); > >> > string outBoundUuid = session.GetVariable("uuid"); > >> > if (string.IsNullOrEmpty(outBoundUuid)) > >> > { // Log error; > >> > return; > >> > } > >> > freeswitch.bridge(inboundSession, session); > >> > > >> > Also I have tried setting hangup_after_bridge=false. > >> > > >> > Regards > >> > Srini > >> > > >> > > >> > > >> >> Date: Thu, 9 Feb 2012 18:17:45 -0600 > >> > > >> >> From: anthony.minessale at gmail.com > >> >> To: freeswitch-users at lists.freeswitch.org > >> >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after > >> >> receiving > >> >> DTMF from the Callee. > >> >> > >> >> the obvious reason would be setting bridge_terminate_key=1 > >> >> but I don't know where your bridge is being started based on your > >> >> description or what version of the code you are using. > >> >> > >> >> On Thu, Feb 9, 2012 at 6:06 PM, Srini K wrote: > >> >> > Any reason that it didn't return success. Anything Iam missing it > >> >> > out. > >> >> > > >> >> > > >> >> >> Date: Thu, 9 Feb 2012 17:33:01 -0600 > >> >> >> From: anthony.minessale at gmail.com > >> >> >> To: freeswitch-users at lists.freeswitch.org > >> >> >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after > >> >> >> receiving > >> >> >> DTMF from the Callee. > >> >> > > >> >> >> > >> >> >> bridge with input_callback and did not return success. > >> >> >> > >> >> >> On Thu, Feb 9, 2012 at 5:28 PM, Srini K wrote: > >> >> >> > Hi, > >> >> >> > I have configured FreeSWITCH to receive DTMF by > >> >> >> > > >> >> >> > > >> >> >> > > >> >> >> > > >> >> >> > in both sip_profile internal and external files. > >> >> >> > > >> >> >> > Iam using mod managed and I have subscribed to receive DTMF > >> >> >> > events. > >> >> >> > I can process DTMF from Caller without any problem. When I receive > >> >> >> > DTMF > >> >> >> > from > >> >> >> > the callee as Sip Info, I do receive the DTMF event from the > >> >> >> > FreeSWITCH > >> >> >> > and > >> >> >> > immediately FreeSWITCH disconnects the call. > >> >> >> > > >> >> >> > Snapshot of the log is... > >> >> >> > > >> >> >> > [DEBUG] sofia.c:7229 INFO DTMF(1) > >> >> >> > [DEBUG] sofia.c:7307 dispatched freeswitch event for INFO > >> >> >> > [DEBUG] switch_core_session.c:875 Send signal > >> >> >> > sofia/external/yyyyyyyyyy > >> >> >> > [BREAK] > >> >> >> > [DEBUG] switch_ivr_bridge.c:383 sofia/external/yyyyyyyyyy ended > >> >> >> > call > >> >> >> > via > >> >> >> > DTMF > >> >> >> > [DEBUG] switch_ivr_bridge.c:384 Send signal > >> >> >> > sofia/external/xxxxxxxxxx at 10.10.2.3 [BREAK] > >> >> >> > [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE > >> >> >> > [sofia/external/yyyyyyyyyy] > >> >> >> > > >> >> >> > Whether Iam missing anything in the config? > >> >> >> > > >> >> >> > Thanks in advance. > >> >> >> > 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 > >> >> >> > > >> >> >> > FreeSWITCH-users mailing list > >> >> >> > FreeSWITCH-users at lists.freeswitch.org > >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> >> > > >> >> >> > > >> >> >> > UNSUBSCRIBE:http://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 > >> >> >> > >> >> >> FreeSWITCH-users mailing list > >> >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> >> > >> >> >> > >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> >> http://www.freeswitch.org > >> >> > > >> >> > > >> >> > > >> >> > _________________________________________________________________________ > >> >> > Professional FreeSWITCH Consulting Services: > >> >> > 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 > >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > > >> >> > UNSUBSCRIBE:http://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 > >> >> > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> > > >> > > >> > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > 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 > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://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 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > 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 > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/27a5520b/attachment-0001.html From justlikeef at gmail.com Fri Feb 10 23:46:57 2012 From: justlikeef at gmail.com (Rob Hutton) Date: Fri, 10 Feb 2012 15:46:57 -0500 Subject: [Freeswitch-users] Help with attended transfer / att_xfer In-Reply-To: <067b01cce804$16741000$435c3000$@launch3.net> References: <029e01cce5ef$aaadc0c0$00094240$@launch3.net> <399BEF93D8F8443BA14E1A90B8443AC0@freeswitch.org> <03d901cce6bf$bcc682d0$36538870$@launch3.net> <52A04E7E1B414CD0AF0D33A279B68C4F@freeswitch.org> <067b01cce804$16741000$435c3000$@launch3.net> Message-ID: We're using 2100s also, and the MOH plays when doing an attended transfer. If you look at the process, pressing the second line button actually puts the first on hold. On Fri, Feb 10, 2012 at 9:55 AM, Brett Wilson wrote: > I have the MOH setup correctly I think. When I do a blind transfer through > the phone, and the other extension is ringing and not picked up yet, MOH > does work. Only the phone?s attended transfer exhibits the strange > behavior. I?ll have to get around to doing a packet cap one day.**** > > ** ** > > *Brett Wilson* > > *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: Blogger-logo][image: > Description: Description: FaceBook-Logo][image: > Description: Description: Twitter-Logo] > **** > > ** ** > > *From:* Anthony Minessale [mailto:anthony.minessale at gmail.com] > *Sent:* Thursday, February 09, 2012 11:47 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help with attended transfer / att_xfer** > ** > > ** ** > > normal sip transfers should support MOH, you just have to set hold_music > to the filename of your moh or install the default config and sound sets > and swap out the sounds in the moh local_stream**** > > 2012/2/8 Jo?o Mesquita **** > > If I am reading code correctly, you should be able to insert the required > play code around line #2105 of mod_dptools.c**** > > ** ** > > I might add this patch tomorrow because I believe it is a pretty cool > feature to have with a configurable tone to be played.**** > > ** ** > > Regards,**** > > ** ** > > -- **** > > Jo?o Mesquita**** > > Sent with Sparrow **** > > ** ** > > On Wednesday, February 8, 2012 at 9:39 PM, Michael Collins wrote:**** > > I agree w/ jmesquita - this probably needs a patch, although I'd need to > dig further to see if there's a non-patchy workaround. > > As far as FusionPBX goes, you should be able to do a git pull and make > install to keep yourself updated. Just confirm with Mark Crane (IRC: > mcrane) or one of the other fusionpbx guys about any caveats with things > like the target install directory, etc. > > -MC**** > > On Wed, Feb 8, 2012 at 4:14 PM, Brett Wilson wrote: > > **** > > Thanks guys I am using the FS att_xfer along with the meta bind. I tried > attended transfer on my gxp2100 phones and it works but I am under the > impression that the phone is actually handling the sip switching itself for > that functionality, and our MOH was not being played while the two internal > parties were speaking, before putting through the transfer. It was just > dead silence which did not sit well with our customers. So I switched to > the FS feature, and it works great. Only problem is that we don?t know when > to start speaking after the transfer!**** > > **** > > Also I have another question. I have FS installed via the FusionPBX linux > disk image. What is the best way to keep FS updated? Do I need to pull from > git and build from source or what?**** > > **** > > *Brett Wilson***** > > *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: Blogger-logo][image: > Description: Description: FaceBook-Logo][image: > Description: Description: Twitter-Logo] > **** > > **** > > *From:* Jo?o Mesquita [mailto:jmesquita at freeswitch.org] > *Sent:* Wednesday, February 08, 2012 5:08 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Help with attended transfer / att_xfer** > ** > > **** > > I believe he is using the feature code, therefore att_xfer so this would > require a patch of some kind. I am this on my todo list which is quite > large nowadays and I don't even know if it is feasible yet. I can take a > look at the code later on.**** > > **** > > Regards,**** > > **** > > -- **** > > Jo?o Mesquita**** > > Sent with Sparrow **** > > **** > > On Wednesday, February 8, 2012 at 2:40 PM, Michael Collins wrote:**** > > Brett, > > No worries on the double-post - we have a first-time sender moderation > filter so I had to allow your messages through. You should be go to go from > here on out. > > As to your question, are you using the default configuration or are you > doing something different? Also, how are you executing the transfer - using > the transfer button on the telephone or the *1 feature code? > > Thanks, > MC**** > > On Tue, Feb 7, 2012 at 3:24 PM, Brett Wilson wrote:*** > * > > Hey guys,**** > > I need a pointer. I?m trying to get some kind of functionality where after > a attended transfer is completed, ie. The ouside caller is connected to the > 2nd phone after the 1st phone has hung up or chosen to continue the > transfer. I would like a beep to sound for the 2nd phone or for both > legs, just to let the recipient of the transfer know that it has gone > through and they can start talking.**** > > Any ideas?**** > > **** > > Thanks**** > > **** > > **** > > *Brett Wilson***** > > *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: Blogger-logo][image: > Description: Description: FaceBook-Logo][image: > Description: Description: Twitter-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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > **** > > _________________________________________________________________________* > *** > > Professional FreeSWITCH Consulting Services:**** > > 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**** > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users**** > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users** > ** > > http://www.freeswitch.org**** > > **** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________* > *** > > Professional FreeSWITCH Consulting Services:**** > > 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**** > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users**** > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users** > ** > > http://www.freeswitch.org**** > > ** ** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/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/jpeg Size: 1680 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/570e4820/attachment-0005.jpe From holger at freyther.de Fri Feb 10 21:28:52 2012 From: holger at freyther.de (Holger Freyther) Date: Fri, 10 Feb 2012 18:28:52 +0000 (UTC) Subject: [Freeswitch-users] How to inspect the jitterbuffer? References: Message-ID: Anthony Minessale writes: > > If there is no RTP then the TDM should be generating its own audio. > > Try > > MTU = 80 > IDLE_FLAG = 0xFF Writing about audio is difficult. Imagine one says the word "Two".. and one called a mailbox. Opening the file in audacity gives you the wave of what you said, you zoom in and you see that in the middle of the word the the amplitude is zero for a bit (where it should not be zero). As echoing works on FreeTDM, there is no packet loss in RTP, the bug has the whole audio... it means something is broken in FreeSWITCH to get the audio to FreeTDM in time? Any ideas? From rzumaeta at yahoo.com Fri Feb 10 22:24:11 2012 From: rzumaeta at yahoo.com (=?iso-8859-1?Q?Rodrigo_Andr=E9s_Zumaeta?=) Date: Fri, 10 Feb 2012 16:24:11 -0300 Subject: =?iso-8859-1?q?_=5BFreeswitch-users=5D_=16Buffer_size_sanity_che?= =?iso-8859-1?q?ck_failed!?= Message-ID: Hello, this is my first message to the list! when connecting one specific group of users to Freeswitch, I get this message: [CRIT] switch_core_codec.c:760 Buffer size sanity check failed! The call is then dropped. That is, on 1.0.6. When I upgrade to the latest git version, I still get the same message, but the call is not dropped anymore. However, I get only one way audio. As aforementioned, I just get this odd behavior with one group of users. I will be happy to provide any relevant information. From philq at qsystemsengineering.com Fri Feb 10 23:50:20 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Fri, 10 Feb 2012 15:50:20 -0500 Subject: [Freeswitch-users] Registration problem after attempting to install mod_opal and updating to latest git pull Message-ID: <01f801cce835$9ab5faf0$d021f0d0$@com> Ok, here's part of the sip trace as requested. I won't have time to sanitize the whole thing until later but in the meantime, here's an interesting excerpt that I wanted to make a few comments on: First, notice the "Unauthorized" response to the first registration attempt but the next attempt is successful. This has actually been going on as long as I can remember with this particular provider. ------------------------------------------------------------------------ 2012-02-10 12:14:32.174952 [WARNING] sofia.c:4717 Unregister HL7611 2012-02-10 12:14:32.174952 [WARNING] sofia.c:4730 Ping failed HL7611 with code 900 - count -1/-1/1, state UP send 854 bytes to udp/[140.239.143.5]:5060 at 17:14:33.595032: ------------------------------------------------------------------------ REGISTER sip:sip16.vitelity.net;transport=udp SIP/2.0 Via: SIP/2.0/UDP 71.179.xx.xx:5080;rport;branch=z9hG4bK7857r78Xt8pKg Max-Forwards: 70 From: ;tag=D0jHcrrtFjtym To: Call-ID: 04cac6f5-9608-4019-9658-6c0e6d1fd7af CSeq: 24054543 REGISTER Contact: Expires: 120 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4c07a00 2012-02-08 16-52-13 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Authorization: Digest username="thisis7611", realm="asterisk", nonce="xxxxxxxx", algorithm=MD5, uri="sip:sip16.vitelity.net;transport=udp", response="d7ecbd5585a7bef456be36b8e5fe0b8e" Content-Length: 0 ------------------------------------------------------------------------ recv 463 bytes from udp/[140.239.143.5]:5060 at 17:14:33.656879: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 71.179.xx.xx:5080;branch=z9hG4bK7857r78Xt8pKg;received=71.179.xx.xx;rport=50 80 From: ;tag=D0jHcrrtFjtym To: Call-ID: 04cac6f5-9608-4019-9658-6c0e6d1fd7af CSeq: 24054543 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 ------------------------------------------------------------------------ recv 560 bytes from udp/[140.239.143.5]:5060 at 17:14:33.656961: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 71.179.xx.xx:5080;branch=z9hG4bK7857r78Xt8pKg;received=71.179.xx.xx;rport=50 80 From: ;tag=D0jHcrrtFjtym To: ;tag=as117f4b90 Call-ID: 04cac6f5-9608-4019-9658-6c0e6d1fd7af CSeq: 24054543 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="xxxxxxxx" Content-Length: 0 ------------------------------------------------------------------------ send 854 bytes to udp/[140.239.143.5]:5060 at 17:14:33.657072: ------------------------------------------------------------------------ REGISTER sip:sip16.vitelity.net;transport=udp SIP/2.0 Via: SIP/2.0/UDP 71.179.xx.xx:5080;rport;branch=z9hG4bK8HZ0t2S1QHD6B Max-Forwards: 70 From: ;tag=D0jHcrrtFjtym To: Call-ID: 04cac6f5-9608-4019-9658-6c0e6d1fd7af CSeq: 24054544 REGISTER Contact: Expires: 120 User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4c07a00 2012-02-08 16-52-13 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Authorization: Digest username="thisis7611", realm="asterisk", nonce="xxxxxxxx", algorithm=MD5, uri="sip:sip16.vitelity.net;transport=udp", response="d82a35caa676c34e817dd63a658d34ac" Content-Length: 0 ------------------------------------------------------------------------ recv 463 bytes from udp/[140.239.143.5]:5060 at 17:14:33.719551: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 71.179.xx.xx:5080;branch=z9hG4bK8HZ0t2S1QHD6B;received=71.179.xx.xx;rport=50 80 From: ;tag=D0jHcrrtFjtym To: Call-ID: 04cac6f5-9608-4019-9658-6c0e6d1fd7af CSeq: 24054544 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Content-Length: 0 ------------------------------------------------------------------------ recv 604 bytes from udp/[140.239.143.5]:5060 at 17:14:33.721743: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 71.179.xx.xx:5080;branch=z9hG4bK8HZ0t2S1QHD6B;received=71.179.xx.xx;rport=50 80 From: ;tag=D0jHcrrtFjtym To: ;tag=as117f4b90 Call-ID: 04cac6f5-9608-4019-9658-6c0e6d1fd7af CSeq: 24054544 REGISTER User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO Supported: replaces Expires: 60 Contact: ;expires=60 Date: Fri, 10 Feb 2012 17:14:33 GMT Content-Length: 0 Next, what's with the "Bad event", the time discrepancy, and "sip:ping at invalid" in the following exchange? ------------------------------------------------------------------------ recv 683 bytes from udp/[192.168.1.4]:5060 at 19:33:11.486617: ------------------------------------------------------------------------ SUBSCRIBE sip:102 at 192.168.1.6:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.4;branch=z9hG4bKfe095021fa5ec8e60;rport Route: Max-Forwards: 70 From: "Phil" ;tag=a4efbafcef To: Call-ID: 17f321dbe885cc25 CSeq: 24812 SUBSCRIBE Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Contact: "Phil" ;+sip.instance="" Event: as-feature-event Expires: 3600 Supported: path User-Agent: Aastra 9480i/3.2.2.1136 Content-Length: 0 ------------------------------------------------------------------------ send 653 bytes to udp/[192.168.1.4]:5060 at 19:33:11.486724: ------------------------------------------------------------------------ SIP/2.0 489 Bad Event Via: SIP/2.0/UDP 192.168.1.4;branch=z9hG4bKfe095021fa5ec8e60;rport=5060 From: "Phil" ;tag=a4efbafcef To: ;tag=gS042N7B17j3j Call-ID: 17f321dbe885cc25 CSeq: 24812 SUBSCRIBE User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4c07a00 2012-02-08 16-52-13 -0600 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-Length: 0 ------------------------------------------------------------------------ recv 958 bytes from udp/[74.93.xx.xx]:1060 at 19:33:11.837860: ------------------------------------------------------------------------ REGISTER sip:qsystemseng.mydomain.org:5060 SIP/2.0 Via: SIP/2.0/UDP 74.93.xx.xx:51660;branch=z9hG4bKaaab13558da89266a;rport Route: Max-Forwards: 70 From: "Tina" ;tag=5dfb41956c To: "Tina" Call-ID: 69748b87cb4b8d1d CSeq: 35981 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="257",realm="qsystemseng.mydomain.org",nonce="xxxxxxxx",uri="sip:qs ystemseng.mydomain.org:5060",response="48feb96d103538b600c37e6bef966799",alg orithm=MD5,qop=auth,cnonce="xxxxxxxx",nc=00000004 Contact: "Tina" ;+sip.instance="";expires=90 Supported: path, gruu User-Agent: Aastra 9143i/3.2.2.1136 Content-Length: 0 ------------------------------------------------------------------------ send 640 bytes to udp/[74.93.xx.xx]:1060 at 19:33:11.839412: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 74.93.xx.xx:51660;branch=z9hG4bKaaab13558da89266a;rport=1060 From: "Tina" ;tag=5dfb41956c To: "Tina" ;tag=H2SX4grFyg9Ne Call-ID: 69748b87cb4b8d1d CSeq: 35981 REGISTER Contact: ;expires=90 Date: Fri, 10 Feb 2012 19:33:11 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4c07a00 2012-02-08 16-52-13 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 2012-02-10 14:33:12.381474 [WARNING] sofia.c:4717 Unregister HL7611 2012-02-10 14:33:12.381474 [WARNING] sofia.c:4730 Ping failed HL7611 with code 900 - count -1/-1/1, state UP 2012-02-10 14:33:13.389496 [WARNING] sofia_reg.c:474 HL7611 Failed Registration [0], setting retry to 90 seconds. recv 917 bytes from udp/[74.93.xx.xx]:1064 at 19:33:13.465516: ------------------------------------------------------------------------ REGISTER sip:qsystemseng.mydomain.org:5060 SIP/2.0 Via: SIP/2.0/UDP 74.93.xx.xx:51620;branch=z9hG4bKc6b28c84c8b2a8b8e;rport Max-Forwards: 70 From: "Laura" ;tag=87516cda24 To: "Laura" Call-ID: 09856e1e62e97740 CSeq: 27466 REGISTER Allow: INVITE, ACK, CANCEL, BYE, NOTIFY, REFER, OPTIONS, UPDATE, PRACK, SUBSCRIBE, INFO Allow-Events: talk, hold, conference, LocalModeStatus Authorization: Digest username="227",realm="qsystemseng.mydomain.org",nonce="xxxxxxxx",uri="sip:qs ystemseng.mydomain.org:5060",response="05a8d65372aef7cbd3d7968ba5844fa8",alg orithm=MD5,qop=auth,cnonce="xxxxxxxx",nc=00000003 Contact: "Laura" ;+sip.instance="";expires=90 Supported: path, gruu User-Agent: Aastra 9143i/3.2.2.1136 Content-Length: 0 ------------------------------------------------------------------------ send 642 bytes to udp/[74.93.xx.xx]:1064 at 19:33:13.467000: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 74.93.xx.xx:51620;branch=z9hG4bKc6b28c84c8b2a8b8e;rport=1064 From: "Laura" ;tag=87516cda24 To: "Laura" ;tag=jBKp6B9jUSZ8S Call-ID: 09856e1e62e97740 CSeq: 27466 REGISTER Contact: ;expires=90 Date: Fri, 10 Feb 2012 19:33:13 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4c07a00 2012-02-08 16-52-13 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ recv 450 bytes from udp/[216.115.69.144]:5060 at 19:33:15.332881: ------------------------------------------------------------------------ OPTIONS sip:71.179.xx.xx:5080 SIP/2.0 Max-Forwards: 10 Record-Route: Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bKfc94.cc57baa771388125df1b041bbd688213.0 Via: SIP/2.0/UDP 70.167.153.136:5060;branch=0 Route: From: sip:ping at invalid;tag=6c8e4582 To: sip:71.179.xx.xx:5080 Call-ID: aa72439-f0144116-cf3b885 at 70.167.153.136 CSeq: 1 OPTIONS Content-Length: 0 ------------------------------------------------------------------------ send 670 bytes to udp/[216.115.69.144]:5060 at 19:33:15.333175: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 216.115.69.144;branch=z9hG4bKfc94.cc57baa771388125df1b041bbd688213.0 Via: SIP/2.0/UDP 70.167.153.136:5060;branch=0 From: sip:ping at invalid;tag=6c8e4582 To: ;tag=21ecKKFFNK3cj Call-ID: aa72439-f0144116-cf3b885 at 70.167.153.136 CSeq: 1 OPTIONS Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4c07a00 2012-02-08 16-52-13 -0600 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 Interestingly, I can't reproduce the SIPBroker problem after the last update, even though I re-enabled ping. Ping is not necessary with SIPBroker as it's only for outgoing calls and you don't need to register with them. I'll try to sanitize and post a complete trace here later tonight. Thanks again, - Phil -----Original Message----- From: Phil Quesinberry [mailto:philq at qsystemsengineering.com] Sent: Thursday, February 09, 2012 12:51 AM To: 'freeswitch-users at lists.freeswitch.org' Subject: Registration problem after attempting to install mod_opal and updating to latest git pull After a recent unsuccessful attempt to install mod_opal and updating to a recent git, I now get some odd reports of gateway registration failures, although everything appears to be working ok call-wise for the moment. Doing a 'make uninstall' from /root/opal and updating to the latest git tonight yielded no improvement. Here are some excerpts of failures reported in fs_cli: 2012-02-09 00:12:59.992099 [WARNING] sofia.c:4717 Unregister QS8002 2012-02-09 00:12:59.992099 [WARNING] sofia.c:4730 Ping failed QS8002 with code 900 - count -1/-1/1, state UP 2012-02-09 00:12:59.992099 [WARNING] sofia.c:4717 Unregister HL7612 2012-02-09 00:12:59.992099 [WARNING] sofia.c:4730 Ping failed HL7612 with code 900 - count -1/-1/1, state UP 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4717 Unregister HL7611 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4730 Ping failed HL7611 with code 900 - count -1/-1/1, state UP 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4717 Unregister HL7519 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4730 Ping failed HL7519 with code 900 - count -1/-1/1, state UP 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4717 Unregister CallCentric_T.38 2012-02-09 00:13:00.011988 [WARNING] sofia.c:4730 Ping failed CallCentric_T.38 with code 900 - count -1/-1/1, state UP 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 QS8002 Failed Registration [0], setting retry to 60 seconds. 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 HL7612 Failed Registration [0], setting retry to 90 seconds. 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 HL7611 Failed Registration [0], setting retry to 90 seconds. 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 HL7519 Failed Registration [0], setting retry to 60 seconds. 2012-02-09 00:13:01.992050 [WARNING] sofia_reg.c:474 CallCentric_T.38 Failed Registration [0], setting retry to 60 seconds. To give a sense of the timing, the three groups of warnings below are contiguous, from a single fs_cli session: 2012-02-09 00:21:18.984384 [WARNING] sofia.c:4717 Unregister HL7611 2012-02-09 00:21:18.984384 [WARNING] sofia.c:4730 Ping failed HL7611 with code 900 - count -1/-1/1, state UP 2012-02-09 00:21:20.988379 [WARNING] sofia_reg.c:474 HL7611 Failed Registration [0], setting retry to 90 seconds. 2012-02-09 00:21:22.988382 [WARNING] sofia.c:4717 Unregister HL7519 2012-02-09 00:21:22.988382 [WARNING] sofia.c:4730 Ping failed HL7519 with code 900 - count -1/-1/1, state UP 2012-02-09 00:21:24.988318 [WARNING] sofia_reg.c:474 HL7519 Failed Registration [0], setting retry to 60 seconds. 2012-02-09 00:22:00.447778 [WARNING] sofia_reg.c:1422 SIP auth challenge (REGISTER) on sofia profile 'internal' for [226 at qsystemseng.no-ip.org] from ip 74.93.222.182 2012-02-09 00:22:04.147688 [WARNING] sofia.c:4717 Unregister HL7612 2012-02-09 00:22:04.147688 [WARNING] sofia.c:4730 Ping failed HL7612 with code 900 - count -1/-1/1, state UP 2012-02-09 00:22:04.167699 [WARNING] sofia.c:4717 Unregister HL7611 2012-02-09 00:22:04.167699 [WARNING] sofia.c:4730 Ping failed HL7611 with code 900 - count -1/-1/1, state UP 2012-02-09 00:22:04.167699 [WARNING] sofia.c:4717 Unregister CallCentric_T.38 2012-02-09 00:22:04.167699 [WARNING] sofia.c:4730 Ping failed CallCentric_T.38 with code 900 - count -1/-1/1, state UP 2012-02-09 00:22:05.167744 [WARNING] sofia_reg.c:474 HL7611 Failed Registration [0], setting retry to 90 seconds. 2012-02-09 00:22:05.167744 [WARNING] sofia_reg.c:474 CallCentric_T.38 Failed Registration [0], setting retry to 60 seconds. 2012-02-09 00:22:05.167744 [WARNING] sofia.c:4717 Unregister QS8002 2012-02-09 00:22:05.167744 [WARNING] sofia.c:4730 Ping failed QS8002 with code 900 - count -1/-1/1, state UP 2012-02-09 00:22:07.171680 [WARNING] sofia_reg.c:474 QS8002 Failed Registration [0], setting retry to 60 seconds. 2012-02-09 00:22:07.171680 [WARNING] sofia.c:4717 Unregister HL7519 2012-02-09 00:22:07.171680 [WARNING] sofia.c:4730 Ping failed HL7519 with code 900 - count -1/-1/1, state UP One of the gateways is SIPBroker, and calls through that gate starting failing after the update. I realized that the gateway was configured to ping SIPBroker unnecessarily and restarting the gateway would restore service until the ping. I disabled the ping for that gateway and calls kept working after that. Any ideas as to what might be going on? There was no such problem before the FS update/attempt to use mod_opal. I can provide additional/more detailed info if needed. 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/20120210/fa70ebff/attachment-0001.html From msc at freeswitch.org Sat Feb 11 00:01:55 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 10 Feb 2012 13:01:55 -0800 Subject: [Freeswitch-users] Registration problem after attempting to install mod_opal and updating to latest git pull In-Reply-To: <01f801cce835$9ab5faf0$d021f0d0$@com> References: <01f801cce835$9ab5faf0$d021f0d0$@com> Message-ID: On Fri, Feb 10, 2012 at 12:50 PM, Phil Quesinberry < philq at qsystemsengineering.com> wrote: > ** > > *Ok, here's part of the sip trace as requested. I won't have time to > sanitize the whole thing until later but in the meantime, here's an > interesting excerpt that I wanted to make a few comments on:* > > *First, notice the "Unauthorized" response to the first registration > attempt but the next attempt is successful. This has actually been going > on as long as I can remember with this particular provider.* > As far as the auth goes I believe it is required to have the registrar send out the 401 first because it contains a nonce that assists in keeping the communication relatively secure. If I could send a single REG and magically authenticate then that would make a SIP replay attack really easy. If I understand all this correctly, the registrar sending a 401 "Unauthorized" does not mean, "Go away." Rather it means, "I'm not gonna let you in unless you give me the magic password. Here's a nonce to help you calculate the proper digest. I'm waiting for your next REGISTER message with the appropriate Authorization header." I'll have to defer to those more experienced than I on the rest of the post. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/beb1d98b/attachment.html From anthony.minessale at gmail.com Sat Feb 11 00:15:46 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 10 Feb 2012 15:15:46 -0600 Subject: [Freeswitch-users] Registration problem after attempting to install mod_opal and updating to latest git pull In-Reply-To: References: <01f801cce835$9ab5faf0$d021f0d0$@com> Message-ID: try latest GIT On Fri, Feb 10, 2012 at 3:01 PM, Michael Collins wrote: > > > On Fri, Feb 10, 2012 at 12:50 PM, Phil Quesinberry > wrote: >> >> Ok, here's part of the sip trace as requested.? I won't have time to >> sanitize the whole thing until later but in the meantime, here's an >> interesting excerpt that I wanted to make a few comments on: >> >> First, notice the "Unauthorized" response to the first registration >> attempt but the next attempt is successful.? This has actually been going on >> as long as I can remember with this particular provider. > > > As far as the auth goes I believe it is required to have the registrar send > out the 401 first because it contains a nonce that assists in keeping the > communication relatively secure. If I could send a single REG and magically > authenticate then that would make a SIP replay attack really easy. > > If I understand all this correctly, the registrar sending a 401 > "Unauthorized" does not mean, "Go away." Rather it means, "I'm not gonna let > you in unless you give me the magic password. Here's a nonce to help you > calculate the proper digest. I'm waiting for your next REGISTER message with > the appropriate Authorization header." > > I'll have to defer to those more experienced than I on the rest of the post. > > -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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 krice at freeswitch.org Sat Feb 11 00:17:37 2012 From: krice at freeswitch.org (Ken Rice) Date: Fri, 10 Feb 2012 15:17:37 -0600 Subject: =?iso-8859-1?q?Re=3A_=5BFreeswitch-users=5D_=16Buffer_size_sanit?= =?iso-8859-1?q?y_check_failed!?= In-Reply-To: Message-ID: Can you provide more information on that group of users? Can you duplicate this at will? ie: is it a specific hard phone or softphone using a specific codec? Also console logs and network traces would be helpful... On 2/10/12 1:24 PM, "Rodrigo Andr?s Zumaeta" wrote: > Hello, this is my first message to the list! > when connecting one specific group of users to Freeswitch, I get this message: > > [CRIT] switch_core_codec.c:760 Buffer size sanity check failed! > > The call is then dropped. That is, on 1.0.6. When I upgrade to the latest git > version, I still get the same message, but the call is not dropped anymore. > However, I get only one way audio. > > As aforementioned, I just get this odd behavior with one group of users. > > I will be happy to provide any relevant information. > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > 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 Feb 11 00:24:24 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 10 Feb 2012 15:24:24 -0600 Subject: [Freeswitch-users] How to inspect the jitterbuffer? In-Reply-To: References: Message-ID: That made no sense in regards to my last message. I asked you to try particular wanpipe parameters, so you should try them and restart your wanpipe and report the results. Your MTU is very high and may cause timing issues for interop with voip. set MTU to 80 (10ms) set IDLE_FLAG to 0xFF (send the 255 byte on idle stream) 1) Turn off FreeSWITCH 2) wanrouter stop 3) wanrouter start 4) start FreeSWITCH You are pushing RTP up against TDM, RTP can start and stop, TDM cannot, When there is no RTP the TDM will carry on sending the IDLE_FLAG byte over the wire. I made both the RTP stack and the TDM endpoint from scratch so you should assume I may have some insight. On Fri, Feb 10, 2012 at 12:28 PM, Holger Freyther wrote: > Anthony Minessale writes: > >> >> If there is no RTP then the TDM should be generating its own audio. >> >> Try >> >> MTU ? ? ? ? ? = 80 >> IDLE_FLAG ? ? = 0xFF > > Writing about audio is difficult. Imagine one says the word "Two".. > and one called a mailbox. Opening the file in audacity gives you > the wave of what you said, you zoom in and you see that in the > middle of the word the the amplitude is zero for a bit (where it > should not be zero). As echoing works on FreeTDM, there is no packet > loss in RTP, the bug has the whole audio... it means something is > broken in FreeSWITCH to get the audio to FreeTDM in time? > > Any ideas? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 nasida at live.ru Sat Feb 11 01:06:30 2012 From: nasida at live.ru (Yuriy Nasida) Date: Sat, 11 Feb 2012 02:06:30 +0400 Subject: [Freeswitch-users] voicemail_say_phone_number In-Reply-To: References: , , <4F3060D2.7050505@earthspike.net>, , <6574CFF3DDE54C118E0E9250AC19EDD7@DWP>, <01b601cce54e$f865c250$e93146f0$@com>, <26ADC64FC37D4049A68787B3A0DCF4FA@DWP>, , , Message-ID: Michael, I have tried voicemail with latest git. Looks good, macro works now, thank you. But I see many WARNING messages in fs_cli when i try to check or leave voicemail. Not sure why. It appear when I don't use vm_announce_cid as well. '2012-02-11 00:49:20.169712 [WARNING] switch_xml.c:2329 Invalid UTF-8 character to ampersand, skip it' Thanks. Date: Tue, 7 Feb 2012 10:28:22 -0800 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] voicemail_say_phone_number Okay, try latest git. I changed mod_voicemail to use the macro and I set vm_announce_cid to be true/false instead of using a file name. Try it with various caller ID number values and let us know how it goes. -MC On Tue, Feb 7, 2012 at 8:11 AM, Michael Collins wrote: Yeah, this was my bad. I started on this and got interrupted, then came back like an hour later and totally forgot about the fact that of mod_voicemail.c not using the macro. I think today I can do a quick test on that and if it works I'll commit it to git master and then you all can test it. Thanks, MC On Tue, Feb 7, 2012 at 2:22 AM, Yuriy Nasida wrote: Thank you guys! It works. The example with 'vm_announce_cid=ivr/ivr-this_is_a_call_from.wav' is more correct. Otherwise FS tries to play message true.wav :)So, voicemail module doesn't use macro "voicemail_say_phone_number" and as far as I see the source code of mod_voicemail.c explains this behaviour. Yes, ability for playing of some wav file if I will have vm from annonymous would be very useful. Probably I can add it independently but I believe that your modifying will more correctly :)Darcy please let me know if you plan to add this feature in the near future. Anyway thanks again! From: darcy at primrose.ws To: freeswitch-users at lists.freeswitch.org Date: Mon, 6 Feb 2012 23:27:23 -0500 Subject: Re: [Freeswitch-users] voicemail_say_phone_number If John?s works, use it, I could not make it work but the example I show below played the greeting before the clid, I have not added anything to play an annonymous greeting yet as I have not had time. in mod_voicemail.c you have the following code: if (!zstr(cbt->cid_number) && (vm_announce_cid = switch_channel_get_variable(channel, "vm_announce_cid"))) { switch_ivr_play_file(session, NULL, vm_announce_cid, NULL); switch_ivr_sleep(session, 500, SWITCH_TRUE, NULL); switch_ivr_say(session, cbt->cid_number, NULL, "name_spelled", "pronounced", NULL, NULL); } Which indicates you will play the file in variable ?vm_announce_cid?. Darcy In Sunny Ottawa Canada From: Bote Man Sent: Monday, February 06, 2012 11:14 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] voicemail_say_phone_number Well, now I'm cornfused. The original example by John used vm_announce_cid as a Boolean switch. But you've tested it as a string that points to the desired sound file? I have no means to test this nor access to sources right now and I just added John's example to the wiki. I should back it out or correct it based on your results. Please advise. John Boteler Bote Communications in rainy Fort Lauderdale, FL From: Darcy Sent: Monday, 06 February, 2012 22:02 The message, this_is_a_call_from actually has to be set in the dialplan it appears, the fs plays the file set in vm_announce_cid, a simple dial plan below reflects one way of doing this, tested and it works. Needs more time to suit the total requirements, but this makes it a little more professional by adding the message in front of the number. Darcy ? On Mon, Feb 6, 2012 at 3:22 PM, John wrote: Michael, It does work, but it's a bit 'rough': all it does it speak the number just before the date. So I can understand why it was not documented... ? John PS. For those who want to know where to insert this, put the line marked with + into your conf/dialplan/default.xml file: + On 06/02/12 22:00, Michael Collins wrote: Yuriy, ?Please set vm_announce_cid to true prior to checking voicemail and see if it works. If it does, please let me know. If you can add it to the wiki then do so, otherwise one of our intrepid community members will do it. -MC 2012/2/6 Yuriy Nasida Hello list, I would like to have one simple ability to listen the phone number of caller when I check my voicemail. At present I listen date of message only. I have found the macro "voicemail_say_phone_number" in conf/lang/en/vm/sounds.xml. But... when FS uses it ? Can I get it without modifying of source code of voicemail module ? Please advise. Thanks. _________________________________________________________________________ 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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120211/f9e78f46/attachment-0001.html From rzumaeta at yahoo.com Sat Feb 11 00:57:48 2012 From: rzumaeta at yahoo.com (=?iso-8859-1?Q?Rodrigo_Andr=E9s_Zumaeta?=) Date: Fri, 10 Feb 2012 18:57:48 -0300 Subject: =?iso-8859-1?q?Re=3A_=5BFreeswitch-users=5D_=16Buffer_size_sanit?= =?iso-8859-1?q?y_check_failed!?= In-Reply-To: References: Message-ID: <672EBD90-192F-4907-B700-20BDAD8F7950@yahoo.com> Ken, thanks for your reply, let's see: 1. there is a group of users using analog phones. 2. there is another group using digital phones. 3. there is a third group that is not relevant for this specific problems. All of these groups are connected to a Nortel, and we get H.323 traffic from them. The first group of users, the ones using analog phones, is the one reporting this problem. It happens with 100% of their calls. I am attaching a trace and console log... The only noticeable difference in the flow of the call (comparing to successful calls from digital phones) is the presence of CN RTP packets. -------------- next part -------------- A non-text attachment was scrubbed... Name: analog_fail3.pcap Type: application/octet-stream Size: 6259 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/42de91e4/attachment-0002.obj -------------- next part -------------- A non-text attachment was scrubbed... Name: analog_fail.log Type: application/octet-stream Size: 166773 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/42de91e4/attachment-0003.obj -------------- next part -------------- On Feb 10, 2012, at 6:17 PM, Ken Rice wrote: > Can you provide more information on that group of users? Can you duplicate > this at will? ie: is it a specific hard phone or softphone using a specific > codec? > > Also console logs and network traces would be helpful... > > > On 2/10/12 1:24 PM, "Rodrigo Andr?s Zumaeta" wrote: > >> Hello, this is my first message to the list! >> when connecting one specific group of users to Freeswitch, I get this message: >> >> [CRIT] switch_core_codec.c:760 Buffer size sanity check failed! >> >> The call is then dropped. That is, on 1.0.6. When I upgrade to the latest git >> version, I still get the same message, but the call is not dropped anymore. >> However, I get only one way audio. >> >> As aforementioned, I just get this odd behavior with one group of users. >> >> I will be happy to provide any relevant information. >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > 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 Sat Feb 11 02:15:42 2012 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Fri, 10 Feb 2012 18:15:42 -0500 Subject: [Freeswitch-users] Registration problem after attempting to install mod_opal and updating to latest git pull Message-ID: <000001cce849$ebc67410$c3535c30$@com> That appears to have fixed the problem. Thanks again. - Phil _____________________________________________ From: Phil Quesinberry Sent: Friday, February 10, 2012 3:50 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: RE: Registration problem after attempting to install mod_opal and updating to latest git pull Ok, here's part of the sip trace as requested. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/b0316c53/attachment.html From msc at freeswitch.org Sat Feb 11 02:17:38 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 10 Feb 2012 15:17:38 -0800 Subject: [Freeswitch-users] voicemail_say_phone_number In-Reply-To: References: <4F3060D2.7050505@earthspike.net> <6574CFF3DDE54C118E0E9250AC19EDD7@DWP> <01b601cce54e$f865c250$e93146f0$@com> <26ADC64FC37D4049A68787B3A0DCF4FA@DWP> Message-ID: The German xml file has poorly formatted umlaut characters. Try deleting the files in conf/lang/de/ and you should be fine. -MC On Fri, Feb 10, 2012 at 2:06 PM, Yuriy Nasida wrote: > Michael, > > I have tried voicemail with latest git. Looks good, macro works now, thank > you. But I see many WARNING messages in fs_cli when i try to check or leave > voicemail. Not sure why. It appear when I don't use vm_announce_cid as well. > > '2012-02-11 00:49:20.169712 [WARNING] switch_xml.c:2329 Invalid UTF-8 > character to ampersand, skip it' > > Thanks. > > ------------------------------ > Date: Tue, 7 Feb 2012 10:28:22 -0800 > From: msc at freeswitch.org > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] voicemail_say_phone_number > > Okay, try latest git. I changed mod_voicemail to use the macro and I set > vm_announce_cid to be true/false instead of using a file name. Try it with > various caller ID number values and let us know how it goes. > > -MC > > On Tue, Feb 7, 2012 at 8:11 AM, Michael Collins wrote: > > Yeah, this was my bad. I started on this and got interrupted, then came > back like an hour later and totally forgot about the fact that of > mod_voicemail.c not using the macro. I think today I can do a quick test on > that and if it works I'll commit it to git master and then you all can test > it. > > Thanks, > MC > > > On Tue, Feb 7, 2012 at 2:22 AM, Yuriy Nasida wrote: > > Thank you guys! > > It works. The example with > 'vm_announce_cid=ivr/ivr-this_is_a_call_from.wav' is more correct. > Otherwise FS tries to play message true.wav :) > So, voicemail module doesn't use macro "voicemail_say_phone_number" and as > far as I see the source code of mod_voicemail.c explains this behaviour. > Yes, ability for playing of some wav file if I will have vm from > annonymous would be very useful. > Probably I can add it independently but I believe that your modifying > will more correctly :) > Darcy please let me know if you plan to add this feature in the near > future. > > Anyway thanks again! > > ------------------------------ > From: darcy at primrose.ws > To: freeswitch-users at lists.freeswitch.org > Date: Mon, 6 Feb 2012 23:27:23 -0500 > Subject: Re: [Freeswitch-users] voicemail_say_phone_number > > > If John?s works, use it, I could not make it work but the example I show > below played the greeting before the clid, I have not added anything to > play an annonymous greeting yet as I have not had time. > > in mod_voicemail.c you have the following code: > if (!zstr(cbt->cid_number) && (vm_announce_cid = > switch_channel_get_variable(channel, "vm_announce_cid"))) { > switch_ivr_play_file(session, NULL, vm_announce_cid, NULL); > switch_ivr_sleep(session, 500, SWITCH_TRUE, NULL); > switch_ivr_say(session, cbt->cid_number, NULL, "name_spelled", > "pronounced", NULL, NULL); > } > Which indicates you will play the file in variable ?vm_announce_cid?. > > Darcy > In Sunny Ottawa Canada > > *From:* Bote Man > *Sent:* Monday, February 06, 2012 11:14 PM > *To:* 'FreeSWITCH Users Help' > *Subject:* Re: [Freeswitch-users] voicemail_say_phone_number > > Well, now I'm cornfused. The original example by John used > vm_announce_cid as a Boolean switch. But you've tested it as a string that > points to the desired sound file? > > I have no means to test this nor access to sources right now and I just > added John's example to the wiki. I should back it out or correct it based > on your results. > > Please advise. > > > John Boteler > Bote Communications > in rainy Fort Lauderdale, FL > > > > *From:* Darcy > *Sent:* Monday, 06 February, 2012 22:02 > > > The message, this_is_a_call_from actually has to be set in the > dialplan it appears, the fs plays the file set in vm_announce_cid, a simple > dial plan below reflects one way of doing this, tested and it works. > Needs more time to suit the total requirements, but this makes it a little > more professional by adding the message in front of the number. > > > > > data="vm_announce_cid=ivr/ivr-this_is_a_call_from.wav"/> > > > > > > Darcy > > > *?* > > On Mon, Feb 6, 2012 at 3:22 PM, John wrote: > Michael, > > It does work, but it's a bit 'rough': all it does it speak the number just > before the date. So I can understand why it was not documented... > ? > John > > PS. For those who want to know where to insert this, put the line marked > with + into your conf/dialplan/default.xml file: > > > > expression="^vmain$|^4000$|^\*98$"> > > > + > > > > > > On 06/02/12 22:00, Michael Collins wrote: > > Yuriy, > > ?Please set vm_announce_cid to true prior to checking voicemail and see > if it works. If it does, please let me know. If you can add it to the wiki > then do so, otherwise one of our intrepid community members will do it. > > -MC > 2012/2/6 Yuriy Nasida > Hello list, > > I would like to have one simple ability to listen the phone number of > caller when I check my voicemail. At present I listen date of message only. > I have found the macro "voicemail_say_phone_number" in > conf/lang/en/vm/sounds.xml. But... when FS uses it ? Can I get it without > modifying of source code of voicemail module ? > > Please advise. > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: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 FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE: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 FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/a802cbef/attachment-0001.html From anthony.minessale at gmail.com Sat Feb 11 02:28:07 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 10 Feb 2012 17:28:07 -0600 Subject: [Freeswitch-users] Buffer size sanity check failed! In-Reply-To: <672EBD90-192F-4907-B700-20BDAD8F7950@yahoo.com> References: <672EBD90-192F-4907-B700-20BDAD8F7950@yahoo.com> Message-ID: update again and reproduce the sanity check err and send the new value now with more info. On Fri, Feb 10, 2012 at 3:57 PM, Rodrigo Andr?s Zumaeta wrote: > Ken, > thanks for your reply, let's see: > 1. there is a group of users using analog phones. > 2. there is another group using digital phones. > 3. there is a third group that is not relevant for this specific problems. > > All of these groups are connected to a Nortel, and we get H.323 traffic from them. The first group of users, the ones using analog phones, is the one reporting this problem. It happens with 100% of their calls. I am attaching a trace and console log... The only noticeable difference in the flow of the call (comparing to successful calls from digital phones) is the presence of CN RTP packets. > > > > > > > > > > On Feb 10, 2012, at 6:17 PM, Ken Rice wrote: > >> Can you provide more information on that group of users? Can you duplicate >> this at will? ie: is it a specific hard phone or softphone using a specific >> codec? >> >> Also console logs and network traces would be helpful... >> >> >> On 2/10/12 1:24 PM, "Rodrigo Andr?s Zumaeta" wrote: >> >>> Hello, this is my first message to the list! >>> when connecting one specific group of users to Freeswitch, I get this message: >>> >>> ?[CRIT] switch_core_codec.c:760 Buffer size sanity check failed! >>> >>> The call is then dropped. That is, on 1.0.6. When I upgrade to the latest git >>> version, I still get the same message, but the call is not dropped anymore. >>> However, I get only one way audio. >>> >>> As aforementioned, I just get this odd behavior with one group of users. >>> >>> I will be happy to provide any relevant information. >>> >>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> 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 >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 freeswitch at earthspike.net Sat Feb 11 02:38:27 2012 From: freeswitch at earthspike.net (John) Date: Fri, 10 Feb 2012 23:38:27 +0000 Subject: [Freeswitch-users] How to inspect the jitterbuffer? In-Reply-To: References: Message-ID: <4F35AA73.9040205@earthspike.net> Holger, Anthony, I'm seeing a similar/identical 'feature' where the voice is intermittently broken on ISDN; I've raised ticket #1310 with Sangoma. My MTU is still on the default 80 but there is no IDLE_FLAG setting (default 0, I guess?). Interestingly(!), it is intermittent and unpredictable, but when it manifests there is no clear pattern as to what will be disrupted; A wav audio played by a Lua IVR will be fine, then on the same call the MOH will be broken, as will the voicemail outgoing message and finally the internal user's speech. The path from ISDN into FreeSWITCH is never disrupted (no surprise). Record_session doesn't show the breaks and there's nothing in the FS logs. I have sample audio taken from another (SIP only) FreeSWITCH instance which demonstrates the problem, but as the fault is intermittent and on a production system, debugging is taking some time. Sangoma has asked me to take samples at the ISDN (S?) interface and the echo canceller on the B700 board. When I next have the problem manifest, and grab those traces, I should be able to share the results once Sangoma has decoded them. I can make available the audio samples I already have if that would help. John On 10/02/12 21:24, Anthony Minessale wrote: > That made no sense in regards to my last message. > > I asked you to try particular wanpipe parameters, so you should try > them and restart your wanpipe and report the results. > Your MTU is very high and may cause timing issues for interop with voip. > > set MTU to 80 (10ms) > set IDLE_FLAG to 0xFF (send the 255 byte on idle stream) > > > 1) Turn off FreeSWITCH > 2) wanrouter stop > 3) wanrouter start > 4) start FreeSWITCH > > > You are pushing RTP up against TDM, RTP can start and stop, TDM > cannot, When there is no RTP the TDM will carry on sending the > IDLE_FLAG byte over the wire. > > I made both the RTP stack and the TDM endpoint from scratch so you > should assume I may have some insight. > > > On Fri, Feb 10, 2012 at 12:28 PM, Holger Freyther wrote: >> Anthony Minessale writes: >> >>> If there is no RTP then the TDM should be generating its own audio. >>> >>> Try >>> >>> MTU = 80 >>> IDLE_FLAG = 0xFF >> Writing about audio is difficult. Imagine one says the word "Two".. >> and one called a mailbox. Opening the file in audacity gives you >> the wave of what you said, you zoom in and you see that in the >> middle of the word the the amplitude is zero for a bit (where it >> should not be zero). As echoing works on FreeTDM, there is no packet >> loss in RTP, the bug has the whole audio... it means something is >> broken in FreeSWITCH to get the audio to FreeTDM in time? >> >> Any ideas? >> >> From anthony.minessale at gmail.com Sat Feb 11 02:46:13 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 10 Feb 2012 17:46:13 -0600 Subject: [Freeswitch-users] FreeSWITCH ends the calls after receiving DTMF from the Callee. In-Reply-To: References: Message-ID: Try this patch to print more debug info. It seems somewhere in the managed code something is assigning an input callback and its being executed and returns false which makes the call end. On Fri, Feb 10, 2012 at 2:13 PM, Srini K wrote: > > The Lua script works fine. No problem in capturing the DTMF from both the > call legs. > > Only in Mod_Managed I? have issue. Its simple 3 lines of code nothing > fancy... > > public void Run(AppContext context) { > ?context.Session.Answer(); > ?var newSession = new ManagedSession("sofia/gateway/foo/callenumber"); > ?freeswitch.bridge(newSession, context.Session); > } > > FreeSWITCH version is > ?FreeSWITCH Version 1.0.head (git-f477404 2012-02-09 11-08-52 -0600) > > Thanks > Srini >> Date: Fri, 10 Feb 2012 12:32:03 -0600 > >> From: anthony.minessale at gmail.com >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after receiving >> DTMF from the Callee. >> >> I tried this lua script which should be the same thing: >> >> b_leg = freeswitch.Session("sofia/internal/foo at bar.com", session); >> freeswitch.bridge(session, b_leg); >> >> See if you get the same problem running this lua code, I was passing dtmf >> fine. >> >> >> Internally, the code should not even use a callback capable of ending >> the call as you described unless your program has an input callback of >> dtmf handler function. >> >> >> Also make sure you are on a recent build of FS. >> >> >> >> On Fri, Feb 10, 2012 at 12:21 PM, Srini K wrote: >> > Anthony, >> > Thanks for your response. >> > I have not defined any of these input_callback or dtmf_callback. >> > >> > I have attached the log with debug level. >> > http://pastebin.freeswitch.org/18360 >> > >> > Thanks in advance. >> > >> > Regards >> > Srini >> > >> > >> > >> >> Date: Thu, 9 Feb 2012 19:06:39 -0600 >> > >> >> From: anthony.minessale at gmail.com >> >> To: freeswitch-users at lists.freeswitch.org >> >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after >> >> receiving >> >> DTMF from the Callee. >> >> >> >> do you have input_callback or dtmf_callback or anything defined? >> >> >> >> On Thu, Feb 9, 2012 at 6:37 PM, Srini K wrote: >> >> > Iam using the latest FreeSWITCh version (updated today morning). >> >> > >> >> > FS terminates on receiving any DTMF digits from the callee. I have >> >> > not >> >> > set >> >> > bridge_terminate_key. >> >> > >> >> > I have created an oubound session and bridging the inbound and >> >> > oubound >> > >> >> > session. Code snippet is >> >> > >> >> > var session = new ManagedSession("{origination_caller_id_number=" + >> >> > callerIdNumber + ",originate_timeout=8" + >> >> > "}sofia/gateway/408xxxxyyyy"); >> >> > string outBoundUuid = session.GetVariable("uuid"); >> >> > if (string.IsNullOrEmpty(outBoundUuid)) >> >> > { // Log error; >> >> > ??? return; >> >> > } >> >> > freeswitch.bridge(inboundSession, session); >> >> > >> >> > Also I have tried setting hangup_after_bridge=false. >> >> > >> >> > Regards >> >> > Srini >> >> > >> >> > >> >> > >> >> >> Date: Thu, 9 Feb 2012 18:17:45 -0600 >> >> > >> >> >> From: anthony.minessale at gmail.com >> >> >> To: freeswitch-users at lists.freeswitch.org >> >> >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after >> >> >> receiving >> >> >> DTMF from the Callee. >> >> >> >> >> >> the obvious reason would be setting bridge_terminate_key=1 >> >> >> but I don't know where your bridge is being started based on your >> >> >> description or what version of the code you are using. >> >> >> >> >> >> On Thu, Feb 9, 2012 at 6:06 PM, Srini K wrote: >> >> >> > Any reason that it didn't return success. Anything Iam missing it >> >> >> > out. >> >> >> > >> >> >> > >> >> >> >> Date: Thu, 9 Feb 2012 17:33:01 -0600 >> >> >> >> From: anthony.minessale at gmail.com >> >> >> >> To: freeswitch-users at lists.freeswitch.org >> >> >> >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after >> >> >> >> receiving >> >> >> >> DTMF from the Callee. >> >> >> > >> >> >> >> >> >> >> >> bridge with input_callback and did not return success. >> >> >> >> >> >> >> >> On Thu, Feb 9, 2012 at 5:28 PM, Srini K >> >> >> >> wrote: >> >> >> >> > Hi, >> >> >> >> > I have configured FreeSWITCH to receive DTMF by >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > in both sip_profile internal and external files. >> >> >> >> > >> >> >> >> > Iam using mod managed and I have subscribed to receive DTMF >> >> >> >> > events. >> >> >> >> > I can process DTMF from Caller without any problem. When I >> >> >> >> > receive >> >> >> >> > DTMF >> >> >> >> > from >> >> >> >> > the callee as Sip Info, I do receive the DTMF event from the >> >> >> >> > FreeSWITCH >> >> >> >> > and >> >> >> >> > immediately FreeSWITCH disconnects the call. >> >> >> >> > >> >> >> >> > Snapshot of the log is... >> >> >> >> > >> >> >> >> > [DEBUG] sofia.c:7229 INFO DTMF(1) >> >> >> >> > [DEBUG] sofia.c:7307 dispatched freeswitch event for INFO >> >> >> >> > [DEBUG] switch_core_session.c:875 Send signal >> >> >> >> > sofia/external/yyyyyyyyyy >> >> >> >> > [BREAK] >> >> >> >> > [DEBUG] switch_ivr_bridge.c:383 sofia/external/yyyyyyyyyy ended >> >> >> >> > call >> >> >> >> > via >> >> >> >> > DTMF >> >> >> >> > [DEBUG] switch_ivr_bridge.c:384 Send signal >> >> >> >> > sofia/external/xxxxxxxxxx at 10.10.2.3 [BREAK] >> >> >> >> > [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE >> >> >> >> > [sofia/external/yyyyyyyyyy] >> >> >> >> > >> >> >> >> > Whether Iam missing anything in the config? >> >> >> >> > >> >> >> >> > Thanks in advance. >> >> >> >> > 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 >> >> >> >> > >> >> >> >> > FreeSWITCH-users mailing list >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org >> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > UNSUBSCRIBE:http://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 >> >> >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> >> http://www.freeswitch.org >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > _________________________________________________________________________ >> >> >> > Professional FreeSWITCH Consulting Services: >> >> >> > 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 >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> > >> >> >> > >> >> >> > UNSUBSCRIBE:http://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 >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ >> >> > Professional FreeSWITCH Consulting Services: >> >> > 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 >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > >> >> > UNSUBSCRIBE:http://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 >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > 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 >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 -------------- A non-text attachment was scrubbed... Name: debug.diff Type: application/octet-stream Size: 1179 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/9af3517e/attachment-0001.obj From robert.longfield at gmail.com Sat Feb 11 02:27:29 2012 From: robert.longfield at gmail.com (Robert) Date: Fri, 10 Feb 2012 23:27:29 +0000 (UTC) Subject: [Freeswitch-users] xmpp outbound caller ID Message-ID: I just setup FS to be able to make phone calls through Gtalk and I was wondering if there is a way to set the outbound caller ID? I've tried various forms of the following the dialplan effective_caller_id_number=${###} On the receiving phone the Google number still shows up. From holger at freyther.de Sat Feb 11 02:31:33 2012 From: holger at freyther.de (Holger Freyther) Date: Fri, 10 Feb 2012 23:31:33 +0000 (UTC) Subject: [Freeswitch-users] How to inspect the jitterbuffer? References: Message-ID: Anthony Minessale writes: > You are pushing RTP up against TDM, RTP can start and stop, TDM > cannot, When there is no RTP the TDM will carry on sending the > IDLE_FLAG byte over the wire. > > I made both the RTP stack and the TDM endpoint from scratch so you > should assume I may have some insight. What I am arguing with... is that if I say a word like 'two' and RTP looks a bit like this. RTP: Wall Time Seq.No Time 'Sound' 0.0 1 0 'T' 20ms* 2 160 'W' 40ms* 3 320 'O' ... some skew 100 100*160 'T' +20ms 101 101*160 'W' +20ms 102 102*160 'O' and on the remote it arrives as: 'TW(?)O'..'T'..'O' something is wrong with the playback speed/queue/buffer of the audio stack. All I ask for is (because I didn't see anything in the wiki about that): 1.) How do I check if the jitter buffer is enabled on a call. 2.) How do I check if arriving RTP packets arrived too late for the speed of the FreeTDM voice queue. More details: I started with a setup like this: RTP/AMR -> FreeSWITCH (sangoma transcoder) -> FreeTDM.. For debugging I changed it to: AMR -> FreeSWITCH -> RTP/PCMA -> FreeSWITCH -> FreeTDM. (SIP on 5060) (SIP on 5062) The idea was that the first FreeSWITCH has to deal with the jitter and the second receives a clean linear in wall clock time RTP stream. My sofia sip profile includes these params: The resulting output of the frontend FreeSWITCH is not linear though (so no idea what rtp-timer-name soft should do): But the resulting stream has packets like this: P.No Wall Time Codec 723 12.761045 G.711 PCMA, Seq=18533, Time=90720 731 12.841107 G.711 PCMA, Seq=18534, Time=90880 so somehow the expected arrival time was 17.78 but it ended up being 12.84 and I think FreeSWITCH/FreeTDM/STFU do not deal well with that. any ideas? holger * add a little jitter From anthony.minessale at gmail.com Sat Feb 11 03:02:55 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 10 Feb 2012 18:02:55 -0600 Subject: [Freeswitch-users] How to inspect the jitterbuffer? In-Reply-To: References: Message-ID: You are arrogantly refusing to cooperate with my questions (why are people so stubborn!!!!) My patience is limited when dealing with someone playing games like this intentionally ignoring all my questions then writing a whole new paragraph of other non-related statements ....... This is what causes people to be mean on forums and mailing lists and I am doing my best to hold on to my patience. I am asking you question to try and weed out what your problem is, not to argue with you in any way. I bet when the cable guy tells you to reboot your device, you only pretend to and tell him you did it....... Why are you adding the manual-rtp-bugs if you don't even know what they do? They are all documented in the header file RTP_BUG_SEND_LINEAR_TIMESTAMPS = (1 << 3), /* Our friends at Sonus get real mad when the timestamps are not in perfect sequence even during periods of silence. With this flag, we will only increment the timestamp when write packets even if they are eons apart. */ RTP_BUG_NEVER_SEND_MARKER = (1 << 5), /* Our friends at Sonus are on a roll, They also get easily dumbfounded by marker bits. This flag will never send any. Sheesh.... */ If your packets are out of order it means your network is saturated or your network stack on your box is overloaded. Are you taking packet captures from your box sending the packets or the one receiving of a switch in between or both? Somewhere you have a problem...... if you want to use a jitter buffer to fix it, you can but you need it on the receiving box by using the jitterbuffer_ms variable set to the desired size in ms 60-80 recommended. BEFORE you answer the call. once its up, at the cli you can do uuid_jitterbuffer debug:DEBUG to get live stats...... This mailing list thread has turned into a bug report about a bug with no data to even look at and I hate putting issues on this mailing list. ?Within a few hours this email will be 5 pages outside the top of my inbox and I will have a hard time keeping track of it. I think I write 3 emails a day begging people to keep issues on Jira. That way we can at least get some confirmation what version the FS is and all the other On Fri, Feb 10, 2012 at 5:31 PM, Holger Freyther wrote: > Anthony Minessale writes: > > >> You are pushing RTP up against TDM, RTP can start and stop, TDM >> cannot, When there is no RTP the TDM will carry on sending the >> IDLE_FLAG byte over the wire. >> >> I made both the RTP stack and the TDM endpoint from scratch so you >> should assume I may have some insight. > > What I am arguing with... is that if I say a word like 'two' and > RTP looks a bit like this. > > RTP: > Wall Time ? Seq.No ? Time ? 'Sound' > 0.0 ? ? ? ? 1 ? ? ? ?0 ? ? ? 'T' > 20ms* ? ? 2 ? ? ? ?160 ? ? ? 'W' > 40ms* ? ? 3 ? ? ? ?320 ? ? ? 'O' > ... > some skew ? 100 ? ? ?100*160 ?'T' > +20ms ? ? ? 101 ? ? ?101*160 ?'W' > +20ms ? ? ? 102 ? ? ?102*160 ?'O' > > > and on the remote it arrives as: > 'TW(?)O'..'T'..'O' > > something is wrong with the playback speed/queue/buffer of the > audio stack. All I ask for is (because I didn't see anything > in the wiki about that): > > 1.) How do I check if the jitter buffer is enabled on a call. > 2.) How do I check if arriving RTP packets arrived too late for > ? ?the speed of the FreeTDM voice queue. > > > More details: > I started with a setup like this: > > RTP/AMR -> FreeSWITCH (sangoma transcoder) -> FreeTDM.. > > For debugging I changed it to: > > AMR -> FreeSWITCH -> RTP/PCMA -> FreeSWITCH -> FreeTDM. > ? ? ? (SIP on 5060) ? ? ? ? ? ? (SIP on 5062) > > The idea was that the first FreeSWITCH has to deal with > the jitter and the second receives a clean linear in wall > clock time RTP stream. My sofia sip profile includes these > params: > ? ? > ? ? ? ?value="SEND_LINEAR_TIMESTAMPS|NEVER_SEND_MARKER|\n > ? ?IGNORE_DTMF_DURATION" /> > ? ? > > > The resulting output of the frontend FreeSWITCH is not > linear though (so no idea what rtp-timer-name soft should > do): > But the resulting stream has packets like this: > P.No ?Wall Time ? ? ? Codec > 723 12.761045 G.711 PCMA, Seq=18533, Time=90720 > 731 12.841107 G.711 PCMA, Seq=18534, Time=90880 > > so somehow the expected arrival time was 17.78 but it > ended up being 12.84 and I think FreeSWITCH/FreeTDM/STFU > do not deal well with that. > > > any ideas? > ?holger > > > > * add a little jitter > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 Sat Feb 11 03:54:49 2012 From: msc at freeswitch.org (Michael Collins) Date: Fri, 10 Feb 2012 16:54:49 -0800 Subject: [Freeswitch-users] xmpp outbound caller ID In-Reply-To: References: Message-ID: Does GTalk allow you to make per-call CID changes? -MC On Fri, Feb 10, 2012 at 3:27 PM, Robert wrote: > I just setup FS to be able to make phone calls through Gtalk and I was > wondering > if there is a way to set the outbound caller ID? > > I've tried various forms of the following the dialplan > > effective_caller_id_number=${###} > > On the receiving phone the Google number still shows 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/6dbb2b6f/attachment.html From aksrini at hotmail.com Sat Feb 11 04:20:45 2012 From: aksrini at hotmail.com (Srini K) Date: Fri, 10 Feb 2012 17:20:45 -0800 Subject: [Freeswitch-users] FreeSWITCH ends the calls after receiving DTMF from the Callee. In-Reply-To: References: , , , , , , , , , Message-ID: Anthony,I do see the two prints as... [CRIT] switch_cpp.cpp:697 ASSIGN INPUT CALLBACK 003F2559 [CRIT] switch_cpp.cpp:1274 BRIDGE input callback func: 003F2559 Complete debug log is attached and I have highlighted them.http://pastebin.freeswitch.org/18362 ThanksSrini Date: Fri, 10 Feb 2012 17:46:13 -0600 From: anthony.minessale at gmail.com To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after receiving DTMF from the Callee. Try this patch to print more debug info. It seems somewhere in the managed code something is assigning an input callback and its being executed and returns false which makes the call end. On Fri, Feb 10, 2012 at 2:13 PM, Srini K wrote: > > The Lua script works fine. No problem in capturing the DTMF from both the > call legs. > > Only in Mod_Managed I have issue. Its simple 3 lines of code nothing > fancy... > > public void Run(AppContext context) { > context.Session.Answer(); > var newSession = new ManagedSession("sofia/gateway/foo/callenumber"); > freeswitch.bridge(newSession, context.Session); > } > > FreeSWITCH version is > FreeSWITCH Version 1.0.head (git-f477404 2012-02-09 11-08-52 -0600) > > Thanks > Srini >> Date: Fri, 10 Feb 2012 12:32:03 -0600 > >> From: anthony.minessale at gmail.com >> To: freeswitch-users at lists.freeswitch.org >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after receiving >> DTMF from the Callee. >> >> I tried this lua script which should be the same thing: >> >> b_leg = freeswitch.Session("sofia/internal/foo at bar.com", session); >> freeswitch.bridge(session, b_leg); >> >> See if you get the same problem running this lua code, I was passing dtmf >> fine. >> >> >> Internally, the code should not even use a callback capable of ending >> the call as you described unless your program has an input callback of >> dtmf handler function. >> >> >> Also make sure you are on a recent build of FS. >> >> >> >> On Fri, Feb 10, 2012 at 12:21 PM, Srini K wrote: >> > Anthony, >> > Thanks for your response. >> > I have not defined any of these input_callback or dtmf_callback. >> > >> > I have attached the log with debug level. >> > http://pastebin.freeswitch.org/18360 >> > >> > Thanks in advance. >> > >> > Regards >> > Srini >> > >> > >> > >> >> Date: Thu, 9 Feb 2012 19:06:39 -0600 >> > >> >> From: anthony.minessale at gmail.com >> >> To: freeswitch-users at lists.freeswitch.org >> >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after >> >> receiving >> >> DTMF from the Callee. >> >> >> >> do you have input_callback or dtmf_callback or anything defined? >> >> >> >> On Thu, Feb 9, 2012 at 6:37 PM, Srini K wrote: >> >> > Iam using the latest FreeSWITCh version (updated today morning). >> >> > >> >> > FS terminates on receiving any DTMF digits from the callee. I have >> >> > not >> >> > set >> >> > bridge_terminate_key. >> >> > >> >> > I have created an oubound session and bridging the inbound and >> >> > oubound >> > >> >> > session. Code snippet is >> >> > >> >> > var session = new ManagedSession("{origination_caller_id_number=" + >> >> > callerIdNumber + ",originate_timeout=8" + >> >> > "}sofia/gateway/408xxxxyyyy"); >> >> > string outBoundUuid = session.GetVariable("uuid"); >> >> > if (string.IsNullOrEmpty(outBoundUuid)) >> >> > { // Log error; >> >> > return; >> >> > } >> >> > freeswitch.bridge(inboundSession, session); >> >> > >> >> > Also I have tried setting hangup_after_bridge=false. >> >> > >> >> > Regards >> >> > Srini >> >> > >> >> > >> >> > >> >> >> Date: Thu, 9 Feb 2012 18:17:45 -0600 >> >> > >> >> >> From: anthony.minessale at gmail.com >> >> >> To: freeswitch-users at lists.freeswitch.org >> >> >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after >> >> >> receiving >> >> >> DTMF from the Callee. >> >> >> >> >> >> the obvious reason would be setting bridge_terminate_key=1 >> >> >> but I don't know where your bridge is being started based on your >> >> >> description or what version of the code you are using. >> >> >> >> >> >> On Thu, Feb 9, 2012 at 6:06 PM, Srini K wrote: >> >> >> > Any reason that it didn't return success. Anything Iam missing it >> >> >> > out. >> >> >> > >> >> >> > >> >> >> >> Date: Thu, 9 Feb 2012 17:33:01 -0600 >> >> >> >> From: anthony.minessale at gmail.com >> >> >> >> To: freeswitch-users at lists.freeswitch.org >> >> >> >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after >> >> >> >> receiving >> >> >> >> DTMF from the Callee. >> >> >> > >> >> >> >> >> >> >> >> bridge with input_callback and did not return success. >> >> >> >> >> >> >> >> On Thu, Feb 9, 2012 at 5:28 PM, Srini K >> >> >> >> wrote: >> >> >> >> > Hi, >> >> >> >> > I have configured FreeSWITCH to receive DTMF by >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > in both sip_profile internal and external files. >> >> >> >> > >> >> >> >> > Iam using mod managed and I have subscribed to receive DTMF >> >> >> >> > events. >> >> >> >> > I can process DTMF from Caller without any problem. When I >> >> >> >> > receive >> >> >> >> > DTMF >> >> >> >> > from >> >> >> >> > the callee as Sip Info, I do receive the DTMF event from the >> >> >> >> > FreeSWITCH >> >> >> >> > and >> >> >> >> > immediately FreeSWITCH disconnects the call. >> >> >> >> > >> >> >> >> > Snapshot of the log is... >> >> >> >> > >> >> >> >> > [DEBUG] sofia.c:7229 INFO DTMF(1) >> >> >> >> > [DEBUG] sofia.c:7307 dispatched freeswitch event for INFO >> >> >> >> > [DEBUG] switch_core_session.c:875 Send signal >> >> >> >> > sofia/external/yyyyyyyyyy >> >> >> >> > [BREAK] >> >> >> >> > [DEBUG] switch_ivr_bridge.c:383 sofia/external/yyyyyyyyyy ended >> >> >> >> > call >> >> >> >> > via >> >> >> >> > DTMF >> >> >> >> > [DEBUG] switch_ivr_bridge.c:384 Send signal >> >> >> >> > sofia/external/xxxxxxxxxx at 10.10.2.3 [BREAK] >> >> >> >> > [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE >> >> >> >> > [sofia/external/yyyyyyyyyy] >> >> >> >> > >> >> >> >> > Whether Iam missing anything in the config? >> >> >> >> > >> >> >> >> > Thanks in advance. >> >> >> >> > 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 >> >> >> >> > >> >> >> >> > FreeSWITCH-users mailing list >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org >> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> > UNSUBSCRIBE:http://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 >> >> >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> >> http://www.freeswitch.org >> >> >> > >> >> >> > >> >> >> > >> >> >> > >> >> >> > _________________________________________________________________________ >> >> >> > Professional FreeSWITCH Consulting Services: >> >> >> > 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 >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> > >> >> >> > >> >> >> > UNSUBSCRIBE:http://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 >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> >> FreeSWITCH-users at lists.freeswitch.org >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> >> >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> >> http://www.freeswitch.org >> >> > >> >> > >> >> > >> >> > _________________________________________________________________________ >> >> > Professional FreeSWITCH Consulting Services: >> >> > 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 >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > >> >> > UNSUBSCRIBE:http://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 >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > 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 >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/31c8f3da/attachment-0001.html From aromberg at gmail.com Sat Feb 11 04:37:31 2012 From: aromberg at gmail.com (aromberg at gmail.com) Date: Fri, 10 Feb 2012 19:37:31 -0600 Subject: [Freeswitch-users] Choppy Audio Recordings In-Reply-To: References: <9D59C823-7B8A-451F-8CDD-D5F46335903E@freeswitch.org> <1FFF97C269757C458224B7C895F35F1503BA1D@cantor.std.visionutv.se> Message-ID: GIT head as of the 7th or so. I'll update and report back to make sure I am on the latest. On Fri, Feb 10, 2012 at 11:43 AM, Anthony Minessale wrote: > Are you on very latest FS code? ?We did a large refactoring of > recording code recently and it was just complete last week sometime. > > > On Thu, Feb 9, 2012 at 11:05 PM, ? wrote: >>> I'm not 100% sure how your setup is. Are you running FS in the actual Host OS, or in a virtual (Hyper-V) machine? If running FS virtual > you will get these kinds of troubles, beacuse of not so accurate timing in those setups. >>> If you want things to work without problems, use real hardware. >> >> It's on a Hyper-V machine. ?I would have believed your timing >> reason/excuse if the issue would have persisted on the Hyper-V box and >> not on the CentOS6 machine (which is a dedicated non-virtualized >> machine). ?However, GV works near perfect (~99% with some small blips) >> and anything routed thru mod_sofia is choppy on the Hyper-V box, and >> the same is also true on the CentOS6 box. >> >> I receive perfect audio on my endpoint, it's just getting that audio >> written to the disk that's causing me the headache! >> >> For reference, here are my setups again: >> >> #1) Windows 2008 R2 x64 Dual Xeon 2.8, 4G of ram, Hyper-V instance, IIS >> running a few sites, nothing major. >> #2) Debian 6, dual core 2.13GHz Xeon, 512mb dedicated 1gb burst, OpenVZ >> instance, freeswitch + small apache instance >> #3) CentOS 5, 8 core 2.27 Xeon, 512MB RAM, Xen instance, freeswitch + >> apache (doing nothing) >> #4) Centos6, single core 3.33Ghz 512MB ram, dedicated machine (no >> instance), nohz setting enabled per the wiki, only freeswitch running >> >> 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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Feb 11 07:16:03 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 10 Feb 2012 22:16:03 -0600 Subject: [Freeswitch-users] FreeSWITCH ends the calls after receiving DTMF from the Callee. In-Reply-To: References: Message-ID: This illustrates that something in your code or something close to it is setting the input callback. Maybe a lib you are using on top of managed? On Feb 10, 2012 7:21 PM, "Srini K" wrote: > Anthony, > I do see the two prints as... > > [CRIT] switch_cpp.cpp:697 ASSIGN INPUT CALLBACK 003F2559 > [CRIT] switch_cpp.cpp:1274 BRIDGE input callback func: 003F2559 > > Complete debug log is attached and I have highlighted them. > http://pastebin.freeswitch.org/18362 > > Thanks > Srini > > Date: Fri, 10 Feb 2012 17:46:13 -0600 > From: anthony.minessale at gmail.com > To: freeswitch-users at lists.freeswitch.org > Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after receiving > DTMF from the Callee. > > Try this patch to print more debug info. > > It seems somewhere in the managed code something is assigning an input > callback and its being executed and returns false which makes the call > end. > > On Fri, Feb 10, 2012 at 2:13 PM, Srini K wrote: > > > > The Lua script works fine. No problem in capturing the DTMF from both the > > call legs. > > > > Only in Mod_Managed I have issue. Its simple 3 lines of code nothing > > fancy... > > > > public void Run(AppContext context) { > > context.Session.Answer(); > > var newSession = new ManagedSession("sofia/gateway/foo/callenumber"); > > freeswitch.bridge(newSession, context.Session); > > } > > > > FreeSWITCH version is > > FreeSWITCH Version 1.0.head (git-f477404 2012-02-09 11-08-52 -0600) > > > > Thanks > > Srini > >> Date: Fri, 10 Feb 2012 12:32:03 -0600 > > > >> From: anthony.minessale at gmail.com > >> To: freeswitch-users at lists.freeswitch.org > >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after receiving > >> DTMF from the Callee. > >> > >> I tried this lua script which should be the same thing: > >> > >> b_leg = freeswitch.Session("sofia/internal/foo at bar.com", session); > >> freeswitch.bridge(session, b_leg); > >> > >> See if you get the same problem running this lua code, I was passing dtmf > >> fine. > >> > >> > >> Internally, the code should not even use a callback capable of ending > >> the call as you described unless your program has an input callback of > >> dtmf handler function. > >> > >> > >> Also make sure you are on a recent build of FS. > >> > >> > >> > >> On Fri, Feb 10, 2012 at 12:21 PM, Srini K wrote: > >> > Anthony, > >> > Thanks for your response. > >> > I have not defined any of these input_callback or dtmf_callback. > >> > > >> > I have attached the log with debug level. > >> > http://pastebin.freeswitch.org/18360 > >> > > >> > Thanks in advance. > >> > > >> > Regards > >> > Srini > >> > > >> > > >> > > >> >> Date: Thu, 9 Feb 2012 19:06:39 -0600 > >> > > >> >> From: anthony.minessale at gmail.com > >> >> To: freeswitch-users at lists.freeswitch.org > >> >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after > >> >> receiving > >> >> DTMF from the Callee. > >> >> > >> >> do you have input_callback or dtmf_callback or anything defined? > >> >> > >> >> On Thu, Feb 9, 2012 at 6:37 PM, Srini K wrote: > >> >> > Iam using the latest FreeSWITCh version (updated today morning). > >> >> > > >> >> > FS terminates on receiving any DTMF digits from the callee. I have > >> >> > not > >> >> > set > >> >> > bridge_terminate_key. > >> >> > > >> >> > I have created an oubound session and bridging the inbound and > >> >> > oubound > >> > > >> >> > session. Code snippet is > >> >> > > >> >> > var session = new ManagedSession("{origination_caller_id_number=" + > >> >> > callerIdNumber + ",originate_timeout=8" + > >> >> > "}sofia/gateway/408xxxxyyyy"); > >> >> > string outBoundUuid = session.GetVariable("uuid"); > >> >> > if (string.IsNullOrEmpty(outBoundUuid)) > >> >> > { // Log error; > >> >> > return; > >> >> > } > >> >> > freeswitch.bridge(inboundSession, session); > >> >> > > >> >> > Also I have tried setting hangup_after_bridge=false. > >> >> > > >> >> > Regards > >> >> > Srini > >> >> > > >> >> > > >> >> > > >> >> >> Date: Thu, 9 Feb 2012 18:17:45 -0600 > >> >> > > >> >> >> From: anthony.minessale at gmail.com > >> >> >> To: freeswitch-users at lists.freeswitch.org > >> >> >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after > >> >> >> receiving > >> >> >> DTMF from the Callee. > >> >> >> > >> >> >> the obvious reason would be setting bridge_terminate_key=1 > >> >> >> but I don't know where your bridge is being started based on your > >> >> >> description or what version of the code you are using. > >> >> >> > >> >> >> On Thu, Feb 9, 2012 at 6:06 PM, Srini K wrote: > >> >> >> > Any reason that it didn't return success. Anything Iam missing it > >> >> >> > out. > >> >> >> > > >> >> >> > > >> >> >> >> Date: Thu, 9 Feb 2012 17:33:01 -0600 > >> >> >> >> From: anthony.minessale at gmail.com > >> >> >> >> To: freeswitch-users at lists.freeswitch.org > >> >> >> >> Subject: Re: [Freeswitch-users] FreeSWITCH ends the calls after > >> >> >> >> receiving > >> >> >> >> DTMF from the Callee. > >> >> >> > > >> >> >> >> > >> >> >> >> bridge with input_callback and did not return success. > >> >> >> >> > >> >> >> >> On Thu, Feb 9, 2012 at 5:28 PM, Srini K > >> >> >> >> wrote: > >> >> >> >> > Hi, > >> >> >> >> > I have configured FreeSWITCH to receive DTMF by > >> >> >> >> > > >> >> >> >> > > >> >> >> >> > > >> >> >> >> > > >> >> >> >> > in both sip_profile internal and external files. > >> >> >> >> > > >> >> >> >> > Iam using mod managed and I have subscribed to receive DTMF > >> >> >> >> > events. > >> >> >> >> > I can process DTMF from Caller without any problem. When I > >> >> >> >> > receive > >> >> >> >> > DTMF > >> >> >> >> > from > >> >> >> >> > the callee as Sip Info, I do receive the DTMF event from the > >> >> >> >> > FreeSWITCH > >> >> >> >> > and > >> >> >> >> > immediately FreeSWITCH disconnects the call. > >> >> >> >> > > >> >> >> >> > Snapshot of the log is... > >> >> >> >> > > >> >> >> >> > [DEBUG] sofia.c:7229 INFO DTMF(1) > >> >> >> >> > [DEBUG] sofia.c:7307 dispatched freeswitch event for INFO > >> >> >> >> > [DEBUG] switch_core_session.c:875 Send signal > >> >> >> >> > sofia/external/yyyyyyyyyy > >> >> >> >> > [BREAK] > >> >> >> >> > [DEBUG] switch_ivr_bridge.c:383 sofia/external/yyyyyyyyyy ended > >> >> >> >> > call > >> >> >> >> > via > >> >> >> >> > DTMF > >> >> >> >> > [DEBUG] switch_ivr_bridge.c:384 Send signal > >> >> >> >> > sofia/external/xxxxxxxxxx at 10.10.2.3 [BREAK] > >> >> >> >> > [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE > >> >> >> >> > [sofia/external/yyyyyyyyyy] > >> >> >> >> > > >> >> >> >> > Whether Iam missing anything in the config? > >> >> >> >> > > >> >> >> >> > Thanks in advance. > >> >> >> >> > 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 > >> >> >> >> > > >> >> >> >> > FreeSWITCH-users mailing list > >> >> >> >> > FreeSWITCH-users at lists.freeswitch.org > >> >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> >> >> > > >> >> >> >> > > >> >> >> >> > > >> >> >> >> > UNSUBSCRIBE:http://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 > >> >> >> >> > >> >> >> >> FreeSWITCH-users mailing list > >> >> >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> >> >> > >> >> >> >> > >> >> >> >> > >> >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> >> >> http://www.freeswitch.org > >> >> >> > > >> >> >> > > >> >> >> > > >> >> >> > > >> >> >> > _________________________________________________________________________ > >> >> >> > Professional FreeSWITCH Consulting Services: > >> >> >> > 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 > >> >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> >> > > >> >> >> > > >> >> >> > UNSUBSCRIBE:http://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 > >> >> >> > >> >> >> FreeSWITCH-users mailing list > >> >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> >> > >> >> >> > >> >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> >> http://www.freeswitch.org > >> >> > > >> >> > > >> >> > > >> >> > _________________________________________________________________________ > >> >> > Professional FreeSWITCH Consulting Services: > >> >> > 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 > >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > > >> >> > UNSUBSCRIBE:http://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 > >> >> > >> >> FreeSWITCH-users mailing list > >> >> FreeSWITCH-users at lists.freeswitch.org > >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> >> > >> >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> >> http://www.freeswitch.org > >> > > >> > > >> > _________________________________________________________________________ > >> > Professional FreeSWITCH Consulting Services: > >> > 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 > >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> > UNSUBSCRIBE:http://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 > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > 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 > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://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 FreeSWITCH-powered IP PBX: The CudaTel > Communication Server 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > 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URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120210/d8ebecd7/attachment-0001.html From robert.longfield at gmail.com Sat Feb 11 06:23:29 2012 From: robert.longfield at gmail.com (Robert) Date: Sat, 11 Feb 2012 03:23:29 +0000 (UTC) Subject: [Freeswitch-users] xmpp outbound caller ID References: Message-ID: Michael Collins writes: > > > Does GTalk allow you to make per-call CID changes? -MC hmm good question, I didn't think that they might not allow that. I'll have to do some checking. From holger at freyther.de Sat Feb 11 12:21:26 2012 From: holger at freyther.de (Holger Freyther) Date: Sat, 11 Feb 2012 09:21:26 +0000 (UTC) Subject: [Freeswitch-users] How to inspect the jitterbuffer? References: Message-ID: Anthony Minessale writes: > > You are arrogantly refusing to cooperate with my questions (why are > people so stubborn!!!!) I am sorry you have this impression. See, I am not coming here with "Please fix my problem, right now", the Sangoma techdesk helped me to verify the wanpipe/FreeTDM setup but the problem was not there. What I wanted and asked for is in the title of this email and in the body, I wanted to know how to inspect the jitter buffer as I couldn't find anything about that in the wiki. You were nice enough to answer that question in your last email. So thank you for that. > > Why are you adding the manual-rtp-bugs if you don't even know what they do? Well, I read the wiki page (otherwise I would not have found the options). For me knowing referred to actually know how this impacts switch_rtp.c and others. cheers holger From sherifomran2000 at yahoo.com Sat Feb 11 13:35:12 2012 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sat, 11 Feb 2012 02:35:12 -0800 (PST) Subject: [Freeswitch-users] FreeSwitch does not start in the background In-Reply-To: <4F35051C.7050801@belrosbank.by> Message-ID: <1328956512.2981.YahooMailClassic@web110814.mail.gq1.yahoo.com> try to see what FS says cd /usr/local/freeswitch/bin ./fs_cli If it does not connect, then you ve a problem in FS. It seems an XML is not correct --- On Fri, 2/10/12, Eugene Shcherbatyuk wrote: From: Eugene Shcherbatyuk Subject: [Freeswitch-users] FreeSwitch does not start in the background To: freeswitch-users at lists.freeswitch.org Date: Friday, February 10, 2012, 1:53 PM Hello, The purpose was to evaluate FreeSwitch and decide whether to replace my Asterisks with it or not. I installed FS1.0.6 from git on Ubuntu Server 10.04.3 LTS. Everything went smoothly until FS did not start with the system today. There are no errors in logs. FS starts in console fine. FS does not start in background mode nor from console (with -nc option) nor from init script. Even worse: FS can unexpectedly start (rarely) or do not start (often) with "backgrounding" message. Have a look at the console output below, please. > root at fs:/usr/local/freeswitch/bin# /etc/init.d/freeswitch start > # ******* FS started without a message ********* > root at fs:/usr/local/freeswitch/bin# netstat -ant > Active Internet connections (servers and established) > Proto Recv-Q Send-Q Local Address? Foreign Address State > tcp? ? ? ? 0 0 172.18.0.130:5060 0.0.0.0:* LISTEN > tcp? ? ? ? 0 0 127.0.0.1:8021? ? 0.0.0.0:* LISTEN > tcp? ? ? ? 0 0 0.0.0.0:22? ? ? ? 0.0.0.0:* LISTEN > tcp? ? ? ? 0 0 172.18.0.130:5080 0.0.0.0:* LISTEN > tcp6? ? ???0 0 ::1:5060? ? ? ? ? ? ???:::* LISTEN > tcp6? ? ???0 0 :::22? ? ? ? ? ? ? ? ? :::* LISTEN > root at fs:/usr/local/freeswitch/bin# /etc/init.d/freeswitch stop > Killing: 850 > root at fs:/usr/local/freeswitch/bin# /etc/init.d/freeswitch start > root at fs:/usr/local/freeswitch/bin# 923 Backgrounding. > # ****** Note message above ********* > root at fs:/usr/local/freeswitch/bin# netstat -ant > Active Internet connections (servers and established) > Proto Recv-Q Send-Q Local Address Foreign Address State > tcp? ? ? ? 0 0 0.0.0.0:22 0.0.0.0:* LISTEN > tcp6? ? ???0 0 :::22 :::* LISTEN Your opinions, please? Thank you in advance, Eugene ========================================================= ?????? ????????? ? ????? ???????? (???????????) ???????? ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? ?????????????. ========================================================= 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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120211/b9537f84/attachment.html From nasida at live.ru Sat Feb 11 13:47:53 2012 From: nasida at live.ru (Yuriy Nasida) Date: Sat, 11 Feb 2012 14:47:53 +0400 Subject: [Freeswitch-users] voicemail_say_phone_number In-Reply-To: References: , , <4F3060D2.7050505@earthspike.net>, , <6574CFF3DDE54C118E0E9250AC19EDD7@DWP>, <01b601cce54e$f865c250$e93146f0$@com>, <26ADC64FC37D4049A68787B3A0DCF4FA@DWP>, , , , , Message-ID: It helped. Thanks. Date: Fri, 10 Feb 2012 15:17:38 -0800 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] voicemail_say_phone_number The German xml file has poorly formatted umlaut characters. Try deleting the files in conf/lang/de/ and you should be fine. -MC On Fri, Feb 10, 2012 at 2:06 PM, Yuriy Nasida wrote: Michael, I have tried voicemail with latest git. Looks good, macro works now, thank you. But I see many WARNING messages in fs_cli when i try to check or leave voicemail. Not sure why. It appear when I don't use vm_announce_cid as well. '2012-02-11 00:49:20.169712 [WARNING] switch_xml.c:2329 Invalid UTF-8 character to ampersand, skip it' Thanks. Date: Tue, 7 Feb 2012 10:28:22 -0800 From: msc at freeswitch.org To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] voicemail_say_phone_number Okay, try latest git. I changed mod_voicemail to use the macro and I set vm_announce_cid to be true/false instead of using a file name. Try it with various caller ID number values and let us know how it goes. -MC On Tue, Feb 7, 2012 at 8:11 AM, Michael Collins wrote: Yeah, this was my bad. I started on this and got interrupted, then came back like an hour later and totally forgot about the fact that of mod_voicemail.c not using the macro. I think today I can do a quick test on that and if it works I'll commit it to git master and then you all can test it. Thanks, MC On Tue, Feb 7, 2012 at 2:22 AM, Yuriy Nasida wrote: Thank you guys! It works. The example with 'vm_announce_cid=ivr/ivr-this_is_a_call_from.wav' is more correct. Otherwise FS tries to play message true.wav :)So, voicemail module doesn't use macro "voicemail_say_phone_number" and as far as I see the source code of mod_voicemail.c explains this behaviour. Yes, ability for playing of some wav file if I will have vm from annonymous would be very useful. Probably I can add it independently but I believe that your modifying will more correctly :)Darcy please let me know if you plan to add this feature in the near future. Anyway thanks again! From: darcy at primrose.ws To: freeswitch-users at lists.freeswitch.org Date: Mon, 6 Feb 2012 23:27:23 -0500 Subject: Re: [Freeswitch-users] voicemail_say_phone_number If John?s works, use it, I could not make it work but the example I show below played the greeting before the clid, I have not added anything to play an annonymous greeting yet as I have not had time. in mod_voicemail.c you have the following code: if (!zstr(cbt->cid_number) && (vm_announce_cid = switch_channel_get_variable(channel, "vm_announce_cid"))) { switch_ivr_play_file(session, NULL, vm_announce_cid, NULL); switch_ivr_sleep(session, 500, SWITCH_TRUE, NULL); switch_ivr_say(session, cbt->cid_number, NULL, "name_spelled", "pronounced", NULL, NULL); } Which indicates you will play the file in variable ?vm_announce_cid?. Darcy In Sunny Ottawa Canada From: Bote Man Sent: Monday, February 06, 2012 11:14 PM To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] voicemail_say_phone_number Well, now I'm cornfused. The original example by John used vm_announce_cid as a Boolean switch. But you've tested it as a string that points to the desired sound file? I have no means to test this nor access to sources right now and I just added John's example to the wiki. I should back it out or correct it based on your results. Please advise. John Boteler Bote Communications in rainy Fort Lauderdale, FL From: Darcy Sent: Monday, 06 February, 2012 22:02 The message, this_is_a_call_from actually has to be set in the dialplan it appears, the fs plays the file set in vm_announce_cid, a simple dial plan below reflects one way of doing this, tested and it works. Needs more time to suit the total requirements, but this makes it a little more professional by adding the message in front of the number. Darcy ? On Mon, Feb 6, 2012 at 3:22 PM, John wrote: Michael, It does work, but it's a bit 'rough': all it does it speak the number just before the date. So I can understand why it was not documented... ? John PS. For those who want to know where to insert this, put the line marked with + into your conf/dialplan/default.xml file: + On 06/02/12 22:00, Michael Collins wrote: Yuriy, ?Please set vm_announce_cid to true prior to checking voicemail and see if it works. If it does, please let me know. If you can add it to the wiki then do so, otherwise one of our intrepid community members will do it. -MC 2012/2/6 Yuriy Nasida Hello list, I would like to have one simple ability to listen the phone number of caller when I check my voicemail. At present I listen date of message only. I have found the macro "voicemail_say_phone_number" in conf/lang/en/vm/sounds.xml. But... when FS uses it ? Can I get it without modifying of source code of voicemail module ? Please advise. Thanks. _________________________________________________________________________ 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 FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: 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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120211/93a9a008/attachment-0001.html From garbytrash at gmail.com Sat Feb 11 14:12:01 2012 From: garbytrash at gmail.com (Zenny) Date: Sat, 11 Feb 2012 12:12:01 +0100 Subject: [Freeswitch-users] life360 feature with FreeSWITCH?? Message-ID: Hi: I am just wondering whether life360.com kind of feature be achieved with FreeSWITCH? Any inputs like tracking family members or group members in a team using GPS? It is important in order to protect privacy from handing over to third parties rather create an inhouse system like life360? Any inputs to achieve the above including relevant modules will be appreciated? PS: How does FreeSWITCH fares behind a NAT now? Thanks! /z From b2m at a-cti.com Sat Feb 11 17:23:44 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Sat, 11 Feb 2012 19:53:44 +0530 Subject: [Freeswitch-users] Skype module killing Message-ID: Skype module crashed the FS freeswitch at internal> load mod_skypopen +OK Reloading XML -ERR [module load file routine returned an error] 2012-02-11 14:18:55.142045 [INFO] mod_pocketsphinx.c:482 PocketSphinx Reloaded freeswitch at internal> 2012-02-11 14:18:55.142045 [INFO] mod_enum.c:812 ENUM Reloaded 2012-02-11 14:18:55.142045 [INFO] switch_time.c:1035 Timezone reloaded 530 definitions 2012-02-11 14:18:55.162036 [WARNING] mod_skypopen.c:1738 [406c21c|1b4c78b] [WARNINGA 1738 ][skype101 ][IDLE,IDLE] STARTING interface_id=1 2012-02-11 14:18:55.162036 [ERR] skypopen_protocol.c:1549 [406c21c|1b4c78b] [ERRORA 1549 ][none ][N/A,N/A] Received error code 5 from X Server 2012-02-11 14:18:55.162036 [ERR] skypopen_protocol.c:1550 [406c21c|1b4c78b] [ERRORA 1550 ][none ][N/A,N/A] Display error for 1, :101 2012-02-11 14:18:55.162036 [ERR] skypopen_protocol.c:1747 [406c21c|1b4c78b] [ERRORA 1747 ][skype101 ][DEAD,IDLE] Fatal display error for :101 - successed to jump 2012-02-11 14:18:55.162036 [WARNING] skypopen_protocol.c:1772 [406c21c|1b4c78b] [WARNINGA 1772 ][skype101 ][DEAD,IDLE] Removing skype interface #skype101 2012-02-11 14:18:55.362032 [NOTICE] mod_skypopen.c:1768 [406c21c|1b4c78b] [NOTICA 1768 ][skype101 ][DEAD,IDLE] WAITING roughly 10 seconds to find a running Skype client and connect to its SKYPE API for interface_id=1 2012-02-11 14:19:05.362034 [ERR] mod_skypopen.c:1781 [406c21c|1b4c78b] [ERRORA 1781 ][skype101 ][DEAD,IDLE] Failed to connect to a SKYPE API for interface_id=1, no SKYPE client running, please (re)start Skype client. Skypopen exiting 2012-02-11 14:19:05.362034 [CRIT] switch_loadable_module.c:1281 Error Loading module /usr/local/freeswitch/mod/mod_skypopen.so **Module load routine returned an error** freeswitch at internal> load mod_skypopen Socket interrupted, bye! 5 XSELINUXs still allocated at reset SCREEN: 0 objects of 168 bytes = 0 total bytes 0 private allocs DEVICE: 4 objects of 48 bytes = 192 total bytes 0 private allocs CLIENT: 0 objects of 160 bytes = 0 total bytes 0 private allocs WINDOW: 0 objects of 32 bytes = 0 total bytes 0 private allocs PIXMAP: 1 objects of 16 bytes = 16 total bytes 0 private allocs GC: 0 objects of 56 bytes = 0 total bytes 0 private allocs CURSOR: 0 objects of 8 bytes = 0 total bytes 0 private allocs CURSOR_BITS: 0 objects of 8 bytes = 0 total bytes 0 private allocs DBE_WINDOW: 0 objects of 24 bytes = 0 total bytes 0 private allocs TOTAL: 5 objects, 208 bytes, 0 allocs 4 DEVICEs still allocated at reset DEVICE: 4 objects of 48 bytes = 192 total bytes 0 private allocs CLIENT: 0 objects of 160 bytes = 0 total bytes 0 private allocs WINDOW: 0 objects of 32 bytes = 0 total bytes 0 private allocs PIXMAP: 1 objects of 16 bytes = 16 total bytes 0 private allocs GC: 0 objects of 56 bytes = 0 total bytes 0 private allocs CURSOR: 0 objects of 8 bytes = 0 total bytes 0 private allocs CURSOR_BITS: 0 objects of 8 bytes = 0 total bytes 0 private allocs DBE_WINDOW: 0 objects of 24 bytes = 0 total bytes 0 private allocs TOTAL: 5 objects, 208 bytes, 0 allocs 1 PIXMAPs still allocated at reset PIXMAP: 1 objects of 16 bytes = 16 total bytes 0 private allocs GC: 0 objects of 56 bytes = 0 total bytes 0 private allocs CURSOR: 0 objects of 8 bytes = 0 total bytes 0 private allocs CURSOR_BITS: 0 objects of 8 bytes = 0 total bytes 0 private allocs DBE_WINDOW: 0 objects of 24 bytes = 0 total bytes 0 private allocs TOTAL: 1 objects, 16 bytes, 0 allocs Thanks, Bala -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120211/dc5c1bd9/attachment.html From rzumaeta at yahoo.com Sat Feb 11 17:32:26 2012 From: rzumaeta at yahoo.com (=?iso-8859-1?Q?Rodrigo_Andr=E9s_Zumaeta?=) Date: Sat, 11 Feb 2012 11:32:26 -0300 Subject: [Freeswitch-users] Buffer size sanity check failed! In-Reply-To: References: <672EBD90-192F-4907-B700-20BDAD8F7950@yahoo.com> Message-ID: <87BB1F33-B6A8-4A75-91C9-D03848B776A0@yahoo.com> Hi Anthony, I will update again when I get to be close to my FS box again, but I updated yesterday, is there anything new to update to? On Feb 10, 2012, at 8:28 PM, Anthony Minessale wrote: > update again and reproduce the sanity check err and send the new value > now with more info. > > > On Fri, Feb 10, 2012 at 3:57 PM, Rodrigo Andr?s Zumaeta > wrote: >> Ken, >> thanks for your reply, let's see: >> 1. there is a group of users using analog phones. >> 2. there is another group using digital phones. >> 3. there is a third group that is not relevant for this specific problems. >> >> All of these groups are connected to a Nortel, and we get H.323 traffic from them. The first group of users, the ones using analog phones, is the one reporting this problem. It happens with 100% of their calls. I am attaching a trace and console log... The only noticeable difference in the flow of the call (comparing to successful calls from digital phones) is the presence of CN RTP packets. >> >> >> >> >> >> >> >> >> >> On Feb 10, 2012, at 6:17 PM, Ken Rice wrote: >> >>> Can you provide more information on that group of users? Can you duplicate >>> this at will? ie: is it a specific hard phone or softphone using a specific >>> codec? >>> >>> Also console logs and network traces would be helpful... >>> >>> >>> On 2/10/12 1:24 PM, "Rodrigo Andr?s Zumaeta" wrote: >>> >>>> Hello, this is my first message to the list! >>>> when connecting one specific group of users to Freeswitch, I get this message: >>>> >>>> [CRIT] switch_core_codec.c:760 Buffer size sanity check failed! >>>> >>>> The call is then dropped. That is, on 1.0.6. When I upgrade to the latest git >>>> version, I still get the same message, but the call is not dropped anymore. >>>> However, I get only one way audio. >>>> >>>> As aforementioned, I just get this odd behavior with one group of users. >>>> >>>> I will be happy to provide any relevant information. >>>> >>>> >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> 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 >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> 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 >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From sat at calgaryit.com Sat Feb 11 18:24:24 2012 From: sat at calgaryit.com (George Sapak) Date: Sat, 11 Feb 2012 08:24:24 -0700 (MST) Subject: [Freeswitch-users] IVR greet-long repeat Message-ID: <585993538.2396.1328973864874.JavaMail.root@server3> any way to repeat the greet-long message a few time without user needing to press any buttons? Thank You, George From b2m at a-cti.com Sat Feb 11 19:10:39 2012 From: b2m at a-cti.com (Balamurugan Mahendran) Date: Sat, 11 Feb 2012 21:40:39 +0530 Subject: [Freeswitch-users] FS: handling Text messages Message-ID: Hi Team, Please guide me to the right direction for handling SMS, Service provider(bandwidhth) looks supporting sms. IS there any know solution to forward sms? Thanks, Bala -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120211/5505a864/attachment.html From errotan at elder.hu Sat Feb 11 21:33:35 2012 From: errotan at elder.hu (=?ISO-8859-1?Q?Pusk=E1s_Zsolt?=) Date: Sat, 11 Feb 2012 19:33:35 +0100 Subject: [Freeswitch-users] Skype module killing In-Reply-To: References: Message-ID: <4F36B47F.3050708@elder.hu> Please make a backtrace file of the crash and post a ticket on Jira. Read this page for more info: http://wiki.freeswitch.org/wiki/Reporting_Bugs 2012-02-11 15:23 keltez?ssel, Balamurugan Mahendran ?rta: > Skype module crashed the FS > > > freeswitch at internal> load mod_skypopen > +OK Reloading XML > -ERR [module load file routine returned an error] > > 2012-02-11 14:18:55.142045 [INFO] mod_pocketsphinx.c:482 PocketSphinx > Reloaded > freeswitch at internal> 2012-02-11 14:18:55.142045 [INFO] mod_enum.c:812 > ENUM Reloaded > 2012-02-11 14:18:55.142045 [INFO] switch_time.c:1035 Timezone reloaded > 530 definitions > 2012-02-11 14:18:55.162036 [WARNING] mod_skypopen.c:1738 > [406c21c|1b4c78b] [WARNINGA 1738 ][skype101 ][IDLE,IDLE] > STARTING interface_id=1 > 2012-02-11 14:18:55.162036 [ERR] skypopen_protocol.c:1549 > [406c21c|1b4c78b] [ERRORA 1549 ][none ][N/A,N/A] > Received error code 5 from X Server > > 2012-02-11 14:18:55.162036 [ERR] skypopen_protocol.c:1550 > [406c21c|1b4c78b] [ERRORA 1550 ][none ][N/A,N/A] > Display error for 1, :101 > 2012-02-11 14:18:55.162036 [ERR] skypopen_protocol.c:1747 > [406c21c|1b4c78b] [ERRORA 1747 ][skype101 ][DEAD,IDLE] > Fatal display error for :101 - successed to jump > 2012-02-11 14:18:55.162036 [WARNING] skypopen_protocol.c:1772 > [406c21c|1b4c78b] [WARNINGA 1772 ][skype101 ][DEAD,IDLE] > Removing skype interface #skype101 > 2012-02-11 14:18:55.362032 [NOTICE] mod_skypopen.c:1768 > [406c21c|1b4c78b] [NOTICA 1768 ][skype101 ][DEAD,IDLE] > WAITING roughly 10 seconds to find a running Skype client and connect > to its SKYPE API for interface_id=1 > 2012-02-11 14:19:05.362034 [ERR] mod_skypopen.c:1781 > [406c21c|1b4c78b] [ERRORA 1781 ][skype101 ][DEAD,IDLE] > Failed to connect to a SKYPE API for interface_id=1, no SKYPE client > running, please (re)start Skype client. Skypopen exiting > 2012-02-11 14:19:05.362034 [CRIT] switch_loadable_module.c:1281 Error > Loading module /usr/local/freeswitch/mod/mod_skypopen.so > **Module load routine returned an error** > > freeswitch at internal> load mod_skypopen > Socket interrupted, bye! > 5 XSELINUXs still allocated at reset > SCREEN: 0 objects of 168 bytes = 0 total bytes 0 private allocs > DEVICE: 4 objects of 48 bytes = 192 total bytes 0 private allocs > CLIENT: 0 objects of 160 bytes = 0 total bytes 0 private allocs > WINDOW: 0 objects of 32 bytes = 0 total bytes 0 private allocs > PIXMAP: 1 objects of 16 bytes = 16 total bytes 0 private allocs > GC: 0 objects of 56 bytes = 0 total bytes 0 private allocs > CURSOR: 0 objects of 8 bytes = 0 total bytes 0 private allocs > CURSOR_BITS: 0 objects of 8 bytes = 0 total bytes 0 private allocs > DBE_WINDOW: 0 objects of 24 bytes = 0 total bytes 0 private allocs > TOTAL: 5 objects, 208 bytes, 0 allocs > 4 DEVICEs still allocated at reset > DEVICE: 4 objects of 48 bytes = 192 total bytes 0 private allocs > CLIENT: 0 objects of 160 bytes = 0 total bytes 0 private allocs > WINDOW: 0 objects of 32 bytes = 0 total bytes 0 private allocs > PIXMAP: 1 objects of 16 bytes = 16 total bytes 0 private allocs > GC: 0 objects of 56 bytes = 0 total bytes 0 private allocs > CURSOR: 0 objects of 8 bytes = 0 total bytes 0 private allocs > CURSOR_BITS: 0 objects of 8 bytes = 0 total bytes 0 private allocs > DBE_WINDOW: 0 objects of 24 bytes = 0 total bytes 0 private allocs > TOTAL: 5 objects, 208 bytes, 0 allocs > 1 PIXMAPs still allocated at reset > PIXMAP: 1 objects of 16 bytes = 16 total bytes 0 private allocs > GC: 0 objects of 56 bytes = 0 total bytes 0 private allocs > CURSOR: 0 objects of 8 bytes = 0 total bytes 0 private allocs > CURSOR_BITS: 0 objects of 8 bytes = 0 total bytes 0 private allocs > DBE_WINDOW: 0 objects of 24 bytes = 0 total bytes 0 private allocs > TOTAL: 1 objects, 16 bytes, 0 allocs > > Thanks, > Bala > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120211/7402ab67/attachment-0001.html From errotan at elder.hu Sat Feb 11 21:54:50 2012 From: errotan at elder.hu (=?ISO-8859-1?Q?Pusk=E1s_Zsolt?=) Date: Sat, 11 Feb 2012 19:54:50 +0100 Subject: [Freeswitch-users] unable to compile mod_shout on latest git In-Reply-To: References: Message-ID: <4F36B97A.60603@elder.hu> According to packages.ubuntu.com the libvorbis.la file is located at /usr/lib/i386-linux-gnu/libvorbis.la or /usr/lib/x86_64-linux-gnu/libvorbis.la . But the posted log shows: /usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la => /usr/lib/libvorbis.la so it's not the right path for the file. On debian 6 "squeeze" amd64 it compiles without problems if libvorbis-dev is installed: root at home:/usr/src/freeswitch/src/mod/formats/mod_shout/.libs# ldd mod_shout.so linux-vdso.so.1 => (0x00007fffd73ff000) libz.so.1 => /usr/lib/libz.so.1 (0x00007f38e34d3000) libfreeswitch.so.1 => /usr/src/freeswitch/.libs/libfreeswitch.so.1 (0x00007f38e309e000) libvorbis.so.0 => /usr/lib/libvorbis.so.0 (0x00007f38e2e71000) libm.so.6 => /lib/libm.so.6 (0x00007f38e2bef000) libogg.so.0 => /usr/lib/libogg.so.0 (0x00007f38e29e9000) libpthread.so.0 => /lib/libpthread.so.0 (0x00007f38e27cc000) libc.so.6 => /lib/libc.so.6 (0x00007f38e246a000) libdl.so.2 => /lib/libdl.so.2 (0x00007f38e2266000) libcrypt.so.1 => /lib/libcrypt.so.1 (0x00007f38e202e000) librt.so.1 => /lib/librt.so.1 (0x00007f38e1e26000) libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f38e1bd1000) libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8 (0x00007f38e182f000) libncurses.so.5 => /lib/libncurses.so.5 (0x00007f38e15e9000) libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f38e12d5000) libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f38e10be000) /lib64/ld-linux-x86-64.so.2 (0x00007f38e39c2000) 2012-02-09 20:21 keltez?ssel, Michael Collins ?rta: > Doh! I didn't see your other email. Evidently it's not finding your > libvorbis for some reason. Can someone who is familiar with these > kinds of issues on Debian-based systems chime in? > > -MC > > On Thu, Feb 9, 2012 at 4:22 AM, Neil Patel > wrote: > > I'm on Ubuntu LTS and just did a pull from git. I've installed all > the dependencies > , > and am getting this error: > > making all mod_shout > Compiling /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c... > quiet_libtool: compile: gcc > -I/usr/src/freeswitch/libs/libshout-2.2.2/include > -I/usr/src/freeswitch/libs/lame-3.98.4/include > -I/usr/src/freeswitch/libs/mpg123-1.13.2/src > -I/usr/src/freeswitch/libs/mpg123-1.13.2/src > -I/usr/src/freeswitch/libs/libshout-2.2.2/include > -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 -g -O2 -Wall -std=c99 -pedantic > -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c -fPIC > -DPIC -o .libs/mod_shout.o > quiet_libtool: compile: gcc > -I/usr/src/freeswitch/libs/libshout-2.2.2/include > -I/usr/src/freeswitch/libs/lame-3.98.4/include > -I/usr/src/freeswitch/libs/mpg123-1.13.2/src > -I/usr/src/freeswitch/libs/mpg123-1.13.2/src > -I/usr/src/freeswitch/libs/libshout-2.2.2/include > -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 -g -O2 -Wall -std=c99 -pedantic > -Wdeclaration-after-statement -D_GNU_SOURCE -DHAVE_CONFIG_H -c > /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c -o > mod_shout.o >/dev/null 2>&1 > Creating mod_shout.la... > quiet_libtool: link: cannot find the library > `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la > ' or unhandled argument > `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la > ' > make[5]: *** [mod_shout.la ] Error 1 > make[4]: *** [all] Error 1 > make[3]: *** [mod_shout-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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120211/a03e6296/attachment.html From mario_fs at mgtech.com Sun Feb 12 01:10:55 2012 From: mario_fs at mgtech.com (Mario G) Date: Sat, 11 Feb 2012 14:10:55 -0800 Subject: [Freeswitch-users] Anyone have info on multicast paging FS and Linksys SPA phones Message-ID: Scoured web sites, traces, etc. for a year no luck. Anyone out there with some info? Can't get paging to work on Linksys SPA962s, I have check the multicast address and it matches FS. Here is what happens: If I connect my old Linksys SPA9000 PBX and power it on, then turn it off, FreeSwitch paging works on all the phone! If a phone is powered off, FreeSwitch paging no longer works on that phone. Power on/off SPA9000 and all is well. Anyone out there have it working that could tell me the secret? I am using *11 to page to trigger the rtp_multicast_page extension. Thanks in advance! MarioG -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120211/fef37a33/attachment.html From anthony.minessale at gmail.com Sun Feb 12 02:01:50 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 11 Feb 2012 17:01:50 -0600 Subject: [Freeswitch-users] unable to compile mod_shout on latest git In-Reply-To: <4F36B97A.60603@elder.hu> References: <4F36B97A.60603@elder.hu> Message-ID: IIRC there is a patch in the makefile for mod shout to strip out the vorbis depend completely there is a build bug in the lib where they offer a flag to disable forbid but then depend on it explicitly even with the flag set. On Feb 11, 2012 12:56 PM, "Pusk?s Zsolt" wrote: > > According to packages.ubuntu.com the libvorbis.la file is located at > /usr/lib/i386-linux-gnu/libvorbis.la or /usr/lib/x86_64-linux-gnu/ > libvorbis.la . > > But the posted log shows: > /usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la => > /usr/lib/libvorbis.la so it's not the right path for the file. > > On debian 6 "squeeze" amd64 it compiles without problems if libvorbis-dev > is installed: > > root at home:/usr/src/freeswitch/src/mod/formats/mod_shout/.libs# ldd > mod_shout.so > linux-vdso.so.1 => (0x00007fffd73ff000) > libz.so.1 => /usr/lib/libz.so.1 (0x00007f38e34d3000) > libfreeswitch.so.1 => /usr/src/freeswitch/.libs/libfreeswitch.so.1 > (0x00007f38e309e000) > libvorbis.so.0 => /usr/lib/libvorbis.so.0 (0x00007f38e2e71000) > libm.so.6 => /lib/libm.so.6 (0x00007f38e2bef000) > libogg.so.0 => /usr/lib/libogg.so.0 (0x00007f38e29e9000) > libpthread.so.0 => /lib/libpthread.so.0 (0x00007f38e27cc000) > libc.so.6 => /lib/libc.so.6 (0x00007f38e246a000) > libdl.so.2 => /lib/libdl.so.2 (0x00007f38e2266000) > libcrypt.so.1 => /lib/libcrypt.so.1 (0x00007f38e202e000) > librt.so.1 => /lib/librt.so.1 (0x00007f38e1e26000) > libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f38e1bd1000) > libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8 (0x00007f38e182f000) > libncurses.so.5 => /lib/libncurses.so.5 (0x00007f38e15e9000) > libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f38e12d5000) > libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f38e10be000) > /lib64/ld-linux-x86-64.so.2 (0x00007f38e39c2000) > > > 2012-02-09 20:21 keltez?ssel, Michael Collins ?rta: > > Doh! I didn't see your other email. Evidently it's not finding your > libvorbis for some reason. Can someone who is familiar with these kinds of > issues on Debian-based systems chime in? > > -MC > > On Thu, Feb 9, 2012 at 4:22 AM, Neil Patel wrote: > >> I'm on Ubuntu LTS and just did a pull from git. I've installed all the >> dependencies, >> and am getting this error: >> >> making all mod_shout >> Compiling /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c... >> quiet_libtool: compile: gcc >> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >> -I/usr/src/freeswitch/libs/lame-3.98.4/include >> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >> -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 >> -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE >> -DHAVE_CONFIG_H -c >> /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c -fPIC -DPIC -o >> .libs/mod_shout.o >> quiet_libtool: compile: gcc >> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >> -I/usr/src/freeswitch/libs/lame-3.98.4/include >> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >> -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 >> -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE >> -DHAVE_CONFIG_H -c >> /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c -o mod_shout.o >> >/dev/null 2>&1 >> Creating mod_shout.la... >> quiet_libtool: link: cannot find the library >> `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la' or >> unhandled argument `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib// >> libvorbis.la' >> make[5]: *** [mod_shout.la] Error 1 >> make[4]: *** [all] Error 1 >> make[3]: *** [mod_shout-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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/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 > > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120211/19174632/attachment-0001.html From anthony.minessale at gmail.com Sun Feb 12 02:04:22 2012 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Sat, 11 Feb 2012 17:04:22 -0600 Subject: [Freeswitch-users] How to inspect the jitterbuffer? In-Reply-To: References: Message-ID: Also do not set the rewrite timestamps on the upstream box or the jitterbuffer will be useless. You need to preserve the originals or the jitter cannot be unraveled. On Feb 11, 2012 3:27 AM, "Holger Freyther" wrote: > Anthony Minessale writes: > > > > > You are arrogantly refusing to cooperate with my questions (why are > > people so stubborn!!!!) > > I am sorry you have this impression. See, I am not coming here with > "Please fix my problem, right now", the Sangoma techdesk helped me > to verify the wanpipe/FreeTDM setup but the problem was not there. > > What I wanted and asked for is in the title of this email and in the > body, I wanted to know how to inspect the jitter buffer as I couldn't > find anything about that in the wiki. You were nice enough to answer > that question in your last email. So thank you for that. > > > > > Why are you adding the manual-rtp-bugs if you don't even know what they > do? > > Well, I read the wiki page (otherwise I would not have found the > options). For me knowing referred to actually know how this impacts > switch_rtp.c and others. > > cheers > holger > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120211/2c750682/attachment.html From gael at kelta.net Sun Feb 12 02:30:24 2012 From: gael at kelta.net (Gael Martin) Date: Sun, 12 Feb 2012 00:30:24 +0100 Subject: [Freeswitch-users] Access the Remote-Party-ID header from a SIP invite. Message-ID: Hi, I need to read the content of the Remote-Party-ID header in a SIP invite from a provide but I can't seem to find the right channel variable that has the content I tried: But it's always blank, even so I can see the Header in the SIP invite. Any ideas? Gael From gael at kelta.net Sun Feb 12 02:35:09 2012 From: gael at kelta.net (Gael Martin) Date: Sun, 12 Feb 2012 00:35:09 +0100 Subject: [Freeswitch-users] Access the Remote-Party-ID header from a SIP invite. In-Reply-To: References: Message-ID: Solved, I looked into the source and found the one I did not tried: sip_Remote-Party-ID Gael On Sun, Feb 12, 2012 at 12:30 AM, Gael Martin wrote: > Hi, I need to read the content of the Remote-Party-ID header in a SIP > invite from a provide but I can't seem to find the right channel > variable that has the content > > I tried: > > > > > > > But it's always blank, even so I can see the Header in the SIP invite. > > Any ideas? > > Gael From curriegrad2004 at gmail.com Sun Feb 12 04:18:58 2012 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sat, 11 Feb 2012 17:18:58 -0800 Subject: [Freeswitch-users] unable to compile mod_shout on latest git In-Reply-To: References: <4F36B97A.60603@elder.hu> Message-ID: Libshout doesn't even support vorbis on windows, let alone ogg. On 2012-02-11 3:03 PM, "Anthony Minessale" wrote: > IIRC there is a patch in the makefile for mod shout to strip out the > vorbis depend completely there is a build bug in the lib where they offer a > flag to disable forbid but then depend on it explicitly even with the flag > set. > On Feb 11, 2012 12:56 PM, "Pusk?s Zsolt" wrote: > >> >> According to packages.ubuntu.com the libvorbis.la file is located at >> /usr/lib/i386-linux-gnu/libvorbis.la or /usr/lib/x86_64-linux-gnu/ >> libvorbis.la . >> >> But the posted log shows: >> /usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la => >> /usr/lib/libvorbis.la so it's not the right path for the file. >> >> On debian 6 "squeeze" amd64 it compiles without problems if libvorbis-dev >> is installed: >> >> root at home:/usr/src/freeswitch/src/mod/formats/mod_shout/.libs# ldd >> mod_shout.so >> linux-vdso.so.1 => (0x00007fffd73ff000) >> libz.so.1 => /usr/lib/libz.so.1 (0x00007f38e34d3000) >> libfreeswitch.so.1 => /usr/src/freeswitch/.libs/libfreeswitch.so.1 >> (0x00007f38e309e000) >> libvorbis.so.0 => /usr/lib/libvorbis.so.0 (0x00007f38e2e71000) >> libm.so.6 => /lib/libm.so.6 (0x00007f38e2bef000) >> libogg.so.0 => /usr/lib/libogg.so.0 (0x00007f38e29e9000) >> libpthread.so.0 => /lib/libpthread.so.0 (0x00007f38e27cc000) >> libc.so.6 => /lib/libc.so.6 (0x00007f38e246a000) >> libdl.so.2 => /lib/libdl.so.2 (0x00007f38e2266000) >> libcrypt.so.1 => /lib/libcrypt.so.1 (0x00007f38e202e000) >> librt.so.1 => /lib/librt.so.1 (0x00007f38e1e26000) >> libssl.so.0.9.8 => /usr/lib/libssl.so.0.9.8 (0x00007f38e1bd1000) >> libcrypto.so.0.9.8 => /usr/lib/libcrypto.so.0.9.8 (0x00007f38e182f000) >> libncurses.so.5 => /lib/libncurses.so.5 (0x00007f38e15e9000) >> libstdc++.so.6 => /usr/lib/libstdc++.so.6 (0x00007f38e12d5000) >> libgcc_s.so.1 => /lib/libgcc_s.so.1 (0x00007f38e10be000) >> /lib64/ld-linux-x86-64.so.2 (0x00007f38e39c2000) >> >> >> 2012-02-09 20:21 keltez?ssel, Michael Collins ?rta: >> >> Doh! I didn't see your other email. Evidently it's not finding your >> libvorbis for some reason. Can someone who is familiar with these kinds of >> issues on Debian-based systems chime in? >> >> -MC >> >> On Thu, Feb 9, 2012 at 4:22 AM, Neil Patel wrote: >> >>> I'm on Ubuntu LTS and just did a pull from git. I've installed all the >>> dependencies, >>> and am getting this error: >>> >>> making all mod_shout >>> Compiling /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c... >>> quiet_libtool: compile: gcc >>> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >>> -I/usr/src/freeswitch/libs/lame-3.98.4/include >>> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >>> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >>> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >>> -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 >>> -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE >>> -DHAVE_CONFIG_H -c >>> /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c -fPIC -DPIC -o >>> .libs/mod_shout.o >>> quiet_libtool: compile: gcc >>> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >>> -I/usr/src/freeswitch/libs/lame-3.98.4/include >>> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >>> -I/usr/src/freeswitch/libs/mpg123-1.13.2/src >>> -I/usr/src/freeswitch/libs/libshout-2.2.2/include >>> -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 >>> -g -O2 -Wall -std=c99 -pedantic -Wdeclaration-after-statement -D_GNU_SOURCE >>> -DHAVE_CONFIG_H -c >>> /usr/src/freeswitch/src/mod/formats/mod_shout/mod_shout.c -o mod_shout.o >>> >/dev/null 2>&1 >>> Creating mod_shout.la... >>> quiet_libtool: link: cannot find the library >>> `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib//libvorbis.la' or >>> unhandled argument `/usr/lib/gcc/i486-linux-gnu/4.3.3/../../../../lib// >>> libvorbis.la' >>> make[5]: *** [mod_shout.la] Error 1 >>> make[4]: *** [all] Error 1 >>> make[3]: *** [mod_shout-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 >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/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 >> >> 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 >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120211/a4454f6e/attachment-0001.html From bdfoster at endigotech.com Sun Feb 12 04:25:32 2012 From: bdfoster at endigotech.com (Brian Foster) Date: Sat, 11 Feb 2012 20:25:32 -0500 Subject: [Freeswitch-users] xmpp outbound caller ID In-Reply-To: References: Message-ID: They don't. On Feb 11, 2012 12:35 AM, "Robert" wrote: > Michael Collins writes: > > > > > > > Does GTalk allow you to make per-call CID changes? -MC > > hmm good question, I didn't think that they might not allow that. > I'll have to do some checking. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120211/dc893a88/attachment.html From errotan at elder.hu Sun Feb 12 14:40:55 2012 From: errotan at elder.hu (=?ISO-8859-1?Q?Pusk=E1s_Zsolt?=) Date: Sun, 12 Feb 2012 12:40:55 +0100 Subject: [Freeswitch-users] Anyone have info on multicast paging FS and Linksys SPA phones In-Reply-To: References: Message-ID: <4F37A547.9050806@elder.hu> Hi. If you can get an old HUB ( http://en.wikipedia.org/wiki/Ethernet_hub ) you can capture the packets using wireshark between the spa962 and the spa9000. Then compare the packets between your spa962 and FS box. Put your call and sip trace from FS to pastebin so we can help to find the problem. Mail me the pcap file of the working and failed call if you can't find the differences between them. I tried with a SPA502G and also can't make it work. I added the route ( ip route add 224.168.168.168 dev eth0:0 src 192.168.1.5 ) but in my case the interface has 2 IP address and multicast is sent from the main IP ( eth0 ) and not from eth0:0. Maybe this is the problem for me. I try different config options on the phone to make it work and post results to the list and wikify them if needed. 2012-02-11 23:10 keltez?ssel, Mario G ?rta: > Scoured web sites, traces, etc. for a year no luck. Anyone out there > with some info? > > Can't get paging to work on Linksys SPA962s, I have check the > multicast address and it matches FS. Here is what happens: If I > connect my old Linksys SPA9000 PBX and power it on, then turn it off, > FreeSwitch paging works on all the phone! If a phone is powered off, > FreeSwitch paging no longer works on that phone. Power on/off SPA9000 > and all is well. Anyone out there have it working that could tell me > the secret? I am using *11 to page to trigger the rtp_multicast_page > extension. Thanks in advance! > MarioG > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120212/e48b9a4c/attachment.html From errotan at elder.hu Sun Feb 12 16:27:27 2012 From: errotan at elder.hu (=?ISO-8859-1?Q?Pusk=E1s_Zsolt?=) Date: Sun, 12 Feb 2012 14:27:27 +0100 Subject: [Freeswitch-users] Anyone have info on multicast paging FS and Linksys SPA phones In-Reply-To: <4F37A547.9050806@elder.hu> References: <4F37A547.9050806@elder.hu> Message-ID: <4F37BE3F.1000709@elder.hu> Ooops the problem was the "Linksys Key System" option is set to No changing to Yes makes it work. I can't verify it work fully because the phone is at an office and nobody is in today :) but the web interface shows an active call. Now the only problem is when the calling phone hangup the call the channel is not destroyed and FS keeps sending the multicast packets. I'm looking at the source code to find the problem. 2012-02-12 12:40 keltez?ssel, Pusk?s Zsolt ?rta: > Hi. > > If you can get an old HUB ( http://en.wikipedia.org/wiki/Ethernet_hub > ) you can capture the packets using wireshark between the spa962 and > the spa9000. Then compare the packets between your spa962 and FS box. > Put your call and sip trace from FS to pastebin so we can help to find > the problem. Mail me the pcap file of the working and failed call if > you can't find the differences between them. > > I tried with a SPA502G and also can't make it work. I added the route > ( ip route add 224.168.168.168 dev eth0:0 src 192.168.1.5 ) but in my > case the interface has 2 IP address and multicast is sent from the > main IP ( eth0 ) and not from eth0:0. Maybe this is the problem for > me. I try different config options on the phone to make it work and > post results to the list and wikify them if needed. > > > 2012-02-11 23:10 keltez?ssel, Mario G ?rta: >> Scoured web sites, traces, etc. for a year no luck. Anyone out there >> with some info? >> >> Can't get paging to work on Linksys SPA962s, I have check the >> multicast address and it matches FS. Here is what happens: If I >> connect my old Linksys SPA9000 PBX and power it on, then turn it off, >> FreeSwitch paging works on all the phone! If a phone is powered off, >> FreeSwitch paging no longer works on that phone. Power on/off SPA9000 >> and all is well. Anyone out there have it working that could tell me >> the secret? I am using *11 to page to trigger the rtp_multicast_page >> extension. Thanks in advance! >> MarioG >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120212/df2b643d/attachment.html From mario_fs at mgtech.com Sun Feb 12 20:03:26 2012 From: mario_fs at mgtech.com (Mario G) Date: Sun, 12 Feb 2012 09:03:26 -0800 Subject: [Freeswitch-users] Anyone have info on multicast paging FS and Linksys SPA phones In-Reply-To: <4F37BE3F.1000709@elder.hu> References: <4F37A547.9050806@elder.hu> <4F37BE3F.1000709@elder.hu> Message-ID: <29B6A252-6261-4D0E-AA82-9ED6B0EAB2D8@mgtech.com> Far out, I knew I wasn't crazy. I did run traces but I am not a SIP expert to see the difference. I thought I tried the "Linksys Key System" option yes and no. As I said, been trying to get this to work for a year. You're the first to help. Keep in mind FS will work but only if the SPA9000 is connected to the phones at least once. Thanks! On Feb 12, 2012, at 5:27 AM, Pusk?s Zsolt wrote: > > Ooops the problem was the "Linksys Key System" option is set to No changing to Yes makes it work. I can't verify it work fully because the phone is at an office and nobody is in today :) but the web interface shows an active call. Now the only problem is when the calling phone hangup the call the channel is not destroyed and FS keeps sending the multicast packets. I'm looking at the source code to find the problem. > > > 2012-02-12 12:40 keltez?ssel, Pusk?s Zsolt ?rta: >> >> Hi. >> >> If you can get an old HUB ( http://en.wikipedia.org/wiki/Ethernet_hub ) you can capture the packets using wireshark between the spa962 and the spa9000. Then compare the packets between your spa962 and FS box. Put your call and sip trace from FS to pastebin so we can help to find the problem. Mail me the pcap file of the working and failed call if you can't find the differences between them. >> >> I tried with a SPA502G and also can't make it work. I added the route ( ip route add 224.168.168.168 dev eth0:0 src 192.168.1.5 ) but in my case the interface has 2 IP address and multicast is sent from the main IP ( eth0 ) and not from eth0:0. Maybe this is the problem for me. I try different config options on the phone to make it work and post results to the list and wikify them if needed. >> >> >> 2012-02-11 23:10 keltez?ssel, Mario G ?rta: >>> >>> Scoured web sites, traces, etc. for a year no luck. Anyone out there with some info? >>> >>> Can't get paging to work on Linksys SPA962s, I have check the multicast address and it matches FS. Here is what happens: If I connect my old Linksys SPA9000 PBX and power it on, then turn it off, FreeSwitch paging works on all the phone! If a phone is powered off, FreeSwitch paging no longer works on that phone. Power on/off SPA9000 and all is well. Anyone out there have it working that could tell me the secret? I am using *11 to page to trigger the rtp_multicast_page extension. Thanks in advance! >>> MarioG >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> 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 >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120212/054c6aed/attachment-0001.html From aakashviswam at gmail.com Sun Feb 12 20:05:57 2012 From: aakashviswam at gmail.com (Aakash) Date: Sun, 12 Feb 2012 09:05:57 -0800 (PST) Subject: [Freeswitch-users] skipping vm and plays enter the pwd In-Reply-To: References: <1328730591696-7267135.post@n2.nabble.com> Message-ID: <1329066357159-7278154.post@n2.nabble.com> Hi MC, I have pasted the logs in the pastebin. http://pastebin.freeswitch.org/18371 Note: i have tried twice calling our inbound number ,the first time the call lands perfectly to the user recorded voicemail.On the second time its ask me to enter the password. Regards, Aakash -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/skipping-vm-and-plays-enter-the-pwd-tp7267135p7278154.html Sent from the freeswitch-users mailing list archive at Nabble.com. From shaon.khan at gmail.com Sun Feb 12 21:28:55 2012 From: shaon.khan at gmail.com (Arafath-uz-zaman khan) Date: Mon, 13 Feb 2012 00:28:55 +0600 Subject: [Freeswitch-users] setting higher ptime with sangoma transcoding card Message-ID: Hello All I am using FS with Sangoma transcoding card. Can anyone suggest me how can i user 60 or more ptime for G729 codec? -- Arafath-uz-zaman khan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120213/8d8c03d4/attachment.html From henrikaagaardsorensen at gmail.com Sun Feb 12 13:13:24 2012 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Sun, 12 Feb 2012 11:13:24 +0100 Subject: [Freeswitch-users] Rotate cdr-records as root user (when FS is running from freeswitch user). Message-ID: I'm having FS running from the user freeswitch on CentOS 6. When calling (as the user root): /usr/local/freeswitch/bin/fs_cli -x 'cdr_csv rotate' >/dev/null nothing happens. The cdr-records does not get rotated. Can anyone help me on why this is happening and how I can rotate the files as root? From wstephen80 at gmail.com Sun Feb 12 23:35:33 2012 From: wstephen80 at gmail.com (Stephen Wilde) Date: Sun, 12 Feb 2012 21:35:33 +0100 Subject: [Freeswitch-users] setting higher ptime with sangoma transcoding card In-Reply-To: References: Message-ID: The maximum supported ptime for G729 is 50ms. On Sun, Feb 12, 2012 at 7:28 PM, Arafath-uz-zaman khan wrote: > Hello All > > I am using FS with Sangoma transcoding card. Can anyone suggest me how can > i user 60 or more ptime for G729 codec? > > -- > Arafath-uz-zaman khan > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120212/10735f31/attachment.html From robert.longfield at gmail.com Sun Feb 12 23:57:02 2012 From: robert.longfield at gmail.com (Robert) Date: Sun, 12 Feb 2012 20:57:02 +0000 (UTC) Subject: [Freeswitch-users] xmpp outbound caller ID References: Message-ID: Brian Foster writes: > > > They don't. Thanks for the confirmation. Not the answer I was hoping for but it is an answer. From david at styleflare.com Mon Feb 13 00:53:29 2012 From: david at styleflare.com (David J) Date: Sun, 12 Feb 2012 16:53:29 -0500 Subject: [Freeswitch-users] xmpp outbound caller ID In-Reply-To: References: Message-ID: I guess if they let you do it then carriers could theoretically just send all traffic thru google. On Feb 12, 2012 4:47 PM, "Robert" wrote: > Brian Foster writes: > > > > > > > They don't. > > > Thanks for the confirmation. Not the answer I was hoping for but it is an > answer. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120212/c8bf6b6a/attachment.html From vetali100 at gmail.com Mon Feb 13 05:05:57 2012 From: vetali100 at gmail.com (Vitalie Colosov) Date: Sun, 12 Feb 2012 18:05:57 -0800 Subject: [Freeswitch-users] Rotate cdr-records as root user (when FS is running from freeswitch user). In-Reply-To: References: Message-ID: Ensure the following line is uncommented in cdr_csv.conf.xml To rotate, execute from commandline: killall -HUP freeswitch You can add this command to cron so it will be executed automatically. 2012/2/12 Henrik Aagaard S?rensen > I'm having FS running from the user freeswitch on CentOS 6. > > When calling (as the user root): > /usr/local/freeswitch/bin/fs_cli -x 'cdr_csv rotate' >/dev/null > nothing happens. The cdr-records does not get rotated. > > Can anyone help me on why this is happening and how I can rotate the > files as root? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120212/ea1b696e/attachment.html From mays.david at gmail.com Mon Feb 13 07:29:12 2012 From: mays.david at gmail.com (dma) Date: Sun, 12 Feb 2012 20:29:12 -0800 (PST) Subject: [Freeswitch-users] Question regarding "uuid_bridge needs at least one leg to be answered" Message-ID: <1329107352468-7279309.post@n2.nabble.com> Hi All, On the wiki page for uuid_bridge, there is following description: -------------------------------------- uuid_bridge Bridge two call legs together. Usage: uuid_bridge uuid_bridge needs atleast any one leg to be answered. -------------------------------------- Here, "uuid_bridge needs at least any one leg to be answered". Does this limit also apply to dialplan bridge and/or session:bridge in lua script? Thanks, D.Ma -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Question-regarding-uuid-bridge-needs-at-least-one-leg-to-be-answered-tp7279309p7279309.html Sent from the freeswitch-users mailing list archive at Nabble.com. From shcherbatyuk at belrosbank.by Mon Feb 13 09:22:20 2012 From: shcherbatyuk at belrosbank.by (Eugene Shcherbatyuk) Date: Mon, 13 Feb 2012 09:22:20 +0300 Subject: [Freeswitch-users] FreeSwitch does not start in the background -- bypassed In-Reply-To: References: <4F35051C.7050801@belrosbank.by> Message-ID: <4F38AC1C.7050701@belrosbank.by> Thank you for your answer! Well, I use the latest git and init script from debian directory in the sources. Obviously, the problem is not solved at its roots yet, but I managed to bypass it by 'chowning' FS directory and setting user to 'root' in init script. On 10/02/12 20:38, Anton Kvashenkin wrote: > The first suggestion would be to use latest git and init script from > http://wiki.freeswitch.org/wiki/Freeswitch_init. > > 10 ??????? 2012 ?. 15:53 ???????????? Eugene Shcherbatyuk > > ???????: > > Hello, > > The purpose was to evaluate FreeSwitch and decide whether to replace my > Asterisks with it or not. I installed FS1.0.6 from git on Ubuntu Server > 10.04.3 LTS. Everything went smoothly until FS did not start with the > system today. > > There are no errors in logs. FS starts in console fine. FS does not > start in background mode nor from console (with -nc option) nor from > init script. Even worse: FS can unexpectedly start (rarely) or do not > start (often) with "backgrounding" message. Have a look at the console > output below, please. > > > root at fs:/usr/local/freeswitch/bin# /etc/init.d/freeswitch start > > # ******* FS started without a message ********* > > root at fs:/usr/local/freeswitch/bin# netstat -ant > > Active Internet connections (servers and established) > > Proto Recv-Q Send-Q Local Address Foreign Address State > > tcp 0 0 172.18.0.130:5060 0.0.0.0:* LISTEN > > tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN > > tcp 0 0 0.0.0.0:22 0.0.0.0:* LISTEN > > tcp 0 0 172.18.0.130:5080 0.0.0.0:* LISTEN > > tcp6 0 0 ::1:5060 :::* LISTEN > > tcp6 0 0 :::22 :::* LISTEN > > root at fs:/usr/local/freeswitch/bin# /etc/init.d/freeswitch stop > > Killing: 850 > > root at fs:/usr/local/freeswitch/bin# /etc/init.d/freeswitch start > > root at fs:/usr/local/freeswitch/bin# 923 Backgrounding. > > # ****** Note message above ********* > > root at fs:/usr/local/freeswitch/bin# netstat -ant > > Active Internet connections (servers and established) > > Proto Recv-Q Send-Q Local Address Foreign Address State > > tcp 0 0 0.0.0.0:22 0.0.0.0:* LISTEN > > tcp6 0 0 :::22 :::* LISTEN > > Your opinions, please? > > Thank you in advance, > Eugene > ========================================================= ?????? ????????? ? ????? ???????? (???????????) ???????? ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? ?????????????. ========================================================= 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. ========================================================= From miha at softnet.si Mon Feb 13 11:13:06 2012 From: miha at softnet.si (Miha Zoubek) Date: Mon, 13 Feb 2012 09:13:06 +0100 Subject: [Freeswitch-users] setting variable Message-ID: <4F38C612.9050802@softnet.si> Hi, I need a little help. First I have set in my public dialplan one of variables to false. But for few calls I must have it on true but I must set this variable on true after the variable as already set on false ( I get information for condition further in dialplan). Problem is that when I set it on false I can not set it on true, as variable do net set. Paste bin. variable: disable_radius_stop=false http://pastebin.freeswitch.org/18377 Thanks!!! -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. From anton.jugatsu at gmail.com Mon Feb 13 14:59:41 2012 From: anton.jugatsu at gmail.com (Anton Kvashenkin) Date: Mon, 13 Feb 2012 15:59:41 +0400 Subject: [Freeswitch-users] FreeSwitch does not start in the background -- bypassed In-Reply-To: <4F38AC1C.7050701@belrosbank.by> References: <4F35051C.7050801@belrosbank.by> <4F38AC1C.7050701@belrosbank.by> Message-ID: ? ? ??? ?????? ????????, ????? ???????? ????????????, ??????????. ????? ??????? ?????????? ? ???????? FS ?????? ?? ????????? ?????? ??? ?????? ???? ????????. ???????? ????????? freeswitch -nc, ????? ??????? ?? pasterbin. 13 ??????? 2012 ?. 10:22 ???????????? Eugene Shcherbatyuk < shcherbatyuk at belrosbank.by> ???????: > Thank you for your answer! > Well, I use the latest git and init script from debian directory in the > sources. > > Obviously, the problem is not solved at its roots yet, but I managed to > bypass it by 'chowning' FS directory and setting user to 'root' in init > script. > > On 10/02/12 20:38, Anton Kvashenkin wrote: > > The first suggestion would be to use latest git and init script from > > http://wiki.freeswitch.org/wiki/Freeswitch_init. > > > > 10 ??????? 2012 ?. 15:53 ???????????? Eugene Shcherbatyuk > > > > ???????: > > > > Hello, > > > > The purpose was to evaluate FreeSwitch and decide whether to replace > my > > Asterisks with it or not. I installed FS1.0.6 from git on Ubuntu > Server > > 10.04.3 LTS. Everything went smoothly until FS did not start with the > > system today. > > > > There are no errors in logs. FS starts in console fine. FS does not > > start in background mode nor from console (with -nc option) nor from > > init script. Even worse: FS can unexpectedly start (rarely) or do not > > start (often) with "backgrounding" message. Have a look at the > console > > output below, please. > > > > > root at fs:/usr/local/freeswitch/bin# /etc/init.d/freeswitch start > > > # ******* FS started without a message ********* > > > root at fs:/usr/local/freeswitch/bin# netstat -ant > > > Active Internet connections (servers and established) > > > Proto Recv-Q Send-Q Local Address Foreign Address State > > > tcp 0 0 172.18.0.130:5060 0.0.0.0:* > LISTEN > > > tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN > > > tcp 0 0 0.0.0.0:22 0.0.0.0:* LISTEN > > > tcp 0 0 172.18.0.130:5080 0.0.0.0:* > LISTEN > > > tcp6 0 0 ::1:5060 :::* LISTEN > > > tcp6 0 0 :::22 :::* LISTEN > > > root at fs:/usr/local/freeswitch/bin# /etc/init.d/freeswitch stop > > > Killing: 850 > > > root at fs:/usr/local/freeswitch/bin# /etc/init.d/freeswitch start > > > root at fs:/usr/local/freeswitch/bin# 923 Backgrounding. > > > # ****** Note message above ********* > > > root at fs:/usr/local/freeswitch/bin# netstat -ant > > > Active Internet connections (servers and established) > > > Proto Recv-Q Send-Q Local Address Foreign Address State > > > tcp 0 0 0.0.0.0:22 0.0.0.0:* LISTEN > > > tcp6 0 0 :::22 :::* LISTEN > > > > Your opinions, please? > > > > Thank you in advance, > > Eugene > > > > > ========================================================= > > ?????? ????????? ? ????? ???????? (<>) ???????? > ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? > ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? > ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. > ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? > "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? > ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? > ?????????????. > > ========================================================= > > 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120213/f13060ec/attachment.html From gopalakrishnan.an at gmail.com Mon Feb 13 15:54:49 2012 From: gopalakrishnan.an at gmail.com (Gopalakrishnan N) Date: Mon, 13 Feb 2012 18:24:49 +0530 Subject: [Freeswitch-users] Internal profile not showing In-Reply-To: <2CFA639E-4530-4B8C-BF67-E2DA60BC2319@jerris.com> References: <4F1DDDA0.1080805@elder.hu> <1CFF2112-65D9-4A36-84A0-57662F929BC6@gmail.com> <2CFA639E-4530-4B8C-BF67-E2DA60BC2319@jerris.com> Message-ID: I enabled the nta debug, even tried as per mentioned here http://wiki.freeswitch.org/wiki/Configuring_SIP but no use. Not able to register a softphone to FS, since its a basic one, which i am not able to. On Wed, Jan 25, 2012 at 7:59 PM, Michael Jerris wrote: > you probably need to turn on nua or nta debug to see the real error here, > but it is almost always an error binding to the IP or port due to IP not > existing, or port being in use. > > On Jan 25, 2012, at 12:06 AM, Gopalakrishnan N wrote: > > Also in my fs cli i got some error message while starting freeswitch which > i pasted here http://pastebin.com/BLegHUGG > > On Wed, Jan 25, 2012 at 10:34 AM, Gopalakrishnan N < > gopalakrishnan.an at gmail.com> wrote: > >> I have pasted the output here http://pastebin.com/54mxJCsT >> >> >> On Tue, Jan 24, 2012 at 11:07 PM, Michael Collins wrote: >> >>> Try: >>> >>> sofia profile internal reload >>> >>> If you get any errors on output then put them in pastebin and let us >>> know. >>> -MC >>> >>> >>> On Tue, Jan 24, 2012 at 1:47 AM, Gopalakrishnan N < >>> gopalakrishnan.an at gmail.com> wrote: >>> >>>> My "sofia status" output is, >>>> freeswitch at ubuntu> sofia status >>>> >>>> Name Type >>>> Data State >>>> >>>> ================================================================================================= >>>> external profile >>>> sip:mod_sofia at 192.168.0.153:5080 RUNNING (0) >>>> external::example.com gateway >>>> sip:joeuser at example.com NOREG >>>> >>>> ================================================================================================= >>>> 1 profile 0 aliases >>>> >>>> freeswitch at ubuntu> >>>> >>>> >>>> On Tue, Jan 24, 2012 at 3:05 PM, Gopalakrishnan N < >>>> gopalakrishnan.an at gmail.com> wrote: >>>> >>>>> I am not able to run "sofia status profile internal reg", In my fs_cli >>>>> I am getting till sofia status profile external not internal, I think my >>>>> internal profile is not activated. >>>>> >>>>> >>>>> On Tue, Jan 24, 2012 at 9:47 AM, Vitaly Colosov wrote: >>>>> >>>>>> After recent change you need to type "sofia status profile internal >>>>>> reg" >>>>>> >>>>>> Sent from my iPad >>>>>> >>>>>> On Jan 23, 2012, at 2:22 PM, Pusk?s Zsolt wrote: >>>>>> >>>>>> Hi, >>>>>> >>>>>> Does your softphone runs on the same machine as freeswitch ? Maybe >>>>>> your softphone already binds to the 5060 port and freeswitch can't start >>>>>> the internal profile which also listens on 5060. >>>>>> >>>>>> What it is the output of "sofia status" command ? >>>>>> >>>>>> >>>>>> 2012-01-23 13:12 keltez?ssel, Gopalakrishnan N ?rta: >>>>>> >>>>>> Hi, >>>>>> >>>>>> I have installed Freeswitch from git in my Ubuntu 11.10. There are >>>>>> some default sip users in prefix/conf/directory/default/1000.xml to >>>>>> 1019.xml. But when I try to register one extension to IP Phone or Softphone >>>>>> the account is not registered and responding as Authentication failuere. >>>>>> 403 Forbidden error, even though I have changed the password in 1000.xml >>>>>> file. >>>>>> >>>>>> Also in FS CLI I am not able to see the output for "sofia status >>>>>> profile 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120213/0ee51bec/attachment-0001.html From shcherbatyuk at belrosbank.by Mon Feb 13 16:10:00 2012 From: shcherbatyuk at belrosbank.by (Eugene Shcherbatyuk) Date: Mon, 13 Feb 2012 16:10:00 +0300 Subject: [Freeswitch-users] FreeSwitch does not start in the background -- bypassed In-Reply-To: References: <4F35051C.7050801@belrosbank.by> <4F38AC1C.7050701@belrosbank.by> Message-ID: <4F390BA8.6010908@belrosbank.by> For the sake of the rest of the list subscribers I would like to answer in English. More elaborate description of the problem follows. Installation went smooth. FS was configured to start as a daemon with freeswitch user privelegies and started fine at OS boot. Couple a days later I had downloaded some extra sound files and copied them into sounds directory. And all of sudden FS did not start after issuing '/etc/init.d/freeswitch restart' command. And it did not start after clean OS reboot. And it did not start from console in background mode (freeswitch -nc). And there was no output nor in system logs nor in FS own log file. Soon enough I have realised that copied files as root and ruined file permissions. After restoring file permissions command '/etc/init.d/freeswitch start' worked. But it worked only once: subsequent attempts to [re]start failed. The situation was reproducible. Finally I could get rid of the problem by setting FS directory ownership to root and starting FS as root in init script. On 13/02/12 14:59, Anton Kvashenkin wrote: > ? ? ??? ?????? ????????, ????? ???????? ????????????, ??????????. ????? > ??????? ?????????? ? ???????? FS ?????? ?? ????????? ?????? ??? ?????? > ???? ????????. ???????? ????????? freeswitch -nc, ????? ??????? ?? > pasterbin. > > 13 ??????? 2012 ?. 10:22 ???????????? Eugene Shcherbatyuk > > ???????: > > Thank you for your answer! > Well, I use the latest git and init script from debian directory in the > sources. > > Obviously, the problem is not solved at its roots yet, but I managed to > bypass it by 'chowning' FS directory and setting user to 'root' in init > script. > > On 10/02/12 20:38, Anton Kvashenkin wrote: > > The first suggestion would be to use latest git and init script from > > http://wiki.freeswitch.org/wiki/Freeswitch_init. > > > > 10 ??????? 2012 ?. 15:53 ???????????? Eugene Shcherbatyuk > > > >> ???????: > > > > Hello, > > > > The purpose was to evaluate FreeSwitch and decide whether to > replace my > > Asterisks with it or not. I installed FS1.0.6 from git on Ubuntu > Server > > 10.04.3 LTS. Everything went smoothly until FS did not start with the > > system today. > > > > There are no errors in logs. FS starts in console fine. FS does not > > start in background mode nor from console (with -nc option) nor from > > init script. Even worse: FS can unexpectedly start (rarely) or do not > > start (often) with "backgrounding" message. Have a look at the > console > > output below, please. > > > > > root at fs:/usr/local/freeswitch/bin# /etc/init.d/freeswitch start > > > # ******* FS started without a message ********* > > > root at fs:/usr/local/freeswitch/bin# netstat -ant > > > Active Internet connections (servers and established) > > > Proto Recv-Q Send-Q Local Address Foreign Address State > > > tcp 0 0 172.18.0.130:5060 > 0.0.0.0:* LISTEN > > > tcp 0 0 127.0.0.1:8021 > 0.0.0.0:* LISTEN > > > tcp 0 0 0.0.0.0:22 > 0.0.0.0:* LISTEN > > > tcp 0 0 172.18.0.130:5080 > 0.0.0.0:* LISTEN > > > tcp6 0 0 ::1:5060 :::* LISTEN > > > tcp6 0 0 :::22 :::* LISTEN > > > root at fs:/usr/local/freeswitch/bin# /etc/init.d/freeswitch stop > > > Killing: 850 > > > root at fs:/usr/local/freeswitch/bin# /etc/init.d/freeswitch start > > > root at fs:/usr/local/freeswitch/bin# 923 Backgrounding. > > > # ****** Note message above ********* > > > root at fs:/usr/local/freeswitch/bin# netstat -ant > > > Active Internet connections (servers and established) > > > Proto Recv-Q Send-Q Local Address Foreign Address State > > > tcp 0 0 0.0.0.0:22 > 0.0.0.0:* LISTEN > > > tcp6 0 0 :::22 :::* LISTEN > > > > Your opinions, please? > > > > Thank you in advance, > > Eugene > > > ========================================================= ?????? ????????? ? ????? ???????? (???????????) ???????? ?????????????????, ???????????????? ????????????? ??? ?????????, ? ????? ????????? ????????, ?????????? ? ???????????? ? ?????????????????. ????? ??????????????????? ????????????? ??? ??????????????? ????????? ?????????. ??????????? ????? ?? ??????????? ??????????? ?????????. ?? ??? "??? "??????????", ?? ????? ??? ????????????? ??? ???????? ??????????? ?? ????? ??????????????? ?? ????????? ? ?????? ??? ???????, ????????? ???? ?????????????. ========================================================= 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. ========================================================= From nasida at live.ru Mon Feb 13 16:48:04 2012 From: nasida at live.ru (Yuriy Nasida) Date: Mon, 13 Feb 2012 17:48:04 +0400 Subject: [Freeswitch-users] Registration Count Message-ID: Hello guys! Is there the standart method to limit count of simultaneous registrations for some user? I wiki I have found only one similar ability http://wiki.freeswitch.org/wiki/API_sofia_count_reg But... It is for dialplan. I use curl module with php scripts for dynamic directory.So I can use this API from php script before sending the xml to FS. But is there more simple ways ? Btw, API 'sofia_count_reg' doesn't work with 'new' git'FreeSWITCH Version 1.0.head (git-e6bfa11 2012-02-09 16-47-32 -0600)' But! API 'sofia_count_reg' works with 'old' gitFreeSWITCH Version 1.0.head (git-54b4b08 2011-11-04 16-58-35 -0500) Please adviseThanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120213/91c0d707/attachment.html From peter.olsson at visionutveckling.se Mon Feb 13 18:43:28 2012 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 13 Feb 2012 15:43:28 +0000 Subject: [Freeswitch-users] Question regarding "uuid_bridge needs at least one leg to be answered" In-Reply-To: <1329107352468-7279309.post@n2.nabble.com> References: <1329107352468-7279309.post@n2.nabble.com> Message-ID: <1FFF97C269757C458224B7C895F35F1503E70F@cantor.std.visionutv.se> No, "bridge" is a totally different application. "uuid_bridge" connects two existing calls, "bridge" connects one existing call to a new call. The call doesn't need to be answered before this. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r dma Skickat: den 13 februari 2012 05:29 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] Question regarding "uuid_bridge needs at least one leg to be answered" Hi All, On the wiki page for uuid_bridge, there is following description: -------------------------------------- uuid_bridge Bridge two call legs together. Usage: uuid_bridge uuid_bridge needs atleast any one leg to be answered. -------------------------------------- Here, "uuid_bridge needs at least any one leg to be answered". Does this limit also apply to dialplan bridge and/or session:bridge in lua script? Thanks, D.Ma -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Question-regarding-uuid-bridge-needs-at-least-one-leg-to-be-answered-tp7279309p7279309.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 FreeSWITCH-users mailing list FreeSWITCH-users at 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:4f38910632766717953067! From fs-list at communicatefreely.net Mon Feb 13 19:32:58 2012 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Mon, 13 Feb 2012 11:32:58 -0500 Subject: [Freeswitch-users] High CPU usage with only 30 channels? Message-ID: <4F393B3A.5090301@communicatefreely.net> Hello, I'm having some scaling problems with our Freeswitch system. The machine running Freeswitch seems to be working very hard during the day, and we don't have that big a load. I need to be able to double our subscriber base, and I don't really want to go to a cluster just yet. Here's what top -CP displays during busy hour: tim at stefan%top -PC last pid: 75968; load averages: 4.64, 2.92, 1.74 up 10+08:54:14 11:17:25 129 processes: 1 running, 128 sleeping CPU 0: 4.5% user, 0.0% nice, 1.9% system, 0.0% interrupt, 93.6% idle CPU 1: 3.8% user, 0.0% nice, 2.6% system, 0.0% interrupt, 93.6% idle Mem: 632M Active, 2093M Inact, 322M Wired, 83M Cache, 384M Buf, 785M Free Swap: 4096M Total, 36K Used, 4096M Free PID USERNAME THR PRI NICE SIZE RES STATE C TIME CPU COMMAND 58521 freeswitch 98 74 r30 749M 648M ucond 0 528:54 0.10% freeswitch 983 root 1 44 0 10824K 2272K select 1 0:40 0.00% ntpd As you can see, the load average is pretty high, at least for something not that big. FS often has up to about 110 threads going. At night, this will settle right down to a 0.05 load average. CPU load never gets that high, but the machine still seems busy. Here's some other details: 40 channels, nearly all of them in CS_EXECUTE or CS_EXCHANGE_MEDIA We see only a few CPS, maybe 10 if it's really busy, but most calls are longer. Nearly all our customers are regular office users. There are 340 phones registered. About 300 of those are behind NAT getting an options-ping every 30s. They re-register every 5 minutes, so we see around 1 registration attempt per second, although it can be a little bursty. Each phone has at least 5 BLF subscriptions, There are 1156 rows in the sip_subscriptions table. All our trunks are SIP. Trunk side is g.711u and endpoints can do either g.711u or g.722 We use ODBC in the core - as well as for sofia, voicemail, and anything else that can use it. Directory and dialplan are done using CURL and handled by another server. The database is also on another server. Our MySQL system sits at about 100 Queries per second during the day , down to about 80 qps at night. This system started to choke until I changed a lot of the tables to the MEMORY engine and installed SSDs as the primary storage devices. Now the database is pretty happy - about 0.5 load average at the moment. It was up to 2 with 60% CPU going to mysqld before I went to memory tables. I keep hearing about FS systems handling thousands of users on one box. I'm nowhere near that and it seems to be maxed out. This system has a dual core Xeon 5100 at 2.6 GHz and 4GB of ram. Memory doesn't seem to be the issue, as there is still some unused. Operating system is FreeBSD 8.0-RELEASE (amd64) Freeswitch is Version 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500) I'm trying to update, but newer version break functionality, so I have to find a release that work before I can upgrade. What should I look for to remedy this, or is this considered normal and I just need more power? I could easily upgrade this box and install a pair of quad core chips. Any other information that would be helpful? From mojo1736 at privatedemail.net Mon Feb 13 19:35:56 2012 From: mojo1736 at privatedemail.net (Josh) Date: Mon, 13 Feb 2012 16:35:56 +0000 Subject: [Freeswitch-users] questions about modules.conf Message-ID: <4F393BEC.9060502@privatedemail.net> According to the Installation/Build Guide (http://wiki.freeswitch.org/wiki/Installation_Guide), modules.conf gets generated during the execution of bootstrap.sh. Should I assume that the options specified in this file - both in terms of what modules are included as well as whether or not they are enabled - depends on my own configuration (the presence of required packages & libraries) as well as the presence of the source code/makefile for a particular module or is the content of this file "predefined" and filled in with a set of "standard" modules regardless of what bootstrap has found (I hope the answer is the former, but need to make absolutely sure)? Also, the same wiki page recommends that I also fetch freeswitch-contrib.git as well as freeswitch-sample-configs.git, but it is not clear to me whether I should dump the contents of these two projects within the main FS tree, where freeswitch.git was placed, or whether I should compile/build these separately - in a completely separate tree? Thanks! From krice at freeswitch.org Mon Feb 13 19:40:33 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 13 Feb 2012 10:40:33 -0600 Subject: [Freeswitch-users] High CPU usage with only 30 channels? In-Reply-To: <4F393B3A.5090301@communicatefreely.net> Message-ID: How is that working hard? That thing is only using about 4% of the cpu Most programs arent as heavily threaded as FS so the load avg never really gets that high... You can safely ignore it... The real thing to pay attention to is how much idle time you have on the cpu... And w0w its 93.6% on both cores.... That should tell you something On 2/13/12 10:32 AM, "Tim St. Pierre" wrote: > Hello, > > I'm having some scaling problems with our Freeswitch system. The > machine running Freeswitch seems to be working very hard during the day, > and we don't have that big a load. I need to be able to double our > subscriber base, and I don't really want to go to a cluster just yet. > Here's what top -CP displays during busy hour: > > tim at stefan%top -PC > last pid: 75968; load averages: 4.64, 2.92, > 1.74 > up 10+08:54:14 11:17:25 > 129 processes: 1 running, 128 sleeping > CPU 0: 4.5% user, 0.0% nice, 1.9% system, 0.0% interrupt, 93.6% idle > CPU 1: 3.8% user, 0.0% nice, 2.6% system, 0.0% interrupt, 93.6% idle > Mem: 632M Active, 2093M Inact, 322M Wired, 83M Cache, 384M Buf, 785M Free > Swap: 4096M Total, 36K Used, 4096M Free > > PID USERNAME THR PRI NICE SIZE RES STATE C TIME CPU COMMAND > 58521 freeswitch 98 74 r30 749M 648M ucond 0 528:54 0.10% > freeswitch > 983 root 1 44 0 10824K 2272K select 1 0:40 0.00% ntpd > > As you can see, the load average is pretty high, at least for something > not that big. FS often has up to about 110 threads going. At night, > this will settle right down to a 0.05 load average. CPU load never gets > that high, but the machine still seems busy. > > Here's some other details: > > 40 channels, nearly all of them in CS_EXECUTE or CS_EXCHANGE_MEDIA We > see only a few CPS, maybe 10 if it's really busy, but most calls are > longer. Nearly all our customers are regular office users. > > There are 340 phones registered. About 300 of those are behind NAT > getting an options-ping every 30s. They re-register every 5 minutes, so > we see around 1 registration attempt per second, although it can be a > little bursty. > > Each phone has at least 5 BLF subscriptions, There are 1156 rows in > the sip_subscriptions table. > > All our trunks are SIP. Trunk side is g.711u and endpoints can do > either g.711u or g.722 > > We use ODBC in the core - as well as for sofia, voicemail, and anything > else that can use it. Directory and dialplan are done using CURL and > handled by another server. The database is also on another server. > > Our MySQL system sits at about 100 Queries per second during the day , > down to about 80 qps at night. This system started to choke until I > changed a lot of the tables to the MEMORY engine and installed SSDs as > the primary storage devices. Now the database is pretty happy - about > 0.5 load average at the moment. It was up to 2 with 60% CPU going to > mysqld before I went to memory tables. > > I keep hearing about FS systems handling thousands of users on one box. > I'm nowhere near that and it seems to be maxed out. > > This system has a dual core Xeon 5100 at 2.6 GHz and 4GB of ram. > Memory doesn't seem to be the issue, as there is still some unused. > Operating system is FreeBSD 8.0-RELEASE (amd64) Freeswitch is Version > 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500) I'm trying to update, > but newer version break functionality, so I have to find a release that > work before I can upgrade. > > What should I look for to remedy this, or is this considered normal and > I just need more power? I could easily upgrade this box and install a > pair of quad core chips. > > Any other information that would be helpful? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > 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 Mon Feb 13 19:43:37 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 13 Feb 2012 10:43:37 -0600 Subject: [Freeswitch-users] questions about modules.conf In-Reply-To: <4F393BEC.9060502@privatedemail.net> Message-ID: Modules.conf is generated at bootstrap time from modules.conf.in The modules that are endabled by default should build if you installed the pre-reqs as defined on the wiki page you reference... Commented out modules either require additional dependancies or are just not needed in the stock example configs.... You don't need to check out the freeswitch-sample-configs.git tree at all... What you need is in the main freeswitch.git tree K On 2/13/12 10:35 AM, "Josh" wrote: > According to the Installation/Build Guide > (http://wiki.freeswitch.org/wiki/Installation_Guide), modules.conf gets > generated during the execution of bootstrap.sh. > > Should I assume that the options specified in this file - both in terms > of what modules are included as well as whether or not they are enabled > - depends on my own configuration (the presence of required packages & > libraries) as well as the presence of the source code/makefile for a > particular module or is the content of this file "predefined" and filled > in with a set of "standard" modules regardless of what bootstrap has > found (I hope the answer is the former, but need to make absolutely sure)? > > Also, the same wiki page recommends that I also fetch > freeswitch-contrib.git as well as freeswitch-sample-configs.git, but it > is not clear to me whether I should dump the contents of these two > projects within the main FS tree, where freeswitch.git was placed, or > whether I should compile/build these separately - in a completely > separate tree? 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 > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fs-list at communicatefreely.net Mon Feb 13 19:52:55 2012 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Mon, 13 Feb 2012 11:52:55 -0500 Subject: [Freeswitch-users] High CPU usage with only 30 channels? In-Reply-To: References: Message-ID: <4F393FE7.6020203@communicatefreely.net> Thanks Ken, I did wonder if it was due to the thread count. If I can safely ignore it, then maybe I shouldn't worry until the CPU usage is higher. We had some spotty service earlier this week that got me scared - phones taking a long time to respond, etc. I think that was on account of the database being choked though. Thanks! -Tim Ken Rice wrote: > How is that working hard? That thing is only using about 4% of the cpu > > Most programs arent as heavily threaded as FS so the load avg never really > gets that high... You can safely ignore it... The real thing to pay > attention to is how much idle time you have on the cpu... And w0w its 93.6% > on both cores.... That should tell you something > > > On 2/13/12 10:32 AM, "Tim St. Pierre" wrote: > > >> Hello, >> >> I'm having some scaling problems with our Freeswitch system. The >> machine running Freeswitch seems to be working very hard during the day, >> and we don't have that big a load. I need to be able to double our >> subscriber base, and I don't really want to go to a cluster just yet. >> Here's what top -CP displays during busy hour: >> >> tim at stefan%top -PC >> last pid: 75968; load averages: 4.64, 2.92, >> 1.74 >> up 10+08:54:14 11:17:25 >> 129 processes: 1 running, 128 sleeping >> CPU 0: 4.5% user, 0.0% nice, 1.9% system, 0.0% interrupt, 93.6% idle >> CPU 1: 3.8% user, 0.0% nice, 2.6% system, 0.0% interrupt, 93.6% idle >> Mem: 632M Active, 2093M Inact, 322M Wired, 83M Cache, 384M Buf, 785M Free >> Swap: 4096M Total, 36K Used, 4096M Free >> >> PID USERNAME THR PRI NICE SIZE RES STATE C TIME CPU COMMAND >> 58521 freeswitch 98 74 r30 749M 648M ucond 0 528:54 0.10% >> freeswitch >> 983 root 1 44 0 10824K 2272K select 1 0:40 0.00% ntpd >> >> As you can see, the load average is pretty high, at least for something >> not that big. FS often has up to about 110 threads going. At night, >> this will settle right down to a 0.05 load average. CPU load never gets >> that high, but the machine still seems busy. >> >> Here's some other details: >> >> 40 channels, nearly all of them in CS_EXECUTE or CS_EXCHANGE_MEDIA We >> see only a few CPS, maybe 10 if it's really busy, but most calls are >> longer. Nearly all our customers are regular office users. >> >> There are 340 phones registered. About 300 of those are behind NAT >> getting an options-ping every 30s. They re-register every 5 minutes, so >> we see around 1 registration attempt per second, although it can be a >> little bursty. >> >> Each phone has at least 5 BLF subscriptions, There are 1156 rows in >> the sip_subscriptions table. >> >> All our trunks are SIP. Trunk side is g.711u and endpoints can do >> either g.711u or g.722 >> >> We use ODBC in the core - as well as for sofia, voicemail, and anything >> else that can use it. Directory and dialplan are done using CURL and >> handled by another server. The database is also on another server. >> >> Our MySQL system sits at about 100 Queries per second during the day , >> down to about 80 qps at night. This system started to choke until I >> changed a lot of the tables to the MEMORY engine and installed SSDs as >> the primary storage devices. Now the database is pretty happy - about >> 0.5 load average at the moment. It was up to 2 with 60% CPU going to >> mysqld before I went to memory tables. >> >> I keep hearing about FS systems handling thousands of users on one box. >> I'm nowhere near that and it seems to be maxed out. >> >> This system has a dual core Xeon 5100 at 2.6 GHz and 4GB of ram. >> Memory doesn't seem to be the issue, as there is still some unused. >> Operating system is FreeBSD 8.0-RELEASE (amd64) Freeswitch is Version >> 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500) I'm trying to update, >> but newer version break functionality, so I have to find a release that >> work before I can upgrade. >> >> What should I look for to remedy this, or is this considered normal and >> I just need more power? I could easily upgrade this box and install a >> pair of quad core chips. >> >> Any other information that would be helpful? >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From godson.g at gmail.com Mon Feb 13 19:56:56 2012 From: godson.g at gmail.com (Godson Gera) Date: Mon, 13 Feb 2012 22:26:56 +0530 Subject: [Freeswitch-users] IVR greet-long repeat In-Reply-To: <585993538.2396.1328973864874.JavaMail.root@server3> References: <585993538.2396.1328973864874.JavaMail.root@server3> Message-ID: Thats the job of greet-short, if you want same file i.e in greet-long to be played out when caller presses nothing then give same file name in greet-short as well On Sat, Feb 11, 2012 at 8:54 PM, George Sapak wrote: > any way to repeat the greet-long message a few time without user needing > to press any buttons? > > Thank You, > George > -- Thanks & Regards, Godson Gera IVR India -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20120213/7fcee4f3/attachment.html From krice at freeswitch.org Mon Feb 13 19:57:49 2012 From: krice at freeswitch.org (Ken Rice) Date: Mon, 13 Feb 2012 10:57:49 -0600 Subject: [Freeswitch-users] High CPU usage with only 30 channels? In-Reply-To: <4F393FE7.6020203@communicatefreely.net> Message-ID: Also keep in mind that since FS is highly threaded its designed to take advantage of more cores... ie: a 8 core 2.3ghz box should easily out perform a 2core 3ghz box... This isnt the case with all software... Most software due to legacy coding restrictions more cores help a fair bit, but not as much as increasing clock speeds due certain parts of the software requiring single threaded (or atleast 1 thread at a time) accessing key linked lists which drags down over all system performance.... K On 2/13/12 10:52 AM, "Tim St. Pierre" wrote: > Thanks Ken, > > I did wonder if it was due to the thread count. If I can safely ignore > it, then maybe I shouldn't worry until the CPU usage is higher. We had > some spotty service earlier this week that got me scared - phones taking > a long time to respond, etc. I think that was on account of the > database being choked though. > > Thanks! > > -Tim > > Ken Rice wrote: >> How is that working hard? That thing is only using about 4% of the cpu >> >> Most programs arent as heavily threaded as FS so the load avg never really >> gets that high... You can safely ignore it... The real thing to pay >> attention to is how much idle time you have on the cpu... And w0w its 93.6% >> on both cores.... That should tell you something >> >> >> On 2/13/12 10:32 AM, "Tim St. Pierre" wrote: >> >> >>> Hello, >>> >>> I'm having some scaling problems with our Freeswitch system. The >>> machine running Freeswitch seems to be working very hard during the day, >>> and we don't have that big a load. I need to be able to double our >>> subscriber base, and I don't really want to go to a cluster just yet. >>> Here's what top -CP displays during busy hour: >>> >>> tim at stefan%top -PC >>> last pid: 75968; load averages: 4.64, 2.92, >>> 1.74 >>> up 10+08:54:14 11:17:25 >>> 129 processes: 1 running, 128 sleeping >>> CPU 0: 4.5% user, 0.0% nice, 1.9% system, 0.0% interrupt, 93.6% idle >>> CPU 1: 3.8% user, 0.0% nice, 2.6% system, 0.0% interrupt, 93.6% idle >>> Mem: 632M Active, 2093M Inact, 322M Wired, 83M Cache, 384M Buf, 785M Free >>> Swap: 4096M Total, 36K Used, 4096M Free >>> >>> PID USERNAME THR PRI NICE SIZE RES STATE C TIME CPU COMMAND >>> 58521 freeswitch 98 74 r30 749M 648M ucond 0 528:54 0.10% >>> freeswitch >>> 983 root 1 44 0 10824K 2272K select 1 0:40 0.00% ntpd >>> >>> As you can see, the load average is pretty high, at least for something >>> not that big. FS often has up to about 110 threads going. At night, >>> this will settle right down to a 0.05 load average. CPU load never gets >>> that high, but the machine still seems busy. >>> >>> Here's some other details: >>> >>> 40 channels, nearly all of them in CS_EXECUTE or CS_EXCHANGE_MEDIA We >>> see only a few CPS, maybe 10 if it's really busy, but most calls are >>> longer. Nearly all our customers are regular office users. >>> >>> There are 340 phones registered. About 300 of those are behind NAT >>> getting an options-ping every 30s. They re-register every 5 minutes, so >>> we see around 1 registration attempt per second, although it can be a >>> little bursty. >>> >>> Each phone has at least 5 BLF subscriptions, There are 1156 rows in >>> the sip_subscriptions table. >>> >>> All our trunks are SIP. Trunk side is g.711u and endpoints can do >>> either g.711u or g.722 >>> >>> We use ODBC in the core - as well as for sofia, voicemail, and anything >>> else that can use it. Directory and dialplan are done using CURL and >>> handled by another server. The database is also on another server. >>> >>> Our MySQL system sits at about 100 Queries per second during the day , >>> down to about 80 qps at night. This system started to choke until I >>> changed a lot of the tables to the MEMORY engine and installed SSDs as >>> the primary storage devices. Now the database is pretty happy - about >>> 0.5 load average at the moment. It was up to 2 with 60% CPU going to >>> mysqld before I went to memory tables. >>> >>> I keep hearing about FS systems handling thousands of users on one box. >>> I'm nowhere near that and it seems to be maxed out. >>> >>> This system has a dual core Xeon 5100 at 2.6 GHz and 4GB of ram. >>> Memory doesn't seem to be the issue, as there is still some unused. >>> Operating system is FreeBSD 8.0-RELEASE (amd64) Freeswitch is Version >>> 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500) I'm trying to update, >>> but newer version break functionality, so I have to find a release that >>> work before I can upgrade. >>> >>> What should I look for to remedy this, or is this considered normal and >>> I just need more power? I could easily upgrade this box and install a >>> pair of quad core chips. >>> >>> Any other information that would be helpful? >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> 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 >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> 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 >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > 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 > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fs-list at communicatefreely.net Mon Feb 13 19:59:40 2012 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Mon, 13 Feb 2012 11:59:40 -0500 Subject: [Freeswitch-users] bypass media only if on same subnet? Message-ID: <4F39417C.3070308@communicatefreely.net> Hello, In order to eliminate NAT issues, we have built routable tunnels to our larger customers so that their phones are on a private subnet that is routable to a private subnet at our datacenter. Each Freeswitch system has a profile called nonat that uses an interface and address bound to this network. The remaining customers register to the "internal" profile, which is bound to a public IP address. At the moment, everything works just great - no nat issues, instant failover between the primary and secondary (shared registrations in DB), but all media flows through our network. If someone in office A calls another phone in office A, I would like FS to instruct the phones to send their media direct. The addresses and ports are all correct in this case. If someone in office A calls a phone in office B, I want Freeswitch to stay in the media path, as these two offices are not routable to each other, even though each is routable to Freeswitch. Is there a way to set up a profile (or dialplan) so that FS will bypass media only if the two endpoints are on the same subnet? An ACL isn't really the right thing, since it would require an exponential number of ACLs. Also, many calls go to ring groups, where several phones ring and we don't know which one will answer until it actually does (eliminating some sort of dialplan code using the rtp variables). Is this possible? From henrikaagaardsorensen at gmail.com Mon Feb 13 11:08:48 2012 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Mon, 13 Feb 2012 09:08:48 +0100 Subject: [Freeswitch-users] Rotate cdr-records as root user (when FS is running from freeswitch user). In-Reply-To: References: Message-ID: I just tested the killall -HUB freeswitch as well, and the cdr records does not get rotated. This is my cdr_csv.conf.xml (the dots ... is just so it didn't take up the entire screen when copy-paste):