From darknesslabs at gmail.com Thu Dec 1 00:55:08 2011 From: darknesslabs at gmail.com (Karol) Date: Wed, 30 Nov 2011 16:55:08 -0500 Subject: [Freeswitch-users] NORTEL IP Phone 1535 with Video In-Reply-To: References: <4ED67544.3020503@gmail.com> Message-ID: I had it working once, for like 5 minutes. On Wed, Nov 30, 2011 at 2:54 PM, curriegrad2004 wrote: > mod_h323 would be the place I'd start looking at for setting up for video > on those nortel/avaya phones. Referring to the manufacturers docs is also > strongly recommended. > On 2011-11-30 11:23 AM, "Timothy Bolton" wrote: > >> Does anyone have any experience in setting up video to work with these >> Nortel phones? >> >> I am completely lost. I've searched the wiki and went to the IRC >> channels too. Any help would be glorious! >> >> -- >> 'We who cut mere stones must always be envisioning cathedrals.' >> Quarry Worker's Creed >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111130/2a146249/attachment.html From ifoundthetao at gmail.com Thu Dec 1 00:58:12 2011 From: ifoundthetao at gmail.com (Timothy Bolton) Date: Wed, 30 Nov 2011 15:58:12 -0600 Subject: [Freeswitch-users] NORTEL IP Phone 1535 with Video In-Reply-To: References: <4ED67544.3020503@gmail.com> Message-ID: <4ED6A6F4.9080104@gmail.com> Was it beautiful? Is it something worthwhile? 'We who cut mere stones must always be envisioning cathedrals.' Quarry Worker's Creed On 11/30/2011 3:55 PM, Karol wrote: > I had it working once, for like 5 minutes. > > On Wed, Nov 30, 2011 at 2:54 PM, curriegrad2004 > > wrote: > > mod_h323 would be the place I'd start looking at for setting up > for video on those nortel/avaya phones. Referring to the > manufacturers docs is also strongly recommended. > > On 2011-11-30 11:23 AM, "Timothy Bolton" > wrote: > > Does anyone have any experience in setting up video to work > with these > Nortel phones? > > I am completely lost. I've searched the wiki and went to the IRC > channels too. Any help would be glorious! > > -- > 'We who cut mere stones must always be envisioning cathedrals.' > Quarry Worker's Creed > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111130/3508b9eb/attachment.html From darknesslabs at gmail.com Thu Dec 1 01:16:34 2011 From: darknesslabs at gmail.com (Karol) Date: Wed, 30 Nov 2011 17:16:34 -0500 Subject: [Freeswitch-users] NORTEL IP Phone 1535 with Video In-Reply-To: <4ED6A6F4.9080104@gmail.com> References: <4ED67544.3020503@gmail.com> <4ED6A6F4.9080104@gmail.com> Message-ID: Yes. On Wed, Nov 30, 2011 at 4:58 PM, Timothy Bolton wrote: > Was it beautiful? Is it something worthwhile? > > 'We who cut mere stones must always be envisioning cathedrals.' > Quarry Worker's Creed > > > On 11/30/2011 3:55 PM, Karol wrote: > > I had it working once, for like 5 minutes. > > On Wed, Nov 30, 2011 at 2:54 PM, curriegrad2004 wrote: > >> mod_h323 would be the place I'd start looking at for setting up for video >> on those nortel/avaya phones. Referring to the manufacturers docs is also >> strongly recommended. >> On 2011-11-30 11:23 AM, "Timothy Bolton" >> wrote: >> >>> Does anyone have any experience in setting up video to work with these >>> Nortel phones? >>> >>> I am completely lost. I've searched the wiki and went to the IRC >>> channels too. Any help would be glorious! >>> >>> -- >>> 'We who cut mere stones must always be envisioning cathedrals.' >>> Quarry Worker's Creed >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/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/20111130/9d813c6c/attachment-0001.html From jeff at jefflenk.com Thu Dec 1 03:55:17 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 30 Nov 2011 16:55:17 -0800 (PST) Subject: [Freeswitch-users] Asterisk TDM410 card and windows In-Reply-To: References: Message-ID: <1322700917423-7049065.post@n2.nabble.com> Look into the Sangoma line of cards they work nicely with Windows! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Asterisk-TDM410-card-and-windows-tp7045776p7049065.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ahe.sanath at gmail.com Thu Dec 1 05:19:22 2011 From: ahe.sanath at gmail.com (Sanath Prasanna) Date: Thu, 1 Dec 2011 07:49:22 +0530 Subject: [Freeswitch-users] Mysql unixODBC problem Message-ID: Hi all, I write lua program to connect with mysql db through ODBC & play the things. But it is not work due to /usr/local/lib/libmyodbc5.so' : file not found.But it is there. Pls advice to solve this problem. [root at test]# ll /usr/local/lib/libmyodbc5.so -rwxr-xr-x 1 root root 318366 Dec 1 07:06 /usr/local/lib/libmyodbc5.so freeswitch.log => 2011-12-01 07:30:15.386020 [DEBUG] switch_rtp.c:2699 Correct ip/port confirmed. 2011-12-01 07:30:17.206663 [INFO] switch_cpp.cpp:1190 TEST_DB - db_connection2011-12-01 07:30:17.206663 [ERR] switch_odbc.c:365 STATE: 01000 CODE 0 ERROR: [unixODBC][Driver Manager]Can't open lib '/usr/local/lib/libmyodbc5.so' : file not found 2011-12-01 07:30:17.206663 [CRIT] switch_core_sqldb.c:381 Failure! 2011-12-01 07:30:17.206663 [INFO] switch_cpp.cpp:1190 TEST_DB - cannot connect to database /usr/local/etc/odbcinst.ini [MySQL] Description=ODBC for MySQL Driver=/usr/local/lib/libmyodbc5.so UsageCount=2 /usr/local/etc/odbc.ini [freeswitch_odbc] Driver=MySQL SERVER=localhost PORT=3306 DATABASE=vm Socket = /tmp/mysql.sock [root at test]# isql -v freeswitch_odbc root test +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> [root at test]# mysql -V mysql Ver 14.14 Distrib 5.5.18, for Linux (i686) using EditLine wrapper Br Sanath -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111201/f942c1ca/attachment.html From admin at blindi.net Thu Dec 1 05:39:39 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Thu, 1 Dec 2011 03:39:39 +0100 (CET) Subject: [Freeswitch-users] Problem send_dtmf send only to a-leg In-Reply-To: References: Message-ID: Hi Michael, Thanks for you reply. Now the following problem occurs: The queue_dtmf sent to bleg. But queue_dtmf send the digits before the channel is answerd. The characters are sent to the channel, while it ringing. sent the tones to aleg. send_dtmf with this construct, is not working. --------------- 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 philq at qsystemsengineering.com Thu Dec 1 07:26:36 2011 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Wed, 30 Nov 2011 23:26:36 -0500 Subject: [Freeswitch-users] How does one make dialplan changes current without restarting FS? L16 codec? Message-ID: <007501ccafe1$6c9c3260$45d49720$@com> Every time I make a dialplan change I have to restart FS to put it into effect. FusionPBX was nice for getting started/familiar with FS and seems to be able to make changes current immediately but I'm discovering that you need to roll up your sleeves and edit the XML files manually to get real work done. Since I'm doing that more and more now, I'd like to be able to make those changes current without restarting. Also, I've noticed that FS seems to be using the L16 codec for early media on the local internal leg lately (I think it's early media where it's being used) - not sure if it was a recent build or changes I made to enable late negotiation and codec inheritance. I do not have that codec specified in any codec lists but everything certainly seems to sound better since I made those changes. What's going on? 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/20111130/9bd7a5b3/attachment.html From farhan.husain at csebuet.org Thu Dec 1 07:56:57 2011 From: farhan.husain at csebuet.org (Farhan Husain) Date: Wed, 30 Nov 2011 20:56:57 -0800 Subject: [Freeswitch-users] Build error: undefined reference to `Curl_setopt' In-Reply-To: <4ED5EEB8.9030709@chaschperli.ch> References: <4ED5EEB8.9030709@chaschperli.ch> Message-ID: Unfortunately this did not help. I installed the library. The Makefile already had -lcurl in several places. I tried to add -lcurl4-dev, -lcurl-dev and -lcurldev. The linker could not find any of those. Is there any other name for the library or -lcurl includes the libcurl4-dev library? On Wed, Nov 30, 2011 at 12:52 AM, Thomas Mueller wrote: > On 30.11.2011 08:14, Farhan Husain wrote: > >> Hi all, >> >> I am trying to build the latest source from Git in an Ubuntu machine. The >> error I am getting is this: >> >> Making all in . >> quiet_libtool: link: gcc -I/usr/local/src/freeswitch/**src/include >> -I/usr/local/src/freeswitch/**src/include -I/usr/local/src/freeswitch/**libs/libteletone/src >> -fPIC -Werror -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 >> -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 -pedantic >> -Wdeclaration-after-statement -g -O2 -pthread -DLINUX=2 -D_REENTRANT >> -D_GNU_SOURCE -I/usr/local/src/freeswitch/**libs/apr/include >> -I/usr/local/src/freeswitch/**libs/apr-util/include >> -I/usr/local/src/freeswitch/**libs/apr-util/xml/expat/lib >> -I/usr/local/src/freeswitch/**libs/stfu -I/usr/local/src/freeswitch/**libs/sqlite >> -I/usr/local/src/freeswitch/**libs/pcre -I/usr/local/src/freeswitch/**libs/speex/include >> -Ilibs/speex/include -I/usr/local/src/freeswitch/**libs/srtp/include >> -I/usr/local/src/freeswitch/**libs/srtp/crypto/include >> -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/**libs/spandsp/src >> -I/usr/local/src/freeswitch/**libs/tiff-3.8.2/libtiff -DENABLE_SRTP >> -I/usr/local/src/freeswitch/**libs/libedit/src -DSWITCH_HAVE_LIBEDIT >> -Ilibs/libedit/src -DSWITCH_HAVE_LIBEDIT -g -O2 -o .libs/freeswitch >> freeswitch-switch.o /usr/lib/libcurl.so -lm -lz ./.libs/libfreeswitch.so >> libs/apr/.libs/libapr-1.a -luuid -lrt -ldl -lcrypt -lpthread >> libs/libedit/src/.libs/**libedit.a -lssl -lcrypto -lncurses -pthread >> -Wl,-rpath -Wl,/usr/local/freeswitch/lib >> >> ./.libs/libfreeswitch.so: undefined reference to `Curl_setopt' >> collect2: ld returned 1 exit status >> make[2]: *** [freeswitch] Error 1 >> >> I have installed php-curl (sudo apt-get install php5-curl) but to no >> avail. Does anyone have any solution to this? >> > > seems like the curl dev package is missing ( libcurl4-dev or libcurl3-dev) > -> sudo aptitude install libcurl4-dev > > - Thomas > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111130/b977f262/attachment.html From vetali100 at gmail.com Thu Dec 1 09:32:58 2011 From: vetali100 at gmail.com (Vitali Colosov) Date: Wed, 30 Nov 2011 22:32:58 -0800 Subject: [Freeswitch-users] How does one make dialplan changes current without restarting FS? L16 codec? In-Reply-To: <007501ccafe1$6c9c3260$45d49720$@com> References: <007501ccafe1$6c9c3260$45d49720$@com> Message-ID: To reload the dialplan without restarting FS, you can just execute "reloadxml" from the fs_cli. Regarding L16, someone else will advise. On Nov 30, 2011, at 8:26 PM, Phil Quesinberry wrote: > Every time I make a dialplan change I have to restart FS to put it into effect. FusionPBX was nice for getting started/familiar with FS and seems to be able to make changes current immediately but I?m discovering that you need to roll up your sleeves and edit the XML files manually to get real work done. Since I?m doing that more and more now, I?d like to be able to make those changes current without restarting. > > Also, I?ve noticed that FS seems to be using the L16 codec for early media on the local internal leg lately (I think it?s early media where it?s being used) ? not sure if it was a recent build or changes I made to enable late negotiation and codec inheritance. I do not have that codec specified in any codec lists but everything certainly seems to sound better since I made those changes. What?s going on? > > 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111130/7c761d15/attachment-0001.html From gmaruzz at gmail.com Thu Dec 1 10:39:20 2011 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Thu, 1 Dec 2011 08:39:20 +0100 Subject: [Freeswitch-users] How does one make dialplan changes current without restarting FS? L16 codec? In-Reply-To: References: <007501ccafe1$6c9c3260$45d49720$@com> Message-ID: L16 is an internal format (signed linear 16 bit) used by FS. Is not a codec, and you don't need to worry about it. -giovanni On 12/1/11, Vitali Colosov wrote: > To reload the dialplan without restarting FS, you can just execute > "reloadxml" from the fs_cli. > > Regarding L16, someone else will advise. > > > > On Nov 30, 2011, at 8:26 PM, Phil Quesinberry wrote: > >> Every time I make a dialplan change I have to restart FS to put it into >> effect. FusionPBX was nice for getting started/familiar with FS and seems >> to be able to make changes current immediately but I?m discovering that >> you need to roll up your sleeves and edit the XML files manually to get >> real work done. Since I?m doing that more and more now, I?d like to be >> able to make those changes current without restarting. >> >> Also, I?ve noticed that FS seems to be using the L16 codec for early media >> on the local internal leg lately (I think it?s early media where it?s >> being used) ? not sure if it was a recent build or changes I made to >> enable late negotiation and codec inheritance. I do not have that codec >> specified in any codec lists but everything certainly seems to sound >> better since I made those changes. What?s going on? >> >> 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 > > -- Sent from my mobile device Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From tonybecq at yahoo.fr Thu Dec 1 12:57:32 2011 From: tonybecq at yahoo.fr (obbyone) Date: Thu, 1 Dec 2011 01:57:32 -0800 (PST) Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... Message-ID: <1322733452381-7049935.post@n2.nabble.com> Hi, I've just installed Freeswitch from GIT on Linux UBUNTU 8.04 LTS. Registering is ok, so every softphone or ATA are registered on the server BUT, as I try to place a call from a softphone to another, it doesn't work. Has someone a solution to my problem, Thanks, Tony -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-installed-ATA-registered-but-no-call-are-possible-tp7049935p7049935.html Sent from the freeswitch-users mailing list archive at Nabble.com. From avi at avimarcus.net Thu Dec 1 13:33:58 2011 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 1 Dec 2011 12:33:58 +0200 Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... In-Reply-To: <1322733452381-7049935.post@n2.nabble.com> References: <1322733452381-7049935.post@n2.nabble.com> Message-ID: You need to start with explain what exactly you mean by "it doesn't work". Also, paste-binning a trace from fs_cli will help explain what's going on. -Avi Marcus On Thu, Dec 1, 2011 at 11:57 AM, obbyone wrote: > Hi, > > I've just installed Freeswitch from GIT on Linux UBUNTU 8.04 LTS. > Registering is ok, so every softphone or ATA are registered on the server > BUT, as I try to place a call from a softphone to another, it doesn't work. > > Has someone a solution to my problem, > > Thanks, > > Tony > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-installed-ATA-registered-but-no-call-are-possible-tp7049935p7049935.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/20111201/2a3589de/attachment.html From tonybecq at yahoo.fr Thu Dec 1 15:46:26 2011 From: tonybecq at yahoo.fr (obbyone) Date: Thu, 1 Dec 2011 04:46:26 -0800 (PST) Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... In-Reply-To: References: <1322733452381-7049935.post@n2.nabble.com> Message-ID: <1322743586826-7050389.post@n2.nabble.com> When I say it doesn't work, I mean that I can only connect the "answering machine". It doesn't ring. Following the fs_cli trace... (from logs): ... 2011-12-01 13:42:18.918274 [DEBUG] sofia.c:7267 IP 90.36.1.89 Rejected by acl "domains". Falling back to Digest auth. 2011-12-01 13:42:29.258256 [DEBUG] sofia.c:7267 IP 90.36.1.89 Rejected by acl "domains". Falling back to Digest auth. 2011-12-01 13:42:29.258256 [NOTICE] switch_channel.c:920 New Channel sofia/internal/1004 at connexur.dyndns.org [ee2049ee-1c19-11e1-abcd-1f494360f62d] 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_NEW 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:380 (sofia/internal/1004 at connexur.dyndns.org) State NEW 2011-12-01 13:42:29.268425 [DEBUG] sofia.c:8186 Setting NAT mode based on nat.auto 2011-12-01 13:42:29.268425 [DEBUG] sofia.c:5282 Channel sofia/internal/1004 at connexur.dyndns.org entering state [received][100] 2011-12-01 13:42:29.268425 [DEBUG] sofia.c:5293 Remote SDP: v=0 o=- 9 2 IN IP4 192.168.1.10 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.10 t=0 0 m=audio 64860 RTP/AVP 107 119 0 98 8 3 101 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 1 : PBDGzwKZ EILmrBFJ 192.168.1.10 64860 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[G7221:115:32000:20:48000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[G7221:107:16000:20:32000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[G722:9:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[PCMU:0:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[PCMA:8:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[GSM:3:8000:20:13200] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[PCMA:8:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[GSM:3:8000:20:13200] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32-FEC:119:16000:20:0]/[G7221:115:32000:20:48000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32-FEC:119:16000:20:0]/[G7221:107:16000:20:32000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32-FEC:119:16000:20:0]/[G722:9:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32-FEC:119:16000:20:0]/[PCMU:0:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32-FEC:119:16000:20:0]/[PCMA:8:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32-FEC:119:16000:20:0]/[GSM:3:8000:20:13200] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:2864 Set Codec sofia/internal/1004 at connexur.dyndns.org PCMU/8000 20 ms 160 samples 64000 bits 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4871 Set 2833 dtmf send/recv payload to 101 2011-12-01 13:42:29.268425 [DEBUG] sofia.c:5505 (sofia/internal/1004 at connexur.dyndns.org) State Change CS_NEW -> CS_INIT 2011-12-01 13:42:29.268425 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_INIT 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1004 at connexur.dyndns.org) State INIT 2011-12-01 13:42:29.268425 [DEBUG] mod_sofia.c:85 sofia/internal/1004 at connexur.dyndns.org SOFIA INIT 2011-12-01 13:42:29.268425 [DEBUG] mod_sofia.c:125 (sofia/internal/1004 at connexur.dyndns.org) State Change CS_INIT -> CS_ROUTING 2011-12-01 13:42:29.268425 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1004 at connexur.dyndns.org) State INIT going to sleep 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_ROUTING 2011-12-01 13:42:29.268425 [DEBUG] switch_channel.c:1871 (sofia/internal/1004 at connexur.dyndns.org) Callstate Change DOWN -> RINGING 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1004 at connexur.dyndns.org) State ROUTING 2011-12-01 13:42:29.268425 [DEBUG] mod_sofia.c:148 sofia/internal/1004 at connexur.dyndns.org SOFIA ROUTING 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1004 at connexur.dyndns.org Standard ROUTING 2011-12-01 13:42:29.268425 [INFO] mod_dialplan_xml.c:481 Processing 1004 <1004>->1001 in context default Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->unloop] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->tod_example] continue=true Dialplan: sofia/internal/1004 at connexur.dyndns.org Date/Time Match (PASS) [tod_example] break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(open=true) Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->holiday_example] continue=true Dialplan: sofia/internal/1004 at connexur.dyndns.org Date/TimeMatch (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->global-intercept] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [global-intercept] destination_number(1001) =~ /^886$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->group-intercept] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [group-intercept] destination_number(1001) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [intercept-ext] destination_number(1001) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->redial] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [redial] destination_number(1001) =~ /^(redial|870)$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->global] continue=true Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=n$ Dialplan: sofia/internal/1004 at connexur.dyndns.org Absolute Condition [global] Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [snom-demo-2] destination_number(1001) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [snom-demo-2] destination_number(1001) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [snom-demo-1] destination_number(1001) =~ /^9000$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^779$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->call_return] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [call_return] destination_number(1001) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->del-group] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [del-group] destination_number(1001) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->add-group] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [add-group] destination_number(1001) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [call-group-simo] destination_number(1001) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->call-group-order] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [call-group-order] destination_number(1001) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [extension-intercom] destination_number(1001) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (PASS) [Local_Extension] destination_number(1001) =~ /^(10[01][0-9]|1100)$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(dialed_extension=1001) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action export(dialed_extension=1001) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime$ Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(ringback=${us-ring}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(call_timeout=30) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(hangup_after_bridge=true) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(continue_on_fail=true) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action answer() Dialplan: sofia/internal/1004 at connexur.dyndns.org Action sleep(1000) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/1004 at connexur.dyndns.org) State Change CS_ROUTING -> CS_EXECUTE 2011-12-01 13:42:29.268425 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1004 at connexur.dyndns.org) State ROUTING going to sleep 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_EXECUTE 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1004 at connexur.dyndns.org) State EXECUTE 2011-12-01 13:42:29.268425 [DEBUG] mod_sofia.c:241 sofia/internal/1004 at connexur.dyndns.org SOFIA EXECUTE 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:192 sofia/internal/1004 at connexur.dyndns.org Standard EXECUTE EXECUTE sofia/internal/1004 at connexur.dyndns.org set(open=true) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [open]=[true] EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-spymap/1004/ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial/1004/1001) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial/global/ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org set(RFC2822_DATE=Thu, 01 Dec 2011 13:42:29 +0100) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial/global/ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org set(RFC2822_DATE=Thu, 01 Dec 2011 13:42:29 +0100) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [RFC2822_DATE]=[Thu, 01 Dec 2011 13:42:29 +0100] EXECUTE sofia/internal/1004 at connexur.dyndns.org set(dialed_extension=1001) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [dialed_extension]=[1001] EXECUTE sofia/internal/1004 at connexur.dyndns.org export(dialed_extension=1001) 2011-12-01 13:42:29.268425 [DEBUG] switch_channel.c:1087 EXPORT (export_vars) [dialed_extension]=[1001] EXECUTE sofia/internal/1004 at connexur.dyndns.org bind_meta_app(1 b s execute_extension::dx XML features) 2011-12-01 13:42:29.268425 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE sofia/internal/1004 at connexur.dyndns.org bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/1004.2011-12-01-13-42-29.wav) 2011-12-01 13:42:29.268425 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1004.2011-12-01-13-42-29.wav EXECUTE sofia/internal/1004 at connexur.dyndns.org bind_meta_app(3 b s execute_extension::cf XML features) 2011-12-01 13:42:29.268425 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE sofia/internal/1004 at connexur.dyndns.org bind_meta_app(4 b s execute_extension::att_xfer XML features) 2011-12-01 13:42:29.268425 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *4 execute_extension::att_xfer XML features EXECUTE sofia/internal/1004 at connexur.dyndns.org set(ringback=%(2000,4000,440.0,480.0)) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [ringback]=[%(2000,4000,440.0,480.0)] EXECUTE sofia/internal/1004 at connexur.dyndns.org set(transfer_ringback=local_stream://moh) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/1004 at connexur.dyndns.org set(call_timeout=30) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [call_timeout]=[30] EXECUTE sofia/internal/1004 at connexur.dyndns.org set(hangup_after_bridge=true) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/1004 at connexur.dyndns.org set(continue_on_fail=true) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [continue_on_fail]=[true] EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-call_return/1001/1004) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial_ext/1001/ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial_ext//ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial_ext/global/ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org set(called_party_callgroup=techsupport) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [called_party_callgroup]=[techsupport] EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial/techsupport/ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org bridge(user/1001 at 91.204.116.116) 2011-12-01 13:42:29.268425 [DEBUG] switch_channel.c:1041 sofia/internal/1004 at connexur.dyndns.org EXPORTING[export_vars] [dialed_extension]=[1001] to event 2011-12-01 13:42:29.268425 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2011-12-01 13:42:29.268425 [DEBUG] switch_channel.c:1041 sofia/internal/1004 at connexur.dyndns.org EXPORTING[export_vars] [dialed_extension]=[1001] to event 2011-12-01 13:42:29.268425 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2011-12-01 13:42:29.268425 [DEBUG] switch_event.c:1521 Parsing variable [sip_invite_domain]=[91.204.116.116] 2011-12-01 13:42:29.268425 [DEBUG] switch_event.c:1521 Parsing variable [presence_id]=[1001 at 91.204.116.116] 2011-12-01 13:42:29.268425 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2011-12-01 13:42:29.268425 [DEBUG] switch_ivr_originate.c:3367 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2011-12-01 13:42:29.268425 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2011-12-01 13:42:29.268425 [DEBUG] switch_ivr_originate.c:3367 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2011-12-01 13:42:29.268425 [INFO] mod_dptools.c:2838 Originate Failed. Cause: USER_NOT_REGISTERED EXECUTE sofia/internal/1004 at connexur.dyndns.org answer() 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:3116 AUDIO RTP [sofia/internal/1004 at connexur.dyndns.org] 91.204.116.116 port 21454 -> 192.168.1.10 port 6486$ 2011-12-01 13:42:29.268425 [DEBUG] switch_rtp.c:1642 Starting timer [soft] 160 bytes per 20ms 2011-12-01 13:42:29.293791 [DEBUG] sofia_glue.c:3382 Set 2833 dtmf send payload to 101 2011-12-01 13:42:29.293791 [DEBUG] sofia_glue.c:3388 Set 2833 dtmf receive payload to 101 2011-12-01 13:42:29.293791 [DEBUG] mod_sofia.c:746 Local SDP sofia/internal/1004 at connexur.dyndns.org: v=0 o=FreeSWITCH 1322721895 1322721896 IN IP4 91.204.116.116 s=FreeSWITCH c=IN IP4 91.204.116.116 t=0 0 c=IN IP4 91.204.116.116 t=0 0 m=audio 21454 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 2011-12-01 13:42:29.293791 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.293791 [DEBUG] sofia.c:5282 Channel sofia/internal/1004 at connexur.dyndns.org entering state [completed][200] 2011-12-01 13:42:29.293791 [DEBUG] switch_core_session.c:726 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.293791 [DEBUG] switch_channel.c:3175 (sofia/internal/1004 at connexur.dyndns.org) Callstate Change RINGING -> ACTIVE 2011-12-01 13:42:29.293791 [NOTICE] mod_dptools.c:1071 Channel [sofia/internal/1004 at connexur.dyndns.org] has been answered EXECUTE sofia/internal/1004 at connexur.dyndns.org sleep(1000) 2011-12-01 13:42:29.568264 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.568264 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.568264 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.578421 [DEBUG] sofia.c:5282 Channel sofia/internal/1004 at connexur.dyndns.org entering state [ready][200] 2011-12-01 13:42:29.658301 [INFO] switch_rtp.c:3170 Auto Changing port from 192.168.1.10:64860 to 90.36.1.89:64860 EXECUTE sofia/internal/1004 at connexur.dyndns.org bridge(loopback/app=voicemail:default 91.204.116.116 1001) 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:1041 sofia/internal/1004 at connexur.dyndns.org EXPORTING[export_vars] [dialed_extension]=[1001] to event 2011-12-01 13:42:30.298263 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2011-12-01 13:42:30.298263 [NOTICE] switch_channel.c:920 New Channel loopback/app=voicemail:default 91.204.116.116 1001-a [eebdaf40-1c19-11e1-abd8-1f494360f$ 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:145 loopback/app=voicemail:default 91.204.116.116 1001-a setup codec PCMU/8000/20 2011-12-01 13:42:30.298263 [NOTICE] switch_channel.c:918 Rename Channel loopback/app=voicemail:default 91.204.116.116 1001-a->loopback/voicemail-a [eebdaf40$ 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:973 (loopback/voicemail-a) State Change CS_NEW -> CS_INIT 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-a) Running State Change CS_INIT 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:401 (loopback/voicemail-a) State INIT 2011-12-01 13:42:30.298263 [NOTICE] switch_channel.c:920 New Channel loopback/voicemail-b [eebdc6ba-1c19-11e1-abdc-1f494360f62d] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:145 loopback/voicemail-b setup codec PCMU/8000/20 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:258 (loopback/voicemail-b) State Change CS_NEW -> CS_INIT 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-b) Running State Change CS_INIT 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:401 (loopback/voicemail-b) State INIT 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:304 (loopback/voicemail-b) State Change CS_INIT -> CS_ROUTING 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:401 (loopback/voicemail-b) State INIT going to sleep 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-b) Running State Change CS_ROUTING 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:1871 (loopback/voicemail-b) Callstate Change DOWN -> RINGING 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:410 (loopback/voicemail-b) State ROUTING 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:336 loopback/voicemail-b CHANNEL ROUTING 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:355 (loopback/voicemail-b) State Change CS_ROUTING -> CS_EXECUTE 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:410 (loopback/voicemail-b) State ROUTING going to sleep 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-b) Running State Change CS_EXECUTE 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:417 (loopback/voicemail-b) State EXECUTE 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:375 loopback/voicemail-b CHANNEL EXECUTE 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:192 loopback/voicemail-b Standard EXECUTE 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:375 loopback/voicemail-b CHANNEL EXECUTE 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:192 loopback/voicemail-b Standard EXECUTE EXECUTE loopback/voicemail-b pre_answer() 2011-12-01 13:42:30.298263 [NOTICE] mod_loopback.c:760 Pre-Answer loopback/voicemail-a! 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:2917 (loopback/voicemail-a) Callstate Change DOWN -> EARLY 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:2959 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:726 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:30.298263 [NOTICE] mod_dptools.c:1097 Pre-Answer loopback/voicemail-b! 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:2917 (loopback/voicemail-b) Callstate Change RINGING -> EARLY 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:2959 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] EXECUTE loopback/voicemail-b voicemail(default 91.204.116.116 1001) 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:304 (loopback/voicemail-a) State Change CS_INIT -> CS_ROUTING 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:401 (loopback/voicemail-a) State INIT going to sleep 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-a) Running State Change CS_ROUTING 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:1871 (loopback/voicemail-a) Callstate Change EARLY -> RINGING 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:410 (loopback/voicemail-a) State ROUTING 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:336 loopback/voicemail-a CHANNEL ROUTING 2011-12-01 13:42:30.298263 [DEBUG] switch_ivr_originate.c:66 (loopback/voicemail-a) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:410 (loopback/voicemail-a) State ROUTING going to sleep 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-a) Running State Change CS_CONSUME_MEDIA 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:1875 (loopback/voicemail-a) Callstate Change RINGING -> EARLY 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:429 (loopback/voicemail-a) State CONSUME_MEDIA 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:535 CHANNEL CONSUME_MEDIA 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:429 (loopback/voicemail-a) State CONSUME_MEDIA going to sleep 2011-12-01 13:42:30.308562 [DEBUG] switch_ivr_originate.c:3269 Originate Resulted in Success: [loopback/voicemail-a] 2011-12-01 13:42:30.308562 [DEBUG] switch_core_session.c:726 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:30.308562 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:30.308562 [DEBUG] switch_core_session.c:726 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:30.308562 [DEBUG] switch_ivr_bridge.c:1270 (loopback/voicemail-a) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2011-12-01 13:42:30.308562 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:30.308562 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:30.308562 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-a) Running State Change CS_EXCHANGE_MEDIA 2011-12-01 13:42:30.308562 [DEBUG] switch_core_state_machine.c:420 (loopback/voicemail-a) State EXCHANGE_MEDIA 2011-12-01 13:42:30.308562 [DEBUG] mod_loopback.c:497 CHANNEL LOOPBACK 2011-12-01 13:42:30.398283 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en] 2011-12-01 13:42:30.409312 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-person.wav] (en:en) 2011-12-01 13:42:30.409312 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1 channels 20ms 2011-12-01 13:42:31.798263 [DEBUG] switch_ivr_play_say.c:1672 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-person.wav 2011-12-01 13:42:31.898264 [DEBUG] switch_ivr_play_say.c:244 Handle say:[1001] (en:en) 2011-12-01 13:42:31.898264 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1 channels 20ms 2011-12-01 13:42:33.398281 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:33.418917 [DEBUG] switch_channel.c:2833 (sofia/internal/1004 at connexur.dyndns.org) Callstate Change ACTIVE -> HANGUP 2011-12-01 13:42:33.418917 [NOTICE] sofia.c:572 Hangup sofia/internal/1004 at connexur.dyndns.org [CS_EXECUTE] [NORMAL_CLEARING] 2011-12-01 13:42:33.418917 [DEBUG] switch_channel.c:2856 Send signal sofia/internal/1004 at connexur.dyndns.org [KILL] 2011-12-01 13:42:33.418917 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:33.418917 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/1004 at connexur.dyndns.org] 2011-12-01 13:42:33.418917 [DEBUG] switch_ivr_bridge.c:611 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:33.418917 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [loopback/voicemail-a] 2011-12-01 13:42:33.438243 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:33.438243 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [loopback/voicemail-a] 2011-12-01 13:42:33.438243 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:33.438243 [DEBUG] switch_channel.c:2833 (loopback/voicemail-a) Callstate Change EARLY -> HANGUP 2011-12-01 13:42:33.438243 [NOTICE] switch_ivr_bridge.c:666 Hangup loopback/voicemail-a [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2011-12-01 13:42:33.438243 [DEBUG] switch_channel.c:2856 Send signal loopback/voicemail-a [KILL] 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:420 (loopback/voicemail-a) State EXCHANGE_MEDIA going to sleep 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-a) Running State Change CS_HANGUP 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:602 (loopback/voicemail-a) State HANGUP 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:427 loopback/voicemail-a CHANNEL HANGUP 2011-12-01 13:42:33.438243 [DEBUG] switch_channel.c:2833 (loopback/voicemail-b) Callstate Change EARLY -> HANGUP 2011-12-01 13:42:33.438243 [NOTICE] mod_loopback.c:438 Hangup loopback/voicemail-b [CS_EXECUTE] [NORMAL_CLEARING] 2011-12-01 13:42:33.438243 [DEBUG] switch_channel.c:2856 Send signal loopback/voicemail-b [KILL] 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:47 loopback/voicemail-a Standard HANGUP, cause: NORMAL_CLEARING 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:602 (loopback/voicemail-a) State HANGUP going to sleep 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:393 (loopback/voicemail-a) State Change CS_HANGUP -> CS_REPORTING 2011-12-01 13:42:33.438243 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-a) Running State Change CS_REPORTING 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:662 (loopback/voicemail-a) State REPORTING 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:79 loopback/voicemail-a Standard REPORTING, cause: NORMAL_CLEARING 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:662 (loopback/voicemail-a) State REPORTING going to sleep 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:387 (loopback/voicemail-a) State Change CS_REPORTING -> CS_DESTROY 2011-12-01 13:42:33.438243 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_core_session.c:1377 Session 145 (loopback/voicemail-a) Locked, Waiting on external entities 2011-12-01 13:42:33.450220 [DEBUG] switch_ivr_bridge.c:1348 sofia/internal/1004 at connexur.dyndns.org skip receive message [UNBRIDGE] (channel is hungup alrea$ 2011-12-01 13:42:33.450220 [DEBUG] switch_core_session.c:2272 sofia/internal/1004 at connexur.dyndns.org skip receive message [APPLICATION_EXEC_COMPLETE] (chan$ 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1004 at connexur.dyndns.org) State EXECUTE going to sleep 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_HANGUP 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1004 at connexur.dyndns.org) State HANGUP 2011-12-01 13:42:33.450220 [DEBUG] mod_sofia.c:459 sofia/internal/1004 at connexur.dyndns.org Overriding SIP cause 480 with 200 from the other leg 2011-12-01 13:42:33.450220 [DEBUG] mod_sofia.c:465 Channel sofia/internal/1004 at connexur.dyndns.org hanging up, cause: NORMAL_CLEARING 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1004 at connexur.dyndns.org Standard HANGUP, cause: NORMAL_CLEARING 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1004 at connexur.dyndns.org) State HANGUP going to sleep 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1004 at connexur.dyndns.org) State Change CS_HANGUP -> CS_REPORTING 2011-12-01 13:42:33.450220 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_REPORTING 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1004 at connexur.dyndns.org) State REPORTING 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1004 at connexur.dyndns.org Standard REPORTING, cause: NORMAL_CLEARING 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1004 at connexur.dyndns.org) State REPORTING going to sleep 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1004 at connexur.dyndns.org) State Change CS_REPORTING -> CS_DESTROY 2011-12-01 13:42:33.450220 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:33.450220 [DEBUG] switch_core_session.c:1377 Session 144 (sofia/internal/1004 at connexur.dyndns.org) Locked, Waiting on external entities 2011-12-01 13:42:33.450220 [NOTICE] switch_core_session.c:1395 Session 144 (sofia/internal/1004 at connexur.dyndns.org) Ended 2011-12-01 13:42:33.450220 [NOTICE] switch_core_session.c:1397 Close Channel sofia/internal/1004 at connexur.dyndns.org [CS_DESTROY] 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1004 at connexur.dyndns.org) Callstate Change HANGUP -> DOWN 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_DESTROY 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1004 at connexur.dyndns.org) State DESTROY 2011-12-01 13:42:33.450220 [DEBUG] mod_sofia.c:370 sofia/internal/1004 at connexur.dyndns.org SOFIA DESTROY 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1004 at connexur.dyndns.org) State DESTROY 2011-12-01 13:42:33.450220 [DEBUG] mod_sofia.c:370 sofia/internal/1004 at connexur.dyndns.org SOFIA DESTROY 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1004 at connexur.dyndns.org Standard DESTROY 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1004 at connexur.dyndns.org) State DESTROY going to sleep 2011-12-01 13:42:33.458337 [DEBUG] switch_ivr_play_say.c:1672 done playing file file_string://digits/1.wav!digits/0.wav!digits/0.wav!digits/1.wav 2011-12-01 13:42:33.548273 [DEBUG] switch_core_session.c:2272 loopback/voicemail-b skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup alrea$ 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:417 (loopback/voicemail-b) State EXECUTE going to sleep 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-b) Running State Change CS_HANGUP 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:602 (loopback/voicemail-b) State HANGUP 2011-12-01 13:42:33.548273 [DEBUG] mod_loopback.c:427 loopback/voicemail-b CHANNEL HANGUP 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:47 loopback/voicemail-b Standard HANGUP, cause: NORMAL_CLEARING 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:602 (loopback/voicemail-b) State HANGUP going to sleep 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:393 (loopback/voicemail-b) State Change CS_HANGUP -> CS_REPORTING 2011-12-01 13:42:33.548273 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:33.548273 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-b) Running State Change CS_REPORTING 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:662 (loopback/voicemail-b) State REPORTING 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:79 loopback/voicemail-b Standard REPORTING, cause: NORMAL_CLEARING 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:662 (loopback/voicemail-b) State REPORTING going to sleep 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:387 (loopback/voicemail-b) State Change CS_REPORTING -> CS_DESTROY 2011-12-01 13:42:33.548273 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:33.548273 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:33.548273 [DEBUG] switch_core_session.c:1377 Session 146 (loopback/voicemail-b) Locked, Waiting on external entities 2011-12-01 13:42:33.548273 [NOTICE] switch_core_session.c:1395 Session 146 (loopback/voicemail-b) Ended 2011-12-01 13:42:33.548273 [NOTICE] switch_core_session.c:1397 Close Channel loopback/voicemail-b [CS_DESTROY] 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:491 (loopback/voicemail-b) Callstate Change HANGUP -> DOWN 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:494 (loopback/voicemail-b) Running State Change CS_DESTROY 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:504 (loopback/voicemail-b) State DESTROY 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:86 loopback/voicemail-b Standard DESTROY 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:504 (loopback/voicemail-b) State DESTROY going to sleep 2011-12-01 13:42:33.559551 [NOTICE] switch_core_session.c:1395 Session 145 (loopback/voicemail-a) Ended 2011-12-01 13:42:33.559551 [NOTICE] switch_core_session.c:1397 Close Channel loopback/voicemail-a [CS_DESTROY] 2011-12-01 13:42:33.559551 [DEBUG] switch_core_state_machine.c:491 (loopback/voicemail-a) Callstate Change HANGUP -> DOWN 2011-12-01 13:42:33.559551 [DEBUG] switch_core_state_machine.c:494 (loopback/voicemail-a) Running State Change CS_DESTROY 2011-12-01 13:42:33.559551 [DEBUG] switch_core_state_machine.c:504 (loopback/voicemail-a) State DESTROY 2011-12-01 13:42:33.559551 [DEBUG] switch_core_state_machine.c:86 loopback/voicemail-a Standard DESTROY 2011-12-01 13:42:33.559551 [DEBUG] switch_core_state_machine.c:504 (loopback/voicemail-a) State DESTROY going to sleep -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-installed-ATA-registered-but-no-call-are-possible-tp7049935p7050389.html Sent from the freeswitch-users mailing list archive at Nabble.com. From cstomi.levlist at gmail.com Thu Dec 1 16:22:49 2011 From: cstomi.levlist at gmail.com (Tamas.Cseke ) Date: Thu, 01 Dec 2011 14:22:49 +0100 Subject: [Freeswitch-users] Build error: undefined reference to `Curl_setopt' In-Reply-To: References: <4ED5EEB8.9030709@chaschperli.ch> Message-ID: <4ED77FA9.5020308@gmail.com> It has been fixed. try latest git On 2011-12-01 05:56, Farhan Husain wrote: > Unfortunately this did not help. I installed the library. The Makefile > already had -lcurl in several places. I tried to add -lcurl4-dev, > -lcurl-dev and -lcurldev. The linker could not find any of those. Is > there any other name for the library or -lcurl includes the > libcurl4-dev library? > > On Wed, Nov 30, 2011 at 12:52 AM, Thomas Mueller > > wrote: > > On 30.11.2011 08:14, Farhan Husain wrote: > > Hi all, > > I am trying to build the latest source from Git in an Ubuntu > machine. The error I am getting is this: > > Making all in . > quiet_libtool: link: gcc > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/src/include > -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 > -DHAVE_VISIBILITY=1 -g -ggdb -DHAVE_OPENSSL -Wall -std=c99 > -pedantic -Wdeclaration-after-statement -g -O2 -pthread > -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE > -I/usr/local/src/freeswitch/libs/apr/include > -I/usr/local/src/freeswitch/libs/apr-util/include > -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib > -I/usr/local/src/freeswitch/libs/stfu > -I/usr/local/src/freeswitch/libs/sqlite > -I/usr/local/src/freeswitch/libs/pcre > -I/usr/local/src/freeswitch/libs/speex/include > -Ilibs/speex/include > -I/usr/local/src/freeswitch/libs/srtp/include > -I/usr/local/src/freeswitch/libs/srtp/crypto/include > -Ilibs/srtp/crypto/include > -I/usr/local/src/freeswitch/libs/spandsp/src > -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff > -DENABLE_SRTP -I/usr/local/src/freeswitch/libs/libedit/src > -DSWITCH_HAVE_LIBEDIT -Ilibs/libedit/src -DSWITCH_HAVE_LIBEDIT > -g -O2 -o .libs/freeswitch freeswitch-switch.o > /usr/lib/libcurl.so -lm -lz ./.libs/libfreeswitch.so > libs/apr/.libs/libapr-1.a -luuid -lrt -ldl -lcrypt -lpthread > libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lncurses > -pthread -Wl,-rpath -Wl,/usr/local/freeswitch/lib > > ./.libs/libfreeswitch.so: undefined reference to `Curl_setopt' > collect2: ld returned 1 exit status > make[2]: *** [freeswitch] Error 1 > > I have installed php-curl (sudo apt-get install php5-curl) but > to no avail. Does anyone have any solution to this? > > > seems like the curl dev package is missing ( libcurl4-dev or > libcurl3-dev) -> sudo aptitude install libcurl4-dev > > - Thomas > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111201/362c582e/attachment.html From acrow at integrafin.co.uk Thu Dec 1 16:29:38 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Thu, 01 Dec 2011 13:29:38 +0000 Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... In-Reply-To: <1322733452381-7049935.post@n2.nabble.com> References: <1322733452381-7049935.post@n2.nabble.com> Message-ID: <4ED78142.6070707@integrafin.co.uk> Hi, I think Tony might have something here. I built a new FS from git of 29/11/2011 and calling between registered extensions is broken. The called phone will ring and I get ringing on the caller, but as soon and the callee goes offhook the caller gets sent to voicemail. Here are is a log snippet of such a call: 2011-12-01 12:58:48.206623 [DEBUG] switch_channel.c:3175 (sofia/internal/sip:1000 at 192.168.44.150:5060) Callstate Change RINGING -> ACTIVE 2011-12-01 12:58:48.206623 [DEBUG] switch_channel.c:3187 Send signal sofia/internal/1001 at 192.168.44.171 [BREAK] 2011-12-01 12:58:48.206623 [NOTICE] sofia.c:6077 Channel [sofia/internal/sip:1000 at 192.168.44.150:5060] has been answered 2011-12-01 12:58:48.206623 [DEBUG] sofia_glue.c:3140 AUDIO RTP [sofia/internal/1001 at 192.168.44.171] 192.168.44.171 port 19398 -> 192.168.44.145 port 5034 codec: 0 ms: 20 2011-12-01 12:58:48.206623 [DEBUG] switch_rtp.c:1642 Starting timer [soft] 160 bytes per 20ms 2011-12-01 12:58:48.206623 [DEBUG] sofia_glue.c:3404 Set 2833 dtmf send payload to 101 2011-12-01 12:58:48.206623 [DEBUG] sofia_glue.c:3410 Set 2833 dtmf receive payload to 101 2011-12-01 12:58:48.206623 [DEBUG] mod_sofia.c:746 Local SDP sofia/internal/1001 at 192.168.44.171: v=0 o=FreeSWITCH 1322724930 1322724931 IN IP4 192.168.44.171 s=FreeSWITCH c=IN IP4 192.168.44.171 t=0 0 m=audio 19398 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 2011-12-01 12:58:48.206623 [DEBUG] switch_core_session.c:726 Send signal sofia/internal/1001 at 192.168.44.171 [BREAK] 2011-12-01 12:58:48.206623 [DEBUG] switch_channel.c:3175 (sofia/internal/1001 at 192.168.44.171) Callstate Change RINGING -> ACTIVE 2011-12-01 12:58:48.206623 [NOTICE] switch_ivr_originate.c:3209 Channel [sofia/internal/1001 at 192.168.44.171] has been answered 2011-12-01 12:58:48.206623 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1001 at 192.168.44.171 [BREAK] 2011-12-01 12:58:48.206623 [DEBUG] sofia.c:5368 Channel sofia/internal/1001 at 192.168.44.171 entering state [completed][200] 2011-12-01 12:58:48.206623 [DEBUG] switch_ivr_originate.c:3269 Originate Resulted in Success: [sofia/internal/sip:1000 at 192.168.44.150:5060] 2011-12-01 12:58:48.206623 [DEBUG] switch_ivr_originate.c:3514 (sofia/internal/sip:1000 at 192.168.44.150:5060) State Change CS_CONSUME_MEDIA -> CS_RESET 2011-12-01 12:58:48.206623 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/sip:1000 at 192.168.44.150:5060 [BREAK] 2011-12-01 12:58:48.206623 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1000 at 192.168.44.150:5060) Running State Change CS_RESET 2011-12-01 12:58:48.206623 [DEBUG] switch_core_state_machine.c:413 (sofia/internal/sip:1000 at 192.168.44.150:5060) State RESET 2011-12-01 12:58:48.206623 [DEBUG] mod_sofia.c:166 sofia/internal/sip:1000 at 192.168.44.150:5060 SOFIA RESET 2011-12-01 12:58:48.206623 [DEBUG] switch_core_state_machine.c:93 sofia/internal/sip:1000 at 192.168.44.150:5060 Standard RESET 2011-12-01 12:58:48.206623 [DEBUG] switch_core_state_machine.c:413 (sofia/internal/sip:1000 at 192.168.44.150:5060) State RESET going to sleep 2011-12-01 12:58:48.226638 [DEBUG] switch_ivr_originate.c:3367 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2011-12-01 12:58:48.226638 [INFO] mod_dptools.c:2897 Originate Failed. Cause: ORIGINATOR_CANCEL This also happens on a call in from freetdm to an extension, as soon as the destination picks up the caller gets voicemail. Very odd. Cheers Alex On 01/12/11 09:57, obbyone wrote: > Hi, > > I've just installed Freeswitch from GIT on Linux UBUNTU 8.04 LTS. > Registering is ok, so every softphone or ATA are registered on the server > BUT, as I try to place a call from a softphone to another, it doesn't work. > > Has someone a solution to my problem, > > Thanks, > > Tony > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-installed-ATA-registered-but-no-call-are-possible-tp7049935p7049935.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 > -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From peter.olsson at visionutveckling.se Thu Dec 1 16:37:04 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 1 Dec 2011 14:37:04 +0100 Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... In-Reply-To: <4ED78142.6070707@integrafin.co.uk> References: <1322733452381-7049935.post@n2.nabble.com> <4ED78142.6070707@integrafin.co.uk> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5B277233E0@cooper> I think this is fixed in latest git. I had a similar problem, and it was fixed yesterday. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Alex Crow Skickat: den 1 december 2011 14:30 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... Hi, I think Tony might have something here. I built a new FS from git of 29/11/2011 and calling between registered extensions is broken. The called phone will ring and I get ringing on the caller, but as soon and the callee goes offhook the caller gets sent to voicemail. Here are is a log snippet of such a call: 2011-12-01 12:58:48.206623 [DEBUG] switch_channel.c:3175 (sofia/internal/sip:1000 at 192.168.44.150:5060) Callstate Change RINGING -> ACTIVE 2011-12-01 12:58:48.206623 [DEBUG] switch_channel.c:3187 Send signal sofia/internal/1001 at 192.168.44.171 [BREAK] 2011-12-01 12:58:48.206623 [NOTICE] sofia.c:6077 Channel [sofia/internal/sip:1000 at 192.168.44.150:5060] has been answered 2011-12-01 12:58:48.206623 [DEBUG] sofia_glue.c:3140 AUDIO RTP [sofia/internal/1001 at 192.168.44.171] 192.168.44.171 port 19398 -> 192.168.44.145 port 5034 codec: 0 ms: 20 2011-12-01 12:58:48.206623 [DEBUG] switch_rtp.c:1642 Starting timer [soft] 160 bytes per 20ms 2011-12-01 12:58:48.206623 [DEBUG] sofia_glue.c:3404 Set 2833 dtmf send payload to 101 2011-12-01 12:58:48.206623 [DEBUG] sofia_glue.c:3410 Set 2833 dtmf receive payload to 101 2011-12-01 12:58:48.206623 [DEBUG] mod_sofia.c:746 Local SDP sofia/internal/1001 at 192.168.44.171: v=0 o=FreeSWITCH 1322724930 1322724931 IN IP4 192.168.44.171 s=FreeSWITCH c=IN IP4 192.168.44.171 t=0 0 m=audio 19398 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 2011-12-01 12:58:48.206623 [DEBUG] switch_core_session.c:726 Send signal sofia/internal/1001 at 192.168.44.171 [BREAK] 2011-12-01 12:58:48.206623 [DEBUG] switch_channel.c:3175 (sofia/internal/1001 at 192.168.44.171) Callstate Change RINGING -> ACTIVE 2011-12-01 12:58:48.206623 [NOTICE] switch_ivr_originate.c:3209 Channel [sofia/internal/1001 at 192.168.44.171] has been answered 2011-12-01 12:58:48.206623 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1001 at 192.168.44.171 [BREAK] 2011-12-01 12:58:48.206623 [DEBUG] sofia.c:5368 Channel sofia/internal/1001 at 192.168.44.171 entering state [completed][200] 2011-12-01 12:58:48.206623 [DEBUG] switch_ivr_originate.c:3269 Originate Resulted in Success: [sofia/internal/sip:1000 at 192.168.44.150:5060] 2011-12-01 12:58:48.206623 [DEBUG] switch_ivr_originate.c:3514 (sofia/internal/sip:1000 at 192.168.44.150:5060) State Change CS_CONSUME_MEDIA -> CS_RESET 2011-12-01 12:58:48.206623 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/sip:1000 at 192.168.44.150:5060 [BREAK] 2011-12-01 12:58:48.206623 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1000 at 192.168.44.150:5060) Running State Change CS_RESET 2011-12-01 12:58:48.206623 [DEBUG] switch_core_state_machine.c:413 (sofia/internal/sip:1000 at 192.168.44.150:5060) State RESET 2011-12-01 12:58:48.206623 [DEBUG] mod_sofia.c:166 sofia/internal/sip:1000 at 192.168.44.150:5060 SOFIA RESET 2011-12-01 12:58:48.206623 [DEBUG] switch_core_state_machine.c:93 sofia/internal/sip:1000 at 192.168.44.150:5060 Standard RESET 2011-12-01 12:58:48.206623 [DEBUG] switch_core_state_machine.c:413 (sofia/internal/sip:1000 at 192.168.44.150:5060) State RESET going to sleep 2011-12-01 12:58:48.226638 [DEBUG] switch_ivr_originate.c:3367 Originate Resulted in Error Cause: 487 [ORIGINATOR_CANCEL] 2011-12-01 12:58:48.226638 [INFO] mod_dptools.c:2897 Originate Failed. Cause: ORIGINATOR_CANCEL This also happens on a call in from freetdm to an extension, as soon as the destination picks up the caller gets voicemail. Very odd. Cheers Alex On 01/12/11 09:57, obbyone wrote: > Hi, > > I've just installed Freeswitch from GIT on Linux UBUNTU 8.04 LTS. > Registering is ok, so every softphone or ATA are registered on the > server BUT, as I try to place a call from a softphone to another, it doesn't work. > > Has someone a solution to my problem, > > Thanks, > > Tony > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-installed-ATA > -registered-but-no-call-are-possible-tp7049935p7049935.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-use > rs > http://www.freeswitch.org > -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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:4ed7807532761918531564! From acrow at integrafin.co.uk Thu Dec 1 16:47:33 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Thu, 01 Dec 2011 13:47:33 +0000 Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5B277233E0@cooper> References: <1322733452381-7049935.post@n2.nabble.com> <4ED78142.6070707@integrafin.co.uk> <549CFEF87AEDE841A38E9D15EAB4C04C5B277233E0@cooper> Message-ID: <4ED78575.2080702@integrafin.co.uk> On 01/12/11 13:37, Peter Olsson wrote: > I think this is fixed in latest git. I had a similar problem, and it was fixed yesterday. > > /Peter > > > -----Ursprungligt meddelande----- Peter, Yes, I just found FS3727. Cheers! Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From daveh at beachdognet.com Thu Dec 1 18:02:56 2011 From: daveh at beachdognet.com (Dave Horton) Date: Thu, 1 Dec 2011 10:02:56 -0500 Subject: [Freeswitch-users] is 100rel still a bad idea? In-Reply-To: <40941536-3617-4EA4-8C24-B5B441B19CE9@beachdognet.com> References: <40941536-3617-4EA4-8C24-B5B441B19CE9@beachdognet.com> Message-ID: <81BD8082-92E3-47B1-9D4D-21BFA3BD088B@beachdognet.com> anyone...? On Nov 30, 2011, at 11:48 AM, Dave Horton wrote: I saw a few threads from a couple years back indicating the enabling reliable provisional responses could cause a FS crash. In fact, I am working with a customer who was told some time ago (by "freeswitch support", not sure exactly who they are referring to) to turn off 100rel (this while they were troubleshooting an unrelated issue). Just looking for an update on this, because I prefer to run my SIP gear with 100rel supported --- is there still a known bug (or even, suspected issue) with 100rel support on FS? From freeswitch at peely.com Thu Dec 1 18:15:33 2011 From: freeswitch at peely.com (peely) Date: Thu, 1 Dec 2011 07:15:33 -0800 (PST) Subject: [Freeswitch-users] is 100rel still a bad idea? In-Reply-To: <81BD8082-92E3-47B1-9D4D-21BFA3BD088B@beachdognet.com> References: <40941536-3617-4EA4-8C24-B5B441B19CE9@beachdognet.com> <81BD8082-92E3-47B1-9D4D-21BFA3BD088B@beachdognet.com> Message-ID: <1322752533542-7050888.post@n2.nabble.com> I haven't seen any issues with it in terms of the FreeSWITCH side, but regularly have to turn it off because it screws up many other device's view of the offer / answer model. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/is-100rel-still-a-bad-idea-tp7047542p7050888.html Sent from the freeswitch-users mailing list archive at Nabble.com. From freeswitch at peely.com Thu Dec 1 18:24:16 2011 From: freeswitch at peely.com (peely) Date: Thu, 1 Dec 2011 07:24:16 -0800 (PST) Subject: [Freeswitch-users] mod_rtmp: "Read error" when attempting calls from FS to RTMP client. In-Reply-To: <1322081549623-7026117.post@n2.nabble.com> References: <1322081549623-7026117.post@n2.nabble.com> Message-ID: <1322753056997-7050910.post@n2.nabble.com> Sorry, can I bump this? I've tried everything I'm capable of, but can't get a call from mod_rtmp out to the client without keeping the box logging in debug mode! As soon as I bring the log level down I get the issue, but of course can't get any more debug for this specific issue. I've looked at the code but don't know enough C to make any changes. Thanks, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-rtmp-Read-error-when-attempting-calls-from-FS-to-RTMP-client-tp7026117p7050910.html Sent from the freeswitch-users mailing list archive at Nabble.com. From curriegrad2004 at gmail.com Thu Dec 1 18:30:37 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Thu, 1 Dec 2011 07:30:37 -0800 Subject: [Freeswitch-users] Build error: undefined reference to `Curl_setopt' In-Reply-To: <4ED77FA9.5020308@gmail.com> References: <4ED5EEB8.9030709@chaschperli.ch> <4ED77FA9.5020308@gmail.com> Message-ID: Or what you could do is configure the code with the --without-libcurl switch. This makes FreeSWITCH to ignore the system installed CURL library and use it's own bundled libcurl instead. I'd personally recommend you to do this instead as the later versions of libcurl may have broken a few things that FS requires On Thu, Dec 1, 2011 at 5:22 AM, Tamas.Cseke wrote: > It has been fixed. try latest git > > > On 2011-12-01 05:56, Farhan Husain wrote: > > Unfortunately this did not help. I installed the library. The Makefile > already had -lcurl in several places. I tried to add -lcurl4-dev, -lcurl-dev > and -lcurldev. The linker could not find any of those. Is there any other > name for the library or -lcurl includes the libcurl4-dev library? > > On Wed, Nov 30, 2011 at 12:52 AM, Thomas Mueller > wrote: >> >> On 30.11.2011 08:14, Farhan Husain wrote: >>> >>> Hi all, >>> >>> I am trying to build the latest source from Git in an Ubuntu machine. The >>> error I am getting is this: >>> >>> Making all in . >>> quiet_libtool: link: gcc -I/usr/local/src/freeswitch/src/include >>> -I/usr/local/src/freeswitch/src/include >>> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror >>> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g -ggdb >>> -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement -g -O2 >>> -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE >>> -I/usr/local/src/freeswitch/libs/apr/include >>> -I/usr/local/src/freeswitch/libs/apr-util/include >>> -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib >>> -I/usr/local/src/freeswitch/libs/stfu >>> -I/usr/local/src/freeswitch/libs/sqlite >>> -I/usr/local/src/freeswitch/libs/pcre >>> -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include >>> -I/usr/local/src/freeswitch/libs/srtp/include >>> -I/usr/local/src/freeswitch/libs/srtp/crypto/include >>> -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src >>> -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -DENABLE_SRTP >>> -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT >>> -Ilibs/libedit/src -DSWITCH_HAVE_LIBEDIT -g -O2 -o .libs/freeswitch >>> freeswitch-switch.o ?/usr/lib/libcurl.so -lm -lz ./.libs/libfreeswitch.so >>> libs/apr/.libs/libapr-1.a -luuid -lrt -ldl -lcrypt -lpthread >>> libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lncurses -pthread >>> -Wl,-rpath -Wl,/usr/local/freeswitch/lib >>> >>> ./.libs/libfreeswitch.so: undefined reference to `Curl_setopt' >>> collect2: ld returned 1 exit status >>> make[2]: *** [freeswitch] Error 1 >>> >>> I have installed php-curl (sudo apt-get install php5-curl) but to no >>> avail. Does anyone have any solution to this? >> >> >> seems like the curl dev package is missing ( libcurl4-dev or libcurl3-dev) >> -> sudo aptitude install libcurl4-dev >> >> - Thomas > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Dec 1 18:50:51 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 1 Dec 2011 09:50:51 -0600 Subject: [Freeswitch-users] Build error: undefined reference to `Curl_setopt' In-Reply-To: References: <4ED5EEB8.9030709@chaschperli.ch> <4ED77FA9.5020308@gmail.com> Message-ID: latest GIT now detects this missing symbol and auto-enables our built in one. On Thu, Dec 1, 2011 at 9:30 AM, curriegrad2004 wrote: > Or what you could do is configure the code with the --without-libcurl > switch. This makes FreeSWITCH to ignore the system installed CURL > library and use it's own bundled libcurl instead. I'd personally > recommend you to do this instead as the later versions of libcurl may > have broken a few things that FS requires > > On Thu, Dec 1, 2011 at 5:22 AM, Tamas.Cseke > wrote: > > It has been fixed. try latest git > > > > > > On 2011-12-01 05:56, Farhan Husain wrote: > > > > Unfortunately this did not help. I installed the library. The Makefile > > already had -lcurl in several places. I tried to add -lcurl4-dev, > -lcurl-dev > > and -lcurldev. The linker could not find any of those. Is there any other > > name for the library or -lcurl includes the libcurl4-dev library? > > > > On Wed, Nov 30, 2011 at 12:52 AM, Thomas Mueller > > wrote: > >> > >> On 30.11.2011 08:14, Farhan Husain wrote: > >>> > >>> Hi all, > >>> > >>> I am trying to build the latest source from Git in an Ubuntu machine. > The > >>> error I am getting is this: > >>> > >>> Making all in . > >>> quiet_libtool: link: gcc -I/usr/local/src/freeswitch/src/include > >>> -I/usr/local/src/freeswitch/src/include > >>> -I/usr/local/src/freeswitch/libs/libteletone/src -fPIC -Werror > >>> -fvisibility=hidden -DSWITCH_API_VISIBILITY=1 -DHAVE_VISIBILITY=1 -g > -ggdb > >>> -DHAVE_OPENSSL -Wall -std=c99 -pedantic -Wdeclaration-after-statement > -g -O2 > >>> -pthread -DLINUX=2 -D_REENTRANT -D_GNU_SOURCE > >>> -I/usr/local/src/freeswitch/libs/apr/include > >>> -I/usr/local/src/freeswitch/libs/apr-util/include > >>> -I/usr/local/src/freeswitch/libs/apr-util/xml/expat/lib > >>> -I/usr/local/src/freeswitch/libs/stfu > >>> -I/usr/local/src/freeswitch/libs/sqlite > >>> -I/usr/local/src/freeswitch/libs/pcre > >>> -I/usr/local/src/freeswitch/libs/speex/include -Ilibs/speex/include > >>> -I/usr/local/src/freeswitch/libs/srtp/include > >>> -I/usr/local/src/freeswitch/libs/srtp/crypto/include > >>> -Ilibs/srtp/crypto/include -I/usr/local/src/freeswitch/libs/spandsp/src > >>> -I/usr/local/src/freeswitch/libs/tiff-3.8.2/libtiff -DENABLE_SRTP > >>> -I/usr/local/src/freeswitch/libs/libedit/src -DSWITCH_HAVE_LIBEDIT > >>> -Ilibs/libedit/src -DSWITCH_HAVE_LIBEDIT -g -O2 -o .libs/freeswitch > >>> freeswitch-switch.o /usr/lib/libcurl.so -lm -lz > ./.libs/libfreeswitch.so > >>> libs/apr/.libs/libapr-1.a -luuid -lrt -ldl -lcrypt -lpthread > >>> libs/libedit/src/.libs/libedit.a -lssl -lcrypto -lncurses -pthread > >>> -Wl,-rpath -Wl,/usr/local/freeswitch/lib > >>> > >>> ./.libs/libfreeswitch.so: undefined reference to `Curl_setopt' > >>> collect2: ld returned 1 exit status > >>> make[2]: *** [freeswitch] Error 1 > >>> > >>> I have installed php-curl (sudo apt-get install php5-curl) but to no > >>> avail. Does anyone have any solution to this? > >> > >> > >> seems like the curl dev package is missing ( libcurl4-dev or > libcurl3-dev) > >> -> sudo aptitude install libcurl4-dev > >> > >> - Thomas > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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/20111201/f9b50fb6/attachment.html From charlie.orford at attackplan.net Thu Dec 1 19:17:19 2011 From: charlie.orford at attackplan.net (Charlie Orford) Date: Thu, 01 Dec 2011 17:17:19 +0100 Subject: [Freeswitch-users] "481 Call/Transaction Does Not Exist" when hanging up before call connects Message-ID: <4ED7A88F.1010302@attackplan.net> Hi list, When we make a call from an FS extension to a PSTN number (via our ITSP gateway provider) and hang-up before the call completes, FS replies to the CANCEL request with "481 Call/Transaction Does Not Exist" and the call continues to ring on the remote end. If we hangup after the call has connected, it works with no problem. I have looked through a SIP trace of this happening and to my (untrained eye) nothing seems obviously wrong (i.e. tag, call id and branch values all seem to be correct). I'm using the latest git snapshot from 2011-11-30 18-14-24 -0600. For a sip trace showing the problem, see: http://pastebin.freeswitch.org/17908 For a sip trace showing a successful hangup, see: http://pastebin.freeswitch.org/17909 *Note: ip addresses, domain and called number have been altered for privacy. Any help or insight is much appreciated. Kind Regards, Charlie From daveh at beachdognet.com Thu Dec 1 19:28:16 2011 From: daveh at beachdognet.com (Dave Horton) Date: Thu, 1 Dec 2011 11:28:16 -0500 Subject: [Freeswitch-users] is 100rel still a bad idea? In-Reply-To: <81BD8082-92E3-47B1-9D4D-21BFA3BD088B@beachdognet.com> References: <40941536-3617-4EA4-8C24-B5B441B19CE9@beachdognet.com> <81BD8082-92E3-47B1-9D4D-21BFA3BD088B@beachdognet.com> Message-ID: I guess I am worried about this reference on the current wiki page: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#enable-100rel The offhand "Enabling this may cause FreeSWITCH to crash" is not a confidence-builder..... On Nov 30, 2011, at 11:48 AM, Dave Horton wrote: I saw a few threads from a couple years back indicating the enabling reliable provisional responses could cause a FS crash. In fact, I am working with a customer who was told some time ago (by "freeswitch support", not sure exactly who they are referring to) to turn off 100rel (this while they were troubleshooting an unrelated issue). Just looking for an update on this, because I prefer to run my SIP gear with 100rel supported --- is there still a known bug (or even, suspected issue) with 100rel support on FS? From mstockton at harqen.com Thu Dec 1 19:50:26 2011 From: mstockton at harqen.com (Matt Stockton) Date: Thu, 1 Dec 2011 10:50:26 -0600 Subject: [Freeswitch-users] mod_rtmp: "Read error" when attempting calls from FS to RTMP client. In-Reply-To: <1322753056997-7050910.post@n2.nabble.com> References: <1322081549623-7026117.post@n2.nabble.com> <1322753056997-7050910.post@n2.nabble.com> Message-ID: For what it's worth, I am getting the exact same read error and it is closing my socket when I dial into FS from RTMP client, but only in certain browsers it seems. It's a read error on the same line of code. Description of the issue and debug lines are listed in this JIRA: http://jira.freeswitch.org/browse/FS-3729 On Thu, Dec 1, 2011 at 9:24 AM, peely wrote: > Sorry, can I bump this? > > I've tried everything I'm capable of, but can't get a call from mod_rtmp > out > to the client without keeping the box logging in debug mode! As soon as I > bring the log level down I get the issue, but of course can't get any more > debug for this specific issue. > > I've looked at the code but don't know enough C to make any changes. > > > Thanks, > > > Neil. > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/mod-rtmp-Read-error-when-attempting-calls-from-FS-to-RTMP-client-tp7026117p7050910.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/20111201/ef95ed57/attachment.html From msc at freeswitch.org Thu Dec 1 20:58:07 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 1 Dec 2011 09:58:07 -0800 Subject: [Freeswitch-users] Problem send_dtmf send only to a-leg In-Reply-To: References: Message-ID: Sounds like the early media is fooling queue_dtmf. You could try using execute_on_answer and send_dtmf together. -MC On Wed, Nov 30, 2011 at 6:39 PM, Thomas Hoellriegel wrote: > Hi Michael, > > Thanks for you reply. > Now the following problem occurs: > The queue_dtmf sent to bleg. But queue_dtmf send the digits before the > channel is answerd. > The characters are sent to the channel, while it ringing. > > sent the tones to aleg. > send_dtmf with this construct, is not working. > > > > --------------- > 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/20111201/cfce5047/attachment.html From tonybecq at yahoo.fr Thu Dec 1 22:11:46 2011 From: tonybecq at yahoo.fr (obbyone) Date: Thu, 1 Dec 2011 11:11:46 -0800 (PST) Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... In-Reply-To: <4ED78142.6070707@integrafin.co.uk> References: <1322733452381-7049935.post@n2.nabble.com> <4ED78142.6070707@integrafin.co.uk> Message-ID: <1322766706097-7051703.post@n2.nabble.com> I'm sorry but in my case, the phone that is called doesn't ring. It only goes to the voicemail -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-installed-ATA-registered-but-no-call-are-possible-tp7049935p7051703.html Sent from the freeswitch-users mailing list archive at Nabble.com. From huw.selley at netdev.co.uk Thu Dec 1 14:06:44 2011 From: huw.selley at netdev.co.uk (Huw Selley) Date: Thu, 1 Dec 2011 11:06:44 +0000 Subject: [Freeswitch-users] Max sessions for a javascript app? Message-ID: Hi, This is my first post here so please be gentle :) First of all thanks for making such an awesome product! I have a question about the maximum number of sessions with media that I can support from a javascript app. My freeswitch box is CentOS 5.6 x86_64 running on a dell R410 with dual quad core X5650 @ 2.67GHz and 32GB of ram. I am using freeswitch commit 1ea5b3cf62ff99f07a21a70196f8a684bd5333e3 which I built yesterday. I have modified switch.xml to raise max-sessions and sessions-per-second to quite high values (100000 and 1000 respectively). Freeswitch is being started with the -nonat and -hp flags and I have applied the ulimits recommendedinhttp://wiki.freeswitch.org/wiki/Performance_testing_and_configurations. The call sender and receivers are both on slightly older dell machines with dual quad core xeon cpus and 16GB RAM. The servers are all on a gigabit ethernet network. I have a simple test setup with a sipp uac sending ingress into freeswitch, the dialplan executes a simple javascript app that just performs a bridge. The outbound gateway points at a sipp uas to answer the call. Each call has a hold time of 300s then the uac ends the call. My uac script is sending in 10cps to a total of 12000 calls. The javascript app is: session.execute("bridge", "sofia/gateway/sipp/argv[0]"); I have noticed that when using the js app I don't appear to be able to get >3001 inbound calls. Once I have reached 3001 inbound calls (so 6002 sessions including the b-leg) freeswitch responds to each new INVITE with a 100 trying message but the call doesn't progress. I have also performed the same test but using the bridge application in the dialplan instead of calling the javascript and it happily runs through to 12000 calls. The reason I am querying this is because I want to use a more complex js app so I can't just use the bridge dialplan app. Am i doing it wrong? Any advice gratefully received. Thanks Huw From Giovanni.Visciano at italtel.it Thu Dec 1 18:35:54 2011 From: Giovanni.Visciano at italtel.it (Visciano Giovanni) Date: Thu, 1 Dec 2011 16:35:54 +0100 Subject: [Freeswitch-users] R: Sofia late-negotiation on re-INVITE(codec-modification) References: <3E991E54-B2A4-4B97-B74D-3B0EC2227B29@freeswitch.org> Message-ID: Finally we are back to our test. I updated my FS installation to last GIT (FreeSWITCH Version 1.0.head (git-eae86e0 2011-11-30 18-14-24 -0600)) TEST ---- SIP vs SIP basic audio call, then re-INVITE for codec modification. FS configuration B2B, avoid transcoding. CONF ---- In sofia SIP profile I have: and in my dialplan XML I hit: Loaded codec modules: freeswitch at internal> show codec type,name,ikey codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.729,mod_g729 codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE TEST 1) ------- 1001 ----invite(pcma)----> FS --invite(pcma)--> 1000 1001 <----200OK(pcma)---- FS <--200OK(pcma)-- 1000 1001 --re/invite(g729)---> FS 1001 <----488------------ FS Full log: http://pastebin.freeswitch.org/17906 Note: FS does not negotiate end to end the reINVITE O/A codec modification. It is closed locally on the 1001->FS leg. From the log I see 2011-12-01 15:54:13.871332 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2011-12-01 15:54:13.871332 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] 2011-12-01 15:54:13.871332 [ERR] sofia.c:5876 Reinvite Codec Error! TEST 2) ------- 1000 ----invite(pcma)----> FS --invite(pcma)--> 1001 1000 <----200OK(pcma)---- FS <--200OK(pcma)-- 1001 1000 FS <--re/invite(g729)- 1001 1000 FS ------------------> 1001 Full log: http://pastebin.freeswitch.org/17907 Note: FS does not negotiate end to end the reINVITE O/A codec modification. It is closed locally on the 1001->FS leg. FS select locally G729 but this result in transcoding! I hate transcoding! >From the log I see 2011-12-01 16:00:12.971244 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2011-12-01 16:00:12.971244 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [G729:18:8000:20:8000]/[G729:18:8000:20:8000] 2011-12-01 16:00:12.971244 [DEBUG] sofia_glue.c:2806 Changing Codec from PCMA at 20ms@8000hz to G729 at 20ms@8000hz 2011-12-01 16:00:13.011337 [DEBUG] sofia_glue.c:2888 Set Codec sofia/internal/1001 at 138.132.110.64:5070 G729/8000 20 ms 160 samples 8000 bits 2011-12-01 16:00:13.011337 [DEBUG] switch_core_codec.c:116 sofia/internal/1001 at 138.132.110.64:5070 Push codec G729:18 2011-12-01 16:00:13.031252 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode! 2011-12-01 16:00:13.031252 [ERR] switch_core_io.c:1077 Codec G.729 encoder error! Regard, Giovanni PS: next week I won't be at work. My collegue Nevio will follow the problem, so if you need more data/details just tell us. Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111201/93585706/attachment-0001.html From monroyricardo at gmail.com Thu Dec 1 09:05:47 2011 From: monroyricardo at gmail.com (Ricardo Monroy) Date: Thu, 1 Dec 2011 06:05:47 +0000 (UTC) Subject: [Freeswitch-users] =?utf-8?q?google_voice_connection_going_offlin?= =?utf-8?b?ZQkobW9kX2RpbmdhbGluZyk=?= References: <1317265876.9229.YahooMailNeo@web39703.mail.mud.yahoo.com> <1317308654166-6844212.post@n2.nabble.com> Message-ID: Hi Federico and Anthony, Were you guys able to come up with a solution for this? I had mine working for about a month without any problems. I'm running the git version from October 1, 2011. I just ran into the same problem after changing my router since I changed my Internet connection from DSL to fiber optic. Reading the comments from this post: http://www.personal.psu.edu/wcs131/blogs/psuvoip/2010/10/using_freeswitch_to_add _google.html I guess this is an issue with Freeswitch having problems when there are Internet disconnections. I guess one way to solve this would be to change my router, but I don't really want to buy another ONT router, besides I don't even know if my ISP would let me use a different ONT since the ones that they provide are all "customized". It is really hard to tell when it has disconnected. Even when typing "dingaling status" from the fs_cli, it will show AUTHORIZED when it is disconnected. The only way to tell if it's been disconnected is checking whether I can make calls or not. To solve this ,without having to restart freeswitch I would "disconnect" (according to FS) from my gtalk profile typing "dl_logout gtalk" and reconnect with "dl_login profile=gtalk". I also had the idea to use the "cron" approach and thought this was going to be the solution to all my problems. Of course, I had to test what would happen if the cron job would happen to run in the middle of a call. I manually typed the commands on the fs_cli while I was on a call and my call did not die!! Great!! I had the solution to the problem!! I then finished the call and ooohh surprise!! I was suddenly disconnected from the fs_cli, FS would die, close unexpectedly and create a core dump. I would really like to help to solve this so let me know if there's any way I can help out. I would like to try the chat approach to see if that works. Did you figure out what is the right syntax to chat? Greetings, Ricardo From monroyricardo at gmail.com Thu Dec 1 19:47:27 2011 From: monroyricardo at gmail.com (Ricardo Monroy) Date: Thu, 1 Dec 2011 16:47:27 +0000 (UTC) Subject: [Freeswitch-users] =?utf-8?q?google_voice_connection_going_offlin?= =?utf-8?b?ZQkobW9kX2RpbmdhbGluZyk=?= References: <1317265876.9229.YahooMailNeo@web39703.mail.mud.yahoo.com> <1317308654166-6844212.post@n2.nabble.com> Message-ID: Federico Beffa writes: > > Hi Anthony, > > thanks for the clarification and the suggestion. As an immediate > workaround I would like to try the "cron" approach you suggested. I > have therefore experimented a little bit with the chat api command. > > My jingle_profiles/client.xml looks essentially as the one provided > with the example configuration: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > When I try to use the chat command at the fs_cli prompt I'm unable to > form a full jid and always get an error message from the remote > server. Here an example: > > chat jingle|888 xmppc|user2 at ...|test > > Sent > freeswitch ubuntu-vbox-t5500> 2011-10-08 14:22:09.804596 [NOTICE] > libdingaling.c:1373 SecSEND: > ------------------------------------------------------------------------------ - > > test > > > 2011-10-08 14:22:09.844698 [INFO] libdingaling.c:1371 SecRECV: > ------------------------------------------------------------------------------ - > > test > > > If set, the > 'from' attribute must be set to the user's full JID. > > > > "xmppc" is the value of the variable "xmpp_client_profile" as in the > default config. I get the same error independently from what I set > before "@xmppc" in the from_jid. Differently from this, if as the > from_jid I set my explicit jid "user1 at .../talk" as in the > client.xml file I get the error: > > 2011-10-08 14:20:50.446696 [ERR] mod_dingaling.c:544 Invalid Profile gmail.com > > Despite the fact that even a wrong message reaching the remote server > could work as a "keep alive" signal, I still would like to understand > the proper syntax. > I would be grateful if somebody could explain me what is the correct > syntax to use with chat and jingle? > > Thank you! > Fede > > P.S.: By the way, I'm now working on HEAD. > > Hi Federico and Anthony, Were you guys able to come up with a solution for this? I had mine working for about a month without any problems. I'm running the git version from October 1, 2011. I just ran into the same problem after changing my router since I changed my Internet connection from DSL to fiber optic. Reading the comments from this post: http://www.personal.psu.edu/wcs131/blogs/psuvoip/2010/10/using_freeswitch_to_add _google.html I guess this is an issue with Freeswitch having problems when there are Internet disconnections. I guess one way to solve this would be to change my router, but I don't really want to buy another ONT router, besides I don't even know if my ISP would let me use a different ONT since the ones that they provide are all "customized". It is really hard to tell when it has disconnected. Even when typing "dingaling status" from the fs_cli, it will show AUTHORIZED when it is disconnected. The only way to tell if it's been disconnected is checking whether I can make calls or not. To solve this ,without having to restart freeswitch I would "disconnect" (according to FS) from my gtalk profile typing "dl_logout gtalk" and reconnect with "dl_login profile=gtalk". I also had the idea to use the "cron" approach and thought this was going to be the solution to all my problems. Of course, I had to test what would happen if the cron job would happen to run in the middle of a call. I manually typed the commands on the fs_cli while I was on a call and my call did not die!! Great!! I had the solution to the problem!! I then finished the call and ooohh surprise!! I was suddenly disconnected from the fs_cli, FS would die, close unexpectedly and create a core dump. I would really like to help to solve this so let me know if there's any way I can help out. I would like to try the chat approach to see if that works. Did you figure out what is the right syntax to chat? Greetings, Ricardo From anthony.minessale at gmail.com Thu Dec 1 23:53:22 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 1 Dec 2011 14:53:22 -0600 Subject: [Freeswitch-users] "481 Call/Transaction Does Not Exist" when hanging up before call connects In-Reply-To: <4ED7A88F.1010302@attackplan.net> References: <4ED7A88F.1010302@attackplan.net> Message-ID: try same failed call test with sofia loglevel all 9 Also, try some other phone or device that does not have the problem, and create a cancel situation the same way and see if you can find a difference. Finally, have you tried the latest firmware on the phone? On Thu, Dec 1, 2011 at 10:17 AM, Charlie Orford < charlie.orford at attackplan.net> wrote: > Hi list, > > When we make a call from an FS extension to a PSTN number (via our ITSP > gateway provider) and hang-up before the call completes, FS replies to > the CANCEL request with "481 Call/Transaction Does Not Exist" and the > call continues to ring on the remote end. If we hangup after the call > has connected, it works with no problem. > > I have looked through a SIP trace of this happening and to my (untrained > eye) nothing seems obviously wrong (i.e. tag, call id and branch values > all seem to be correct). I'm using the latest git snapshot from > 2011-11-30 18-14-24 -0600. > > For a sip trace showing the problem, see: > http://pastebin.freeswitch.org/17908 > > For a sip trace showing a successful hangup, see: > http://pastebin.freeswitch.org/17909 > > *Note: ip addresses, domain and called number have been altered for > privacy. > > Any help or insight is much appreciated. > > Kind Regards, > Charlie > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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/20111201/6af08e93/attachment.html From mstockton at harqen.com Fri Dec 2 01:35:39 2011 From: mstockton at harqen.com (Matt Stockton) Date: Thu, 1 Dec 2011 16:35:39 -0600 Subject: [Freeswitch-users] mod_rtmp: "Read error" when attempting calls from FS to RTMP client. In-Reply-To: References: <1322081549623-7026117.post@n2.nabble.com> <1322753056997-7050910.post@n2.nabble.com> Message-ID: Also worthwhile to note that the 'fsctl loglevel 9' does not make it work for me, and the read error occurs still: 2011-12-01 16:34:39.226836 [ERR] rtmp.c:713 Read error Also, even on Chrome (which is working), the read error does occur upon hangup On Thu, Dec 1, 2011 at 10:50 AM, Matt Stockton wrote: > For what it's worth, I am getting the exact same read error and it is > closing my socket when I dial into FS from RTMP client, but only in certain > browsers it seems. It's a read error on the same line of code. > > Description of the issue and debug lines are listed in this JIRA: > http://jira.freeswitch.org/browse/FS-3729 > > > On Thu, Dec 1, 2011 at 9:24 AM, peely wrote: > >> Sorry, can I bump this? >> >> I've tried everything I'm capable of, but can't get a call from mod_rtmp >> out >> to the client without keeping the box logging in debug mode! As soon as I >> bring the log level down I get the issue, but of course can't get any more >> debug for this specific issue. >> >> I've looked at the code but don't know enough C to make any changes. >> >> >> Thanks, >> >> >> Neil. >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/mod-rtmp-Read-error-when-attempting-calls-from-FS-to-RTMP-client-tp7026117p7050910.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/20111201/1bb50223/attachment-0001.html From djbinter at gmail.com Fri Dec 2 02:34:36 2011 From: djbinter at gmail.com (DJB International) Date: Thu, 1 Dec 2011 15:34:36 -0800 Subject: [Freeswitch-users] MOH in multi-tenant PBX Message-ID: I ran into a problem when I want to have a separate music on hold for multi-tenant only when using hold button on Polycom phone because when polycom phone send re-invite to FS with no media for call hold; for some reason, FS takes the default hold-music value from sip profile parameter. I have set the hold_music variable on top of the dialplan, and it works for other stuffs like transfer_ringback, etc. But, the problem only happened when using hold button from the phone. Any suggestions would be appreciate. -djbinter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111201/93d0ab7a/attachment.html From brian at freeswitch.org Fri Dec 2 02:41:58 2011 From: brian at freeswitch.org (Brian West) Date: Thu, 1 Dec 2011 17:41:58 -0600 Subject: [Freeswitch-users] MOH in multi-tenant PBX In-Reply-To: References: Message-ID: <1B62CD93-ABD5-43C6-B0C3-7D564C0FAA27@freeswitch.org> bridge_export the value. /b On Dec 1, 2011, at 5:34 PM, DJB International wrote: > I ran into a problem when I want to have a separate music on hold for > multi-tenant only when using hold button on Polycom phone because when > polycom phone send re-invite to FS with no media for call hold; for some > reason, FS takes the default hold-music value from sip profile parameter. > I have set the hold_music variable on top of the dialplan, and it works for > other stuffs like transfer_ringback, etc. But, the problem only happened > when using hold button from the phone. Any suggestions would be appreciate. > > -djbinter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111201/03d44f25/attachment.html From djbinter at gmail.com Fri Dec 2 03:06:12 2011 From: djbinter at gmail.com (DJB International) Date: Thu, 1 Dec 2011 16:06:12 -0800 Subject: [Freeswitch-users] MOH in multi-tenant PBX In-Reply-To: <1B62CD93-ABD5-43C6-B0C3-7D564C0FAA27@freeswitch.org> References: <1B62CD93-ABD5-43C6-B0C3-7D564C0FAA27@freeswitch.org> Message-ID: Thank you, Brian. It works great. I was unaware about this bridge_export :) Wiki has also been updated: http://wiki.freeswitch.org/wiki/Variable_hold_music#hold_music -djbinter On Thu, Dec 1, 2011 at 3:41 PM, Brian West wrote: > bridge_export the value. > > /b > > On Dec 1, 2011, at 5:34 PM, DJB International wrote: > > I ran into a problem when I want to have a separate music on hold for > multi-tenant only when using hold button on Polycom phone because when > polycom phone send re-invite to FS with no media for call hold; for some > reason, FS takes the default hold-music value from sip profile parameter. > I have set the hold_music variable on top of the dialplan, and it works for > other stuffs like transfer_ringback, etc. But, the problem only happened > when using hold button from the phone. Any suggestions would be > appreciate. > > -djbinter > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111201/03221ab1/attachment.html From brian at freeswitch.org Fri Dec 2 03:12:33 2011 From: brian at freeswitch.org (Brian West) Date: Thu, 1 Dec 2011 18:12:33 -0600 Subject: [Freeswitch-users] MOH in multi-tenant PBX In-Reply-To: References: <1B62CD93-ABD5-43C6-B0C3-7D564C0FAA27@freeswitch.org> Message-ID: <8D793E27-F481-4585-BD51-07129518B3A4@freeswitch.org> See question one on the FAQ! /b On Dec 1, 2011, at 6:06 PM, DJB International wrote: > Thank you, Brian. It works great. I was unaware about this bridge_export > :) > > Wiki has also been updated: > http://wiki.freeswitch.org/wiki/Variable_hold_music#hold_music > > -djbinter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111201/2b6c6606/attachment.html From tonybecq at yahoo.fr Fri Dec 2 03:55:02 2011 From: tonybecq at yahoo.fr (obbyone) Date: Thu, 1 Dec 2011 16:55:02 -0800 (PST) Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5B277233E0@cooper> References: <1322733452381-7049935.post@n2.nabble.com> <4ED78142.6070707@integrafin.co.uk> <549CFEF87AEDE841A38E9D15EAB4C04C5B277233E0@cooper> Message-ID: <1322787302866-7053026.post@n2.nabble.com> I'm sorry but in my case, the phone that is called doesn't ring. It only goes to the voicemail. Thanks in advance... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-installed-ATA-registered-but-no-call-are-possible-tp7049935p7053026.html Sent from the freeswitch-users mailing list archive at Nabble.com. From tonybecq at yahoo.fr Fri Dec 2 03:56:29 2011 From: tonybecq at yahoo.fr (obbyone) Date: Thu, 1 Dec 2011 16:56:29 -0800 (PST) Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... In-Reply-To: References: <1322733452381-7049935.post@n2.nabble.com> Message-ID: <1322787389063-7053032.post@n2.nabble.com> Can it come from a bad handling of ACL ? Thanks... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-installed-ATA-registered-but-no-call-are-possible-tp7049935p7053032.html Sent from the freeswitch-users mailing list archive at Nabble.com. From tonybecq at yahoo.fr Fri Dec 2 11:54:16 2011 From: tonybecq at yahoo.fr (obbyone) Date: Fri, 2 Dec 2011 00:54:16 -0800 (PST) Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... In-Reply-To: <1322733452381-7049935.post@n2.nabble.com> References: <1322733452381-7049935.post@n2.nabble.com> Message-ID: <1322816056363-7053958.post@n2.nabble.com> Hi, Just a detail. Freeswitch is running on a VMWare virtual machine. Is it the problem ? Thanks... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-installed-ATA-registered-but-no-call-are-possible-tp7049935p7053958.html Sent from the freeswitch-users mailing list archive at Nabble.com. From peter.olsson at visionutveckling.se Fri Dec 2 12:05:14 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 2 Dec 2011 10:05:14 +0100 Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... In-Reply-To: <1322816056363-7053958.post@n2.nabble.com> References: <1322733452381-7049935.post@n2.nabble.com> <1322816056363-7053958.post@n2.nabble.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5B277A2021@cooper> I think we need other kind of details... Logs, SIP traces etc.. :) /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r obbyone Skickat: den 2 december 2011 09:54 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... Hi, Just a detail. Freeswitch is running on a VMWare virtual machine. Is it the problem ? Thanks... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-installed-ATA-registered-but-no-call-are-possible-tp7049935p7053958.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:4ed8925932761429410865! From beffa at ieee.org Fri Dec 2 12:13:26 2011 From: beffa at ieee.org (Federico Beffa) Date: Fri, 2 Dec 2011 10:13:26 +0100 Subject: [Freeswitch-users] google voice connection going offline (mod_dingaling) In-Reply-To: References: <1317265876.9229.YahooMailNeo@web39703.mail.mud.yahoo.com> <1317308654166-6844212.post@n2.nabble.com> Message-ID: Dear Ricardo, unfortunately the short answer is no, I still have intermittent disconnection problems. Some more details: * Sending a chat message from mod_dingaling to gtalk was problematic as google assigns you an from JID including a random number. However, to avoid having to find out this random number, in communicating with google talk, the from JID can be left blank.The special keywork "auto_from" has recently been added to mod_dingaling to make use of this functionality with the chat application. Please see http://jira.freeswitch.org/browse/FS-3611?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel for more details. * To mod_dingaling a "keep alive" functionality has been added. In practice this sends a small packet to google talk at regular intervals. The functionality has been integrated in the git repository, but not the latest patch: http://jira.freeswitch.org/browse/FS-3612?page=com.atlassian.jira.plugin.system.issuetabpanels:all-tabpanel Not sure why the latest patch has not yet been integrated. Maybe lack of time on the part of the developers. Disappointingly this does not solve my problem. I also tested sending "presence status" messages as keep alive packets, but this does not help either. I've dug a little bit further into the code to find out exactly where the problem occurs, but I've not found a solution. See http://lists.freeswitch.org/pipermail/freeswitch-dev/2011-October/005341.html Regards, Fede On Thu, Dec 1, 2011 at 7:05 AM, Ricardo Monroy wrote: > Hi Federico and Anthony, > > Were you guys able to come up with a solution for this? I had mine working for > about a month without any problems. I'm running the git version from October 1, > 2011. I just ran into the same problem after changing my router since I changed > my Internet connection from DSL to fiber optic. Reading the comments from this > post: > http://www.personal.psu.edu/wcs131/blogs/psuvoip/2010/10/using_freeswitch_to_add > _google.html > I guess this is an issue with Freeswitch having problems when there are Internet > disconnections. I guess one way to solve this would be to change my router, but > I don't really want to buy another ONT router, besides I don't even know if my > ISP would let me use a different ONT since the ones that they provide are all > "customized". > It is really hard to tell when it has disconnected. Even when typing "dingaling > status" from the fs_cli, it will show AUTHORIZED when it is disconnected. The > only way to tell if it's been disconnected is checking whether I can make calls > or not. > To solve this ,without having to restart freeswitch I would "disconnect" > (according to FS) from my gtalk profile typing "dl_logout gtalk" and reconnect > with "dl_login profile=gtalk". I also had the idea to use the "cron" approach > and thought this was going to be the solution to all my problems. Of course, I > had to test what would happen if the cron job would happen to run in the middle > of a call. I manually typed the commands on the fs_cli while I was on a call and > my call did not die!! Great!! I had the solution to the problem!! I then > finished the call and ooohh surprise!! I was suddenly disconnected from the > fs_cli, FS would die, close unexpectedly and create a core dump. > I would really like to help to solve this so let me know if there's any way I > can help out. > I would like to try the chat approach to see if that works. Did you figure out > what is the right syntax to chat? > > Greetings, > > Ricardo > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From tonybecq at yahoo.fr Fri Dec 2 12:15:00 2011 From: tonybecq at yahoo.fr (obbyone) Date: Fri, 2 Dec 2011 01:15:00 -0800 (PST) Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5B277A2021@cooper> References: <1322733452381-7049935.post@n2.nabble.com> <1322816056363-7053958.post@n2.nabble.com> <549CFEF87AEDE841A38E9D15EAB4C04C5B277A2021@cooper> Message-ID: <1322817300742-7054021.post@n2.nabble.com> Hi, Logs have already been sent on my second message. I send it back here (thanks) : 2011-12-01 13:42:18.918274 [DEBUG] sofia.c:7267 IP 90.36.1.89 Rejected by acl "domains". Falling back to Digest auth. 2011-12-01 13:42:29.258256 [DEBUG] sofia.c:7267 IP 90.36.1.89 Rejected by acl "domains". Falling back to Digest auth. 2011-12-01 13:42:29.258256 [NOTICE] switch_channel.c:920 New Channel sofia/internal/1004 at connexur.dyndns.org [ee2049ee-1c19-11e1-abcd-1f494360f62d] 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_NEW 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:380 (sofia/internal/1004 at connexur.dyndns.org) State NEW 2011-12-01 13:42:29.268425 [DEBUG] sofia.c:8186 Setting NAT mode based on nat.auto 2011-12-01 13:42:29.268425 [DEBUG] sofia.c:5282 Channel sofia/internal/1004 at connexur.dyndns.org entering state [received][100] 2011-12-01 13:42:29.268425 [DEBUG] sofia.c:5293 Remote SDP: v=0 o=- 9 2 IN IP4 192.168.1.10 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.10 t=0 0 m=audio 64860 RTP/AVP 107 119 0 98 8 3 101 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 1 : PBDGzwKZ EILmrBFJ 192.168.1.10 64860 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[G7221:115:32000:20:48000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[G7221:107:16000:20:32000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[G722:9:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[PCMU:0:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[PCMA:8:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[GSM:3:8000:20:13200] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[PCMA:8:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[GSM:3:8000:20:13200] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32-FEC:119:16000:20:0]/[G7221:115:32000:20:48000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32-FEC:119:16000:20:0]/[G7221:107:16000:20:32000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32-FEC:119:16000:20:0]/[G722:9:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32-FEC:119:16000:20:0]/[PCMU:0:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32-FEC:119:16000:20:0]/[PCMA:8:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32-FEC:119:16000:20:0]/[GSM:3:8000:20:13200] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:2864 Set Codec sofia/internal/1004 at connexur.dyndns.org PCMU/8000 20 ms 160 samples 64000 bits 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4871 Set 2833 dtmf send/recv payload to 101 2011-12-01 13:42:29.268425 [DEBUG] sofia.c:5505 (sofia/internal/1004 at connexur.dyndns.org) State Change CS_NEW -> CS_INIT 2011-12-01 13:42:29.268425 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_INIT 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1004 at connexur.dyndns.org) State INIT 2011-12-01 13:42:29.268425 [DEBUG] mod_sofia.c:85 sofia/internal/1004 at connexur.dyndns.org SOFIA INIT 2011-12-01 13:42:29.268425 [DEBUG] mod_sofia.c:125 (sofia/internal/1004 at connexur.dyndns.org) State Change CS_INIT -> CS_ROUTING 2011-12-01 13:42:29.268425 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1004 at connexur.dyndns.org) State INIT going to sleep 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_ROUTING 2011-12-01 13:42:29.268425 [DEBUG] switch_channel.c:1871 (sofia/internal/1004 at connexur.dyndns.org) Callstate Change DOWN -> RINGING 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1004 at connexur.dyndns.org) State ROUTING 2011-12-01 13:42:29.268425 [DEBUG] mod_sofia.c:148 sofia/internal/1004 at connexur.dyndns.org SOFIA ROUTING 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1004 at connexur.dyndns.org Standard ROUTING 2011-12-01 13:42:29.268425 [INFO] mod_dialplan_xml.c:481 Processing 1004 <1004>->1001 in context default Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->unloop] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->tod_example] continue=true Dialplan: sofia/internal/1004 at connexur.dyndns.org Date/Time Match (PASS) [tod_example] break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(open=true) Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->holiday_example] continue=true Dialplan: sofia/internal/1004 at connexur.dyndns.org Date/TimeMatch (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->global-intercept] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [global-intercept] destination_number(1001) =~ /^886$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->group-intercept] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [group-intercept] destination_number(1001) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [intercept-ext] destination_number(1001) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->redial] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [redial] destination_number(1001) =~ /^(redial|870)$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->global] continue=true Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=n$ Dialplan: sofia/internal/1004 at connexur.dyndns.org Absolute Condition [global] Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [snom-demo-2] destination_number(1001) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [snom-demo-2] destination_number(1001) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [snom-demo-1] destination_number(1001) =~ /^9000$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^779$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->call_return] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [call_return] destination_number(1001) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->del-group] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [del-group] destination_number(1001) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->add-group] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [add-group] destination_number(1001) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [call-group-simo] destination_number(1001) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->call-group-order] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [call-group-order] destination_number(1001) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [extension-intercom] destination_number(1001) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (PASS) [Local_Extension] destination_number(1001) =~ /^(10[01][0-9]|1100)$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(dialed_extension=1001) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action export(dialed_extension=1001) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime$ Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(ringback=${us-ring}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(call_timeout=30) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(hangup_after_bridge=true) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(continue_on_fail=true) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action answer() Dialplan: sofia/internal/1004 at connexur.dyndns.org Action sleep(1000) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/1004 at connexur.dyndns.org) State Change CS_ROUTING -> CS_EXECUTE 2011-12-01 13:42:29.268425 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1004 at connexur.dyndns.org) State ROUTING going to sleep 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_EXECUTE 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1004 at connexur.dyndns.org) State EXECUTE 2011-12-01 13:42:29.268425 [DEBUG] mod_sofia.c:241 sofia/internal/1004 at connexur.dyndns.org SOFIA EXECUTE 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:192 sofia/internal/1004 at connexur.dyndns.org Standard EXECUTE EXECUTE sofia/internal/1004 at connexur.dyndns.org set(open=true) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [open]=[true] EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-spymap/1004/ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial/1004/1001) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial/global/ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org set(RFC2822_DATE=Thu, 01 Dec 2011 13:42:29 +0100) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial/global/ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org set(RFC2822_DATE=Thu, 01 Dec 2011 13:42:29 +0100) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [RFC2822_DATE]=[Thu, 01 Dec 2011 13:42:29 +0100] EXECUTE sofia/internal/1004 at connexur.dyndns.org set(dialed_extension=1001) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [dialed_extension]=[1001] EXECUTE sofia/internal/1004 at connexur.dyndns.org export(dialed_extension=1001) 2011-12-01 13:42:29.268425 [DEBUG] switch_channel.c:1087 EXPORT (export_vars) [dialed_extension]=[1001] EXECUTE sofia/internal/1004 at connexur.dyndns.org bind_meta_app(1 b s execute_extension::dx XML features) 2011-12-01 13:42:29.268425 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE sofia/internal/1004 at connexur.dyndns.org bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/1004.2011-12-01-13-42-29.wav) 2011-12-01 13:42:29.268425 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1004.2011-12-01-13-42-29.wav EXECUTE sofia/internal/1004 at connexur.dyndns.org bind_meta_app(3 b s execute_extension::cf XML features) 2011-12-01 13:42:29.268425 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE sofia/internal/1004 at connexur.dyndns.org bind_meta_app(4 b s execute_extension::att_xfer XML features) 2011-12-01 13:42:29.268425 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *4 execute_extension::att_xfer XML features EXECUTE sofia/internal/1004 at connexur.dyndns.org set(ringback=%(2000,4000,440.0,480.0)) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [ringback]=[%(2000,4000,440.0,480.0)] EXECUTE sofia/internal/1004 at connexur.dyndns.org set(transfer_ringback=local_stream://moh) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/1004 at connexur.dyndns.org set(call_timeout=30) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [call_timeout]=[30] EXECUTE sofia/internal/1004 at connexur.dyndns.org set(hangup_after_bridge=true) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/1004 at connexur.dyndns.org set(continue_on_fail=true) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [continue_on_fail]=[true] EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-call_return/1001/1004) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial_ext/1001/ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial_ext//ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial_ext/global/ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org set(called_party_callgroup=techsupport) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [called_party_callgroup]=[techsupport] EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial/techsupport/ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org bridge(user/1001 at 91.204.116.116) 2011-12-01 13:42:29.268425 [DEBUG] switch_channel.c:1041 sofia/internal/1004 at connexur.dyndns.org EXPORTING[export_vars] [dialed_extension]=[1001] to event 2011-12-01 13:42:29.268425 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2011-12-01 13:42:29.268425 [DEBUG] switch_channel.c:1041 sofia/internal/1004 at connexur.dyndns.org EXPORTING[export_vars] [dialed_extension]=[1001] to event 2011-12-01 13:42:29.268425 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2011-12-01 13:42:29.268425 [DEBUG] switch_event.c:1521 Parsing variable [sip_invite_domain]=[91.204.116.116] 2011-12-01 13:42:29.268425 [DEBUG] switch_event.c:1521 Parsing variable [presence_id]=[1001 at 91.204.116.116] 2011-12-01 13:42:29.268425 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2011-12-01 13:42:29.268425 [DEBUG] switch_ivr_originate.c:3367 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2011-12-01 13:42:29.268425 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2011-12-01 13:42:29.268425 [DEBUG] switch_ivr_originate.c:3367 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2011-12-01 13:42:29.268425 [INFO] mod_dptools.c:2838 Originate Failed. Cause: USER_NOT_REGISTERED EXECUTE sofia/internal/1004 at connexur.dyndns.org answer() 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:3116 AUDIO RTP [sofia/internal/1004 at connexur.dyndns.org] 91.204.116.116 port 21454 -> 192.168.1.10 port 6486$ 2011-12-01 13:42:29.268425 [DEBUG] switch_rtp.c:1642 Starting timer [soft] 160 bytes per 20ms 2011-12-01 13:42:29.293791 [DEBUG] sofia_glue.c:3382 Set 2833 dtmf send payload to 101 2011-12-01 13:42:29.293791 [DEBUG] sofia_glue.c:3388 Set 2833 dtmf receive payload to 101 2011-12-01 13:42:29.293791 [DEBUG] mod_sofia.c:746 Local SDP sofia/internal/1004 at connexur.dyndns.org: v=0 o=FreeSWITCH 1322721895 1322721896 IN IP4 91.204.116.116 s=FreeSWITCH c=IN IP4 91.204.116.116 t=0 0 c=IN IP4 91.204.116.116 t=0 0 m=audio 21454 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 2011-12-01 13:42:29.293791 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.293791 [DEBUG] sofia.c:5282 Channel sofia/internal/1004 at connexur.dyndns.org entering state [completed][200] 2011-12-01 13:42:29.293791 [DEBUG] switch_core_session.c:726 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.293791 [DEBUG] switch_channel.c:3175 (sofia/internal/1004 at connexur.dyndns.org) Callstate Change RINGING -> ACTIVE 2011-12-01 13:42:29.293791 [NOTICE] mod_dptools.c:1071 Channel [sofia/internal/1004 at connexur.dyndns.org] has been answered EXECUTE sofia/internal/1004 at connexur.dyndns.org sleep(1000) 2011-12-01 13:42:29.568264 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.568264 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.568264 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.578421 [DEBUG] sofia.c:5282 Channel sofia/internal/1004 at connexur.dyndns.org entering state [ready][200] 2011-12-01 13:42:29.658301 [INFO] switch_rtp.c:3170 Auto Changing port from 192.168.1.10:64860 to 90.36.1.89:64860 EXECUTE sofia/internal/1004 at connexur.dyndns.org bridge(loopback/app=voicemail:default 91.204.116.116 1001) 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:1041 sofia/internal/1004 at connexur.dyndns.org EXPORTING[export_vars] [dialed_extension]=[1001] to event 2011-12-01 13:42:30.298263 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2011-12-01 13:42:30.298263 [NOTICE] switch_channel.c:920 New Channel loopback/app=voicemail:default 91.204.116.116 1001-a [eebdaf40-1c19-11e1-abd8-1f494360f$ 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:145 loopback/app=voicemail:default 91.204.116.116 1001-a setup codec PCMU/8000/20 2011-12-01 13:42:30.298263 [NOTICE] switch_channel.c:918 Rename Channel loopback/app=voicemail:default 91.204.116.116 1001-a->loopback/voicemail-a [eebdaf40$ 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:973 (loopback/voicemail-a) State Change CS_NEW -> CS_INIT 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-a) Running State Change CS_INIT 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:401 (loopback/voicemail-a) State INIT 2011-12-01 13:42:30.298263 [NOTICE] switch_channel.c:920 New Channel loopback/voicemail-b [eebdc6ba-1c19-11e1-abdc-1f494360f62d] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:145 loopback/voicemail-b setup codec PCMU/8000/20 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:258 (loopback/voicemail-b) State Change CS_NEW -> CS_INIT 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-b) Running State Change CS_INIT 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:401 (loopback/voicemail-b) State INIT 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:304 (loopback/voicemail-b) State Change CS_INIT -> CS_ROUTING 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:401 (loopback/voicemail-b) State INIT going to sleep 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-b) Running State Change CS_ROUTING 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:1871 (loopback/voicemail-b) Callstate Change DOWN -> RINGING 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:410 (loopback/voicemail-b) State ROUTING 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:336 loopback/voicemail-b CHANNEL ROUTING 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:355 (loopback/voicemail-b) State Change CS_ROUTING -> CS_EXECUTE 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:410 (loopback/voicemail-b) State ROUTING going to sleep 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-b) Running State Change CS_EXECUTE 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:417 (loopback/voicemail-b) State EXECUTE 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:375 loopback/voicemail-b CHANNEL EXECUTE 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:192 loopback/voicemail-b Standard EXECUTE 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:375 loopback/voicemail-b CHANNEL EXECUTE 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:192 loopback/voicemail-b Standard EXECUTE EXECUTE loopback/voicemail-b pre_answer() 2011-12-01 13:42:30.298263 [NOTICE] mod_loopback.c:760 Pre-Answer loopback/voicemail-a! 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:2917 (loopback/voicemail-a) Callstate Change DOWN -> EARLY 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:2959 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:726 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:30.298263 [NOTICE] mod_dptools.c:1097 Pre-Answer loopback/voicemail-b! 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:2917 (loopback/voicemail-b) Callstate Change RINGING -> EARLY 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:2959 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] EXECUTE loopback/voicemail-b voicemail(default 91.204.116.116 1001) 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:304 (loopback/voicemail-a) State Change CS_INIT -> CS_ROUTING 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:401 (loopback/voicemail-a) State INIT going to sleep 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-a) Running State Change CS_ROUTING 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:1871 (loopback/voicemail-a) Callstate Change EARLY -> RINGING 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:410 (loopback/voicemail-a) State ROUTING 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:336 loopback/voicemail-a CHANNEL ROUTING 2011-12-01 13:42:30.298263 [DEBUG] switch_ivr_originate.c:66 (loopback/voicemail-a) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:410 (loopback/voicemail-a) State ROUTING going to sleep 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-a) Running State Change CS_CONSUME_MEDIA 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:1875 (loopback/voicemail-a) Callstate Change RINGING -> EARLY 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:429 (loopback/voicemail-a) State CONSUME_MEDIA 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:535 CHANNEL CONSUME_MEDIA 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:429 (loopback/voicemail-a) State CONSUME_MEDIA going to sleep 2011-12-01 13:42:30.308562 [DEBUG] switch_ivr_originate.c:3269 Originate Resulted in Success: [loopback/voicemail-a] 2011-12-01 13:42:30.308562 [DEBUG] switch_core_session.c:726 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:30.308562 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:30.308562 [DEBUG] switch_core_session.c:726 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:30.308562 [DEBUG] switch_ivr_bridge.c:1270 (loopback/voicemail-a) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2011-12-01 13:42:30.308562 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:30.308562 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:30.308562 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-a) Running State Change CS_EXCHANGE_MEDIA 2011-12-01 13:42:30.308562 [DEBUG] switch_core_state_machine.c:420 (loopback/voicemail-a) State EXCHANGE_MEDIA 2011-12-01 13:42:30.308562 [DEBUG] mod_loopback.c:497 CHANNEL LOOPBACK 2011-12-01 13:42:30.398283 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en] 2011-12-01 13:42:30.409312 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-person.wav] (en:en) 2011-12-01 13:42:30.409312 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1 channels 20ms 2011-12-01 13:42:31.798263 [DEBUG] switch_ivr_play_say.c:1672 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-person.wav 2011-12-01 13:42:31.898264 [DEBUG] switch_ivr_play_say.c:244 Handle say:[1001] (en:en) 2011-12-01 13:42:31.898264 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1 channels 20ms 2011-12-01 13:42:33.398281 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:33.418917 [DEBUG] switch_channel.c:2833 (sofia/internal/1004 at connexur.dyndns.org) Callstate Change ACTIVE -> HANGUP 2011-12-01 13:42:33.418917 [NOTICE] sofia.c:572 Hangup sofia/internal/1004 at connexur.dyndns.org [CS_EXECUTE] [NORMAL_CLEARING] 2011-12-01 13:42:33.418917 [DEBUG] switch_channel.c:2856 Send signal sofia/internal/1004 at connexur.dyndns.org [KILL] 2011-12-01 13:42:33.418917 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:33.418917 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/1004 at connexur.dyndns.org] 2011-12-01 13:42:33.418917 [DEBUG] switch_ivr_bridge.c:611 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:33.418917 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [loopback/voicemail-a] 2011-12-01 13:42:33.438243 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:33.438243 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [loopback/voicemail-a] 2011-12-01 13:42:33.438243 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:33.438243 [DEBUG] switch_channel.c:2833 (loopback/voicemail-a) Callstate Change EARLY -> HANGUP 2011-12-01 13:42:33.438243 [NOTICE] switch_ivr_bridge.c:666 Hangup loopback/voicemail-a [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2011-12-01 13:42:33.438243 [DEBUG] switch_channel.c:2856 Send signal loopback/voicemail-a [KILL] 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:420 (loopback/voicemail-a) State EXCHANGE_MEDIA going to sleep 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-a) Running State Change CS_HANGUP 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:602 (loopback/voicemail-a) State HANGUP 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:427 loopback/voicemail-a CHANNEL HANGUP 2011-12-01 13:42:33.438243 [DEBUG] switch_channel.c:2833 (loopback/voicemail-b) Callstate Change EARLY -> HANGUP 2011-12-01 13:42:33.438243 [NOTICE] mod_loopback.c:438 Hangup loopback/voicemail-b [CS_EXECUTE] [NORMAL_CLEARING] 2011-12-01 13:42:33.438243 [DEBUG] switch_channel.c:2856 Send signal loopback/voicemail-b [KILL] 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:47 loopback/voicemail-a Standard HANGUP, cause: NORMAL_CLEARING 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:602 (loopback/voicemail-a) State HANGUP going to sleep 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:393 (loopback/voicemail-a) State Change CS_HANGUP -> CS_REPORTING 2011-12-01 13:42:33.438243 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-a) Running State Change CS_REPORTING 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:662 (loopback/voicemail-a) State REPORTING 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:79 loopback/voicemail-a Standard REPORTING, cause: NORMAL_CLEARING 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:662 (loopback/voicemail-a) State REPORTING going to sleep 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:387 (loopback/voicemail-a) State Change CS_REPORTING -> CS_DESTROY 2011-12-01 13:42:33.438243 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_core_session.c:1377 Session 145 (loopback/voicemail-a) Locked, Waiting on external entities 2011-12-01 13:42:33.450220 [DEBUG] switch_ivr_bridge.c:1348 sofia/internal/1004 at connexur.dyndns.org skip receive message [UNBRIDGE] (channel is hungup alrea$ 2011-12-01 13:42:33.450220 [DEBUG] switch_core_session.c:2272 sofia/internal/1004 at connexur.dyndns.org skip receive message [APPLICATION_EXEC_COMPLETE] (chan$ 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1004 at connexur.dyndns.org) State EXECUTE going to sleep 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_HANGUP 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1004 at connexur.dyndns.org) State HANGUP 2011-12-01 13:42:33.450220 [DEBUG] mod_sofia.c:459 sofia/internal/1004 at connexur.dyndns.org Overriding SIP cause 480 with 200 from the other leg 2011-12-01 13:42:33.450220 [DEBUG] mod_sofia.c:465 Channel sofia/internal/1004 at connexur.dyndns.org hanging up, cause: NORMAL_CLEARING 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1004 at connexur.dyndns.org Standard HANGUP, cause: NORMAL_CLEARING 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1004 at connexur.dyndns.org) State HANGUP going to sleep 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1004 at connexur.dyndns.org) State Change CS_HANGUP -> CS_REPORTING 2011-12-01 13:42:33.450220 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_REPORTING 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1004 at connexur.dyndns.org) State REPORTING 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1004 at connexur.dyndns.org Standard REPORTING, cause: NORMAL_CLEARING 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1004 at connexur.dyndns.org) State REPORTING going to sleep 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1004 at connexur.dyndns.org) State Change CS_REPORTING -> CS_DESTROY 2011-12-01 13:42:33.450220 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:33.450220 [DEBUG] switch_core_session.c:1377 Session 144 (sofia/internal/1004 at connexur.dyndns.org) Locked, Waiting on external entities 2011-12-01 13:42:33.450220 [NOTICE] switch_core_session.c:1395 Session 144 (sofia/internal/1004 at connexur.dyndns.org) Ended 2011-12-01 13:42:33.450220 [NOTICE] switch_core_session.c:1397 Close Channel sofia/internal/1004 at connexur.dyndns.org [CS_DESTROY] 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1004 at connexur.dyndns.org) Callstate Change HANGUP -> DOWN 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_DESTROY 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1004 at connexur.dyndns.org) State DESTROY 2011-12-01 13:42:33.450220 [DEBUG] mod_sofia.c:370 sofia/internal/1004 at connexur.dyndns.org SOFIA DESTROY 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1004 at connexur.dyndns.org) State DESTROY 2011-12-01 13:42:33.450220 [DEBUG] mod_sofia.c:370 sofia/internal/1004 at connexur.dyndns.org SOFIA DESTROY 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1004 at connexur.dyndns.org Standard DESTROY 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1004 at connexur.dyndns.org) State DESTROY going to sleep 2011-12-01 13:42:33.458337 [DEBUG] switch_ivr_play_say.c:1672 done playing file file_string://digits/1.wav!digits/0.wav!digits/0.wav!digits/1.wav 2011-12-01 13:42:33.548273 [DEBUG] switch_core_session.c:2272 loopback/voicemail-b skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup alrea$ 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:417 (loopback/voicemail-b) State EXECUTE going to sleep 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-b) Running State Change CS_HANGUP 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:602 (loopback/voicemail-b) State HANGUP 2011-12-01 13:42:33.548273 [DEBUG] mod_loopback.c:427 loopback/voicemail-b CHANNEL HANGUP 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:47 loopback/voicemail-b Standard HANGUP, cause: NORMAL_CLEARING 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:602 (loopback/voicemail-b) State HANGUP going to sleep 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:393 (loopback/voicemail-b) State Change CS_HANGUP -> CS_REPORTING 2011-12-01 13:42:33.548273 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:33.548273 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-b) Running State Change CS_REPORTING 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:662 (loopback/voicemail-b) State REPORTING 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:79 loopback/voicemail-b Standard REPORTING, cause: NORMAL_CLEARING 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:662 (loopback/voicemail-b) State REPORTING going to sleep 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:387 (loopback/voicemail-b) State Change CS_REPORTING -> CS_DESTROY 2011-12-01 13:42:33.548273 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:33.548273 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:33.548273 [DEBUG] switch_core_session.c:1377 Session 146 (loopback/voicemail-b) Locked, Waiting on external entities 2011-12-01 13:42:33.548273 [NOTICE] switch_core_session.c:1395 Session 146 (loopback/voicemail-b) Ended 2011-12-01 13:42:33.548273 [NOTICE] switch_core_session.c:1397 Close Channel loopback/voicemail-b [CS_DESTROY] 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:491 (loopback/voicemail-b) Callstate Change HANGUP -> DOWN 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:494 (loopback/voicemail-b) Running State Change CS_DESTROY 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:504 (loopback/voicemail-b) State DESTROY 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:86 loopback/voicemail-b Standard DESTROY 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:504 (loopback/voicemail-b) State DESTROY going to sleep 2011-12-01 13:42:33.559551 [NOTICE] switch_core_session.c:1395 Session 145 (loopback/voicemail-a) Ended 2011-12-01 13:42:33.559551 [NOTICE] switch_core_session.c:1397 Close Channel loopback/voicemail-a [CS_DESTROY] 2011-12-01 13:42:33.559551 [DEBUG] switch_core_state_machine.c:491 (loopback/voicemail-a) Callstate Change HANGUP -> DOWN 2011-12-01 13:42:33.559551 [DEBUG] switch_core_state_machine.c:494 (loopback/voicemail-a) Running State Change CS_DESTROY 2011-12-01 13:42:33.559551 [DEBUG] switch_core_state_machine.c:504 (loopback/voicemail-a) State DESTROY 2011-12-01 13:42:33.559551 [DEBUG] switch_core_state_machine.c:86 loopback/voicemail-a Standard DESTROY 2011-12-01 13:42:33.559551 [DEBUG] switch_core_state_machine.c:504 (loopback/voicemail-a) State DESTROY going to sleep ... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-installed-ATA-registered-but-no-call-are-possible-tp7049935p7054021.html Sent from the freeswitch-users mailing list archive at Nabble.com. From davidwaf at gmail.com Fri Dec 2 12:22:30 2011 From: davidwaf at gmail.com (David Wafula) Date: Fri, 2 Dec 2011 11:22:30 +0200 Subject: [Freeswitch-users] mod_rtmp: "Read error" when attempting calls from FS to RTMP client. In-Reply-To: References: <1322081549623-7026117.post@n2.nabble.com> <1322753056997-7050910.post@n2.nabble.com> Message-ID: On Fri, Dec 2, 2011 at 12:35 AM, Matt Stockton wrote: > > Also, even on Chrome (which is working), the read error does occur upon > hangup > > Oh i also ran into this. I got a "Your Linode, linodexxxx, has exceeded the notification threshold (90) for CPU Usage by averaging 104.6% for the last 2 hours" and that is when i noticed it in the logs. -- David Wafula -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111202/602686d4/attachment.html From peter.olsson at visionutveckling.se Fri Dec 2 12:49:04 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Fri, 2 Dec 2011 10:49:04 +0100 Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... In-Reply-To: <1322817300742-7054021.post@n2.nabble.com> References: <1322733452381-7049935.post@n2.nabble.com> <1322816056363-7053958.post@n2.nabble.com> <549CFEF87AEDE841A38E9D15EAB4C04C5B277A2021@cooper> <1322817300742-7054021.post@n2.nabble.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5B277A206F@cooper> It's pretty clear in the logs :) The user is not registered (USER_NOT_REGISTERED). So you will need to get the user registered before calling it. /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r obbyone Skickat: den 2 december 2011 10:15 Till: freeswitch-users at lists.freeswitch.org ?mne: Re: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... Hi, Logs have already been sent on my second message. I send it back here (thanks) : 2011-12-01 13:42:18.918274 [DEBUG] sofia.c:7267 IP 90.36.1.89 Rejected by acl "domains". Falling back to Digest auth. 2011-12-01 13:42:29.258256 [DEBUG] sofia.c:7267 IP 90.36.1.89 Rejected by acl "domains". Falling back to Digest auth. 2011-12-01 13:42:29.258256 [NOTICE] switch_channel.c:920 New Channel sofia/internal/1004 at connexur.dyndns.org [ee2049ee-1c19-11e1-abcd-1f494360f62d] 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_NEW 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:380 (sofia/internal/1004 at connexur.dyndns.org) State NEW 2011-12-01 13:42:29.268425 [DEBUG] sofia.c:8186 Setting NAT mode based on nat.auto 2011-12-01 13:42:29.268425 [DEBUG] sofia.c:5282 Channel sofia/internal/1004 at connexur.dyndns.org entering state [received][100] 2011-12-01 13:42:29.268425 [DEBUG] sofia.c:5293 Remote SDP: v=0 o=- 9 2 IN IP4 192.168.1.10 s=CounterPath X-Lite 3.0 c=IN IP4 192.168.1.10 t=0 0 m=audio 64860 RTP/AVP 107 119 0 98 8 3 101 a=rtpmap:107 BV32/16000 a=rtpmap:119 BV32-FEC/16000 a=rtpmap:98 iLBC/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=alt:1 1 : PBDGzwKZ EILmrBFJ 192.168.1.10 64860 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[G7221:115:32000:20:48000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[G7221:107:16000:20:32000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[G722:9:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[PCMU:0:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[PCMA:8:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[GSM:3:8000:20:13200] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[PCMA:8:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32:107:16000:20:0]/[GSM:3:8000:20:13200] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32-FEC:119:16000:20:0]/[G7221:115:32000:20:48000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32-FEC:119:16000:20:0]/[G7221:107:16000:20:32000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32-FEC:119:16000:20:0]/[G722:9:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32-FEC:119:16000:20:0]/[PCMU:0:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32-FEC:119:16000:20:0]/[PCMA:8:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [BV32-FEC:119:16000:20:0]/[GSM:3:8000:20:13200] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:115:32000:20:48000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [PCMU:0:8000:20:64000]/[G7221:107:16000:20:32000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [PCMU:0:8000:20:64000]/[G722:9:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4750 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:2864 Set Codec sofia/internal/1004 at connexur.dyndns.org PCMU/8000 20 ms 160 samples 64000 bits 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:4871 Set 2833 dtmf send/recv payload to 101 2011-12-01 13:42:29.268425 [DEBUG] sofia.c:5505 (sofia/internal/1004 at connexur.dyndns.org) State Change CS_NEW -> CS_INIT 2011-12-01 13:42:29.268425 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_INIT 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1004 at connexur.dyndns.org) State INIT 2011-12-01 13:42:29.268425 [DEBUG] mod_sofia.c:85 sofia/internal/1004 at connexur.dyndns.org SOFIA INIT 2011-12-01 13:42:29.268425 [DEBUG] mod_sofia.c:125 (sofia/internal/1004 at connexur.dyndns.org) State Change CS_INIT -> CS_ROUTING 2011-12-01 13:42:29.268425 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/1004 at connexur.dyndns.org) State INIT going to sleep 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_ROUTING 2011-12-01 13:42:29.268425 [DEBUG] switch_channel.c:1871 (sofia/internal/1004 at connexur.dyndns.org) Callstate Change DOWN -> RINGING 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1004 at connexur.dyndns.org) State ROUTING 2011-12-01 13:42:29.268425 [DEBUG] mod_sofia.c:148 sofia/internal/1004 at connexur.dyndns.org SOFIA ROUTING 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:104 sofia/internal/1004 at connexur.dyndns.org Standard ROUTING 2011-12-01 13:42:29.268425 [INFO] mod_dialplan_xml.c:481 Processing 1004 <1004>->1001 in context default Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->unloop] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->tod_example] continue=true Dialplan: sofia/internal/1004 at connexur.dyndns.org Date/Time Match (PASS) [tod_example] break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(open=true) Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->holiday_example] continue=true Dialplan: sofia/internal/1004 at connexur.dyndns.org Date/TimeMatch (FAIL) [holiday_example] break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->global-intercept] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [global-intercept] destination_number(1001) =~ /^886$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->group-intercept] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [group-intercept] destination_number(1001) =~ /^\*8$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->intercept-ext] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [intercept-ext] destination_number(1001) =~ /^\*\*(\d+)$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->redial] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [redial] destination_number(1001) =~ /^(redial|870)$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->global] continue=true Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [global] ${call_debug}(false) =~ /^true$/ break=never Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [global] ${sip_has_crypto}() =~ /^(AES_CM_128_HMAC_SHA1_32|AES_CM_128_HMAC_SHA1_80)$/ break=n$ Dialplan: sofia/internal/1004 at connexur.dyndns.org Absolute Condition [global] Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(RFC2822_DATE=${strftime(%a, %d %b %Y %T %z)}) Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [snom-demo-2] destination_number(1001) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [snom-demo-2] destination_number(1001) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [snom-demo-1] destination_number(1001) =~ /^9000$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^88(\d{4})$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^779$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->call_return] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [call_return] destination_number(1001) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->del-group] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [del-group] destination_number(1001) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->add-group] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [add-group] destination_number(1001) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [call-group-simo] destination_number(1001) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->call-group-order] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [call-group-order] destination_number(1001) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [extension-intercom] destination_number(1001) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (PASS) [Local_Extension] destination_number(1001) =~ /^(10[01][0-9]|1100)$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(dialed_extension=1001) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action export(dialed_extension=1001) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime$ Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bind_meta_app(4 b s execute_extension::att_xfer XML features) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(ringback=${us-ring}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(call_timeout=30) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(hangup_after_bridge=true) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(continue_on_fail=true) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial_ext/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial_ext/global/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action answer() Dialplan: sofia/internal/1004 at connexur.dyndns.org Action sleep(1000) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bridge(loopback/app=voicemail:default ${domain_name} ${dialed_extension}) 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/1004 at connexur.dyndns.org) State Change CS_ROUTING -> CS_EXECUTE 2011-12-01 13:42:29.268425 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1004 at connexur.dyndns.org) State ROUTING going to sleep 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_EXECUTE 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1004 at connexur.dyndns.org) State EXECUTE 2011-12-01 13:42:29.268425 [DEBUG] mod_sofia.c:241 sofia/internal/1004 at connexur.dyndns.org SOFIA EXECUTE 2011-12-01 13:42:29.268425 [DEBUG] switch_core_state_machine.c:192 sofia/internal/1004 at connexur.dyndns.org Standard EXECUTE EXECUTE sofia/internal/1004 at connexur.dyndns.org set(open=true) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [open]=[true] EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-spymap/1004/ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial/1004/1001) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial/global/ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org set(RFC2822_DATE=Thu, 01 Dec 2011 13:42:29 +0100) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial/global/ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org set(RFC2822_DATE=Thu, 01 Dec 2011 13:42:29 +0100) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [RFC2822_DATE]=[Thu, 01 Dec 2011 13:42:29 +0100] EXECUTE sofia/internal/1004 at connexur.dyndns.org set(dialed_extension=1001) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [dialed_extension]=[1001] EXECUTE sofia/internal/1004 at connexur.dyndns.org export(dialed_extension=1001) 2011-12-01 13:42:29.268425 [DEBUG] switch_channel.c:1087 EXPORT (export_vars) [dialed_extension]=[1001] EXECUTE sofia/internal/1004 at connexur.dyndns.org bind_meta_app(1 b s execute_extension::dx XML features) 2011-12-01 13:42:29.268425 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE sofia/internal/1004 at connexur.dyndns.org bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/1004.2011-12-01-13-42-29.wav) 2011-12-01 13:42:29.268425 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1004.2011-12-01-13-42-29.wav EXECUTE sofia/internal/1004 at connexur.dyndns.org bind_meta_app(3 b s execute_extension::cf XML features) 2011-12-01 13:42:29.268425 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE sofia/internal/1004 at connexur.dyndns.org bind_meta_app(4 b s execute_extension::att_xfer XML features) 2011-12-01 13:42:29.268425 [INFO] switch_ivr_async.c:3130 Bound B-Leg: *4 execute_extension::att_xfer XML features EXECUTE sofia/internal/1004 at connexur.dyndns.org set(ringback=%(2000,4000,440.0,480.0)) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [ringback]=[%(2000,4000,440.0,480.0)] EXECUTE sofia/internal/1004 at connexur.dyndns.org set(transfer_ringback=local_stream://moh) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/1004 at connexur.dyndns.org set(call_timeout=30) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [call_timeout]=[30] EXECUTE sofia/internal/1004 at connexur.dyndns.org set(hangup_after_bridge=true) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/1004 at connexur.dyndns.org set(continue_on_fail=true) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [continue_on_fail]=[true] EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-call_return/1001/1004) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial_ext/1001/ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial_ext//ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial_ext/global/ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org set(called_party_callgroup=techsupport) 2011-12-01 13:42:29.268425 [DEBUG] mod_dptools.c:1204 sofia/internal/1004 at connexur.dyndns.org SET [called_party_callgroup]=[techsupport] EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial/techsupport/ee2049ee-1c19-11e1-abcd-1f494360f62d) EXECUTE sofia/internal/1004 at connexur.dyndns.org bridge(user/1001 at 91.204.116.116) 2011-12-01 13:42:29.268425 [DEBUG] switch_channel.c:1041 sofia/internal/1004 at connexur.dyndns.org EXPORTING[export_vars] [dialed_extension]=[1001] to event 2011-12-01 13:42:29.268425 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2011-12-01 13:42:29.268425 [DEBUG] switch_channel.c:1041 sofia/internal/1004 at connexur.dyndns.org EXPORTING[export_vars] [dialed_extension]=[1001] to event 2011-12-01 13:42:29.268425 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2011-12-01 13:42:29.268425 [DEBUG] switch_event.c:1521 Parsing variable [sip_invite_domain]=[91.204.116.116] 2011-12-01 13:42:29.268425 [DEBUG] switch_event.c:1521 Parsing variable [presence_id]=[1001 at 91.204.116.116] 2011-12-01 13:42:29.268425 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [error] cause: [USER_NOT_REGISTERED] 2011-12-01 13:42:29.268425 [DEBUG] switch_ivr_originate.c:3367 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2011-12-01 13:42:29.268425 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [user] cause: [USER_NOT_REGISTERED] 2011-12-01 13:42:29.268425 [DEBUG] switch_ivr_originate.c:3367 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] 2011-12-01 13:42:29.268425 [INFO] mod_dptools.c:2838 Originate Failed. Cause: USER_NOT_REGISTERED EXECUTE sofia/internal/1004 at connexur.dyndns.org answer() 2011-12-01 13:42:29.268425 [DEBUG] sofia_glue.c:3116 AUDIO RTP [sofia/internal/1004 at connexur.dyndns.org] 91.204.116.116 port 21454 -> 192.168.1.10 port 6486$ 2011-12-01 13:42:29.268425 [DEBUG] switch_rtp.c:1642 Starting timer [soft] 160 bytes per 20ms 2011-12-01 13:42:29.293791 [DEBUG] sofia_glue.c:3382 Set 2833 dtmf send payload to 101 2011-12-01 13:42:29.293791 [DEBUG] sofia_glue.c:3388 Set 2833 dtmf receive payload to 101 2011-12-01 13:42:29.293791 [DEBUG] mod_sofia.c:746 Local SDP sofia/internal/1004 at connexur.dyndns.org: v=0 o=FreeSWITCH 1322721895 1322721896 IN IP4 91.204.116.116 s=FreeSWITCH c=IN IP4 91.204.116.116 t=0 0 c=IN IP4 91.204.116.116 t=0 0 m=audio 21454 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 2011-12-01 13:42:29.293791 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.293791 [DEBUG] sofia.c:5282 Channel sofia/internal/1004 at connexur.dyndns.org entering state [completed][200] 2011-12-01 13:42:29.293791 [DEBUG] switch_core_session.c:726 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.293791 [DEBUG] switch_channel.c:3175 (sofia/internal/1004 at connexur.dyndns.org) Callstate Change RINGING -> ACTIVE 2011-12-01 13:42:29.293791 [NOTICE] mod_dptools.c:1071 Channel [sofia/internal/1004 at connexur.dyndns.org] has been answered EXECUTE sofia/internal/1004 at connexur.dyndns.org sleep(1000) 2011-12-01 13:42:29.568264 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.568264 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.568264 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:29.578421 [DEBUG] sofia.c:5282 Channel sofia/internal/1004 at connexur.dyndns.org entering state [ready][200] 2011-12-01 13:42:29.658301 [INFO] switch_rtp.c:3170 Auto Changing port from 192.168.1.10:64860 to 90.36.1.89:64860 EXECUTE sofia/internal/1004 at connexur.dyndns.org bridge(loopback/app=voicemail:default 91.204.116.116 1001) 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:1041 sofia/internal/1004 at connexur.dyndns.org EXPORTING[export_vars] [dialed_extension]=[1001] to event 2011-12-01 13:42:30.298263 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2011-12-01 13:42:30.298263 [NOTICE] switch_channel.c:920 New Channel loopback/app=voicemail:default 91.204.116.116 1001-a [eebdaf40-1c19-11e1-abd8-1f494360f$ 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:145 loopback/app=voicemail:default 91.204.116.116 1001-a setup codec PCMU/8000/20 2011-12-01 13:42:30.298263 [NOTICE] switch_channel.c:918 Rename Channel loopback/app=voicemail:default 91.204.116.116 1001-a->loopback/voicemail-a [eebdaf40$ 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:973 (loopback/voicemail-a) State Change CS_NEW -> CS_INIT 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-a) Running State Change CS_INIT 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:401 (loopback/voicemail-a) State INIT 2011-12-01 13:42:30.298263 [NOTICE] switch_channel.c:920 New Channel loopback/voicemail-b [eebdc6ba-1c19-11e1-abdc-1f494360f62d] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:145 loopback/voicemail-b setup codec PCMU/8000/20 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:258 (loopback/voicemail-b) State Change CS_NEW -> CS_INIT 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-b) Running State Change CS_INIT 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:401 (loopback/voicemail-b) State INIT 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:304 (loopback/voicemail-b) State Change CS_INIT -> CS_ROUTING 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:401 (loopback/voicemail-b) State INIT going to sleep 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-b) Running State Change CS_ROUTING 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:1871 (loopback/voicemail-b) Callstate Change DOWN -> RINGING 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:410 (loopback/voicemail-b) State ROUTING 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:336 loopback/voicemail-b CHANNEL ROUTING 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:355 (loopback/voicemail-b) State Change CS_ROUTING -> CS_EXECUTE 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:410 (loopback/voicemail-b) State ROUTING going to sleep 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-b) Running State Change CS_EXECUTE 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:417 (loopback/voicemail-b) State EXECUTE 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:375 loopback/voicemail-b CHANNEL EXECUTE 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:192 loopback/voicemail-b Standard EXECUTE 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:375 loopback/voicemail-b CHANNEL EXECUTE 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:192 loopback/voicemail-b Standard EXECUTE EXECUTE loopback/voicemail-b pre_answer() 2011-12-01 13:42:30.298263 [NOTICE] mod_loopback.c:760 Pre-Answer loopback/voicemail-a! 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:2917 (loopback/voicemail-a) Callstate Change DOWN -> EARLY 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:2959 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:726 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:30.298263 [NOTICE] mod_dptools.c:1097 Pre-Answer loopback/voicemail-b! 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:2917 (loopback/voicemail-b) Callstate Change RINGING -> EARLY 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:2959 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] EXECUTE loopback/voicemail-b voicemail(default 91.204.116.116 1001) 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:304 (loopback/voicemail-a) State Change CS_INIT -> CS_ROUTING 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:401 (loopback/voicemail-a) State INIT going to sleep 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-a) Running State Change CS_ROUTING 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:1871 (loopback/voicemail-a) Callstate Change EARLY -> RINGING 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:410 (loopback/voicemail-a) State ROUTING 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:336 loopback/voicemail-a CHANNEL ROUTING 2011-12-01 13:42:30.298263 [DEBUG] switch_ivr_originate.c:66 (loopback/voicemail-a) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2011-12-01 13:42:30.298263 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:410 (loopback/voicemail-a) State ROUTING going to sleep 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-a) Running State Change CS_CONSUME_MEDIA 2011-12-01 13:42:30.298263 [DEBUG] switch_channel.c:1875 (loopback/voicemail-a) Callstate Change RINGING -> EARLY 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:429 (loopback/voicemail-a) State CONSUME_MEDIA 2011-12-01 13:42:30.298263 [DEBUG] mod_loopback.c:535 CHANNEL CONSUME_MEDIA 2011-12-01 13:42:30.298263 [DEBUG] switch_core_state_machine.c:429 (loopback/voicemail-a) State CONSUME_MEDIA going to sleep 2011-12-01 13:42:30.308562 [DEBUG] switch_ivr_originate.c:3269 Originate Resulted in Success: [loopback/voicemail-a] 2011-12-01 13:42:30.308562 [DEBUG] switch_core_session.c:726 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:30.308562 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:30.308562 [DEBUG] switch_core_session.c:726 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:30.308562 [DEBUG] switch_ivr_bridge.c:1270 (loopback/voicemail-a) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2011-12-01 13:42:30.308562 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:30.308562 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:30.308562 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-a) Running State Change CS_EXCHANGE_MEDIA 2011-12-01 13:42:30.308562 [DEBUG] switch_core_state_machine.c:420 (loopback/voicemail-a) State EXCHANGE_MEDIA 2011-12-01 13:42:30.308562 [DEBUG] mod_loopback.c:497 CHANNEL LOOPBACK 2011-12-01 13:42:30.398283 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en] 2011-12-01 13:42:30.409312 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-person.wav] (en:en) 2011-12-01 13:42:30.409312 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1 channels 20ms 2011-12-01 13:42:31.798263 [DEBUG] switch_ivr_play_say.c:1672 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-person.wav 2011-12-01 13:42:31.898264 [DEBUG] switch_ivr_play_say.c:244 Handle say:[1001] (en:en) 2011-12-01 13:42:31.898264 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1 channels 20ms 2011-12-01 13:42:33.398281 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:33.418917 [DEBUG] switch_channel.c:2833 (sofia/internal/1004 at connexur.dyndns.org) Callstate Change ACTIVE -> HANGUP 2011-12-01 13:42:33.418917 [NOTICE] sofia.c:572 Hangup sofia/internal/1004 at connexur.dyndns.org [CS_EXECUTE] [NORMAL_CLEARING] 2011-12-01 13:42:33.418917 [DEBUG] switch_channel.c:2856 Send signal sofia/internal/1004 at connexur.dyndns.org [KILL] 2011-12-01 13:42:33.418917 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:33.418917 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/1004 at connexur.dyndns.org] 2011-12-01 13:42:33.418917 [DEBUG] switch_ivr_bridge.c:611 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:33.418917 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [loopback/voicemail-a] 2011-12-01 13:42:33.438243 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:33.438243 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [loopback/voicemail-a] 2011-12-01 13:42:33.438243 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:33.438243 [DEBUG] switch_channel.c:2833 (loopback/voicemail-a) Callstate Change EARLY -> HANGUP 2011-12-01 13:42:33.438243 [NOTICE] switch_ivr_bridge.c:666 Hangup loopback/voicemail-a [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2011-12-01 13:42:33.438243 [DEBUG] switch_channel.c:2856 Send signal loopback/voicemail-a [KILL] 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:420 (loopback/voicemail-a) State EXCHANGE_MEDIA going to sleep 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-a) Running State Change CS_HANGUP 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:602 (loopback/voicemail-a) State HANGUP 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:427 loopback/voicemail-a CHANNEL HANGUP 2011-12-01 13:42:33.438243 [DEBUG] switch_channel.c:2833 (loopback/voicemail-b) Callstate Change EARLY -> HANGUP 2011-12-01 13:42:33.438243 [NOTICE] mod_loopback.c:438 Hangup loopback/voicemail-b [CS_EXECUTE] [NORMAL_CLEARING] 2011-12-01 13:42:33.438243 [DEBUG] switch_channel.c:2856 Send signal loopback/voicemail-b [KILL] 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:47 loopback/voicemail-a Standard HANGUP, cause: NORMAL_CLEARING 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:602 (loopback/voicemail-a) State HANGUP going to sleep 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:393 (loopback/voicemail-a) State Change CS_HANGUP -> CS_REPORTING 2011-12-01 13:42:33.438243 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-a) Running State Change CS_REPORTING 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:662 (loopback/voicemail-a) State REPORTING 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:79 loopback/voicemail-a Standard REPORTING, cause: NORMAL_CLEARING 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:662 (loopback/voicemail-a) State REPORTING going to sleep 2011-12-01 13:42:33.438243 [DEBUG] switch_core_state_machine.c:387 (loopback/voicemail-a) State Change CS_REPORTING -> CS_DESTROY 2011-12-01 13:42:33.438243 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-a [BREAK] 2011-12-01 13:42:33.438243 [DEBUG] mod_loopback.c:475 loopback/voicemail-a CHANNEL KILL 2011-12-01 13:42:33.438243 [DEBUG] switch_core_session.c:1377 Session 145 (loopback/voicemail-a) Locked, Waiting on external entities 2011-12-01 13:42:33.450220 [DEBUG] switch_ivr_bridge.c:1348 sofia/internal/1004 at connexur.dyndns.org skip receive message [UNBRIDGE] (channel is hungup alrea$ 2011-12-01 13:42:33.450220 [DEBUG] switch_core_session.c:2272 sofia/internal/1004 at connexur.dyndns.org skip receive message [APPLICATION_EXEC_COMPLETE] (chan$ 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1004 at connexur.dyndns.org) State EXECUTE going to sleep 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_HANGUP 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1004 at connexur.dyndns.org) State HANGUP 2011-12-01 13:42:33.450220 [DEBUG] mod_sofia.c:459 sofia/internal/1004 at connexur.dyndns.org Overriding SIP cause 480 with 200 from the other leg 2011-12-01 13:42:33.450220 [DEBUG] mod_sofia.c:465 Channel sofia/internal/1004 at connexur.dyndns.org hanging up, cause: NORMAL_CLEARING 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1004 at connexur.dyndns.org Standard HANGUP, cause: NORMAL_CLEARING 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1004 at connexur.dyndns.org) State HANGUP going to sleep 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1004 at connexur.dyndns.org) State Change CS_HANGUP -> CS_REPORTING 2011-12-01 13:42:33.450220 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_REPORTING 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1004 at connexur.dyndns.org) State REPORTING 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1004 at connexur.dyndns.org Standard REPORTING, cause: NORMAL_CLEARING 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1004 at connexur.dyndns.org) State REPORTING going to sleep 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1004 at connexur.dyndns.org) State Change CS_REPORTING -> CS_DESTROY 2011-12-01 13:42:33.450220 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-01 13:42:33.450220 [DEBUG] switch_core_session.c:1377 Session 144 (sofia/internal/1004 at connexur.dyndns.org) Locked, Waiting on external entities 2011-12-01 13:42:33.450220 [NOTICE] switch_core_session.c:1395 Session 144 (sofia/internal/1004 at connexur.dyndns.org) Ended 2011-12-01 13:42:33.450220 [NOTICE] switch_core_session.c:1397 Close Channel sofia/internal/1004 at connexur.dyndns.org [CS_DESTROY] 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1004 at connexur.dyndns.org) Callstate Change HANGUP -> DOWN 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_DESTROY 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1004 at connexur.dyndns.org) State DESTROY 2011-12-01 13:42:33.450220 [DEBUG] mod_sofia.c:370 sofia/internal/1004 at connexur.dyndns.org SOFIA DESTROY 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1004 at connexur.dyndns.org) State DESTROY 2011-12-01 13:42:33.450220 [DEBUG] mod_sofia.c:370 sofia/internal/1004 at connexur.dyndns.org SOFIA DESTROY 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1004 at connexur.dyndns.org Standard DESTROY 2011-12-01 13:42:33.450220 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1004 at connexur.dyndns.org) State DESTROY going to sleep 2011-12-01 13:42:33.458337 [DEBUG] switch_ivr_play_say.c:1672 done playing file file_string://digits/1.wav!digits/0.wav!digits/0.wav!digits/1.wav 2011-12-01 13:42:33.548273 [DEBUG] switch_core_session.c:2272 loopback/voicemail-b skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup alrea$ 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:417 (loopback/voicemail-b) State EXECUTE going to sleep 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-b) Running State Change CS_HANGUP 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:602 (loopback/voicemail-b) State HANGUP 2011-12-01 13:42:33.548273 [DEBUG] mod_loopback.c:427 loopback/voicemail-b CHANNEL HANGUP 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:47 loopback/voicemail-b Standard HANGUP, cause: NORMAL_CLEARING 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:602 (loopback/voicemail-b) State HANGUP going to sleep 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:393 (loopback/voicemail-b) State Change CS_HANGUP -> CS_REPORTING 2011-12-01 13:42:33.548273 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:33.548273 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:362 (loopback/voicemail-b) Running State Change CS_REPORTING 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:662 (loopback/voicemail-b) State REPORTING 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:79 loopback/voicemail-b Standard REPORTING, cause: NORMAL_CLEARING 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:662 (loopback/voicemail-b) State REPORTING going to sleep 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:387 (loopback/voicemail-b) State Change CS_REPORTING -> CS_DESTROY 2011-12-01 13:42:33.548273 [DEBUG] switch_core_session.c:1177 Send signal loopback/voicemail-b [BREAK] 2011-12-01 13:42:33.548273 [DEBUG] mod_loopback.c:475 loopback/voicemail-b CHANNEL KILL 2011-12-01 13:42:33.548273 [DEBUG] switch_core_session.c:1377 Session 146 (loopback/voicemail-b) Locked, Waiting on external entities 2011-12-01 13:42:33.548273 [NOTICE] switch_core_session.c:1395 Session 146 (loopback/voicemail-b) Ended 2011-12-01 13:42:33.548273 [NOTICE] switch_core_session.c:1397 Close Channel loopback/voicemail-b [CS_DESTROY] 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:491 (loopback/voicemail-b) Callstate Change HANGUP -> DOWN 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:494 (loopback/voicemail-b) Running State Change CS_DESTROY 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:504 (loopback/voicemail-b) State DESTROY 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:86 loopback/voicemail-b Standard DESTROY 2011-12-01 13:42:33.548273 [DEBUG] switch_core_state_machine.c:504 (loopback/voicemail-b) State DESTROY going to sleep 2011-12-01 13:42:33.559551 [NOTICE] switch_core_session.c:1395 Session 145 (loopback/voicemail-a) Ended 2011-12-01 13:42:33.559551 [NOTICE] switch_core_session.c:1397 Close Channel loopback/voicemail-a [CS_DESTROY] 2011-12-01 13:42:33.559551 [DEBUG] switch_core_state_machine.c:491 (loopback/voicemail-a) Callstate Change HANGUP -> DOWN 2011-12-01 13:42:33.559551 [DEBUG] switch_core_state_machine.c:494 (loopback/voicemail-a) Running State Change CS_DESTROY 2011-12-01 13:42:33.559551 [DEBUG] switch_core_state_machine.c:504 (loopback/voicemail-a) State DESTROY 2011-12-01 13:42:33.559551 [DEBUG] switch_core_state_machine.c:86 loopback/voicemail-a Standard DESTROY 2011-12-01 13:42:33.559551 [DEBUG] switch_core_state_machine.c:504 (loopback/voicemail-a) State DESTROY going to sleep ... -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-installed-ATA-registered-but-no-call-are-possible-tp7049935p7054021.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:4ed8965c32769010918511! From charlie.orford at attackplan.net Fri Dec 2 14:19:50 2011 From: charlie.orford at attackplan.net (Charlie Orford) Date: Fri, 02 Dec 2011 12:19:50 +0100 Subject: [Freeswitch-users] "481 Call/Transaction Does Not Exist" when hanging up before call connects In-Reply-To: References: <4ED7A88F.1010302@attackplan.net> Message-ID: <4ED8B456.9000505@attackplan.net> Hi Anthony We are using Aastra 57i phones with the latest firmware (v3.2.2.56 from June 2011). However, it looks like a specific issue with the Aastra 57i as I have just replicated the call and cancel situation using a 3CX softphone and in this case the CANCEL request gets honoured by FS. Here's a copy of a failed CANCEL test using the Aastra phone: http://pastebin.freeswitch.org/17922 Here's a copy of a successful CANCEL test using the 3CX softphone: http://pastebin.freeswitch.org/17923 sofia loglevel all 9 is on for both tests. Charlie On 01/12/2011 21:53, Anthony Minessale wrote: > try same failed call test with sofia loglevel all 9 > > Also, try some other phone or device that does not have the problem, > and create a cancel situation the same way and see if you can find a > difference. > > Finally, have you tried the latest firmware on the phone? > > > On Thu, Dec 1, 2011 at 10:17 AM, Charlie Orford > > > wrote: > > Hi list, > > When we make a call from an FS extension to a PSTN number (via our > ITSP > gateway provider) and hang-up before the call completes, FS replies to > the CANCEL request with "481 Call/Transaction Does Not Exist" and the > call continues to ring on the remote end. If we hangup after the call > has connected, it works with no problem. > > I have looked through a SIP trace of this happening and to my > (untrained > eye) nothing seems obviously wrong (i.e. tag, call id and branch > values > all seem to be correct). I'm using the latest git snapshot from > 2011-11-30 18-14-24 -0600. > > For a sip trace showing the problem, see: > http://pastebin.freeswitch.org/17908 > > For a sip trace showing a successful hangup, see: > http://pastebin.freeswitch.org/17909 > > *Note: ip addresses, domain and called number have been altered > for privacy. > > Any help or insight is much appreciated. > > Kind Regards, > Charlie > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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/20111202/41072418/attachment.html From jbaclor at ezuce.com Fri Dec 2 14:50:11 2011 From: jbaclor at ezuce.com (Joegen Baclor) Date: Fri, 02 Dec 2011 19:50:11 +0800 Subject: [Freeswitch-users] "481 Call/Transaction Does Not Exist" when hanging up before call connects In-Reply-To: <4ED8B456.9000505@attackplan.net> References: <4ED7A88F.1010302@attackplan.net> <4ED8B456.9000505@attackplan.net> Message-ID: <4ED8BB73.8080401@ezuce.com> For some reason, the aastra phone is sending a different contact and via address after it gets challenged by freeswitch. Original address was Via: SIP/2.0/UDP7.7.7.7:6060;branch=z9hG4bK2ff03b684c51c6e9a.84815b35f84df80f4;rport (LINE# 35) Then it changes to Via: SIP/2.0/UDP192.168.0.175:6060;branch=z9hG4bK085dcaff3e1ae346b.e0c0723efae179659;rport (LINE# 187) It then sends a CANCEL using Via: SIP/2.0/UDP7.7.7.7:6060 (LINE# 961) which freeswitch isn't able to match to a transaction because it is probably expecting 192.168.0.175. On 12/02/2011 07:19 PM, Charlie Orford wrote: > Hi Anthony > > We are using Aastra 57i phones with the latest firmware (v3.2.2.56 > from June 2011). > > However, it looks like a specific issue with the Aastra 57i as I have > just replicated the call and cancel situation using a 3CX softphone > and in this case the CANCEL request gets honoured by FS. > > Here's a copy of a failed CANCEL test using the Aastra phone: > http://pastebin.freeswitch.org/17922 > > Here's a copy of a successful CANCEL test using the 3CX softphone: > http://pastebin.freeswitch.org/17923 > > sofia loglevel all 9 is on for both tests. > > Charlie > > > On 01/12/2011 21:53, Anthony Minessale wrote: >> try same failed call test with sofia loglevel all 9 >> >> Also, try some other phone or device that does not have the problem, >> and create a cancel situation the same way and see if you can find a >> difference. >> >> Finally, have you tried the latest firmware on the phone? >> >> >> On Thu, Dec 1, 2011 at 10:17 AM, Charlie Orford >> > > wrote: >> >> Hi list, >> >> When we make a call from an FS extension to a PSTN number (via >> our ITSP >> gateway provider) and hang-up before the call completes, FS >> replies to >> the CANCEL request with "481 Call/Transaction Does Not Exist" and the >> call continues to ring on the remote end. If we hangup after the call >> has connected, it works with no problem. >> >> I have looked through a SIP trace of this happening and to my >> (untrained >> eye) nothing seems obviously wrong (i.e. tag, call id and branch >> values >> all seem to be correct). I'm using the latest git snapshot from >> 2011-11-30 18-14-24 -0600. >> >> For a sip trace showing the problem, see: >> http://pastebin.freeswitch.org/17908 >> >> For a sip trace showing a successful hangup, see: >> http://pastebin.freeswitch.org/17909 >> >> *Note: ip addresses, domain and called number have been altered >> for privacy. >> >> Any help or insight is much appreciated. >> >> Kind Regards, >> Charlie >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111202/96b3028a/attachment-0001.html From charlie.orford at attackplan.net Fri Dec 2 15:18:14 2011 From: charlie.orford at attackplan.net (Charlie Orford) Date: Fri, 02 Dec 2011 13:18:14 +0100 Subject: [Freeswitch-users] "481 Call/Transaction Does Not Exist" when hanging up before call connects In-Reply-To: <4ED8BB73.8080401@ezuce.com> References: <4ED7A88F.1010302@attackplan.net> <4ED8B456.9000505@attackplan.net> <4ED8BB73.8080401@ezuce.com> Message-ID: <4ED8C206.5010502@attackplan.net> Thanks Joegen, I hadn't picked up on that. Is there any valid reason why the phone would swap to using it's internal IP (192.168.0.175) for the Via and Contact headers when responding to the auth challenge? Or does this look more like a bug with the phone? On 02/12/2011 12:50, Joegen Baclor wrote: > For some reason, the aastra phone is sending a different contact and > via address after it gets challenged by freeswitch. > > Original address was > > Via: > SIP/2.0/UDP7.7.7.7:6060;branch=z9hG4bK2ff03b684c51c6e9a.84815b35f84df80f4;rport > (LINE# 35) > > Then it changes to > > Via: > SIP/2.0/UDP192.168.0.175:6060;branch=z9hG4bK085dcaff3e1ae346b.e0c0723efae179659;rport > (LINE# 187) > > It then sends a CANCEL using Via: SIP/2.0/UDP7.7.7.7:6060 (LINE# 961) > which freeswitch isn't able to match to a transaction because it is > probably expecting 192.168.0.175. > > > > On 12/02/2011 07:19 PM, Charlie Orford wrote: >> Hi Anthony >> >> We are using Aastra 57i phones with the latest firmware (v3.2.2.56 >> from June 2011). >> >> However, it looks like a specific issue with the Aastra 57i as I have >> just replicated the call and cancel situation using a 3CX softphone >> and in this case the CANCEL request gets honoured by FS. >> >> Here's a copy of a failed CANCEL test using the Aastra phone: >> http://pastebin.freeswitch.org/17922 >> >> Here's a copy of a successful CANCEL test using the 3CX softphone: >> http://pastebin.freeswitch.org/17923 >> >> sofia loglevel all 9 is on for both tests. >> >> Charlie >> >> >> On 01/12/2011 21:53, Anthony Minessale wrote: >>> try same failed call test with sofia loglevel all 9 >>> >>> Also, try some other phone or device that does not have the problem, >>> and create a cancel situation the same way and see if you can find a >>> difference. >>> >>> Finally, have you tried the latest firmware on the phone? >>> >>> >>> On Thu, Dec 1, 2011 at 10:17 AM, Charlie Orford >>> >> > wrote: >>> >>> Hi list, >>> >>> When we make a call from an FS extension to a PSTN number (via >>> our ITSP >>> gateway provider) and hang-up before the call completes, FS >>> replies to >>> the CANCEL request with "481 Call/Transaction Does Not Exist" >>> and the >>> call continues to ring on the remote end. If we hangup after the >>> call >>> has connected, it works with no problem. >>> >>> I have looked through a SIP trace of this happening and to my >>> (untrained >>> eye) nothing seems obviously wrong (i.e. tag, call id and branch >>> values >>> all seem to be correct). I'm using the latest git snapshot from >>> 2011-11-30 18-14-24 -0600. >>> >>> For a sip trace showing the problem, see: >>> http://pastebin.freeswitch.org/17908 >>> >>> For a sip trace showing a successful hangup, see: >>> http://pastebin.freeswitch.org/17909 >>> >>> *Note: ip addresses, domain and called number have been altered >>> for privacy. >>> >>> Any help or insight is much appreciated. >>> >>> Kind Regards, >>> Charlie >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111202/c7092c0c/attachment-0001.html From admin at blindi.net Fri Dec 2 17:56:41 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Fri, 2 Dec 2011 15:56:41 +0100 (CET) Subject: [Freeswitch-users] Problem send_dtmf send only to a-leg In-Reply-To: References: Message-ID: Hi Michael, thanks for you nice help!!! i.m. so glad!!! Nice weekend --------------- 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 Dec 2 18:37:45 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Fri, 2 Dec 2011 16:37:45 +0100 (CET) Subject: [Freeswitch-users] send_dtmf and queue_dtmf very slowly In-Reply-To: References: Message-ID: Hi all, I send various dtmftoenes. But the tones are sent very slowly. I've tried the following parameters: or: or: But unfortunately it does not work. I've tried it the following: In the external / internal probile: or: or: But unfortunately it does not work. The tones are given still slow. I mean pay the length and the distance between. Can your help please? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From jeferson at eox.com.br Fri Dec 2 14:48:45 2011 From: jeferson at eox.com.br (Jeferson Rodrigo Almeida) Date: Fri, 2 Dec 2011 09:48:45 -0200 Subject: [Freeswitch-users] Callcenter ringback - help Message-ID: Hi... I'm using mod_callcenter I need to send a ringback (or any other sound, different of the default moh) to the caller when a agent phone is ringing, and I don't know how to do it. Basically, I need to playback other sound to the caller when agent-offering event is fired. Is there any way to configure a dialplan after the call is bridged to the agent? Is there any way to modify the moh for this call in real-time? (I can capture the event...) Any help will be appreciated. Ps.: Sorry for any mistake in my "school" english. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111202/81753924/attachment.html From jeferson at eox.com.br Fri Dec 2 18:56:56 2011 From: jeferson at eox.com.br (Jeferson Rodrigo Almeida) Date: Fri, 2 Dec 2011 13:56:56 -0200 Subject: [Freeswitch-users] mod_callcenter ringback - help Message-ID: Hi... I'm using mod_callcenter I need to send a ringback (or any other sound, different of the default moh) to the caller when a agent phone is ringing, and I don't know how to do it. Basically, I need to playback other sound to the caller when agent-offering event is fired. Is there any way to configure a dialplan after the call is bridged to the agent? Is there any way to modify the moh for this call in real-time? (I can capture the event...) Any help will be appreciated. Ps.: Sorry for any mistake in my "school" english. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111202/8ae41c7d/attachment.html From jefersonparanaense at gmail.com Fri Dec 2 19:17:32 2011 From: jefersonparanaense at gmail.com (Jeferson Rodrigo Almeida) Date: Fri, 2 Dec 2011 14:17:32 -0200 Subject: [Freeswitch-users] mod_callcenter ringback - help In-Reply-To: References: Message-ID: > > Hi... > > I'm using mod_callcenter > > I need to send a ringback (or any other sound, different of the default > moh) to the caller when a agent phone is ringing, and I don't know how to > do it. > > Basically, I need to playback other sound to the caller when > agent-offering event is fired. > > Is there any way to configure a dialplan after the call is bridged to the > agent? > Is there any way to modify the moh for this call in real-time? (I can > capture the event...) > > Any help will be appreciated. > > Ps.: Sorry for any mistake in my "school" english. > > Thanks > > -- > Jeferson Rodrigo Almeida > Engenheiro de Computa??o > jefersonparanaense at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111202/8dcf57b3/attachment.html From brad at tech21.com Fri Dec 2 20:38:27 2011 From: brad at tech21.com (Brad Mina) Date: Fri, 2 Dec 2011 09:38:27 -0800 Subject: [Freeswitch-users] mod_callcenter ringback - help In-Reply-To: References: Message-ID: Jeferson, I'd recommend you look at the following links: http://wiki.freeswitch.org/wiki/Mod_callcenter#moh-sound http://wiki.freeswitch.org/wiki/Mod_local_stream On Fri, Dec 2, 2011 at 8:17 AM, Jeferson Rodrigo Almeida < jefersonparanaense at gmail.com> wrote: > Hi... >> >> I'm using mod_callcenter >> >> I need to send a ringback (or any other sound, different of the default >> moh) to the caller when a agent phone is ringing, and I don't know how to >> do it. >> >> Basically, I need to playback other sound to the caller when >> agent-offering event is fired. >> >> Is there any way to configure a dialplan after the call is bridged to the >> agent? >> Is there any way to modify the moh for this call in real-time? (I can >> capture the event...) >> >> Any help will be appreciated. >> >> Ps.: Sorry for any mistake in my "school" english. >> >> Thanks >> >> -- >> Jeferson Rodrigo Almeida >> Engenheiro de Computa??o >> jefersonparanaense at gmail.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/20111202/f0328abf/attachment.html From jefersonparanaense at gmail.com Fri Dec 2 21:04:06 2011 From: jefersonparanaense at gmail.com (Jeferson Rodrigo Almeida) Date: Fri, 2 Dec 2011 16:04:06 -0200 Subject: [Freeswitch-users] mod_callcenter ringback - help In-Reply-To: References: Message-ID: Thanks for the reply Brad, but my problem is other, I'll try to be clearer... When a member enters in a queue, and all agents are busy, he stays listening to the moh sound. When a agent goes to Waiting state, the call is bridged to him. In this moment, I want that the member star to listen a ringback (to know that his call is going to an agent). The link that you sent only explains how to set a moh sound, that will play continuously, doesn't matter if the agent is ringing or not. Is there any way to make this? Thanks again for the attention... 2011/12/2 Brad Mina > Jeferson, > > I'd recommend you look at the following links: > http://wiki.freeswitch.org/wiki/Mod_callcenter#moh-sound > > http://wiki.freeswitch.org/wiki/Mod_local_stream > > On Fri, Dec 2, 2011 at 8:17 AM, Jeferson Rodrigo Almeida < > jefersonparanaense at gmail.com> wrote: > >> Hi... >>> >>> I'm using mod_callcenter >>> >>> I need to send a ringback (or any other sound, different of the default >>> moh) to the caller when a agent phone is ringing, and I don't know how to >>> do it. >>> >>> Basically, I need to playback other sound to the caller when >>> agent-offering event is fired. >>> >>> Is there any way to configure a dialplan after the call is bridged to >>> the agent? >>> Is there any way to modify the moh for this call in real-time? (I can >>> capture the event...) >>> >>> Any help will be appreciated. >>> >>> Ps.: Sorry for any mistake in my "school" english. >>> >>> Thanks >>> >>> -- >>> Jeferson Rodrigo Almeida >>> Engenheiro de Computa??o >>> jefersonparanaense at gmail.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 > > -- Jeferson Rodrigo Almeida Engenheiro de Computa??o jefersonparanaense at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111202/6e0680ec/attachment-0001.html From dgarcia at anew.com.ve Fri Dec 2 22:44:24 2011 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Fri, 02 Dec 2011 15:14:24 -0430 Subject: [Freeswitch-users] mod_callcenter ringback - help In-Reply-To: References: Message-ID: <4ED92A98.2050905@anew.com.ve> try use this param: cc_warning_tone On 12/2/2011 1:34 PM, Jeferson Rodrigo Almeida wrote: > Thanks for the reply Brad, but my problem is other, I'll try to be > clearer... > > When a member enters in a queue, and all agents are busy, he stays > listening to the moh sound. When a agent goes to Waiting state, the > call is bridged to him. In this moment, I want that the member star to > listen a ringback (to know that his call is going to an agent). > > The link that you sent only explains how to set a moh sound, that will > play continuously, doesn't matter if the agent is ringing or not. > > Is there any way to make this? > > Thanks again for the attention... > > 2011/12/2 Brad Mina > > > Jeferson, > > I'd recommend you look at the following links: > http://wiki.freeswitch.org/wiki/Mod_callcenter#moh-sound > > http://wiki.freeswitch.org/wiki/Mod_local_stream > > On Fri, Dec 2, 2011 at 8:17 AM, Jeferson Rodrigo Almeida > > wrote: > > Hi... > > I'm using mod_callcenter > > I need to send a ringback (or any other sound, different > of the default moh) to the caller when a agent phone is > ringing, and I don't know how to do it. > > Basically, I need to playback other sound to the caller > when agent-offering event is fired. > > Is there any way to configure a dialplan after the call is > bridged to the agent? > Is there any way to modify the moh for this call in > real-time? (I can capture the event...) > > Any help will be appreciated. > > Ps.: Sorry for any mistake in my "school" english. > > Thanks > > -- > Jeferson Rodrigo Almeida > Engenheiro de Computa??o > jefersonparanaense at gmail.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 > > > > > -- > Jeferson Rodrigo Almeida > Engenheiro de Computa??o > jefersonparanaense at gmail.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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1873 / Virus Database: 2102/4652 - Release Date: 12/02/11 > -- 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/20111202/2c452272/attachment.html From dgarcia at anew.com.ve Fri Dec 2 23:55:32 2011 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Fri, 02 Dec 2011 16:25:32 -0430 Subject: [Freeswitch-users] mod_callcenter ringback - help In-Reply-To: References: Message-ID: <4ED93B44.3000108@anew.com.ve> Sorry, I mistake Try this: On 12/2/2011 1:34 PM, Jeferson Rodrigo Almeida wrote: > Thanks for the reply Brad, but my problem is other, I'll try to be > clearer... > > When a member enters in a queue, and all agents are busy, he stays > listening to the moh sound. When a agent goes to Waiting state, the > call is bridged to him. In this moment, I want that the member star to > listen a ringback (to know that his call is going to an agent). > > The link that you sent only explains how to set a moh sound, that will > play continuously, doesn't matter if the agent is ringing or not. > > Is there any way to make this? > > Thanks again for the attention... > > 2011/12/2 Brad Mina > > > Jeferson, > > I'd recommend you look at the following links: > http://wiki.freeswitch.org/wiki/Mod_callcenter#moh-sound > > http://wiki.freeswitch.org/wiki/Mod_local_stream > > On Fri, Dec 2, 2011 at 8:17 AM, Jeferson Rodrigo Almeida > > wrote: > > Hi... > > I'm using mod_callcenter > > I need to send a ringback (or any other sound, different > of the default moh) to the caller when a agent phone is > ringing, and I don't know how to do it. > > Basically, I need to playback other sound to the caller > when agent-offering event is fired. > > Is there any way to configure a dialplan after the call is > bridged to the agent? > Is there any way to modify the moh for this call in > real-time? (I can capture the event...) > > Any help will be appreciated. > > Ps.: Sorry for any mistake in my "school" english. > > Thanks > > -- > Jeferson Rodrigo Almeida > Engenheiro de Computa??o > jefersonparanaense at gmail.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 > > > > > -- > Jeferson Rodrigo Almeida > Engenheiro de Computa??o > jefersonparanaense at gmail.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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1873 / Virus Database: 2102/4652 - Release Date: 12/02/11 > -- 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/20111202/0475b8cf/attachment-0001.html From dgarcia at anew.com.ve Sat Dec 3 00:04:53 2011 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Fri, 02 Dec 2011 16:34:53 -0430 Subject: [Freeswitch-users] mod_callcenter ringback - help In-Reply-To: References: Message-ID: <4ED93D75.9010204@anew.com.ve> Also, check: http://freeswitch-users.2379917.n2.nabble.com/Using-fifo-orbit-announce-For-On-hook-Agent-td5222158.html On 12/2/2011 1:34 PM, Jeferson Rodrigo Almeida wrote: > Thanks for the reply Brad, but my problem is other, I'll try to be > clearer... > > When a member enters in a queue, and all agents are busy, he stays > listening to the moh sound. When a agent goes to Waiting state, the > call is bridged to him. In this moment, I want that the member star to > listen a ringback (to know that his call is going to an agent). > > The link that you sent only explains how to set a moh sound, that will > play continuously, doesn't matter if the agent is ringing or not. > > Is there any way to make this? > > Thanks again for the attention... > > 2011/12/2 Brad Mina > > > Jeferson, > > I'd recommend you look at the following links: > http://wiki.freeswitch.org/wiki/Mod_callcenter#moh-sound > > http://wiki.freeswitch.org/wiki/Mod_local_stream > > On Fri, Dec 2, 2011 at 8:17 AM, Jeferson Rodrigo Almeida > > wrote: > > Hi... > > I'm using mod_callcenter > > I need to send a ringback (or any other sound, different > of the default moh) to the caller when a agent phone is > ringing, and I don't know how to do it. > > Basically, I need to playback other sound to the caller > when agent-offering event is fired. > > Is there any way to configure a dialplan after the call is > bridged to the agent? > Is there any way to modify the moh for this call in > real-time? (I can capture the event...) > > Any help will be appreciated. > > Ps.: Sorry for any mistake in my "school" english. > > Thanks > > -- > Jeferson Rodrigo Almeida > Engenheiro de Computa??o > jefersonparanaense at gmail.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 > > > > > -- > Jeferson Rodrigo Almeida > Engenheiro de Computa??o > jefersonparanaense at gmail.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 > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1873 / Virus Database: 2102/4652 - Release Date: 12/02/11 > -- 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/20111202/7e8aaac4/attachment.html From jaknap34 at gmail.com Sat Dec 3 00:40:19 2011 From: jaknap34 at gmail.com (Pankaj Upadhyay) Date: Sat, 3 Dec 2011 03:10:19 +0530 Subject: [Freeswitch-users] Recording with music (playing in background) Message-ID: Hi all, I am new freeswitch user. I need to record the user voice/songs with music that selected earlier and this music plays in background . That is results comes as mix up of both songs i.e music in background and voice records in foreground . I want to dedicate this mix up file to other user.. Can I patchup these two files seperately..into one file ? or simultaneouly do recording with playing music in background ......and if so ..then please guide me..how can i do it ? Any help will be highly appreciated. Thanks . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111203/f5c7f47c/attachment.html From jack at livecall.com Sat Dec 3 00:49:33 2011 From: jack at livecall.com (Jack) Date: Fri, 02 Dec 2011 13:49:33 -0800 Subject: [Freeswitch-users] mod_rtmp: "Read error" when attempting calls from FS to RTMP client. In-Reply-To: References: <1322081549623-7026117.post@n2.nabble.com> <1322753056997-7050910.post@n2.nabble.com> Message-ID: <4ED947ED.2010208@livecall.com> I also get : 2011-11-30 16:04:55.108875 [ERR] rtmp.c:678 Read error each time an rtmp session from a browser FS Flex client is ended. It doesn't seem to bother anything. Jack On 12/1/2011 2:35 PM, Matt Stockton wrote: > Also worthwhile to note that the 'fsctl loglevel 9' does not make it > work for me, and the read error occurs still: > 2011-12-01 16:34:39.226836 [ERR] rtmp.c:713 Read error > > Also, even on Chrome (which is working), the read error does occur > upon hangup > > > On Thu, Dec 1, 2011 at 10:50 AM, Matt Stockton > wrote: > > For what it's worth, I am getting the exact same read error and it > is closing my socket when I dial into FS from RTMP client, but > only in certain browsers it seems. It's a read error on the same > line of code. > > Description of the issue and debug lines are listed in this JIRA: > http://jira.freeswitch.org/browse/FS-3729 > > > On Thu, Dec 1, 2011 at 9:24 AM, peely > wrote: > > Sorry, can I bump this? > > I've tried everything I'm capable of, but can't get a call > from mod_rtmp out > to the client without keeping the box logging in debug mode! > As soon as I > bring the log level down I get the issue, but of course can't > get any more > debug for this specific issue. > > I've looked at the code but don't know enough C to make any > changes. > > > Thanks, > > > Neil. > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/mod-rtmp-Read-error-when-attempting-calls-from-FS-to-RTMP-client-tp7026117p7050910.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 > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111202/51de6a98/attachment-0001.html From darcy at primrose.ws Sat Dec 3 01:01:25 2011 From: darcy at primrose.ws (Darcy) Date: Fri, 2 Dec 2011 17:01:25 -0500 Subject: [Freeswitch-users] acl validation Message-ID: <31CBF2DFBF224915A5FBB11AC495D24C@DWP> Looking through my call log, I see the following item in the log, this IP is not in the acl Domain List, it is not a profile and it is not registered on the switch, Is there something I am missing on this that needs to be set. These guys are attempting calls to Jamaica. 2011-10-31 09:11:31.801420 [DEBUG] sofia.c:5798 IP 91.212.226.30 Approved by acl "domains[]". Access Granted. I have 5 more entries like so in this switch. etc. Darcy Primrose -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111202/aaad2aff/attachment.html From brian at freeswitch.org Sat Dec 3 01:06:59 2011 From: brian at freeswitch.org (Brian West) Date: Fri, 2 Dec 2011 16:06:59 -0600 Subject: [Freeswitch-users] acl validation In-Reply-To: <31CBF2DFBF224915A5FBB11AC495D24C@DWP> References: <31CBF2DFBF224915A5FBB11AC495D24C@DWP> Message-ID: do you have a cidr=91.212.226.30 tag on a user in your directory? /b On Dec 2, 2011, at 4:01 PM, Darcy wrote: > Looking through my call log, I see the following item in the log, this IP is not in the acl Domain List, it is not a profile and it is not registered on the switch, Is there something I am missing on this that needs to be set. These guys are attempting calls to Jamaica. > > 2011-10-31 09:11:31.801420 [DEBUG] sofia.c:5798 IP 91.212.226.30 Approved by acl "domains[]". Access Granted. > > > > I have 5 more entries like so in this switch. > etc. > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111202/d0457a6a/attachment.html From msc at freeswitch.org Sat Dec 3 01:15:30 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 2 Dec 2011 14:15:30 -0800 Subject: [Freeswitch-users] Recording with music (playing in background) In-Reply-To: References: Message-ID: I would mix them separately using sox. Sox will let you change the volume level of the music so that you can make it sound like it's in the background. Something like this: sox -v .5 music.wav human.wav output.wav It's probably going to be a bit more complicated than that but since you're smart enough to be using FreeSWITCH then you should be able to figure it out. ;) If you need more help then hop on the #freeswitch channel on irc.freenode.net. -MC On Fri, Dec 2, 2011 at 1:40 PM, Pankaj Upadhyay wrote: > Hi all, > > I am new freeswitch user. > > I need to record the user voice/songs with music that selected earlier > and this music plays in background . That is results comes as mix up of > both songs > i.e music in background and voice records in foreground . > > I want to dedicate this mix up file to other user.. > Can I patchup these two files seperately..into one file ? or > simultaneouly do recording > with playing music in background ......and if so ..then please guide > me..how can i do it ? > > Any help will be highly appreciated. > > 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/20111202/ec4f4e24/attachment.html From jaknap34 at gmail.com Sat Dec 3 03:59:56 2011 From: jaknap34 at gmail.com (Pankaj Upadhyay) Date: Sat, 3 Dec 2011 06:29:56 +0530 Subject: [Freeswitch-users] Fwd: Recording with music (playing in background) In-Reply-To: References: Message-ID: Hi Michael, Thanks for the reply ..but my problem is slightly different. I explain here.. The output.wav file contain both files but play one by one ....Inorder to listen the output file one by one ..I need to play both files (of output files) at a time , that is music file play in background, at same time recorded file play in foreground...(so that it seems a one complete song with (music and recorded user songs)). and also output files adjusted it duration period (12 sec) according to the duration period of max( music.wav(10 sec), record.wav(12 sec)) not add up of two files. May be it is possible...... to play music files in one legs..and at same time record user voice song in other leg and thus produce recorded file have both songs ( because music file play at the same time as recording triggered..) actually i have no any idea.....please clarify...it. Or any patching method is there..or we doing some changing in dialplan with playback or other....? Thanks again for your reply sir.. Any help will be highly appreciated.... Regards Pankaj Upadhyay. ---------- Forwarded message ---------- From: Pankaj Upadhyay Date: Sat, Dec 3, 2011 at 3:10 AM Subject: [Freeswitch-users] Recording with music (playing in background) To: freeswitch-users at lists.freeswitch.org Hi all, I am new freeswitch user. I need to record the user voice/songs with music that selected earlier and this music plays in background . That is results comes as mix up of both songs i.e music in background and voice records in foreground . I want to dedicate this mix up file to other user.. Can I patchup these two files seperately..into one file ? or simultaneouly do recording with playing music in background ......and if so ..then please guide me..how can i do it ? Any help will be highly appreciated. Thanks . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111203/9f2a4340/attachment.html From darcy at primrose.ws Sat Dec 3 05:41:47 2011 From: darcy at primrose.ws (Darcy) Date: Fri, 2 Dec 2011 21:41:47 -0500 Subject: [Freeswitch-users] acl validation In-Reply-To: References: <31CBF2DFBF224915A5FBB11AC495D24C@DWP> Message-ID: <04248452835A42AB856DA88719A97DEE@DWP> Brian, thanks for the input, I have researched this exhaustively to be sure I don?t. We ran an extensive search on the system and could not find this IP address anywhere on the system, but yet they could make calls. This is an IP address from russia, they hit the network for around $20k in October then out of the blue they started being blocked, I am really worried they will find another hole in the system, if there is indeed one. Darcy From: Brian West Sent: Friday, December 02, 2011 5:06 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] acl validation do you have a cidr=91.212.226.30 tag on a user in your directory? /b On Dec 2, 2011, at 4:01 PM, Darcy wrote: Looking through my call log, I see the following item in the log, this IP is not in the acl Domain List, it is not a profile and it is not registered on the switch, Is there something I am missing on this that needs to be set. These guys are attempting calls to Jamaica. 2011-10-31 09:11:31.801420 [DEBUG] sofia.c:5798 IP 91.212.226.30 Approved by acl "domains[]". Access Granted. I have 5 more entries like so in this switch. etc. -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111202/9528fb4b/attachment-0001.html From brian at freeswitch.org Sat Dec 3 06:58:41 2011 From: brian at freeswitch.org (Brian West) Date: Fri, 2 Dec 2011 21:58:41 -0600 Subject: [Freeswitch-users] acl validation In-Reply-To: <04248452835A42AB856DA88719A97DEE@DWP> References: <31CBF2DFBF224915A5FBB11AC495D24C@DWP> <04248452835A42AB856DA88719A97DEE@DWP> Message-ID: <1FD86437-9E47-4B8A-ABE4-7C818DAF3B86@freeswitch.org> Well if you paypal me 5k I can help ya! :P /b On Dec 2, 2011, at 8:41 PM, Darcy wrote: > Brian, thanks for the input, I have researched this exhaustively to be sure I don?t. We ran an extensive search on the system and could not find this IP address anywhere on the system, but yet they could make calls. This is an IP address from russia, they hit the network for around $20k in October then out of the blue they started being blocked, I am really worried they will find another hole in the system, if there is indeed one. > > Darcy > > From: Brian West > Sent: Friday, December 02, 2011 5:06 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] acl validation > > do you have a cidr=91.212.226.30 tag on a user in your directory? > > / -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111202/afc0e25a/attachment.html From darcy at primrose.ws Sat Dec 3 07:20:10 2011 From: darcy at primrose.ws (Darcy) Date: Fri, 2 Dec 2011 23:20:10 -0500 Subject: [Freeswitch-users] acl validation In-Reply-To: <1FD86437-9E47-4B8A-ABE4-7C818DAF3B86@freeswitch.org> References: <31CBF2DFBF224915A5FBB11AC495D24C@DWP><04248452835A42AB856DA88719A97DEE@DWP> <1FD86437-9E47-4B8A-ABE4-7C818DAF3B86@freeswitch.org> Message-ID: That is a nice offer, but most likely for you, not us. What I have done is build a front to the dial plan that basically duplicates the acl list, it inspects all calls made to the freeswitch and looks at our version of the acl list, discards what it does not like, took a little work but I have not been able to bust it, gives us a little overhead, would like it to be done automatically in the freeswitch, however, this is a good work around. We use the freeswitch as a tandem between pbxnsip hosted servers and the pstn, that is why someone could generate a lot of traffic before we were alerted, now it is secure. Thanks in any case. Darcy From: Brian West Sent: Friday, December 02, 2011 10:58 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] acl validation Well if you paypal me 5k I can help ya! :P /b On Dec 2, 2011, at 8:41 PM, Darcy wrote: Brian, thanks for the input, I have researched this exhaustively to be sure I don?t. We ran an extensive search on the system and could not find this IP address anywhere on the system, but yet they could make calls. This is an IP address from russia, they hit the network for around $20k in October then out of the blue they started being blocked, I am really worried they will find another hole in the system, if there is indeed one. Darcy From: Brian West Sent: Friday, December 02, 2011 5:06 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] acl validation do you have a cidr=91.212.226.30 tag on a user in your directory? / -------------------------------------------------------------------------------- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111202/2cfec419/attachment.html From jbaclor at ezuce.com Sat Dec 3 08:29:12 2011 From: jbaclor at ezuce.com (Joegen Baclor) Date: Sat, 03 Dec 2011 13:29:12 +0800 Subject: [Freeswitch-users] "481 Call/Transaction Does Not Exist" when hanging up before call connects In-Reply-To: <4ED8C206.5010502@attackplan.net> References: <4ED7A88F.1010302@attackplan.net> <4ED8B456.9000505@attackplan.net> <4ED8BB73.8080401@ezuce.com> <4ED8C206.5010502@attackplan.net> Message-ID: <4ED9B3A8.4040707@ezuce.com> It's definitely a bug on the phone. I would try setting all Aastra NAT related features off and see if it stops getting confused like this. Freeswitch should be able to handle NAT from the far end. On 12/02/2011 08:18 PM, Charlie Orford wrote: > Thanks Joegen, I hadn't picked up on that. > > Is there any valid reason why the phone would swap to using it's > internal IP (192.168.0.175) for the Via and Contact headers when > responding to the auth challenge? Or does this look more like a bug > with the phone? > > > > On 02/12/2011 12:50, Joegen Baclor wrote: >> For some reason, the aastra phone is sending a different contact and >> via address after it gets challenged by freeswitch. >> >> Original address was >> >> Via: >> SIP/2.0/UDP7.7.7.7:6060;branch=z9hG4bK2ff03b684c51c6e9a.84815b35f84df80f4;rport >> (LINE# 35) >> >> Then it changes to >> >> Via: >> SIP/2.0/UDP192.168.0.175:6060;branch=z9hG4bK085dcaff3e1ae346b.e0c0723efae179659;rport >> (LINE# 187) >> >> It then sends a CANCEL using Via: SIP/2.0/UDP7.7.7.7:6060 (LINE# 961) >> which freeswitch isn't able to match to a transaction because it is >> probably expecting 192.168.0.175. >> >> >> >> On 12/02/2011 07:19 PM, Charlie Orford wrote: >>> Hi Anthony >>> >>> We are using Aastra 57i phones with the latest firmware (v3.2.2.56 >>> from June 2011). >>> >>> However, it looks like a specific issue with the Aastra 57i as I >>> have just replicated the call and cancel situation using a 3CX >>> softphone and in this case the CANCEL request gets honoured by FS. >>> >>> Here's a copy of a failed CANCEL test using the Aastra phone: >>> http://pastebin.freeswitch.org/17922 >>> >>> Here's a copy of a successful CANCEL test using the 3CX softphone: >>> http://pastebin.freeswitch.org/17923 >>> >>> sofia loglevel all 9 is on for both tests. >>> >>> Charlie >>> >>> >>> On 01/12/2011 21:53, Anthony Minessale wrote: >>>> try same failed call test with sofia loglevel all 9 >>>> >>>> Also, try some other phone or device that does not have the >>>> problem, and create a cancel situation the same way and see if you >>>> can find a difference. >>>> >>>> Finally, have you tried the latest firmware on the phone? >>>> >>>> >>>> On Thu, Dec 1, 2011 at 10:17 AM, Charlie Orford >>>> >>> > wrote: >>>> >>>> Hi list, >>>> >>>> When we make a call from an FS extension to a PSTN number (via >>>> our ITSP >>>> gateway provider) and hang-up before the call completes, FS >>>> replies to >>>> the CANCEL request with "481 Call/Transaction Does Not Exist" >>>> and the >>>> call continues to ring on the remote end. If we hangup after >>>> the call >>>> has connected, it works with no problem. >>>> >>>> I have looked through a SIP trace of this happening and to my >>>> (untrained >>>> eye) nothing seems obviously wrong (i.e. tag, call id and >>>> branch values >>>> all seem to be correct). I'm using the latest git snapshot from >>>> 2011-11-30 18-14-24 -0600. >>>> >>>> For a sip trace showing the problem, see: >>>> http://pastebin.freeswitch.org/17908 >>>> >>>> For a sip trace showing a successful hangup, see: >>>> http://pastebin.freeswitch.org/17909 >>>> >>>> *Note: ip addresses, domain and called number have been altered >>>> for privacy. >>>> >>>> Any help or insight is much appreciated. >>>> >>>> Kind Regards, >>>> Charlie >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE:http://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 >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111203/10949d3c/attachment-0001.html From msc at freeswitch.org Sat Dec 3 08:36:21 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 2 Dec 2011 21:36:21 -0800 Subject: [Freeswitch-users] Fwd: Recording with music (playing in background) In-Reply-To: References: Message-ID: Now I'm really confused... So you have just a single sound file to start with, or you have two different sound files to start with? And your ultimate goal is to have how many sound files? -MC On Fri, Dec 2, 2011 at 4:59 PM, Pankaj Upadhyay wrote: > Hi Michael, > > Thanks for the reply ..but my problem is slightly different. > > I explain here.. > > The output.wav file contain both files but play one by one ....Inorder to listen the output file one by one ..I need to play both files (of output files) at a time , that is music file play in background, at same time recorded file play in foreground...(so that it seems a one complete song with (music and recorded user songs)). > > and also output files adjusted it duration period (12 sec) according to the duration period of max( music.wav(10 sec), record.wav(12 sec)) not add up of two files. > > May be it is possible...... to play music files in one legs..and at same time record user voice song in other leg and thus produce recorded file have both songs ( because music file play at the same time as recording triggered..) > > actually i have no any idea.....please clarify...it. Or any patching method is there..or we doing some changing in dialplan with playback or other....? > > Thanks again for your reply sir.. > > Any help will be highly appreciated.... > > Regards > > Pankaj Upadhyay. > > ---------- Forwarded message ---------- > From: Pankaj Upadhyay > Date: Sat, Dec 3, 2011 at 3:10 AM > Subject: [Freeswitch-users] Recording with music (playing in background) > To: freeswitch-users at lists.freeswitch.org > > > Hi all, > > I am new freeswitch user. > > I need to record the user voice/songs with music that selected earlier > and this music plays in background . That is results comes as mix up of > both songs > i.e music in background and voice records in foreground . > > I want to dedicate this mix up file to other user.. > Can I patchup these two files seperately..into one file ? or > simultaneouly do recording > with playing music in background ......and if so ..then please guide > me..how can i do it ? > > Any help will be highly appreciated. > > 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/20111202/b526d51f/attachment.html From jaknap34 at gmail.com Sat Dec 3 12:39:25 2011 From: jaknap34 at gmail.com (Pankaj Upadhyay) Date: Sat, 3 Dec 2011 15:09:25 +0530 Subject: [Freeswitch-users] Fwd: Recording with music (playing in background) In-Reply-To: References: Message-ID: Hi Michael, Sorry for that confusion... actually i have one single music sound file to start with and my ultimate goal is to also have one single output sound file. now i have to do recording such that our current recording mux with original single music sound file and produce one single output file. ..it's like mobile karaoke services. thanks for your responding. Any help will be highly appreciated. Pankaj. ---------- Forwarded message ---------- From: Pankaj Upadhyay Date: Sat, Dec 3, 2011 at 6:29 AM Subject: Fwd: [Freeswitch-users] Recording with music (playing in background) To: freeswitch-users at lists.freeswitch.org Hi Michael, Thanks for the reply ..but my problem is slightly different. I explain here.. The output.wav file contain both files but play one by one ....Inorder to listen the output file one by one ..I need to play both files (of output files) at a time , that is music file play in background, at same time recorded file play in foreground...(so that it seems a one complete song with (music and recorded user songs)). and also output files adjusted it duration period (12 sec) according to the duration period of max( music.wav(10 sec), record.wav(12 sec)) not add up of two files. May be it is possible...... to play music files in one legs..and at same time record user voice song in other leg and thus produce recorded file have both songs ( because music file play at the same time as recording triggered..) actually i have no any idea.....please clarify...it. Or any patching method is there..or we doing some changing in dialplan with playback or other....? Thanks again for your reply sir.. Any help will be highly appreciated.... Regards Pankaj Upadhyay. ---------- Forwarded message ---------- From: Pankaj Upadhyay Date: Sat, Dec 3, 2011 at 3:10 AM Subject: [Freeswitch-users] Recording with music (playing in background) To: freeswitch-users at lists.freeswitch.org Hi all, I am new freeswitch user. I need to record the user voice/songs with music that selected earlier and this music plays in background . That is results comes as mix up of both songs i.e music in background and voice records in foreground . I want to dedicate this mix up file to other user.. Can I patchup these two files seperately..into one file ? or simultaneouly do recording with playing music in background ......and if so ..then please guide me..how can i do it ? Any help will be highly appreciated. Thanks . -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111203/edc3b9a7/attachment.html From charlie.orford at attackplan.net Sat Dec 3 14:37:32 2011 From: charlie.orford at attackplan.net (Charlie Orford) Date: Sat, 03 Dec 2011 12:37:32 +0100 Subject: [Freeswitch-users] "481 Call/Transaction Does Not Exist" when hanging up before call connects In-Reply-To: <4ED9B3A8.4040707@ezuce.com> References: <4ED7A88F.1010302@attackplan.net> <4ED8B456.9000505@attackplan.net> <4ED8BB73.8080401@ezuce.com> <4ED8C206.5010502@attackplan.net> <4ED9B3A8.4040707@ezuce.com> Message-ID: <4EDA09FC.9000504@attackplan.net> Bummer - we've got an office full of these phones :( As you suggest, I'll turn off all the NAT stuff when I'm there on Monday and see if that fixes it - hope so, otherwise I guess it looks like I'll have to try and get Aastra to look at the problem (doubt they are going to be very interested though!). Thanks, Charlie On 03/12/2011 06:29, Joegen Baclor wrote: > It's definitely a bug on the phone. I would try setting all Aastra > NAT related features off and see if it stops getting confused like > this. Freeswitch should be able to handle NAT from the far end. > > On 12/02/2011 08:18 PM, Charlie Orford wrote: >> Thanks Joegen, I hadn't picked up on that. >> >> Is there any valid reason why the phone would swap to using it's >> internal IP (192.168.0.175) for the Via and Contact headers when >> responding to the auth challenge? Or does this look more like a bug >> with the phone? >> >> >> >> On 02/12/2011 12:50, Joegen Baclor wrote: >>> For some reason, the aastra phone is sending a different contact >>> and via address after it gets challenged by freeswitch. >>> >>> Original address was >>> >>> Via: >>> SIP/2.0/UDP7.7.7.7:6060;branch=z9hG4bK2ff03b684c51c6e9a.84815b35f84df80f4;rport >>> (LINE# 35) >>> >>> Then it changes to >>> >>> Via: >>> SIP/2.0/UDP192.168.0.175:6060;branch=z9hG4bK085dcaff3e1ae346b.e0c0723efae179659;rport >>> (LINE# 187) >>> >>> It then sends a CANCEL using Via: SIP/2.0/UDP7.7.7.7:6060 (LINE# >>> 961) which freeswitch isn't able to match to a transaction because >>> it is probably expecting 192.168.0.175. >>> >>> >>> >>> On 12/02/2011 07:19 PM, Charlie Orford wrote: >>>> Hi Anthony >>>> >>>> We are using Aastra 57i phones with the latest firmware (v3.2.2.56 >>>> from June 2011). >>>> >>>> However, it looks like a specific issue with the Aastra 57i as I >>>> have just replicated the call and cancel situation using a 3CX >>>> softphone and in this case the CANCEL request gets honoured by FS. >>>> >>>> Here's a copy of a failed CANCEL test using the Aastra phone: >>>> http://pastebin.freeswitch.org/17922 >>>> >>>> Here's a copy of a successful CANCEL test using the 3CX softphone: >>>> http://pastebin.freeswitch.org/17923 >>>> >>>> sofia loglevel all 9 is on for both tests. >>>> >>>> Charlie >>>> >>>> >>>> On 01/12/2011 21:53, Anthony Minessale wrote: >>>>> try same failed call test with sofia loglevel all 9 >>>>> >>>>> Also, try some other phone or device that does not have the >>>>> problem, and create a cancel situation the same way and see if you >>>>> can find a difference. >>>>> >>>>> Finally, have you tried the latest firmware on the phone? >>>>> >>>>> >>>>> On Thu, Dec 1, 2011 at 10:17 AM, Charlie Orford >>>>> >>>> > wrote: >>>>> >>>>> Hi list, >>>>> >>>>> When we make a call from an FS extension to a PSTN number (via >>>>> our ITSP >>>>> gateway provider) and hang-up before the call completes, FS >>>>> replies to >>>>> the CANCEL request with "481 Call/Transaction Does Not Exist" >>>>> and the >>>>> call continues to ring on the remote end. If we hangup after >>>>> the call >>>>> has connected, it works with no problem. >>>>> >>>>> I have looked through a SIP trace of this happening and to my >>>>> (untrained >>>>> eye) nothing seems obviously wrong (i.e. tag, call id and >>>>> branch values >>>>> all seem to be correct). I'm using the latest git snapshot from >>>>> 2011-11-30 18-14-24 -0600. >>>>> >>>>> For a sip trace showing the problem, see: >>>>> http://pastebin.freeswitch.org/17908 >>>>> >>>>> For a sip trace showing a successful hangup, see: >>>>> http://pastebin.freeswitch.org/17909 >>>>> >>>>> *Note: ip addresses, domain and called number have been >>>>> altered for privacy. >>>>> >>>>> Any help or insight is much appreciated. >>>>> >>>>> Kind Regards, >>>>> Charlie >>>>> >>>>> >>>>> _________________________________________________________________________ >>>>> Professional FreeSWITCH Consulting Services: >>>>> consulting at freeswitch.org >>>>> http://www.freeswitchsolutions.com >>>>> >>>>> >>>>> >>>>> >>>>> Official FreeSWITCH Sites >>>>> http://www.freeswitch.org >>>>> http://wiki.freeswitch.org >>>>> http://www.cluecon.com >>>>> >>>>> FreeSWITCH-users mailing list >>>>> FreeSWITCH-users at lists.freeswitch.org >>>>> >>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>> UNSUBSCRIBE:http://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 >>> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111203/427487f6/attachment-0001.html From freeswitch-list at puzzled.xs4all.nl Sat Dec 3 18:33:41 2011 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Sat, 03 Dec 2011 16:33:41 +0100 Subject: [Freeswitch-users] acl validation In-Reply-To: <04248452835A42AB856DA88719A97DEE@DWP> References: <31CBF2DFBF224915A5FBB11AC495D24C@DWP> <04248452835A42AB856DA88719A97DEE@DWP> Message-ID: <4EDA4155.2070205@puzzled.xs4all.nl> On 12/03/2011 03:41 AM, Darcy wrote: > Brian, thanks for the input, I have researched this exhaustively to be > sure I don?t. We ran an extensive search on the system and could not > find this IP address anywhere on the system, but yet they could make > calls. This is an IP address from russia, they hit the network for > around $20k in October then out of the blue they started being blocked, > I am really worried they will find another hole in the system, if there > is indeed one. Don't know if you already have more countermeasures in place but I would also add firewall (iptables) rules to *only* allow certain IP ranges so you not only rely on the FreeSWITCH ACL. The 91.212.226.0/24 network is a Provider Independent range owned by some shady Russian outfit (missing city+country in the RIPE reg?) and judging from various traceroutes seems to be currently hosted by as5577.net in Luxembourg (root.lu?): http://trace.die.net/search/?q=91.212.226.23 Regards, Patrick From ayobami at programmer.net Sat Dec 3 19:07:21 2011 From: ayobami at programmer.net (ayobami) Date: Sat, 3 Dec 2011 08:07:21 -0800 (PST) Subject: [Freeswitch-users] Softphones could not call each other but could register on the FS server Message-ID: <1322928441467-7058171.post@n2.nabble.com> I just git pulled the Freeswitch source code and compiled, everything went well, I mean the compilation, but the problem that arose now is sip phones could not call each other again, now the soft phones could register on the FS server because I could use them to call 5000, 4000 and some of my custom numbers, but when I tried to call an extension from another extension, the operator keep saying that the extension is not available, I have troubleshooted and dont know what to do, please people help me out. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Softphones-could-not-call-each-other-but-could-register-on-the-FS-server-tp7058171p7058171.html Sent from the freeswitch-users mailing list archive at Nabble.com. From tonybecq at yahoo.fr Sat Dec 3 19:31:51 2011 From: tonybecq at yahoo.fr (obbyone) Date: Sat, 3 Dec 2011 08:31:51 -0800 (PST) Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... In-Reply-To: <1322733452381-7049935.post@n2.nabble.com> References: <1322733452381-7049935.post@n2.nabble.com> Message-ID: <1322929911507-7058216.post@n2.nabble.com> Hi, The trace I gave was not a good one. The one I show now Is a good sample. Any attempt to dial "1001" results only in a response by voicemail. Any answer ? Thanks /The trace :/ Dialplan: sofia/internal/1004 at connexur.dyndns.org Absolute Condition [global] Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-spymap/${caller_id_number}/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial/${caller_id_number}/${destination_number}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial/global/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->snom-demo-2] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [snom-demo-2] destination_number(1001) =~ /^9001$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->snom-demo-1] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [snom-demo-1] destination_number(1001) =~ /^9000$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^88(.*)$|^\*0(.*)$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->eavesdrop] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [eavesdrop] destination_number(1001) =~ /^779$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->call_return] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [call_return] destination_number(1001) =~ /^\*69$|^869$|^lcr$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->del-group] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [del-group] destination_number(1001) =~ /^80(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->add-group] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [add-group] destination_number(1001) =~ /^81(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->call-group-simo] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [call-group-simo] destination_number(1001) =~ /^82(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->call-group-order] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [call-group-order] destination_number(1001) =~ /^83(\d{2})$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->extension-intercom] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (FAIL) [extension-intercom] destination_number(1001) =~ /^8(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org parsing [default->Local_Extension] continue=false Dialplan: sofia/internal/1004 at connexur.dyndns.org Regex (PASS) [Local_Extension] destination_number(1001) =~ /^(10[01][0-9])$/ break=on-false Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(dialed_extension=1001) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action export(dialed_extension=1001) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bind_meta_app(1 b s execute_extension::dx XML features) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bind_meta_app(3 b s execute_extension::cf XML features) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(ringback=${us-ring}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(transfer_ringback=local_stream://moh) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(call_timeout=30) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(hangup_after_bridge=true) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(continue_on_fail=true) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action set(called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action hash(insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action bridge(user/${dialed_extension}@${domain_name}) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action answer() Dialplan: sofia/internal/1004 at connexur.dyndns.org Action sleep(1000) Dialplan: sofia/internal/1004 at connexur.dyndns.org Action voicemail(default ${domain_name} ${dialed_extension}) 2011-12-03 17:28:50.405366 [DEBUG] switch_core_state_machine.c:154 (sofia/internal/1004 at connexur.dyndns.org) State Change CS_ROUTING -> CS_EXECUTE 2011-12-03 17:28:50.405366 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-03 17:28:50.405366 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/1004 at connexur.dyndns.org) State ROUTING going to sleep 2011-12-03 17:28:50.405366 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_EXECUTE 2011-12-03 17:28:50.405366 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1004 at connexur.dyndns.org) State EXECUTE 2011-12-03 17:28:50.405366 [DEBUG] mod_sofia.c:241 sofia/internal/1004 at connexur.dyndns.org SOFIA EXECUTE 2011-12-03 17:28:50.405366 [DEBUG] switch_core_state_machine.c:192 sofia/internal/1004 at connexur.dyndns.org Standard EXECUTE EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-spymap/1004/e1f1595c-1dcb-11e1-b3b3-b530f603990b) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial/1004/1001) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial/global/e1f1595c-1dcb-11e1-b3b3-b530f603990b) EXECUTE sofia/internal/1004 at connexur.dyndns.org set(dialed_extension=1001) 2011-12-03 17:28:50.405366 [DEBUG] mod_dptools.c:1263 sofia/internal/1004 at connexur.dyndns.org SET [dialed_extension]=[1001] EXECUTE sofia/internal/1004 at connexur.dyndns.org export(dialed_extension=1001) 2011-12-03 17:28:50.405366 [DEBUG] switch_channel.c:1087 EXPORT (export_vars) [dialed_extension]=[1001] EXECUTE sofia/internal/1004 at connexur.dyndns.org bind_meta_app(1 b s execute_extension::dx XML features) 2011-12-03 17:28:50.405366 [INFO] switch_ivr_async.c:3164 Bound B-Leg: *1 execute_extension::dx XML features EXECUTE sofia/internal/1004 at connexur.dyndns.org bind_meta_app(2 b s record_session::/usr/local/freeswitch/recordings/1004.2011-12-03-17-28-50.wav) 2011-12-03 17:28:50.405366 [INFO] switch_ivr_async.c:3164 Bound B-Leg: *2 record_session::/usr/local/freeswitch/recordings/1004.2011-12-03-17-28-50.wav EXECUTE sofia/internal/1004 at connexur.dyndns.org bind_meta_app(3 b s execute_extension::cf XML features) 2011-12-03 17:28:50.405366 [INFO] switch_ivr_async.c:3164 Bound B-Leg: *3 execute_extension::cf XML features EXECUTE sofia/internal/1004 at connexur.dyndns.org set(ringback=%(2000,4000,440.0,480.0)) 2011-12-03 17:28:50.405366 [DEBUG] mod_dptools.c:1263 sofia/internal/1004 at connexur.dyndns.org SET [ringback]=[%(2000,4000,440.0,480.0)] EXECUTE sofia/internal/1004 at connexur.dyndns.org set(transfer_ringback=local_stream://moh) 2011-12-03 17:28:50.405366 [DEBUG] mod_dptools.c:1263 sofia/internal/1004 at connexur.dyndns.org SET [transfer_ringback]=[local_stream://moh] EXECUTE sofia/internal/1004 at connexur.dyndns.org set(call_timeout=30) 2011-12-03 17:28:50.405366 [DEBUG] mod_dptools.c:1263 sofia/internal/1004 at connexur.dyndns.org SET [call_timeout]=[30] EXECUTE sofia/internal/1004 at connexur.dyndns.org set(hangup_after_bridge=true) 2011-12-03 17:28:50.405366 [DEBUG] mod_dptools.c:1263 sofia/internal/1004 at connexur.dyndns.org SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/1004 at connexur.dyndns.org set(continue_on_fail=true) 2011-12-03 17:28:50.405366 [DEBUG] mod_dptools.c:1263 sofia/internal/1004 at connexur.dyndns.org SET [continue_on_fail]=[true] EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-call_return/1001/1004) EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial_ext/1001/e1f1595c-1dcb-11e1-b3b3-b530f603990b) EXECUTE sofia/internal/1004 at connexur.dyndns.org set(called_party_callgroup=techsupport) 2011-12-03 17:28:50.405366 [DEBUG] mod_dptools.c:1263 sofia/internal/1004 at connexur.dyndns.org SET [called_party_callgroup]=[techsupport] EXECUTE sofia/internal/1004 at connexur.dyndns.org hash(insert/91.204.116.116-last_dial/techsupport/e1f1595c-1dcb-11e1-b3b3-b530f603990b) EXECUTE sofia/internal/1004 at connexur.dyndns.org bridge(user/1001 at 91.204.116.116) 2011-12-03 17:28:50.405366 [DEBUG] switch_channel.c:1041 sofia/internal/1004 at connexur.dyndns.org EXPORTING[export_vars] [dialed_extension]=[1001] to event 2011-12-03 17:28:50.405366 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2011-12-03 17:28:50.405366 [DEBUG] switch_channel.c:1041 sofia/internal/1004 at connexur.dyndns.org EXPORTING[export_vars] [dialed_extension]=[1001] to event 2011-12-03 17:28:50.405366 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2011-12-03 17:28:50.405366 [DEBUG] switch_event.c:1521 Parsing variable [presence_id]=[1001 at 91.204.116.116] 2011-12-03 17:28:50.405366 [NOTICE] switch_channel.c:920 New Channel sofia/internal/sip:1001 at 192.168.0.43:5080 [e1f3aea0-1dcb-11e1-b3bb-b530f603990b] 2011-12-03 17:28:50.405366 [DEBUG] mod_sofia.c:4557 (sofia/internal/sip:1001 at 192.168.0.43:5080) State Change CS_NEW -> CS_INIT 2011-12-03 17:28:50.405366 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/sip:1001 at 192.168.0.43:5080 [BREAK] 2011-12-03 17:28:50.405366 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at 192.168.0.43:5080) Running State Change CS_INIT 2011-12-03 17:28:50.405366 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:1001 at 192.168.0.43:5080) State INIT 2011-12-03 17:28:50.405366 [DEBUG] mod_sofia.c:85 sofia/internal/sip:1001 at 192.168.0.43:5080 SOFIA INIT 2011-12-03 17:28:50.405366 [DEBUG] sofia_glue.c:2448 sip:1001 at 93.20.214.53:5080 Setting proxy route to sofia/internal/sip:1001 at 192.168.0.43:5080 2011-12-03 17:28:50.405366 [DEBUG] mod_sofia.c:125 (sofia/internal/sip:1001 at 192.168.0.43:5080) State Change CS_INIT -> CS_ROUTING 2011-12-03 17:28:50.405366 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/sip:1001 at 192.168.0.43:5080 [BREAK] 2011-12-03 17:28:50.405366 [DEBUG] switch_core_state_machine.c:401 (sofia/internal/sip:1001 at 192.168.0.43:5080) State INIT going to sleep 2011-12-03 17:28:50.405366 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at 192.168.0.43:5080) Running State Change CS_ROUTING 2011-12-03 17:28:50.405366 [DEBUG] switch_channel.c:1871 (sofia/internal/sip:1001 at 192.168.0.43:5080) Callstate Change DOWN -> RINGING 2011-12-03 17:28:50.405366 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at 192.168.0.43:5080) State ROUTING 2011-12-03 17:28:50.405366 [DEBUG] mod_sofia.c:148 sofia/internal/sip:1001 at 192.168.0.43:5080 SOFIA ROUTING 2011-12-03 17:28:50.405366 [DEBUG] switch_ivr_originate.c:66 (sofia/internal/sip:1001 at 192.168.0.43:5080) State Change CS_ROUTING -> CS_CONSUME_MEDIA 2011-12-03 17:28:50.405366 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/sip:1001 at 192.168.0.43:5080 [BREAK] 2011-12-03 17:28:50.405366 [DEBUG] switch_core_state_machine.c:410 (sofia/internal/sip:1001 at 192.168.0.43:5080) State ROUTING going to sleep 2011-12-03 17:28:50.405366 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at 192.168.0.43:5080) Running State Change CS_CONSUME_MEDIA 2011-12-03 17:28:50.405366 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:1001 at 192.168.0.43:5080) State CONSUME_MEDIA 2011-12-03 17:28:50.405366 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:1001 at 192.168.0.43:5080) State CONSUME_MEDIA going to sleep 2011-12-03 17:28:50.405366 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/sip:1001 at 192.168.0.43:5080 [BREAK] 2011-12-03 17:28:50.405366 [DEBUG] sofia.c:5368 Channel sofia/internal/sip:1001 at 192.168.0.43:5080 entering state [calling][0] 2011-12-03 17:28:50.525351 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/sip:1001 at 192.168.0.43:5080 [BREAK] 2011-12-03 17:28:50.525351 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/sip:1001 at 192.168.0.43:5080 [BREAK] 2011-12-03 17:28:50.525351 [INFO] sofia.c:830 sofia/internal/sip:1001 at 192.168.0.43:5080 Update Callee ID to "1001" 2011-12-03 17:28:50.525351 [DEBUG] sofia.c:5368 Channel sofia/internal/sip:1001 at 192.168.0.43:5080 entering state [completing][200] 2011-12-03 17:28:50.525351 [DEBUG] sofia.c:5379 Remote SDP: v=0 o=FreeSWITCH 1322901237 1322901238 IN IP4 192.168.0.22 s=FreeSWITCH c=IN IP4 192.168.0.22 t=0 0 m=audio 28016 RTP/AVP 0 101 13 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=rtpmap:13 CN/8000 a=ptime:20 2011-12-03 17:28:50.525351 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/sip:1001 at 192.168.0.43:5080 [BREAK] 2011-12-03 17:28:50.525351 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/sip:1001 at 192.168.0.43:5080 [BREAK] 2011-12-03 17:28:50.525351 [DEBUG] sofia.c:5368 Channel sofia/internal/sip:1001 at 192.168.0.43:5080 entering state [ready][200] 2011-12-03 17:28:50.525351 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [PCMU:0:8000:20:64000]/[PCMU:0:8000:20:64000] 2011-12-03 17:28:50.525351 [DEBUG] sofia_glue.c:2888 Set Codec sofia/internal/sip:1001 at 192.168.0.43:5080 PCMU/8000 20 ms 160 samples 64000 bits 2011-12-03 17:28:50.525351 [DEBUG] sofia_glue.c:4881 Set 2833 dtmf send payload to 101 2011-12-03 17:28:50.525351 [DEBUG] sofia_glue.c:3140 AUDIO RTP [sofia/internal/sip:1001 at 192.168.0.43:5080] 91.204.116.116 port 28276 -> 192.168.0.22 port 28016 codec: 0 ms: 20 2011-12-03 17:28:50.525351 [DEBUG] switch_rtp.c:1642 Starting timer [soft] 160 bytes per 20ms 2011-12-03 17:28:50.525351 [DEBUG] sofia_glue.c:3404 Set 2833 dtmf send payload to 101 2011-12-03 17:28:50.525351 [DEBUG] sofia_glue.c:3410 Set 2833 dtmf receive payload to 101 2011-12-03 17:28:50.525351 [DEBUG] sofia_glue.c:3429 Set comfort noise payload to 13 2011-12-03 17:28:50.525351 [DEBUG] switch_channel.c:3175 (sofia/internal/sip:1001 at 192.168.0.43:5080) Callstate Change RINGING -> ACTIVE 2011-12-03 17:28:50.525351 [DEBUG] switch_channel.c:3187 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-03 17:28:50.525351 [NOTICE] sofia.c:6077 Channel [sofia/internal/sip:1001 at 192.168.0.43:5080] has been answered 2011-12-03 17:28:50.546367 [DEBUG] sofia_glue.c:3140 AUDIO RTP [sofia/internal/1004 at connexur.dyndns.org] 91.204.116.116 port 21574 -> 192.168.1.10 port 45044 codec: 0 ms: 20 2011-12-03 17:28:50.546367 [DEBUG] switch_rtp.c:1642 Starting timer [soft] 160 bytes per 20ms 2011-12-03 17:28:50.546367 [DEBUG] sofia_glue.c:3404 Set 2833 dtmf send payload to 101 2011-12-03 17:28:50.546367 [DEBUG] sofia_glue.c:3410 Set 2833 dtmf receive payload to 101 2011-12-03 17:28:50.546367 [DEBUG] mod_sofia.c:746 Local SDP sofia/internal/1004 at connexur.dyndns.org: v=0 o=FreeSWITCH 1322908156 1322908157 IN IP4 91.204.116.116 s=FreeSWITCH c=IN IP4 91.204.116.116 t=0 0 m=audio 21574 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 2011-12-03 17:28:50.546367 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-03 17:28:50.546367 [DEBUG] sofia.c:5368 Channel sofia/internal/1004 at connexur.dyndns.org entering state [completed][200] 2011-12-03 17:28:50.546367 [DEBUG] switch_core_session.c:726 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-03 17:28:50.546367 [DEBUG] switch_channel.c:3175 (sofia/internal/1004 at connexur.dyndns.org) Callstate Change RINGING -> ACTIVE 2011-12-03 17:28:50.546367 [NOTICE] switch_ivr_originate.c:3209 Channel [sofia/internal/1004 at connexur.dyndns.org] has been answered 2011-12-03 17:28:50.546367 [DEBUG] switch_ivr_originate.c:3269 Originate Resulted in Success: [sofia/internal/sip:1001 at 192.168.0.43:5080] 2011-12-03 17:28:50.546367 [DEBUG] switch_ivr_originate.c:3514 (sofia/internal/sip:1001 at 192.168.0.43:5080) State Change CS_CONSUME_MEDIA -> CS_HIBERNATE 2011-12-03 17:28:50.546367 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/sip:1001 at 192.168.0.43:5080 [BREAK] 2011-12-03 17:28:50.546367 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at 192.168.0.43:5080) Running State Change CS_HIBERNATE 2011-12-03 17:28:50.546367 [DEBUG] switch_core_state_machine.c:432 (sofia/internal/sip:1001 at 192.168.0.43:5080) State HIBERNATE 2011-12-03 17:28:50.546367 [DEBUG] mod_sofia.c:222 sofia/internal/sip:1001 at 192.168.0.43:5080 SOFIA HIBERNATE 2011-12-03 17:28:50.546367 [DEBUG] switch_core_state_machine.c:261 sofia/internal/sip:1001 at 192.168.0.43:5080 Standard HIBERNATE 2011-12-03 17:28:50.546367 [DEBUG] switch_core_state_machine.c:432 (sofia/internal/sip:1001 at 192.168.0.43:5080) State HIBERNATE going to sleep 2011-12-03 17:28:50.546367 [DEBUG] switch_ivr_originate.c:3269 Originate Resulted in Success: [sofia/internal/sip:1001 at 192.168.0.43:5080] 2011-12-03 17:28:50.546367 [DEBUG] switch_ivr_bridge.c:1197 (sofia/internal/sip:1001 at 192.168.0.43:5080) State Change CS_HIBERNATE -> CS_CONSUME_MEDIA 2011-12-03 17:28:50.546367 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/sip:1001 at 192.168.0.43:5080 [BREAK] 2011-12-03 17:28:50.546367 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at 192.168.0.43:5080) Running State Change CS_CONSUME_MEDIA 2011-12-03 17:28:50.546367 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:1001 at 192.168.0.43:5080) State CONSUME_MEDIA 2011-12-03 17:28:50.546367 [DEBUG] switch_core_state_machine.c:429 (sofia/internal/sip:1001 at 192.168.0.43:5080) State CONSUME_MEDIA going to sleep 2011-12-03 17:28:50.546367 [DEBUG] switch_core_session.c:726 Send signal sofia/internal/sip:1001 at 192.168.0.43:5080 [BREAK] 2011-12-03 17:28:50.546367 [DEBUG] switch_core_session.c:726 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-03 17:28:50.546367 [DEBUG] switch_ivr_bridge.c:1289 (sofia/internal/sip:1001 at 192.168.0.43:5080) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 2011-12-03 17:28:50.546367 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/sip:1001 at 192.168.0.43:5080 [BREAK] 2011-12-03 17:28:50.546367 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at 192.168.0.43:5080) Running State Change CS_EXCHANGE_MEDIA 2011-12-03 17:28:50.546367 [DEBUG] switch_core_state_machine.c:420 (sofia/internal/sip:1001 at 192.168.0.43:5080) State EXCHANGE_MEDIA 2011-12-03 17:28:50.546367 [DEBUG] mod_sofia.c:574 SOFIA EXCHANGE_MEDIA 2011-12-03 17:28:50.805377 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-03 17:28:50.805377 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-03 17:28:50.805377 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-03 17:28:50.825358 [DEBUG] sofia.c:5368 Channel sofia/internal/1004 at connexur.dyndns.org entering state [ready][200] 2011-12-03 17:28:50.825358 [DEBUG] switch_core_session.c:788 Send signal sofia/internal/sip:1001 at 192.168.0.43:5080 [BREAK] 2011-12-03 17:28:50.825358 [DEBUG] switch_core_session.c:788 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-03 17:28:50.905374 [INFO] switch_rtp.c:3170 Auto Changing port from 192.168.1.10:45044 to 92.145.203.187:45044 2011-12-03 17:28:50.905374 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/sip:1001 at 192.168.0.43:5080 [BREAK] 2011-12-03 17:28:51.885388 [INFO] switch_rtp.c:3170 Auto Changing port from 192.168.0.22:28016 to 93.20.214.53:28016 2011-12-03 17:29:00.775407 [DEBUG] switch_core_session.c:872 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-03 17:29:00.795373 [DEBUG] switch_channel.c:2833 (sofia/internal/1004 at connexur.dyndns.org) Callstate Change ACTIVE -> HANGUP 2011-12-03 17:29:00.795373 [NOTICE] sofia.c:634 Hangup sofia/internal/1004 at connexur.dyndns.org [CS_EXECUTE] [NORMAL_CLEARING] 2011-12-03 17:29:00.795373 [DEBUG] switch_channel.c:2856 Send signal sofia/internal/1004 at connexur.dyndns.org [KILL] 2011-12-03 17:29:00.795373 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-03 17:29:00.795373 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/1004 at connexur.dyndns.org] 2011-12-03 17:29:00.795373 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/internal/sip:1001 at 192.168.0.43:5080 [BREAK] 2011-12-03 17:29:00.815375 [DEBUG] switch_ivr_bridge.c:586 BRIDGE THREAD DONE [sofia/internal/sip:1001 at 192.168.0.43:5080] 2011-12-03 17:29:00.815375 [DEBUG] switch_ivr_bridge.c:611 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-03 17:29:00.815375 [DEBUG] switch_channel.c:2833 (sofia/internal/sip:1001 at 192.168.0.43:5080) Callstate Change ACTIVE -> HANGUP 2011-12-03 17:29:00.815375 [NOTICE] switch_ivr_bridge.c:669 Hangup sofia/internal/sip:1001 at 192.168.0.43:5080 [CS_EXCHANGE_MEDIA] [NORMAL_CLEARING] 2011-12-03 17:29:00.815375 [DEBUG] switch_channel.c:2856 Send signal sofia/internal/sip:1001 at 192.168.0.43:5080 [KILL] 2011-12-03 17:29:00.815375 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/sip:1001 at 192.168.0.43:5080 [BREAK] 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:420 (sofia/internal/sip:1001 at 192.168.0.43:5080) State EXCHANGE_MEDIA going to sleep 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at 192.168.0.43:5080) Running State Change CS_HANGUP 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1001 at 192.168.0.43:5080) State HANGUP 2011-12-03 17:29:00.815375 [DEBUG] mod_sofia.c:465 Channel sofia/internal/sip:1001 at 192.168.0.43:5080 hanging up, cause: NORMAL_CLEARING 2011-12-03 17:29:00.815375 [DEBUG] mod_sofia.c:509 Sending BYE to sofia/internal/sip:1001 at 192.168.0.43:5080 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:47 sofia/internal/sip:1001 at 192.168.0.43:5080 Standard HANGUP, cause: NORMAL_CLEARING 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/sip:1001 at 192.168.0.43:5080) State HANGUP going to sleep 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/sip:1001 at 192.168.0.43:5080) State Change CS_HANGUP -> CS_REPORTING 2011-12-03 17:29:00.815375 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/sip:1001 at 192.168.0.43:5080 [BREAK] 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/sip:1001 at 192.168.0.43:5080) Running State Change CS_REPORTING 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1001 at 192.168.0.43:5080) State REPORTING 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:79 sofia/internal/sip:1001 at 192.168.0.43:5080 Standard REPORTING, cause: NORMAL_CLEARING 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/sip:1001 at 192.168.0.43:5080) State REPORTING going to sleep 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/sip:1001 at 192.168.0.43:5080) State Change CS_REPORTING -> CS_DESTROY 2011-12-03 17:29:00.815375 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/sip:1001 at 192.168.0.43:5080 [BREAK] 2011-12-03 17:29:00.815375 [DEBUG] switch_core_session.c:1377 Session 2 (sofia/internal/sip:1001 at 192.168.0.43:5080) Locked, Waiting on external entities 2011-12-03 17:29:00.815375 [DEBUG] switch_ivr_bridge.c:1367 sofia/internal/1004 at connexur.dyndns.org skip receive message [UNBRIDGE] (channel is hungup already) 2011-12-03 17:29:00.815375 [DEBUG] switch_core_session.c:2272 sofia/internal/1004 at connexur.dyndns.org skip receive message [APPLICATION_EXEC_COMPLETE] (channel is hungup already) 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:417 (sofia/internal/1004 at connexur.dyndns.org) State EXECUTE going to sleep 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_HANGUP 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1004 at connexur.dyndns.org) State HANGUP 2011-12-03 17:29:00.815375 [DEBUG] mod_sofia.c:459 sofia/internal/1004 at connexur.dyndns.org Overriding SIP cause 480 with 200 from the other leg 2011-12-03 17:29:00.815375 [DEBUG] mod_sofia.c:465 Channel sofia/internal/1004 at connexur.dyndns.org hanging up, cause: NORMAL_CLEARING 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:47 sofia/internal/1004 at connexur.dyndns.org Standard HANGUP, cause: NORMAL_CLEARING 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:602 (sofia/internal/1004 at connexur.dyndns.org) State HANGUP going to sleep 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:393 (sofia/internal/1004 at connexur.dyndns.org) State Change CS_HANGUP -> CS_REPORTING 2011-12-03 17:29:00.815375 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:362 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_REPORTING 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1004 at connexur.dyndns.org) State REPORTING 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:79 sofia/internal/1004 at connexur.dyndns.org Standard REPORTING, cause: NORMAL_CLEARING 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:662 (sofia/internal/1004 at connexur.dyndns.org) State REPORTING going to sleep 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:387 (sofia/internal/1004 at connexur.dyndns.org) State Change CS_REPORTING -> CS_DESTROY 2011-12-03 17:29:00.815375 [DEBUG] switch_core_session.c:1177 Send signal sofia/internal/1004 at connexur.dyndns.org [BREAK] 2011-12-03 17:29:00.815375 [DEBUG] switch_core_session.c:1377 Session 1 (sofia/internal/1004 at connexur.dyndns.org) Locked, Waiting on external entities 2011-12-03 17:29:00.815375 [NOTICE] switch_core_session.c:1395 Session 1 (sofia/internal/1004 at connexur.dyndns.org) Ended 2011-12-03 17:29:00.815375 [NOTICE] switch_core_session.c:1397 Close Channel sofia/internal/1004 at connexur.dyndns.org [CS_DESTROY] 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/1004 at connexur.dyndns.org) Callstate Change HANGUP -> DOWN 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/1004 at connexur.dyndns.org) Running State Change CS_DESTROY 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1004 at connexur.dyndns.org) State DESTROY 2011-12-03 17:29:00.815375 [DEBUG] mod_sofia.c:370 sofia/internal/1004 at connexur.dyndns.org SOFIA DESTROY 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:86 sofia/internal/1004 at connexur.dyndns.org Standard DESTROY 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/1004 at connexur.dyndns.org) State DESTROY going to sleep 2011-12-03 17:29:00.815375 [NOTICE] switch_core_session.c:1395 Session 2 (sofia/internal/sip:1001 at 192.168.0.43:5080) Ended 2011-12-03 17:29:00.815375 [NOTICE] switch_core_session.c:1397 Close Channel sofia/internal/sip:1001 at 192.168.0.43:5080 [CS_DESTROY] 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:491 (sofia/internal/sip:1001 at 192.168.0.43:5080) Callstate Change HANGUP -> DOWN 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:494 (sofia/internal/sip:1001 at 192.168.0.43:5080) Running State Change CS_DESTROY 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:1001 at 192.168.0.43:5080) State DESTROY 2011-12-03 17:29:00.815375 [DEBUG] mod_sofia.c:370 sofia/internal/sip:1001 at 192.168.0.43:5080 SOFIA DESTROY 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:86 sofia/internal/sip:1001 at 192.168.0.43:5080 Standard DESTROY 2011-12-03 17:29:00.815375 [DEBUG] switch_core_state_machine.c:504 (sofia/internal/sip:1001 at 192.168.0.43:5080) State DESTROY going to sleep -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-installed-ATA-registered-but-no-call-are-possible-tp7049935p7058216.html Sent from the freeswitch-users mailing list archive at Nabble.com. From elliott at zoogmedia.com Sat Dec 3 20:28:25 2011 From: elliott at zoogmedia.com (Elliott Vogel) Date: Sat, 3 Dec 2011 17:28:25 +0000 Subject: [Freeswitch-users] dialplan Message-ID: Hello, How can I bridge out calls via another bridge if the first bridge fails? - let's say the first bridge returns a 404 or 503 error -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111203/2f8109e1/attachment.html From paul at cupis.co.uk Sat Dec 3 22:33:10 2011 From: paul at cupis.co.uk (Paul Cupis) Date: Sat, 03 Dec 2011 19:33:10 +0000 Subject: [Freeswitch-users] dialplan In-Reply-To: References: Message-ID: <4EDA7976.1020803@cupis.co.uk> On 03/12/11 17:28, Elliott Vogel wrote: > How can I bridge out calls via another bridge if the first bridge fails? - let's say the first bridge returns a 404 or 503 error Have a look at the following: http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_7:_Action_failover_on_failed_action http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail Regards, From mstockton at harqen.com Sat Dec 3 23:49:11 2011 From: mstockton at harqen.com (Matt Stockton) Date: Sat, 3 Dec 2011 14:49:11 -0600 Subject: [Freeswitch-users] mod_rtmp: "Read error" when attempting calls from FS to RTMP client. In-Reply-To: <4ED947ED.2010208@livecall.com> References: <1322081549623-7026117.post@n2.nabble.com> <1322753056997-7050910.post@n2.nabble.com> <4ED947ED.2010208@livecall.com> Message-ID: I figured out what was happening for my specific problem that was closing the NetConnection. See my comment here: http://jira.freeswitch.org/browse/FS-3729 Basically, I was moving the flash widget to an off-screen location, and it doesn't look like Firefox likes this - Chrome and IE do not seem to care --- so there must be different rules in browsers for when Flash can operate --- Maybe your situation is similar? On Fri, Dec 2, 2011 at 3:49 PM, Jack wrote: > I also get : 2011-11-30 16:04:55.108875 [ERR] rtmp.c:678 Read error > each time an rtmp session from a browser FS Flex client is ended. It > doesn't seem to bother anything. > Jack > > > On 12/1/2011 2:35 PM, Matt Stockton wrote: > > Also worthwhile to note that the 'fsctl loglevel 9' does not make it work > for me, and the read error occurs still: > 2011-12-01 16:34:39.226836 [ERR] rtmp.c:713 Read error > > Also, even on Chrome (which is working), the read error does occur upon > hangup > > > On Thu, Dec 1, 2011 at 10:50 AM, Matt Stockton wrote: > >> For what it's worth, I am getting the exact same read error and it is >> closing my socket when I dial into FS from RTMP client, but only in certain >> browsers it seems. It's a read error on the same line of code. >> >> Description of the issue and debug lines are listed in this JIRA: >> http://jira.freeswitch.org/browse/FS-3729 >> >> >> On Thu, Dec 1, 2011 at 9:24 AM, peely wrote: >> >>> Sorry, can I bump this? >>> >>> I've tried everything I'm capable of, but can't get a call from mod_rtmp >>> out >>> to the client without keeping the box logging in debug mode! As soon as I >>> bring the log level down I get the issue, but of course can't get any >>> more >>> debug for this specific issue. >>> >>> I've looked at the code but don't know enough C to make any changes. >>> >>> >>> Thanks, >>> >>> >>> Neil. >>> >>> -- >>> View this message in context: >>> http://freeswitch-users.2379917.n2.nabble.com/mod-rtmp-Read-error-when-attempting-calls-from-FS-to-RTMP-client-tp7026117p7050910.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 >>> >> >> > > > _________________________________________________________________________ > 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/20111203/4d181686/attachment.html From acrow at integrafin.co.uk Sun Dec 4 00:28:36 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Sat, 03 Dec 2011 21:28:36 +0000 Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... In-Reply-To: <1322929911507-7058216.post@n2.nabble.com> References: <1322733452381-7049935.post@n2.nabble.com> <1322929911507-7058216.post@n2.nabble.com> Message-ID: <4EDA9484.9000000@integrafin.co.uk> On 03/12/11 16:31, obbyone wrote: > Hi, > > The trace I gave was not a good one. The one I show now Is a good sample. > Any attempt to dial "1001" results only in a response by voicemail. Any > answer ? > > Thanks > From your trace, it looks like your ATAs are connecting to your external profile (something.dyndns.org:5080). Are you trying to connect your ATAs somewhere "out on the internet", possibly NAT'ed to a NAT'ed or otherwise Freeswitch server? If the ATAs are on the LAN local to your FS box you should be using the internal profile and default SIP port of 5060 on the ATAs. If so you need to read up on the NAT scenarios if you don't UPNP compatible routers at both ends of the connection (or don't want to use STUN at both ends). It's tricky but it is possible to get it working even with NAT on both sides. I just got it working. Cheers Alex From moises.silva at gmail.com Sun Dec 4 00:48:45 2011 From: moises.silva at gmail.com (Moises Silva) Date: Sat, 3 Dec 2011 16:48:45 -0500 Subject: [Freeswitch-users] FreeTDM does not work In-Reply-To: References: Message-ID: On Sun, Nov 27, 2011 at 8:45 PM, Valery Kalinin wrote: > I upgraded freeswitch from git. > Nothing changed in the configuration! > Suddenly stopped working freetdm. Why? > Your email is dated Nov 27, but the error messages you mention were modified in Nov 10. You are clearly not at latest git. *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 VegaStream is now part of Sangoma! Ask us about both Gateway Appliances and Internal Gateways -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111203/44797fc2/attachment-0001.html From freeswitch at peely.com Sun Dec 4 01:18:14 2011 From: freeswitch at peely.com (peely) Date: Sat, 3 Dec 2011 14:18:14 -0800 (PST) Subject: [Freeswitch-users] mod_rtmp: "Read error" when attempting calls from FS to RTMP client. In-Reply-To: References: <1322081549623-7026117.post@n2.nabble.com> <1322753056997-7050910.post@n2.nabble.com> <4ED947ED.2010208@livecall.com> Message-ID: <1322950694672-7058838.post@n2.nabble.com> I doubt it, as stated in my OP, my control works perfectly when the debug level is set top maximum on FreeSWITCH, it's only when the debug level is lowered. Plus I use exactly the same control for making and receiving calls, making calls from the client is fine, but receiving a call fails. In both situations I don;t move the control after initialising it, everything else is controlled through JS with in headless. I'm really stuck and am having to rethink how I use the control to get around this as there doesn't seem to be any activity. I'm thinking even when I want to receive a call I'll have to somehow initiate a call and park it, then switch an incoming call into that session or something. It's definitely something going on in mod_rtmp. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-rtmp-Read-error-when-attempting-calls-from-FS-to-RTMP-client-tp7026117p7058838.html Sent from the freeswitch-users mailing list archive at Nabble.com. From saeedahmad1981 at gmail.com Sun Dec 4 15:33:00 2011 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Sun, 4 Dec 2011 13:33:00 +0100 Subject: [Freeswitch-users] get cepstral volume lower In-Reply-To: <854F4E8A03274B099B3DD8819DE0721B@e1705> References: <854F4E8A03274B099B3DD8819DE0721B@e1705> Message-ID: Trying to do the same with no success. have you got it working? On Mon, Jan 17, 2011 at 5:43 AM, Madovsky wrote: > ** > I'm trying the SSML example from mod_cepstral wiki > > > > but the voice doesn't say the sentence but says something like "slash > blablabla...". > > Any idea why I can't embed SSML tag in the cepstral sentence ? > > Thanks > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111204/958a8065/attachment.html From notlikeme75 at yahoo.com Sun Dec 4 22:44:37 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Sun, 4 Dec 2011 11:44:37 -0800 (PST) Subject: [Freeswitch-users] play and get digits with dynamic conference Message-ID: <1323027877.27264.YahooMailNeo@web65303.mail.ac2.yahoo.com> I have tried the nb_conference example in the default.xml but I think I am doing something wrong. when i transfer to that extension it drops me into a single static conference of the variable . what do i need to add or change to prompt my callers transfered to the nb_conference to enter a 4 digit conference number without pin that gives them the same options as my static conferences on the default profile? I know play and get digits must be involved but would appreciate the help. thanks. ??? ????? ??? ??? ????? ??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111204/faa63e56/attachment.html From brian at freeswitch.org Mon Dec 5 03:46:27 2011 From: brian at freeswitch.org (Brian West) Date: Sun, 4 Dec 2011 18:46:27 -0600 Subject: [Freeswitch-users] get cepstral volume lower In-Reply-To: References: <854F4E8A03274B099B3DD8819DE0721B@e1705> Message-ID: <6B99B350-D6D4-40FF-8073-9482A867A93B@freeswitch.org> Because thats invalid XML :P Try something like this: Thank you for holding]]> /b On Dec 4, 2011, at 6:33 AM, Saeed Ahmed wrote: > Trying to do the same with no success. > > have you got it working? > > On Mon, Jan 17, 2011 at 5:43 AM, Madovsky wrote: > >> ** >> I'm trying the SSML example from mod_cepstral wiki >> >> >> >> but the voice doesn't say the sentence but says something like "slash >> blablabla...". -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111204/1c1396ce/attachment.html From admin at blindi.net Mon Dec 5 07:14:13 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Mon, 5 Dec 2011 05:14:13 +0100 (CET) Subject: [Freeswitch-users] Answerconfirmation with ivr not working In-Reply-To: References: Message-ID: Hi all, i create a callscreening extension with: My ivrmenu: Glows when the phone and I answer it, then I hear my outgoing message. But i can.t transfer both channels to the conference. i here my menuprompt in the conference. But not the caller. What is wrong? Can your help please? thanks. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From miha at softnet.si Mon Dec 5 13:04:20 2011 From: miha at softnet.si (Miha Zoubek) Date: Mon, 05 Dec 2011 11:04:20 +0100 Subject: [Freeswitch-users] Segmentation fault (core dumped) Message-ID: <4EDC9724.90603@softnet.si> Hi, I have added this in mod_radius_cdr.c and run make&make install. if (channel){ const char *no_radius_start = switch_channel_get_variable(channel, "no_radius_start"); if (switch_true(no_radius_start)){ switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "[mod_radius_cdr] Skipping Radius Start\n"); return SWITCH_STATUS_SUCCESS; } } and for radius_stop. if (channel){ const char *no_radius_stop = switch_channel_get_variable(channel, "no_radius_stop"); if (switch_true(no_radius_stop)){ switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "[mod_radius_cdr] Skipping Radius Stop\n"); return SWITCH_STATUS_SUCCESS; } } When I run call a get this: Segmentation fault (core dumped) Do you know why I am getting this? Regrdas, Miha -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/fc194d0a/attachment-0001.html From acrow at integrafin.co.uk Mon Dec 5 15:09:12 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Mon, 05 Dec 2011 12:09:12 +0000 Subject: [Freeswitch-users] Created new profile for double-natted phones, calling these fails with USER_NOT_REGISTERED Message-ID: <4EDCB468.2060001@integrafin.co.uk> Hi all, I have set up a new profile for phones behind NAT connecting to an FS box behind NAT. Calls in from a NAT'ed phone works fine, however calls out to one fail with USER_NOT_REGISTERED. All the phones are set up in /opt/freeswitch/conf/directory/default/. I have added the following in the new profile config hoping it would resolve the issue but it does not: Any clues? Thanks Alex From charlie.orford at attackplan.net Mon Dec 5 15:33:54 2011 From: charlie.orford at attackplan.net (Charlie Orford) Date: Mon, 05 Dec 2011 13:33:54 +0100 Subject: [Freeswitch-users] "481 Call/Transaction Does Not Exist" when hanging up before call connects In-Reply-To: <4EDA09FC.9000504@attackplan.net> References: <4ED7A88F.1010302@attackplan.net> <4ED8B456.9000505@attackplan.net> <4ED8BB73.8080401@ezuce.com> <4ED8C206.5010502@attackplan.net> <4ED9B3A8.4040707@ezuce.com> <4EDA09FC.9000504@attackplan.net> Message-ID: <4EDCBA32.4090608@attackplan.net> Hi Joegen, I just wanted to say a quick thank you for your advice - it worked. As soon as I turned off all the NAT related stuff on the Aastra phone (basically turned off rport and removed the stun and turn servers), the phone stopped changing its VIA header and CANCEL worked as advertised. In case it's useful to anyone else in a similar position, I have the following NAT related stuff enabled on the internal FS sip profile (note: my FS server is on a public IP i.e. not itself behind NAT): Charlie On 03/12/2011 12:37, Charlie Orford wrote: > Bummer - we've got an office full of these phones :( > > As you suggest, I'll turn off all the NAT stuff when I'm there on > Monday and see if that fixes it - hope so, otherwise I guess it looks > like I'll have to try and get Aastra to look at the problem (doubt > they are going to be very interested though!). > > Thanks, > Charlie > > > On 03/12/2011 06:29, Joegen Baclor wrote: >> It's definitely a bug on the phone. I would try setting all Aastra >> NAT related features off and see if it stops getting confused like >> this. Freeswitch should be able to handle NAT from the far end. >> >> On 12/02/2011 08:18 PM, Charlie Orford wrote: >>> Thanks Joegen, I hadn't picked up on that. >>> >>> Is there any valid reason why the phone would swap to using it's >>> internal IP (192.168.0.175) for the Via and Contact headers when >>> responding to the auth challenge? Or does this look more like a bug >>> with the phone? >>> >>> >>> >>> On 02/12/2011 12:50, Joegen Baclor wrote: >>>> For some reason, the aastra phone is sending a different contact >>>> and via address after it gets challenged by freeswitch. >>>> >>>> Original address was >>>> >>>> Via: >>>> SIP/2.0/UDP7.7.7.7:6060;branch=z9hG4bK2ff03b684c51c6e9a.84815b35f84df80f4;rport >>>> (LINE# 35) >>>> >>>> Then it changes to >>>> >>>> Via: >>>> SIP/2.0/UDP192.168.0.175:6060;branch=z9hG4bK085dcaff3e1ae346b.e0c0723efae179659;rport >>>> (LINE# 187) >>>> >>>> It then sends a CANCEL using Via: SIP/2.0/UDP7.7.7.7:6060 (LINE# >>>> 961) which freeswitch isn't able to match to a transaction because >>>> it is probably expecting 192.168.0.175. >>>> >>>> >>>> >>>> On 12/02/2011 07:19 PM, Charlie Orford wrote: >>>>> Hi Anthony >>>>> >>>>> We are using Aastra 57i phones with the latest firmware (v3.2.2.56 >>>>> from June 2011). >>>>> >>>>> However, it looks like a specific issue with the Aastra 57i as I >>>>> have just replicated the call and cancel situation using a 3CX >>>>> softphone and in this case the CANCEL request gets honoured by FS. >>>>> >>>>> Here's a copy of a failed CANCEL test using the Aastra phone: >>>>> http://pastebin.freeswitch.org/17922 >>>>> >>>>> Here's a copy of a successful CANCEL test using the 3CX softphone: >>>>> http://pastebin.freeswitch.org/17923 >>>>> >>>>> sofia loglevel all 9 is on for both tests. >>>>> >>>>> Charlie >>>>> >>>>> >>>>> On 01/12/2011 21:53, Anthony Minessale wrote: >>>>>> try same failed call test with sofia loglevel all 9 >>>>>> >>>>>> Also, try some other phone or device that does not have the >>>>>> problem, and create a cancel situation the same way and see if >>>>>> you can find a difference. >>>>>> >>>>>> Finally, have you tried the latest firmware on the phone? >>>>>> >>>>>> >>>>>> On Thu, Dec 1, 2011 at 10:17 AM, Charlie Orford >>>>>> >>>>> > wrote: >>>>>> >>>>>> Hi list, >>>>>> >>>>>> When we make a call from an FS extension to a PSTN number >>>>>> (via our ITSP >>>>>> gateway provider) and hang-up before the call completes, FS >>>>>> replies to >>>>>> the CANCEL request with "481 Call/Transaction Does Not Exist" >>>>>> and the >>>>>> call continues to ring on the remote end. If we hangup after >>>>>> the call >>>>>> has connected, it works with no problem. >>>>>> >>>>>> I have looked through a SIP trace of this happening and to my >>>>>> (untrained >>>>>> eye) nothing seems obviously wrong (i.e. tag, call id and >>>>>> branch values >>>>>> all seem to be correct). I'm using the latest git snapshot from >>>>>> 2011-11-30 18-14-24 -0600. >>>>>> >>>>>> For a sip trace showing the problem, see: >>>>>> http://pastebin.freeswitch.org/17908 >>>>>> >>>>>> For a sip trace showing a successful hangup, see: >>>>>> http://pastebin.freeswitch.org/17909 >>>>>> >>>>>> *Note: ip addresses, domain and called number have been >>>>>> altered for privacy. >>>>>> >>>>>> Any help or insight is much appreciated. >>>>>> >>>>>> Kind Regards, >>>>>> Charlie >>>>>> >>>>>> >>>>>> _________________________________________________________________________ >>>>>> Professional FreeSWITCH Consulting Services: >>>>>> consulting at freeswitch.org >>>>>> http://www.freeswitchsolutions.com >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> Official FreeSWITCH Sites >>>>>> http://www.freeswitch.org >>>>>> http://wiki.freeswitch.org >>>>>> http://www.cluecon.com >>>>>> >>>>>> FreeSWITCH-users mailing list >>>>>> FreeSWITCH-users at lists.freeswitch.org >>>>>> >>>>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>>>> UNSUBSCRIBE:http://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 >>>> >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/0cfcbb96/attachment-0001.html From acosgrov at gmail.com Mon Dec 5 16:04:56 2011 From: acosgrov at gmail.com (Anthony Cosgrove) Date: Mon, 5 Dec 2011 08:04:56 -0500 Subject: [Freeswitch-users] Created new profile for double-natted phones, calling these fails with USER_NOT_REGISTERED In-Reply-To: <4EDCB468.2060001@integrafin.co.uk> References: <4EDCB468.2060001@integrafin.co.uk> Message-ID: <9E6ADFB3-5BB6-4D45-8CF1-5C0DA06BB76D@gmail.com> Alex, Does FS *have* to be behind a NAT? Are you doing a straight 1:1 from public to private (FS side)? What kind of router/firewall are you using on the FS side and endpoint sides? You'll will probably want to lower the endpoint registration times from the normal default of 1 hour to 180 seconds (3 minutes) or even lower depending on router/firewall. You may also want to turn on keep-alive packets to keep the data flowing in/out. What you are running into is either one or both sides are closing off 5060/udp early, FS is marking the registration as dead. Best thing to do is to place FS in a DMZ or install a public facing SBC to take the registrations and forward them on to FS internally. Anthony On Dec 5, 2011, at 7:09 AM, Alex Crow wrote: > Hi all, > > I have set up a new profile for phones behind NAT connecting to an FS > box behind NAT. Calls in from a NAT'ed phone works fine, however calls > out to one fail with USER_NOT_REGISTERED. All the phones are set up in > /opt/freeswitch/conf/directory/default/. > > I have added the following in the new profile config hoping it would > resolve the issue but it does not: > > > > > > > Any clues? > > Thanks > > Alex > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From dgarcia at anew.com.ve Mon Dec 5 16:12:53 2011 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Mon, 05 Dec 2011 08:42:53 -0430 Subject: [Freeswitch-users] acl validation In-Reply-To: <31CBF2DFBF224915A5FBB11AC495D24C@DWP> References: <31CBF2DFBF224915A5FBB11AC495D24C@DWP> Message-ID: <4EDCC355.9090600@anew.com.ve> Hi, Take a look to this reference http://arstechnica.com/tech-policy/news/2011/11/how-filipino-phreakers-turned-pbx-systems-into-cash-machines-for-terrorists.ars Hacker, exploiters, phreakers, etc, etc are reals. Try to secure your network, specially if you handle sensible data (ex. customers info, accounts, etc). Add firewalls, Web App Firewalls, vlan, and other security to your network. Is not good idea to have your FS host machine connected directly to internet relaying in basic protection. On 12/2/2011 5:31 PM, Darcy wrote: > Looking through my call log, I see the following item in the log, > this IP is not in the acl Domain List, it is not a profile and it is > not registered on the switch, Is there something I am missing on this > that needs to be set. These guys are attempting calls to Jamaica. > 2011-10-31 09:11:31.801420 [DEBUG] sofia.c:5798 IP 91.212.226.30 > Approved by acl "domains[]". Access Granted. > > > I have 5 more entries like so in this switch. > etc. > > Darcy Primrose > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: 2012.0.1873 / Virus Database: 2102/4658 - Release Date: 12/05/11 > -- 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/20111205/b0194329/attachment.html From acrow at integrafin.co.uk Mon Dec 5 16:19:33 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Mon, 05 Dec 2011 13:19:33 +0000 Subject: [Freeswitch-users] Created new profile for double-natted phones, calling these fails with USER_NOT_REGISTERED In-Reply-To: <9E6ADFB3-5BB6-4D45-8CF1-5C0DA06BB76D@gmail.com> References: <4EDCB468.2060001@integrafin.co.uk> <9E6ADFB3-5BB6-4D45-8CF1-5C0DA06BB76D@gmail.com> Message-ID: <4EDCC4E5.803@integrafin.co.uk> On 05/12/11 13:04, Anthony Cosgrove wrote: > Alex, > > Does FS *have* to be behind a NAT? Are you doing a straight 1:1 from public to private (FS side)? What kind of router/firewall are you using on the FS side and endpoint sides? You'll will probably want to lower the endpoint registration times from the normal default of 1 hour to 180 seconds (3 minutes) or even lower depending on router/firewall. You may also want to turn on keep-alive packets to keep the data flowing in/out. What you are running into is either one or both sides are closing off 5060/udp early, FS is marking the registration as dead. > Anthony, No, it's definitely not that - the registration is alive and well. The issue is that FS sees that the user is not registered on the internal sip profile and doesn't check the doublenat profile. The router/firewall is iptables (shorewall) and I'm doing a DNAT of the relevant ports to the FS box. What I have found is that I can specify the dialstring for the external users in the directory entries, eg: > Best thing to do is to place FS in a DMZ or install a public facing SBC to take the registrations and forward them on to FS internally. > Yes, it would probably be better that way, but this is really mostly for internal use. Cheers Alex > > Anthony > > On Dec 5, 2011, at 7:09 AM, Alex Crow wrote: > >> Hi all, >> >> I have set up a new profile for phones behind NAT connecting to an FS >> box behind NAT. Calls in from a NAT'ed phone works fine, however calls >> out to one fail with USER_NOT_REGISTERED. All the phones are set up in >> /opt/freeswitch/conf/directory/default/. >> >> I have added the following in the new profile config hoping it would >> resolve the issue but it does not: >> >> >> >> >> >> >> Any clues? >> >> Thanks >> >> Alex >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From saeedahmad1981 at gmail.com Mon Dec 5 17:53:16 2011 From: saeedahmad1981 at gmail.com (Saeed Ahmed) Date: Mon, 5 Dec 2011 15:53:16 +0100 Subject: [Freeswitch-users] get cepstral volume lower In-Reply-To: <6B99B350-D6D4-40FF-8073-9482A867A93B@freeswitch.org> References: <854F4E8A03274B099B3DD8819DE0721B@e1705> <6B99B350-D6D4-40FF-8073-9482A867A93B@freeswitch.org> Message-ID: Hi Brian, It looks better but still doesn't work: Ich kann deutsch sprechen."]]> Logs shows: Speaking text: Hallo, Ich kann deutsch sprechen. *ssml: expected "="* So it only says 'Hallo" and then disconnect. Thanks. On Mon, Dec 5, 2011 at 1:46 AM, Brian West wrote: > Because thats invalid XML :P > > Try something like this: > > volume=13>Thank you for holding]]> > > /b > > > On Dec 4, 2011, at 6:33 AM, Saeed Ahmed wrote: > > Trying to do the same with no success. > > have you got it working? > > On Mon, Jan 17, 2011 at 5:43 AM, Madovsky wrote: > > ** > > I'm trying the SSML example from mod_cepstral wiki > > > > > > but the voice doesn't say the sentence but says something like "slash > > blablabla...". > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/f92c7da4/attachment.html From beppe.grillo at gmail.com Mon Dec 5 18:12:28 2011 From: beppe.grillo at gmail.com (Beppe Grillo) Date: Mon, 5 Dec 2011 16:12:28 +0100 Subject: [Freeswitch-users] REGISTER using the draft-sip-outbound Message-ID: Hy all I'd like send a REGISTER using the draft-sip-outbound. However, the contact header generated by freeswitch, hasn't the reg-id and +sip.instance params. I have last git (2011-11-28). Can your help please? Thanks Giuseppe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/5b206c30/attachment.html From brian at freeswitch.org Mon Dec 5 18:43:31 2011 From: brian at freeswitch.org (Brian West) Date: Mon, 5 Dec 2011 09:43:31 -0600 Subject: [Freeswitch-users] REGISTER using the draft-sip-outbound In-Reply-To: References: Message-ID: write a patch and put it on jira.freeswitch.org, I don't think we have that feature yet. /b On Dec 5, 2011, at 9:12 AM, Beppe Grillo wrote: > Can your help please? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/3959d4ba/attachment-0001.html From turqmr2 at gmail.com Mon Dec 5 15:41:27 2011 From: turqmr2 at gmail.com (Jacob Smith) Date: Mon, 05 Dec 2011 07:41:27 -0500 Subject: [Freeswitch-users] Almost there... Dingaling help please Message-ID: <4EDCBBF7.30205@gmail.com> At this point, I can make calls using Dingaling through Google Voice. The problem is receiving: the phone rings, but when I answer there is silence and after a few seconds, disconnect. Here is a log from one attempt: http://pastebin.freeswitch.org/17937 The Jingle profile and dialplan is just like http://wiki.freeswitch.org/wiki/Google_Voice Hoping that this is an easy fix, I have one more question. How can I answer the call with an IVR instead of sending the call straight to an extension? The only forwarding option I can find is in the Jingle profile where you specify the extension. Thank you, Jacob From msc at freeswitch.org Mon Dec 5 19:08:51 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 5 Dec 2011 08:08:51 -0800 Subject: [Freeswitch-users] Segmentation fault (core dumped) In-Reply-To: <4EDC9724.90603@softnet.si> References: <4EDC9724.90603@softnet.si> Message-ID: FYI, This message is a bit intense for the -users list. I recommend sending to the -dev list instead. -MC On Mon, Dec 5, 2011 at 2:04 AM, Miha Zoubek wrote: > Hi, > > I have added this in mod_radius_cdr.c and run make&make install. > > if (channel){ > const char *no_radius_start = > switch_channel_get_variable(channel, "no_radius_start"); > if (switch_true(no_radius_start)){ > switch_log_printf(SWITCH_CHANNEL_LOG, > SWITCH_LOG_DEBUG, "[mod_radius_cdr] Skipping Radius Start\n"); > return SWITCH_STATUS_SUCCESS; > } > } > > > and for radius_stop. > > if (channel){ > const char *no_radius_stop = > switch_channel_get_variable(channel, "no_radius_stop"); > if (switch_true(no_radius_stop)){ > switch_log_printf(SWITCH_CHANNEL_LOG, > SWITCH_LOG_DEBUG, "[mod_radius_cdr] Skipping Radius Stop\n"); > return SWITCH_STATUS_SUCCESS; > } > } > > When I run call a get this: > > Segmentation fault (core dumped) > > > Do you know why I am getting this? > > Regrdas, > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/260761b2/attachment.html From msc at freeswitch.org Mon Dec 5 19:21:41 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 5 Dec 2011 08:21:41 -0800 Subject: [Freeswitch-users] Softphones could not call each other but could register on the FS server In-Reply-To: <1322928441467-7058171.post@n2.nabble.com> References: <1322928441467-7058171.post@n2.nabble.com> Message-ID: Use pastebin to show us the console log of the call failing. Capture the output from these commands: sofia status sofia status profile sofia status profile reg Then enable siptrace and make a test call, capturing the console output: sofia global siptrace on Put all that stuff in pastebin.freeswitch.org using "FreeSWITCH Log" as the syntax highlighting. Reply to this thread with the URL and the gang here will take a look. -MC On Sat, Dec 3, 2011 at 8:07 AM, ayobami wrote: > I just git pulled the Freeswitch source code and compiled, everything went > well, I mean the compilation, but the problem that arose now is sip phones > could not call each other again, now the soft phones could register on the > FS server because I could use them to call 5000, 4000 and some of my custom > numbers, but when I tried to call an extension from another extension, the > operator keep saying that the extension is not available, I have > troubleshooted and dont know what to do, please people help me out. > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Softphones-could-not-call-each-other-but-could-register-on-the-FS-server-tp7058171p7058171.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/20111205/09ae7993/attachment.html From msc at freeswitch.org Mon Dec 5 19:34:51 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 5 Dec 2011 08:34:51 -0800 Subject: [Freeswitch-users] play and get digits with dynamic conference In-Reply-To: <1323027877.27264.YahooMailNeo@web65303.mail.ac2.yahoo.com> References: <1323027877.27264.YahooMailNeo@web65303.mail.ac2.yahoo.com> Message-ID: I believe you just need to call play_and_get_digits (PAGD) prior to dropping them into the conference. If I understand the question correctly, the four-digits that the caller enters represents the "conference number" or whatever you call it. You could do this: naturally you'll need to read up on PAGD so that you understand what all that stuff is doing. Also, be sure to specify real sound files, not the fake ones that I used. The ${digits} value is what the caller actually dials. In a production environment you'll need to handle the scenario where the caller never actually enters a valid 4-digit number. -MC On Sun, Dec 4, 2011 at 11:44 AM, Rodney wrote: > I have tried the nb_conference example in the default.xml but I think I am > doing something wrong. when i transfer to that extension it drops me into a > single static conference of the variable . what do i need to add or change > to prompt my callers transfered to the nb_conference to enter a 4 digit > conference number without pin that gives them the same options as my static > conferences on the default profile? I know play and get digits must be > involved but would appreciate the help. 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/20111205/885055fc/attachment-0001.html From freeswitch at peely.com Mon Dec 5 19:40:54 2011 From: freeswitch at peely.com (peely) Date: Mon, 5 Dec 2011 08:40:54 -0800 (PST) Subject: [Freeswitch-users] REGISTER using the draft-sip-outbound In-Reply-To: References: Message-ID: <1323103254684-7063523.post@n2.nabble.com> Isn't this now RFC5626? This is more of a client-side standard, are you wanting FreeSWITCH to register a gateway using this standard? Doesn't seem that appropriate to me. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/REGISTER-using-the-draft-sip-outbound-tp7063199p7063523.html Sent from the freeswitch-users mailing list archive at Nabble.com. From dhairya.blogs at gmail.com Mon Dec 5 19:50:39 2011 From: dhairya.blogs at gmail.com (Dhairya Vora) Date: Mon, 5 Dec 2011 22:20:39 +0530 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error Message-ID: Hi group, I am getting this error while generating outbound calls using freeswitch. *freeswitch at server> 2011-12-05 22:01:44.760933 [ERR] mod_sofia.c:4279 Invalid URL 2011-12-05 22:01:44.760933 [NOTICE] mod_sofia.c:4660 Close Channel N/A [CS_NEW] 2011-12-05 22:01:44.760933 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2011-12-05 22:03:50.208776 [ERR] mod_sofia.c:4279 Invalid URL* Can anyone help me with this? -Dhairya -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/dd374753/attachment.html From brian at freeswitch.org Mon Dec 5 20:00:26 2011 From: brian at freeswitch.org (Brian West) Date: Mon, 5 Dec 2011 11:00:26 -0600 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error In-Reply-To: References: Message-ID: Can you give us more to go on? You seem to have cut out the info required to make any sort of recommendation.. how are you trying to place a call? Could the remote just not like your number format? /b On Dec 5, 2011, at 10:50 AM, Dhairya Vora wrote: > Hi group, > I am getting this error while generating outbound calls using freeswitch. > *freeswitch at server> 2011-12-05 22:01:44.760933 [ERR] mod_sofia.c:4279 > Invalid URL > 2011-12-05 22:01:44.760933 [NOTICE] mod_sofia.c:4660 Close Channel N/A > [CS_NEW] > 2011-12-05 22:01:44.760933 [NOTICE] switch_ivr_originate.c:2459 Cannot > create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] > 2011-12-05 22:03:50.208776 [ERR] mod_sofia.c:4279 Invalid URL* > > Can anyone help me with this? > > -Dhairya -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/00c74c6f/attachment.html From anita.hall at simmortel.com Mon Dec 5 20:02:05 2011 From: anita.hall at simmortel.com (Anita Hall) Date: Mon, 5 Dec 2011 22:32:05 +0530 Subject: [Freeswitch-users] SS7 on FreeSWITCH Message-ID: Hi all ! I have to support SS7 on FreeSWITCH for a client. He told me that Asterisk is _not_ good enough for him :) He has built entire infrastructure on FS. libss7, chan_ss7, openss7, ss7box, Sangoma Media Gateway ? What all options do I have ? If none of these existing options will work with FreeSWITCH, is there anyone who could modify chan_ss7 for FreeSWITCH or provide some other alternative ? Thanks, Anita. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/2d2681c9/attachment.html From msc at freeswitch.org Mon Dec 5 20:05:15 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 5 Dec 2011 09:05:15 -0800 Subject: [Freeswitch-users] SS7 on FreeSWITCH In-Reply-To: References: Message-ID: Obviously your client is well-informed. ;) Check with the guys at Sangoma - they're the SS7 gatekeepers. -MC On Mon, Dec 5, 2011 at 9:02 AM, Anita Hall wrote: > Hi all ! > > I have to support SS7 on FreeSWITCH for a client. He told me that Asterisk > is _not_ good enough for him :) He has built entire infrastructure on FS. > > libss7, chan_ss7, openss7, ss7box, Sangoma Media Gateway ? What all > options do I have ? > > If none of these existing options will work with FreeSWITCH, is there > anyone who could modify chan_ss7 for FreeSWITCH or provide some other > alternative ? > > Thanks, > Anita. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/0811c9a7/attachment.html From msc at freeswitch.org Mon Dec 5 20:07:02 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 5 Dec 2011 09:07:02 -0800 Subject: [Freeswitch-users] Almost there... Dingaling help please In-Reply-To: <4EDCBBF7.30205@gmail.com> References: <4EDCBBF7.30205@gmail.com> Message-ID: The extension that you specify in your jingle profile does not have to be a "user" extension - it can be any extension, including an IVR. If you have the default dialplan you could send the call to 5000 and have it use the demo IVR as a test to make sure it works. -MC On Mon, Dec 5, 2011 at 4:41 AM, Jacob Smith wrote: > At this point, I can make calls using Dingaling through Google Voice. > The problem is receiving: the phone rings, but when I answer there is > silence and after a few seconds, disconnect. > > Here is a log from one attempt: http://pastebin.freeswitch.org/17937 > > The Jingle profile and dialplan is just like > http://wiki.freeswitch.org/wiki/Google_Voice > > Hoping that this is an easy fix, I have one more question. How can I > answer the call with an IVR instead of sending the call straight to an > extension? The only forwarding option I can find is in the Jingle > profile where you specify the extension. > > Thank you, > Jacob > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/e32ec6d0/attachment.html From dhairya.blogs at gmail.com Mon Dec 5 20:08:19 2011 From: dhairya.blogs at gmail.com (Dhairya Vora) Date: Mon, 5 Dec 2011 22:38:19 +0530 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error In-Reply-To: References: Message-ID: Hi Brian, Thanks for reply. Here are more details. *2011-12-05 22:30:31.541222 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable [plivo_request_uuid]=[a3d8554c-1f62-11e1-b2be-4040af60d0d9] 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable [plivo_answer_url]=[http://dev.MMMMMM.net/namo/index.php] 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable [plivo_ring_url]=[http://dev.MMMMMM.net/namo/ring.php] 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable [plivo_hangup_url]=[http://dev.MMMMMM.net/namo/hangUp.php] 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable [origination_caller_id_number]=[0012018200770] 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable [plivo_from]=[0012018200770] 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable [plivo_to]=[00919426973436] 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable [plivo_app]=[true] 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable [absolute_codec_string]=[PCMU] 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable [originate_timeout]=[60] 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable [ignore_early_media]=[true] 2011-12-05 22:30:31.541222 [ERR] mod_sofia.c:4279 Invalid URL 2011-12-05 22:30:31.541222 [NOTICE] mod_sofia.c:4660 Close Channel N/A [CS_NEW] 2011-12-05 22:30:31.541222 [DEBUG] switch_core_state_machine.c:494 () Running State Change CS_DESTROY 2011-12-05 22:30:31.541222 [DEBUG] switch_core_state_machine.c:504 (N/A) State DESTROY 2011-12-05 22:30:31.541222 [DEBUG] mod_sofia.c:370 N/A SOFIA DESTROY 2011-12-05 22:30:31.541222 [DEBUG] switch_core_state_machine.c:504 (N/A) State DESTROY going to sleep 2011-12-05 22:30:31.541222 [NOTICE] switch_ivr_originate.c:2459 Cannot create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] 2011-12-05 22:30:31.541222 [DEBUG] switch_ivr_originate.c:3367 Originate Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]* I am placing the call using plivo. Please let me know if any other information would be useful. Thanks, Dhairya On Mon, Dec 5, 2011 at 10:30 PM, Brian West wrote: > Can you give us more to go on? You seem to have cut out the info required > to make any sort of recommendation.. how are you trying to place a call? > Could the remote just not like your number format? > > /b > > On Dec 5, 2011, at 10:50 AM, Dhairya Vora wrote: > > Hi group, > I am getting this error while generating outbound calls using freeswitch. > *freeswitch at server> 2011-12-05 22:01:44.760933 [ERR] mod_sofia.c:4279 > > Invalid URL > 2011-12-05 22:01:44.760933 [NOTICE] mod_sofia.c:4660 Close Channel N/A > [CS_NEW] > 2011-12-05 22:01:44.760933 [NOTICE] switch_ivr_originate.c:2459 Cannot > create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] > 2011-12-05 22:03:50.208776 [ERR] mod_sofia.c:4279 Invalid URL* > > > Can anyone help me with this? > > -Dhairya > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/a739e47b/attachment-0001.html From jefersonparanaense at gmail.com Mon Dec 5 20:14:23 2011 From: jefersonparanaense at gmail.com (Jeferson Rodrigo Almeida) Date: Mon, 5 Dec 2011 15:14:23 -0200 Subject: [Freeswitch-users] mod_callcenter ringback - help In-Reply-To: <4ED93D75.9010204@anew.com.ve> References: <4ED93D75.9010204@anew.com.ve> Message-ID: Thanks for the help. I'll try to do this configurations, and than I put here the results. 2011/12/2 Saugort Dario Garcia Tovar > Also, check: > > > http://freeswitch-users.2379917.n2.nabble.com/Using-fifo-orbit-announce-For-On-hook-Agent-td5222158.html > > > > On 12/2/2011 1:34 PM, Jeferson Rodrigo Almeida wrote: > > Thanks for the reply Brad, but my problem is other, I'll try to be > clearer... > > When a member enters in a queue, and all agents are busy, he stays > listening to the moh sound. When a agent goes to Waiting state, the call is > bridged to him. In this moment, I want that the member star to listen a > ringback (to know that his call is going to an agent). > > The link that you sent only explains how to set a moh sound, that will > play continuously, doesn't matter if the agent is ringing or not. > > Is there any way to make this? > > Thanks again for the attention... > > 2011/12/2 Brad Mina > >> Jeferson, >> >> I'd recommend you look at the following links: >> http://wiki.freeswitch.org/wiki/Mod_callcenter#moh-sound >> >> http://wiki.freeswitch.org/wiki/Mod_local_stream >> >> On Fri, Dec 2, 2011 at 8:17 AM, Jeferson Rodrigo Almeida < >> jefersonparanaense at gmail.com> wrote: >> >>> Hi... >>>> >>>> I'm using mod_callcenter >>>> >>>> I need to send a ringback (or any other sound, different of the >>>> default moh) to the caller when a agent phone is ringing, and I don't know >>>> how to do it. >>>> >>>> Basically, I need to playback other sound to the caller when >>>> agent-offering event is fired. >>>> >>>> Is there any way to configure a dialplan after the call is bridged to >>>> the agent? >>>> Is there any way to modify the moh for this call in real-time? (I can >>>> capture the event...) >>>> >>>> Any help will be appreciated. >>>> >>>> Ps.: Sorry for any mistake in my "school" english. >>>> >>>> Thanks >>>> >>>> -- >>>> Jeferson Rodrigo Almeida >>>> Engenheiro de Computa??o >>>> jefersonparanaense at gmail.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 >> >> > > > -- > Jeferson Rodrigo Almeida > Engenheiro de Computa??o > jefersonparanaense at gmail.com > > > _________________________________________________________________________ > 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 > > > > No virus found in this message. > Checked by AVG - www.avg.com > Version: 2012.0.1873 / Virus Database: 2102/4652 - Release Date: 12/02/11 > > > > -- > 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 > > -- Jeferson Rodrigo Almeida Engenheiro de Computa??o jefersonparanaense at gmail.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/5fedfde0/attachment.html From dhairya.blogs at gmail.com Mon Dec 5 20:17:11 2011 From: dhairya.blogs at gmail.com (Dhairya Vora) Date: Mon, 5 Dec 2011 22:47:11 +0530 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error In-Reply-To: References: Message-ID: And this is the version information: FreeSWITCH Version 1.0.head (git-e641045 2011-12-05 09-35-00 +0100) On Mon, Dec 5, 2011 at 10:38 PM, Dhairya Vora wrote: > Hi Brian, > Thanks for reply. Here are more details. > > *2011-12-05 22:30:31.541222 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [plivo_request_uuid]=[a3d8554c-1f62-11e1-b2be-4040af60d0d9] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [plivo_answer_url]=[http://dev.MMMMMM.net/namo/index.php] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [plivo_ring_url]=[http://dev.MMMMMM.net/namo/ring.php] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [plivo_hangup_url]=[http://dev.MMMMMM.net/namo/hangUp.php] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [origination_caller_id_number]=[0012018200770] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [plivo_from]=[0012018200770] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [plivo_to]=[00919426973436] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [plivo_app]=[true] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [absolute_codec_string]=[PCMU] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [originate_timeout]=[60] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [ignore_early_media]=[true] > 2011-12-05 22:30:31.541222 [ERR] mod_sofia.c:4279 Invalid URL > 2011-12-05 22:30:31.541222 [NOTICE] mod_sofia.c:4660 Close Channel N/A > [CS_NEW] > 2011-12-05 22:30:31.541222 [DEBUG] switch_core_state_machine.c:494 () > Running State Change CS_DESTROY > 2011-12-05 22:30:31.541222 [DEBUG] switch_core_state_machine.c:504 (N/A) > State DESTROY > 2011-12-05 22:30:31.541222 [DEBUG] mod_sofia.c:370 N/A SOFIA DESTROY > 2011-12-05 22:30:31.541222 [DEBUG] switch_core_state_machine.c:504 (N/A) > State DESTROY going to sleep > 2011-12-05 22:30:31.541222 [NOTICE] switch_ivr_originate.c:2459 Cannot > create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] > 2011-12-05 22:30:31.541222 [DEBUG] switch_ivr_originate.c:3367 Originate > Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]* > > I am placing the call using plivo. Please let me know if any other > information would be useful. > > Thanks, > Dhairya > > > On Mon, Dec 5, 2011 at 10:30 PM, Brian West wrote: > >> Can you give us more to go on? You seem to have cut out the info >> required to make any sort of recommendation.. how are you trying to place a >> call? Could the remote just not like your number format? >> >> /b >> >> On Dec 5, 2011, at 10:50 AM, Dhairya Vora wrote: >> >> Hi group, >> I am getting this error while generating outbound calls using freeswitch. >> *freeswitch at server> 2011-12-05 22:01:44.760933 [ERR] mod_sofia.c:4279 >> >> Invalid URL >> 2011-12-05 22:01:44.760933 [NOTICE] mod_sofia.c:4660 Close Channel N/A >> [CS_NEW] >> 2011-12-05 22:01:44.760933 [NOTICE] switch_ivr_originate.c:2459 Cannot >> create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] >> 2011-12-05 22:03:50.208776 [ERR] mod_sofia.c:4279 Invalid URL* >> >> >> Can anyone help me with this? >> >> -Dhairya >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/aabd49d9/attachment-0001.html From msc at freeswitch.org Mon Dec 5 20:41:56 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 5 Dec 2011 09:41:56 -0800 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error In-Reply-To: References: Message-ID: Still not enough information, although there is a clue in the line from mod_sofia that says invalid URL. You need to check the phone number you are dialing. What is that number? Also, use pastebin.freeswitch.org and use "FreeSWITCH Log" as the syntax highlighting as this is getting to be much too large to put into an email thread. -MC On Mon, Dec 5, 2011 at 9:08 AM, Dhairya Vora wrote: > Hi Brian, > Thanks for reply. Here are more details. > > *2011-12-05 22:30:31.541222 [DEBUG] switch_ivr_originate.c:1884 Parsing > global variables > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [plivo_request_uuid]=[a3d8554c-1f62-11e1-b2be-4040af60d0d9] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [plivo_answer_url]=[http://dev.MMMMMM.net/namo/index.php] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [plivo_ring_url]=[http://dev.MMMMMM.net/namo/ring.php] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [plivo_hangup_url]=[http://dev.MMMMMM.net/namo/hangUp.php] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [origination_caller_id_number]=[0012018200770] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [plivo_from]=[0012018200770] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [plivo_to]=[00919426973436] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [plivo_app]=[true] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [absolute_codec_string]=[PCMU] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [originate_timeout]=[60] > 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable > [ignore_early_media]=[true] > 2011-12-05 22:30:31.541222 [ERR] mod_sofia.c:4279 Invalid URL > 2011-12-05 22:30:31.541222 [NOTICE] mod_sofia.c:4660 Close Channel N/A > [CS_NEW] > 2011-12-05 22:30:31.541222 [DEBUG] switch_core_state_machine.c:494 () > Running State Change CS_DESTROY > 2011-12-05 22:30:31.541222 [DEBUG] switch_core_state_machine.c:504 (N/A) > State DESTROY > 2011-12-05 22:30:31.541222 [DEBUG] mod_sofia.c:370 N/A SOFIA DESTROY > 2011-12-05 22:30:31.541222 [DEBUG] switch_core_state_machine.c:504 (N/A) > State DESTROY going to sleep > 2011-12-05 22:30:31.541222 [NOTICE] switch_ivr_originate.c:2459 Cannot > create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] > 2011-12-05 22:30:31.541222 [DEBUG] switch_ivr_originate.c:3367 Originate > Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]* > > I am placing the call using plivo. Please let me know if any other > information would be useful. > > Thanks, > Dhairya > > > On Mon, Dec 5, 2011 at 10:30 PM, Brian West wrote: > >> Can you give us more to go on? You seem to have cut out the info >> required to make any sort of recommendation.. how are you trying to place a >> call? Could the remote just not like your number format? >> >> /b >> >> On Dec 5, 2011, at 10:50 AM, Dhairya Vora wrote: >> >> Hi group, >> I am getting this error while generating outbound calls using freeswitch. >> *freeswitch at server> 2011-12-05 22:01:44.760933 [ERR] mod_sofia.c:4279 >> >> Invalid URL >> 2011-12-05 22:01:44.760933 [NOTICE] mod_sofia.c:4660 Close Channel N/A >> [CS_NEW] >> 2011-12-05 22:01:44.760933 [NOTICE] switch_ivr_originate.c:2459 Cannot >> create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] >> 2011-12-05 22:03:50.208776 [ERR] mod_sofia.c:4279 Invalid URL* >> >> >> Can anyone help me with this? >> >> -Dhairya >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/df1c706a/attachment.html From ayobami at programmer.net Mon Dec 5 20:42:53 2011 From: ayobami at programmer.net (ayobami) Date: Mon, 5 Dec 2011 09:42:53 -0800 (PST) Subject: [Freeswitch-users] Softphones could not call each other but could register on the FS server In-Reply-To: References: <1322928441467-7058171.post@n2.nabble.com> Message-ID: <1323106973748-7063811.post@n2.nabble.com> thanks for your time, I think the problem was from the git head that I pulled, I have re-pulled the latest version and recompiled it, and the problem is now gone -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Softphones-could-not-call-each-other-but-could-register-on-the-FS-server-tp7058171p7063811.html Sent from the freeswitch-users mailing list archive at Nabble.com. From dhairya.blogs at gmail.com Mon Dec 5 20:50:08 2011 From: dhairya.blogs at gmail.com (Dhairya Vora) Date: Mon, 5 Dec 2011 23:20:08 +0530 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error In-Reply-To: References: Message-ID: Hi Michael, For a completely new call, I am pasting out from freeswitch.log to pastebin. http://pastebin.freeswitch.org/17938 is the link for it. Where can i found more information (this information is copied from /usr/local/freeswitch/log/freeswitch.log) Thanks, Dhairya On Mon, Dec 5, 2011 at 11:11 PM, Michael Collins wrote: > Still not enough information, although there is a clue in the line from > mod_sofia that says invalid URL. You need to check the phone number you are > dialing. What is that number? Also, use pastebin.freeswitch.org and use > "FreeSWITCH Log" as the syntax highlighting as this is getting to be much > too large to put into an email thread. > > -MC > > > On Mon, Dec 5, 2011 at 9:08 AM, Dhairya Vora wrote: > >> Hi Brian, >> Thanks for reply. Here are more details. >> >> *2011-12-05 22:30:31.541222 [DEBUG] switch_ivr_originate.c:1884 Parsing >> global variables >> 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable >> [plivo_request_uuid]=[a3d8554c-1f62-11e1-b2be-4040af60d0d9] >> 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable >> [plivo_answer_url]=[http://dev.MMMMMM.net/namo/index.php] >> 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable >> [plivo_ring_url]=[http://dev.MMMMMM.net/namo/ring.php] >> 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable >> [plivo_hangup_url]=[http://dev.MMMMMM.net/namo/hangUp.php] >> 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable >> [origination_caller_id_number]=[0012018200770] >> 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable >> [plivo_from]=[0012018200770] >> 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable >> [plivo_to]=[00919426973436] >> 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable >> [plivo_app]=[true] >> 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable >> [absolute_codec_string]=[PCMU] >> 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable >> [originate_timeout]=[60] >> 2011-12-05 22:30:31.541222 [DEBUG] switch_event.c:1521 Parsing variable >> [ignore_early_media]=[true] >> 2011-12-05 22:30:31.541222 [ERR] mod_sofia.c:4279 Invalid URL >> 2011-12-05 22:30:31.541222 [NOTICE] mod_sofia.c:4660 Close Channel N/A >> [CS_NEW] >> 2011-12-05 22:30:31.541222 [DEBUG] switch_core_state_machine.c:494 () >> Running State Change CS_DESTROY >> 2011-12-05 22:30:31.541222 [DEBUG] switch_core_state_machine.c:504 (N/A) >> State DESTROY >> 2011-12-05 22:30:31.541222 [DEBUG] mod_sofia.c:370 N/A SOFIA DESTROY >> 2011-12-05 22:30:31.541222 [DEBUG] switch_core_state_machine.c:504 (N/A) >> State DESTROY going to sleep >> 2011-12-05 22:30:31.541222 [NOTICE] switch_ivr_originate.c:2459 Cannot >> create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] >> 2011-12-05 22:30:31.541222 [DEBUG] switch_ivr_originate.c:3367 Originate >> Resulted in Error Cause: 28 [INVALID_NUMBER_FORMAT]* >> >> I am placing the call using plivo. Please let me know if any other >> information would be useful. >> >> Thanks, >> Dhairya >> >> >> On Mon, Dec 5, 2011 at 10:30 PM, Brian West wrote: >> >>> Can you give us more to go on? You seem to have cut out the info >>> required to make any sort of recommendation.. how are you trying to place a >>> call? Could the remote just not like your number format? >>> >>> /b >>> >>> On Dec 5, 2011, at 10:50 AM, Dhairya Vora wrote: >>> >>> Hi group, >>> I am getting this error while generating outbound calls using freeswitch. >>> *freeswitch at server> 2011-12-05 22:01:44.760933 [ERR] mod_sofia.c:4279 >>> >>> Invalid URL >>> 2011-12-05 22:01:44.760933 [NOTICE] mod_sofia.c:4660 Close Channel N/A >>> [CS_NEW] >>> 2011-12-05 22:01:44.760933 [NOTICE] switch_ivr_originate.c:2459 Cannot >>> create outgoing channel of type [sofia] cause: [INVALID_NUMBER_FORMAT] >>> 2011-12-05 22:03:50.208776 [ERR] mod_sofia.c:4279 Invalid URL* >>> >>> >>> Can anyone help me with this? >>> >>> -Dhairya >>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/729195af/attachment-0001.html From justlikeef at gmail.com Mon Dec 5 20:55:32 2011 From: justlikeef at gmail.com (Rob Hutton) Date: Mon, 5 Dec 2011 12:55:32 -0500 Subject: [Freeswitch-users] Created new profile for double-natted phones, calling these fails with USER_NOT_REGISTERED In-Reply-To: <4EDCC4E5.803@integrafin.co.uk> References: <4EDCB468.2060001@integrafin.co.uk> <9E6ADFB3-5BB6-4D45-8CF1-5C0DA06BB76D@gmail.com> <4EDCC4E5.803@integrafin.co.uk> Message-ID: <201112051255.33240.justlikeef@gmail.com> Alex - Check out the presense settings. Make sure both profiles are using the same DB: On Monday 05 December 2011 08:19:33 Alex Crow wrote: > On 05/12/11 13:04, Anthony Cosgrove wrote: > > Alex, > > > > Does FS *have* to be behind a NAT? Are you doing a straight 1:1 from public to private (FS side)? What kind of router/firewall are you using on the FS side and endpoint sides? You'll will probably want to lower the endpoint registration times from the normal default of 1 hour to 180 seconds (3 minutes) or even lower depending on router/firewall. You may also want to turn on keep-alive packets to keep the data flowing in/out. What you are running into is either one or both sides are closing off 5060/udp early, FS is marking the registration as dead. > > > > Anthony, > > No, it's definitely not that - the registration is alive and well. The > issue is that FS sees that the user is not registered on the internal > sip profile and doesn't check the doublenat profile. > > The router/firewall is iptables (shorewall) and I'm doing a DNAT of the > relevant ports to the FS box. > > What I have found is that I can specify the dialstring for the external > users in the directory entries, eg: > > value="{sip_invite_domain=${domain_name},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(doublenat/${dialed_user}@${dialed_domain})}"/> > > > > Best thing to do is to place FS in a DMZ or install a public facing SBC to take the registrations and forward them on to FS internally. > > > > Yes, it would probably be better that way, but this is really mostly for > internal use. > > Cheers > > Alex > > > > > Anthony > > > > On Dec 5, 2011, at 7:09 AM, Alex Crow wrote: > > > >> Hi all, > >> > >> I have set up a new profile for phones behind NAT connecting to an FS > >> box behind NAT. Calls in from a NAT'ed phone works fine, however calls > >> out to one fail with USER_NOT_REGISTERED. All the phones are set up in > >> /opt/freeswitch/conf/directory/default/. > >> > >> I have added the following in the new profile config hoping it would > >> resolve the issue but it does not: > >> > >> > >> > >> > >> > >> > >> Any clues? > >> > >> Thanks > >> > >> Alex > >> > >> _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/3769faf3/attachment.html From justlikeef at gmail.com Mon Dec 5 20:56:05 2011 From: justlikeef at gmail.com (Rob Hutton) Date: Mon, 5 Dec 2011 12:56:05 -0500 Subject: [Freeswitch-users] Created new profile for double-natted phones, calling these fails with USER_NOT_REGISTERED In-Reply-To: <201112051255.33240.justlikeef@gmail.com> References: <4EDCB468.2060001@integrafin.co.uk> <4EDCC4E5.803@integrafin.co.uk> <201112051255.33240.justlikeef@gmail.com> Message-ID: <201112051256.05883.justlikeef@gmail.com> http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Presence On Monday 05 December 2011 12:55:32 Rob Hutton wrote: > Alex - > > Check out the presense settings. Make sure both profiles are using the same DB: > > > > On Monday 05 December 2011 08:19:33 Alex Crow wrote: > > On 05/12/11 13:04, Anthony Cosgrove wrote: > > > Alex, > > > > > > Does FS *have* to be behind a NAT? Are you doing a straight 1:1 from public to private (FS side)? What kind of router/firewall are you using on the FS side and endpoint sides? You'll will probably want to lower the endpoint registration times from the normal default of 1 hour to 180 seconds (3 minutes) or even lower depending on router/firewall. You may also want to turn on keep-alive packets to keep the data flowing in/out. What you are running into is either one or both sides are closing off 5060/udp early, FS is marking the registration as dead. > > > > > > > Anthony, > > > > No, it's definitely not that - the registration is alive and well. The > > issue is that FS sees that the user is not registered on the internal > > sip profile and doesn't check the doublenat profile. > > > > The router/firewall is iptables (shorewall) and I'm doing a DNAT of the > > relevant ports to the FS box. > > > > What I have found is that I can specify the dialstring for the external > > users in the directory entries, eg: > > > > > value="{sip_invite_domain=${domain_name},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(doublenat/${dialed_user}@${dialed_domain})}"/> > > > > > > > Best thing to do is to place FS in a DMZ or install a public facing SBC to take the registrations and forward them on to FS internally. > > > > > > > Yes, it would probably be better that way, but this is really mostly for > > internal use. > > > > Cheers > > > > Alex > > > > > > > > Anthony > > > > > > On Dec 5, 2011, at 7:09 AM, Alex Crow wrote: > > > > > >> Hi all, > > >> > > >> I have set up a new profile for phones behind NAT connecting to an FS > > >> box behind NAT. Calls in from a NAT'ed phone works fine, however calls > > >> out to one fail with USER_NOT_REGISTERED. All the phones are set up in > > >> /opt/freeswitch/conf/directory/default/. > > >> > > >> I have added the following in the new profile config hoping it would > > >> resolve the issue but it does not: > > >> > > >> > > >> > > >> > > >> > > >> > > >> Any clues? > > >> > > >> Thanks > > >> > > >> Alex > > >> > > >> _________________________________________________________________________ > > >> Professional FreeSWITCH Consulting Services: > > >> consulting at freeswitch.org > > >> http://www.freeswitchsolutions.com > > >> > > >> > > >> > > >> > > >> Official FreeSWITCH Sites > > >> http://www.freeswitch.org > > >> http://wiki.freeswitch.org > > >> http://www.cluecon.com > > >> > > >> FreeSWITCH-users mailing list > > >> FreeSWITCH-users at lists.freeswitch.org > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > >> http://www.freeswitch.org > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/b2cab088/attachment-0001.html From sos at sokhapkin.dyndns.org Mon Dec 5 21:03:35 2011 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 5 Dec 2011 13:03:35 -0500 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error In-Reply-To: References: Message-ID: <201112051303.35723.sos@sokhapkin.dyndns.org> The log is incomplete. Paste log starting with "bridge" application execution. On Monday 05 December 2011, Dhairya Vora wrote: > Hi Michael, > > For a completely new call, I am pasting out from freeswitch.log to > pastebin. http://pastebin.freeswitch.org/17938 is the link for it. > > Where can i found more information (this information is copied from > /usr/local/freeswitch/log/freeswitch.log) > > Thanks, From dhairya.blogs at gmail.com Mon Dec 5 21:09:05 2011 From: dhairya.blogs at gmail.com (Dhairya Vora) Date: Mon, 5 Dec 2011 23:39:05 +0530 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error In-Reply-To: <201112051303.35723.sos@sokhapkin.dyndns.org> References: <201112051303.35723.sos@sokhapkin.dyndns.org> Message-ID: Hi Sergey, can you please tell me how to get such log? from which file? and what other settings are required to be done? Thanks, Dhairya On Mon, Dec 5, 2011 at 11:33 PM, Sergey Okhapkin wrote: > The log is incomplete. Paste log starting with "bridge" application > execution. > > On Monday 05 December 2011, Dhairya Vora wrote: > > Hi Michael, > > > > For a completely new call, I am pasting out from freeswitch.log to > > pastebin. http://pastebin.freeswitch.org/17938 is the link for it. > > > > Where can i found more information (this information is copied from > > /usr/local/freeswitch/log/freeswitch.log) > > > > 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/20111205/1b490a8c/attachment.html From sos at sokhapkin.dyndns.org Mon Dec 5 21:17:11 2011 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 5 Dec 2011 13:17:11 -0500 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error In-Reply-To: References: <201112051303.35723.sos@sokhapkin.dyndns.org> Message-ID: <201112051317.11965.sos@sokhapkin.dyndns.org> From the same file you're using, /usr/local/freeswitch/log/freeswitch.log On Monday 05 December 2011, Dhairya Vora wrote: > Hi Sergey, > > can you please tell me how to get such log? > from which file? and what other settings are required to be done? > > Thanks, > Dhairya > > On Mon, Dec 5, 2011 at 11:33 PM, Sergey Okhapkin > > wrote: > > The log is incomplete. Paste log starting with "bridge" application > > execution. > > > > On Monday 05 December 2011, Dhairya Vora wrote: > > > Hi Michael, > > > > > > For a completely new call, I am pasting out from freeswitch.log to > > > pastebin. http://pastebin.freeswitch.org/17938 is the link for it. > > > > > > Where can i found more information (this information is copied from > > > /usr/local/freeswitch/log/freeswitch.log) > > > > > > 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 acrow at integrafin.co.uk Mon Dec 5 21:43:02 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Mon, 05 Dec 2011 18:43:02 +0000 Subject: [Freeswitch-users] Created new profile for double-natted phones, calling these fails with USER_NOT_REGISTERED In-Reply-To: <201112051255.33240.justlikeef@gmail.com> References: <4EDCB468.2060001@integrafin.co.uk> <9E6ADFB3-5BB6-4D45-8CF1-5C0DA06BB76D@gmail.com> <4EDCC4E5.803@integrafin.co.uk> <201112051255.33240.justlikeef@gmail.com> Message-ID: <4EDD10B6.7060503@integrafin.co.uk> Rob, Thanks. Ah, OK. I thought presence was just for things like BLF and the like, but I will give it a go. Probably should be added to the wiki if this fixes it (which I'm sure it will). I'd rather not have to add special entries in my directory for external phones. Cheers Alex On 05/12/11 17:55, Rob Hutton wrote: > > Alex - > > > Check out the presense settings. Make sure both profiles are using the > same DB: > > > > > On Monday 05 December 2011 08:19:33 Alex Crow wrote: > > > On 05/12/11 13:04, Anthony Cosgrove wrote: > > > > Alex, > > > > > > > > Does FS *have* to be behind a NAT? Are you doing a straight 1:1 > from public to private (FS side)? What kind of router/firewall are you > using on the FS side and endpoint sides? You'll will probably want to > lower the endpoint registration times from the normal default of 1 > hour to 180 seconds (3 minutes) or even lower depending on > router/firewall. You may also want to turn on keep-alive packets to > keep the data flowing in/out. What you are running into is either one > or both sides are closing off 5060/udp early, FS is marking the > registration as dead. > > > > > > > > > > Anthony, > > > > > > No, it's definitely not that - the registration is alive and well. The > > > issue is that FS sees that the user is not registered on the internal > > > sip profile and doesn't check the doublenat profile. > > > > > > The router/firewall is iptables (shorewall) and I'm doing a DNAT of the > > > relevant ports to the FS box. > > > > > > What I have found is that I can specify the dialstring for the external > > > users in the directory entries, eg: > > > > > > > > > value="{sip_invite_domain=${domain_name},presence_id=${dialed_user}@${dialed_domain}}${sofia_contact(doublenat/${dialed_user}@${dialed_domain})}"/> > > > > > > > > > > Best thing to do is to place FS in a DMZ or install a public > facing SBC to take the registrations and forward them on to FS internally. > > > > > > > > > > Yes, it would probably be better that way, but this is really mostly > for > > > internal use. > > > > > > Cheers > > > > > > Alex > > > > > > > > > > > Anthony > > > > > > > > On Dec 5, 2011, at 7:09 AM, Alex Crow wrote: > > > > > > > >> Hi all, > > > >> > > > >> I have set up a new profile for phones behind NAT connecting to an FS > > > >> box behind NAT. Calls in from a NAT'ed phone works fine, however > calls > > > >> out to one fail with USER_NOT_REGISTERED. All the phones are set > up in > > > >> /opt/freeswitch/conf/directory/default/. > > > >> > > > >> I have added the following in the new profile config hoping it would > > > >> resolve the issue but it does not: > > > >> > > > >> > > > >> > > > >> > > > >> > > > >> > > > >> Any clues? > > > >> > > > >> Thanks > > > >> > > > >> Alex > > > >> > > > >> > _________________________________________________________________________ > > > >> Professional FreeSWITCH Consulting Services: > > > >> consulting at freeswitch.org > > > >> http://www.freeswitchsolutions.com > > > >> > > > >> > > > >> > > > >> > > > >> Official FreeSWITCH Sites > > > >> http://www.freeswitch.org > > > >> http://wiki.freeswitch.org > > > >> http://www.cluecon.com > > > >> > > > >> FreeSWITCH-users mailing list > > > >> FreeSWITCH-users at lists.freeswitch.org > > > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > >> http://www.freeswitch.org > > > > > > > > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > > > consulting at freeswitch.org > > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > > http://www.freeswitch.org > > > > http://wiki.freeswitch.org > > > > http://www.cluecon.com > > > > > > > > FreeSWITCH-users mailing list > > > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > > http://www.freeswitch.org > > > > > > > > > > > > > > > > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/a9d6d2db/attachment-0001.html From dhairya.blogs at gmail.com Mon Dec 5 22:12:19 2011 From: dhairya.blogs at gmail.com (Dhairya Vora) Date: Tue, 6 Dec 2011 00:42:19 +0530 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error In-Reply-To: <201112051317.11965.sos@sokhapkin.dyndns.org> References: <201112051303.35723.sos@sokhapkin.dyndns.org> <201112051317.11965.sos@sokhapkin.dyndns.org> Message-ID: Hi everyone, freeswitch.log really doesn't show anything else. Please see http://pastebin.freeswitch.org/17940 . Please notice that first line is for the previous call and new call is started from second line. I am really thankful for your help but can't find what you are looking for. Sorry for inconvenience and thanks for replies, Dhairya On Mon, Dec 5, 2011 at 11:47 PM, Sergey Okhapkin wrote: > >From the same file you're using, /usr/local/freeswitch/log/freeswitch.log > > On Monday 05 December 2011, Dhairya Vora wrote: > > Hi Sergey, > > > > can you please tell me how to get such log? > > from which file? and what other settings are required to be done? > > > > Thanks, > > Dhairya > > > > On Mon, Dec 5, 2011 at 11:33 PM, Sergey Okhapkin > > > > wrote: > > > The log is incomplete. Paste log starting with "bridge" application > > > execution. > > > > > > On Monday 05 December 2011, Dhairya Vora wrote: > > > > Hi Michael, > > > > > > > > For a completely new call, I am pasting out from freeswitch.log to > > > > pastebin. http://pastebin.freeswitch.org/17938 is the link for it. > > > > > > > > Where can i found more information (this information is copied from > > > > /usr/local/freeswitch/log/freeswitch.log) > > > > > > > > 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/20111206/ac187850/attachment.html From sos at sokhapkin.dyndns.org Mon Dec 5 22:22:17 2011 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 5 Dec 2011 14:22:17 -0500 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error In-Reply-To: References: <201112051317.11965.sos@sokhapkin.dyndns.org> Message-ID: <201112051422.17815.sos@sokhapkin.dyndns.org> How call origination is called, from dialplan or from ESL? You have to provide more information. On Monday 05 December 2011, Dhairya Vora wrote: > Hi everyone, > > freeswitch.log really doesn't show anything else. > Please see http://pastebin.freeswitch.org/17940 . Please notice that first > line is for the previous call and new call is started from second line. > I am really thankful for your help but can't find what you are looking for. > > Sorry for inconvenience and thanks for replies, > Dhairya > > On Mon, Dec 5, 2011 at 11:47 PM, Sergey Okhapkin > > wrote: > > >From the same file you're using, > > >/usr/local/freeswitch/log/freeswitch.log > > > > On Monday 05 December 2011, Dhairya Vora wrote: > > > Hi Sergey, > > > > > > can you please tell me how to get such log? > > > from which file? and what other settings are required to be done? > > > > > > Thanks, > > > Dhairya > > > > > > On Mon, Dec 5, 2011 at 11:33 PM, Sergey Okhapkin > > > > > > wrote: > > > > The log is incomplete. Paste log starting with "bridge" application > > > > execution. > > > > > > > > On Monday 05 December 2011, Dhairya Vora wrote: > > > > > Hi Michael, > > > > > > > > > > For a completely new call, I am pasting out from freeswitch.log to > > > > > pastebin. http://pastebin.freeswitch.org/17938 is the link for it. > > > > > > > > > > Where can i found more information (this information is copied from > > > > > /usr/local/freeswitch/log/freeswitch.log) > > > > > > > > > > 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 dhairya.blogs at gmail.com Mon Dec 5 22:44:21 2011 From: dhairya.blogs at gmail.com (Dhairya Vora) Date: Tue, 6 Dec 2011 01:14:21 +0530 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error In-Reply-To: <201112051422.17815.sos@sokhapkin.dyndns.org> References: <201112051317.11965.sos@sokhapkin.dyndns.org> <201112051422.17815.sos@sokhapkin.dyndns.org> Message-ID: Hi Sergey, I am using Plivo to generate calls. So, I can say my calls are generated using ESL. Does it mean that there is some issue with plivo? Thanks, -Dhairya On Tue, Dec 6, 2011 at 12:52 AM, Sergey Okhapkin wrote: > How call origination is called, from dialplan or from ESL? You have to > provide > more information. > > On Monday 05 December 2011, Dhairya Vora wrote: > > Hi everyone, > > > > freeswitch.log really doesn't show anything else. > > Please see http://pastebin.freeswitch.org/17940 . Please notice that > first > > line is for the previous call and new call is started from second line. > > I am really thankful for your help but can't find what you are looking > for. > > > > Sorry for inconvenience and thanks for replies, > > Dhairya > > > > On Mon, Dec 5, 2011 at 11:47 PM, Sergey Okhapkin > > > > wrote: > > > >From the same file you're using, > > > >/usr/local/freeswitch/log/freeswitch.log > > > > > > On Monday 05 December 2011, Dhairya Vora wrote: > > > > Hi Sergey, > > > > > > > > can you please tell me how to get such log? > > > > from which file? and what other settings are required to be done? > > > > > > > > Thanks, > > > > Dhairya > > > > > > > > On Mon, Dec 5, 2011 at 11:33 PM, Sergey Okhapkin > > > > > > > > wrote: > > > > > The log is incomplete. Paste log starting with "bridge" application > > > > > execution. > > > > > > > > > > On Monday 05 December 2011, Dhairya Vora wrote: > > > > > > Hi Michael, > > > > > > > > > > > > For a completely new call, I am pasting out from freeswitch.log > to > > > > > > pastebin. http://pastebin.freeswitch.org/17938 is the link for > it. > > > > > > > > > > > > Where can i found more information (this information is copied > from > > > > > > /usr/local/freeswitch/log/freeswitch.log) > > > > > > > > > > > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/73d8995b/attachment.html From sos at sokhapkin.dyndns.org Mon Dec 5 22:54:14 2011 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 5 Dec 2011 14:54:14 -0500 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error In-Reply-To: References: <201112051422.17815.sos@sokhapkin.dyndns.org> Message-ID: <201112051454.14507.sos@sokhapkin.dyndns.org> Most likely plivo issue, looks like it issues originate command with wrong arguments. This is not FS issue. What is plivo, BTW? On Monday 05 December 2011, Dhairya Vora wrote: > Hi Sergey, > > I am using Plivo to generate calls. > So, I can say my calls are generated using ESL. > Does it mean that there is some issue with plivo? > > Thanks, > -Dhairya > > On Tue, Dec 6, 2011 at 12:52 AM, Sergey Okhapkin > > wrote: > > How call origination is called, from dialplan or from ESL? You have to > > provide > > more information. > > > > On Monday 05 December 2011, Dhairya Vora wrote: > > > Hi everyone, > > > > > > freeswitch.log really doesn't show anything else. > > > Please see http://pastebin.freeswitch.org/17940 . Please notice that > > > > first > > > > > line is for the previous call and new call is started from second line. > > > I am really thankful for your help but can't find what you are looking > > > > for. > > > > > Sorry for inconvenience and thanks for replies, > > > Dhairya > > > > > > On Mon, Dec 5, 2011 at 11:47 PM, Sergey Okhapkin > > > > > > wrote: > > > > >From the same file you're using, > > > > >/usr/local/freeswitch/log/freeswitch.log > > > > > > > > On Monday 05 December 2011, Dhairya Vora wrote: > > > > > Hi Sergey, > > > > > > > > > > can you please tell me how to get such log? > > > > > from which file? and what other settings are required to be done? > > > > > > > > > > Thanks, > > > > > Dhairya > > > > > > > > > > On Mon, Dec 5, 2011 at 11:33 PM, Sergey Okhapkin > > > > > > > > > > wrote: > > > > > > The log is incomplete. Paste log starting with "bridge" > > > > > > application execution. > > > > > > > > > > > > On Monday 05 December 2011, Dhairya Vora wrote: > > > > > > > Hi Michael, > > > > > > > > > > > > > > For a completely new call, I am pasting out from freeswitch.log > > > > to > > > > > > > > > pastebin. http://pastebin.freeswitch.org/17938 is the link for > > > > it. > > > > > > > > > Where can i found more information (this information is copied > > > > from > > > > > > > > > /usr/local/freeswitch/log/freeswitch.log) > > > > > > > > > > > > > > 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 From dhairya.blogs at gmail.com Mon Dec 5 22:56:28 2011 From: dhairya.blogs at gmail.com (Dhairya Vora) Date: Tue, 6 Dec 2011 01:26:28 +0530 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error In-Reply-To: <201112051454.14507.sos@sokhapkin.dyndns.org> References: <201112051422.17815.sos@sokhapkin.dyndns.org> <201112051454.14507.sos@sokhapkin.dyndns.org> Message-ID: Hi Sergey, Plivo is a framework on top of freeswitch to generate outbound calls and receive inbound calls. With installed freeswitch and plivo, inbound calls are working but outbound calls are not working. Thanks, -Dhairya On Tue, Dec 6, 2011 at 1:24 AM, Sergey Okhapkin wrote: > Most likely plivo issue, looks like it issues originate command with wrong > arguments. This is not FS issue. > > What is plivo, BTW? > > On Monday 05 December 2011, Dhairya Vora wrote: > > Hi Sergey, > > > > I am using Plivo to generate calls. > > So, I can say my calls are generated using ESL. > > Does it mean that there is some issue with plivo? > > > > Thanks, > > -Dhairya > > > > On Tue, Dec 6, 2011 at 12:52 AM, Sergey Okhapkin > > > > wrote: > > > How call origination is called, from dialplan or from ESL? You have to > > > provide > > > more information. > > > > > > On Monday 05 December 2011, Dhairya Vora wrote: > > > > Hi everyone, > > > > > > > > freeswitch.log really doesn't show anything else. > > > > Please see http://pastebin.freeswitch.org/17940 . Please notice that > > > > > > first > > > > > > > line is for the previous call and new call is started from second > line. > > > > I am really thankful for your help but can't find what you are > looking > > > > > > for. > > > > > > > Sorry for inconvenience and thanks for replies, > > > > Dhairya > > > > > > > > On Mon, Dec 5, 2011 at 11:47 PM, Sergey Okhapkin > > > > > > > > wrote: > > > > > >From the same file you're using, > > > > > >/usr/local/freeswitch/log/freeswitch.log > > > > > > > > > > On Monday 05 December 2011, Dhairya Vora wrote: > > > > > > Hi Sergey, > > > > > > > > > > > > can you please tell me how to get such log? > > > > > > from which file? and what other settings are required to be done? > > > > > > > > > > > > Thanks, > > > > > > Dhairya > > > > > > > > > > > > On Mon, Dec 5, 2011 at 11:33 PM, Sergey Okhapkin > > > > > > > > > > > > wrote: > > > > > > > The log is incomplete. Paste log starting with "bridge" > > > > > > > application execution. > > > > > > > > > > > > > > On Monday 05 December 2011, Dhairya Vora wrote: > > > > > > > > Hi Michael, > > > > > > > > > > > > > > > > For a completely new call, I am pasting out from > freeswitch.log > > > > > > to > > > > > > > > > > > pastebin. http://pastebin.freeswitch.org/17938 is the link > for > > > > > > it. > > > > > > > > > > > Where can i found more information (this information is > copied > > > > > > from > > > > > > > > > > > /usr/local/freeswitch/log/freeswitch.log) > > > > > > > > > > > > > > > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/3a60f2a4/attachment.html From msc at freeswitch.org Mon Dec 5 22:57:29 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 5 Dec 2011 11:57:29 -0800 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error In-Reply-To: <201112051454.14507.sos@sokhapkin.dyndns.org> References: <201112051422.17815.sos@sokhapkin.dyndns.org> <201112051454.14507.sos@sokhapkin.dyndns.org> Message-ID: See plivo.org for more information there. In this case I would do whatever troubleshooting steps the plivo guys suggest. -MC On Mon, Dec 5, 2011 at 11:54 AM, Sergey Okhapkin wrote: > Most likely plivo issue, looks like it issues originate command with wrong > arguments. This is not FS issue. > > What is plivo, BTW? > > On Monday 05 December 2011, Dhairya Vora wrote: > > Hi Sergey, > > > > I am using Plivo to generate calls. > > So, I can say my calls are generated using ESL. > > Does it mean that there is some issue with plivo? > > > > Thanks, > > -Dhairya > > > > On Tue, Dec 6, 2011 at 12:52 AM, Sergey Okhapkin > > > > wrote: > > > How call origination is called, from dialplan or from ESL? You have to > > > provide > > > more information. > > > > > > On Monday 05 December 2011, Dhairya Vora wrote: > > > > Hi everyone, > > > > > > > > freeswitch.log really doesn't show anything else. > > > > Please see http://pastebin.freeswitch.org/17940 . Please notice that > > > > > > first > > > > > > > line is for the previous call and new call is started from second > line. > > > > I am really thankful for your help but can't find what you are > looking > > > > > > for. > > > > > > > Sorry for inconvenience and thanks for replies, > > > > Dhairya > > > > > > > > On Mon, Dec 5, 2011 at 11:47 PM, Sergey Okhapkin > > > > > > > > wrote: > > > > > >From the same file you're using, > > > > > >/usr/local/freeswitch/log/freeswitch.log > > > > > > > > > > On Monday 05 December 2011, Dhairya Vora wrote: > > > > > > Hi Sergey, > > > > > > > > > > > > can you please tell me how to get such log? > > > > > > from which file? and what other settings are required to be done? > > > > > > > > > > > > Thanks, > > > > > > Dhairya > > > > > > > > > > > > On Mon, Dec 5, 2011 at 11:33 PM, Sergey Okhapkin > > > > > > > > > > > > wrote: > > > > > > > The log is incomplete. Paste log starting with "bridge" > > > > > > > application execution. > > > > > > > > > > > > > > On Monday 05 December 2011, Dhairya Vora wrote: > > > > > > > > Hi Michael, > > > > > > > > > > > > > > > > For a completely new call, I am pasting out from > freeswitch.log > > > > > > to > > > > > > > > > > > pastebin. http://pastebin.freeswitch.org/17938 is the link > for > > > > > > it. > > > > > > > > > > > Where can i found more information (this information is > copied > > > > > > from > > > > > > > > > > > /usr/local/freeswitch/log/freeswitch.log) > > > > > > > > > > > > > > > > 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 > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/2adb2f86/attachment-0001.html From dhairya.blogs at gmail.com Mon Dec 5 23:00:59 2011 From: dhairya.blogs at gmail.com (Dhairya Vora) Date: Tue, 6 Dec 2011 01:30:59 +0530 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error In-Reply-To: References: <201112051422.17815.sos@sokhapkin.dyndns.org> <201112051454.14507.sos@sokhapkin.dyndns.org> Message-ID: Thanks MC. Will do it. -Dhairya On Tue, Dec 6, 2011 at 1:27 AM, Michael Collins wrote: > See plivo.org for more information there. In this case I would do > whatever troubleshooting steps the plivo guys suggest. > > -MC > > > On Mon, Dec 5, 2011 at 11:54 AM, Sergey Okhapkin > wrote: > >> Most likely plivo issue, looks like it issues originate command with wrong >> arguments. This is not FS issue. >> >> What is plivo, BTW? >> >> On Monday 05 December 2011, Dhairya Vora wrote: >> > Hi Sergey, >> > >> > I am using Plivo to generate calls. >> > So, I can say my calls are generated using ESL. >> > Does it mean that there is some issue with plivo? >> > >> > Thanks, >> > -Dhairya >> > >> > On Tue, Dec 6, 2011 at 12:52 AM, Sergey Okhapkin >> > >> > wrote: >> > > How call origination is called, from dialplan or from ESL? You have to >> > > provide >> > > more information. >> > > >> > > On Monday 05 December 2011, Dhairya Vora wrote: >> > > > Hi everyone, >> > > > >> > > > freeswitch.log really doesn't show anything else. >> > > > Please see http://pastebin.freeswitch.org/17940 . Please notice >> that >> > > >> > > first >> > > >> > > > line is for the previous call and new call is started from second >> line. >> > > > I am really thankful for your help but can't find what you are >> looking >> > > >> > > for. >> > > >> > > > Sorry for inconvenience and thanks for replies, >> > > > Dhairya >> > > > >> > > > On Mon, Dec 5, 2011 at 11:47 PM, Sergey Okhapkin >> > > > >> > > > wrote: >> > > > > >From the same file you're using, >> > > > > >/usr/local/freeswitch/log/freeswitch.log >> > > > > >> > > > > On Monday 05 December 2011, Dhairya Vora wrote: >> > > > > > Hi Sergey, >> > > > > > >> > > > > > can you please tell me how to get such log? >> > > > > > from which file? and what other settings are required to be >> done? >> > > > > > >> > > > > > Thanks, >> > > > > > Dhairya >> > > > > > >> > > > > > On Mon, Dec 5, 2011 at 11:33 PM, Sergey Okhapkin >> > > > > > >> > > > > > wrote: >> > > > > > > The log is incomplete. Paste log starting with "bridge" >> > > > > > > application execution. >> > > > > > > >> > > > > > > On Monday 05 December 2011, Dhairya Vora wrote: >> > > > > > > > Hi Michael, >> > > > > > > > >> > > > > > > > For a completely new call, I am pasting out from >> freeswitch.log >> > > >> > > to >> > > >> > > > > > > > pastebin. http://pastebin.freeswitch.org/17938 is the link >> for >> > > >> > > it. >> > > >> > > > > > > > Where can i found more information (this information is >> copied >> > > >> > > from >> > > >> > > > > > > > /usr/local/freeswitch/log/freeswitch.log) >> > > > > > > > >> > > > > > > > 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 >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/c59eb04a/attachment.html From sos at sokhapkin.dyndns.org Mon Dec 5 23:07:48 2011 From: sos at sokhapkin.dyndns.org (Sergey Okhapkin) Date: Mon, 5 Dec 2011 15:07:48 -0500 Subject: [Freeswitch-users] INVALID_NUMBER_FORMAT error In-Reply-To: References: <201112051454.14507.sos@sokhapkin.dyndns.org> Message-ID: <201112051507.48688.sos@sokhapkin.dyndns.org> It would be nice to log dialstring to debug log from ESL originate, BTW. Will simplify troubleshooting. On Monday 05 December 2011, Michael Collins wrote: > See plivo.org for more information there. In this case I would do whatever > troubleshooting steps the plivo guys suggest. > > -MC > > On Mon, Dec 5, 2011 at 11:54 AM, Sergey Okhapkin > > wrote: > > Most likely plivo issue, looks like it issues originate command with > > wrong arguments. This is not FS issue. > > > > What is plivo, BTW? > > > > On Monday 05 December 2011, Dhairya Vora wrote: > > > Hi Sergey, > > > > > > I am using Plivo to generate calls. > > > So, I can say my calls are generated using ESL. > > > Does it mean that there is some issue with plivo? > > > > > > Thanks, > > > -Dhairya > > > > > > On Tue, Dec 6, 2011 at 12:52 AM, Sergey Okhapkin > > > > > > wrote: > > > > How call origination is called, from dialplan or from ESL? You have > > > > to provide > > > > more information. > > > > > > > > On Monday 05 December 2011, Dhairya Vora wrote: > > > > > Hi everyone, > > > > > > > > > > freeswitch.log really doesn't show anything else. > > > > > Please see http://pastebin.freeswitch.org/17940 . Please notice > > > > > that > > > > > > > > first > > > > > > > > > line is for the previous call and new call is started from second > > > > line. > > > > > > > I am really thankful for your help but can't find what you are > > > > looking > > > > > > for. > > > > > > > > > Sorry for inconvenience and thanks for replies, > > > > > Dhairya > > > > > > > > > > On Mon, Dec 5, 2011 at 11:47 PM, Sergey Okhapkin > > > > > > > > > > wrote: > > > > > > >From the same file you're using, > > > > > > >/usr/local/freeswitch/log/freeswitch.log > > > > > > > > > > > > On Monday 05 December 2011, Dhairya Vora wrote: > > > > > > > Hi Sergey, > > > > > > > > > > > > > > can you please tell me how to get such log? > > > > > > > from which file? and what other settings are required to be > > > > > > > done? > > > > > > > > > > > > > > Thanks, > > > > > > > Dhairya > > > > > > > > > > > > > > On Mon, Dec 5, 2011 at 11:33 PM, Sergey Okhapkin > > > > > > > > > > > > > > wrote: > > > > > > > > The log is incomplete. Paste log starting with "bridge" > > > > > > > > application execution. > > > > > > > > > > > > > > > > On Monday 05 December 2011, Dhairya Vora wrote: > > > > > > > > > Hi Michael, > > > > > > > > > > > > > > > > > > For a completely new call, I am pasting out from > > > > freeswitch.log > > > > > > to > > > > > > > > > > > > > pastebin. http://pastebin.freeswitch.org/17938 is the link > > > > for > > > > > > it. > > > > > > > > > > > > > Where can i found more information (this information is > > > > copied > > > > > > from > > > > > > > > > > > > > /usr/local/freeswitch/log/freeswitch.log) > > > > > > > > > > > > > > > > > > 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 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org From jerry.richards at teotech.com Mon Dec 5 23:25:45 2011 From: jerry.richards at teotech.com (Jerry Richards) Date: Mon, 5 Dec 2011 12:25:45 -0800 Subject: [Freeswitch-users] In Bypass Mode Freeswitch Changes SDP? In-Reply-To: References: <2BF7FB90DF25EA4485949F3AF2B9D69633169C807C@VA3DIAXVS351.RED001.local> Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D696343AC748F4@VA3DIAXVS351.RED001.local> I confirmed this is not a problem with the latest GIT. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, November 23, 2011 11:22 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] In Bypass Mode Freeswitch Changes SDP? Do you have a trace of this taking place on latest GIT? I tested this and it appears to not touch the codecs.... On Wed, Nov 23, 2011 at 1:08 PM, Jerry Richards > wrote: Hello, I noticed an interop issue when using a Bria softphone and inbound-bypass-media is true in the sip_profile. When answering an inbound call, the Bria softphone's SDP Answer includes the list of codecs it supports (audio and video). I noticed that, even in bypass media mode, Freeswitch filters out all of the codecs in the list after the first one. I think this is not right. According to RFC 3264 (Section 7 Offerer Processing of the Answer), it is valid to provide a list of codecs in an SDP Answer and the endpoint SHOULD use the first one (but it might not). Why is Freeswitch altering the SDP in the 200 OK? I think it should send the SDP unmodified. Thanks, Jerry _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://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/20111205/8adcb500/attachment-0001.html From ajohnston at blimessaging.com Mon Dec 5 23:06:21 2011 From: ajohnston at blimessaging.com (Adam Johnston) Date: Mon, 5 Dec 2011 15:06:21 -0500 Subject: [Freeswitch-users] myevents json and xml arguments throwing errors Message-ID: Hi everyone, I've been trying to use the myevents command to subscribe to the events for an api call, but I've been having some trouble. When I call: myevents from my app (or even from a telnet client), I receive back all the event data just fine and the socket closes once the call has completed. However, if I pass a format argument, i.e: myevents json I receive a response of: ERR invalid uuid My uuids are definitely valid. The command breaks for all format arguments too (xml, json and plain). Has anyone encountered this problem before? Documentation for myevents is a little light, and web searches so far have been fruitless. Thanks, Adam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/f84eb60b/attachment.html From anthony.minessale at gmail.com Tue Dec 6 00:03:17 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 5 Dec 2011 15:03:17 -0600 Subject: [Freeswitch-users] myevents json and xml arguments throwing errors In-Reply-To: References: Message-ID: Did you try it reversed? On Dec 5, 2011 2:50 PM, "Adam Johnston" wrote: > Hi everyone, > > I've been trying to use the myevents command to subscribe to the events > for an api call, but I've been having some trouble. When I call: > myevents > from my app (or even from a telnet client), I receive back all the event > data just fine and the socket closes once the call has completed. However, > if I pass a format argument, i.e: > myevents json > I receive a response of: > ERR invalid uuid > > My uuids are definitely valid. The command breaks for all format arguments > too (xml, json and plain). Has anyone encountered this problem before? > Documentation for myevents is a little light, and web searches so far > have been fruitless. > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/84347661/attachment.html From ajohnston at blimessaging.com Tue Dec 6 00:33:55 2011 From: ajohnston at blimessaging.com (Adam Johnston) Date: Mon, 5 Dec 2011 16:33:55 -0500 Subject: [Freeswitch-users] myevents json and xml arguments throwing errors In-Reply-To: References: Message-ID: I have tried with syntax reversed, (myevents json and myevents xml) with the same result: ERR invalid uuid On Mon, Dec 5, 2011 at 4:03 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Did you try it reversed? > On Dec 5, 2011 2:50 PM, "Adam Johnston" > wrote: > >> Hi everyone, >> >> I've been trying to use the myevents command to subscribe to the events >> for an api call, but I've been having some trouble. When I call: >> myevents >> from my app (or even from a telnet client), I receive back all the event >> data just fine and the socket closes once the call has completed. However, >> if I pass a format argument, i.e: >> myevents json >> I receive a response of: >> ERR invalid uuid >> >> My uuids are definitely valid. The command breaks for all format >> arguments too (xml, json and plain). Has anyone encountered this problem >> before? Documentation for myevents is a little light, and web searches >> so far have been fruitless. >> >> 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 >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/ea71a742/attachment.html From jort.bloem at btg.co.nz Tue Dec 6 01:02:29 2011 From: jort.bloem at btg.co.nz (Jort Bloem) Date: Tue, 06 Dec 2011 11:02:29 +1300 Subject: [Freeswitch-users] Identify the gateway Message-ID: <4EDD3F75.9080303@btg.co.nz> Hi all, I have a setup with several gateways coming in on one profile. The Realms, unfortunately, are by DNS, and move from time to time. Is there any way to identify, in the dialplan, which gateway the call comes from, for incoming calls? I know I can get the ip address, but the ip address is subject to change. I know that the trunk provider can set various things, but I don't have control of the far end of the trunk. If there is a way to set variables on a per-gateway (NOT per-profile) basis, that would be perfect. "RTFM" gladly accepted with a hint about which part of which FM to R. -- Jaybee- From msc at freeswitch.org Tue Dec 6 01:45:27 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 5 Dec 2011 14:45:27 -0800 Subject: [Freeswitch-users] Identify the gateway In-Reply-To: <4EDD3F75.9080303@btg.co.nz> References: <4EDD3F75.9080303@btg.co.nz> Message-ID: Try adding one of these to the dialplan that handles inbound calls: Then, watch the console while an inbound comes in and you'll see quite a lot of information. Anything in that info dump is available to you for "figuring out" what's going on. If you have any questions then drop that info output to a pastebin and give us the URL here. -MC On Mon, Dec 5, 2011 at 2:02 PM, Jort Bloem wrote: > Hi all, > > I have a setup with several gateways coming in on one profile. The > Realms, unfortunately, are by DNS, and move from time to time. > > Is there any way to identify, in the dialplan, which gateway the call > comes from, for incoming calls? > > I know I can get the ip address, but the ip address is subject to change. > > I know that the trunk provider can set various things, but I don't have > control of the far end of the trunk. > > If there is a way to set variables on a per-gateway (NOT per-profile) > basis, that would be perfect. > > "RTFM" gladly accepted with a hint about which part of which FM to R. > > -- > Jaybee- > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/3a405b2d/attachment.html From avi at avimarcus.net Tue Dec 6 01:57:46 2011 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 6 Dec 2011 00:57:46 +0200 Subject: [Freeswitch-users] Identify the gateway In-Reply-To: References: <4EDD3F75.9080303@btg.co.nz> Message-ID: http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055508.html Hmm, gateways on the wiki should be cleaned up.. I couldn't find it on the wiki, only via google. -Avi On Tue, Dec 6, 2011 at 12:45 AM, Michael Collins wrote: > Try adding one of these to the dialplan that handles inbound calls: > > > > Then, watch the console while an inbound comes in and you'll see quite a > lot of information. Anything in that info dump is available to you for > "figuring out" what's going on. If you have any questions then drop that > info output to a pastebin and give us the URL here. > > -MC > > > On Mon, Dec 5, 2011 at 2:02 PM, Jort Bloem wrote: > >> Hi all, >> >> I have a setup with several gateways coming in on one profile. The >> Realms, unfortunately, are by DNS, and move from time to time. >> >> Is there any way to identify, in the dialplan, which gateway the call >> comes from, for incoming calls? >> >> I know I can get the ip address, but the ip address is subject to change. >> >> I know that the trunk provider can set various things, but I don't have >> control of the far end of the trunk. >> >> If there is a way to set variables on a per-gateway (NOT per-profile) >> basis, that would be perfect. >> >> "RTFM" gladly accepted with a hint about which part of which FM to R. >> >> -- >> Jaybee- >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/d8b54a93/attachment-0001.html From jort.bloem at btg.co.nz Tue Dec 6 03:26:39 2011 From: jort.bloem at btg.co.nz (Jort Bloem) Date: Tue, 06 Dec 2011 13:26:39 +1300 Subject: [Freeswitch-users] Identify the gateway In-Reply-To: References: <4EDD3F75.9080303@btg.co.nz> Message-ID: <4EDD613F.7060609@btg.co.nz> Hi Avi & Michael, Michael, I'm familiar with the info application, and have been using this to test my progress. It's gotten me a long way, but not past this hurdle. Avi, I tried that, or at least my interpretation of that, and it didn't work. I found the gateway in question - it's one of only two references to the trunk - in sip_profiles/. The declaration, including my addition of the variables clause, looks like this: I then called in via that interface, using a DDI, with the info application. I got lots of information, but neither foo nor bar. I did change the direction from "inbound" to "both", to make it as likely as possible to work. The log output I got had lots of stuff in it (see http://pastebin.freeswitch.org/17943 ) but no foo, no bar, and no trunk_1. It did have several occurrences of the ip address of talk2, 10.10.9.1, but that can change. I also see several references to the ip address (10.10.10.149) and sofia profile name (sipinterface_3), but all gateways will be on the same profile. Any further help would be appreciated. Jaybee- On 06/12/11 11:57, Avi Marcus wrote: > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055508.html > > Hmm, gateways on the wiki should be cleaned up.. I couldn't find it on > the wiki, only via google. > > -Avi > > > On Tue, Dec 6, 2011 at 12:45 AM, Michael Collins > wrote: > > Try adding one of these to the dialplan that handles inbound calls: > > > > Then, watch the console while an inbound comes in and you'll see > quite a lot of information. Anything in that info dump is > available to you for "figuring out" what's going on. If you have > any questions then drop that info output to a pastebin and give us > the URL here. > > -MC > > > On Mon, Dec 5, 2011 at 2:02 PM, Jort Bloem > wrote: > > Hi all, > > I have a setup with several gateways coming in on one profile. The > Realms, unfortunately, are by DNS, and move from time to time. > > Is there any way to identify, in the dialplan, which gateway > the call > comes from, for incoming calls? > > I know I can get the ip address, but the ip address is subject > to change. > > I know that the trunk provider can set various things, but I > don't have > control of the far end of the trunk. > > If there is a way to set variables on a per-gateway (NOT > per-profile) > basis, that would be perfect. > > "RTFM" gladly accepted with a hint about which part of which > FM to R. > > -- > Jaybee- > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Jort Bloem Technical Engineer - Auckland Business Technology Group P: +64 9 5801374 x 9884 Sent from Mozilla Thunderbird -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/5fe52151/attachment.html From admin at blindi.net Tue Dec 6 06:36:05 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Tue, 6 Dec 2011 04:36:05 +0100 (CET) Subject: [Freeswitch-users] lua error in session:transfer In-Reply-To: References: Message-ID: Hi all, i have a luascript. I hear the prompt, but the transfer fails. My script is: session:answer(); freeswitch.console_log("info", "Announce\n"); announce='/usr/local/freeswitch/sounds/callscreen/callscrrenopts.wav'; number = session:read(1, 1, announce, 5000, "#"); freeswitch.consoleLog("info", "Got number: ".. number .. "\n"); if number == argv[1] then target_number=argv[2]; elseif number == argv[3] then target_number=argv[4]; elseif number == argv[5] then target_number=argv[6]; elseif number == argv[7] then target_number=argv[8]; else end freeswitch.consoleLog("info", "Transfer to target number: ".. target_number .. " XML default\n"); session:transfer("target_number", "XML", "default"); in my dialplan reads: Then i press 1 for example, i become on the cli: [ERR] mod_lua.cpp:196 /usr/local/freeswitch/scripts/thomas.lua:16: attempt to concatenate global 'target_number' (a nil value)M What is wrong please? Thanks for your help. --------------- Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: http://www.blindi.net/callback homepage: http://www.blindi.net blinde-misc mailingliste f?r blinde. anmeldung unter: http://www.blindi.net/mailman/listinfo/blinde-misc From dujinfang at gmail.com Tue Dec 6 06:40:42 2011 From: dujinfang at gmail.com (Seven Du) Date: Tue, 6 Dec 2011 11:40:42 +0800 Subject: [Freeswitch-users] NAT traversal, NDLB-force-rport per user? Message-ID: <370B31E2D1EA48F2AFB36B3BFE0B7317@gmail.com> Hi, To help some UA for NAT I need set in user profile?however looks like the NDLB-force-rport doesn't work in a per user way, when I set in profile settings it works. 1) So questions is, is it possible to make it work a per-user way? I don't to do the risk to set it on a profile for all users. 2) Also, another question, looks both params fails to work when FS is behind kamailio(maybe opensips), my settings is: kamailio at the front listening at 5060 and forward anything to 5070 - the FS internal profile rewriteport("5070"); route(RELAY); exit; I'm not a kamailio guru, any possible way to make it work behind kamailio? Or should there some equivalent module/params in kamailio for this behavior? 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/20111206/8314f07f/attachment.html From darcy at primrose.ws Tue Dec 6 07:49:03 2011 From: darcy at primrose.ws (Darcy) Date: Mon, 5 Dec 2011 23:49:03 -0500 Subject: [Freeswitch-users] dray tek vigor 2920 Message-ID: <77C2266829D440CD8AE6AB043D9047A8@DWP> Has anyone had any experience with spa2102 atas behind a dray tek vigor 2920 router, I cannot get the gateways to register to the freeswitch, but they do register to a pbxnsip on the same subnet. Darcy Primrose -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/83989ff2/attachment-0001.html From boris at tagnet.ru Tue Dec 6 08:39:44 2011 From: boris at tagnet.ru (Boris Kovalenko) Date: Tue, 06 Dec 2011 11:39:44 +0600 Subject: [Freeswitch-users] Does I need jitter buffer Message-ID: <4EDDAAA0.4040004@tagnet.ru> Hello! Would You please explain does I need jitter or not? As I wrote about a month ago I offten have an echo when calling local_user -- freeswitch -- localuser. And _never_ have an echo when calling local_user -- freeswitch -- remote_gateway. My local network is gigabit based. remote_gateway is on internet. Will the jitter help me and what settings should I start from? -- Regards, Boris From notlikeme75 at yahoo.com Tue Dec 6 10:41:14 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Mon, 5 Dec 2011 23:41:14 -0800 (PST) Subject: [Freeswitch-users] pagd dynamic conference In-Reply-To: References: Message-ID: <1323157274.53780.YahooMailNeo@web65305.mail.ac2.yahoo.com> >>> Michael, I tried what you suggest and it just plays conf-pin.wav then goes right to bad-pin.wav then hangups without actually putting me in conference.before i tried this i tried using "read" instead of pagd and it accepted the pin and put me into the conference but didn't allow me any conference options or the ability to back out to main ivr.? I am still stuck :( . my system is ivr based incoming calls only so this feature is essential. all help is appreciated. ?condition??? ? destination_number??? ? ^3001$??? ? ? action??? ? answer??? ? ? ? action??? ? play_and_get_digits??? ? 4 4 3 # 2000 conf-pin.wav conf-bad-pin.wav \d+?? ? action??? ? conference??? ? ${digits}-${domain}@default I believe you just need to call play_and_get_digits (PAGD) prior to dropping them into the conference. If I understand the question correctly, the four-digits that the caller enters represents the "conference number" or whatever you call it. You could do this: naturally you'll need to read up on PAGD so that you understand what all that stuff is doing. Also, be sure to specify real sound files, not the fake ones that I used. The ${digits} value is what the caller actually dials. In a production environment you'll need to handle the scenario where the caller never actually enters a valid 4-digit number. -MC On Sun, Dec 4, 2011 at 11:44 AM, Rodney wrote: I have tried the nb_conference example in the default.xml but I think I am doing something wrong. when i transfer to that extension it drops me into a single static conference of the variable . what do i need to add or change to prompt my callers transfered to the nb_conference to enter a 4 digit conference number without pin that gives them the same options as my static conferences on the default profile? I know play and get digits must be involved but would appreciate the help. 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 > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111205/6deb25a7/attachment.html From avi at avimarcus.net Tue Dec 6 12:01:16 2011 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 6 Dec 2011 11:01:16 +0200 Subject: [Freeswitch-users] Identify the gateway In-Reply-To: <4EDD613F.7060609@btg.co.nz> References: <4EDD3F75.9080303@btg.co.nz> <4EDD613F.7060609@btg.co.nz> Message-ID: Did you kill and restart the gateway after adding the variable? (Or restart the entire interface, or restart FS?) -Avi On Tue, Dec 6, 2011 at 2:26 AM, Jort Bloem wrote: > Hi Avi & Michael, > > Michael, I'm familiar with the info application, and have been using this > to test my progress. It's gotten me a long way, but not past this hurdle. > > Avi, I tried that, or at least my interpretation of that, and it didn't > work. > > I found the gateway in question - it's one of only two references to the > trunk - in sip_profiles/. The declaration, including my addition of the > variables clause, looks like this: > > > > > > > > > > > > > I then called in via that interface, using a DDI, with the info > application. I got lots of information, but neither foo nor bar. I did > change the direction from "inbound" to "both", to make it as likely as > possible to work. > > The log output I got had lots of stuff in it (see > http://pastebin.freeswitch.org/17943 ) but no foo, no bar, and no > trunk_1. It did have several occurrences of the ip address of talk2, > 10.10.9.1, but that can change. I also see several references to the ip > address (10.10.10.149) and sofia profile name (sipinterface_3), but all > gateways will be on the same profile. > > Any further help would be appreciated. > > Jaybee- > > > > On 06/12/11 11:57, Avi Marcus wrote: > > > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055508.html > > Hmm, gateways on the wiki should be cleaned up.. I couldn't find it on > the wiki, only via google. > > -Avi > > > On Tue, Dec 6, 2011 at 12:45 AM, Michael Collins wrote: > >> Try adding one of these to the dialplan that handles inbound calls: >> >> >> >> Then, watch the console while an inbound comes in and you'll see quite >> a lot of information. Anything in that info dump is available to you for >> "figuring out" what's going on. If you have any questions then drop that >> info output to a pastebin and give us the URL here. >> >> -MC >> >> >> On Mon, Dec 5, 2011 at 2:02 PM, Jort Bloem wrote: >> >>> Hi all, >>> >>> I have a setup with several gateways coming in on one profile. The >>> Realms, unfortunately, are by DNS, and move from time to time. >>> >>> Is there any way to identify, in the dialplan, which gateway the call >>> comes from, for incoming calls? >>> >>> I know I can get the ip address, but the ip address is subject to change. >>> >>> I know that the trunk provider can set various things, but I don't have >>> control of the far end of the trunk. >>> >>> If there is a way to set variables on a per-gateway (NOT per-profile) >>> basis, that would be perfect. >>> >>> "RTFM" gladly accepted with a hint about which part of which FM to R. >>> >>> -- >>> Jaybee- >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/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 > > > -- > Jort Bloem > > > Technical Engineer - Auckland > Business Technology Group > > P: +64 9 5801374 x 9884 > > Sent from Mozilla Thunderbird > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/1dd07325/attachment-0001.html From beppe.grillo at gmail.com Tue Dec 6 12:11:21 2011 From: beppe.grillo at gmail.com (Beppe Grillo) Date: Tue, 6 Dec 2011 10:11:21 +0100 Subject: [Freeswitch-users] REGISTER using the draft-sip-outbound In-Reply-To: <1323103254684-7063523.post@n2.nabble.com> References: <1323103254684-7063523.post@n2.nabble.com> Message-ID: Yes, the draft-sip-outbound its now RFC5626. FreeSWITCH supported this is RFC ? 2011/12/5 peely > Isn't this now RFC5626? > > This is more of a client-side standard, are you wanting FreeSWITCH to > register a gateway using this standard? Doesn't seem that appropriate to > me. > > > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/REGISTER-using-the-draft-sip-outbound-tp7063199p7063523.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/20111206/9d8f9cfc/attachment.html From beppe.grillo at gmail.com Tue Dec 6 14:17:46 2011 From: beppe.grillo at gmail.com (Beppe Grillo) Date: Tue, 6 Dec 2011 12:17:46 +0100 Subject: [Freeswitch-users] REGISTER using the draft-sip-outbound In-Reply-To: References: Message-ID: Hy Brian, I have to change the software of mod_sofia or stack lib nua ? Nua supported RFC 5626 ? 2011/12/5 Brian West > write a patch and put it on jira.freeswitch.org, I don't think we have > that feature yet. > > /b > > On Dec 5, 2011, at 9:12 AM, Beppe Grillo wrote: > > Can your help please? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/f5c499ff/attachment.html From adam.kelloway at newpace.ca Tue Dec 6 16:03:05 2011 From: adam.kelloway at newpace.ca (Adam Kelloway) Date: Tue, 06 Dec 2011 09:03:05 -0400 Subject: [Freeswitch-users] lua error in session:transfer In-Reply-To: References: Message-ID: <4EDE1289.1060807@newpace.ca> Just a hunch, but I would guess command line input is being treated as ints. Note that, in lua, 1 == 1 evaluates to true, while "1" == 1 evaluates to false. I believe session:read returns a string. http://www.lua.org/manual/5.1/manual.html#2.5.2 Also, the last two elseif statements could also potentially set target_number to nil, as you are only passing the script 4 arguments. Likewise for the empty else at the end. Your transfer will work once this is fixed, assuming your have an extension called "target_number". Something tells me you actually want the extension number you are setting the variable to though, so you might want to remove the quotes around target_number when calling transfer. Add some print statements in you if/elseif clauses to see which one resolve to true. Hope those ideas help, Adam On 3:59 PM, Thomas Hoellriegel wrote: > Hi all, > i have a luascript. I hear the prompt, but the transfer fails. > My script is: > > session:answer(); > freeswitch.console_log("info", "Announce\n"); > announce='/usr/local/freeswitch/sounds/callscreen/callscrrenopts.wav'; > number = session:read(1, 1, announce, 5000, "#"); > freeswitch.consoleLog("info", "Got number: ".. number .. "\n"); > if number == argv[1] then > target_number=argv[2]; > elseif number == argv[3] then > target_number=argv[4]; > elseif number == argv[5] then > target_number=argv[6]; > elseif number == argv[7] then > target_number=argv[8]; > else > end > freeswitch.consoleLog("info", "Transfer to target number: ".. > target_number .. " XML default\n"); > session:transfer("target_number", "XML", "default"); > > in my dialplan reads: > > > Then i press 1 for example, i become on the cli: > [ERR] mod_lua.cpp:196 /usr/local/freeswitch/scripts/thomas.lua:16: > attempt to concatenate global 'target_number' (a nil value)M > > What is wrong please? > Thanks for your help. > > --------------- > 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 freeswitch at peely.com Tue Dec 6 16:34:23 2011 From: freeswitch at peely.com (peely) Date: Tue, 6 Dec 2011 05:34:23 -0800 (PST) Subject: [Freeswitch-users] dray tek vigor 2920 In-Reply-To: <77C2266829D440CD8AE6AB043D9047A8@DWP> References: <77C2266829D440CD8AE6AB043D9047A8@DWP> Message-ID: <1323178463232-7066799.post@n2.nabble.com> I have a 2820 and find I need change the SIP Port of each UA on the LAN to something other than 5060 to get responses back from SIP Servers on the Internet. I usually change them to something like 16384. I haven;t found any settings on the 2820 which switches off ALG behavior. Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Re-dray-tek-vigor-2920-tp7065611p7066799.html Sent from the freeswitch-users mailing list archive at Nabble.com. From freeswitch at peely.com Tue Dec 6 16:47:44 2011 From: freeswitch at peely.com (peely) Date: Tue, 6 Dec 2011 05:47:44 -0800 (PST) Subject: [Freeswitch-users] Max sessions for a javascript app? In-Reply-To: References: Message-ID: <1323179264324-7066833.post@n2.nabble.com> Have you checked you have enough RTP ports available in vars.conf.xml? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Max-sessions-for-a-javascript-app-tp7051877p7066833.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jeff at jefflenk.com Tue Dec 6 18:48:22 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Tue, 6 Dec 2011 07:48:22 -0800 (PST) Subject: [Freeswitch-users] Does I need jitter buffer In-Reply-To: <4EDDAAA0.4040004@tagnet.ru> References: <4EDDAAA0.4040004@tagnet.ru> Message-ID: <1323186502931-7067246.post@n2.nabble.com> Its not possible to have echo in a pure VOIP (properly functioning) implementation. The only source of echo would be from acoustic echo on your phones which is most likely what you are experiencing. A Jitter buffer wont help this problem. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Does-I-need-jitter-buffer-tp7065665p7067246.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ryan at kaevee.com Tue Dec 6 06:52:05 2011 From: ryan at kaevee.com (Ryan V) Date: Tue, 6 Dec 2011 09:22:05 +0530 Subject: [Freeswitch-users] PRI Line Configuration Message-ID: Hi, I am trying to configure Sangoma A101DE card with Bharati Airtel (India) PRI Line. Here is the output of "wanrouter status" Devices currently active: wanpipe1 Wanpipe Config: Device name | Protocol Map | Adapter | IRQ | Slot/IO | If's | CLK | Baud rate | wanpipe1 | N/A | A101/1D/A102/2D/4/4D/8| 16 | 4 | 1 | N/A | 0 | Wanrouter Status: Device name | Protocol | Station | Status | wanpipe1 | AFT TE1 | N/A | Connected | I have configured the card with following settings. Clock = Normal SwitchType = EuroISDN/ETSI Signalling = PRI NET Dialplan Context = Public DTMF Detection = Yes Hardware Fax Detection = No Freeswitch recognized the card and here is the output of ftdm list +OK span: 1 (wp1) type: Sangoma (ISDN) physical_status: ok signaling_status: UP chan_count: 31 dialplan: XML context: public dial_regex: fail_dial_regex: hold_music: analog_options: none I am stuck here. I don't receive any incoming calls and get a busy message when we dial our main DID no. Please tell me where I am going wrong? Thanks, Ryan. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/d5d181ae/attachment.html From chad at apartmentlines.com Tue Dec 6 19:10:03 2011 From: chad at apartmentlines.com (Chad Phillips -- Apartment Lines) Date: Tue, 6 Dec 2011 08:10:03 -0800 Subject: [Freeswitch-users] How best to use event system for many originates with listeners Message-ID: <4FD95AAF-424A-4A9C-B037-511DFA32C129@apartmentlines.com> I'm designing an application that will originate many calls via the event system which I also want to attach event listeners to. i want each event listener to only listen for events particular to each originated call. Since I want FreeSWITCH to control the call after I originate it, it seems that I should use an inbound connection. The two possible workflows I see are: a) Issue an originate command via inbound, then turn it into an outbound-type connection via a 'myevents ' call (or 'handlecall' using mod_erlang_event) once the channel is up. With this method, it seems that I could miss critical channel events between the time the call is originated and the listener is attached. b) Set up an inbound listener filtered on a pre-determined UUID for the call, then originate the call with that UUID. This would seem to guarantee that no events are missed by the listener, but I'm wondering about the performance of having potentially hundreds of inbound connections with filters going at the same time. So my questions are: 1) Is there a way to accomplish a) without potentially missing any channel events? 2) Is b) vastly less efficient than a)? From moises.silva at gmail.com Tue Dec 6 19:46:23 2011 From: moises.silva at gmail.com (Moises Silva) Date: Tue, 6 Dec 2011 11:46:23 -0500 Subject: [Freeswitch-users] SS7 on FreeSWITCH In-Reply-To: References: Message-ID: On Mon, Dec 5, 2011 at 12:02 PM, Anita Hall wrote: > Hi all ! > > I have to support SS7 on FreeSWITCH for a client. He told me that Asterisk > is _not_ good enough for him :) He has built entire infrastructure on FS. > > libss7, chan_ss7, openss7, ss7box, Sangoma Media Gateway ? What all > options do I have ? > > If none of these existing options will work with FreeSWITCH, is there > anyone who could modify chan_ss7 for FreeSWITCH or provide some other > alternative ? > > We have many installations all over the world with SS7 and FreeSWITCH. The Sangoma Media Gateway product was recently rebranded to Netborder SS7 Gateway to be consistent with our other gateway products. All of this FreeSWITCH based. You can choose between ISO installation or binary installation to choose your preferred Linux distribution. Take a look here: http://sangoma.com/products/software_products/ss7_solutions/ss7_voip_media_gateway.html *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 VegaStream is now part of Sangoma! Ask us about both Gateway Appliances and Internal Gateways -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/a0189e56/attachment-0001.html From tonybecq at yahoo.fr Tue Dec 6 19:50:06 2011 From: tonybecq at yahoo.fr (obbyone) Date: Tue, 6 Dec 2011 08:50:06 -0800 (PST) Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... In-Reply-To: <4EDA9484.9000000@integrafin.co.uk> References: <1322733452381-7049935.post@n2.nabble.com> <1322929911507-7058216.post@n2.nabble.com> <4EDA9484.9000000@integrafin.co.uk> Message-ID: <1323190206664-7067482.post@n2.nabble.com> Hi, In fact, freeswitch is installed on a virtual serveur (VMWare) that has a fixed external IP. Every ATA's are behind a different ADSL modem same as if it was also on an external IP. I'm a newbie on freeswitch so is there someone who can take me by the hand and help me to handle this issue... Thanks On 03/12/11 16:31, obbyone wrote: > Hi, > > The trace I gave was not a good one. The one I show now Is a good sample. > Any attempt to dial "1001" results only in a response by voicemail. Any > answer ? > > Thanks > From your trace, it looks like your ATAs are connecting to your external profile (something.dyndns.org:5080). Are you trying to connect your ATAs somewhere "out on the internet", possibly NAT'ed to a NAT'ed or otherwise Freeswitch server? If the ATAs are on the LAN local to your FS box you should be using the internal profile and default SIP port of 5060 on the ATAs. If so you need to read up on the NAT scenarios if you don't UPNP compatible routers at both ends of the connection (or don't want to use STUN at both ends). It's tricky but it is possible to get it working even with NAT on both sides. I just got it working. Cheers Alex -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-installed-ATA-registered-but-no-call-are-possible-tp7049935p7067482.html Sent from the freeswitch-users mailing list archive at Nabble.com. From moises.silva at gmail.com Tue Dec 6 19:50:19 2011 From: moises.silva at gmail.com (Moises Silva) Date: Tue, 6 Dec 2011 11:50:19 -0500 Subject: [Freeswitch-users] PRI Line Configuration In-Reply-To: References: Message-ID: On Mon, Dec 5, 2011 at 10:52 PM, Ryan V wrote: > Freeswitch recognized the card and here is the output of ftdm list > > +OK > span: 1 (wp1) > type: Sangoma (ISDN) > physical_status: ok > signaling_status: UP > chan_count: 31 > dialplan: XML > context: public > dial_regex: > fail_dial_regex: > hold_music: > analog_options: none > > I am stuck here. I don't receive any incoming calls and get a busy message > when we dial our main DID no. > > Please tell me where I am going wrong? > > That looks pretty good to me. Try enabling all debugging output. You can also try to trace the d-channel and dump it to a pcap file to see if you receive any messages from the telco indicating a new call. http://wiki.sangoma.com/wanpipe-wireshark-pcap-pri-bri-wan-t1-e1-tracing What about outgoing calls? *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 VegaStream is now part of Sangoma! Ask us about both Gateway Appliances and Internal Gateways -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/7379f696/attachment.html From ryan at kaevee.com Tue Dec 6 20:13:42 2011 From: ryan at kaevee.com (Ryan V) Date: Tue, 6 Dec 2011 22:43:42 +0530 Subject: [Freeswitch-users] PRI Line Configuration In-Reply-To: References: Message-ID: On Tue, Dec 6, 2011 at 10:20 PM, Moises Silva wrote: > On Mon, Dec 5, 2011 at 10:52 PM, Ryan V wrote: > >> Freeswitch recognized the card and here is the output of ftdm list >> >> +OK >> span: 1 (wp1) >> type: Sangoma (ISDN) >> physical_status: ok >> signaling_status: UP >> chan_count: 31 >> dialplan: XML >> context: public >> dial_regex: >> fail_dial_regex: >> hold_music: >> analog_options: none >> >> I am stuck here. I don't receive any incoming calls and get a busy >> message when we dial our main DID no. >> >> Please tell me where I am going wrong? >> >> > That looks pretty good to me. Try enabling all debugging output. You can > also try to trace the d-channel and dump it to a pcap file to see if you > receive any messages from the telco indicating a new call. > > http://wiki.sangoma.com/wanpipe-wireshark-pcap-pri-bri-wan-t1-e1-tracing > > What about outgoing calls? > > I made a stupid mistake of connecting to our existing EPABX instead of connecting to modem :( I figured it out couple of hours back.. and connected it to our Telco modem. I am now receiving the calls. As I am totally newbie to FreeSwitch or soft PBX, I have to figure out how to setup outgoing calls :) In fact, I was about to write to list and update my email soon. Thanks, Venkatesh K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/d7ced6df/attachment.html From acrow at integrafin.co.uk Tue Dec 6 20:30:44 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Tue, 06 Dec 2011 17:30:44 +0000 Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... In-Reply-To: <1323190206664-7067482.post@n2.nabble.com> References: <1322733452381-7049935.post@n2.nabble.com> <1322929911507-7058216.post@n2.nabble.com> <4EDA9484.9000000@integrafin.co.uk> <1323190206664-7067482.post@n2.nabble.com> Message-ID: <4EDE5144.4020703@integrafin.co.uk> On 06/12/11 16:50, obbyone wrote: > Hi, > In fact, freeswitch is installed on a virtual serveur (VMWare) that has a > fixed external IP. Every ATA's are behind a different ADSL modem same as if > it was also on an external IP. I'm a newbie on freeswitch so is there > someone who can take me by the hand and help me to handle this issue... > > Thanks Hi, Does the FS VM have a fixed IP or is it natted? I suspect the latter. I would follow the guide on the Wiki about "NAT scenarios", specifically the one about double NAT. You will need to create another profile for this: http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios You also need to forward your chosen SIP port for the profile (eg 5090) and the RTP ports (16384-32768) to the real IP of the FS VM. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From jort.bloem at btg.co.nz Tue Dec 6 20:42:48 2011 From: jort.bloem at btg.co.nz (Jort Bloem) Date: Wed, 07 Dec 2011 06:42:48 +1300 Subject: [Freeswitch-users] Identify the gateway In-Reply-To: References: <4EDD3F75.9080303@btg.co.nz> <4EDD613F.7060609@btg.co.nz> Message-ID: <4EDE5418.5050703@btg.co.nz> I fully shut it down freeswitch, and then started it up again (I've found that the time wasted doing a complete shutdown is more than offset by the hours you can waste trying to figure out why your changes don't take effect ;-) ) On 06/12/11 22:01, Avi Marcus wrote: > Did you kill and restart the gateway after adding the variable? (Or > restart the entire interface, or restart FS?) > > -Avi > > > On Tue, Dec 6, 2011 at 2:26 AM, Jort Bloem > wrote: > > Hi Avi & Michael, > > Michael, I'm familiar with the info application, and have been > using this to test my progress. It's gotten me a long way, but not > past this hurdle. > > Avi, I tried that, or at least my interpretation of that, and it > didn't work. > > I found the gateway in question - it's one of only two references > to the trunk - in sip_profiles/. The declaration, including my > addition of the variables clause, looks like this: > > > > > > > > > > > > > I then called in via that interface, using a DDI, with the info > application. I got lots of information, but neither foo nor bar. I > did change the direction from "inbound" to "both", to make it as > likely as possible to work. > > The log output I got had lots of stuff in it (see > http://pastebin.freeswitch.org/17943 ) but no foo, no bar, and no > trunk_1. It did have several occurrences of the ip address of > talk2, 10.10.9.1, but that can change. I also see several > references to the ip address (10.10.10.149) and sofia profile name > (sipinterface_3), but all gateways will be on the same profile. > > Any further help would be appreciated. > > Jaybee- > > > > On 06/12/11 11:57, Avi Marcus wrote: >> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055508.html >> >> Hmm, gateways on the wiki should be cleaned up.. I couldn't find >> it on the wiki, only via google. >> >> -Avi >> >> >> On Tue, Dec 6, 2011 at 12:45 AM, Michael Collins >> > wrote: >> >> Try adding one of these to the dialplan that handles inbound >> calls: >> >> >> >> Then, watch the console while an inbound comes in and you'll >> see quite a lot of information. Anything in that info dump is >> available to you for "figuring out" what's going on. If you >> have any questions then drop that info output to a pastebin >> and give us the URL here. >> >> -MC >> >> >> On Mon, Dec 5, 2011 at 2:02 PM, Jort Bloem >> > wrote: >> >> Hi all, >> >> I have a setup with several gateways coming in on one >> profile. The >> Realms, unfortunately, are by DNS, and move from time to >> time. >> >> Is there any way to identify, in the dialplan, which >> gateway the call >> comes from, for incoming calls? >> >> I know I can get the ip address, but the ip address is >> subject to change. >> >> I know that the trunk provider can set various things, >> but I don't have >> control of the far end of the trunk. >> >> If there is a way to set variables on a per-gateway (NOT >> per-profile) >> basis, that would be perfect. >> >> "RTFM" gladly accepted with a hint about which part of >> which FM to R. >> >> -- >> Jaybee- >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > Jort Bloem > > > Technical Engineer - Auckland > Business Technology Group > > P:+64 9 5801374 x 9884 > > Sent from Mozilla Thunderbird > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Jaybee -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111207/e797d5c6/attachment-0001.html From Alessandro.Crespi at italtel.it Tue Dec 6 19:59:44 2011 From: Alessandro.Crespi at italtel.it (Crespi Alessandro) Date: Tue, 6 Dec 2011 17:59:44 +0100 Subject: [Freeswitch-users] SIP UPDATE Support Message-ID: <8E6DA0ECC9F8EE4483CBF31602B1537E0541BF3A@BESONE.corp.dom> Hi All, I am testing the FS behavior receiving the SIP UPDATE message in a SIP to SIP call in early media state for media modification. FS does not negotiate end to end the UPDATE/SDP offer but answer 200 OK locally with the SDP answer. In addition It seems that UPDATE doesn't have any impact on media handling. These are two examples: - UPDATE with codec modification: result -> 200OK with SDP answer without audio codec - UPDATE with IP address modification -> 200 OK SDP answer Do you know if FS support UPDATE for media modification? Below the full log. Thanks Alessandro UPDATE with codec modification freeswitch at internal > recv 818 bytes from udp/[138.132.110.64]:5065 at 08:10:45.038063: ------------------------------------------------------------------------ INVITE sip:1001 at 138.132.105.40 SIP/2.0 Via: SIP/2.0/UDP 138.132.110.64:5065;branch=z9hG4bKxTP2KbyJab0564 To: "1001" > From: "1000" >;tag=3bad0564 Call-ID: 05646d49df5d85548e08b348fa46b455 at 138.132.110.64 CSeq: 1 INVITE Max-Forwards: 70 Contact: "1000" > User-Agent: SIP TP2000 Emu (Build: 240464) Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE Supported: 100rel Content-Type: application/sdp Content-Length: 255 v=0 o=TI-WLAN-PHONE 0 0 IN IP4 138.132.110.64 s=TI-WLAN-CALL c=IN IP4 138.132.110.64 t=0 0 m=audio 10136 RTP/AVP 8 18 101 102 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:102 G.729a/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 331 bytes to udp/[138.132.110.64]:5065 at 08:10:45.047354: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 138.132.110.64:5065;branch=z9hG4bKxTP2KbyJab0564 From: "1000" >;tag=3bad0564 To: "1001" > Call-ID: 05646d49df5d85548e08b348fa46b455 at 138.132.110.64 CSeq: 1 INVITE User-Agent: Italtel-WeGate/HW_V1.0.0/FW_V1.0.0/SW_V1.0.0 Content-Length: 0 ------------------------------------------------------------------------ 1970-01-06 08:10:45.041214 [NOTICE] switch_channel.c:920 New Channel sofia/external/1000 at 138.132.105.40 [6837d3fd-27eb-4832-9835-6cf7074dc859] 1970-01-06 08:10:45.060669 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1000 at 138.132.105.40 ) Running State Change CS_NEW 1970-01-06 08:10:45.060669 [DEBUG] switch_core_state_machine.c:380 (sofia/external/1000 at 138.132.105.40 ) State NEW 1970-01-06 08:10:45.060669 [DEBUG] sofia.c:5390 Channel sofia/external/1000 at 138.132.105.40 entering state [received][100] 1970-01-06 08:10:45.060669 [DEBUG] sofia.c:5401 Remote SDP: v=0 o=TI-WLAN-PHONE 0 0 IN IP4 138.132.110.64 s=TI-WLAN-CALL c=IN IP4 138.132.110.64 t=0 0 m=audio 10136 RTP/AVP 8 18 101 102 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:102 G.729a/8000 1970-01-06 08:10:45.060669 [DEBUG] sofia.c:5594 (sofia/external/1000 at 138.132.105.40 ) State Change CS_NEW -> CS_INIT 1970-01-06 08:10:45.060669 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] 1970-01-06 08:10:45.060669 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1000 at 138.132.105.40 ) Running State Change CS_INIT 1970-01-06 08:10:45.060669 [DEBUG] switch_core_state_machine.c:401 (sofia/external/1000 at 138.132.105.40 ) State INIT 1970-01-06 08:10:45.060669 [DEBUG] mod_sofia.c:85 sofia/external/1000 at 138.132.105.40 SOFIA INIT 1970-01-06 08:10:45.060669 [DEBUG] mod_sofia.c:125 (sofia/external/1000 at 138.132.105.40 ) State Change CS_INIT -> CS_ROUTING 1970-01-06 08:10:45.060669 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] 1970-01-06 08:10:45.060669 [DEBUG] switch_core_state_machine.c:401 (sofia/external/1000 at 138.132.105.40 ) State INIT going to sleep 1970-01-06 08:10:45.060669 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1000 at 138.132.105.40 ) Running State Change CS_ROUTING 1970-01-06 08:10:45.060669 [DEBUG] switch_channel.c:1871 (sofia/external/1000 at 138.132.105.40 ) Callstate Change DOWN -> RINGING 1970-01-06 08:10:45.060669 [DEBUG] switch_core_state_machine.c:410 (sofia/external/1000 at 138.132.105.40 ) State ROUTING 1970-01-06 08:10:45.060669 [DEBUG] mod_sofia.c:148 sofia/external/1000 at 138.132.105.40 SOFIA ROUTING 1970-01-06 08:10:45.060669 [DEBUG] switch_core_state_machine.c:104 sofia/external/1000 at 138.132.105.40 Standard ROUTING 1970-01-06 08:10:45.060669 [INFO] mod_dialplan_xml.c:481 Processing 1000 <1000>->1001 in context public Dialplan: sofia/external/1000 at 138.132.105.40 parsing [public->dial_from_gateway1] continue=false Dialplan: sofia/external/1000 at 138.132.105.40 Regex (FAIL) [dial_from_gateway1] destination_number(1001) =~ /1000/ break=on-false Dialplan: sofia/external/1000 at 138.132.105.40 parsing [public->dial_from_gateway1] continue=false Dialplan: sofia/external/1000 at 138.132.105.40 Regex (PASS) [dial_from_gateway1] destination_number(1001) =~ /1001/ break=on-false Dialplan: sofia/external/1000 at 138.132.105.40 Action set(call_timeout=400) Dialplan: sofia/external/1000 at 138.132.105.40 Action set(inherit_codec=true) Dialplan: sofia/external/1000 at 138.132.105.40 Action bridge(sofia/external/1001 at 138.132.110.64:5070 ) 1970-01-06 08:10:45.101525 [DEBUG] switch_core_state_machine.c:154 (sofia/external/1000 at 138.132.105.40 ) State Change CS_ROUTING -> CS_EXECUTE 1970-01-06 08:10:45.101525 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] 1970-01-06 08:10:45.101525 [DEBUG] switch_core_state_machine.c:410 (sofia/external/1000 at 138.132.105.40 ) State ROUTING going to sleep 1970-01-06 08:10:45.101525 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1000 at 138.132.105.40 ) Running State Change CS_EXECUTE 1970-01-06 08:10:45.101525 [DEBUG] switch_core_state_machine.c:417 (sofia/external/1000 at 138.132.105.40 ) State EXECUTE 1970-01-06 08:10:45.101525 [DEBUG] mod_sofia.c:241 sofia/external/1000 at 138.132.105.40 SOFIA EXECUTE 1970-01-06 08:10:45.101525 [DEBUG] switch_core_state_machine.c:192 sofia/external/1000 at 138.132.105.40 Standard EXECUTE EXECUTE sofia/external/1000 at 138.132.105.40 set(call_timeout=400) 1970-01-06 08:10:45.101525 [DEBUG] mod_dptools.c:1263 sofia/external/1000 at 138.132.105.40 SET [call_timeout]=[400] EXECUTE sofia/external/1000 at 138.132.105.40 set(inherit_codec=true) 1970-01-06 08:10:45.101525 [DEBUG] mod_dptools.c:1263 sofia/external/1000 at 138.132.105.40 SET [inherit_codec]=[true] EXECUTE sofia/external/1000 at 138.132.105.40 bridge(sofia/external/1001 at 138.132.110.64:5070 ) 1970-01-06 08:10:45.101525 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 1970-01-06 08:10:45.101525 [NOTICE] switch_channel.c:920 New Channel sofia/external/1001 at 138.132.110.64:5070 [478e7aac-980d-43b5-834a-4f9e38102547] 1970-01-06 08:10:45.101525 [DEBUG] mod_sofia.c:4539 (sofia/external/1001 at 138.132.110.64:5070 ) State Change CS_NEW -> CS_INIT 1970-01-06 08:10:45.101525 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/1001 at 138.132.110.64:5070 [BREAK] 1970-01-06 08:10:45.101525 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1001 at 138.132.110.64:5070 ) Running State Change CS_INIT 1970-01-06 08:10:45.101525 [DEBUG] switch_core_state_machine.c:401 (sofia/external/1001 at 138.132.110.64:5070 ) State INIT 1970-01-06 08:10:45.101525 [DEBUG] mod_sofia.c:85 sofia/external/1001 at 138.132.110.64:5070 SOFIA INIT 1970-01-06 08:10:45.141164 [DEBUG] mod_sofia.c:125 (sofia/external/1001 at 138.132.110.64:5070 ) State Change CS_INIT -> CS_ROUTING 1970-01-06 08:10:45.141164 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/1001 at 138.132.110.64:5070 [BREAK] 1970-01-06 08:10:45.141164 [DEBUG] switch_core_state_machine.c:401 (sofia/external/1001 at 138.132.110.64:5070 ) State INIT going to sleep 1970-01-06 08:10:45.141164 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1001 at 138.132.110.64:5070 ) Running State Change CS_ROUTING 1970-01-06 08:10:45.141164 [DEBUG] switch_channel.c:1871 (sofia/external/1001 at 138.132.110.64:5070 ) Callstate Change DOWN -> RINGING send 1033 bytes to udp/[138.132.110.64]:5070 at 08:10:45.152568: ------------------------------------------------------------------------ INVITE sip:1001 at 138.132.110.64:5070 SIP/2.0 Via: SIP/2.0/UDP 138.132.105.40;rport;branch=z9hG4bK5FS2tNSXeU48a Max-Forwards: 69 From: "1000" >;tag=gX2j9p6vFN8BN To: > Call-ID: 4d9c002d-9ca7-1200-1ca0-0020da862374 CSeq: 30983298 INVITE Contact: > User-Agent: Italtel-WeGate/HW_V1.0.0/FW_V1.0.0/SW_V1.0.0 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY Supported: 100rel, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 254 X-FS-Support: update_display Remote-Party-ID: "1000" >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 436277 436278 IN IP4 138.132.105.40 s=FreeSWITCH c=IN IP4 138.132.105.40 t=0 0 m=audio 25168 RTP/AVP 8 18 98 101 13 a=rtpmap:98 AMR-WB/16000 a=fmtp:98 octet-align=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 1970-01-06 08:10:45.141164 [DEBUG] switch_core_session.c:872 Send signal sofia/external/1001 at 138.132.110.64:5070 [BREAK] 1970-01-06 08:10:45.141164 [DEBUG] switch_core_state_machine.c:410 (sofia/external/1001 at 138.132.110.64:5070 ) State ROUTING 1970-01-06 08:10:45.141164 [DEBUG] mod_sofia.c:148 sofia/external/1001 at 138.132.110.64:5070 SOFIA ROUTING 1970-01-06 08:10:45.141164 [DEBUG] switch_ivr_originate.c:66 (sofia/external/1001 at 138.132.110.64:5070 ) State Change CS_ROUTING -> CS_CONSUME_MEDIA 1970-01-06 08:10:45.141164 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/1001 at 138.132.110.64:5070 [BREAK] 1970-01-06 08:10:45.141164 [DEBUG] switch_core_state_machine.c:410 (sofia/external/1001 at 138.132.110.64:5070 ) State ROUTING going to sleep 1970-01-06 08:10:45.141164 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1001 at 138.132.110.64:5070 ) Running State Change CS_CONSUME_MEDIA 1970-01-06 08:10:45.141164 [DEBUG] switch_core_state_machine.c:429 (sofia/external/1001 at 138.132.110.64:5070 ) State CONSUME_MEDIA 1970-01-06 08:10:45.141164 [DEBUG] switch_core_state_machine.c:429 (sofia/external/1001 at 138.132.110.64:5070 ) State CONSUME_MEDIA going to sleep 1970-01-06 08:10:45.141164 [DEBUG] sofia.c:5390 Channel sofia/external/1001 at 138.132.110.64:5070 entering state [calling][0] recv 299 bytes from udp/[138.132.110.64]:5070 at 08:10:45.281081: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 138.132.105.40;rport;branch=z9hG4bK5FS2tNSXeU48a To: > From: "1000" >;tag=gX2j9p6vFN8BN Call-ID: 4d9c002d-9ca7-1200-1ca0-0020da862374 CSeq: 30983298 INVITE User Agent: TP2000 By JAB Content-Length: 0 ------------------------------------------------------------------------ recv 682 bytes from udp/[138.132.110.64]:5070 at 08:10:45.285542: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 138.132.105.40;rport;branch=z9hG4bK5FS2tNSXeU48a To: >;tag=beef4243 From: "1000" >;tag=gX2j9p6vFN8BN Call-ID: 4d9c002d-9ca7-1200-1ca0-0020da862374 CSeq: 30983298 INVITE Contact: > Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE Require: 100rel RSeq: 84 Content-Type: application/sdp Content-Length: 178 v=0 o=TI-WLAN-PHONE 0 1 IN IP4 138.132.110.64 s=TI-WLAN-CALL c=IN IP4 138.132.110.64 t=0 0 m=audio 10136 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 602 bytes to udp/[138.132.110.64]:5070 at 08:10:45.288805: ------------------------------------------------------------------------ PRACK sip:1001 at 138.132.110.64:5070 SIP/2.0 Via: SIP/2.0/UDP 138.132.105.40;rport;branch=z9hG4bKyNFXv3H4a4XBD Max-Forwards: 70 From: "1000" >;tag=gX2j9p6vFN8BN To: >;tag=beef4243 Call-ID: 4d9c002d-9ca7-1200-1ca0-0020da862374 CSeq: 30983299 PRACK Contact: > RAck: 84 30983298 INVITE User-Agent: Italtel-WeGate/HW_V1.0.0/FW_V1.0.0/SW_V1.0.0 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY Supported: 100rel, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 1970-01-06 08:10:45.281994 [DEBUG] switch_core_session.c:872 Send signal sofia/external/1001 at 138.132.110.64:5070 [BREAK] 1970-01-06 08:10:45.281994 [DEBUG] switch_core_session.c:872 Send signal sofia/external/1001 at 138.132.110.64:5070 [BREAK] 1970-01-06 08:10:45.281994 [DEBUG] sofia.c:5390 Channel sofia/external/1001 at 138.132.110.64:5070 entering state [proceeding][183] 1970-01-06 08:10:45.281994 [DEBUG] sofia.c:5401 Remote SDP: v=0 o=TI-WLAN-PHONE 0 1 IN IP4 138.132.110.64 s=TI-WLAN-CALL c=IN IP4 138.132.110.64 t=0 0 m=audio 10136 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 1970-01-06 08:10:45.281994 [DEBUG] sofia.c:5516 [GIO] sofia_handle_sip_i_state() : 1970-01-06 08:10:45.281994 [DEBUG] sofia_glue.c:3904 [GIO] sofia_glue_tech_media() : 1970-01-06 08:10:45.281994 [DEBUG] sofia_glue.c:4304 [GIO] sofia_glue_negotiate_sdp() : 1970-01-06 08:10:45.281994 [DEBUG] sofia_glue.c:4768 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 1970-01-06 08:10:45.281994 [DEBUG] sofia_glue.c:4768 Audio Codec Compare [G729:18:8000:20:8000]/[G729:18:8000:20:8000] 1970-01-06 08:10:45.281994 [DEBUG] sofia_glue.c:2887 Set Codec sofia/external/1001 at 138.132.110.64:5070 G729/8000 20 ms 160 samples 8000 bits 1970-01-06 08:10:45.300933 [DEBUG] sofia_glue.c:4882 Set 2833 dtmf send payload to 101 1970-01-06 08:10:45.300933 [DEBUG] sofia_glue.c:3139 AUDIO RTP [sofia/external/1001 at 138.132.110.64:5070] 138.132.105.40 port 25168 -> 138.132.110.64 port 10136 codec: 18 ms: 20 1970-01-06 08:10:45.300933 [DEBUG] switch_rtp.c:1642 Starting timer [soft] 160 bytes per 20ms 1970-01-06 08:10:45.300933 [DEBUG] sofia_glue.c:3403 Set 2833 dtmf send payload to 101 1970-01-06 08:10:45.300933 [DEBUG] sofia_glue.c:3409 Set 2833 dtmf receive payload to 101 1970-01-06 08:10:45.300933 [NOTICE] sofia_glue.c:3915 Pre-Answer sofia/external/1001 at 138.132.110.64:5070! 1970-01-06 08:10:45.300933 [DEBUG] switch_channel.c:2917 (sofia/external/1001 at 138.132.110.64:5070 ) Callstate Change RINGING -> EARLY 1970-01-06 08:10:45.340087 [DEBUG] switch_ivr_originate.c:405 Setting codec string on sofia/external/1000 at 138.132.105.40 to G729 at 8000h@20i 1970-01-06 08:10:45.340087 [INFO] switch_ivr_originate.c:3215 Sending early media 1970-01-06 08:10:45.340087 [DEBUG] sofia_glue.c:3904 [GIO] sofia_glue_tech_media() : 1970-01-06 08:10:45.340087 [DEBUG] sofia_glue.c:4304 [GIO] sofia_glue_negotiate_sdp() : 1970-01-06 08:10:45.340087 [DEBUG] sofia_glue.c:4768 Audio Codec Compare [PCMA:8:8000:20:64000]/[G729:18:8000:20:8000] 1970-01-06 08:10:45.340087 [DEBUG] sofia_glue.c:4768 Audio Codec Compare [G729:18:8000:20:8000]/[G729:18:8000:20:8000] 1970-01-06 08:10:45.340087 [DEBUG] sofia_glue.c:2887 Set Codec sofia/external/1000 at 138.132.105.40 G729/8000 20 ms 160 samples 8000 bits 1970-01-06 08:10:45.340087 [DEBUG] sofia_glue.c:4889 Set 2833 dtmf send/recv payload to 101 1970-01-06 08:10:45.340087 [DEBUG] sofia_glue.c:3139 AUDIO RTP [sofia/external/1000 at 138.132.105.40] 138.132.105.40 port 18270 -> 138.132.110.64 port 10136 codec: 18 ms: 20 1970-01-06 08:10:45.340087 [DEBUG] switch_rtp.c:1642 Starting timer [soft] 160 bytes per 20ms 1970-01-06 08:10:45.360383 [DEBUG] sofia_glue.c:3403 Set 2833 dtmf send payload to 101 1970-01-06 08:10:45.360383 [DEBUG] sofia_glue.c:3409 Set 2833 dtmf receive payload to 101 1970-01-06 08:10:45.360383 [NOTICE] sofia_glue.c:3915 Pre-Answer sofia/external/1000 at 138.132.105.40! 1970-01-06 08:10:45.360383 [DEBUG] switch_channel.c:2917 (sofia/external/1000 at 138.132.105.40 ) Callstate Change RINGING -> EARLY 1970-01-06 08:10:45.420089 [DEBUG] switch_channel.c:2959 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] 1970-01-06 08:10:45.420089 [DEBUG] mod_sofia.c:2500 Ring SDP: v=0 o=FreeSWITCH 0000443175 0000443176 IN IP4 138.132.105.40 s=FreeSWITCH c=IN IP4 138.132.105.40 t=0 0 m=audio 18270 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv send 1048 bytes to udp/[138.132.110.64]:5065 at 08:10:45.428139: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 138.132.110.64:5065;branch=z9hG4bKxTP2KbyJab0564 From: "1000" >;tag=3bad0564 To: "1001" >;tag=QQcp78HpKcF9j Call-ID: 05646d49df5d85548e08b348fa46b455 at 138.132.110.64 CSeq: 1 INVITE Contact: > RSeq: 1101569577 User-Agent: Italtel-WeGate/HW_V1.0.0/FW_V1.0.0/SW_V1.0.0 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY Require: 100rel Supported: 100rel, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 245 Remote-Party-ID: "1001" >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 443175 443176 IN IP4 138.132.105.40 s=FreeSWITCH c=IN IP4 138.132.105.40 t=0 0 m=audio 18270 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 1970-01-06 08:10:45.420089 [DEBUG] switch_core_session.c:872 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] recv 367 bytes from udp/[138.132.110.64]:5065 at 08:10:45.436792: ------------------------------------------------------------------------ PRACK sip:1001 at 138.132.105.40:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 138.132.110.64:5065;branch=z9hG4bKe27d3961 To: "1001" >;tag=QQcp78HpKcF9j From: "1000" >;tag=3bad0564 Call-ID: 05646d49df5d85548e08b348fa46b455 at 138.132.110.64 CSeq: 2 PRACK Max-Forwards: 70 RAck: 1101569577 1 INVITE Content-Length: 0 ------------------------------------------------------------------------ send 564 bytes to udp/[138.132.110.64]:5065 at 08:10:45.438378: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 138.132.110.64:5065;branch=z9hG4bKe27d3961 From: "1000" >;tag=3bad0564 To: "1001" >;tag=QQcp78HpKcF9j Call-ID: 05646d49df5d85548e08b348fa46b455 at 138.132.110.64 CSeq: 2 PRACK Contact: > User-Agent: Italtel-WeGate/HW_V1.0.0/FW_V1.0.0/SW_V1.0.0 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY Supported: 100rel, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 1970-01-06 08:10:45.420089 [DEBUG] switch_core_session.c:872 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] recv 339 bytes from udp/[138.132.110.64]:5070 at 08:10:45.440556: ------------------------------------------------------------------------ SIP/2.0 200 OK Call-ID: 4d9c002d-9ca7-1200-1ca0-0020da862374 Via: SIP/2.0/UDP 138.132.105.40;rport;branch=z9hG4bKyNFXv3H4a4XBD To: >;tag=beef4243 From: "1000" >;tag=gX2j9p6vFN8BN CSeq: 30983299 PRACK Contact: > Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 1970-01-06 08:10:45.442044 [DEBUG] switch_core_session.c:872 Send signal sofia/external/1001 at 138.132.110.64:5070 [BREAK] 1970-01-06 08:10:45.442044 [DEBUG] sofia.c:5383 Channel sofia/external/1000 at 138.132.105.40 skipping state [early][183] 1970-01-06 08:10:45.442044 [DEBUG] switch_core_session.c:726 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] 1970-01-06 08:10:45.442044 [DEBUG] switch_ivr_originate.c:3269 Originate Resulted in Success: [sofia/external/1001 at 138.132.110.64:5070] 1970-01-06 08:10:45.442044 [DEBUG] switch_core_session.c:726 Send signal sofia/external/1001 at 138.132.110.64:5070 [BREAK] 1970-01-06 08:10:45.442044 [DEBUG] switch_core_session.c:726 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] 1970-01-06 08:10:45.442044 [DEBUG] switch_ivr_bridge.c:1270 (sofia/external/1001 at 138.132.110.64:5070 ) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 1970-01-06 08:10:45.442044 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/1001 at 138.132.110.64:5070 [BREAK] 1970-01-06 08:10:45.442044 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1001 at 138.132.110.64:5070 ) Running State Change CS_EXCHANGE_MEDIA 1970-01-06 08:10:45.442044 [DEBUG] switch_core_state_machine.c:420 (sofia/external/1001 at 138.132.110.64:5070 ) State EXCHANGE_MEDIA 1970-01-06 08:10:45.442044 [DEBUG] mod_sofia.c:574 SOFIA EXCHANGE_MEDIA recv 593 bytes from udp/[138.132.110.64]:5065 at 08:10:46.445851: ------------------------------------------------------------------------ UPDATE sip:1001 at 138.132.105.40:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 138.132.110.64:5065;branch=z9hG4bKe27d3597 To: "1001" >;tag=QQcp78HpKcF9j From: "1000" >;tag=3bad0564 Call-ID: 05646d49df5d85548e08b348fa46b455 at 138.132.110.64 CSeq: 3 UPDATE Max-Forwards: 70 Contact: > Content-Type: application/sdp Content-Length: 177 v=0 o=TI-WLAN-PHONE 0 2 IN IP4 138.132.110.64 s=TI-WLAN-CALL c=IN IP4 138.132.110.64 t=0 0 m=audio 10136 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 847 bytes to udp/[138.132.110.64]:5065 at 08:10:46.448561: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 138.132.110.64:5065;branch=z9hG4bKe27d3597 From: "1000" >;tag=3bad0564 To: "1001" >;tag=QQcp78HpKcF9j Call-ID: 05646d49df5d85548e08b348fa46b455 at 138.132.110.64 CSeq: 3 UPDATE Contact: > User-Agent: Italtel-WeGate/HW_V1.0.0/FW_V1.0.0/SW_V1.0.0 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY Supported: 100rel, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 219 v=0 o=FreeSWITCH 443175 443177 IN IP4 138.132.105.40 s=FreeSWITCH c=IN IP4 138.132.105.40 t=0 0 m=audio 18270 RTP/AVP 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 1970-01-06 08:10:46.440055 [DEBUG] switch_core_session.c:872 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] 1970-01-06 08:10:46.460090 [DEBUG] switch_core_session.c:872 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] 1970-01-06 08:10:46.480073 [DEBUG] sofia.c:5390 Channel sofia/external/1000 at 138.132.105.40 entering state [early][200] 1970-01-06 08:10:46.480073 [DEBUG] sofia.c:5401 Remote SDP: v=0 o=TI-WLAN-PHONE 0 2 IN IP4 138.132.110.64 s=TI-WLAN-CALL c=IN IP4 138.132.110.64 t=0 0 m=audio 10136 RTP/AVP 8 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 UPDATE with IP address modification freeswitch at internal > recv 818 bytes from udp/[138.132.110.64]:5065 at 08:19:06.920845: ------------------------------------------------------------------------ INVITE sip:1001 at 138.132.105.40 SIP/2.0 Via: SIP/2.0/UDP 138.132.110.64:5065;branch=z9hG4bKxTP2KbyJab6373 To: "1001" > From: "1000" >;tag=3bad6373 Call-ID: 63736d49df5d85548e08b348fa46b455 at 138.132.110.64 CSeq: 1 INVITE Max-Forwards: 70 Contact: "1000" > User-Agent: SIP TP2000 Emu (Build: 240464) Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE Supported: 100rel Content-Type: application/sdp Content-Length: 255 v=0 o=TI-WLAN-PHONE 0 0 IN IP4 138.132.110.64 s=TI-WLAN-CALL c=IN IP4 138.132.110.64 t=0 0 m=audio 10136 RTP/AVP 8 18 101 102 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:102 G.729a/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 331 bytes to udp/[138.132.110.64]:5065 at 08:19:06.923992: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 138.132.110.64:5065;branch=z9hG4bKxTP2KbyJab6373 From: "1000" >;tag=3bad6373 To: "1001" > Call-ID: 63736d49df5d85548e08b348fa46b455 at 138.132.110.64 CSeq: 1 INVITE User-Agent: Italtel-WeGate/HW_V1.0.0/FW_V1.0.0/SW_V1.0.0 Content-Length: 0 ------------------------------------------------------------------------ 1970-01-06 08:19:06.920625 [NOTICE] switch_channel.c:920 New Channel sofia/external/1000 at 138.132.105.40 [bb52cd26-2d20-4d0c-ad8a-b2efedd16f6c] 1970-01-06 08:19:06.920625 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1000 at 138.132.105.40 ) Running State Change CS_NEW 1970-01-06 08:19:06.920625 [DEBUG] switch_core_state_machine.c:380 (sofia/external/1000 at 138.132.105.40 ) State NEW 1970-01-06 08:19:06.920625 [DEBUG] sofia.c:5390 Channel sofia/external/1000 at 138.132.105.40 entering state [received][100] 1970-01-06 08:19:06.920625 [DEBUG] sofia.c:5401 Remote SDP: v=0 o=TI-WLAN-PHONE 0 0 IN IP4 138.132.110.64 s=TI-WLAN-CALL c=IN IP4 138.132.110.64 t=0 0 m=audio 10136 RTP/AVP 8 18 101 102 a=rtpmap:8 PCMA/8000 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=rtpmap:102 G.729a/8000 1970-01-06 08:19:06.920625 [DEBUG] sofia.c:5594 (sofia/external/1000 at 138.132.105.40 ) State Change CS_NEW -> CS_INIT 1970-01-06 08:19:06.920625 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] 1970-01-06 08:19:06.920625 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1000 at 138.132.105.40 ) Running State Change CS_INIT 1970-01-06 08:19:06.920625 [DEBUG] switch_core_state_machine.c:401 (sofia/external/1000 at 138.132.105.40 ) State INIT 1970-01-06 08:19:06.920625 [DEBUG] mod_sofia.c:85 sofia/external/1000 at 138.132.105.40 SOFIA INIT 1970-01-06 08:19:06.920625 [DEBUG] mod_sofia.c:125 (sofia/external/1000 at 138.132.105.40 ) State Change CS_INIT -> CS_ROUTING 1970-01-06 08:19:06.920625 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] 1970-01-06 08:19:06.920625 [DEBUG] switch_core_state_machine.c:401 (sofia/external/1000 at 138.132.105.40 ) State INIT going to sleep 1970-01-06 08:19:06.920625 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1000 at 138.132.105.40 ) Running State Change CS_ROUTING 1970-01-06 08:19:06.920625 [DEBUG] switch_channel.c:1871 (sofia/external/1000 at 138.132.105.40 ) Callstate Change DOWN -> RINGING 1970-01-06 08:19:06.920625 [DEBUG] switch_core_state_machine.c:410 (sofia/external/1000 at 138.132.105.40 ) State ROUTING 1970-01-06 08:19:06.920625 [DEBUG] mod_sofia.c:148 sofia/external/1000 at 138.132.105.40 SOFIA ROUTING 1970-01-06 08:19:06.920625 [DEBUG] switch_core_state_machine.c:104 sofia/external/1000 at 138.132.105.40 Standard ROUTING 1970-01-06 08:19:06.920625 [INFO] mod_dialplan_xml.c:481 Processing 1000 <1000>->1001 in context public Dialplan: sofia/external/1000 at 138.132.105.40 parsing [public->dial_from_gateway1] continue=false Dialplan: sofia/external/1000 at 138.132.105.40 Regex (FAIL) [dial_from_gateway1] destination_number(1001) =~ /1000/ break=on-false Dialplan: sofia/external/1000 at 138.132.105.40 parsing [public->dial_from_gateway1] continue=false Dialplan: sofia/external/1000 at 138.132.105.40 Regex (PASS) [dial_from_gateway1] destination_number(1001) =~ /1001/ break=on-false Dialplan: sofia/external/1000 at 138.132.105.40 Action set(call_timeout=400) Dialplan: sofia/external/1000 at 138.132.105.40 Action set(inherit_codec=true) Dialplan: sofia/external/1000 at 138.132.105.40 Action bridge(sofia/external/1001 at 138.132.110.64:5070 ) 1970-01-06 08:19:06.980093 [DEBUG] switch_core_state_machine.c:154 (sofia/external/1000 at 138.132.105.40 ) State Change CS_ROUTING -> CS_EXECUTE 1970-01-06 08:19:06.980093 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] 1970-01-06 08:19:06.980093 [DEBUG] switch_core_state_machine.c:410 (sofia/external/1000 at 138.132.105.40 ) State ROUTING going to sleep 1970-01-06 08:19:06.980093 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1000 at 138.132.105.40 ) Running State Change CS_EXECUTE 1970-01-06 08:19:06.980093 [DEBUG] switch_core_state_machine.c:417 (sofia/external/1000 at 138.132.105.40 ) State EXECUTE 1970-01-06 08:19:06.980093 [DEBUG] mod_sofia.c:241 sofia/external/1000 at 138.132.105.40 SOFIA EXECUTE 1970-01-06 08:19:06.980093 [DEBUG] switch_core_state_machine.c:192 sofia/external/1000 at 138.132.105.40 Standard EXECUTE EXECUTE sofia/external/1000 at 138.132.105.40 set(call_timeout=400) 1970-01-06 08:19:06.980093 [DEBUG] mod_dptools.c:1263 sofia/external/1000 at 138.132.105.40 SET [call_timeout]=[400] EXECUTE sofia/external/1000 at 138.132.105.40 set(inherit_codec=true) 1970-01-06 08:19:06.980093 [DEBUG] mod_dptools.c:1263 sofia/external/1000 at 138.132.105.40 SET [inherit_codec]=[true] EXECUTE sofia/external/1000 at 138.132.105.40 bridge(sofia/external/1001 at 138.132.110.64:5070 ) 1970-01-06 08:19:07.000205 [DEBUG] switch_ivr_originate.c:1884 Parsing global variables 1970-01-06 08:19:07.000205 [NOTICE] switch_channel.c:920 New Channel sofia/external/1001 at 138.132.110.64:5070 [20710684-bb47-4d82-94ec-aca7960bdf05] 1970-01-06 08:19:07.000205 [DEBUG] mod_sofia.c:4539 (sofia/external/1001 at 138.132.110.64:5070 ) State Change CS_NEW -> CS_INIT 1970-01-06 08:19:07.000205 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/1001 at 138.132.110.64:5070 [BREAK] 1970-01-06 08:19:07.020085 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1001 at 138.132.110.64:5070 ) Running State Change CS_INIT 1970-01-06 08:19:07.020085 [DEBUG] switch_core_state_machine.c:401 (sofia/external/1001 at 138.132.110.64:5070 ) State INIT 1970-01-06 08:19:07.020085 [DEBUG] mod_sofia.c:85 sofia/external/1001 at 138.132.110.64:5070 SOFIA INIT 1970-01-06 08:19:07.041130 [DEBUG] mod_sofia.c:125 (sofia/external/1001 at 138.132.110.64:5070 ) State Change CS_INIT -> CS_ROUTING 1970-01-06 08:19:07.041130 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/1001 at 138.132.110.64:5070 [BREAK] 1970-01-06 08:19:07.041130 [DEBUG] switch_core_state_machine.c:401 (sofia/external/1001 at 138.132.110.64:5070 ) State INIT going to sleep 1970-01-06 08:19:07.041130 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1001 at 138.132.110.64:5070 ) Running State Change CS_ROUTING 1970-01-06 08:19:07.041130 [DEBUG] switch_channel.c:1871 (sofia/external/1001 at 138.132.110.64:5070 ) Callstate Change DOWN -> RINGING 1970-01-06 08:19:07.041130 [DEBUG] switch_core_state_machine.c:410 (sofia/external/1001 at 138.132.110.64:5070 ) State ROUTING 1970-01-06 08:19:07.041130 [DEBUG] mod_sofia.c:148 sofia/external/1001 at 138.132.110.64:5070 SOFIA ROUTING 1970-01-06 08:19:07.041130 [DEBUG] switch_ivr_originate.c:66 (sofia/external/1001 at 138.132.110.64:5070 ) State Change CS_ROUTING -> CS_CONSUME_MEDIA 1970-01-06 08:19:07.041130 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/1001 at 138.132.110.64:5070 [BREAK] 1970-01-06 08:19:07.041130 [DEBUG] switch_core_state_machine.c:410 (sofia/external/1001 at 138.132.110.64:5070 ) State ROUTING going to sleep 1970-01-06 08:19:07.041130 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1001 at 138.132.110.64:5070 ) Running State Change CS_CONSUME_MEDIA 1970-01-06 08:19:07.041130 [DEBUG] switch_core_state_machine.c:429 (sofia/external/1001 at 138.132.110.64:5070 ) State CONSUME_MEDIA 1970-01-06 08:19:07.041130 [DEBUG] switch_core_state_machine.c:429 (sofia/external/1001 at 138.132.110.64:5070 ) State CONSUME_MEDIA going to sleep send 1033 bytes to udp/[138.132.110.64]:5070 at 08:19:07.061497: ------------------------------------------------------------------------ INVITE sip:1001 at 138.132.110.64:5070 SIP/2.0 Via: SIP/2.0/UDP 138.132.105.40;rport;branch=z9hG4bKXty0Na85rgvNQ Max-Forwards: 69 From: "1000" >;tag=1Xy95SNcpKSSK To: > Call-ID: 78c3e14f-9ca8-1200-1ca0-0020da862374 CSeq: 30983549 INVITE Contact: > User-Agent: Italtel-WeGate/HW_V1.0.0/FW_V1.0.0/SW_V1.0.0 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY Supported: 100rel, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 254 X-FS-Support: update_display Remote-Party-ID: "1000" >;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 441041 441042 IN IP4 138.132.105.40 s=FreeSWITCH c=IN IP4 138.132.105.40 t=0 0 m=audio 20906 RTP/AVP 8 18 98 101 13 a=rtpmap:98 AMR-WB/16000 a=fmtp:98 octet-align=0 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ------------------------------------------------------------------------ 1970-01-06 08:19:07.080086 [DEBUG] switch_core_session.c:872 Send signal sofia/external/1001 at 138.132.110.64:5070 [BREAK] 1970-01-06 08:19:07.080086 [DEBUG] sofia.c:5390 Channel sofia/external/1001 at 138.132.110.64:5070 entering state [calling][0] recv 299 bytes from udp/[138.132.110.64]:5070 at 08:19:07.292884: ------------------------------------------------------------------------ SIP/2.0 100 Trying Via: SIP/2.0/UDP 138.132.105.40;rport;branch=z9hG4bKXty0Na85rgvNQ To: > From: "1000" >;tag=1Xy95SNcpKSSK Call-ID: 78c3e14f-9ca8-1200-1ca0-0020da862374 CSeq: 30983549 INVITE User Agent: TP2000 By JAB Content-Length: 0 ------------------------------------------------------------------------ recv 682 bytes from udp/[138.132.110.64]:5070 at 08:19:07.307137: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 138.132.105.40;rport;branch=z9hG4bKXty0Na85rgvNQ To: >;tag=beef9284 From: "1000" >;tag=1Xy95SNcpKSSK Call-ID: 78c3e14f-9ca8-1200-1ca0-0020da862374 CSeq: 30983549 INVITE Contact: > Allow: INVITE, ACK, PRACK, CANCEL, BYE, OPTIONS, MESSAGE, NOTIFY, UPDATE, REGISTER, INFO, REFER, SUBSCRIBE Require: 100rel RSeq: 18 Content-Type: application/sdp Content-Length: 178 v=0 o=TI-WLAN-PHONE 0 1 IN IP4 138.132.110.64 s=TI-WLAN-CALL c=IN IP4 138.132.110.64 t=0 0 m=audio 10136 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ 1970-01-06 08:19:07.300076 [DEBUG] switch_core_session.c:872 Send signal sofia/external/1001 at 138.132.110.64:5070 [BREAK] send 602 bytes to udp/[138.132.110.64]:5070 at 08:19:07.323009: ------------------------------------------------------------------------ PRACK sip:1001 at 138.132.110.64:5070 SIP/2.0 Via: SIP/2.0/UDP 138.132.105.40;rport;branch=z9hG4bKp8mU7r0cNSNpS Max-Forwards: 70 From: "1000" >;tag=1Xy95SNcpKSSK To: >;tag=beef9284 Call-ID: 78c3e14f-9ca8-1200-1ca0-0020da862374 CSeq: 30983550 PRACK Contact: > RAck: 18 30983549 INVITE User-Agent: Italtel-WeGate/HW_V1.0.0/FW_V1.0.0/SW_V1.0.0 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY Supported: 100rel, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 1970-01-06 08:19:07.321123 [DEBUG] switch_core_session.c:872 Send signal sofia/external/1001 at 138.132.110.64:5070 [BREAK] 1970-01-06 08:19:07.321123 [DEBUG] sofia.c:5390 Channel sofia/external/1001 at 138.132.110.64:5070 entering state [proceeding][183] 1970-01-06 08:19:07.321123 [DEBUG] sofia.c:5401 Remote SDP: v=0 o=TI-WLAN-PHONE 0 1 IN IP4 138.132.110.64 s=TI-WLAN-CALL c=IN IP4 138.132.110.64 t=0 0 m=audio 10136 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 1970-01-06 08:19:07.321123 [DEBUG] sofia.c:5516 [GIO] sofia_handle_sip_i_state() : 1970-01-06 08:19:07.321123 [DEBUG] sofia_glue.c:3904 [GIO] sofia_glue_tech_media() : 1970-01-06 08:19:07.321123 [DEBUG] sofia_glue.c:4304 [GIO] sofia_glue_negotiate_sdp() : 1970-01-06 08:19:07.321123 [DEBUG] sofia_glue.c:4768 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 1970-01-06 08:19:07.321123 [DEBUG] sofia_glue.c:4768 Audio Codec Compare [G729:18:8000:20:8000]/[G729:18:8000:20:8000] 1970-01-06 08:19:07.321123 [DEBUG] sofia_glue.c:2887 Set Codec sofia/external/1001 at 138.132.110.64:5070 G729/8000 20 ms 160 samples 8000 bits 1970-01-06 08:19:07.321123 [DEBUG] sofia_glue.c:4882 Set 2833 dtmf send payload to 101 1970-01-06 08:19:07.321123 [DEBUG] sofia_glue.c:3139 AUDIO RTP [sofia/external/1001 at 138.132.110.64:5070] 138.132.105.40 port 20906 -> 138.132.110.64 port 10136 codec: 18 ms: 20 1970-01-06 08:19:07.321123 [DEBUG] switch_rtp.c:1642 Starting timer [soft] 160 bytes per 20ms 1970-01-06 08:19:07.321123 [DEBUG] sofia_glue.c:3403 Set 2833 dtmf send payload to 101 1970-01-06 08:19:07.321123 [DEBUG] sofia_glue.c:3409 Set 2833 dtmf receive payload to 101 1970-01-06 08:19:07.321123 [NOTICE] sofia_glue.c:3915 Pre-Answer sofia/external/1001 at 138.132.110.64:5070! 1970-01-06 08:19:07.321123 [DEBUG] switch_channel.c:2917 (sofia/external/1001 at 138.132.110.64:5070 ) Callstate Change RINGING -> EARLY 1970-01-06 08:19:07.321123 [DEBUG] switch_ivr_originate.c:405 Setting codec string on sofia/external/1000 at 138.132.105.40 to G729 at 8000h@20i 1970-01-06 08:19:07.321123 [INFO] switch_ivr_originate.c:3215 Sending early media 1970-01-06 08:19:07.321123 [DEBUG] sofia_glue.c:3904 [GIO] sofia_glue_tech_media() : 1970-01-06 08:19:07.321123 [DEBUG] sofia_glue.c:4304 [GIO] sofia_glue_negotiate_sdp() : 1970-01-06 08:19:07.321123 [DEBUG] sofia_glue.c:4768 Audio Codec Compare [PCMA:8:8000:20:64000]/[G729:18:8000:20:8000] 1970-01-06 08:19:07.321123 [DEBUG] sofia_glue.c:4768 Audio Codec Compare [G729:18:8000:20:8000]/[G729:18:8000:20:8000] 1970-01-06 08:19:07.321123 [DEBUG] sofia_glue.c:2887 Set Codec sofia/external/1000 at 138.132.105.40 G729/8000 20 ms 160 samples 8000 bits 1970-01-06 08:19:07.321123 [DEBUG] sofia_glue.c:4889 Set 2833 dtmf send/recv payload to 101 1970-01-06 08:19:07.321123 [DEBUG] sofia_glue.c:3139 AUDIO RTP [sofia/external/1000 at 138.132.105.40] 138.132.105.40 port 21634 -> 138.132.110.64 port 10136 codec: 18 ms: 20 1970-01-06 08:19:07.321123 [DEBUG] switch_rtp.c:1642 Starting timer [soft] 160 bytes per 20ms 1970-01-06 08:19:07.367755 [DEBUG] sofia_glue.c:3403 Set 2833 dtmf send payload to 101 1970-01-06 08:19:07.367755 [DEBUG] sofia_glue.c:3409 Set 2833 dtmf receive payload to 101 1970-01-06 08:19:07.367755 [NOTICE] sofia_glue.c:3915 Pre-Answer sofia/external/1000 at 138.132.105.40! 1970-01-06 08:19:07.367755 [DEBUG] switch_channel.c:2917 (sofia/external/1000 at 138.132.105.40 ) Callstate Change RINGING -> EARLY 1970-01-06 08:19:07.380617 [DEBUG] mod_sofia.c:2500 Ring SDP: v=0 o=FreeSWITCH 0000440313 0000440314 IN IP4 138.132.105.40 s=FreeSWITCH c=IN IP4 138.132.105.40 t=0 0 m=audio 21634 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv send 1048 bytes to udp/[138.132.110.64]:5065 at 08:19:07.424043: ------------------------------------------------------------------------ SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 138.132.110.64:5065;branch=z9hG4bKxTP2KbyJab6373 From: "1000" >;tag=3bad6373 To: "1001" >;tag=8Q8emBX5ta0pH Call-ID: 63736d49df5d85548e08b348fa46b455 at 138.132.110.64 CSeq: 1 INVITE Contact: > RSeq: 1262621007 User-Agent: Italtel-WeGate/HW_V1.0.0/FW_V1.0.0/SW_V1.0.0 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY Require: 100rel Supported: 100rel, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session Content-Length: 245 Remote-Party-ID: "1001" >;party=calling;privacy=off;screen=no v=0 o=FreeSWITCH 440313 440314 IN IP4 138.132.105.40 s=FreeSWITCH c=IN IP4 138.132.105.40 t=0 0 m=audio 21634 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 1970-01-06 08:19:07.420102 [DEBUG] switch_core_session.c:872 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] recv 367 bytes from udp/[138.132.110.64]:5065 at 08:19:07.430920: ------------------------------------------------------------------------ PRACK sip:1001 at 138.132.105.40:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 138.132.110.64:5065;branch=z9hG4bKe27d4510 To: "1001" >;tag=8Q8emBX5ta0pH From: "1000" >;tag=3bad6373 Call-ID: 63736d49df5d85548e08b348fa46b455 at 138.132.110.64 CSeq: 2 PRACK Max-Forwards: 70 RAck: 1262621007 1 INVITE Content-Length: 0 ------------------------------------------------------------------------ send 564 bytes to udp/[138.132.110.64]:5065 at 08:19:07.432407: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 138.132.110.64:5065;branch=z9hG4bKe27d4510 From: "1000" >;tag=3bad6373 To: "1001" >;tag=8Q8emBX5ta0pH Call-ID: 63736d49df5d85548e08b348fa46b455 at 138.132.110.64 CSeq: 2 PRACK Contact: > User-Agent: Italtel-WeGate/HW_V1.0.0/FW_V1.0.0/SW_V1.0.0 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY Supported: 100rel, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------ 1970-01-06 08:19:07.420102 [DEBUG] switch_core_session.c:872 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] recv 339 bytes from udp/[138.132.110.64]:5070 at 08:19:07.434919: ------------------------------------------------------------------------ SIP/2.0 200 OK Call-ID: 78c3e14f-9ca8-1200-1ca0-0020da862374 Via: SIP/2.0/UDP 138.132.105.40;rport;branch=z9hG4bKp8mU7r0cNSNpS To: >;tag=beef9284 From: "1000" >;tag=1Xy95SNcpKSSK CSeq: 30983550 PRACK Contact: > Max-Forwards: 70 Content-Length: 0 ------------------------------------------------------------------------ 1970-01-06 08:19:07.420102 [DEBUG] switch_core_session.c:872 Send signal sofia/external/1001 at 138.132.110.64:5070 [BREAK] 1970-01-06 08:19:07.420102 [DEBUG] switch_channel.c:2959 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] 1970-01-06 08:19:07.420102 [DEBUG] sofia.c:5383 Channel sofia/external/1000 at 138.132.105.40 skipping state [early][183] 1970-01-06 08:19:07.420102 [DEBUG] switch_core_session.c:726 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] 1970-01-06 08:19:07.420102 [DEBUG] switch_ivr_originate.c:3269 Originate Resulted in Success: [sofia/external/1001 at 138.132.110.64:5070] 1970-01-06 08:19:07.440084 [DEBUG] switch_core_session.c:726 Send signal sofia/external/1001 at 138.132.110.64:5070 [BREAK] 1970-01-06 08:19:07.440084 [DEBUG] switch_core_session.c:726 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] 1970-01-06 08:19:07.440084 [DEBUG] switch_ivr_bridge.c:1270 (sofia/external/1001 at 138.132.110.64:5070 ) State Change CS_CONSUME_MEDIA -> CS_EXCHANGE_MEDIA 1970-01-06 08:19:07.440084 [DEBUG] switch_core_session.c:1177 Send signal sofia/external/1001 at 138.132.110.64:5070 [BREAK] 1970-01-06 08:19:07.440084 [DEBUG] switch_core_state_machine.c:362 (sofia/external/1001 at 138.132.110.64:5070 ) Running State Change CS_EXCHANGE_MEDIA 1970-01-06 08:19:07.440084 [DEBUG] switch_core_state_machine.c:420 (sofia/external/1001 at 138.132.110.64:5070 ) State EXCHANGE_MEDIA 1970-01-06 08:19:07.440084 [DEBUG] mod_sofia.c:574 SOFIA EXCHANGE_MEDIA recv 587 bytes from udp/[138.132.110.64]:5065 at 08:19:08.449739: ------------------------------------------------------------------------ UPDATE sip:1001 at 138.132.105.40:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 138.132.110.64:5065;branch=z9hG4bKe27d7918 To: "1001" >;tag=8Q8emBX5ta0pH From: "1000" >;tag=3bad6373 Call-ID: 63736d49df5d85548e08b348fa46b455 at 138.132.110.64 CSeq: 3 UPDATE Max-Forwards: 70 Contact: > Content-Type: application/sdp Content-Length: 171 v=0 o=TI-WLAN-PHONE 0 2 IN IP4 138.132.110.64 s=TI-WLAN-CALL c=IN IP4 0.0.0.0 t=0 0 m=audio 10136 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 ------------------------------------------------------------------------ send 885 bytes to udp/[138.132.110.64]:5065 at 08:19:08.465202: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 138.132.110.64:5065;branch=z9hG4bKe27d7918 From: "1000" >;tag=3bad6373 To: "1001" >;tag=8Q8emBX5ta0pH Call-ID: 63736d49df5d85548e08b348fa46b455 at 138.132.110.64 CSeq: 3 UPDATE Contact: > User-Agent: Italtel-WeGate/HW_V1.0.0/FW_V1.0.0/SW_V1.0.0 Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, PRACK, NOTIFY Supported: 100rel, precondition, path, replaces Content-Type: application/sdp Content-Disposition: session Content-Length: 257 v=0 o=FreeSWITCH 440313 440315 IN IP4 138.132.105.40 s=FreeSWITCH c=IN IP4 138.132.105.40 t=0 0 m=audio 21634 RTP/AVP 18 101 a=rtpmap:18 G729/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=recvonly a=silenceSupp:off - - - - a=ptime:20 ------------------------------------------------------------------------ 1970-01-06 08:19:08.460083 [DEBUG] switch_core_session.c:872 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] 1970-01-06 08:19:08.460083 [DEBUG] switch_core_session.c:872 Send signal sofia/external/1000 at 138.132.105.40 [BREAK] 1970-01-06 08:19:08.480076 [DEBUG] sofia.c:5390 Channel sofia/external/1000 at 138.132.105.40 entering state [early][200] 1970-01-06 08:19:08.480076 [DEBUG] sofia.c:5401 Remote SDP: v=0 o=TI-WLAN-PHONE 0 2 IN IP4 138.132.110.64 s=TI-WLAN-CALL c=IN IP4 0.0.0.0 t=0 0 m=audio 10136 RTP/AVP 18 101 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendonly Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/7a00f867/attachment-0001.html From avi at avimarcus.net Tue Dec 6 20:49:43 2011 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 6 Dec 2011 19:49:43 +0200 Subject: [Freeswitch-users] Identify the gateway In-Reply-To: <4EDE5418.5050703@btg.co.nz> References: <4EDD3F75.9080303@btg.co.nz> <4EDD613F.7060609@btg.co.nz> <4EDE5418.5050703@btg.co.nz> Message-ID: Maybe in testing, but in production you kinda need to know what you can change live... reloadxml does the dialplan, reload mod_x does that mod, and "sofia $profile killgw $gw" I think kills that gateway, then you can do a rescan.. Anyway, after the reload it still didn't work..? -Avi On Tue, Dec 6, 2011 at 7:42 PM, Jort Bloem wrote: > I fully shut it down freeswitch, and then started it up again (I've found > that the time wasted doing a complete shutdown is more than offset by the > hours you can waste trying to figure out why your changes don't take effect ;-) > ) > > > On 06/12/11 22:01, Avi Marcus wrote: > > Did you kill and restart the gateway after adding the variable? (Or > restart the entire interface, or restart FS?) > > -Avi > > > On Tue, Dec 6, 2011 at 2:26 AM, Jort Bloem wrote: > >> Hi Avi & Michael, >> >> Michael, I'm familiar with the info application, and have been using this >> to test my progress. It's gotten me a long way, but not past this hurdle. >> >> Avi, I tried that, or at least my interpretation of that, and it didn't >> work. >> >> I found the gateway in question - it's one of only two references to the >> trunk - in sip_profiles/. The declaration, including my addition of the >> variables clause, looks like this: >> >> >> >> >> >> >> >> >> >> >> >> >> I then called in via that interface, using a DDI, with the info >> application. I got lots of information, but neither foo nor bar. I did >> change the direction from "inbound" to "both", to make it as likely as >> possible to work. >> >> The log output I got had lots of stuff in it (see >> http://pastebin.freeswitch.org/17943 ) but no foo, no bar, and no >> trunk_1. It did have several occurrences of the ip address of talk2, >> 10.10.9.1, but that can change. I also see several references to the ip >> address (10.10.10.149) and sofia profile name (sipinterface_3), but all >> gateways will be on the same profile. >> >> Any further help would be appreciated. >> >> Jaybee- >> >> >> >> On 06/12/11 11:57, Avi Marcus wrote: >> >> >> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055508.html >> >> Hmm, gateways on the wiki should be cleaned up.. I couldn't find it on >> the wiki, only via google. >> >> -Avi >> >> >> On Tue, Dec 6, 2011 at 12:45 AM, Michael Collins wrote: >> >>> Try adding one of these to the dialplan that handles inbound calls: >>> >>> >>> >>> Then, watch the console while an inbound comes in and you'll see quite >>> a lot of information. Anything in that info dump is available to you for >>> "figuring out" what's going on. If you have any questions then drop that >>> info output to a pastebin and give us the URL here. >>> >>> -MC >>> >>> >>> On Mon, Dec 5, 2011 at 2:02 PM, Jort Bloem wrote: >>> >>>> Hi all, >>>> >>>> I have a setup with several gateways coming in on one profile. The >>>> Realms, unfortunately, are by DNS, and move from time to time. >>>> >>>> Is there any way to identify, in the dialplan, which gateway the call >>>> comes from, for incoming calls? >>>> >>>> I know I can get the ip address, but the ip address is subject to >>>> change. >>>> >>>> I know that the trunk provider can set various things, but I don't have >>>> control of the far end of the trunk. >>>> >>>> If there is a way to set variables on a per-gateway (NOT per-profile) >>>> basis, that would be perfect. >>>> >>>> "RTFM" gladly accepted with a hint about which part of which FM to R. >>>> >>>> -- >>>> Jaybee- >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/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 >> >> >> -- >> Jort Bloem >> >> >> Technical Engineer - Auckland >> Business Technology Group >> >> P: +64 9 5801374 x 9884 >> >> Sent from Mozilla Thunderbird >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/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 > > > -- > Jaybee > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/9a5964d6/attachment-0001.html From jort.bloem at btg.co.nz Tue Dec 6 21:33:46 2011 From: jort.bloem at btg.co.nz (Jort Bloem) Date: Wed, 07 Dec 2011 07:33:46 +1300 Subject: [Freeswitch-users] Identify the gateway In-Reply-To: References: <4EDD3F75.9080303@btg.co.nz> <4EDD613F.7060609@btg.co.nz> <4EDE5418.5050703@btg.co.nz> Message-ID: <4EDE600A.9090902@btg.co.nz> Yes, but I *am* testing [well, development] ;) - besides, before I put things into production, I test them in testing, so I already know it will work [at least, in theory]. After the restart of FS, it still didn't work. On 07/12/11 06:49, Avi Marcus wrote: > Maybe in testing, but in production you kinda need to know what you > can change live... reloadxml does the dialplan, reload mod_x does that > mod, and "sofia $profile killgw $gw" I think kills that gateway, then > you can do a rescan.. > > Anyway, after the reload it still didn't work..? > -Avi > > > On Tue, Dec 6, 2011 at 7:42 PM, Jort Bloem > wrote: > > I fully shut it down freeswitch, and then started it up again > (I've found that the time wasted doing a complete shutdown is more > than offset by the hours you can waste trying to figure out why > your changes don't take effect ;-) ) > > > On 06/12/11 22:01, Avi Marcus wrote: >> Did you kill and restart the gateway after adding the variable? >> (Or restart the entire interface, or restart FS?) >> >> -Avi >> >> >> On Tue, Dec 6, 2011 at 2:26 AM, Jort Bloem > > wrote: >> >> Hi Avi & Michael, >> >> Michael, I'm familiar with the info application, and have >> been using this to test my progress. It's gotten me a long >> way, but not past this hurdle. >> >> Avi, I tried that, or at least my interpretation of that, and >> it didn't work. >> >> I found the gateway in question - it's one of only two >> references to the trunk - in sip_profiles/. The declaration, >> including my addition of the variables clause, looks like this: >> >> >> >> >> >> >> >> >> >> >> >> >> I then called in via that interface, using a DDI, with the >> info application. I got lots of information, but neither foo >> nor bar. I did change the direction from "inbound" to "both", >> to make it as likely as possible to work. >> >> The log output I got had lots of stuff in it (see >> http://pastebin.freeswitch.org/17943 ) but no foo, no bar, >> and no trunk_1. It did have several occurrences of the ip >> address of talk2, 10.10.9.1, but that can change. I also see >> several references to the ip address (10.10.10.149) and sofia >> profile name (sipinterface_3), but all gateways will be on >> the same profile. >> >> Any further help would be appreciated. >> >> Jaybee- >> >> >> >> On 06/12/11 11:57, Avi Marcus wrote: >>> http://lists.freeswitch.org/pipermail/freeswitch-users/2010-March/055508.html >>> >>> Hmm, gateways on the wiki should be cleaned up.. I couldn't >>> find it on the wiki, only via google. >>> >>> -Avi >>> >>> >>> On Tue, Dec 6, 2011 at 12:45 AM, Michael Collins >>> > wrote: >>> >>> Try adding one of these to the dialplan that handles >>> inbound calls: >>> >>> >>> >>> Then, watch the console while an inbound comes in and >>> you'll see quite a lot of information. Anything in that >>> info dump is available to you for "figuring out" what's >>> going on. If you have any questions then drop that info >>> output to a pastebin and give us the URL here. >>> >>> -MC >>> >>> >>> On Mon, Dec 5, 2011 at 2:02 PM, Jort Bloem >>> > wrote: >>> >>> Hi all, >>> >>> I have a setup with several gateways coming in on >>> one profile. The >>> Realms, unfortunately, are by DNS, and move from >>> time to time. >>> >>> Is there any way to identify, in the dialplan, which >>> gateway the call >>> comes from, for incoming calls? >>> >>> I know I can get the ip address, but the ip address >>> is subject to change. >>> >>> I know that the trunk provider can set various >>> things, but I don't have >>> control of the far end of the trunk. >>> >>> If there is a way to set variables on a per-gateway >>> (NOT per-profile) >>> basis, that would be perfect. >>> >>> "RTFM" gladly accepted with a hint about which part >>> of which FM to R. >>> >>> -- >>> Jaybee- >>> >>> >>> _________________________________________________________________________ >>> 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 >> >> -- >> Jort Bloem >> >> >> Technical Engineer - Auckland >> Business Technology Group >> >> P:+64 9 5801374 x 9884 >> >> Sent from Mozilla Thunderbird >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > -- > Jaybee > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Jaybee- -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111207/864cc6b6/attachment-0001.html From msc at freeswitch.org Tue Dec 6 21:52:31 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 6 Dec 2011 10:52:31 -0800 Subject: [Freeswitch-users] pagd dynamic conference In-Reply-To: <1323157274.53780.YahooMailNeo@web65305.mail.ac2.yahoo.com> References: <1323157274.53780.YahooMailNeo@web65305.mail.ac2.yahoo.com> Message-ID: Hmmm, maybe the timeout is too short. Change the 2000 (2 seconds) to 7000 (7 seconds) and see if that makes any difference. -MC On Mon, Dec 5, 2011 at 11:41 PM, Rodney wrote: > >>> Michael, I tried what you suggest and it just plays conf-pin.wav then > goes right to bad-pin.wav then hangups without actually putting me in > conference.before i tried this i tried using "read" instead of pagd and it > accepted the pin and put me into the conference but didn't allow me any > conference options or the ability to back out to main ivr. I am still > stuck :( . my system is ivr based incoming calls only so this feature is > essential. all help is appreciated. > > > condition destination_number ^3001$ > action answer > action play_and_get_digits 4 4 3 # 2000 conf-pin.wav > conf-bad-pin.wav \d+ > action conference ${digits}-${domain}@default > > > > > > > > > > > > > > > > > > > > > > > > > > > I believe you just need to call play_and_get_digits (PAGD) prior to > dropping them into the conference. If I understand the question correctly, > the four-digits that the caller enters represents the "conference number" > or whatever you call it. You could do this: > > > > > naturally you'll need to read up on PAGD so that you understand what all > that stuff is doing. Also, be sure to specify real sound files, not the > fake ones that I used. The ${digits} value is what the caller actually > dials. In a production environment you'll need to handle the scenario where > the caller never actually enters a valid 4-digit number. > > -MC > > On Sun, Dec 4, 2011 at 11:44 AM, Rodney wrote: > > I have tried the nb_conference example in the default.xml but I think I am > doing something wrong. when i transfer to that extension it drops me into a > single static conference of the variable . what do i need to add or change > to prompt my callers transfered to the nb_conference to enter a 4 digit > conference number without pin that gives them the same options as my static > conferences on the default profile? I know play and get digits must be > involved but would appreciate the help. 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 > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/b3b1df24/attachment.html From msc at freeswitch.org Tue Dec 6 22:01:35 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 6 Dec 2011 11:01:35 -0800 Subject: [Freeswitch-users] How best to use event system for many originates with listeners In-Reply-To: <4FD95AAF-424A-4A9C-B037-511DFA32C129@apartmentlines.com> References: <4FD95AAF-424A-4A9C-B037-511DFA32C129@apartmentlines.com> Message-ID: Chad, What is your event listener? Just curious. It seems to me that you could do an inbound event socket to generate the call and then let that call hit the dialplan and send it to the socket app which in turn would send the call to your event listener. (I suppose you could even do something like "bgapi originate sofia/foo/bar at baz &socket("127.0.0.1:8084")" if you wanted to.) -MC On Tue, Dec 6, 2011 at 8:10 AM, Chad Phillips -- Apartment Lines < chad at apartmentlines.com> wrote: > I'm designing an application that will originate many calls via the event > system which I also want to attach event listeners to. i want each event > listener to only listen for events particular to each originated call. > > Since I want FreeSWITCH to control the call after I originate it, it seems > that I should use an inbound connection. The two possible workflows I see > are: > > a) Issue an originate command via inbound, then turn it into an > outbound-type connection via a 'myevents ' call (or 'handlecall' > using mod_erlang_event) once the channel is up. With this method, it seems > that I could miss critical channel events between the time the call is > originated and the listener is attached. > > b) Set up an inbound listener filtered on a pre-determined UUID for the > call, then originate the call with that UUID. This would seem to guarantee > that no events are missed by the listener, but I'm wondering about the > performance of having potentially hundreds of inbound connections with > filters going at the same time. > > So my questions are: > > 1) Is there a way to accomplish a) without potentially missing any > channel events? > > 2) Is b) vastly less efficient than a)? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/2ba48f42/attachment.html From acrow at integrafin.co.uk Tue Dec 6 23:42:34 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Tue, 06 Dec 2011 20:42:34 +0000 Subject: [Freeswitch-users] Created new profile for double-natted phones, calling these fails with USER_NOT_REGISTERED In-Reply-To: <201112051256.05883.justlikeef@gmail.com> References: <4EDCB468.2060001@integrafin.co.uk> <4EDCC4E5.803@integrafin.co.uk> <201112051255.33240.justlikeef@gmail.com> <201112051256.05883.justlikeef@gmail.com> Message-ID: <4EDE7E3A.3040301@integrafin.co.uk> On 05/12/11 17:56, Rob Hutton wrote: > > http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Presence > > > Rob, Indeed, that seems to have done the trick. Many thanks for that! Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/b0e47bd1/attachment-0001.html From admin at blindi.net Wed Dec 7 00:42:09 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Tue, 6 Dec 2011 22:42:09 +0100 (CET) Subject: [Freeswitch-users] lua error in session:transfer In-Reply-To: <4EDE1289.1060807@newpace.ca> References: <4EDE1289.1060807@newpace.ca> Message-ID: Hi Adam, thanks for your reply. It does not work. I set the line before the transfercommand: freeswitch.consoleLog("info", "Transfer t o target number: ".. target_number .. " XML default\n"); The value is entered correctly in the log. But the transfer is not working. I become the same error. Transfer can not handle the numbers of the variable. --------------- 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 avi at avimarcus.net Wed Dec 7 01:00:12 2011 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 7 Dec 2011 00:00:12 +0200 Subject: [Freeswitch-users] Voicemail Transcription Roundup Message-ID: I did some investigating and thought I'd share... Some info on pricing - free (google speech API) to $0.10/minute to approximately $2.80/minute are the options. See here: http://wiki.freeswitch.org/wiki/Transcribing_Voicemail And.. if you use fusionpbx, you can use the google speech API pretty easily with my patch from here: http://wiki.fusionpbx.com/index.php?title=Voicemail#Transcribing_Voicemails_with_google_speech_API Please share your experience or methods... you can update the wiki page. I'm not quite sure how to get the voicemail file to the transcription service with something other than the mailer script in the middle.. unless you are using ESL or polling the database of VMs and keep track if it's been processed yet. -Avi Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111207/b8d48764/attachment.html From tonybecq at yahoo.fr Wed Dec 7 02:07:31 2011 From: tonybecq at yahoo.fr (obbyone) Date: Tue, 6 Dec 2011 15:07:31 -0800 (PST) Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... In-Reply-To: <4EDE5144.4020703@integrafin.co.uk> References: <1322733452381-7049935.post@n2.nabble.com> <1322929911507-7058216.post@n2.nabble.com> <4EDA9484.9000000@integrafin.co.uk> <1323190206664-7067482.post@n2.nabble.com> <4EDE5144.4020703@integrafin.co.uk> Message-ID: <1323212851384-7068781.post@n2.nabble.com> Alex Crow wrote > > On 06/12/11 16:50, obbyone wrote: >> Hi, >> In fact, freeswitch is installed on a virtual serveur (VMWare) that has a >> fixed external IP. Every ATA's are behind a different ADSL modem same as >> if >> it was also on an external IP. I'm a newbie on freeswitch so is there >> someone who can take me by the hand and help me to handle this issue... >> >> Thanks > > Hi, > > Does the FS VM have a fixed IP or is it natted? > > I suspect the latter. I would follow the guide on the Wiki about "NAT > scenarios", specifically the one about double NAT. You will need to > create another profile for this: > http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios > > You also need to forward your chosen SIP port for the profile (eg 5090) > and the RTP ports (16384-32768) to the real IP of the FS VM. > > Cheers > > Alex > > > > -- > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > > "Transact" is operated by Integrated Financial Arrangements plc > Domain House, 5-7 Singer Street, London EC2A 4BQ > Tel: (020) 7608 4900 Fax: (020) 7608 5300 > (Registered office: as above; Registered in England and Wales under > number: 3727592) > Authorised and regulated by the Financial Services Authority (entered on > the FSA Register; number: 190856) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting@ > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at .freeswitch > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > In my case the FS VM has a fixed IP. Do I have to follow the same advice ? Thanks -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Freeswitch-installed-ATA-registered-but-no-call-are-possible-tp7049935p7068781.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Wed Dec 7 02:56:17 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 6 Dec 2011 15:56:17 -0800 Subject: [Freeswitch-users] lua error in session:transfer In-Reply-To: References: <4EDE1289.1060807@newpace.ca> Message-ID: On Tue, Dec 6, 2011 at 1:42 PM, Thomas Hoellriegel wrote: > Hi Adam, > thanks for your reply. > It does not work. > I set the line before the transfercommand: > freeswitch.consoleLog("info", "Transfer t > o target number: > ".. target_number .. " XML default\n"); > > The value is entered correctly in the log. > But the transfer is not working. > I become the same error. > Transfer can not handle the numbers of the variable. Thomas, Pastebin the whole call log, not just a single debug line, otherwise we are just guessing and wasting time. It is virtually impossible to figure this out based on the limited information you've supplied. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/f67dd395/attachment.html From turqmr2 at gmail.com Wed Dec 7 04:06:58 2011 From: turqmr2 at gmail.com (Jacob Smith) Date: Tue, 06 Dec 2011 20:06:58 -0500 Subject: [Freeswitch-users] Google Voice Message-ID: <4EDEBC32.9010203@gmail.com> I am near admitting defeat on this. If anyone can tell me what I am doing wrong, I would really appreciate it. I call my Google voice account, the FS phone (ext 1000) rings, I answer the phone but the calling phone keeps ringing and after 8 or so rings disconnects. I can make outgoing calls with no problems. Here is the full log, all I did was start it up/call/shutdown and replaced the calling phone number with "phone number" http://pastebin.freeswitch.org/17954 Here is my dial plan: http://pastebin.freeswitch.org/17956 Here is my Jingle profile: http://pastebin.freeswitch.org/17955 Thank you! Jacob From boris at tagnet.ru Wed Dec 7 05:49:03 2011 From: boris at tagnet.ru (Boris Kovalenko) Date: Wed, 07 Dec 2011 08:49:03 +0600 Subject: [Freeswitch-users] Does I need jitter buffer In-Reply-To: <1323186502931-7067246.post@n2.nabble.com> References: <4EDDAAA0.4040004@tagnet.ru> <1323186502931-7067246.post@n2.nabble.com> Message-ID: <4EDED41F.1050405@tagnet.ru> Hello! I have tested many phone models with same result. So my VOIP isn't properly functioning? Jeff, may You tell me what should I double check to eliminate echo? > Its not possible to have echo in a pure VOIP (properly functioning) > implementation. The only source of echo would be from acoustic echo on your > phones which is most likely what you are experiencing. A Jitter buffer wont > help this problem. > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Does-I-need-jitter-buffer-tp7065665p7067246.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 -- Regards, Boris From anthony.minessale at gmail.com Wed Dec 7 05:51:10 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 6 Dec 2011 20:51:10 -0600 Subject: [Freeswitch-users] Does I need jitter buffer In-Reply-To: <4EDED41F.1050405@tagnet.ru> References: <4EDDAAA0.4040004@tagnet.ru> <1323186502931-7067246.post@n2.nabble.com> <4EDED41F.1050405@tagnet.ru> Message-ID: Headsets, speaker phones, calling a remote party over high latency. On Tue, Dec 6, 2011 at 8:49 PM, Boris Kovalenko wrote: > Hello! > > I have tested many phone models with same result. So my VOIP isn't > properly functioning? Jeff, may You tell me what should I double check > to eliminate echo? > > > Its not possible to have echo in a pure VOIP (properly functioning) > > implementation. The only source of echo would be from acoustic echo on > your > > phones which is most likely what you are experiencing. A Jitter buffer > wont > > help this problem. > > > > -- > > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Does-I-need-jitter-buffer-tp7065665p7067246.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 > > > -- > Regards, > Boris > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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/20111206/eb5bc704/attachment-0001.html From boris at tagnet.ru Wed Dec 7 05:58:32 2011 From: boris at tagnet.ru (Boris Kovalenko) Date: Wed, 07 Dec 2011 08:58:32 +0600 Subject: [Freeswitch-users] Does I need jitter buffer In-Reply-To: References: <4EDDAAA0.4040004@tagnet.ru> <1323186502931-7067246.post@n2.nabble.com> <4EDED41F.1050405@tagnet.ru> Message-ID: <4EDED658.5060208@tagnet.ru> Hello! We are not using headsets and speaker phones. Only standard analog phones connected via Linksys or Audiocodes devices. I have tested LG, Nortel, Samsung, Panasonic phones with same result. The problem exists only _inside_ my network. So, for example: Audiocodes -- FreeSwitch -- Linksys ... echo present (often) Linksys -- FreeSwitch -- Linksys echo present (rarely) Audiocodes / Linksys -- Freeswitch -- Cisco 5350 -- PSTN (echo never present) Audiocodes / Linksys -- FreeSwitch -- remote calling party over internet (echo never present) > Headsets, speaker phones, calling a remote party over high latency. > > On Tue, Dec 6, 2011 at 8:49 PM, Boris Kovalenko > wrote: > > Hello! > > I have tested many phone models with same result. So my VOIP isn't > properly functioning? Jeff, may You tell me what should I double check > to eliminate echo? > > > Its not possible to have echo in a pure VOIP (properly functioning) > > implementation. The only source of echo would be from acoustic > echo on your > > phones which is most likely what you are experiencing. A Jitter > buffer wont > > help this problem. > > > > -- > > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Does-I-need-jitter-buffer-tp7065665p7067246.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 > > > -- > Regards, > Boris > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 -- Regards, Boris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111207/9414dad3/attachment.html From ljjimenez at gmail.com Wed Dec 7 06:03:22 2011 From: ljjimenez at gmail.com (Luis Jimenez) Date: Wed, 7 Dec 2011 03:03:22 +0000 Subject: [Freeswitch-users] Does I need jitter buffer In-Reply-To: <4EDED658.5060208@tagnet.ru> References: <4EDDAAA0.4040004@tagnet.ru> <1323186502931-7067246.post@n2.nabble.com> <4EDED41F.1050405@tagnet.ru> <4EDED658.5060208@tagnet.ru> Message-ID: <1240910882-1323227002-cardhu_decombobulator_blackberry.rim.net-660486311-@b27.c27.bise6.blackberry> What about network cards on the server Luis Jimenez -----Original Message----- From: Boris Kovalenko Sender: freeswitch-users-bounces at lists.freeswitch.org Date: Wed, 07 Dec 2011 08:58:32 To: FreeSWITCH Users Help Reply-To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Does I need jitter buffer _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From brian at freeswitch.org Wed Dec 7 06:08:07 2011 From: brian at freeswitch.org (Brian West) Date: Tue, 6 Dec 2011 21:08:07 -0600 Subject: [Freeswitch-users] Google Voice In-Reply-To: <4EDEBC32.9010203@gmail.com> References: <4EDEBC32.9010203@gmail.com> Message-ID: <8EABCA23-C257-4078-B712-BC428ABD40E3@freeswitch.org> Try doing an ANSWER before you ring the phones :P /b On Dec 6, 2011, at 7:06 PM, Jacob Smith wrote: > I am near admitting defeat on this. If anyone can tell me what I am > doing wrong, I would really appreciate it. > > I call my Google voice account, the FS phone (ext 1000) rings, I answer > the phone but the calling phone keeps ringing and after 8 or so rings > disconnects. I can make outgoing calls with no problems. > > Here is the full log, all I did was start it up/call/shutdown and > replaced the calling phone number with "phone number" > http://pastebin.freeswitch.org/17954 > > Here is my dial plan: > http://pastebin.freeswitch.org/17956 > > Here is my Jingle profile: > http://pastebin.freeswitch.org/17955 > > Thank you! > Jacob -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/0e258d05/attachment.html From boris at tagnet.ru Wed Dec 7 06:08:44 2011 From: boris at tagnet.ru (Boris Kovalenko) Date: Wed, 07 Dec 2011 09:08:44 +0600 Subject: [Freeswitch-users] Does I need jitter buffer In-Reply-To: <1240910882-1323227002-cardhu_decombobulator_blackberry.rim.net-660486311-@b27.c27.bise6.blackberry> References: <4EDDAAA0.4040004@tagnet.ru> <1323186502931-7067246.post@n2.nabble.com> <4EDED41F.1050405@tagnet.ru> <4EDED658.5060208@tagnet.ru> <1240910882-1323227002-cardhu_decombobulator_blackberry.rim.net-660486311-@b27.c27.bise6.blackberry> Message-ID: <4EDED8BC.1090707@tagnet.ru> Hello! Intel PRO/1000 Network adapter. No errors on interface. > What about network cards on the server > > Luis Jimenez > > -----Original Message----- > From: Boris Kovalenko > Sender: freeswitch-users-bounces at lists.freeswitch.org > Date: Wed, 07 Dec 2011 08:58:32 > To: FreeSWITCH Users Help > Reply-To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] Does I need jitter buffer > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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, Boris From notlikeme75 at yahoo.com Wed Dec 7 06:25:36 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 6 Dec 2011 19:25:36 -0800 (PST) Subject: [Freeswitch-users] pagd dynamic conference In-Reply-To: References: Message-ID: <1323228336.27976.YahooMailNeo@web65301.mail.ac2.yahoo.com> >>>> I changed it to 10000 and 15000 with same results. it does send me to "conference" but not with any numbers. seems the read feature worked without givng me conf options but pagd doesnt really work at all. i must still be missing something. Hmmm, maybe the timeout is too short. Change the 2000 (2 seconds) to 7000 (7 seconds) and see if that makes any difference. -MC On Mon, Dec 5, 2011 at 11:41 PM, Rodney wrote: >>> Michael, I tried what you suggest and it just plays conf-pin.wav then goes right to bad-pin.wav then hangups without actually putting me in conference.before i tried this i tried using "read" instead of pagd and it accepted the pin and put me into the conference but didn't allow me any conference options or the ability to back out to main ivr.? I am still stuck :( . my system is ivr based incoming calls only so this feature is essential. all help is appreciated. > > >?condition??? ? destination_number??? ? ^3001$??? ? >? action??? ? answer??? ? ? >? action??? ? play_and_get_digits??? ? 4 4 3 # 2000 conf-pin.wav conf-bad-pin.wav \d+?? >? action??? ? conference??? ? ${digits}-${domain}@default > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > >I believe you just need to call play_and_get_digits (PAGD) prior to dropping them into the conference. If I understand the question correctly, the four-digits that the caller enters represents the "conference number" or whatever you call it. You could do this: > > > > > > >naturally you'll need to read up on PAGD so that you understand what all that stuff is doing. Also, be sure to specify real sound files, not the fake ones that I used. The ${digits} value is what the caller actually dials. In a production environment you'll need to handle the scenario where the caller never actually enters a valid 4-digit number. > > >-MC > > >On Sun, Dec 4, 2011 at 11:44 AM, Rodney wrote: > >I have tried the nb_conference example in the default.xml but I think I am doing something wrong. when i transfer to that extension it drops me into a single static conference of the variable . what do i need to add or change to prompt my callers transfered to the nb_conference to enter a 4 digit conference number without pin that gives them the same options as my static conferences on the default profile? I know play and get digits must be involved but would appreciate the help. 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 >> >> > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > Chad, What is your event listener? Just curious. It seems to me that you could do an inbound event socket to generate the call and then let that call hit the dialplan and send it to the socket app which in turn would send the call to your event listener. (I suppose you could even do something like "bgapi originate sofia/foo/bar at baz &socket("127.0.0.1:8084")" if you wanted to.) -MC On Tue, Dec 6, 2011 at 8:10 AM, Chad Phillips -- Apartment Lines wrote: I'm designing an application that will originate many calls via the event system which I also want to attach event listeners to. ?i want each event listener to only listen for events particular to each originated call. > >Since I want FreeSWITCH to control the call after I originate it, it seems that I should use an inbound connection. ?The two possible workflows I see are: > >?a) Issue an originate command via inbound, then turn it into an outbound-type connection via a 'myevents ' call (or 'handlecall' using mod_erlang_event) once the channel is up. ?With this method, it seems that I could miss critical channel events between the time the call is originated and the listener is attached. > >?b) Set up an inbound listener filtered on a pre-determined UUID for the call, then originate the call with that UUID. ?This would seem to guarantee that no events are missed by the listener, but I'm wondering about the performance of having potentially hundreds of inbound connections with filters going at the same time. > >So my questions are: > >?1) Is there a way to accomplish a) without potentially missing any channel events? > >?2) Is b) vastly less efficient than a)? > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > On 05/12/11 17:56, Rob Hutton wrote: >http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Presence > > > Rob, Indeed, that seems to have done the trick. Many thanks for that! Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/ec200bcc/attachment-0001.html From turqmr2 at gmail.com Wed Dec 7 07:11:31 2011 From: turqmr2 at gmail.com (Jacob Smith) Date: Tue, 06 Dec 2011 23:11:31 -0500 Subject: [Freeswitch-users] Google Voice In-Reply-To: <8EABCA23-C257-4078-B712-BC428ABD40E3@freeswitch.org> References: <4EDEBC32.9010203@gmail.com> <8EABCA23-C257-4078-B712-BC428ABD40E3@freeswitch.org> Message-ID: <4EDEE773.4020200@gmail.com> In an effort to further the earned :P, do you mind telling me your answer in a painfully specific way? I tried putting "" before, then after the transfer line in my dial plan but both just kept my phone from ringing. I assumed that answering the phone would trigger some kind of answer command. On 12/06/2011 10:08 PM, Brian West wrote: > Try doing an ANSWER before you ring the phones :P > > /b > > On Dec 6, 2011, at 7:06 PM, Jacob Smith wrote: > >> I am near admitting defeat on this. If anyone can tell me what I am >> doing wrong, I would really appreciate it. >> >> I call my Google voice account, the FS phone (ext 1000) rings, I answer >> the phone but the calling phone keeps ringing and after 8 or so rings >> disconnects. I can make outgoing calls with no problems. >> >> Here is the full log, all I did was start it up/call/shutdown and >> replaced the calling phone number with "phone number" >> http://pastebin.freeswitch.org/17954 >> >> Here is my dial plan: >> http://pastebin.freeswitch.org/17956 >> >> Here is my Jingle profile: >> http://pastebin.freeswitch.org/17955 >> >> Thank you! >> Jacob > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111206/6c3ba329/attachment.html From ryan at kaevee.com Wed Dec 7 11:49:57 2011 From: ryan at kaevee.com (Ryan V) Date: Wed, 7 Dec 2011 14:19:57 +0530 Subject: [Freeswitch-users] Need dial plan for Analogue FreeTDM Message-ID: Hi, We have 1 PRI and 3 Analog lines. We are using Sangoma A200 with 4FXS and 4FXO ports. Also, we have a Sangoma A101 card. Incoming calls are working fine. But, we are unable to dial out. Here is our dial plan Here are the logs from mod_freetdm. 2011-12-07 14:03:54.429502 [DEBUG] mod_freetdm.c:1227 Connect outbound channel FreeTDM/3:1/9449905000 2011-12-07 14:03:54.429502 [DEBUG] mod_freetdm.c:1236 (FreeTDM/3:1/9449905000) State Change CS_NEW -> CS_INIT 2011-12-07 14:03:54.429502 [DEBUG] mod_freetdm.c:1253 Attached session 32ad9cea-20ae-11e1-b313-21737ba18370 to channel 3:1 2011-12-07 14:03:54.929503 [DEBUG] mod_freetdm.c:435 (FreeTDM/3:1/9449905000) State Change CS_INIT -> CS_ROUTING 2011-12-07 14:03:54.929503 [DEBUG] mod_freetdm.c:458 FreeTDM/3:1/9449905000 CHANNEL ROUTING 2011-12-07 14:03:59.489509 [DEBUG] mod_freetdm.c:1992 got FXO sig 3:1 [STOP] 2011-12-07 14:03:59.489509 [NOTICE] mod_freetdm.c:2013 Hangup FreeTDM/3:1/9449905000 [CS_CONSUME_MEDIA] [NORMAL_CIRCUIT_CONGESTION] I don't know where we are making mistake. Thanks, Ryan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111207/67e9ba9a/attachment.html From a.afzali2003 at gmail.com Wed Dec 7 12:12:48 2011 From: a.afzali2003 at gmail.com (afshin afzali) Date: Wed, 7 Dec 2011 12:42:48 +0330 Subject: [Freeswitch-users] Compile error after upgrade Message-ID: Hi Guys, After making current, I got compile error as bellow: make[2]: Entering directory `/usr/local/src/freeswitch' mkdir .libs Compiling src/switch_apr.c ... In file included from /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:101, from ./src/include/private/switch_core_pvt.h:35, from src/switch_apr.c:37: /usr/local/src/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:145: error: expected declaration specifiers or '...' before 'tz_t' make[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1 make[2]: Leaving directory `/usr/local/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [current] Error 2 Appreciate, -- afshin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111207/da894d72/attachment.html From acrow at integrafin.co.uk Wed Dec 7 14:35:49 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Wed, 07 Dec 2011 11:35:49 +0000 Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... In-Reply-To: <1323212851384-7068781.post@n2.nabble.com> References: <1322733452381-7049935.post@n2.nabble.com> <1322929911507-7058216.post@n2.nabble.com> <4EDA9484.9000000@integrafin.co.uk> <1323190206664-7067482.post@n2.nabble.com> <4EDE5144.4020703@integrafin.co.uk> <1323212851384-7068781.post@n2.nabble.com> Message-ID: <4EDF4F95.8020507@integrafin.co.uk> On 06/12/11 23:07, obbyone wrote: > Alex Crow wrote >> On 06/12/11 16:50, obbyone wrote: >>> Hi, >>> In fact, freeswitch is installed on a virtual serveur (VMWare) that has a >>> fixed external IP. Every ATA's are behind a different ADSL modem same as >>> if >>> it was also on an external IP. I'm a newbie on freeswitch so is there >>> someone who can take me by the hand and help me to handle this issue... >>> >>> Thanks >> Hi, >> >> Does the FS VM have a fixed IP or is it natted? >> >> I suspect the latter. I would follow the guide on the Wiki about "NAT >> scenarios", specifically the one about double NAT. You will need to >> create another profile for this: >> http://wiki.freeswitch.org/wiki/General_NAT_example_scenarios >> >> You also need to forward your chosen SIP port for the profile (eg 5090) >> and the RTP ports (16384-32768) to the real IP of the FS VM. >> >> Cheers >> >> Alex >> >> >> >> -- >> This message is intended only for the addressee and may contain >> confidential information. Unless you are that person, you may not >> disclose its contents or use it in any way and are requested to delete >> the message along with any attachments and notify us immediately. >> >> "Transact" is operated by Integrated Financial Arrangements plc >> Domain House, 5-7 Singer Street, London EC2A 4BQ >> Tel: (020) 7608 4900 Fax: (020) 7608 5300 >> (Registered office: as above; Registered in England and Wales under >> number: 3727592) >> Authorised and regulated by the Financial Services Authority (entered on >> the FSA Register; number: 190856) >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting@ >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at .freeswitch >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > In my case the FS VM has a fixed IP. Do I have to follow the same advice ? > > Thanks Sorry, I was unclear. I meant a private (RFC1918) IP. If it does, you will have to follow the double-nat scenario (unless the routing on VMWare (is it ESXi?) supports uPNP or NAT-PMP). If you can give the VM a *public* IP, you don't need to do that, then you only have to worry about NAT traversal at the ATA end. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From beppe.grillo at gmail.com Wed Dec 7 17:09:03 2011 From: beppe.grillo at gmail.com (Beppe Grillo) Date: Wed, 7 Dec 2011 15:09:03 +0100 Subject: [Freeswitch-users] FS supports RFC 5626 Message-ID: Hy all, FreeSWITCH supports RFC 5626 ? I FS with the last git. Thanks, Giuseppe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111207/d9931b98/attachment.html From tha_tux at hotmail.com Wed Dec 7 17:15:16 2011 From: tha_tux at hotmail.com (Tux Tux) Date: Wed, 7 Dec 2011 15:15:16 +0100 Subject: [Freeswitch-users] Call Detail Record via Event Socket Layer Message-ID: Hi, Is it possible to request the CDR via a ESL event? I found the possibilities to save it to disk and/or database but not to get it via the ESL. Maybe this is because the CDR can grow large in size, but I want to be certain. Thanks, Nico -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111207/4dfd20d9/attachment.html From anthony.minessale at gmail.com Wed Dec 7 20:18:53 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 7 Dec 2011 11:18:53 -0600 Subject: [Freeswitch-users] Does I need jitter buffer In-Reply-To: <4EDED658.5060208@tagnet.ru> References: <4EDDAAA0.4040004@tagnet.ru> <1323186502931-7067246.post@n2.nabble.com> <4EDED41F.1050405@tagnet.ru> <4EDED658.5060208@tagnet.ru> Message-ID: The analog phones are a common cause of echo. Look for settings on the device to use an echo canceler. On Tue, Dec 6, 2011 at 8:58 PM, Boris Kovalenko wrote: > Hello! > > We are not using headsets and speaker phones. Only standard analog > phones connected via Linksys or Audiocodes devices. I have tested LG, > Nortel, Samsung, Panasonic phones with same result. The problem exists only > _inside_ my network. So, for example: > > Audiocodes -- FreeSwitch -- Linksys ... echo present (often) > Linksys -- FreeSwitch -- Linksys echo present (rarely) > Audiocodes / Linksys -- Freeswitch -- Cisco 5350 -- PSTN (echo never > present) > Audiocodes / Linksys -- FreeSwitch -- remote calling party over internet > (echo never present) > > > > > Headsets, speaker phones, calling a remote party over high latency. > > On Tue, Dec 6, 2011 at 8:49 PM, Boris Kovalenko wrote: > >> Hello! >> >> I have tested many phone models with same result. So my VOIP isn't >> properly functioning? Jeff, may You tell me what should I double check >> to eliminate echo? >> >> > Its not possible to have echo in a pure VOIP (properly functioning) >> > implementation. The only source of echo would be from acoustic echo on >> your >> > phones which is most likely what you are experiencing. A Jitter buffer >> wont >> > help this problem. >> > >> > -- >> > View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Does-I-need-jitter-buffer-tp7065665p7067246.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 >> >> >> -- >> Regards, >> Boris >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://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.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 > > > > -- > Regards, > Boris > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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/20111207/d4163b71/attachment-0001.html From msc at freeswitch.org Wed Dec 7 20:32:50 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 7 Dec 2011 09:32:50 -0800 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today Message-ID: Hello all! We are having a nice conference call today. First off, Brian West has a few items to discuss with the community. After that, I will be doing a brief primer on using the FreeSWITCH event socket and ESL. Agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_12_07 Talk to you soon! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111207/6b21d9e0/attachment.html From jerry.richards at teotech.com Wed Dec 7 22:12:01 2011 From: jerry.richards at teotech.com (Jerry Richards) Date: Wed, 7 Dec 2011 11:12:01 -0800 Subject: [Freeswitch-users] 183 Session Progress Missing Video Codecs Message-ID: <2BF7FB90DF25EA4485949F3AF2B9D696343AC74FF3@VA3DIAXVS351.RED001.local> Using Nov 28 Freeswitch, if I make a Bria video call that includes H.263, H.263-1998 and H.264 in the caller's INVITE, Freeswitch sends 183 Session Progress back to caller with H263/90000 only. I am running will all default configs, except I added "H263-1998,H263,H264" to the list of global_codecs and outbound codecs in vars.xml. I'm thinking this might be a bug? Thanks, Jerry -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111207/b7e00f22/attachment.html From philq at qsystemsengineering.com Wed Dec 7 23:49:16 2011 From: philq at qsystemsengineering.com (Phil Quesinberry) Date: Wed, 07 Dec 2011 15:49:16 -0500 Subject: [Freeswitch-users] Can't quite get call screening to work Message-ID: <013301ccb521$b1e4eca0$15aec5e0$@com> I'm trying to use the call screening example in the wiki and can't get FS to play the caller's name back to the destination extension. example here: http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_13:_Call_Scr eening The initial announcement asking for the caller's name works fine: EXECUTE sofia/internal/102 at 192.168.1.6:5060 phrase(voicemail_record_name) 2011-12-07 14:52:13.715494 [DEBUG] mod_dptools.c:2362 Execute voicemail_record_name() lang 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:67 No language specified - Using [en] 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:244 Handle play-file:[voicemail/vm-record_name1.wav] (en:en) 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1 channels 20ms 2011-12-07 14:52:18.695891 [DEBUG] switch_ivr_play_say.c:1672 done playing file /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_name1.wav EXECUTE sofia/internal/102 at 192.168.1.6:5060 playback(tone_stream://%(500, 0, 640)) 2011-12-07 14:52:18.799791 [DEBUG] switch_ivr_play_say.c:1302 Codec Activated L16 at 8000hz 1 channels 20ms 2011-12-07 14:52:19.299814 [DEBUG] switch_ivr_play_say.c:1672 done playing file tone_stream://%(500, 0, 640) EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(playback_terminators=#*0123456789) 2011-12-07 14:52:19.299814 [DEBUG] mod_dptools.c:1263 sofia/internal/102 at 192.168.1.6:5060 SET [playback_terminators]=[#*0123456789] Then the caller's name is recorded, and I've verified that the recording is indeed saved in /tmp: EXECUTE sofia/internal/102 at 192.168.1.6:5060 record(/tmp/102-name.wav 7 200 2) 2011-12-07 14:52:19.299814 [DEBUG] switch_ivr_play_say.c:585 Raw Codec Activated 2011-12-07 14:52:19.299814 [DEBUG] switch_core_codec.c:116 sofia/internal/102 at 192.168.1.6:5060 Push codec L16:70 . EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(group_confirm_key=1) 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 sofia/internal/102 at 192.168.1.6:5060 SET [group_confirm_key]=[1] EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(fail_on_single_reject=true) 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 sofia/internal/102 at 192.168.1.6:5060 SET [fail_on_single_reject]=[true] EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(group_confirm_file=phrase:screen_confirm:/tmp/102-name.wav) 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 sofia/internal/102 at 192.168.1.6:5060 SET [group_confirm_file]=[phrase:screen_confirm:/tmp/102-name.wav] EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(continue_on_fail=true) 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 sofia/internal/102 at 192.168.1.6:5060 SET [continue_on_fail]=[true] EXECUTE sofia/internal/102 at 192.168.1.6:5060 bridge(user/102) . Then when attempting to play back the Output from the console showing the error is here: 2011-12-07 14:52:26.348406 [ERR] switch_ivr_play_say.c:142 Can't find macro screen_confirm. 2011-12-07 14:52:26.348406 [WARNING] switch_ivr_play_say.c:339 Macro [screen_confirm]: '/tmp/102-name.wav' did not match any patterns 2011-12-07 14:52:26.348406 [ERR] switch_ivr_originate.c:219 sofia/internal/sip:102 at 192.168.1.4:5060 Error Playing File! The call goes right to voicemail once the destination extension attempts to answer it. Where are these macros supposed to be stored? Somewhere under /usr/local/freeswitch/conf/lang/en? Do I need to create a macro for screen_confirm or is it just named incorrectly or in the wrong place? Thanks, Phil Quesinberry Q Systems Engineering, Inc. Electronic Controls and Embedded Systems Development (410) 969-8002 http://www.qsystemsengineering.com From don at confignet.com Thu Dec 8 01:07:03 2011 From: don at confignet.com (Don Coe) Date: Wed, 7 Dec 2011 14:07:03 -0800 Subject: [Freeswitch-users] Where do I find the ANI in SIP packets? Message-ID: I have a couple of carriers that terminate calls for me. Most send my ANI and it shows up, for instance, if I call my cell. However, some carriers are telling me that I am not sending ANI and on those calls I am seeing unknown numbers on my cell. How can I verify I am actually sending ANI and where would I find the ANI in the SIP packets? --Don From msc at freeswitch.org Thu Dec 8 01:51:45 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 7 Dec 2011 14:51:45 -0800 Subject: [Freeswitch-users] Can't quite get call screening to work In-Reply-To: <013301ccb521$b1e4eca0$15aec5e0$@com> References: <013301ccb521$b1e4eca0$15aec5e0$@com> Message-ID: it looks like the person who posted that example did not post their sample phrase macro file. However, do a git pull... commit 9ea3ce666fa7f021b5c2a7e2fbe153eb351c5734 Author: Michael S Collins Date: Wed Dec 7 14:49:16 2011 -0800 config: add screen_confirm macro to lang/en/ivr/sounds.xml snag that config file and drop it into conf/lang/en/ivr/ and then reloadxml. I did this on the fly w/o testing so be sure to test it thoroughly to make sure it works! Also, be sure to use the full path name to the sound file that you are playing back. -MC On Wed, Dec 7, 2011 at 12:49 PM, Phil Quesinberry < philq at qsystemsengineering.com> wrote: > I'm trying to use the call screening example in the wiki and can't get FS > to > play the caller's name back to the destination extension. > example here: > > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_13:_Call_Scr > eening > > The initial announcement asking for the caller's name works fine: > EXECUTE sofia/internal/102 at 192.168.1.6:5060 phrase(voicemail_record_name) > 2011-12-07 14:52:13.715494 [DEBUG] mod_dptools.c:2362 Execute > voicemail_record_name() lang > 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:67 No language > specified - Using [en] > 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:244 Handle > play-file:[voicemail/vm-record_name1.wav] (en:en) > 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:1302 Codec > Activated L16 at 8000hz 1 channels 20ms > 2011-12-07 14:52:18.695891 [DEBUG] switch_ivr_play_say.c:1672 done playing > file > /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_name1.wav > EXECUTE sofia/internal/102 at 192.168.1.6:5060 playback(tone_stream://%(500, > 0, > 640)) > 2011-12-07 14:52:18.799791 [DEBUG] switch_ivr_play_say.c:1302 Codec > Activated L16 at 8000hz 1 channels 20ms > 2011-12-07 14:52:19.299814 [DEBUG] switch_ivr_play_say.c:1672 done playing > file tone_stream://%(500, 0, 640) > EXECUTE sofia/internal/102 at 192.168.1.6:5060 > set(playback_terminators=#*0123456789) > 2011-12-07 14:52:19.299814 [DEBUG] mod_dptools.c:1263 > sofia/internal/102 at 192.168.1.6:5060 SET > [playback_terminators]=[#*0123456789] > > Then the caller's name is recorded, and I've verified that the recording is > indeed saved in /tmp: > EXECUTE sofia/internal/102 at 192.168.1.6:5060 record(/tmp/102-name.wav 7 200 > 2) > 2011-12-07 14:52:19.299814 [DEBUG] switch_ivr_play_say.c:585 Raw Codec > Activated > 2011-12-07 14:52:19.299814 [DEBUG] switch_core_codec.c:116 > sofia/internal/102 at 192.168.1.6:5060 Push codec L16:70 > . > EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(group_confirm_key=1) > 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 > sofia/internal/102 at 192.168.1.6:5060 SET [group_confirm_key]=[1] > EXECUTE sofia/internal/102 at 192.168.1.6:5060set(fail_on_single_reject=true) > 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 > sofia/internal/102 at 192.168.1.6:5060 SET [fail_on_single_reject]=[true] > EXECUTE sofia/internal/102 at 192.168.1.6:5060 > set(group_confirm_file=phrase:screen_confirm:/tmp/102-name.wav) > 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 > sofia/internal/102 at 192.168.1.6:5060 SET > [group_confirm_file]=[phrase:screen_confirm:/tmp/102-name.wav] > EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(continue_on_fail=true) > 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 > sofia/internal/102 at 192.168.1.6:5060 SET [continue_on_fail]=[true] > EXECUTE sofia/internal/102 at 192.168.1.6:5060 bridge(user/102) > . > Then when attempting to play back the Output from the console showing the > error is here: > 2011-12-07 14:52:26.348406 [ERR] switch_ivr_play_say.c:142 Can't find macro > screen_confirm. > 2011-12-07 14:52:26.348406 [WARNING] switch_ivr_play_say.c:339 Macro > [screen_confirm]: '/tmp/102-name.wav' did not match any patterns > 2011-12-07 14:52:26.348406 [ERR] switch_ivr_originate.c:219 > sofia/internal/sip:102 at 192.168.1.4:5060 Error Playing File! > > The call goes right to voicemail once the destination extension attempts to > answer it. > > Where are these macros supposed to be stored? Somewhere under > /usr/local/freeswitch/conf/lang/en? Do I need to create a macro for > screen_confirm or is it just named incorrectly or in the wrong place? > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111207/6f15e517/attachment-0001.html From djbinter at gmail.com Thu Dec 8 03:33:06 2011 From: djbinter at gmail.com (DJB International) Date: Wed, 7 Dec 2011 16:33:06 -0800 Subject: [Freeswitch-users] Compile error after upgrade In-Reply-To: References: Message-ID: make spandsp-reconf && make mod_spandsp && make install -djbinter On Wed, Dec 7, 2011 at 1:12 AM, afshin afzali wrote: > Hi Guys, > > After making current, I got compile error as bellow: > > make[2]: Entering directory `/usr/local/src/freeswitch' > mkdir .libs > Compiling src/switch_apr.c ... > In file included from > /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:101, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch_apr.c:37: > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:145: error: > expected declaration specifiers or '...' before 'tz_t' > make[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1 > make[2]: Leaving directory `/usr/local/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/usr/local/src/freeswitch' > make: *** [current] Error 2 > > Appreciate, > -- afshin > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111207/6adaa995/attachment.html From msc at freeswitch.org Thu Dec 8 03:51:49 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 7 Dec 2011 16:51:49 -0800 Subject: [Freeswitch-users] Call Detail Record via Event Socket Layer In-Reply-To: References: Message-ID: well, actually it is possible because in the upcoming FS cookbook I have this exact recipe! You'll have to buy the book to get the full monty, but here's the example script from that recipe, which will also be freely downloadable from Packt's web site once the CB is done: #!/usr/bin/perl # handle_cdr.pl # Connect to event socket, listen for CHANNEL_HANGUP_COMPLETE events # Uses event data to create custom CDRs use strict; use warnings; use lib '/usr/src/freeswitch.git/libs/esl/perl'; use ESL; my $host = "localhost"; my $port = "8021"; my $pass = "ClueCon"; my $con = new ESL::ESLconnection($host, $port, $pass); if ( ! $con ) { die "Unable to establish connection to FreeSWITCH.\n"; } ## Listen for events, filter in only CHANNEL_HANGUP_COMPLETE $con->events('plain','all'); $con->filter('Event-Name','CHANNEL_HANGUP_COMPLETE'); print "Connected to FreeSWITCH $host:$port and waiting for events...\n\n"; while (1) { my @raw_data = split "\n",$e->serialize(); my %cdr; foreach my $item ( @raw_data ) { #print "$item\n"; my ($header, $value) = split ': ', $item; $header =~ s/^variable_//; $cdr{$header} = $value } # %cdr contains a complete list of channel variables print "New CDR: "; print $cdr{uuid} . ', ' . $cdr{direction} . ', '; print $cdr{answer_epoch} . ', ' . $cdr{end_epoch} . ', '; print $cdr{hangup_cause} . "\n"; } -MC On Wed, Dec 7, 2011 at 6:15 AM, Tux Tux wrote: > Hi, > > Is it possible to request the CDR via a ESL event? > > I found the possibilities to save it to disk and/or database but not to > get it via the ESL. > Maybe this is because the CDR can grow large in size, but I want to be > certain. > > Thanks, > Nico > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111207/7a8b79d8/attachment.html From anthony.minessale at gmail.com Thu Dec 8 04:15:58 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 7 Dec 2011 19:15:58 -0600 Subject: [Freeswitch-users] Call Detail Record via Event Socket Layer In-Reply-To: References: Message-ID: Wait, there's more! if you have channel variable hangup_complete_with_xml=true , then the body of that event will be a full XML CDR On Wed, Dec 7, 2011 at 6:51 PM, Michael Collins wrote: > well, actually it is possible because in the upcoming FS cookbook I have > this exact recipe! You'll have to buy the book to get the full monty, but > here's the example script from that recipe, which will also be freely > downloadable from Packt's web site once the CB is done: > > #!/usr/bin/perl > > > # handle_cdr.pl > > > # Connect to event socket, listen for CHANNEL_HANGUP_COMPLETE events > > > # Uses event data to create custom CDRs > > > use strict; > use warnings; > use lib '/usr/src/freeswitch.git/libs/esl/perl'; > use ESL; > my $host = "localhost"; > my $port = "8021"; > my $pass = "ClueCon"; > my $con = new ESL::ESLconnection($host, $port, $pass); > if ( ! $con ) { > die "Unable to establish connection to FreeSWITCH.\n"; > } > ## Listen for events, filter in only CHANNEL_HANGUP_COMPLETE > > > $con->events('plain','all'); > $con->filter('Event-Name','CHANNEL_HANGUP_COMPLETE'); > print "Connected to FreeSWITCH $host:$port and waiting for events...\n\n"; > while (1) { > my @raw_data = split "\n",$e->serialize(); > my %cdr; > foreach my $item ( @raw_data ) { > #print "$item\n"; > > > my ($header, $value) = split ': ', $item; > $header =~ s/^variable_//; > $cdr{$header} = $value > } > # %cdr contains a complete list of channel variables > print "New CDR: "; > print $cdr{uuid} . ', ' . $cdr{direction} . ', '; > print $cdr{answer_epoch} . ', ' . $cdr{end_epoch} . ', '; > print $cdr{hangup_cause} . "\n"; > } > > > -MC > > On Wed, Dec 7, 2011 at 6:15 AM, Tux Tux wrote: > >> Hi, >> >> Is it possible to request the CDR via a ESL event? >> >> I found the possibilities to save it to disk and/or database but not to >> get it via the ESL. >> Maybe this is because the CDR can grow large in size, but I want to be >> certain. >> >> Thanks, >> Nico >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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/20111207/7b7de0f0/attachment-0001.html From msc at freeswitch.org Thu Dec 8 04:33:11 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 7 Dec 2011 17:33:11 -0800 Subject: [Freeswitch-users] Call Detail Record via Event Socket Layer In-Reply-To: References: Message-ID: Nice! I'll add that to my recipe. :) -MC On Wed, Dec 7, 2011 at 5:15 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Wait, there's more! > > if you have channel variable hangup_complete_with_xml=true , then the body > of that event will be a full XML CDR > > > > On Wed, Dec 7, 2011 at 6:51 PM, Michael Collins wrote: > >> well, actually it is possible because in the upcoming FS cookbook I have >> this exact recipe! You'll have to buy the book to get the full monty, but >> here's the example script from that recipe, which will also be freely >> downloadable from Packt's web site once the CB is done: >> >> #!/usr/bin/perl >> >> >> # handle_cdr.pl >> >> >> # Connect to event socket, listen for CHANNEL_HANGUP_COMPLETE events >> >> >> # Uses event data to create custom CDRs >> >> >> use strict; >> use warnings; >> use lib '/usr/src/freeswitch.git/libs/esl/perl'; >> use ESL; >> my $host = "localhost"; >> my $port = "8021"; >> my $pass = "ClueCon"; >> my $con = new ESL::ESLconnection($host, $port, $pass); >> if ( ! $con ) { >> die "Unable to establish connection to FreeSWITCH.\n"; >> } >> ## Listen for events, filter in only CHANNEL_HANGUP_COMPLETE >> >> >> $con->events('plain','all'); >> $con->filter('Event-Name','CHANNEL_HANGUP_COMPLETE'); >> print "Connected to FreeSWITCH $host:$port and waiting for events...\n\n"; >> while (1) { >> my @raw_data = split "\n",$e->serialize(); >> my %cdr; >> foreach my $item ( @raw_data ) { >> #print "$item\n"; >> >> >> my ($header, $value) = split ': ', $item; >> $header =~ s/^variable_//; >> $cdr{$header} = $value >> } >> # %cdr contains a complete list of channel variables >> print "New CDR: "; >> print $cdr{uuid} . ', ' . $cdr{direction} . ', '; >> print $cdr{answer_epoch} . ', ' . $cdr{end_epoch} . ', '; >> print $cdr{hangup_cause} . "\n"; >> } >> >> >> -MC >> >> On Wed, Dec 7, 2011 at 6:15 AM, Tux Tux wrote: >> >>> Hi, >>> >>> Is it possible to request the CDR via a ESL event? >>> >>> I found the possibilities to save it to disk and/or database but not to >>> get it via the ESL. >>> Maybe this is because the CDR can grow large in size, but I want to be >>> certain. >>> >>> Thanks, >>> Nico >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://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/20111207/5edc40d1/attachment.html From gabe at gundy.org Thu Dec 8 09:03:24 2011 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 7 Dec 2011 23:03:24 -0700 Subject: [Freeswitch-users] Freeswitch installed, ATA registered but no call are possible... In-Reply-To: <1322929911507-7058216.post@n2.nabble.com> References: <1322733452381-7049935.post@n2.nabble.com> <1322929911507-7058216.post@n2.nabble.com> Message-ID: On Sat, Dec 3, 2011 at 9:31 AM, obbyone wrote: > The trace I gave was not a good one. The one I show now Is a good sample. > Any attempt to dial "1001" results only in a response by voicemail. Any > answer ? > > Thanks > > /The trace :/ Let's not forget that there is a pastebin for these types of things. It kind makes it hard to help when the email is cluttered with logs and SIP traces. Best, Gabe From avi at avimarcus.net Thu Dec 8 12:19:53 2011 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 8 Dec 2011 11:19:53 +0200 Subject: [Freeswitch-users] Call Detail Record via Event Socket Layer In-Reply-To: References: Message-ID: I'd just like to mention.. there's already several CDR handler mods written in C that may suit your needs already.. http://wiki.freeswitch.org/wiki/Cdr -Avi On Thu, Dec 8, 2011 at 3:33 AM, Michael Collins wrote: > Nice! I'll add that to my recipe. :) > -MC > > > On Wed, Dec 7, 2011 at 5:15 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> Wait, there's more! >> >> if you have channel variable hangup_complete_with_xml=true , then the >> body of that event will be a full XML CDR >> >> >> >> On Wed, Dec 7, 2011 at 6:51 PM, Michael Collins wrote: >> >>> well, actually it is possible because in the upcoming FS cookbook I have >>> this exact recipe! You'll have to buy the book to get the full monty, but >>> here's the example script from that recipe, which will also be freely >>> downloadable from Packt's web site once the CB is done: >>> >>> #!/usr/bin/perl >>> >>> >>> # handle_cdr.pl >>> >>> >>> # Connect to event socket, listen for CHANNEL_HANGUP_COMPLETE events >>> >>> >>> # Uses event data to create custom CDRs >>> >>> >>> use strict; >>> use warnings; >>> use lib '/usr/src/freeswitch.git/libs/esl/perl'; >>> use ESL; >>> my $host = "localhost"; >>> my $port = "8021"; >>> my $pass = "ClueCon"; >>> my $con = new ESL::ESLconnection($host, $port, $pass); >>> if ( ! $con ) { >>> die "Unable to establish connection to FreeSWITCH.\n"; >>> } >>> ## Listen for events, filter in only CHANNEL_HANGUP_COMPLETE >>> >>> >>> $con->events('plain','all'); >>> $con->filter('Event-Name','CHANNEL_HANGUP_COMPLETE'); >>> print "Connected to FreeSWITCH $host:$port and waiting for >>> events...\n\n"; >>> while (1) { >>> my @raw_data = split "\n",$e->serialize(); >>> my %cdr; >>> foreach my $item ( @raw_data ) { >>> #print "$item\n"; >>> >>> >>> my ($header, $value) = split ': ', $item; >>> $header =~ s/^variable_//; >>> $cdr{$header} = $value >>> } >>> # %cdr contains a complete list of channel variables >>> print "New CDR: "; >>> print $cdr{uuid} . ', ' . $cdr{direction} . ', '; >>> print $cdr{answer_epoch} . ', ' . $cdr{end_epoch} . ', '; >>> print $cdr{hangup_cause} . "\n"; >>> } >>> >>> >>> -MC >>> >>> On Wed, Dec 7, 2011 at 6:15 AM, Tux Tux wrote: >>> >>>> Hi, >>>> >>>> Is it possible to request the CDR via a ESL event? >>>> >>>> I found the possibilities to save it to disk and/or database but not to >>>> get it via the ESL. >>>> Maybe this is because the CDR can grow large in size, but I want to be >>>> certain. >>>> >>>> Thanks, >>>> Nico >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111208/559fa49d/attachment-0001.html From michael.knop at hcu-hamburg.de Thu Dec 8 13:12:17 2011 From: michael.knop at hcu-hamburg.de (michael knop) Date: Thu, 08 Dec 2011 11:12:17 +0100 Subject: [Freeswitch-users] Compile error after upgrade In-Reply-To: References: Message-ID: <4EE08D81.8060301@hcu-hamburg.de> On SUSE SLES 11 SP1 make spandsp-reconf ends with the following error: make[3]: *** [libspandsp.la] Error 1 make[3]: Leaving directory `/usr/local/src/freeswitch/libs/spandsp/src' make[2]: *** [all] Error 2 make[2]: Leaving directory `/usr/local/src/freeswitch/libs/spandsp/src' make[1]: *** [all-recursive] Error 1 make[1]: Leaving directory `/usr/local/src/freeswitch/libs/spandsp' make: *** [spandsp-reconf] Error 2 /micha Am 08.12.2011 01:33, schrieb DJB International: > make spandsp-reconf && make mod_spandsp && make install > > -djbinter > > > > On Wed, Dec 7, 2011 at 1:12 AM, afshin afzali > wrote: > > Hi Guys, > > After making current, I got compile error as bellow: > > make[2]: Entering directory `/usr/local/src/freeswitch' > mkdir .libs > Compiling src/switch_apr.c ... > In file included from > /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:101, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch_apr.c:37: > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:145: > error: expected declaration specifiers or '...' before 'tz_t' > make[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1 > make[2]: Leaving directory `/usr/local/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/usr/local/src/freeswitch' > make: *** [current] Error 2 > > Appreciate, > -- afshin > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 kaevee.com Thu Dec 8 14:44:38 2011 From: ryan at kaevee.com (Ryan V) Date: Thu, 8 Dec 2011 17:14:38 +0530 Subject: [Freeswitch-users] Sangoma A200 FXO Outgoing Problem Message-ID: Hi, I have configured A200 and able to receive incoming calls. I am unable to dial out. Here are the logs. 2011-12-08 16:39:15.990089 [DEBUG] ftmod_analog.c:62 [s3c1][2:1] Changed state from DOWN to DIALING 2011-12-08 16:39:15.990089 [DEBUG] ftmod_analog.c:439 [s3c1][2:1] ANALOG CHANNEL thread starting. 2011-12-08 16:39:15.990089 [DEBUG] ftmod_analog.c:459 [s3c1][2:1] Initialized DTMF detection 2011-12-08 16:39:15.990089 [DEBUG] ftmod_analog.c:640 [s3c1][2:1] Completed state change from DOWN to DIALING in 0ms 2011-12-08 16:39:15.990089 [DEBUG] ftmod_analog.c:646 [s3c1][2:1] Executing state handler on 3:1 for DIALING 2011-12-08 16:39:21.009503 [DEBUG] ftmod_analog.c:505 [s3c1][2:1] Changed state from DIALING to BUSY 2011-12-08 16:39:21.030044 [DEBUG] ftmod_analog.c:640 [s3c1][2:1] Completed state change from DIALING to BUSY in 20ms 2011-12-08 16:39:21.030044 [DEBUG] ftmod_analog.c:646 [s3c1][2:1] Executing state handler on 3:1 for BUSY 2011-12-08 16:39:21.030044 [DEBUG] ftmod_analog.c:802 [s3c1][2:1] Changed state from BUSY to DOWN 2011-12-08 16:39:21.049928 [DEBUG] ftmod_analog.c:640 [s3c1][2:1] Completed state change from BUSY to DOWN in 20ms 2011-12-08 16:39:21.049928 [DEBUG] ftmod_analog.c:646 [s3c1][2:1] Executing state handler on 3:1 for DOWN 2011-12-08 16:39:21.049928 [DEBUG] ftmod_analog.c:944 [s3c1][2:1] Going onhook2011-12-08 16:39:21.049928 [DEBUG] switch_core_state_machine.c:362 (FreeTDM/3:1/9449905000) Running State Change CS_HANGUP Here is our dial plan I suspect FS is not detecting the dial tone on analogue channel. BTW, outgoing on our digital span works fine. Any suggestions? Thanks, Ryan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111208/5b012f94/attachment.html From jeff at jefflenk.com Thu Dec 8 17:31:19 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Thu, 8 Dec 2011 06:31:19 -0800 (PST) Subject: [Freeswitch-users] Sangoma A200 FXO Outgoing Problem In-Reply-To: References: Message-ID: <1323354678820-7074474.post@n2.nabble.com> One thing to check is tones.conf. Do you have that configured correctly? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Sangoma-A200-FXO-Outgoing-Problem-tp7074029p7074474.html Sent from the freeswitch-users mailing list archive at Nabble.com. From Nabble_01394 at slickdeals.endjunk.com Thu Dec 8 17:57:29 2011 From: Nabble_01394 at slickdeals.endjunk.com (mazilo) Date: Thu, 8 Dec 2011 06:57:29 -0800 (PST) Subject: [Freeswitch-users] spandsp/src/dtmf.c: 'duration' isn't a membership of 'dtmf_rx_state_t'? Message-ID: <1323356249725-7074568.post@n2.nabble.com> In recent commit (70c1c03c93f172ea18e33350897d58e3a82028d0), I noticed SPANDSP has undergone some heavy overhaul. Of particularly interesting to me is the src/dtmf.c file (which also got patched). ATM, I just couldn't locate a reference within the scope of spandsp source on the 'duration' membership in 'dtmf_rx_state_t' as shown in the following excerpt of src/dtmf.c patch: Anyone? ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/spandsp-src-dtmf-c-duration-isn-t-a-membership-of-dtmf-rx-state-t-tp7074568p7074568.html Sent from the freeswitch-users mailing list archive at Nabble.com. From steveu at coppice.org Thu Dec 8 19:21:46 2011 From: steveu at coppice.org (Steve Underwood) Date: Fri, 09 Dec 2011 00:21:46 +0800 Subject: [Freeswitch-users] spandsp/src/dtmf.c: 'duration' isn't a membership of 'dtmf_rx_state_t'? In-Reply-To: <1323356249725-7074568.post@n2.nabble.com> References: <1323356249725-7074568.post@n2.nabble.com> Message-ID: <4EE0E41A.4050406@coppice.org> On 12/08/2011 10:57 PM, mazilo wrote: > In recent commit (70c1c03c93f172ea18e33350897d58e3a82028d0), I noticed > SPANDSP has undergone some heavy overhaul. Of particularly interesting to me > is the src/dtmf.c file (which also got patched). ATM, I just couldn't locate > a reference within the scope of spandsp source on the 'duration' membership > in 'dtmf_rx_state_t' as shown in the following excerpt of src/dtmf.c patch: Its there. Steve From ryan at kaevee.com Thu Dec 8 19:47:00 2011 From: ryan at kaevee.com (Ryan V) Date: Thu, 8 Dec 2011 22:17:00 +0530 Subject: [Freeswitch-users] Sangoma A200 FXO Outgoing Problem In-Reply-To: <1323354678820-7074474.post@n2.nabble.com> References: <1323354678820-7074474.post@n2.nabble.com> Message-ID: On Thu, Dec 8, 2011 at 8:01 PM, Jeff Lenk wrote: > One thing to check is tones.conf. Do you have that configured correctly? > There is a tones.conf in config directory. There are entries for India. I have't changed the stock tones.conf file. Thanks, Ryan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111208/41db72a7/attachment.html From davidwaf at gmail.com Thu Dec 8 22:47:35 2011 From: davidwaf at gmail.com (David Wafula) Date: Thu, 8 Dec 2011 21:47:35 +0200 Subject: [Freeswitch-users] =?windows-1252?q?=2E=2E/libs/spandsp/src/spand?= =?windows-1252?q?sp/t4=5Ftx=2Eh=3A145=3A_error=3A_expected_declara?= =?windows-1252?q?tion_specifiers_or_=91=2E=2E=2E=92_before_=91tz?= =?windows-1252?q?=5Ft=92?= Message-ID: Hi all, make current is failing: In file included from /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:101, from ./src/include/private/switch_core_pvt.h:35, from src/switch_apr.c:37: /usr/local/src/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:145: error: expected declaration specifiers or ?...? before ?tz_t? make[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1 make[2]: Leaving directory `/usr/local/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/local/src/freeswitch' make: *** [current] Error 2 -- David Wafula -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111208/bdf7f4a3/attachment.html From msc at freeswitch.org Thu Dec 8 23:06:02 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 8 Dec 2011 12:06:02 -0800 Subject: [Freeswitch-users] =?windows-1252?q?=2E=2E/libs/spandsp/src/spand?= =?windows-1252?q?sp/t4=5Ftx=2Eh=3A145=3A_error=3A_expected_declara?= =?windows-1252?q?tion_specifiers_or_=91=2E=2E=2E=92_before_=91tz?= =?windows-1252?q?=5Ft=92?= In-Reply-To: References: Message-ID: This was answered several times in IRC and on the ML. Look at yesterday's messages, specifically for one from djbinter. -MC On Thu, Dec 8, 2011 at 11:47 AM, David Wafula wrote: > Hi all, > make current is failing: > > In file included from > /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:101, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch_apr.c:37: > /usr/local/src/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:145: error: > expected declaration specifiers or ?...? before ?tz_t? > make[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1 > make[2]: Leaving directory `/usr/local/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/usr/local/src/freeswitch' > make: *** [current] Error 2 > > -- > David Wafula > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111208/65ed4280/attachment-0001.html From msc at freeswitch.org Thu Dec 8 23:08:30 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 8 Dec 2011 12:08:30 -0800 Subject: [Freeswitch-users] Answerconfirmation with ivr not working In-Reply-To: References: Message-ID: So you're trying to have your group confirm file be an ivr? Is that even possible? If so, it's news to me. Where did you hear that this was possible? -MC On Sun, Dec 4, 2011 at 8:14 PM, Thomas Hoellriegel wrote: > Hi all, > i create a callscreening extension with: > > > > > > > > > My ivrmenu: > greet-long="$${base_dir}/**sounds/callscreen/**callscrrenopts.wav" > greet-short="$${base_dir}/**sounds/callscreen/**callscrrenopts.wav" > invalid-sound="$${base_dir}/**sounds/ivr/falsche-eingabe.**wav" > exit-sound="$${base_dir}/**sounds/ivr/goodbye.wav" > confirm-attempts="3" > timeout="10000" > inter-digit-timeout="3000" > max-failures="2" > max-timeouts="3" > digit-len="1"> > action="menu-exec-app" digits="1" param="transfer -both 7676 XML default"/> > > > Glows when the phone and I answer it, then I hear my outgoing message. > But i can.t transfer both channels to the conference. > i here my menuprompt in the conference. But not the caller. > > What is wrong? > Can your help please? > thanks. > > > --------------- > Du kannst mich jederzeit kostenlos per Festnetz erreichen unter: > http://www.blindi.net/callback > homepage: http://www.blindi.net > blinde-misc mailingliste f?r blinde. anmeldung unter: > http://www.blindi.net/mailman/**listinfo/blinde-misc > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111208/c2856142/attachment.html From davidwaf at gmail.com Thu Dec 8 23:16:06 2011 From: davidwaf at gmail.com (David Wafula) Date: Thu, 8 Dec 2011 22:16:06 +0200 Subject: [Freeswitch-users] =?windows-1252?q?=2E=2E/libs/spandsp/src/spand?= =?windows-1252?q?sp/t4=5Ftx=2Eh=3A145=3A_error=3A_expected_declara?= =?windows-1252?q?tion_specifiers_or_=91=2E=2E=2E=92_before_=91tz?= =?windows-1252?q?=5Ft=92?= In-Reply-To: References: Message-ID: Thanks. must have missed it: make spandsp-reconf && make mod_spandsp && make install regards. 2011/12/8 Michael Collins > This was answered several times in IRC and on the ML. Look at yesterday's > messages, specifically for one from djbinter. > > -MC > > On Thu, Dec 8, 2011 at 11:47 AM, David Wafula wrote: > >> Hi all, >> make current is failing: >> >> In file included from >> /usr/local/src/freeswitch/libs/spandsp/src/spandsp.h:101, >> from ./src/include/private/switch_core_pvt.h:35, >> from src/switch_apr.c:37: >> /usr/local/src/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:145: error: >> expected declaration specifiers or ?...? before ?tz_t? >> make[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1 >> make[2]: Leaving directory `/usr/local/src/freeswitch' >> make[1]: *** [all] Error 2 >> make[1]: Leaving directory `/usr/local/src/freeswitch' >> make: *** [current] Error 2 >> >> -- >> David Wafula >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111208/b3fc07b2/attachment.html From brian at freeswitch.org Thu Dec 8 23:43:59 2011 From: brian at freeswitch.org (Brian West) Date: Thu, 8 Dec 2011 14:43:59 -0600 Subject: [Freeswitch-users] Sangoma A200 FXO Outgoing Problem In-Reply-To: References: Message-ID: <14AA2B32-ACE0-429F-AB60-2E9CAF98BD96@freeswitch.org> its freetdm/spanname/channelnumber/numbertodial You can't do a on those as far as I knew. /b On Dec 8, 2011, at 5:44 AM, Ryan V wrote: > > Here is our dial plan > > > expression="^9(\d{10})$"> > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111208/4c1bf789/attachment.html From fs-list at communicatefreely.net Fri Dec 9 00:35:20 2011 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Thu, 08 Dec 2011 16:35:20 -0500 Subject: [Freeswitch-users] BLF gets confused with multiple calls Message-ID: <4EE12D98.8090100@communicatefreely.net> Hello, I'm having all sorts of problems with BLFs not being in the correct state. It works fine in some cases, but others are wrong. Here's what I can see: Single endpoint, single call, everything works fine. Light flashes on ring, goes steady on answer, goes out on hangup. Here's where it gets tricky: If there are two phones registered to the extension, it flashes when they ring, but then goes out when one of them answers. If either phone places an outgoing call, the lamp comes on. If a single phone gets a second call, their lamp flashes again, but then goes out when they answer the second call. I'm constantly getting complaints from users where the lamps are stuck on. It happens more often when they have a lot of phones in ring groups. The lamps work fine for ring groups - they all flash, and whomever picks up the call stays steady while the rest go out. Is there anything I can do to get freeswitch to base the state on whether or not that user has any active calls, rather than just what the last thing that the phone did was? This happens on every model of Aastra phone, and I have all of them. I haven't had a chance to try it yet on Polycom. I'm running FreeSWITCH Version 1.0.head (git-7531fed 2011-08-17 11-27-20 -0500) Any ideas would be more than welcome. -Tim From brian at freeswitch.org Fri Dec 9 00:48:30 2011 From: brian at freeswitch.org (Brian West) Date: Thu, 8 Dec 2011 15:48:30 -0600 Subject: [Freeswitch-users] BLF gets confused with multiple calls In-Reply-To: <4EE12D98.8090100@communicatefreely.net> References: <4EE12D98.8090100@communicatefreely.net> Message-ID: On Dec 8, 2011, at 3:35 PM, Tim St. Pierre wrote: > > Here's where it gets tricky: > > If there are two phones registered to the extension, it flashes when > they ring, but then goes out when one of them answers. > If either phone places an outgoing call, the lamp comes on. I'm going to guess you're not using the user/ from the default directory like the default where it sets the presence ID? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111208/554c83c7/attachment.html From fs-list at communicatefreely.net Fri Dec 9 01:23:21 2011 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Thu, 08 Dec 2011 17:23:21 -0500 Subject: [Freeswitch-users] BLF gets confused with multiple calls In-Reply-To: References: <4EE12D98.8090100@communicatefreely.net> Message-ID: <4EE138D9.7010306@communicatefreely.net> No, we use multiple domains here. The directory is generated by an xml_curl script that sends it back dynamically. In general, BLFs work great, but they seem to be based on the last message that thing the phone did (it just answered a call, or it just hung up a call), as opposed to whether or not it has a channel connected to it or not, and that is where things are going wrong, is when it just hung up, but there is still one more channel left, so it's really still in use. The dial strings are all user/exten at our.domain where our.domain is the same domain that both registrations and subscriptions use. I couldn't get presence to work at all without that. What is the relationship between the user/ domain, the subscription domain, and the registration domain vs. invite domains and whatever else. Should this work correctly, and if so, where should I look in terms of debugging it? Thanks! Brian West wrote: > > On Dec 8, 2011, at 3:35 PM, Tim St. Pierre wrote: >> >> Here's where it gets tricky: >> >> If there are two phones registered to the extension, it flashes when >> they ring, but then goes out when one of them answers. >> If either phone places an outgoing call, the lamp comes on. > > I'm going to guess you're not using the user/ from the default > directory like the default where it sets the presence ID? > > ------------------------------------------------------------------------ > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Fri Dec 9 01:25:03 2011 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Thu, 08 Dec 2011 17:25:03 -0500 Subject: [Freeswitch-users] Need help In-Reply-To: References: <1317356803.21942.YahooMailNeo@web161017.mail.bf1.yahoo.com> Message-ID: <4EE1393F.8080703@communicatefreely.net> That's about as simple as you can get. Have your choice of web script generate this little piece of dial plan based on the content of your database and you have a really easy to maintain system. Sam Govind wrote: > I could only think of these two. I was almost trying the same thing on > my setup some time ago. > > 1- Use redirect application at the end of your prompts back to the > same server. So A-party and C-party being the same server should > bridge the same call back to one. > 2- Use hangup application with some custom header as flag and tell > your partner company to proceed on the basis of that flag. > > hope anybody else comes up with some brilliantly simple idea. > > Regards, > -Sammy > > On Fri, Sep 30, 2011 at 9:26 AM, Sam > wrote: > > Hi, > > I need some help with an idea I am trying to implement with > Freeswitch. I am trying to work with a partner company which will > forward a sip call to my freeswitch server and my freeswitch > server is suppose to playback annoucements to the caller and then > once they are finished listening they should be forwarded back to > the partner company so that the call can continue without my > freeswitch being in the call path. So does any one have any > suggestions/ideas on the best way to implement this? > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.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 Fri Dec 9 01:30:49 2011 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Thu, 08 Dec 2011 17:30:49 -0500 Subject: [Freeswitch-users] Outbound calling with recorded message. In-Reply-To: <9AD65D9A5C774D98B07C8A46D3D9E535@clancysystems.com> References: <9AD65D9A5C774D98B07C8A46D3D9E535@clancysystems.com> Message-ID: <4EE13A99.3000305@communicatefreely.net> Do what MC says below, and use xml_cdr to call your PHP script and update the database. When xml_cdr posts, it gives you pretty much every variable there is, so you could take a look at the number dialed, the duration of the call, and the disposition fields and do a database update to mark that registrant as "contacted" or not. You could even do something fancier with an IVR. All each IVR branch would do would be to set a unique variable, like "liked_the_food=true". When the call hangs up, all these variables will be available in the CDR record, and can be associated with that participant. -Tim Dave wrote: > Thank you very much for the reply. > > Good point about the calls not reaching the target. In thinking about > it it might be better to send a text or email to them. > > I do want to learn about the event socket though, so I'll study up on > that. After I sent the question I noticed the freeSWITCH book has a > chapter on that and there's a lot online about it. > > I really appreciate the help everyone provides. > > Dave > > > > > ----- Original Message ----- > *From:* Michael Collins > *To:* FreeSWITCH Users Help > > *Sent:* Wednesday, September 28, 2011 7:18 PM > *Subject:* Re: [Freeswitch-users] Outbound calling with recorded > message. > > Once you have the calls logged into a database then it is a > relatively simple matter to generate the outbound calls using the > event socket. The real challenge (IMHO) is accounting for the > outbound calls that don't actually reach the target, or that go to > the target's voicemail, etc. > > For the sake of simplicity, let's assume that each person you call > will answer. From there you just need a simple dialplan extension > that does a record app with a specific filename. (You need to > match up the filename recorded with the person you called. You > could use .wav I suppose.) From there it's a matter of > launching the calls. I don't do much with Windows but there are > plenty of folks here who do. If you can establish an event socket > connection then you can execute a bunch of "originate" API calls > to generate your outbound calls. > > Let's say your dialplan extension for recording the name is this: > > expression="^OB_IVR_Record_Name_(\d+)$"> > > > data="please_record_your_name.wav"/> > data="tone_stream://%(1000,0,1500)"/> > > > > > You can generate a call with this API: > > originate sofia/gw/gwname/18005551212 OB_IVR_Record_Name_8005551212 > > If it works then you'll end up with /tmp/8005551212.wav and > hopefully they'll have given you the info you need. > > As for generating these calls, if you don't need something too > fancy you could just use a program written in the scripting > language of your choice. (Perl, Python, Ruby, and PHP are all > suitable for this task.) You could also write "real" program with > Visual Studio if that suits you. The key is that you will need to > keep track of what happens when you make all these calls and be > sure not to keep calling them over and over again. :) > > I've done this sort of thing with just Perl scripts and it works > really well. > > -MC > > On Wed, Sep 28, 2011 at 10:35 AM, Dave > wrote: > > > Hi all, > > I have been using freeSWITCH for a while for inbound calls in > which a person registers for a seminar via IVR. I simply use > the DID, CID number and name to identify the person and put > them in the Database. That part of the application is working > wonderfully. > > The issue I'm presented with now is that we need automate > making calls to a few of these registrants, after each event, > whos caller_id_name comes in as "Unknown" or "Wireless > Caller", Play a recording to let them know we need the name, > and that they can record the information right then, or call > the office. > > I am limited to using a Windows 7 machine. > > What would be the best tool to use to automate these calls > with the use of an IVR? > > Dave Goodwin > > > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------------------------------------------------ > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From spencer at 5ninesolutions.com Fri Dec 9 02:21:09 2011 From: spencer at 5ninesolutions.com (Spencer Thomason) Date: Thu, 8 Dec 2011 15:21:09 -0800 Subject: [Freeswitch-users] BLF gets confused with multiple calls In-Reply-To: <4EE138D9.7010306@communicatefreely.net> References: <4EE12D98.8090100@communicatefreely.net> <4EE138D9.7010306@communicatefreely.net> Message-ID: <444D6932-AA29-4EF7-B855-72BF9F90343A@5ninesolutions.com> Hi Tim, Try this when you generate the XML in the directory: We are using FreeSWITCH in a multi tenant environment and this works great. Thanks, Spencer On Dec 8, 2011, at 2:23 PM, Tim St. Pierre wrote: > No, we use multiple domains here. The directory is generated by an > xml_curl script that sends it back dynamically. > > In general, BLFs work great, but they seem to be based on the last > message that thing the phone did (it just answered a call, or it just > hung up a call), as opposed to whether or not it has a channel connected > to it or not, and that is where things are going wrong, is when it just > hung up, but there is still one more channel left, so it's really still > in use. > > The dial strings are all user/exten at our.domain where our.domain is the > same domain that both registrations and subscriptions use. I couldn't > get presence to work at all without that. > > What is the relationship between the user/ domain, the subscription > domain, and the registration domain vs. invite domains and whatever else. > > Should this work correctly, and if so, where should I look in terms of > debugging it? > > Thanks! > > Brian West wrote: >> >> On Dec 8, 2011, at 3:35 PM, Tim St. Pierre wrote: >>> >>> Here's where it gets tricky: >>> >>> If there are two phones registered to the extension, it flashes when >>> they ring, but then goes out when one of them answers. >>> If either phone places an outgoing call, the lamp comes on. >> >> I'm going to guess you're not using the user/ from the default >> directory like the default where it sets the presence ID? >> >> ------------------------------------------------------------------------ >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From msc at freeswitch.org Fri Dec 9 02:46:41 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 8 Dec 2011 15:46:41 -0800 Subject: [Freeswitch-users] Volunteers Wanted: CDR Help for opencdrrate.org Message-ID: Hello all, I am working with Dean over at opencdrrate.org to help him get his project set up to work with FreeSWITCH. We would like to create a new cdr-csv template specifically for opencdrrate. I am looking for a few volunteers who can do a bit of research and testing based on Dean's input about what fields would constitute the ideal CDR layout. Please email me off list if you can assist. Thanks, Michael S Collins http://www.freeswitch.org http://www.cluecon.com http://www.ostag.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111208/2b6e84e1/attachment.html From anthony.minessale at gmail.com Fri Dec 9 02:48:16 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 8 Dec 2011 17:48:16 -0600 Subject: [Freeswitch-users] BLF gets confused with multiple calls In-Reply-To: <4EE12D98.8090100@communicatefreely.net> References: <4EE12D98.8090100@communicatefreely.net> Message-ID: You are running a version from August. Too bad for you that you are missing a whole autumn worth of updates including a whole bunch of work on presence. while(!current) update(); On Thu, Dec 8, 2011 at 3:35 PM, Tim St. Pierre < fs-list at communicatefreely.net> wrote: > Hello, > > I'm having all sorts of problems with BLFs not being in the correct > state. It works fine in some cases, but others are wrong. Here's what > I can see: > > Single endpoint, single call, everything works fine. Light flashes on > ring, goes steady on answer, goes out on hangup. > > Here's where it gets tricky: > > If there are two phones registered to the extension, it flashes when > they ring, but then goes out when one of them answers. > If either phone places an outgoing call, the lamp comes on. > > If a single phone gets a second call, their lamp flashes again, but then > goes out when they answer the second call. > > I'm constantly getting complaints from users where the lamps are stuck > on. It happens more often when they have a lot of phones in ring > groups. The lamps work fine for ring groups - they all flash, and > whomever picks up the call stays steady while the rest go out. > > Is there anything I can do to get freeswitch to base the state on > whether or not that user has any active calls, rather than just what the > last thing that the phone did was? > > This happens on every model of Aastra phone, and I have all of them. I > haven't had a chance to try it yet on Polycom. > > I'm running FreeSWITCH Version 1.0.head (git-7531fed 2011-08-17 11-27-20 > -0500) > > Any ideas would be more than welcome. > > -Tim > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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/20111208/914a096e/attachment.html From ryan at kaevee.com Fri Dec 9 04:49:45 2011 From: ryan at kaevee.com (Ryan V) Date: Fri, 9 Dec 2011 07:19:45 +0530 Subject: [Freeswitch-users] Sangoma A200 FXO Outgoing Problem In-Reply-To: <14AA2B32-ACE0-429F-AB60-2E9CAF98BD96@freeswitch.org> References: <14AA2B32-ACE0-429F-AB60-2E9CAF98BD96@freeswitch.org> Message-ID: On Fri, Dec 9, 2011 at 2:13 AM, Brian West wrote: > its freetdm/spanname/channelnumber/numbertodial > > You can't do a on those as far as I knew. > > I could see it trying to dial out on span 3 channel 1 even though I had "a" in Here is the excerpt from logs. [DEBUG] switch_core_state_machine.c:362 (FreeTDM/3:1/9449905000) Running State Change CS_HANGUP Thanks, Ryan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111209/f1a84112/attachment-0001.html From fs-list at communicatefreely.net Fri Dec 9 05:53:25 2011 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Thu, 08 Dec 2011 21:53:25 -0500 Subject: [Freeswitch-users] BLF gets confused with multiple calls In-Reply-To: References: <4EE12D98.8090100@communicatefreely.net> Message-ID: <4EE17825.9050305@communicatefreely.net> Yes, but when you have 368 very demanding users, you don't just blindly update systems that mostly work. I do have a devel system, so I will pull the latest git and once I have tested everything I can think of, I will cross my fingers and push it to the production systems. Not that I'm not grateful for continuous development, but it takes a lot of resources to update and test, so I only do it every so often. I'm glad there have been some updates. I look forward to trying them out. -Tim Anthony Minessale wrote: > You are running a version from August. Too bad for you that you are > missing a whole autumn worth of updates including a whole bunch of > work on presence. > > while(!current) update(); > > > > On Thu, Dec 8, 2011 at 3:35 PM, Tim St. Pierre > > > wrote: > > Hello, > > I'm having all sorts of problems with BLFs not being in the correct > state. It works fine in some cases, but others are wrong. Here's > what > I can see: > > Single endpoint, single call, everything works fine. Light flashes on > ring, goes steady on answer, goes out on hangup. > > Here's where it gets tricky: > > If there are two phones registered to the extension, it flashes when > they ring, but then goes out when one of them answers. > If either phone places an outgoing call, the lamp comes on. > > If a single phone gets a second call, their lamp flashes again, > but then > goes out when they answer the second call. > > I'm constantly getting complaints from users where the lamps are stuck > on. It happens more often when they have a lot of phones in ring > groups. The lamps work fine for ring groups - they all flash, and > whomever picks up the call stays steady while the rest go out. > > Is there anything I can do to get freeswitch to base the state on > whether or not that user has any active calls, rather than just > what the > last thing that the phone did was? > > This happens on every model of Aastra phone, and I have all of > them. I > haven't had a chance to try it yet on Polycom. > > I'm running FreeSWITCH Version 1.0.head (git-7531fed 2011-08-17 > 11-27-20 > -0500) > > Any ideas would be more than welcome. > > -Tim > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 yehavi.bourvine at gmail.com Fri Dec 9 07:42:19 2011 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Fri, 9 Dec 2011 06:42:19 +0200 Subject: [Freeswitch-users] BLF gets confused with multiple calls In-Reply-To: <4EE17825.9050305@communicatefreely.net> References: <4EE12D98.8090100@communicatefreely.net> <4EE17825.9050305@communicatefreely.net> Message-ID: Some more input from my side: I do an an upgrade to the current version every 3-5 weeks. BLF worked great on Polycom and SNOM phones up to version FreeSWITCH Version 1.0.head (git-10df279 2011-10-15 07-59-23 -0500) (the last one I tried). In mid-November I've upgraded to the latest GIT and BLFs started getting crazy (phones losing them, stuck in ringing, etc.). I tried the GIT onlast Wednsday but it behaved the same, so I had to revert to the version from 15-Oct. I did not open a JIRA yet as I couldn't gather all the information, but from an intial investigation it looks like a problem with the version number of the NOTIFY message sent on the body of the dialog description: either it starts at 1 instead of 0, or it skips versions. I'll try gathering more information this week. Regards, __Yehavi: 2011/12/9 Tim St. Pierre > Yes, but when you have 368 very demanding users, you don't just blindly > update systems that mostly work. > > I do have a devel system, so I will pull the latest git and once I have > tested everything I can think of, I will cross my fingers and push it to > the production systems. > > Not that I'm not grateful for continuous development, but it takes a lot > of resources to update and test, so I only do it every so often. > > I'm glad there have been some updates. I look forward to trying them out. > > -Tim > > Anthony Minessale wrote: > > You are running a version from August. Too bad for you that you are > > missing a whole autumn worth of updates including a whole bunch of > > work on presence. > > > > while(!current) update(); > > > > > > > > On Thu, Dec 8, 2011 at 3:35 PM, Tim St. Pierre > > > > > wrote: > > > > Hello, > > > > I'm having all sorts of problems with BLFs not being in the correct > > state. It works fine in some cases, but others are wrong. Here's > > what > > I can see: > > > > Single endpoint, single call, everything works fine. Light flashes > on > > ring, goes steady on answer, goes out on hangup. > > > > Here's where it gets tricky: > > > > If there are two phones registered to the extension, it flashes when > > they ring, but then goes out when one of them answers. > > If either phone places an outgoing call, the lamp comes on. > > > > If a single phone gets a second call, their lamp flashes again, > > but then > > goes out when they answer the second call. > > > > I'm constantly getting complaints from users where the lamps are > stuck > > on. It happens more often when they have a lot of phones in ring > > groups. The lamps work fine for ring groups - they all flash, and > > whomever picks up the call stays steady while the rest go out. > > > > Is there anything I can do to get freeswitch to base the state on > > whether or not that user has any active calls, rather than just > > what the > > last thing that the phone did was? > > > > This happens on every model of Aastra phone, and I have all of > > them. I > > haven't had a chance to try it yet on Polycom. > > > > I'm running FreeSWITCH Version 1.0.head (git-7531fed 2011-08-17 > > 11-27-20 > > -0500) > > > > Any ideas would be more than welcome. > > > > -Tim > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE: > http://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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111209/4e98452d/attachment.html From oseslija at gmail.com Fri Dec 9 10:52:47 2011 From: oseslija at gmail.com (Ognjen Seslija) Date: Fri, 9 Dec 2011 08:52:47 +0100 Subject: [Freeswitch-users] BLF gets confused with multiple calls In-Reply-To: <4EE12D98.8090100@communicatefreely.net> References: <4EE12D98.8090100@communicatefreely.net> Message-ID: As Tony said, update. I have concurrent calls presence working just fine nowadays. On 8 Dec 2011 22:38, "Tim St. Pierre" wrote: > > Hello, > > I'm having all sorts of problems with BLFs not being in the correct > state. It works fine in some cases, but others are wrong. Here's what > I can see: > > Single endpoint, single call, everything works fine. Light flashes on > ring, goes steady on answer, goes out on hangup. > > Here's where it gets tricky: > > If there are two phones registered to the extension, it flashes when > they ring, but then goes out when one of them answers. > If either phone places an outgoing call, the lamp comes on. > > If a single phone gets a second call, their lamp flashes again, but then > goes out when they answer the second call. > > I'm constantly getting complaints from users where the lamps are stuck > on. It happens more often when they have a lot of phones in ring > groups. The lamps work fine for ring groups - they all flash, and > whomever picks up the call stays steady while the rest go out. > > Is there anything I can do to get freeswitch to base the state on > whether or not that user has any active calls, rather than just what the > last thing that the phone did was? > > This happens on every model of Aastra phone, and I have all of them. I > haven't had a chance to try it yet on Polycom. > > I'm running FreeSWITCH Version 1.0.head (git-7531fed 2011-08-17 11-27-20 > -0500) > > Any ideas would be more than welcome. > > -Tim > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111209/8c86a13b/attachment-0001.html From Nabble_01394 at slickdeals.endjunk.com Fri Dec 9 14:05:42 2011 From: Nabble_01394 at slickdeals.endjunk.com (mazilo) Date: Fri, 9 Dec 2011 03:05:42 -0800 (PST) Subject: [Freeswitch-users] spandsp/src/dtmf.c: 'duration' isn't a membership of 'dtmf_rx_state_t'? In-Reply-To: <4EE0E41A.4050406@coppice.org> References: <1323356249725-7074568.post@n2.nabble.com> <4EE0E41A.4050406@coppice.org> Message-ID: <1323428742912-7077805.post@n2.nabble.com> Steve, thank you. ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/spandsp-src-dtmf-c-duration-isn-t-a-membership-of-dtmf-rx-state-t-tp7074568p7077805.html Sent from the freeswitch-users mailing list archive at Nabble.com. From acrow at integrafin.co.uk Fri Dec 9 17:46:43 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Fri, 09 Dec 2011 14:46:43 +0000 Subject: [Freeswitch-users] BLF gets confused with multiple calls In-Reply-To: References: <4EE12D98.8090100@communicatefreely.net> Message-ID: <4EE21F53.8080303@integrafin.co.uk> On 09/12/11 07:52, Ognjen Seslija wrote: > > As Tony said, update. I have concurrent calls presence working just > fine nowadays. > On 8 Dec 2011 22:38, "Tim St. Pierre" > wrote: > On Snom, I still don't get flashing for a second call. Snom claim this is as intended, which I think is silly. We get flashing when a monitored extension is held, which is also confusing. Pressing the BLF button steals the held call. On Grandstream, git of about 2 weeks ago, it seems to work perfectly - again though we get flashing on hold and the held call can be stolen. Polycom is also good, but yet again we have the flashing on hold problem, but here the call cannot be stolen, which seems better. I'm very frustrated with the Snom behaviour and the fact they label it as intended. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111209/f208ac00/attachment.html From acrow at integrafin.co.uk Fri Dec 9 21:31:21 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Fri, 09 Dec 2011 18:31:21 +0000 Subject: [Freeswitch-users] Two interfaces (LAN/DMZ), same domain Message-ID: <4EE253F9.9050804@integrafin.co.uk> Hi all, How would I go about setting FS up so that I can register internal extensions (not NAT'ed) and external ones (invariably NAT'ed) to the same domain? I will keep the external profile just for outbound trunks so should not need to change that other than forcing sip-ip and rtp-ip to the DMZ IP. I thought of setting up a third profile (eg, external_phones) and setting the appropriate sip-ip and rtp-ip on all profiles. However sorting out the domain confuses me. Would I need to set for instance: On the external_phones profile to ensure that external devices register to the same domain as internal ones and share presence/registration info? Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From drew_terenzini at wesleycloversolutions.com Fri Dec 9 06:43:21 2011 From: drew_terenzini at wesleycloversolutions.com (Drew Terenzini) Date: Thu, 8 Dec 2011 19:43:21 -0800 Subject: [Freeswitch-users] Multicast paging group not getting conference audio? Message-ID: <75BADEA2E96D4023ACC6E6B8F6B8A958@DREWPC> Good evening, I'm working with a heavily edited version of Freeswitch and I'm trying to get a multicast paging group added to a conference such that any audio in the conference is broadcast to the paging group as well. I've successfully gotten a multicast paging group working in the dialplan as follows: When I use X-Lite registered to FS and call "3456", I get the spoken audio out the paging group correctly. Now I'm trying to link the paging group to a conference. I've tried experimenting with adding it to an existing conference by using "conference_set_auto_outcall", but that's failing: And bridge attempts aren't working as well: I'm new to FS and have been scouring the Wiki and mailing lists for clues on how to accomplish this. Is this something that is not possible or are I missing a configuration step? Any clues would be appreciated, thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111208/f8740544/attachment.html From farooqhussain786 at gmail.com Fri Dec 9 10:36:54 2011 From: farooqhussain786 at gmail.com (Farooq Hussain) Date: Fri, 9 Dec 2011 12:36:54 +0500 Subject: [Freeswitch-users] Fwd: Configure openVOX card FXO card. In-Reply-To: References: Message-ID: Hello, I am new to freeswtich. Please let me know how to configure FXO openvox 4ports card configure on freeswtich. -- Thanks Farooq Hussain -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111209/38674c19/attachment.html From gautambatra24 at gmail.com Fri Dec 9 22:44:00 2011 From: gautambatra24 at gmail.com (gautam) Date: Fri, 9 Dec 2011 11:44:00 -0800 (PST) Subject: [Freeswitch-users] Invalid SAY Interface [en] Message-ID: <1323459840645-7079510.post@n2.nabble.com> Hello, The voicmail and IVR prompts in my system are not saying out the numbers. The rest of the playback is happening correctly. In the log I'm getting the following error for the digits: >>[ERR] switch_ivr_play_say.c:287 Invalid SAY Interface [en]! The prompt that I here is something like this: "The person at extension is not available" and "Press to play the recording, press to listen to the recording. " The files are there in the digits folder, and the voicemail is getting saved correctly. Please help me fix this. Gautam -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Invalid-SAY-Interface-en-tp7079510p7079510.html Sent from the freeswitch-users mailing list archive at Nabble.com. From gautambatra24 at gmail.com Fri Dec 9 22:57:55 2011 From: gautambatra24 at gmail.com (Gautam Batra) Date: Fri, 9 Dec 2011 14:57:55 -0500 Subject: [Freeswitch-users] Invalid SAY Interface [en] Message-ID: Hello, The voicmail and IVR prompts in my system are not saying out the numbers. The rest of the playback is happening correctly. In the log I'm getting the following error for the digits: >>[ERR] switch_ivr_play_say.c:287 Invalid SAY Interface [en]! The prompt that I here is something like this: "The person at extension is not available" and "Press to play the recording, press to listen to the recording. " The files are there in the digits folder, and the voicemail is getting saved correctly. Please help me fix this. Gautam -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111209/93cf30a9/attachment-0001.html From msc at freeswitch.org Fri Dec 9 23:29:16 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 9 Dec 2011 12:29:16 -0800 Subject: [Freeswitch-users] Invalid SAY Interface [en] In-Reply-To: <1323459840645-7079510.post@n2.nabble.com> References: <1323459840645-7079510.post@n2.nabble.com> Message-ID: You need to confirm that mod_say_en is getting loaded. Try this from the fs_cli: load mod_say_en It will say module already loaded, or it will load, or it will throw an error. Either way, you'll know what's going on and where you need to focus your efforts next. -MC On Fri, Dec 9, 2011 at 11:44 AM, gautam wrote: > Hello, > > The voicmail and IVR prompts in my system are not saying out the numbers. > The rest of the playback is happening correctly. In the log I'm getting the > following error for the digits: > >>[ERR] switch_ivr_play_say.c:287 Invalid SAY Interface [en]! > > The prompt that I here is something like this: "The person at extension is > not available" and "Press to play the recording, press to listen to the > recording. " The files are there in the digits folder, and the voicemail is > getting saved correctly. Please help me fix this. > > Gautam > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Invalid-SAY-Interface-en-tp7079510p7079510.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/20111209/75645dec/attachment.html From farooqhussain786 at gmail.com Sat Dec 10 00:44:23 2011 From: farooqhussain786 at gmail.com (Farooq Hussain) Date: Sat, 10 Dec 2011 02:44:23 +0500 Subject: [Freeswitch-users] echo cancellation Message-ID: Hello Everyone, I have just configure a freeswitch with FXO. I may receiving lots of echo when I am dialing number through FXO card. How can i get rid from this echo cancellation. -- Thanks Farooq Hussain -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111210/36fa44dd/attachment.html From gautambatra24 at gmail.com Sat Dec 10 01:04:16 2011 From: gautambatra24 at gmail.com (gautam) Date: Fri, 9 Dec 2011 14:04:16 -0800 (PST) Subject: [Freeswitch-users] Invalid SAY Interface [en] In-Reply-To: References: <1323459840645-7079510.post@n2.nabble.com> Message-ID: Thanks for the reply. It turns out that the say module has disappeared from my system somehow. Is there any way I can add just that module back? -Gautam On Fri, Dec 9, 2011 at 3:45 PM, mercutioviz [via freeswitch-users] < ml-node+s2379917n7079768h89 at n2.nabble.com> wrote: > You need to confirm that mod_say_en is getting loaded. Try this from the > fs_cli: > > load mod_say_en > > It will say module already loaded, or it will load, or it will throw an > error. Either way, you'll know what's going on and where you need to focus > your efforts next. > > -MC > > On Fri, Dec 9, 2011 at 11:44 AM, gautam <[hidden email] > > wrote: > >> Hello, >> >> The voicmail and IVR prompts in my system are not saying out the numbers. >> The rest of the playback is happening correctly. In the log I'm getting >> the >> following error for the digits: >> >>[ERR] switch_ivr_play_say.c:287 Invalid SAY Interface [en]! >> >> The prompt that I here is something like this: "The person at extension is >> not available" and "Press to play the recording, press to listen to the >> recording. " The files are there in the digits folder, and the voicemail >> is >> getting saved correctly. Please help me fix this. >> >> Gautam >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Invalid-SAY-Interface-en-tp7079510p7079510.html >> Sent from the freeswitch-users mailing list archive at Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> [hidden email] >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> [hidden email] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > [hidden email] > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > [hidden email] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > If you reply to this email, your message will be added to the discussion > below: > > http://freeswitch-users.2379917.n2.nabble.com/Invalid-SAY-Interface-en-tp7079510p7079768.html > To unsubscribe from Invalid SAY Interface [en], click here > . > NAML > -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Invalid-SAY-Interface-en-tp7079510p7079984.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111209/0b660a01/attachment.html From msc at freeswitch.org Sat Dec 10 04:28:01 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 9 Dec 2011 17:28:01 -0800 Subject: [Freeswitch-users] Invalid SAY Interface [en] In-Reply-To: References: <1323459840645-7079510.post@n2.nabble.com> Message-ID: at the freeswitch source directory: make mod_say_en-install BTW, all this is on the wiki. The key to using the wiki is to use the search box. I know it can be a little hard to find stuff at first but you'll be glad you learned it. Our wiki is a microcosm of the Internet: the information is out there, you just need to learn some google fu. :) -MC On Fri, Dec 9, 2011 at 2:04 PM, gautam wrote: > Thanks for the reply. It turns out that the say module has disappeared > from my system somehow. Is there any way I can add just that module back? > > -Gautam > > On Fri, Dec 9, 2011 at 3:45 PM, mercutioviz [via freeswitch-users] <[hidden > email] > wrote: > >> You need to confirm that mod_say_en is getting loaded. Try this from the >> fs_cli: >> >> load mod_say_en >> >> It will say module already loaded, or it will load, or it will throw an >> error. Either way, you'll know what's going on and where you need to focus >> your efforts next. >> >> -MC >> >> On Fri, Dec 9, 2011 at 11:44 AM, gautam <[hidden email] >> > wrote: >> >>> Hello, >>> >>> The voicmail and IVR prompts in my system are not saying out the numbers. >>> The rest of the playback is happening correctly. In the log I'm getting >>> the >>> following error for the digits: >>> >>[ERR] switch_ivr_play_say.c:287 Invalid SAY Interface [en]! >>> >>> The prompt that I here is something like this: "The person at extension >>> is >>> not available" and "Press to play the recording, press to listen to the >>> recording. " The files are there in the digits folder, and the voicemail >>> is >>> getting saved correctly. Please help me fix this. >>> >>> Gautam >>> >>> -- >>> View this message in context: >>> http://freeswitch-users.2379917.n2.nabble.com/Invalid-SAY-Interface-en-tp7079510p7079510.html >>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> [hidden email] >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> [hidden email] >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> [hidden email] >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> [hidden email] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> >> ------------------------------ >> If you reply to this email, your message will be added to the >> discussion below: >> >> http://freeswitch-users.2379917.n2.nabble.com/Invalid-SAY-Interface-en-tp7079510p7079768.html >> To unsubscribe from Invalid SAY Interface [en], click here. >> NAML >> > > > ------------------------------ > View this message in context: Re: Invalid SAY Interface [en] > Sent from the freeswitch-users mailing list archiveat Nabble.com. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111209/274de807/attachment-0001.html From anthony.minessale at gmail.com Sat Dec 10 04:35:17 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 9 Dec 2011 19:35:17 -0600 Subject: [Freeswitch-users] spandsp/src/dtmf.c: 'duration' isn't a membership of 'dtmf_rx_state_t'? In-Reply-To: <1323428742912-7077805.post@n2.nabble.com> References: <1323356249725-7074568.post@n2.nabble.com> <4EE0E41A.4050406@coppice.org> <1323428742912-7077805.post@n2.nabble.com> Message-ID: make spandsp-reconf to fix that On Fri, Dec 9, 2011 at 5:05 AM, mazilo wrote: > Steve, thank you. > > ----- > FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 > Watts of electricity. > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/spandsp-src-dtmf-c-duration-isn-t-a-membership-of-dtmf-rx-state-t-tp7074568p7077805.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 > -- 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/20111209/2f3f528c/attachment.html From gautambatra24 at gmail.com Sat Dec 10 05:28:50 2011 From: gautambatra24 at gmail.com (gautam) Date: Fri, 9 Dec 2011 18:28:50 -0800 (PST) Subject: [Freeswitch-users] Invalid SAY Interface [en] In-Reply-To: References: <1323459840645-7079510.post@n2.nabble.com> Message-ID: I had actually found the relevant documentation on the wiki. The problem is that I'm running the system on Alpine linux, and I installed Freeswitch using their package management utility (apk add freeswitch, etc). All the modules were present initially, but now mod_say_en has disappeared somehow. If I try to use "make" it says "no rule to make mod_say_en-install." I would really appreciate it if you could give me any direction to solve this. -Gautam On Fri, Dec 9, 2011 at 8:35 PM, mercutioviz [via freeswitch-users] < ml-node+s2379917n7080389h40 at n2.nabble.com> wrote: > at the freeswitch source directory: > make mod_say_en-install > > BTW, all this is on the wiki. The key to using the wiki is to use the > search box. I know it can be a little hard to find stuff at first but > you'll be glad you learned it. Our wiki is a microcosm of the Internet: the > information is out there, you just need to learn some google fu. :) > > -MC > > On Fri, Dec 9, 2011 at 2:04 PM, gautam <[hidden email] > > wrote: > >> Thanks for the reply. It turns out that the say module has disappeared >> from my system somehow. Is there any way I can add just that module back? >> >> -Gautam >> >> On Fri, Dec 9, 2011 at 3:45 PM, mercutioviz [via freeswitch-users] <[hidden >> email] > wrote: >> >>> You need to confirm that mod_say_en is getting loaded. Try this from the >>> fs_cli: >>> >>> load mod_say_en >>> >>> It will say module already loaded, or it will load, or it will throw an >>> error. Either way, you'll know what's going on and where you need to focus >>> your efforts next. >>> >>> -MC >>> >>> On Fri, Dec 9, 2011 at 11:44 AM, gautam <[hidden email] >>> > wrote: >>> >>>> Hello, >>>> >>>> The voicmail and IVR prompts in my system are not saying out the >>>> numbers. >>>> The rest of the playback is happening correctly. In the log I'm getting >>>> the >>>> following error for the digits: >>>> >>[ERR] switch_ivr_play_say.c:287 Invalid SAY Interface [en]! >>>> >>>> The prompt that I here is something like this: "The person at extension >>>> is >>>> not available" and "Press to play the recording, press to listen to the >>>> recording. " The files are there in the digits folder, and the >>>> voicemail is >>>> getting saved correctly. Please help me fix this. >>>> >>>> Gautam >>>> >>>> -- >>>> View this message in context: >>>> http://freeswitch-users.2379917.n2.nabble.com/Invalid-SAY-Interface-en-tp7079510p7079510.html >>>> Sent from the freeswitch-users mailing list archive at Nabble.com. >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> [hidden email] >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> [hidden email] >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>> >>> >>> _________________________________________________________________________ >>> >>> Professional FreeSWITCH Consulting Services: >>> [hidden email] >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> [hidden email] >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >>> ------------------------------ >>> If you reply to this email, your message will be added to the >>> discussion below: >>> >>> http://freeswitch-users.2379917.n2.nabble.com/Invalid-SAY-Interface-en-tp7079510p7079768.html >>> To unsubscribe from Invalid SAY Interface [en], click here. >>> NAML >>> >> >> >> ------------------------------ >> View this message in context: Re: Invalid SAY Interface [en] >> >> Sent from the freeswitch-users mailing list archiveat Nabble.com. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> [hidden email] >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> [hidden email] >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > [hidden email] > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > [hidden email] > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > ------------------------------ > If you reply to this email, your message will be added to the discussion > below: > > http://freeswitch-users.2379917.n2.nabble.com/Invalid-SAY-Interface-en-tp7079510p7080389.html > To unsubscribe from Invalid SAY Interface [en], click here > . > NAML > -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Invalid-SAY-Interface-en-tp7079510p7080448.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111209/1145867a/attachment-0001.html From xing2kin at yahoo.com Sat Dec 10 09:03:34 2011 From: xing2kin at yahoo.com (king2kin) Date: Fri, 9 Dec 2011 22:03:34 -0800 (PST) Subject: [Freeswitch-users] Question Answertime variable? billsec is not available In-Reply-To: References: Message-ID: <1323497014.35405.YahooMailNeo@web39705.mail.mud.yahoo.com> The channel variables for this purpose seems to be {duration, billsec, answersec, flow_billsec, mduration, billmsec, answermsec, uduration, billusec, answerusec} see: source code "switch_channel.c"; and wiki document on categories: channel variables see: http://wiki.freeswitch.com/index.php?title=Category:Variable&from=Variable+billmsec? ? However, when I tried to access these channel variables from Lua script or dialplan, their values are always "nil" or empty. ? -- dialplan: { ?? ????........ ???? ????? ????? } ? -- Lua script: 2011-12-10 13:39:55.609375 [ERR] mod_lua.cpp:191 C:\c4dev\freeswitch\Release\scr ipts/test1.lua:60: attempt to concatenate global 'ctsec' (a nil value) stack traceback: ??????? C:\c4dev\freeswitch\Release\scripts/test1.lua:60: in main chunk ? see: { session:answer() ......... ctsec = session:getVariable('billsec') ctmsec = session:getVariable('billmsec') freeswitch.consoleLog("INFO","***** Call-Time: sec=" .. ctsec .. "\n") freeswitch.consoleLog("INFO","***** Call-Time: msec=" .. ctmsec .. "\n") session:hangup() } ? From: curriegrad2004 To: FreeSWITCH Users Help Sent: Wednesday, November 30, 2011 11:15 PM Subject: Re: [Freeswitch-users] Question Answertime variable? billseconds or bill_msec would be the one to turn to. On Wed, Nov 30, 2011 at 4:22 AM, Thomas Hoellriegel wrote: > Hi all, > Is there a possibility to determine the caller time for a channel at > the end of a call? > > For example: in asterisk exists a variable: > ${ANSWEREDTIME} > I like to store the Answertime in a Database for example: > Dailly minutes to call a Cellphone. > Can your help plese? > 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 > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111209/fa37ab4a/attachment.html From Nabble_01394 at slickdeals.endjunk.com Sat Dec 10 16:51:01 2011 From: Nabble_01394 at slickdeals.endjunk.com (mazilo) Date: Sat, 10 Dec 2011 05:51:01 -0800 (PST) Subject: [Freeswitch-users] Invalid SAY Interface [en] In-Reply-To: References: <1323459840645-7079510.post@n2.nabble.com> Message-ID: <1323525061803-7081463.post@n2.nabble.com> gautam wrote > > I had actually found the relevant documentation on the wiki. The problem > is > that I'm running the system on Alpine linux, and I installed Freeswitch > using their package management utility (apk add freeswitch, etc). All the > modules were present initially, but now mod_say_en has disappeared > somehow. > If I try to use "make" it says "no rule to make mod_say_en-install." I > would really appreciate it if you could give me any direction to solve > this. > > -Gautam If you do like that, you will probably create more problems to the software package manager on your system. Anyway, I don't use Alpine Linux distro. However, I believe you can rebuild/recompile the FS package using its software package management specs file. On a Linux distro that uses RPM, i.e. RedHat, OpenSuSE, etc., each software package comes with its editable plain-text RPM specs file and can be rebuilt/recompiled using its rpmbuild utility. I believe Alpine Linux distro has its own software package manager utility to do the same thing. Then, you can use the software package manager to upgrade your FS installation with the newly built FS package. ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Invalid-SAY-Interface-en-tp7079510p7081463.html Sent from the freeswitch-users mailing list archive at Nabble.com. From fs-list at communicatefreely.net Sun Dec 11 03:39:51 2011 From: fs-list at communicatefreely.net (Tim St. Pierre) Date: Sat, 10 Dec 2011 19:39:51 -0500 Subject: [Freeswitch-users] Multicast paging group not getting conference audio? In-Reply-To: <75BADEA2E96D4023ACC6E6B8F6B8A958@DREWPC> References: <75BADEA2E96D4023ACC6E6B8F6B8A958@DREWPC> Message-ID: <4EE3FBD7.6020908@communicatefreely.net> It looks to me like you are mixing up the ideas of user vs. extension in a dialplan. When you call bridge with data="sofia/" you are telling freeswitch to use sofia to set up the call. Sofia is trying to initiate a SIP call based on the other arguments, which will try to do a SIP invite to 3456@$${domain} - probably not what you want. Multicast paging is not actually SIP, nor is it even a phone call as far as set up, caller ID, etc. That's why it's done as an application, just like recording or any other audio stream output. You could try a loopback bridge - that might do what you want. This will originate a call back into the dialplan, using the dialplan parts you already have working. Drew Terenzini wrote: > > Good evening, I?m working with a heavily edited version of Freeswitch > and I?m trying to get a multicast paging group added to a conference > such that any audio in the conference is broadcast to the paging group > as well. I?ve successfully gotten a multicast paging group working in > the dialplan as follows: > > > > > > > > > > > > > > > > > > When I use X-Lite registered to FS and call ?3456?, I get the spoken > audio out the paging group correctly. Now I?m trying to link the > paging group to a conference. I?ve tried experimenting with adding it > to an existing conference by using ?conference_set_auto_outcall?, but > that?s failing: > > > > And bridge attempts aren?t working as well: > > > > I?m new to FS and have been scouring the Wiki and mailing lists for > clues on how to accomplish this. Is this something that is not > possible or are I missing a configuration step? Any clues would be > appreciated, 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 elliott at zoogmedia.com Sun Dec 11 22:10:53 2011 From: elliott at zoogmedia.com (Elliott Vogel) Date: Sun, 11 Dec 2011 19:10:53 +0000 Subject: [Freeswitch-users] dialplan In-Reply-To: <4EDA7976.1020803@cupis.co.uk> References: <4EDA7976.1020803@cupis.co.uk> Message-ID: Hello, I seem to have a problem when I try to bridge via " user/1900 at zoogmedia.com,sofia/gateway/TSG/+13124346168" FS stops ringing the user when it rings number on the gateway. Do I not have something configured correctly? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul Cupis Sent: Saturday, December 03, 2011 1:33 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] dialplan On 03/12/11 17:28, Elliott Vogel wrote: > How can I bridge out calls via another bridge if the first bridge > fails? - let's say the first bridge returns a 404 or 503 error Have a look at the following: http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_7:_Action_failover_on_failed_action http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail 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 From avi at avimarcus.net Sun Dec 11 22:39:26 2011 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 11 Dec 2011 21:39:26 +0200 Subject: [Freeswitch-users] dialplan In-Reply-To: References: <4EDA7976.1020803@cupis.co.uk> Message-ID: You only need one ring_ready. My best guess, without seeing a pastebinned actual log, is that the gateway is returning "early media" right away - the ring, even before it connects, which FS by default counts as answering the call. You can add {ignore_early_media=true} to the beginning of the bridge string and try it again. (http://wiki.freeswitch.org/wiki/Variable_ignore_early_media) -Avi On Sun, Dec 11, 2011 at 9:10 PM, Elliott Vogel wrote: > Hello, > > I seem to have a problem when I try to bridge via " user/ > 1900 at zoogmedia.com,sofia/gateway/TSG/+13124346168" FS stops ringing the > user when it rings number on the gateway. Do I not have something > configured correctly? > > > /> > > > > > > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul Cupis > Sent: Saturday, December 03, 2011 1:33 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] dialplan > > On 03/12/11 17:28, Elliott Vogel wrote: > > How can I bridge out calls via another bridge if the first bridge > > fails? - let's say the first bridge returns a 404 or 503 error > > Have a look at the following: > > > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_7:_Action_failover_on_failed_action > > http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail > > 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/20111211/a115228e/attachment-0001.html From elliott at zoogmedia.com Mon Dec 12 01:53:54 2011 From: elliott at zoogmedia.com (Elliott Vogel) Date: Sun, 11 Dec 2011 22:53:54 +0000 Subject: [Freeswitch-users] dialplan In-Reply-To: References: <4EDA7976.1020803@cupis.co.uk> Message-ID: That worked, thanks... is there any problem with always ignoring early media? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Sunday, December 11, 2011 1:39 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] dialplan You only need one ring_ready. My best guess, without seeing a pastebinned actual log, is that the gateway is returning "early media" right away - the ring, even before it connects, which FS by default counts as answering the call. You can add {ignore_early_media=true} to the beginning of the bridge string and try it again. (http://wiki.freeswitch.org/wiki/Variable_ignore_early_media) -Avi On Sun, Dec 11, 2011 at 9:10 PM, Elliott Vogel > wrote: Hello, I seem to have a problem when I try to bridge via " user/1900 at zoogmedia.com,sofia/gateway/TSG/+13124346168" FS stops ringing the user when it rings number on the gateway. Do I not have something configured correctly? -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul Cupis Sent: Saturday, December 03, 2011 1:33 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] dialplan On 03/12/11 17:28, Elliott Vogel wrote: > How can I bridge out calls via another bridge if the first bridge > fails? - let's say the first bridge returns a 404 or 503 error Have a look at the following: http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_7:_Action_failover_on_failed_action http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail 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/20111211/267c7d95/attachment.html From avi at avimarcus.net Mon Dec 12 01:58:36 2011 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 12 Dec 2011 00:58:36 +0200 Subject: [Freeswitch-users] dialplan In-Reply-To: References: <4EDA7976.1020803@cupis.co.uk> Message-ID: Apparently Amex's 800 line has a bunch of messages in early media to save them some money... but in general, you aren't supposed to NEED it... -Avi On Mon, Dec 12, 2011 at 12:53 AM, Elliott Vogel wrote: > That worked, thanks? is there any problem with always ignoring early > media?**** > > ** ** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Sunday, December 11, 2011 1:39 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] dialplan**** > > ** ** > > You only need one ring_ready.**** > > My best guess, without seeing a pastebinned actual log, is that the > gateway is returning "early media" right away - the ring, even before it > connects, which FS by default counts as answering the call.**** > > You can add {ignore_early_media=true} to the beginning of the bridge > string and try it again.**** > > (http://wiki.freeswitch.org/wiki/Variable_ignore_early_media)**** > > ** ** > > -Avi **** > > ** ** > > On Sun, Dec 11, 2011 at 9:10 PM, Elliott Vogel > wrote:**** > > Hello, > > I seem to have a problem when I try to bridge via " user/ > 1900 at zoogmedia.com,sofia/gateway/TSG/+13124346168" FS stops ringing the > user when it rings number on the gateway. Do I not have something > configured correctly? > > > /> > > > > > > **** > > > -----Original Message----- > From: freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Paul Cupis > Sent: Saturday, December 03, 2011 1:33 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] dialplan > > On 03/12/11 17:28, Elliott Vogel wrote: > > How can I bridge out calls via another bridge if the first bridge > > fails? - let's say the first bridge returns a 404 or 503 error > > Have a look at the following: > > > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_7:_Action_failover_on_failed_action > > http://wiki.freeswitch.org/wiki/Channel_Variables#continue_on_fail > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111212/df41691b/attachment-0001.html From sdame at 207me.com Mon Dec 12 02:58:53 2011 From: sdame at 207me.com (Stephen Dame) Date: Sun, 11 Dec 2011 18:58:53 -0500 Subject: [Freeswitch-users] conference xxxx record .mp3 file is shorter than actual time. Message-ID: <021601ccb860$d6d8b7c0$848a2740$@com> Quick question.. Recording 1hr 55min conference is only 1:49 long. No voice data appears to be lost. Not sure if it's a sampling issue, or silence not getting recorded in .mp3.. I have tried a few times and looks like it's about 10% shorter and tried with 16,000 and 48,000 conferences I thought it might have been energy setting, set to 0. I'm also recording out to shout: and can see data 100% of time transmitting in media player. Is there a setting in freeswitch, or lame to ensure exact timing on recorded file. I'm setting recording from fs_cli with conference xxxxx record path/file.mp3 Not sure how to set a different rate in fs_cli for recording. Thanks in advance. Regards, Stephen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111211/0fdf9f57/attachment.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/gif Size: 145 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111211/0fdf9f57/attachment.gif From quiet.ordinary.man at gmail.com Sun Dec 11 13:54:43 2011 From: quiet.ordinary.man at gmail.com (T T) Date: Sun, 11 Dec 2011 12:54:43 +0200 Subject: [Freeswitch-users] Freeswitch on system startup on Fedora 16 Message-ID: Hi, I've installed FreeSWITCH on Fedora 16. I want FreeSWITCH to be launched at system startup. I've put a bash script from http://wiki.freeswitch.org/wiki/Freeswitch_init#Fedora into /etc/init.d directory with corrected paths, and run #chckconfig freeswitch on FreeSWITCH doesnt start at system startup. Running #service freeswitch start Starting freeswitch (via systemctl): [ OK ] ... but FreeSWITCH doesnt actually start. Running #systemctl enable freeswitch.service freeswitch.service is not a native service, redirecting to /sbin/chkconfig. Executing /sbin/chkconfig freeswitch on Warning: unit files do not carry install information. No operation executed. Running # chkconfig --list freeswitch Note: This output shows SysV services only and does not include native systemd services. SysV configuration data might be overridden by native systemd configuration. freeswitch 0:off 1:off 2:on 3:on 4:on 5:on 6:off Running #/usr/local/freeswitch/bin/freeswitch works fine. Please help. Thanks. The init script is below: cat /etc/init.d/freeswitch #! /bin/sh # # freeswitch: ? ? ? Starts the freeswitch Daemon # # chkconfig: 345 96 02 # processname: freeswitch # description: Freeswitch fedora init script \ # config: # Author: gled # Source function library. . /etc/init.d/functions . /etc/sysconfig/network PATH=/sbin:/usr/sbin:/bin:/usr/bin:/usr/local/freeswitch/bin DESC="FreeSwitch Voice Switching System" NAME=freeswitch DAEMON=/usr/local/freeswitch/bin/$NAME DAEMON_ARGS="-nc" PIDFILE=/var/run/freeswitch/$NAME.pid ## SECURITY NOTE: To run as non-root, create a new user for FreeSWITCH and set these variables (FS_GROUP is optional). ## FS_USER=freeswitch #FS_GROUP=freeswitch do_setlimits() { ? ? ? ?ulimit -c unlimited ? ? ? ?ulimit -d unlimited ? ? ? ?ulimit -f unlimited ? ? ? ?ulimit -i unlimited ? ? ? ?ulimit -n 999999 ? ? ? ?ulimit -q unlimited ? ? ? ?ulimit -u unlimited ? ? ? ?ulimit -v unlimited ? ? ? ?ulimit -x unlimited ? ? ? ?ulimit -s 244 ? ? ? ?ulimit -l unlimited ? ? ? ?return 0 } base=${0##*/} do_start() { ? ? ? ?if [ -n "${FS_USER}" ]; then ? ? ? ? ? ? ? ?DAEMON_ARGS="${DAEMON_ARGS} -u ${FS_USER}" ? ? ? ?fi ? ? ? ?if [ -n "${FS_GROUP}" ]; then ? ? ? ? ? ? ? ?DAEMON_ARGS="${DAEMON_ARGS} -g ${FS_GROUP}" ? ? ? ?fi ? ? ? ?do_setlimits ? ? ? ?$DAEMON $DAEMON_ARGS ? ? ? ?RETVAL=$? ? ? ? ?if [ $RETVAL = 0 ]; then ? ? ? ? ? ? ? ?success $"$base startup" ? ? ? ?else ? ? ? ? ? ? ? ?failure $"$base startup" ? ? ? ?fi ? ? ? ?echo ? ? ? ?return $RETVAL } do_stop() { ? ? ? ?$DAEMON -stop ? ? ? ?RETVAL=$? ? ? ? ?[ $RETVAL = 0 ] && success $"$base shutdown" || failure $"$base shutdown" ? ? ? ?rm -f $LOCKFILE ? ? ? ?echo ? ? ? ?return $RETVAL } # See how we were called. case "$1" in ?start) ? ? ? ?do_start ? ? ? ?;; ?stop) ? ? ? ?do_stop ? ? ? ?;; ?restart) ? ? ? ?do_stop ? ? ? ?echo "Waiting for daemon to exit..." ? ? ? ?sleep 5 ? ? ? ?do_start ? ? ? ?;; ?*) ? ? ? ?echo $"Usage: $0 {start|stop}" ? ? ? ?exit 2 ? ? ? ?;; esac exit $RETVAL From ormas at optusnet.com.au Mon Dec 12 08:40:36 2011 From: ormas at optusnet.com.au (Jayden) Date: Sun, 11 Dec 2011 21:40:36 -0800 (PST) Subject: [Freeswitch-users] Calls failing using Second Life Viewer and Parcels Message-ID: <1323668436505-7085039.post@n2.nabble.com> Hi everyone, I am new to Freeswitch and am a first time poster here so any help would be much appreciated. I have tried to search for similar problems but have had no success. I have recently started using Freeswitch 1.0.head (2011-11-30) on Windows because I had trouble compiling the source version obtained from git. I am using Opensim version 0.7.2 and Second Life Viewer 3.2.1 and I am using parcels to restrict communication to certain areas. The problem I am having is that when I move through parcels too quickly the call will fail to reconnect to the Freeswitch service and at this point I have not been able to find a way to reconnect without logging out and back in to the second life client. I realise this is not purely due to Freeswitch, as when using Hippo Viewer (0.6.3) I seem to be unable to "break" the connection in this manner, however Hippo Viewer is not an option because it does not allow the viewing of mesh objects. If anyone can suggest a way that I might be able to get around this by either changing some Freeswitch configurations or Second Life settings that would be great. Otherwise, if there are any ways to simply reconnect to the service without logging in and out of Second Life that would still be of great help, as the project I am working on will be used as a training simulation and logging in and out is not likely to be an acceptable solution by the client. If you would like to help and would like to see any console outputs just let me know and I will try to provide them. Thanks in advance, Jayden -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Calls-failing-using-Second-Life-Viewer-and-Parcels-tp7085039p7085039.html Sent from the freeswitch-users mailing list archive at Nabble.com. From freeswitch-list at puzzled.xs4all.nl Mon Dec 12 16:24:49 2011 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Mon, 12 Dec 2011 14:24:49 +0100 Subject: [Freeswitch-users] Freeswitch on system startup on Fedora 16 In-Reply-To: References: Message-ID: <4EE600A1.2020205@puzzled.xs4all.nl> On 11-12-11 11:54, T T wrote: > Hi, > > I've installed FreeSWITCH on Fedora 16. I want FreeSWITCH to be > launched at system startup. > > I've put a bash script from > http://wiki.freeswitch.org/wiki/Freeswitch_init#Fedora > > into /etc/init.d directory with corrected paths, and run > #chckconfig freeswitch on > > FreeSWITCH doesnt start at system startup. If you have SELinux enabled, did you make sure that the init script has the proper SELinux context? Regards, Patrick From acosgrov at gmail.com Mon Dec 12 16:26:26 2011 From: acosgrov at gmail.com (Anthony Cosgrove) Date: Mon, 12 Dec 2011 08:26:26 -0500 Subject: [Freeswitch-users] Error in my_thread_global_end(): nn threads didn't exit Message-ID: <90DEB7E9-6233-4C3F-B317-A7F524710DBE@gmail.com> Hey folks, So I'm using Lua to process fax calls but I've noticed that after the call has ended and things have settled down I get the following message on the console: Error in my_thread_global_end(): nn threads didn't exit From the testing I've been doing it seems to happen when I use the ODBC functions of freeswitch.Dbh() -- I've posted my debug output on pastebin but it didn't reveal much if anything. Log output @ http://pastebin.freeswitch.org/17995 Lua code being used @ http://pastebin.freeswitch.org/17997 Has anyone else gotten this message with their ODBC functions? Threads not exiting = menos buenos Thanks, Anthony (zorprime) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111212/07bc3ca5/attachment.html From freeswitch-list at puzzled.xs4all.nl Mon Dec 12 16:44:16 2011 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Mon, 12 Dec 2011 14:44:16 +0100 Subject: [Freeswitch-users] Calls failing using Second Life Viewer and Parcels In-Reply-To: <1323668436505-7085039.post@n2.nabble.com> References: <1323668436505-7085039.post@n2.nabble.com> Message-ID: <4EE60530.4020407@puzzled.xs4all.nl> Hi Jayden, I don't use Windows so can be of little help but inline there are some pointers that may make it easier for others to be able to help you. On 12-12-11 06:40, Jayden wrote: [snip] > I have recently started using Freeswitch 1.0.head (2011-11-30) on Windows > because I had trouble compiling the source version obtained from git. I am Did you check the wiki how to build FreeSWITCH on Windows: http://wiki.freeswitch.org/wiki/Installation_for_Windows If you followed that guide and still have issues building latest git then maybe you can send your setup, git version and errors/error logs to the mailing list? If there are no issues reported then it's quite difficult for developers to know if something might need fixing. > using Opensim version 0.7.2 and Second Life Viewer 3.2.1 and I am using > parcels to restrict communication to certain areas. Although I am not I guess those familiar with this stuff know these apps. But for those who don't it helps if you explain it a bit and possibly provide links to Opensim and Second Life Viewer. I had to Google for Opensim and thought that Second Life was dead. > The problem I am having is that when I move through parcels too quickly the > call will fail to reconnect to the Freeswitch service and at this point I No idea what "parcels" are. Again maybe that's just me being unfamiliar with these apps. Maybe you could provide (FreeSWITCH) logs of everything involved which shows the error. Good luck! Regards, Patrick From ted at schober.us Mon Dec 12 16:50:15 2011 From: ted at schober.us (Ted Schober) Date: Mon, 12 Dec 2011 08:50:15 -0500 Subject: [Freeswitch-users] Problems with local phones not registering Message-ID: <4EE60697.8030003@schober.us> I have freeswitch running with the internal and dialplan .xml files near virgin. (I added PASSWORD.xml as suggested in the Wiki One account on each of two of my Grandstream GXP-2000 phones on the eth0 local net of 192.168.61.X register just fine to default extensions of the dialplan. They get their addresses from a home routher by DHCP. I have been unable to get any other accounts on these phones, or any additional phones to register. The second phone registers on port 5062 (the default for account 2) for some reason. It will not register on 5060 even if I turn off account 2. The registered phones call and receive calls from each other and the provider gateway. I have checked the phone options, and they are identical except for the phone IP, the phone extension number. The FS server does have three NIC cards - one on another segment, and one that is disconnected. Any ideas where to look for the problem? Thanks! -------------- next part -------------- A non-text attachment was scrubbed... Name: ted.vcf Type: text/x-vcard Size: 290 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111212/da727283/attachment-0001.vcf From sherifomran2000 at yahoo.com Mon Dec 12 17:22:47 2011 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Mon, 12 Dec 2011 06:22:47 -0800 (PST) Subject: [Freeswitch-users] [INCOMPATIBLE_DESTINATION] Call failed - Fritzbox/Freeswitch - help required Message-ID: <1323699767.23161.YahooMailClassic@web110811.mail.gq1.yahoo.com> Hello I am trying to configure Freeswitch, so that 2 telephones each one is connected to a fritzbox and at different locations can speak. The call rings but when the callee lifts up, call drops. I would appreciate if any body can help. Below is a log of the call Thanks in advance regards, Sherif 2011-12-12 07:38:32.098496 [DEBUG] switch_core_session.c:1164 Session 45 (sofia/sipinterface_1/sip:2004 at 78.138.90.31) Locked, Waiting on external entities 2011-12-12 07:38:32.101287 [DEBUG] switch_core_codec.c:146 sofia/sipinterface_2/2005 at sip.server.com Restore previous codec PCMA:8. 2011-12-12 07:38:32.101287 [DEBUG] switch_ivr_originate.c:3228 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] 2011-12-12 07:38:32.101287 [ERR] switch_ivr_originate.c:2430 Cannot create outgoing channel of type [user] cause: [INCOMPATIBLE_DESTINATION] 2011-12-12 07:38:32.101287 [DEBUG] switch_ivr_originate.c:3228 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] 2011-12-12 07:38:32.101287 [INFO] mod_dptools.c:2355 Originate Failed.? Cause: INCOMPATIBLE_DESTINATION 2011-12-12 07:38:32.101287 [NOTICE] mod_dptools.c:2418 Hangup sofia/sipinterface_2/2005 at sip.server.com [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] 2011-12-12 07:38:32.101287 [DEBUG] switch_channel.c:2102 Send signal sofia/sipinterface_2/2005 at sip.server.com [KILL] 2011-12-12 07:38:32.101287 [DEBUG] switch_core_session.c:1021 Send signal sofia/sipinterface_2/2005 at sip.server.com [BREAK] 2011-12-12 07:38:32.101287 [DEBUG] switch_core_state_machine.c:348 (sofia/sipinterface_2/2005 at sip.server.com) State EXECUTE going to sleep 2011-12-12 07:38:32.101287 [DEBUG] switch_core_state_machine.c:314 (sofia/sipinterface_2/2005 at sip.server.com) Running State Change CS_HANGUP 2011-12-12 07:38:32.101287 [DEBUG] switch_core_state_machine.c:499 (sofia/sipinterface_2/2005 at sip.server.com) State HANGUP 2011-12-12 07:38:32.101287 [DEBUG] mod_sofia.c:408 sofia/sipinterface_2/2005 at sip.server.com Overriding SIP cause 488 with 488 from the other leg 2011-12-12 07:38:32.101287 [DEBUG] mod_sofia.c:414 Channel sofia/sipinterface_2/2005 at sip.server.com hanging up, cause: INCOMPATIBLE_DESTINATION 2011-12-12 07:38:32.101287 [NOTICE] switch_core_session.c:1182 Session 45 (sofia/sipinterface_1/sip:2004 at 78.138.90.31) Ended 2011-12-12 07:38:32.101287 [NOTICE] switch_core_session.c:1184 Close Channel sofia/sipinterface_1/sip:2004 at 78.138.90.31 [CS_DESTROY] 2011-12-12 07:38:32.101287 [DEBUG] switch_core_state_machine.c:428 (sofia/sipinterface_1/sip:2004 at 78.138.90.31) Running State Change CS_DESTROY 2011-12-12 07:38:32.101287 [DEBUG] switch_core_state_machine.c:439 (sofia/sipinterface_1/sip:2004 at 78.138.90.31) State DESTROY 2011-12-12 07:38:32.101287 [DEBUG] mod_sofia.c:341 sofia/sipinterface_1/sip:2004 at 78.138.90.31 SOFIA DESTROY 2011-12-12 07:38:32.101287 [DEBUG] switch_core_state_machine.c:60 sofia/sipinterface_1/sip:2004 at 78.138.90.31 Standard DESTROY 2011-12-12 07:38:32.101287 [DEBUG] switch_core_state_machine.c:439 (sofia/sipinterface_1/sip:2004 at 78.138.90.31) State DESTROY going to sleep 2011-12-12 07:38:32.107730 [DEBUG] mod_sofia.c:476 Responding to INVITE with: 488 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:46 sofia/sipinterface_2/2005 at sip.server.com Standard HANGUP, cause: INCOMPATIBLE_DESTINATION 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:499 (sofia/sipinterface_2/2005 at sip.server.com) State HANGUP going to sleep 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:333 (sofia/sipinterface_2/2005 at sip.server.com) State Change CS_HANGUP -> CS_REPORTING 2011-12-12 07:38:32.107730 [DEBUG] switch_core_session.c:1021 Send signal sofia/sipinterface_2/2005 at sip.server.com [BREAK] 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:314 (sofia/sipinterface_2/2005 at sip.server.com) Running State Change CS_REPORTING 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:590 (sofia/sipinterface_2/2005 at sip.server.com) State REPORTING 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:53 sofia/sipinterface_2/2005 at sip.server.com Standard REPORTING, cause: INCOMPATIBLE_DESTINATION 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:590 (sofia/sipinterface_2/2005 at sip.server.com) State REPORTING going to sleep 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:327 (sofia/sipinterface_2/2005 at sip.server.com) State Change CS_REPORTING -> CS_DESTROY 2011-12-12 07:38:32.107730 [DEBUG] switch_core_session.c:1021 Send signal sofia/sipinterface_2/2005 at sip.server.com [BREAK] 2011-12-12 07:38:32.107730 [DEBUG] switch_core_session.c:1164 Session 44 (sofia/sipinterface_2/2005 at sip.server.com) Locked, Waiting on external entities 2011-12-12 07:38:32.107730 [NOTICE] switch_core_session.c:1182 Session 44 (sofia/sipinterface_2/2005 at sip.server.com) Ended 2011-12-12 07:38:32.107730 [NOTICE] switch_core_session.c:1184 Close Channel sofia/sipinterface_2/2005 at sip.server.com [CS_DESTROY] 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:428 (sofia/sipinterface_2/2005 at sip.server.com) Running State Change CS_DESTROY 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:439 (sofia/sipinterface_2/2005 at sip.server.com) State DESTROY 2011-12-12 07:38:32.107730 [DEBUG] mod_sofia.c:341 sofia/sipinterface_2/2005 at sip.server.com SOFIA DESTROY 2011-12-12 07:38:32.107730 [DEBUG] switch_rtp.c:584? [ zrtp engine]: STOP STREAM ID=10 mode=CLEAR state=NOZRTP. 2011-12-12 07:38:32.107730 [DEBUG] switch_rtp.c:584? [??????? zrtp]: ??? Stream ID=0 UNKNOWN switching ---> . 2011-12-12 07:38:32.107730 [DEBUG] switch_rtp.c:584? [ zrtp engine]: STOP STREAM ID=0 mode=UNKNOWN state=NONE. 2011-12-12 07:38:32.107730 [DEBUG] switch_rtp.c:584? [ zrtp engine]: STOP STREAM ID=0 mode=UNKNOWN state=NONE. 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:60 sofia/sipinterface_2/2005 at sip.server.com Standard DESTROY 2011-12-12 07:38:32.107730 [DEBUG] switch_core_state_machine.c:439 (sofia/sipinterface_2/2005 at sip.server.com) State DESTROY going to sleep -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111212/604b8a53/attachment.html From msc at freeswitch.org Mon Dec 12 19:28:16 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 12 Dec 2011 08:28:16 -0800 Subject: [Freeswitch-users] Invalid SAY Interface [en] In-Reply-To: <1323525061803-7081463.post@n2.nabble.com> References: <1323459840645-7079510.post@n2.nabble.com> <1323525061803-7081463.post@n2.nabble.com> Message-ID: On Sat, Dec 10, 2011 at 5:51 AM, mazilo wrote: > > gautam wrote > > > > I had actually found the relevant documentation on the wiki. The problem > > is > > that I'm running the system on Alpine linux, and I installed Freeswitch > > using their package management utility (apk add freeswitch, etc). All the > > modules were present initially, but now mod_say_en has disappeared > > somehow. > > If I try to use "make" it says "no rule to make mod_say_en-install." I > > would really appreciate it if you could give me any direction to solve > > this. > > > > -Gautam > If you do like that, you will probably create more problems to the software > package manager on your system. Anyway, I don't use Alpine Linux distro. > However, I believe you can rebuild/recompile the FS package using its > software package management specs file. On a Linux distro that uses RPM, > i.e. RedHat, OpenSuSE, etc., each software package comes with its editable > plain-text RPM specs file and can be rebuilt/recompiled using its rpmbuild > utility. I believe Alpine Linux distro has its own software package manager > utility to do the same thing. Then, you can use the software package > manager > to upgrade your FS installation with the newly built FS package. > FWIW, most of us in the FS community build it from the sources. FreeSWITCH isn't a simple little utility - it's a big time piece of software with more than a few dependencies. Personally, I feel much better knowing that I've built FS on the actual hardware on which it will be running. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111212/ba730b14/attachment.html From anthony.minessale at gmail.com Mon Dec 12 19:48:36 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 12 Dec 2011 10:48:36 -0600 Subject: [Freeswitch-users] Error in my_thread_global_end(): nn threads didn't exit In-Reply-To: <90DEB7E9-6233-4C3F-B317-A7F524710DBE@gmail.com> References: <90DEB7E9-6233-4C3F-B317-A7F524710DBE@gmail.com> Message-ID: this is a mysql thing. make sure your odbc driver points at the _r version of the driver in odbcinst.ini On Mon, Dec 12, 2011 at 7:26 AM, Anthony Cosgrove wrote: > Hey folks, > > So I'm using Lua to process fax calls but I've noticed that after the call > has ended and things have settled down I get the following message on the > console: Error in my_thread_global_end(): nn threads didn't exit > > From the testing I've been doing it seems to happen when I use the ODBC > functions of freeswitch.Dbh() -- I've posted my debug output on pastebin > but it didn't reveal much if anything. > > Log output @ http://pastebin.freeswitch.org/17995 > Lua code being used @ http://pastebin.freeswitch.org/17997 > > Has anyone else gotten this message with their ODBC functions? Threads not > exiting = menos buenos > > > Thanks, > > Anthony (zorprime) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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/20111212/7cdf316a/attachment.html From Nabble_01394 at slickdeals.endjunk.com Mon Dec 12 20:37:15 2011 From: Nabble_01394 at slickdeals.endjunk.com (mazilo) Date: Mon, 12 Dec 2011 09:37:15 -0800 (PST) Subject: [Freeswitch-users] spandsp/src/dtmf.c: 'duration' isn't a membership of 'dtmf_rx_state_t'? In-Reply-To: References: <1323356249725-7074568.post@n2.nabble.com> <4EE0E41A.4050406@coppice.org> <1323428742912-7077805.post@n2.nabble.com> Message-ID: <1323711435567-7087024.post@n2.nabble.com> Anthony Minessale wrote > > make spandsp-reconf > to fix that Anthony, Thanks. However, I didn't have a problem to compile spandsp, except I just had a problem to comprehend some of the changes. OK, I followed your suggestion to do a /make spandsp-reconf/. Ironically, I do now see a (cross) compilation problem after the /make spandsp-reconf/ as shown below: ----- FreeSWITCH hosted on a Seagate DockStar with OpenWRT and ONLY consumes 3 Watts of electricity. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/spandsp-src-dtmf-c-duration-isn-t-a-membership-of-dtmf-rx-state-t-tp7074568p7087024.html Sent from the freeswitch-users mailing list archive at Nabble.com. From farooqhussain786 at gmail.com Mon Dec 12 21:36:02 2011 From: farooqhussain786 at gmail.com (Farooq Hussain) Date: Mon, 12 Dec 2011 23:36:02 +0500 Subject: [Freeswitch-users] Please help me Message-ID: hello everyone, I am using FXO Card with freeTDM. Now I can make calls from FXO and I want to receive a call from my cell phone but We are not getting where we went we wrong and sip error code destination out of order. -- Thanks Farooq Hussain -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111212/151eee33/attachment.html From acosgrov at gmail.com Mon Dec 12 22:15:21 2011 From: acosgrov at gmail.com (Anthony Cosgrove) Date: Mon, 12 Dec 2011 14:15:21 -0500 Subject: [Freeswitch-users] Error in my_thread_global_end(): nn threads didn't exit In-Reply-To: References: <90DEB7E9-6233-4C3F-B317-A7F524710DBE@gmail.com> Message-ID: <7CCF16CC-F0ED-4FEE-8D60-2105CC47D421@gmail.com> Hmm... I don't see an _r version of the odbc driver. There is an _r version for the libmysqlclient driver. NPX1601155 unixODBC # cat odbcinst.ini [myodbc-5.1] Description=MySQL ODBC myodbc-5.1.6 Driver Driver=/usr/lib/libmyodbc5.so UsageCount=1 NPX1601155 unixODBC # ls /usr/lib/libmy* -lah -rwxr-xr-x 1 root root 297K Dec 10 08:48 /usr/lib/libmyodbc5-5.1.6.so -rw-r--r-- 1 root root 465K Dec 10 08:48 /usr/lib/libmyodbc5.a -rw-r--r-- 1 root root 1.1K Dec 10 08:48 /usr/lib/libmyodbc5.la lrwxrwxrwx 1 root root 19 Dec 10 08:48 /usr/lib/libmyodbc5.so -> libmyodbc5-5.1.6.so lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient -> mysql/libmysqlclient.so.16.0.0 lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient.so -> mysql/libmysqlclient.so.16.0.0 lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient.so.16 -> mysql/libmysqlclient.so.16.0.0 lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient.so.16.0 -> mysql/libmysqlclient.so.16.0.0 lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient.so.16.0.0 -> mysql/libmysqlclient.so.16.0.0 lrwxrwxrwx 1 root root 32 Dec 10 08:21 /usr/lib/libmysqlclient_r -> mysql/libmysqlclient_r.so.16.0.0 lrwxrwxrwx 1 root root 32 Dec 10 08:21 /usr/lib/libmysqlclient_r.so -> mysql/libmysqlclient_r.so.16.0.0 lrwxrwxrwx 1 root root 32 Dec 10 08:21 /usr/lib/libmysqlclient_r.so.16 -> mysql/libmysqlclient_r.so.16.0.0 lrwxrwxrwx 1 root root 32 Dec 10 08:21 /usr/lib/libmysqlclient_r.so.16.0 -> mysql/libmysqlclient_r.so.16.0.0 lrwxrwxrwx 1 root root 32 Dec 10 08:21 /usr/lib/libmysqlclient_r.so.16.0.0 -> mysql/libmysqlclient_r.so.16.0.0 Anthony C On Dec 12, 2011, at 11:48 AM, Anthony Minessale wrote: > this is a mysql thing. > make sure your odbc driver points at the _r version of the driver in odbcinst.ini > > > On Mon, Dec 12, 2011 at 7:26 AM, Anthony Cosgrove wrote: > Hey folks, > > So I'm using Lua to process fax calls but I've noticed that after the call has ended and things have settled down I get the following message on the console: Error in my_thread_global_end(): nn threads didn't exit > > From the testing I've been doing it seems to happen when I use the ODBC functions of freeswitch.Dbh() -- I've posted my debug output on pastebin but it didn't reveal much if anything. > > Log output @ http://pastebin.freeswitch.org/17995 > Lua code being used @ http://pastebin.freeswitch.org/17997 > > Has anyone else gotten this message with their ODBC functions? Threads not exiting = menos buenos > > > Thanks, > > Anthony (zorprime) > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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/20111212/3b800bf2/attachment.html From brad at tech21.com Mon Dec 12 22:30:22 2011 From: brad at tech21.com (Brad Mina) Date: Mon, 12 Dec 2011 11:30:22 -0800 Subject: [Freeswitch-users] Please help me In-Reply-To: References: Message-ID: First off, you need to escape the + I believe. Second, you're calling the "say" TTS application before you answer, which means you're trying to send media using a channel that has not established media. You've still probably got a bit to go after fixing these, pastebin a log of your call to http://pastebin.freeswitch.org Sent from my iPhone On Dec 12, 2011, at 10:36 AM, Farooq Hussain wrote: > hello everyone, > > I am using FXO Card with freeTDM. Now I can make calls from FXO and > I want to receive a call from my cell phone but We are not getting > where we went we wrong and sip error code destination out of order. > > > > > > > > > > > > -- > Thanks > > Farooq Hussain > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > users > http://www.freeswitch.org From quiet.ordinary.man at gmail.com Mon Dec 12 22:49:03 2011 From: quiet.ordinary.man at gmail.com (T T) Date: Mon, 12 Dec 2011 21:49:03 +0200 Subject: [Freeswitch-users] Freeswitch on system startup on Fedora 16 In-Reply-To: <4EE600A1.2020205@puzzled.xs4all.nl> References: <4EE600A1.2020205@puzzled.xs4all.nl> Message-ID: Hi, SELinux isn't the cause. Setting it to 'permissive' doesn't make it work after reboot. T. On Mon, Dec 12, 2011 at 3:24 PM, Patrick Lists wrote: > On 11-12-11 11:54, T T wrote: >> Hi, >> >> I've installed FreeSWITCH on Fedora 16. I want FreeSWITCH to be >> launched ?at system startup. >> >> I've put a bash script from >> http://wiki.freeswitch.org/wiki/Freeswitch_init#Fedora >> >> into /etc/init.d directory with corrected paths, and run >> #chckconfig freeswitch on >> >> FreeSWITCH doesnt start at system startup. > > If you have SELinux enabled, did you make sure that the init script has > the proper SELinux context? > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From farooqhussain786 at gmail.com Mon Dec 12 23:26:36 2011 From: farooqhussain786 at gmail.com (Farooq Hussain) Date: Tue, 13 Dec 2011 01:26:36 +0500 Subject: [Freeswitch-users] Please help me In-Reply-To: References: Message-ID: Brad Mina, Thanks, for you help. I made changes in dial plan as per your instructions. But still not getting. When I check fs_cli log I am receive call on FXO using freetdm but every time i found in log that 'destination out of order'. One more thing my when I dial my cell number from local network its work fine I am able to receive call and voice work good. But When I try it from different location one of my in usa when they call me on my cell no voice send or receive. Please help me as I have to get this server up and running by the end of tomorrow. Thanks for you support. Farooq On Tue, Dec 13, 2011 at 12:30 AM, Brad Mina wrote: > First off, you need to escape the + I believe. > > Second, you're calling the "say" TTS application before you answer, > which means you're trying to send media using a channel that has not > established media. > > You've still probably got a bit to go after fixing these, pastebin a > log of your call to http://pastebin.freeswitch.org > > Sent from my iPhone > > On Dec 12, 2011, at 10:36 AM, Farooq Hussain > wrote: > > > hello everyone, > > > > I am using FXO Card with freeTDM. Now I can make calls from FXO and > > I want to receive a call from my cell phone but We are not getting > > where we went we wrong and sip error code destination out of order. > > > > > > > > > > > > > > > > > > > > > > > > -- > > Thanks > > > > Farooq Hussain > > _________________________________________________________________________ > > > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- > > users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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 Farooq Hussain -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111213/98b05e41/attachment-0001.html From anthony.minessale at gmail.com Mon Dec 12 23:54:14 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 12 Dec 2011 14:54:14 -0600 Subject: [Freeswitch-users] Error in my_thread_global_end(): nn threads didn't exit In-Reply-To: <7CCF16CC-F0ED-4FEE-8D60-2105CC47D421@gmail.com> References: <90DEB7E9-6233-4C3F-B317-A7F524710DBE@gmail.com> <7CCF16CC-F0ED-4FEE-8D60-2105CC47D421@gmail.com> Message-ID: yes that is the one I was referring to On Mon, Dec 12, 2011 at 1:15 PM, Anthony Cosgrove wrote: > Hmm... I don't see an _r version of the odbc driver. There is an _r > version for the libmysqlclient driver. > > NPX1601155 unixODBC # cat odbcinst.ini > [myodbc-5.1] > Description=MySQL ODBC myodbc-5.1.6 Driver > Driver=/usr/lib/libmyodbc5.so > UsageCount=1 > > NPX1601155 unixODBC # ls /usr/lib/libmy* -lah > -rwxr-xr-x 1 root root 297K Dec 10 08:48 /usr/lib/libmyodbc5-5.1.6.so > -rw-r--r-- 1 root root 465K Dec 10 08:48 /usr/lib/libmyodbc5.a > -rw-r--r-- 1 root root 1.1K Dec 10 08:48 /usr/lib/libmyodbc5.la > lrwxrwxrwx 1 root root 19 Dec 10 08:48 /usr/lib/libmyodbc5.so -> > libmyodbc5-5.1.6.so > lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient -> > mysql/libmysqlclient.so.16.0.0 > lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient.so -> > mysql/libmysqlclient.so.16.0.0 > lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient.so.16 -> > mysql/libmysqlclient.so.16.0.0 > lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient.so.16.0 > -> mysql/libmysqlclient.so.16.0.0 > lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient.so.16.0.0 > -> mysql/libmysqlclient.so.16.0.0 > lrwxrwxrwx 1 root root 32 Dec 10 08:21 /usr/lib/libmysqlclient_r -> > mysql/libmysqlclient_r.so.16.0.0 > lrwxrwxrwx 1 root root 32 Dec 10 08:21 /usr/lib/libmysqlclient_r.so -> > mysql/libmysqlclient_r.so.16.0.0 > lrwxrwxrwx 1 root root 32 Dec 10 08:21 /usr/lib/libmysqlclient_r.so.16 > -> mysql/libmysqlclient_r.so.16.0.0 > lrwxrwxrwx 1 root root 32 Dec 10 08:21 /usr/lib/libmysqlclient_r.so.16.0 > -> mysql/libmysqlclient_r.so.16.0.0 > lrwxrwxrwx 1 root root 32 Dec 10 08:21 > /usr/lib/libmysqlclient_r.so.16.0.0 -> mysql/libmysqlclient_r.so.16.0.0 > > Anthony C > > On Dec 12, 2011, at 11:48 AM, Anthony Minessale wrote: > > this is a mysql thing. > make sure your odbc driver points at the _r version of the driver in > odbcinst.ini > > > On Mon, Dec 12, 2011 at 7:26 AM, Anthony Cosgrove wrote: > >> Hey folks, >> >> So I'm using Lua to process fax calls but I've noticed that after the >> call has ended and things have settled down I get the following message on >> the console: Error in my_thread_global_end(): nn threads didn't exit >> >> From the testing I've been doing it seems to happen when I use the ODBC >> functions of freeswitch.Dbh() -- I've posted my debug output on pastebin >> but it didn't reveal much if anything. >> >> Log output @ http://pastebin.freeswitch.org/17995 >> Lua code being used @ http://pastebin.freeswitch.org/17997 >> >> Has anyone else gotten this message with their ODBC functions? Threads >> not exiting = menos buenos >> >> >> Thanks, >> >> Anthony (zorprime) >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://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 -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111212/34bc50d0/attachment.html From brad at tech21.com Tue Dec 13 00:07:46 2011 From: brad at tech21.com (Brad Mina) Date: Mon, 12 Dec 2011 13:07:46 -0800 Subject: [Freeswitch-users] Please help me In-Reply-To: References: Message-ID: Without a log of the call the description of events is vague. Copy the log starting with that session and paste it to http://pastebin.freeswitch.org Reply back to this thread with your link. On Mon, Dec 12, 2011 at 12:26 PM, Farooq Hussain wrote: > Brad Mina, > > Thanks, for you help. I made changes in dial plan as per your > instructions. But still not getting. When I check fs_cli log I am receive > call on FXO using freetdm but every time i found in log that > 'destination out of order'. One more thing my when I dial my cell number > from local network its work fine I am able to receive call and voice work > good. But When I try it from different location one of my in usa when they > call me on my cell no voice send or receive. > > Please help me as I have to get this server up and running by the end of > tomorrow. > > Thanks for you support. > > Farooq > > > On Tue, Dec 13, 2011 at 12:30 AM, Brad Mina wrote: > >> First off, you need to escape the + I believe. >> >> Second, you're calling the "say" TTS application before you answer, >> which means you're trying to send media using a channel that has not >> established media. >> >> You've still probably got a bit to go after fixing these, pastebin a >> log of your call to http://pastebin.freeswitch.org >> >> Sent from my iPhone >> >> On Dec 12, 2011, at 10:36 AM, Farooq Hussain >> wrote: >> >> > hello everyone, >> > >> > I am using FXO Card with freeTDM. Now I can make calls from FXO and >> > I want to receive a call from my cell phone but We are not getting >> > where we went we wrong and sip error code destination out of order. >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > >> > -- >> > Thanks >> > >> > Farooq Hussain >> > >> _________________________________________________________________________ >> >> >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch- >> > users >> > http://www.freeswitch.org >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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 > > Farooq Hussain > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111212/25c4bac5/attachment-0001.html From acosgrov at gmail.com Tue Dec 13 01:45:04 2011 From: acosgrov at gmail.com (Anthony Cosgrove) Date: Mon, 12 Dec 2011 17:45:04 -0500 Subject: [Freeswitch-users] Error in my_thread_global_end(): nn threads didn't exit In-Reply-To: References: <90DEB7E9-6233-4C3F-B317-A7F524710DBE@gmail.com> <7CCF16CC-F0ED-4FEE-8D60-2105CC47D421@gmail.com> Message-ID: <4AE50826-9174-41FA-B561-34932837FC3E@gmail.com> Bummer... this odbc driver is already linked to the thread-safe mysqlclient driver. Guess I'll need to go postgresql or something. NPX1601155 lib # ldd libmyodbc5-5.1.6.so linux-vdso.so.1 => (0x00007fffbd9ff000) libmysqlclient_r.so.16 => /usr/lib64/mysql/libmysqlclient_r.so.16 (0x00007f56dafed000) libodbcinst.so.1 => /usr/lib64/libodbcinst.so.1 (0x00007f56dadd0000) libltdl.so.7 => /usr/lib64/libltdl.so.7 (0x00007f56dabc6000) libpthread.so.0 => /lib64/libpthread.so.0 (0x00007f56da9a9000) libc.so.6 => /lib64/libc.so.6 (0x00007f56da63d000) libm.so.6 => /lib64/libm.so.6 (0x00007f56da3bb000) libz.so.1 => /lib64/libz.so.1 (0x00007f56da1a3000) libssl.so.1.0.0 => /usr/lib64/libssl.so.1.0.0 (0x00007f56d9f45000) libcrypto.so.1.0.0 => /usr/lib64/libcrypto.so.1.0.0 (0x00007f56d9b94000) libdl.so.2 => /lib64/libdl.so.2 (0x00007f56d9990000) /lib64/ld-linux-x86-64.so.2 (0x00007f56db5c0000) Versions used: unixODBC - 2.30-r1 myodbc - 5.1.6 mysql Ver 14.14 Distrib 5.1.56, for pc-linux-gnu (x86_64) using readline 5.1 Thanks, Anthony C On Dec 12, 2011, at 3:54 PM, Anthony Minessale wrote: > yes that is the one I was referring to > > On Mon, Dec 12, 2011 at 1:15 PM, Anthony Cosgrove wrote: > Hmm... I don't see an _r version of the odbc driver. There is an _r version for the libmysqlclient driver. > > NPX1601155 unixODBC # cat odbcinst.ini > [myodbc-5.1] > Description=MySQL ODBC myodbc-5.1.6 Driver > Driver=/usr/lib/libmyodbc5.so > UsageCount=1 > > NPX1601155 unixODBC # ls /usr/lib/libmy* -lah > -rwxr-xr-x 1 root root 297K Dec 10 08:48 /usr/lib/libmyodbc5-5.1.6.so > -rw-r--r-- 1 root root 465K Dec 10 08:48 /usr/lib/libmyodbc5.a > -rw-r--r-- 1 root root 1.1K Dec 10 08:48 /usr/lib/libmyodbc5.la > lrwxrwxrwx 1 root root 19 Dec 10 08:48 /usr/lib/libmyodbc5.so -> libmyodbc5-5.1.6.so > lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient -> mysql/libmysqlclient.so.16.0.0 > lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient.so -> mysql/libmysqlclient.so.16.0.0 > lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient.so.16 -> mysql/libmysqlclient.so.16.0.0 > lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient.so.16.0 -> mysql/libmysqlclient.so.16.0.0 > lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient.so.16.0.0 -> mysql/libmysqlclient.so.16.0.0 > lrwxrwxrwx 1 root root 32 Dec 10 08:21 /usr/lib/libmysqlclient_r -> mysql/libmysqlclient_r.so.16.0.0 > lrwxrwxrwx 1 root root 32 Dec 10 08:21 /usr/lib/libmysqlclient_r.so -> mysql/libmysqlclient_r.so.16.0.0 > lrwxrwxrwx 1 root root 32 Dec 10 08:21 /usr/lib/libmysqlclient_r.so.16 -> mysql/libmysqlclient_r.so.16.0.0 > lrwxrwxrwx 1 root root 32 Dec 10 08:21 /usr/lib/libmysqlclient_r.so.16.0 -> mysql/libmysqlclient_r.so.16.0.0 > lrwxrwxrwx 1 root root 32 Dec 10 08:21 /usr/lib/libmysqlclient_r.so.16.0.0 -> mysql/libmysqlclient_r.so.16.0.0 > > Anthony C > > On Dec 12, 2011, at 11:48 AM, Anthony Minessale wrote: > >> this is a mysql thing. >> make sure your odbc driver points at the _r version of the driver in odbcinst.ini >> >> >> On Mon, Dec 12, 2011 at 7:26 AM, Anthony Cosgrove wrote: >> Hey folks, >> >> So I'm using Lua to process fax calls but I've noticed that after the call has ended and things have settled down I get the following message on the console: Error in my_thread_global_end(): nn threads didn't exit >> >> From the testing I've been doing it seems to happen when I use the ODBC functions of freeswitch.Dbh() -- I've posted my debug output on pastebin but it didn't reveal much if anything. >> >> Log output @ http://pastebin.freeswitch.org/17995 >> Lua code being used @ http://pastebin.freeswitch.org/17997 >> >> Has anyone else gotten this message with their ODBC functions? Threads not exiting = menos buenos >> >> >> Thanks, >> >> Anthony (zorprime) >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://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/20111212/923c0e81/attachment.html From acosgrov at gmail.com Tue Dec 13 04:57:05 2011 From: acosgrov at gmail.com (Anthony Cosgrove) Date: Mon, 12 Dec 2011 20:57:05 -0500 Subject: [Freeswitch-users] {Solved} Error in my_thread_global_end(): nn threads didn't exit In-Reply-To: <4AE50826-9174-41FA-B561-34932837FC3E@gmail.com> References: <90DEB7E9-6233-4C3F-B317-A7F524710DBE@gmail.com> <7CCF16CC-F0ED-4FEE-8D60-2105CC47D421@gmail.com> <4AE50826-9174-41FA-B561-34932837FC3E@gmail.com> Message-ID: <32C00269-6FBC-483F-AF19-BCE0DE8F00BA@gmail.com> Yup so I landed up switching to postgresql 9.1 on my gentoo box. Been about 10 minutes and haven't seen the error message. I don't see an entry for postgres in the wiki so I shall "do the needful" and pay the tax :) Thanks Anthony AC On Dec 12, 2011, at 5:45 PM, Anthony Cosgrove wrote: > Bummer... this odbc driver is already linked to the thread-safe mysqlclient driver. Guess I'll need to go postgresql or something. > > NPX1601155 lib # ldd libmyodbc5-5.1.6.so > linux-vdso.so.1 => (0x00007fffbd9ff000) > libmysqlclient_r.so.16 => /usr/lib64/mysql/libmysqlclient_r.so.16 (0x00007f56dafed000) > libodbcinst.so.1 => /usr/lib64/libodbcinst.so.1 (0x00007f56dadd0000) > libltdl.so.7 => /usr/lib64/libltdl.so.7 (0x00007f56dabc6000) > libpthread.so.0 => /lib64/libpthread.so.0 (0x00007f56da9a9000) > libc.so.6 => /lib64/libc.so.6 (0x00007f56da63d000) > libm.so.6 => /lib64/libm.so.6 (0x00007f56da3bb000) > libz.so.1 => /lib64/libz.so.1 (0x00007f56da1a3000) > libssl.so.1.0.0 => /usr/lib64/libssl.so.1.0.0 (0x00007f56d9f45000) > libcrypto.so.1.0.0 => /usr/lib64/libcrypto.so.1.0.0 (0x00007f56d9b94000) > libdl.so.2 => /lib64/libdl.so.2 (0x00007f56d9990000) > /lib64/ld-linux-x86-64.so.2 (0x00007f56db5c0000) > > Versions used: > > unixODBC - 2.30-r1 > myodbc - 5.1.6 > mysql Ver 14.14 Distrib 5.1.56, for pc-linux-gnu (x86_64) using readline 5.1 > > Thanks, > > Anthony C > > On Dec 12, 2011, at 3:54 PM, Anthony Minessale wrote: > >> yes that is the one I was referring to >> >> On Mon, Dec 12, 2011 at 1:15 PM, Anthony Cosgrove wrote: >> Hmm... I don't see an _r version of the odbc driver. There is an _r version for the libmysqlclient driver. >> >> NPX1601155 unixODBC # cat odbcinst.ini >> [myodbc-5.1] >> Description=MySQL ODBC myodbc-5.1.6 Driver >> Driver=/usr/lib/libmyodbc5.so >> UsageCount=1 >> >> NPX1601155 unixODBC # ls /usr/lib/libmy* -lah >> -rwxr-xr-x 1 root root 297K Dec 10 08:48 /usr/lib/libmyodbc5-5.1.6.so >> -rw-r--r-- 1 root root 465K Dec 10 08:48 /usr/lib/libmyodbc5.a >> -rw-r--r-- 1 root root 1.1K Dec 10 08:48 /usr/lib/libmyodbc5.la >> lrwxrwxrwx 1 root root 19 Dec 10 08:48 /usr/lib/libmyodbc5.so -> libmyodbc5-5.1.6.so >> lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient -> mysql/libmysqlclient.so.16.0.0 >> lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient.so -> mysql/libmysqlclient.so.16.0.0 >> lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient.so.16 -> mysql/libmysqlclient.so.16.0.0 >> lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient.so.16.0 -> mysql/libmysqlclient.so.16.0.0 >> lrwxrwxrwx 1 root root 30 Dec 10 08:21 /usr/lib/libmysqlclient.so.16.0.0 -> mysql/libmysqlclient.so.16.0.0 >> lrwxrwxrwx 1 root root 32 Dec 10 08:21 /usr/lib/libmysqlclient_r -> mysql/libmysqlclient_r.so.16.0.0 >> lrwxrwxrwx 1 root root 32 Dec 10 08:21 /usr/lib/libmysqlclient_r.so -> mysql/libmysqlclient_r.so.16.0.0 >> lrwxrwxrwx 1 root root 32 Dec 10 08:21 /usr/lib/libmysqlclient_r.so.16 -> mysql/libmysqlclient_r.so.16.0.0 >> lrwxrwxrwx 1 root root 32 Dec 10 08:21 /usr/lib/libmysqlclient_r.so.16.0 -> mysql/libmysqlclient_r.so.16.0.0 >> lrwxrwxrwx 1 root root 32 Dec 10 08:21 /usr/lib/libmysqlclient_r.so.16.0.0 -> mysql/libmysqlclient_r.so.16.0.0 >> >> Anthony C >> >> On Dec 12, 2011, at 11:48 AM, Anthony Minessale wrote: >> >>> this is a mysql thing. >>> make sure your odbc driver points at the _r version of the driver in odbcinst.ini >>> >>> >>> On Mon, Dec 12, 2011 at 7:26 AM, Anthony Cosgrove wrote: >>> Hey folks, >>> >>> So I'm using Lua to process fax calls but I've noticed that after the call has ended and things have settled down I get the following message on the console: Error in my_thread_global_end(): nn threads didn't exit >>> >>> From the testing I've been doing it seems to happen when I use the ODBC functions of freeswitch.Dbh() -- I've posted my debug output on pastebin but it didn't reveal much if anything. >>> >>> Log output @ http://pastebin.freeswitch.org/17995 >>> Lua code being used @ http://pastebin.freeswitch.org/17997 >>> >>> Has anyone else gotten this message with their ODBC functions? Threads not exiting = menos buenos >>> >>> >>> Thanks, >>> >>> Anthony (zorprime) >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://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/20111212/0c6e8fd4/attachment-0001.html From curriegrad2004 at gmail.com Tue Dec 13 18:29:13 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 13 Dec 2011 07:29:13 -0800 Subject: [Freeswitch-users] Freeswitch on system startup on Fedora 16 In-Reply-To: References: <4EE600A1.2020205@puzzled.xs4all.nl> Message-ID: Have you tried to execute FreeSWITCH as normal under Fedora 16? On Mon, Dec 12, 2011 at 11:49 AM, T T wrote: > Hi, > > SELinux isn't the cause. Setting it to 'permissive' doesn't make it > work after reboot. > > T. > > On Mon, Dec 12, 2011 at 3:24 PM, Patrick Lists > wrote: >> On 11-12-11 11:54, T T wrote: >>> Hi, >>> >>> I've installed FreeSWITCH on Fedora 16. I want FreeSWITCH to be >>> launched ?at system startup. >>> >>> I've put a bash script from >>> http://wiki.freeswitch.org/wiki/Freeswitch_init#Fedora >>> >>> into /etc/init.d directory with corrected paths, and run >>> #chckconfig freeswitch on >>> >>> FreeSWITCH doesnt start at system startup. >> >> If you have SELinux enabled, did you make sure that the init script has >> the proper SELinux context? >> >> Regards, >> Patrick >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From freeswitch at pbaines.com Tue Dec 13 14:01:39 2011 From: freeswitch at pbaines.com (Pete) Date: Tue, 13 Dec 2011 11:01:39 +0000 Subject: [Freeswitch-users] mod_say_pt sound file list Message-ID: Hi, Could anybody please provide a list of sound files that mod_say_pt requires ? Regards, Pete -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111213/f555a1e8/attachment-0001.html From arnuld at Phonologies.com Tue Dec 13 16:40:12 2011 From: arnuld at Phonologies.com (arnuld at Phonologies.com) Date: Tue, 13 Dec 2011 19:10:12 +0530 Subject: [Freeswitch-users] FreeSWITCH Logger Message-ID: Currently FreeSWITCH logs to files on disk and you can customize the behavior a bit using logfile.conf.xml. I am working on a project where I need to direct FreeSWITCH to write logs to a socket (like a TCP client). I have written a TCP Server listening for logs on a particular socket. I thought I will change the source code of freeswitch/src/switch_log.c and will add TCP client code to it,? then recomplile it. Is there any better way ? From jdiaz at coinfru.com Tue Dec 13 17:25:59 2011 From: jdiaz at coinfru.com (Josue Diaz Cruz) Date: Tue, 13 Dec 2011 15:25:59 +0100 Subject: [Freeswitch-users] How to change Hang UP Cause and send it to the A-Leg Message-ID: <61A844C3FA734C0D8A1008B0E16B48E4@CROSSTELCOM> I am trying to send a different hangup cause to the caller party. In the Dial plan i am using this public xml file (the asterisks "*" in the code are digits ommited): when a call is made, the dialplan shows me that all is ok This is the Freeswitch server A who play the normal circuit congestion dialplan: 2011-12-13 12:59:37.390188 [NOTICE] switch_channel.c:907 New Channel sofia/external/***********@109.***.28.*** [5a591069-4d76-45c9-93ea-f38e85d17aa2] 2011-12-13 12:59:37.390188 [INFO] mod_dialplan_xml.c:336 Processing *******<********>->5*******#21********** in context public 2011-12-13 12:59:37.390188 [NOTICE] switch_channel.c:907 New Channel sofia/external/5*******#21********** [11fc5e4e-045f-4ce9-81ec-5049c248cb34] 2011-12-13 12:59:37.414191 [NOTICE] sofia.c:5872 Hangup sofia/external/5*******#21********** [CS_CONSUME_MEDIA] [CALL_REJECTED] 2011-12-13 12:59:37.426196 [INFO] mod_dptools.c:2696 Originate Failed. Cause: CALL_REJECTED 2011-12-13 12:59:37.426196 [NOTICE] mod_dptools.c:916 Hangup sofia/external/***********@109.***.28.*** [CS_EXECUTE] [NORMAL_CIRCUIT_CONGESTION] 2011-12-13 12:59:37.426196 [NOTICE] switch_core_session.c:1353 Session 304414 (sofia/external/***********@109.***.28.***) Ended 2011-12-13 12:59:37.426196 [NOTICE] switch_core_session.c:1355 Close Channel sofia/external/***********@109.***.28.*** [CS_DESTROY] 2011-12-13 12:59:37.426196 [NOTICE] switch_core_session.c:1353 Session 304415 (sofia/external/5*******#21**********) Ended 2011-12-13 12:59:37.426196 [NOTICE] switch_core_session.c:1355 Close Channel sofia/external/5*******#21********** [CS_DESTROY] And this is the other freeswitch i am using for testing pourposes to check if the cause code is the right one: 2011-12-13 16:05:13.556630 [NOTICE] switch_channel.c:779 New Channel sofia/external/***********@109.***.28.*** [8a188730-f366-4a49-964e-e4e66fae973c] 2011-12-13 16:05:13.560645 [INFO] mod_dialplan_xml.c:331 Processing 1000 <1000>->500321261452652 in context public 2011-12-13 16:05:13.560645 [NOTICE] switch_channel.c:779 New Channel sofia/external/5*******#21********** [e03f896f-f935-42f9-b83f-49a30de216e7] 2011-12-13 16:05:13.600636 [NOTICE] sofia.c:5051 Hangup sofia/external/5*******#21********** [CS_CONSUME_MEDIA] [NORMAL_TEMPORARY_FAILURE] 2011-12-13 16:05:13.604647 [INFO] mod_dptools.c:2393 Originate Failed. Cause: NORMAL_TEMPORARY_FAILURE 2011-12-13 16:05:13.604647 [NOTICE] mod_dptools.c:2456 Hangup sofia/external/***********@109.***.28.*** [CS_EXECUTE] [NORMAL_TEMPORARY_FAILURE] 2011-12-13 16:05:13.604647 [NOTICE] switch_core_session.c:1228 Session 108345 (sofia/external/***********@109.***.28.***) Ended 2011-12-13 16:05:13.604647 [NOTICE] switch_core_session.c:1230 Close Channel sofia/external/***********@109.***.28.*** [CS_DESTROY] 2011-12-13 16:05:13.604647 [NOTICE] switch_core_session.c:1228 Session 108346 (sofia/external/5*******#21**********) Ended 2011-12-13 16:05:13.604647 [NOTICE] switch_core_session.c:1230 Close Channel sofia/external/5*******#21********** [CS_DESTROY] For me to receive a 34 or a 41 it is not a problem, but for my client they can manage 34 but not 41 error to redirect traffic to other provider. Can someone help me with this issue? I beg you pardon for my english Josue Diaz Cruz Departamento Tecnico y Soporte jdiaz at coinfru.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111213/5a1fb951/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/20111213/5a1fb951/attachment-0001.jpe From jdiaz at coinfru.com Tue Dec 13 18:53:32 2011 From: jdiaz at coinfru.com (Josue Diaz Cruz) Date: Tue, 13 Dec 2011 16:53:32 +0100 Subject: [Freeswitch-users] Freeswitch on system startup on Fedora 16 In-Reply-To: References: <4EE600A1.2020205@puzzled.xs4all.nl> Message-ID: <0D2286349AA8446B80E407420A88E65D@CROSSTELCOM> Try to do this: 1) First of all check this file: nano /etc/init.d/freeswitch 2) if the file does not apears, do this from the freeswitch source folder ( /usr/src/freeswitch/): cd build cp freeswitch.init.redhat /etc/init.d/freeswitch 3) if the file apears check if these sentences are written like these: PID_FILE=${PID_FILE-/opt/freeswitch/run/freeswitch.pid} FS_FILE=${FS_FILE-/opt/freeswitch/bin/freeswitch} FS_HOME=${FS_HOME-/opt/freeswitch} FREESWITCH_ARGS="-nc" 4) Change it to these: PID_FILE=${PID_FILE-/usr/local/freeswitch/run/freeswitch.pid} FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch} FS_HOME=${FS_HOME-/usr/local/freeswitch} FREESWITCH_ARGS="-nc" 5) save and then cd /etc/init.d/ chmod +x freeswitch chkconfig --add freeswitch adduser freeswitch 6) normally works Josue Diaz Cruz Departamento Tecnico y Soporte jdiaz at coinfru.com T 650 694 338 Cl Balsicas 3 Alquerias | 30580 | Murcia www.coinfru.com -----Mensaje original----- De: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] En nombre de curriegrad2004 Enviado el: Tuesday, December 13, 2011 16:29 Para: FreeSWITCH Users Help Asunto: Re: [Freeswitch-users] Freeswitch on system startup on Fedora 16 Have you tried to execute FreeSWITCH as normal under Fedora 16? On Mon, Dec 12, 2011 at 11:49 AM, T T wrote: > Hi, > > SELinux isn't the cause. Setting it to 'permissive' doesn't make it > work after reboot. > > T. > > On Mon, Dec 12, 2011 at 3:24 PM, Patrick Lists > wrote: >> On 11-12-11 11:54, T T wrote: >>> Hi, >>> >>> I've installed FreeSWITCH on Fedora 16. I want FreeSWITCH to be >>> launched ?at system startup. >>> >>> I've put a bash script from >>> http://wiki.freeswitch.org/wiki/Freeswitch_init#Fedora >>> >>> into /etc/init.d directory with corrected paths, and run #chckconfig >>> freeswitch on >>> >>> FreeSWITCH doesnt start at system startup. >> >> If you have SELinux enabled, did you make sure that the init script >> has the proper SELinux context? >> >> Regards, >> Patrick >> >> _____________________________________________________________________ >> ____ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> ers >> http://www.freeswitch.org > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users 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 From avi at avimarcus.net Tue Dec 13 19:10:25 2011 From: avi at avimarcus.net (Avi Marcus) Date: Tue, 13 Dec 2011 18:10:25 +0200 Subject: [Freeswitch-users] FreeSWITCH Logger In-Reply-To: References: Message-ID: ESL? subscribe to events, or use zeromq or some other way of getting them, then log them however you want? -Avi On Tue, Dec 13, 2011 at 3:40 PM, wrote: > > > Currently FreeSWITCH logs to files on disk and you can customize the > behavior a bit using logfile.conf.xml. > > I am working on a > project where I need to direct FreeSWITCH to write logs to a socket (like > a TCP client). I have written a TCP Server listening for logs on a > particular socket. I thought I will change the source code of > freeswitch/src/switch_log.c and will add TCP client code to it, then > recomplile it. > > Is there any better way ? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111213/9b0a6f06/attachment.html From fdelawarde at wirelessmundi.com Tue Dec 13 19:17:03 2011 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Tue, 13 Dec 2011 17:17:03 +0100 Subject: [Freeswitch-users] FreeSWITCH Logger In-Reply-To: References: Message-ID: <1323793023.15906.131.camel@luna.madrid.commsmundi.com> Can't syslog do that? See: http://wiki.freeswitch.org/wiki/Mod_syslog Fran?ois. On Tue, 2011-12-13 at 19:10 +0530, arnuld at Phonologies.com wrote: > > Currently FreeSWITCH logs to files on disk and you can customize the > behavior a bit using logfile.conf.xml. > > I am working on a > project where I need to direct FreeSWITCH to write logs to a socket (like > a TCP client). I have written a TCP Server listening for logs on a > particular socket. I thought I will change the source code of > freeswitch/src/switch_log.c and will add TCP client code to it, then > recomplile it. > > Is there any better way ? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From stuart.mills3 at btopenworld.com Tue Dec 13 19:31:29 2011 From: stuart.mills3 at btopenworld.com (Stuart Mills) Date: Tue, 13 Dec 2011 16:31:29 -0000 Subject: [Freeswitch-users] FreeTDM & libisdn question Message-ID: <67F1171E7AB3444193F564B86EC7C988@hpelite> Hi, I'm new to freeswitch as I'm trying to migrate a system from Asterisk. The Asterisk system has a digium primary rate card in it, so I am trying to get this working, everything else is great it is literally just the card. The model of this card is a 2nd generation Wildcard TE410P and supposedly freetdm does support this when compiled with libisdn, the problem I am having relates to libisdn not compiling properly when I run "make && make install" Has anyone experienced this issue before, or even perhaps tried to do this with success? Regards, Stuart -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111213/81cfcc31/attachment.html From nestor at tiendalinux.com Tue Dec 13 23:17:34 2011 From: nestor at tiendalinux.com (Nestor A Diaz) Date: Tue, 13 Dec 2011 15:17:34 -0500 Subject: [Freeswitch-users] Freeswitch core dump on ftmod_wanpipe Message-ID: <4EE7B2DE.6080906@tiendalinux.com> Hi people, i am experiencing core dumps using freeswitch, they are related to ftmod_wanpipe: The program stop with a core dump atfer: 2011-12-07 00:09:39.908528 [WARNING] ftdm_io.c:4054 [s1c31][1:16] raw I/O read filed 2011-12-07 00:09:40.408534 [WARNING] ftmod_wanpipe.c:1032 [s1c31][1:16] Failed to read 1000 bytes from sangoma device: No buffer space available (-65) 2011-12-07 00:09:40.408534 [WARNING] ftdm_io.c:4054 [s1c31][1:16] raw I/O read filed [...] 2011-12-09 11:57:48.188516 [WARNING] ftmod_wanpipe.c:1032 [s1c31][1:16] Failed to read 1000 bytes from sangoma device: No buffer space available (-65) 2011-12-09 11:57:48.188516 [WARNING] ftdm_io.c:4054 [s1c31][1:16] raw I/O read filed [...] 2011-12-09 22:38:43.508587 [WARNING] ftmod_wanpipe.c:1032 [s1c31][1:16] Failed to read 1000 bytes from sangoma device: No buffer space available (-65) 2011-12-09 22:38:43.508587 [WARNING] ftdm_io.c:4054 [s1c31][1:16] raw I/O read filed 2011-12-09 22:38:46.368554 [DEBUG] ftmod_sangoma_isdn_stack_rcv.c:1000 sng_isdn->s1: Resetting L1 link 2011-12-09 22:38:46.368554 [DEBUG] ftmod_wanpipe.c:900 [s1c31][1:16] First packet write stats: Tx queue len: 0, Tx queue size: 200, Tx idle: 0 2011-12-09 22:38:46.448517 [DEBUG] ftmod_wanpipe.c:964 [s1c31][1:16] First packet read stats: Rx queue len: 0, Rx queue size: 200 and has already happened three times I have the core file, there is anything useful i can do with that ? Slds. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-485-3020 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:211 at tiendalinux.com Email/MSN: nestor at tiendalinux.com http://www.tiendalinux.com/ Bogota, Colombia From gchen00 at insightbb.com Tue Dec 13 23:25:16 2011 From: gchen00 at insightbb.com (Gary Chen) Date: Tue, 13 Dec 2011 15:25:16 -0500 (EST) Subject: [Freeswitch-users] Nee help with error on using ODBC In-Reply-To: <1453814486.13811.1323807838552.JavaMail.root@md09.insight.synacor.com> Message-ID: <1311971789.13834.1323807916315.JavaMail.root@md09.insight.synacor.com> Just download the nightly snapshot and installed. When I tried to start it with ODBC, it has following error: root at lyvt100-2:/usr/local/freeswitch/bin# ./freeswitch 2011-12-13 15:16:30.167564 [INFO] switch_event.c:637 Activate Eventing Engine. 2011-12-13 15:16:30.183610 [DEBUG] switch_event.c:616 Create event dispatch thread 0 2011-12-13 15:16:31.737119 [INFO] switch_nat.c:419 Scanning for NAT 2011-12-13 15:16:31.738025 [DEBUG] switch_nat.c:169 Checking for PMP 1/5 2011-12-13 15:16:31.738330 [ERR] switch_nat.c:200 Error checking for PMP [general error] 2011-12-13 15:16:31.738414 [DEBUG] switch_nat.c:424 Checking for UPnP 2011-12-13 15:16:43.926445 [DEBUG] switch_nat.c:117 No InternetGatewayDevice, using first entry as default (http://204.126.120.212:80/bmlinks/ddf.xml). 2011-12-13 15:16:44.039464 [INFO] switch_nat.c:440 No PMP or UPnP NAT devices detected! 2011-12-13 15:16:44.082799 [INFO] switch_core_sqldb.c:1762 Opening DB 2011-12-13 15:16:44.109064 [ERR] switch_odbc.c:489 ERR: [begin;delete from channels where hostname='';delete from channels where hostname='';commit;] [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 Driver][mysqld-5.0.77]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'delete from channels where hostname='';delete from channels where hostname='';co' at line 1 ] 2011-12-13 15:16:44.109426 [ERR] switch_core_sqldb.c:474 SQL ERR [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 Driver][mysqld-5.0.77]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'delete from channels where hostname='';delete from channels where hostname='';co' at line 1 ] begin;delete from channels where hostname='';delete from channels where hostname='';commit; 2011-12-13 15:16:44.109539 [ERR] switch_core_sqldb.c:1827 Transactions not supported on your DB, disabling ODBC 2011-12-13 15:16:44.119269 [INFO] switch_core_sqldb.c:1762 Opening DB 2011-12-13 15:16:44.272449 [DEBUG] switch_scheduler.c:214 Added task 1 heartbeat (core) to run at 1323807404 2011-12-13 15:16:44.272795 [DEBUG] switch_scheduler.c:214 Added task 2 check_ip (core) to run at 1323807404 2011-12-13 15:16:44.273273 [CONSOLE] switch_core.c:1817 Bringing up environment. 2011-12-13 15:16:44.273372 [CONSOLE] switch_core.c:1818 Loading Modules. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111213/0d4c4657/attachment-0001.html From curriegrad2004 at gmail.com Wed Dec 14 00:23:35 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 13 Dec 2011 13:23:35 -0800 Subject: [Freeswitch-users] Nee help with error on using ODBC In-Reply-To: <1311971789.13834.1323807916315.JavaMail.root@md09.insight.synacor.com> References: <1453814486.13811.1323807838552.JavaMail.root@md09.insight.synacor.com> <1311971789.13834.1323807916315.JavaMail.root@md09.insight.synacor.com> Message-ID: It's testing out your SQL server's capabilities On 2011-12-13 12:26 PM, "Gary Chen" wrote: > Just download the nightly snapshot and installed. > When I tried to start it with ODBC, it has following error: > > root at lyvt100-2:/usr/local/freeswitch/bin# ./freeswitch > 2011-12-13 15:16:30.167564 [INFO] switch_event.c:637 Activate Eventing > Engine. > 2011-12-13 15:16:30.183610 [DEBUG] switch_event.c:616 Create event > dispatch thread 0 > 2011-12-13 15:16:31.737119 [INFO] switch_nat.c:419 Scanning for NAT > 2011-12-13 15:16:31.738025 [DEBUG] switch_nat.c:169 Checking for PMP 1/5 > 2011-12-13 15:16:31.738330 [ERR] switch_nat.c:200 Error checking for PMP > [general error] > 2011-12-13 15:16:31.738414 [DEBUG] switch_nat.c:424 Checking for UPnP > 2011-12-13 15:16:43.926445 [DEBUG] switch_nat.c:117 No > InternetGatewayDevice, using first entry as default ( > http://204.126.120.212:80/bmlinks/ddf.xml). > 2011-12-13 15:16:44.039464 [INFO] switch_nat.c:440 No PMP or UPnP NAT > devices detected! > 2011-12-13 15:16:44.082799 [INFO] switch_core_sqldb.c:1762 Opening DB > 2011-12-13 15:16:44.109064 [ERR] switch_odbc.c:489 ERR: [begin;delete from > channels where hostname='';delete from channels where hostname='';commit;] > [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 > Driver][mysqld-5.0.77]You have an error in your SQL syntax; check the > manual that corresponds to your MySQL server version for the right syntax > to use near 'delete from channels where hostname='';delete from channels > where hostname='';co' at line 1 > ] > 2011-12-13 15:16:44.109426 [ERR] switch_core_sqldb.c:474 SQL ERR [STATE: > 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 Driver][mysqld-5.0.77]You > have an error in your SQL syntax; check the manual that corresponds to your > MySQL server version for the right syntax to use near 'delete from channels > where hostname='';delete from channels where hostname='';co' at line 1 > ] > begin;delete from channels where hostname='';delete from channels where > hostname='';commit; > 2011-12-13 15:16:44.109539 [ERR] switch_core_sqldb.c:1827 Transactions not > supported on your DB, disabling ODBC > 2011-12-13 15:16:44.119269 [INFO] switch_core_sqldb.c:1762 Opening DB > 2011-12-13 15:16:44.272449 [DEBUG] switch_scheduler.c:214 Added task 1 > heartbeat (core) to run at 1323807404 > 2011-12-13 15:16:44.272795 [DEBUG] switch_scheduler.c:214 Added task 2 > check_ip (core) to run at 1323807404 > 2011-12-13 15:16:44.273273 [CONSOLE] switch_core.c:1817 Bringing up > environment. > 2011-12-13 15:16:44.273372 [CONSOLE] switch_core.c:1818 Loading Modules. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111213/89466cb5/attachment.html From gchen00 at insightbb.com Wed Dec 14 00:34:02 2011 From: gchen00 at insightbb.com (Gary Chen) Date: Tue, 13 Dec 2011 16:34:02 -0500 (EST) Subject: [Freeswitch-users] Need help with error on using ODBC In-Reply-To: <1755285317.15528.1323811864675.JavaMail.root@md09.insight.synacor.com> Message-ID: <885201969.15693.1323812042339.JavaMail.root@md09.insight.synacor.com> I added following line odbc.ini and it fixed the problem: OPTION = 67108864 But I got another error: 2011-12-13 16:22:23.150436 [INFO] switch_core_sqldb.c:1762 Opening DB 2011-12-13 16:22:23.244872 [ERR] switch_odbc.c:489 ERR: [create index complete11 on complete (a1,a2,a3,a4,a5,a6,a7,a8,a9,a10,hostname)] [STATE: HY000 CODE 1071 ERROR: [unixODBC][MySQL][ODBC 5.1 Driver][mysqld-5.0.77]Specified key was too long; max key length is 1000 bytes ] 2011-12-13 16:22:23.245184 [ERR] switch_core_sqldb.c:474 SQL ERR [STATE: HY000 CODE 1071 ERROR: [unixODBC][MySQL][ODBC 5.1 Driver][mysqld-5.0.77]Specified key was too long; max key length is 1000 bytes ] create index complete11 on complete (a1,a2,a3,a4,a5,a6,a7,a8,a9,a10,hostname) 2011-12-13 16:22:23.275285 [DEBUG] switch_scheduler.c:214 Added task 1 heartbeat (core) to run at 1323811343 2011-12-13 16:22:23.275660 [DEBUG] switch_scheduler.c:214 Added task 2 check_ip (core) to run at 1323811343 2011-12-13 16:22:23.276173 [CONSOLE] switch_core.c:1817 Bringing up environment. 2011-12-13 16:22:23.276275 [CONSOLE] switch_core.c:1818 Loading Modules. 2011-12-13 16:22:23.280654 [INFO] switch_time.c:1020 Timezone loaded 530 definitions 2011-12-13 16:22:23.280833 [CONSOLE] switch_time.c:1147 Clock calibration disabled. 2011-12-13 16:22:23.280920 [CONSOLE] switch_loadable_module.c:946 Successfully Loaded [CORE_SOFTTIMER_MODULE] 2011-12-13 16:22:23.281046 [NOTICE] switch_loadable_module.c:232 Adding Timer 'soft' 2011-12-13 16:22:23.281656 [CONSOLE] switch_loadable_module.c:946 Successfully Loaded [CORE_PCM_MODULE] 2 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004 Sent: Tuesday, December 13, 2011 4:24 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Nee help with error on using ODBC It's testing out your SQL server's capabilities On 2011-12-13 12:26 PM, "Gary Chen" < gchen00 at insightbb.com > wrote: Just download the nightly snapshot and installed. When I tried to start it with ODBC, it has following error: root at lyvt100-2:/usr/local/freeswitch/bin# ./freeswitch 2011-12-13 15:16:30.167564 [INFO] switch_event.c:637 Activate Eventing Engine. 2011-12-13 15:16:30.183610 [DEBUG] switch_event.c:616 Create event dispatch thread 0 2011-12-13 15:16:31.737119 [INFO] switch_nat.c:419 Scanning for NAT 2011-12-13 15:16:31.738025 [DEBUG] switch_nat.c:169 Checking for PMP 1/5 2011-12-13 15:16:31.738330 [ERR] switch_nat.c:200 Error checking for PMP [general error] 2011-12-13 15:16:31.738414 [DEBUG] switch_nat.c:424 Checking for UPnP 2011-12-13 15:16:43.926445 [DEBUG] switch_nat.c:117 No InternetGatewayDevice, using first entry as default ( http://204.126.120.212:80/bmlinks/ddf.xml ). 2011-12-13 15:16:44.039464 [INFO] switch_nat.c:440 No PMP or UPnP NAT devices detected! 2011-12-13 15:16:44.082799 [INFO] switch_core_sqldb.c:1762 Opening DB 2011-12-13 15:16:44.109064 [ERR] switch_odbc.c:489 ERR: [begin;delete from channels where hostname='';delete from channels where hostname='';commit;] [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 Driver][mysqld-5.0.77]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'delete from channels where hostname='';delete from channels where hostname='';co' at line 1 ] 2011-12-13 15:16:44.109426 [ERR] switch_core_sqldb.c:474 SQL ERR [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 Driver][mysqld-5.0.77]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'delete from channels where hostname='';delete from channels where hostname='';co' at line 1 ] begin;delete from channels where hostname='';delete from channels where hostname='';commit; 2011-12-13 15:16:44.109539 [ERR] switch_core_sqldb.c:1827 Transactions not supported on your DB, disabling ODBC 2011-12-13 15:16:44.119269 [INFO] switch_core_sqldb.c:1762 Opening DB 2011-12-13 15:16:44.272449 [DEBUG] switch_scheduler.c:214 Added task 1 heartbeat (core) to run at 1323807404 2011-12-13 15:16:44.272795 [DEBUG] switch_scheduler.c:214 Added task 2 check_ip (core) to run at 1323807404 2011-12-13 15:16:44.273273 [CONSOLE] switch_core.c:1817 Bringing up environment. 2011-12-13 15:16:44.273372 [CONSOLE] switch_core.c:1818 Loading Modules. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111213/cdd70bec/attachment-0001.html From anthony.minessale at gmail.com Wed Dec 14 00:39:33 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Tue, 13 Dec 2011 15:39:33 -0600 Subject: [Freeswitch-users] Nee help with error on using ODBC In-Reply-To: <1311971789.13834.1323807916315.JavaMail.root@md09.insight.synacor.com> References: <1453814486.13811.1323807838552.JavaMail.root@md09.insight.synacor.com> <1311971789.13834.1323807916315.JavaMail.root@md09.insight.synacor.com> Message-ID: lookup how to enable transaction or multi-line statements for your db On Tue, Dec 13, 2011 at 2:25 PM, Gary Chen wrote: > Just download the nightly snapshot and installed. > When I tried to start it with ODBC, it has following error: > > root at lyvt100-2:/usr/local/freeswitch/bin# ./freeswitch > 2011-12-13 15:16:30.167564 [INFO] switch_event.c:637 Activate Eventing > Engine. > 2011-12-13 15:16:30.183610 [DEBUG] switch_event.c:616 Create event > dispatch thread 0 > 2011-12-13 15:16:31.737119 [INFO] switch_nat.c:419 Scanning for NAT > 2011-12-13 15:16:31.738025 [DEBUG] switch_nat.c:169 Checking for PMP 1/5 > 2011-12-13 15:16:31.738330 [ERR] switch_nat.c:200 Error checking for PMP > [general error] > 2011-12-13 15:16:31.738414 [DEBUG] switch_nat.c:424 Checking for UPnP > 2011-12-13 15:16:43.926445 [DEBUG] switch_nat.c:117 No > InternetGatewayDevice, using first entry as default ( > http://204.126.120.212:80/bmlinks/ddf.xml). > 2011-12-13 15:16:44.039464 [INFO] switch_nat.c:440 No PMP or UPnP NAT > devices detected! > 2011-12-13 15:16:44.082799 [INFO] switch_core_sqldb.c:1762 Opening DB > 2011-12-13 15:16:44.109064 [ERR] switch_odbc.c:489 ERR: [begin;delete from > channels where hostname='';delete from channels where hostname='';commit;] > [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 > Driver][mysqld-5.0.77]You have an error in your SQL syntax; check the > manual that corresponds to your MySQL server version for the right syntax > to use near 'delete from channels where hostname='';delete from channels > where hostname='';co' at line 1 > ] > 2011-12-13 15:16:44.109426 [ERR] switch_core_sqldb.c:474 SQL ERR [STATE: > 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 Driver][mysqld-5.0.77]You > have an error in your SQL syntax; check the manual that corresponds to your > MySQL server version for the right syntax to use near 'delete from channels > where hostname='';delete from channels where hostname='';co' at line 1 > ] > begin;delete from channels where hostname='';delete from channels where > hostname='';commit; > 2011-12-13 15:16:44.109539 [ERR] switch_core_sqldb.c:1827 Transactions not > supported on your DB, disabling ODBC > 2011-12-13 15:16:44.119269 [INFO] switch_core_sqldb.c:1762 Opening DB > 2011-12-13 15:16:44.272449 [DEBUG] switch_scheduler.c:214 Added task 1 > heartbeat (core) to run at 1323807404 > 2011-12-13 15:16:44.272795 [DEBUG] switch_scheduler.c:214 Added task 2 > check_ip (core) to run at 1323807404 > 2011-12-13 15:16:44.273273 [CONSOLE] switch_core.c:1817 Bringing up > environment. > 2011-12-13 15:16:44.273372 [CONSOLE] switch_core.c:1818 Loading Modules. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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/20111213/e4928fa1/attachment.html From msc at freeswitch.org Wed Dec 14 03:38:31 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 13 Dec 2011 16:38:31 -0800 Subject: [Freeswitch-users] Question Answertime variable? billsec is not available In-Reply-To: <1323497014.35405.YahooMailNeo@web39705.mail.mud.yahoo.com> References: <1323497014.35405.YahooMailNeo@web39705.mail.mud.yahoo.com> Message-ID: Are you trying to access these before the channel hangup is complete? If so then yes, they will always be empty. They are not populated until the call actually ends. Are you doing a hangup hook in Lua or something else? -MC On Fri, Dec 9, 2011 at 10:03 PM, king2kin wrote: > The channel variables for this purpose seems to be > {duration, billsec, answersec, flow_billsec, mduration, billmsec, > answermsec, uduration, billusec, answerusec} > see: source code "switch_channel.c"; > and wiki document on categories: channel variables > see: > http://wiki.freeswitch.com/index.php?title=Category:Variable&from=Variable+billmsec > > > However, when I tried to access these channel variables from Lua script or > dialplan, their values are always "nil" or empty. > > -- dialplan: > { > > ........ > > > > } > > -- Lua script: > 2011-12-10 13:39:55.609375 [ERR] mod_lua.cpp:191 > C:\c4dev\freeswitch\Release\scr > ipts/test1.lua:60: attempt to concatenate global 'ctsec' (a nil value) > stack traceback: > C:\c4dev\freeswitch\Release\scripts/test1.lua:60: in main chunk > > see: > { > session:answer() > ......... > ctsec = session:getVariable('billsec') > ctmsec = session:getVariable('billmsec') > freeswitch.consoleLog("INFO","***** Call-Time: sec=" .. ctsec .. "\n") > freeswitch.consoleLog("INFO","***** Call-Time: msec=" .. ctmsec .. "\n") > session:hangup() > } > > > *From:* curriegrad2004 > *To:* FreeSWITCH Users Help > *Sent:* Wednesday, November 30, 2011 11:15 PM > *Subject:* Re: [Freeswitch-users] Question Answertime variable? > > billseconds or bill_msec would be the one to turn to. > > On Wed, Nov 30, 2011 at 4:22 AM, Thomas Hoellriegel > wrote: > > Hi all, > > Is there a possibility to determine the caller time for a channel at > > the end of a call? > > > > For example: in asterisk exists a variable: > > ${ANSWEREDTIME} > > I like to store the Answertime in a Database for example: > > Dailly minutes to call a Cellphone. > > Can your help plese? > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111213/c0a30ec4/attachment.html From msc at freeswitch.org Wed Dec 14 03:43:04 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 13 Dec 2011 16:43:04 -0800 Subject: [Freeswitch-users] Multicast paging group not getting conference audio? In-Reply-To: <75BADEA2E96D4023ACC6E6B8F6B8A958@DREWPC> References: <75BADEA2E96D4023ACC6E6B8F6B8A958@DREWPC> Message-ID: How about setting conference auto out call to just "3456" or "loopback/3456"? -MC On Thu, Dec 8, 2011 at 7:43 PM, Drew Terenzini < drew_terenzini at wesleycloversolutions.com> wrote: > ** > > ** ** > > Good evening, I?m working with a heavily edited version of > Freeswitch and I?m trying to get a multicast paging group added to a > conference such that any audio in the conference is broadcast to the paging > group as well. I?ve successfully gotten a multicast paging group working > in the dialplan as follows:**** > > ** ** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > **** > > ** ** > > When I use X-Lite registered to FS and call ?**3456**?, I get > the spoken audio out the paging group correctly. Now I?m trying to link > the paging group to a conference. I?ve tried experimenting with adding it > to an existing conference by using ?conference_set_auto_outcall?, but > that?s failing:**** > > ** ** > > **** > > ** ** > > And bridge attempts aren?t working as well:**** > > ** ** > > **** > > ** ** > > I?m new to FS and have been scouring the Wiki and mailing > lists for clues on how to accomplish this. Is this something that is not > possible or are I missing a configuration step? Any clues would be > appreciated, 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/20111213/ae82009d/attachment-0001.html From prashant.lamba at gmail.com Wed Dec 14 08:39:47 2011 From: prashant.lamba at gmail.com (Prashant Lamba) Date: Wed, 14 Dec 2011 11:09:47 +0530 Subject: [Freeswitch-users] FreeSWITCH Logger In-Reply-To: References: Message-ID: > On Tue, Dec 13, 2011 at 9:40 PM, Avi Marcus wrote: > > ESL? subscribe to events, or use zeromq or some other way of getting them, > then log them however you want? Im not sure if ESL will give you the output (as detailed) like the FS Logging does. So even if youre going go subscribe to events, its not really logs, but only events. Just like you see log statements being printed in the console, isnt there a way to divert that to a socket? Prashant Phonologies (India) From peter.olsson at visionutveckling.se Wed Dec 14 09:47:19 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 14 Dec 2011 07:47:19 +0100 Subject: [Freeswitch-users] FreeSWITCH Logger Message-ID: <07AA54D1-D8ED-4AD6-A198-7131981EB2E4@visionutveckling.se> I believe this exactly what ESL does for you... The logging in fs_cli is sent exactly that way. If you really want to create your own implementation, just create a new logger module, similar to mod_syslog. /Peter ----- Reply message ----- Fr?n: "Prashant Lamba" Datum: ons, dec 14, 2011 06:48 Rubrik: [Freeswitch-users] FreeSWITCH Logger Till: "FreeSWITCH Users Help" > On Tue, Dec 13, 2011 at 9:40 PM, Avi Marcus wrote: > > ESL? subscribe to events, or use zeromq or some other way of getting them, > then log them however you want? Im not sure if ESL will give you the output (as detailed) like the FS Logging does. So even if youre going go subscribe to events, its not really logs, but only events. Just like you see log statements being printed in the console, isnt there a way to divert that to a socket? Prashant Phonologies (India) _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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:4ee8366b32761120024943! From beppe.grillo at gmail.com Wed Dec 14 13:10:54 2011 From: beppe.grillo at gmail.com (Beppe Grillo) Date: Wed, 14 Dec 2011 11:10:54 +0100 Subject: [Freeswitch-users] REGISTER using the draft-sip-outbound In-Reply-To: References: Message-ID: Hy Brian, I have put the issue and the patch on Jira : http://jira.freeswitch.org/browse/FS-3762 This is my first patch, please be patient with me and let me know if I've done something wrong Regards, Giuseppe 2011/12/5 Brian West > write a patch and put it on jira.freeswitch.org, I don't think we have > that feature yet. > > /b > > On Dec 5, 2011, at 9:12 AM, Beppe Grillo wrote: > > Can your help please? > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111214/47d53a77/attachment.html From arnuld at phonologies.com Wed Dec 14 13:46:04 2011 From: arnuld at phonologies.com (Arnuld Uttre (Phonologies)) Date: Wed, 14 Dec 2011 16:16:04 +0530 Subject: [Freeswitch-users] FreeSWITCH Logger In-Reply-To: <1323793023.15906.131.camel@luna.madrid.commsmundi.com> References: <1323793023.15906.131.camel@luna.madrid.commsmundi.com> Message-ID: <7ab2b4db0aa92ef783ff32fa43c30a7c.squirrel@webmail1.web.com> > Can't syslog do that? > > See: http://wiki.freeswitch.org/wiki/Mod_syslog Unfortunately, there is nothing to see there. I thought I will write there something useful but even google search is unable to provide any useful information on what exactly FreeSWITCH module syslogs does. -- Arnuld Uttre Systems Software Engineer arnuld at Phonologies.COM http://www.phonologies.com Phonologies (India) Private Limited West Wing, Marri Deep, M. C. H. No. 12-5-4, Lallaguda, Secunderabad 500017, INDIA. Ph:+91-40-2701 8993 / 36 Fax:+91-40-2701 8992 From arnuld at phonologies.com Wed Dec 14 13:50:37 2011 From: arnuld at phonologies.com (Arnuld Uttre (Phonologies)) Date: Wed, 14 Dec 2011 16:20:37 +0530 Subject: [Freeswitch-users] FreeSWITCH Logger In-Reply-To: References: Message-ID: <69951c54bd6dd7193c098d639496c4d6.squirrel@webmail1.web.com> > ESL? subscribe to events, or use zeromq or some other way of getting them, > then log them however you want? I see I can use command "log debug/info/etc." through telnet (connected using ESL). I can get useful logs. Only trouble is I have 10 calls in progress and there is no way of knowing which log belongs to which call: In logfile.conf.xml you can use and it will print uuid of each call as first thing before every text line being send to log file. Is there some kind of similar identification I can use in logs coming through ESL ? -- Arnuld Uttre Systems Software Engineer arnuld at Phonologies.COM http://www.phonologies.com Phonologies (India) Private Limited West Wing, Marri Deep, M. C. H. No. 12-5-4, Lallaguda, Secunderabad 500017, INDIA. Ph:+91-40-2701 8993 / 36 Fax:+91-40-2701 8992 From Prometheus001 at gmx.net Wed Dec 14 14:53:25 2011 From: Prometheus001 at gmx.net (Peter P GMX) Date: Wed, 14 Dec 2011 12:53:25 +0100 Subject: [Freeswitch-users] Detect speech in the background with pocketsphinx? Message-ID: <4EE88E35.8040901@gmx.net> Hello, is there a way to detect speech by pocketsphinx in the background? I want to do the following: I want to playback al longer sound file to a caller. Whenever the caller considers, that this part of the file is good, he can say "good", and when he reaches a part of the soundfile which he considers as bad, he says "bad". This means for me, that I have to playback the soundfile to the caller by the XML dialplan and that I have to catch some events in the background through e.g. event socket. During the playing of the soundfile I also would catch these events multiple times, the caller should be able to say "good" and "bad" many times during the soundfile. So this is not just an IVR application where I react on speech and then do something new. In this case the playing of the soundfile must continue after a word is recognized. Questions? - can we achive this with freeswitch? - how can this in gerneral be done? Thanks in advance Peter From fdelawarde at wirelessmundi.com Wed Dec 14 15:35:25 2011 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Wed, 14 Dec 2011 13:35:25 +0100 Subject: [Freeswitch-users] FreeSWITCH Logger In-Reply-To: <7ab2b4db0aa92ef783ff32fa43c30a7c.squirrel@webmail1.web.com> References: <1323793023.15906.131.camel@luna.madrid.commsmundi.com> <7ab2b4db0aa92ef783ff32fa43c30a7c.squirrel@webmail1.web.com> Message-ID: <1323866125.15906.217.camel@luna.madrid.commsmundi.com> On Wed, 2011-12-14 at 16:16 +0530, Arnuld Uttre (Phonologies) wrote: > > Can't syslog do that? > > > > See: http://wiki.freeswitch.org/wiki/Mod_syslog > > Unfortunately, there is nothing to see there. I thought I will write there > something useful but even google search is unable to provide any useful > information on what exactly FreeSWITCH module syslogs does. Here are links to the <200 lines source code and its config file: http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/loggers/mod_syslog/mod_syslog.c?hb=true http://fisheye.freeswitch.org/browse/freeswitch.git/conf/autoload_configs/syslog.conf.xml?hb=true Then you just setup your syslog for remote logging: http://www.deer-run.com/~hal/sysadmin/SSH-SyslogNG.html Fran?ois. From peter.olsson at visionutveckling.se Wed Dec 14 17:22:05 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 14 Dec 2011 15:22:05 +0100 Subject: [Freeswitch-users] Need help with error on using ODBC In-Reply-To: <885201969.15693.1323812042339.JavaMail.root@md09.insight.synacor.com> References: <1755285317.15528.1323811864675.JavaMail.root@md09.insight.synacor.com> <885201969.15693.1323812042339.JavaMail.root@md09.insight.synacor.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5B278E98E4@cooper> MySQL with InnoDB supports key lengths up to 3072 bytes, when using MySQL version 5.0.17 or later (64-bit versions), so you might give that a try. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Gary Chen Skickat: den 13 december 2011 22:34 Till: Freeswitch-user ?mne: Re: [Freeswitch-users] Need help with error on using ODBC I added following line odbc.ini and it fixed the problem: OPTION = 67108864 But I got another error: 2011-12-13 16:22:23.150436 [INFO] switch_core_sqldb.c:1762 Opening DB 2011-12-13 16:22:23.244872 [ERR] switch_odbc.c:489 ERR: [create index complete11 on complete (a1,a2,a3,a4,a5,a6,a7,a8,a9,a10,hostname)] [STATE: HY000 CODE 1071 ERROR: [unixODBC][MySQL][ODBC 5.1 Driver][mysqld-5.0.77]Specified key was too long; max key length is 1000 bytes ] 2011-12-13 16:22:23.245184 [ERR] switch_core_sqldb.c:474 SQL ERR [STATE: HY000 CODE 1071 ERROR: [unixODBC][MySQL][ODBC 5.1 Driver][mysqld-5.0.77]Specified key was too long; max key length is 1000 bytes ] create index complete11 on complete (a1,a2,a3,a4,a5,a6,a7,a8,a9,a10,hostname) 2011-12-13 16:22:23.275285 [DEBUG] switch_scheduler.c:214 Added task 1 heartbeat (core) to run at 1323811343 2011-12-13 16:22:23.275660 [DEBUG] switch_scheduler.c:214 Added task 2 check_ip (core) to run at 1323811343 2011-12-13 16:22:23.276173 [CONSOLE] switch_core.c:1817 Bringing up environment. 2011-12-13 16:22:23.276275 [CONSOLE] switch_core.c:1818 Loading Modules. 2011-12-13 16:22:23.280654 [INFO] switch_time.c:1020 Timezone loaded 530 definitions 2011-12-13 16:22:23.280833 [CONSOLE] switch_time.c:1147 Clock calibration disabled. 2011-12-13 16:22:23.280920 [CONSOLE] switch_loadable_module.c:946 Successfully Loaded [CORE_SOFTTIMER_MODULE] 2011-12-13 16:22:23.281046 [NOTICE] switch_loadable_module.c:232 Adding Timer 'soft' 2011-12-13 16:22:23.281656 [CONSOLE] switch_loadable_module.c:946 Successfully Loaded [CORE_PCM_MODULE] 2 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of curriegrad2004 Sent: Tuesday, December 13, 2011 4:24 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Nee help with error on using ODBC It's testing out your SQL server's capabilities On 2011-12-13 12:26 PM, "Gary Chen" > wrote: Just download the nightly snapshot and installed. When I tried to start it with ODBC, it has following error: root at lyvt100-2:/usr/local/freeswitch/bin# ./freeswitch 2011-12-13 15:16:30.167564 [INFO] switch_event.c:637 Activate Eventing Engine. 2011-12-13 15:16:30.183610 [DEBUG] switch_event.c:616 Create event dispatch thread 0 2011-12-13 15:16:31.737119 [INFO] switch_nat.c:419 Scanning for NAT 2011-12-13 15:16:31.738025 [DEBUG] switch_nat.c:169 Checking for PMP 1/5 2011-12-13 15:16:31.738330 [ERR] switch_nat.c:200 Error checking for PMP [general error] 2011-12-13 15:16:31.738414 [DEBUG] switch_nat.c:424 Checking for UPnP 2011-12-13 15:16:43.926445 [DEBUG] switch_nat.c:117 No InternetGatewayDevice, using first entry as default (http://204.126.120.212:80/bmlinks/ddf.xml). 2011-12-13 15:16:44.039464 [INFO] switch_nat.c:440 No PMP or UPnP NAT devices detected! 2011-12-13 15:16:44.082799 [INFO] switch_core_sqldb.c:1762 Opening DB 2011-12-13 15:16:44.109064 [ERR] switch_odbc.c:489 ERR: [begin;delete from channels where hostname='';delete from channels where hostname='';commit;] [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 Driver][mysqld-5.0.77]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'delete from channels where hostname='';delete from channels where hostname='';co' at line 1 ] 2011-12-13 15:16:44.109426 [ERR] switch_core_sqldb.c:474 SQL ERR [STATE: 42000 CODE 1064 ERROR: [unixODBC][MySQL][ODBC 5.1 Driver][mysqld-5.0.77]You have an error in your SQL syntax; check the manual that corresponds to your MySQL server version for the right syntax to use near 'delete from channels where hostname='';delete from channels where hostname='';co' at line 1 ] begin;delete from channels where hostname='';delete from channels where hostname='';commit; 2011-12-13 15:16:44.109539 [ERR] switch_core_sqldb.c:1827 Transactions not supported on your DB, disabling ODBC 2011-12-13 15:16:44.119269 [INFO] switch_core_sqldb.c:1762 Opening DB 2011-12-13 15:16:44.272449 [DEBUG] switch_scheduler.c:214 Added task 1 heartbeat (core) to run at 1323807404 2011-12-13 15:16:44.272795 [DEBUG] switch_scheduler.c:214 Added task 2 check_ip (core) to run at 1323807404 2011-12-13 15:16:44.273273 [CONSOLE] switch_core.c:1817 Bringing up environment. 2011-12-13 15:16:44.273372 [CONSOLE] switch_core.c:1818 Loading Modules. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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:4ee7c40632764810216920! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111214/1228529d/attachment-0001.html From huw.selley at netdev.co.uk Wed Dec 14 17:26:00 2011 From: huw.selley at netdev.co.uk (Huw Selley) Date: Wed, 14 Dec 2011 14:26:00 +0000 Subject: [Freeswitch-users] Max sessions for a javascript app? In-Reply-To: <1323179264324-7066833.post@n2.nabble.com> References: <1323179264324-7066833.post@n2.nabble.com> Message-ID: Hi Peely, On 6 Dec 2011, at 13:47, peely wrote: > Have you checked you have enough RTP ports available in vars.conf.xml? That was one of the first things I checked :) The fact that I can reach that call volume when using the bridge dialplan app is what made me point at the javascript. I have now replaced the javascript with a pure xml dialplan implementation so this isn't really an issue for me anymore Thanks :) From curriegrad2004 at gmail.com Wed Dec 14 18:09:41 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 14 Dec 2011 07:09:41 -0800 Subject: [Freeswitch-users] FreeSWITCH Logger In-Reply-To: <1323866125.15906.217.camel@luna.madrid.commsmundi.com> References: <1323793023.15906.131.camel@luna.madrid.commsmundi.com> <7ab2b4db0aa92ef783ff32fa43c30a7c.squirrel@webmail1.web.com> <1323866125.15906.217.camel@luna.madrid.commsmundi.com> Message-ID: Unfortunately not everyone can read C :p On Wed, Dec 14, 2011 at 4:35 AM, Fran?ois Delawarde wrote: > On Wed, 2011-12-14 at 16:16 +0530, Arnuld Uttre (Phonologies) wrote: >> > Can't syslog do that? >> > >> > See: http://wiki.freeswitch.org/wiki/Mod_syslog >> >> Unfortunately, there is nothing to see there. I thought I will write there >> something useful but even google search is unable to provide any useful >> information ?on what exactly FreeSWITCH module syslogs does. > > > Here are links to the <200 lines source code and its config file: > > http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/loggers/mod_syslog/mod_syslog.c?hb=true > > http://fisheye.freeswitch.org/browse/freeswitch.git/conf/autoload_configs/syslog.conf.xml?hb=true > > > Then you just setup your syslog for remote logging: > http://www.deer-run.com/~hal/sysadmin/SSH-SyslogNG.html > > > Fran?ois. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From fdelawarde at wirelessmundi.com Wed Dec 14 18:34:18 2011 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Wed, 14 Dec 2011 16:34:18 +0100 Subject: [Freeswitch-users] FreeSWITCH Logger In-Reply-To: References: <1323793023.15906.131.camel@luna.madrid.commsmundi.com> <7ab2b4db0aa92ef783ff32fa43c30a7c.squirrel@webmail1.web.com> <1323866125.15906.217.camel@luna.madrid.commsmundi.com> Message-ID: <1323876858.15906.232.camel@luna.madrid.commsmundi.com> If you plan to do mods in freeswitch/src/switch_log.c, write your own FS module or do some ESL code, you probably can! ;-) On Wed, 2011-12-14 at 07:09 -0800, curriegrad2004 wrote: > Unfortunately not everyone can read C :p > > On Wed, Dec 14, 2011 at 4:35 AM, Fran?ois Delawarde > wrote: > > On Wed, 2011-12-14 at 16:16 +0530, Arnuld Uttre (Phonologies) wrote: > >> > Can't syslog do that? > >> > > >> > See: http://wiki.freeswitch.org/wiki/Mod_syslog > >> > >> Unfortunately, there is nothing to see there. I thought I will write there > >> something useful but even google search is unable to provide any useful > >> information on what exactly FreeSWITCH module syslogs does. > > > > > > Here are links to the <200 lines source code and its config file: > > > > http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/loggers/mod_syslog/mod_syslog.c?hb=true > > > > http://fisheye.freeswitch.org/browse/freeswitch.git/conf/autoload_configs/syslog.conf.xml?hb=true > > > > > > Then you just setup your syslog for remote logging: > > http://www.deer-run.com/~hal/sysadmin/SSH-SyslogNG.html > > > > > > Fran?ois. > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From benkokakao at gmail.com Wed Dec 14 18:46:35 2011 From: benkokakao at gmail.com (Christian Benke) Date: Wed, 14 Dec 2011 16:46:35 +0100 Subject: [Freeswitch-users] "302 Moved Temporarily" invokes ${hold_music} instead of ${transfer_ringback} Message-ID: Hi! I've been tinkering with ringback/transfer_ringback and moh lately and i'm a bit confused about the behaviour in certain cases: I've configured ringback, transfer_ringback and hold_music with three different local-streams(ringback and transfer_ringback are set in the bridging extension, moh is set as a sip_profile-parameter). What i expect to get is hearing "ringback" before the call is answered, "transfer_ringback" when i transfer the answered call from b-leg-endpoint and "moh" wenn the call is set "on hold". Unfortunately this only works when the transfer is initiated via the transfer-command in FreeSWITCH(Default features.xml dx). If the answered calls is transfered by the b-leg-endpoint with "302 Moved Temporarily", FreeSWITCH plays "hold_music". Is there a way to change this behaviour? Best regards Christian From benkokakao at gmail.com Wed Dec 14 18:52:14 2011 From: benkokakao at gmail.com (Christian Benke) Date: Wed, 14 Dec 2011 16:52:14 +0100 Subject: [Freeswitch-users] "302 Moved Temporarily" invokes ${hold_music} instead of ${transfer_ringback} In-Reply-To: References: Message-ID: > transfer-command in FreeSWITCH(Default features.xml dx). > If the answered calls is transfered by the b-leg-endpoint with "302 > Moved Temporarily", FreeSWITCH plays "hold_music". Clarification: When the transfer is invoked by FS with the "transfer"-command, the "transfer_ringback"-stream is played. When the transfer is initiated from the endpoint(302 Moved Temporarily), the "hold_music"-stream is played. From msc at freeswitch.org Wed Dec 14 19:30:41 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 14 Dec 2011 08:30:41 -0800 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today Message-ID: Hello all, Today's conf call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_12_14 We are having Areski from the Newfies Dialer project come and talk to us! Please join us at 1PM EST, 10AM PST. Thanks! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111214/ae82db60/attachment.html From anthony.minessale at gmail.com Wed Dec 14 19:54:34 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 14 Dec 2011 10:54:34 -0600 Subject: [Freeswitch-users] "302 Moved Temporarily" invokes ${hold_music} instead of ${transfer_ringback} In-Reply-To: References: Message-ID: ringback is only generated by the originate process when an a leg is supplied. This means when you run the bridge app. ringback is used on unanswered calls and transfer_ringback is used on answered calls unless its blank then ringback is used. originate, by default terminates when media is reached (aka 183) if {ignore_early_media=true} is specified in the dial string, then it will remain in the originate state until 200ok is received. if ringback var is set, as soon as the b leg sends in ringing indication 180/183 the data from that variable will be played to the inbound leg or A leg. The hold_music is used by the system for any other time something wants to send music while busy. On Wed, Dec 14, 2011 at 9:52 AM, Christian Benke wrote: > > transfer-command in FreeSWITCH(Default features.xml dx). > > If the answered calls is transfered by the b-leg-endpoint with "302 > > Moved Temporarily", FreeSWITCH plays "hold_music". > > Clarification: When the transfer is invoked by FS with the > "transfer"-command, the "transfer_ringback"-stream is played. When the > transfer is initiated from the endpoint(302 Moved Temporarily), the > "hold_music"-stream is played. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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/20111214/b952b10d/attachment.html From henrikaagaardsorensen at gmail.com Wed Dec 14 20:18:10 2011 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Wed, 14 Dec 2011 18:18:10 +0100 Subject: [Freeswitch-users] FreeSWITCH doesn't get SIP registers. Message-ID: I've installed FreeSWITCH on a clean CentOS 6 server. No errors etc. It startup fine, again without any errors. But when I try to connect a SIP client to the sample extension 1000 (or any other) I get a timeout. When I run netstat -tnulp | grep "freeswitch" I get: tcp 0 0 83.221.133.18:5080 0.0.0.0:* LISTEN 1167/freeswitch tcp 0 0 83.221.133.18:5060 0.0.0.0:* LISTEN 1167/freeswitch tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN 1167/freeswitch tcp 0 0 2001:1448:246:9f47:216:5060 :::* LISTEN 1167/freeswitch udp 0 0 83.221.133.18:5080 0.0.0.0:* 1167/freeswitch udp 0 0 83.221.133.18:5060 0.0.0.0:* 1167/freeswitch udp 0 0 2001:1448:246:9f47:5060 :::* 1167/freeswitch In fs_cli, when I run sofia status profile external, I get: Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 83.221.133.18 SIP-IP 83.221.133.18 URL sip:mod_sofia at 83.221.133.18:5080 BIND-URL sip:mod_sofia at 83.221.133.18:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM CODECS OUT PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 When I try to connect and run tshark -f "port 5060" I get: 0.000000 96.224.14.164 -> 83.221.133.18 SIP Request: REGISTER sip:83.221.133.18 3.997431 96.224.14.164 -> 83.221.133.18 SIP Request: REGISTER sip:83.221.133.18 ... I've setup FreeSWITCH on other CentOS 6 distro's and Ubuntu, without any problem. This one I cannot figure out though. Can anyone help? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111214/3240b00b/attachment-0001.html From peter.olsson at visionutveckling.se Wed Dec 14 20:43:14 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 14 Dec 2011 18:43:14 +0100 Subject: [Freeswitch-users] Ang.: FreeSWITCH doesn't get SIP registers. Message-ID: Check the firewall settings on the box, or deactivate the firewall, that's the most probable cause. /Peter ----- Reply message ----- Fr?n: "Henrik Aagaard S?rensen" Datum: ons, dec 14, 2011 18:26 Rubrik: [Freeswitch-users] FreeSWITCH doesn't get SIP registers. Till: "FreeSWITCH Users Help" I've installed FreeSWITCH on a clean CentOS 6 server. No errors etc. It startup fine, again without any errors. But when I try to connect a SIP client to the sample extension 1000 (or any other) I get a timeout. When I run netstat -tnulp | grep "freeswitch" I get: tcp 0 0 83.221.133.18:5080 0.0.0.0:* LISTEN 1167/freeswitch tcp 0 0 83.221.133.18:5060 0.0.0.0:* LISTEN 1167/freeswitch tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN 1167/freeswitch tcp 0 0 2001:1448:246:9f47:216:5060 :::* LISTEN 1167/freeswitch udp 0 0 83.221.133.18:5080 0.0.0.0:* 1167/freeswitch udp 0 0 83.221.133.18:5060 0.0.0.0:* 1167/freeswitch udp 0 0 2001:1448:246:9f47:5060 :::* 1167/freeswitch In fs_cli, when I run sofia status profile external, I get: Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP 83.221.133.18 SIP-IP 83.221.133.18 URL sip:mod_sofia at 83.221.133.18:5080 BIND-URL sip:mod_sofia at 83.221.133.18:5080 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM CODECS OUT PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 0 FAILED-CALLS-IN 0 CALLS-OUT 0 FAILED-CALLS-OUT 0 When I try to connect and run tshark -f "port 5060" I get: 0.000000 96.224.14.164 -> 83.221.133.18 SIP Request: REGISTER sip:83.221.133.18 3.997431 96.224.14.164 -> 83.221.133.18 SIP Request: REGISTER sip:83.221.133.18 ... I've setup FreeSWITCH on other CentOS 6 distro's and Ubuntu, without any problem. This one I cannot figure out though. Can anyone help? !DSPAM:4ee8da2532761595686402! From henrikaagaardsorensen at gmail.com Wed Dec 14 20:51:04 2011 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Wed, 14 Dec 2011 12:51:04 -0500 Subject: [Freeswitch-users] Ang.: FreeSWITCH doesn't get SIP registers. In-Reply-To: References: Message-ID: <-7139615893706902886@unknownmsgid> But I'm able to see incoming SIP on port 5060 with wireshark. And I haven't installed any firewall? On 14/12/2011, at 12.47, Peter Olsson wrote: > Check the firewall settings on the box, or deactivate the firewall, that's the most probable cause. > > /Peter > > ----- Reply message ----- > Fr?n: "Henrik Aagaard S?rensen" > Datum: ons, dec 14, 2011 18:26 > Rubrik: [Freeswitch-users] FreeSWITCH doesn't get SIP registers. > Till: "FreeSWITCH Users Help" > > I've installed FreeSWITCH on a clean CentOS 6 server. No errors etc. > It startup fine, again without any errors. > > But when I try to connect a SIP client to the sample extension 1000 (or any other) I get a timeout. > > When I run netstat -tnulp | grep "freeswitch" I get: > tcp 0 0 83.221.133.18:5080 0.0.0.0:* LISTEN 1167/freeswitch > tcp 0 0 83.221.133.18:5060 0.0.0.0:* LISTEN 1167/freeswitch > tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN 1167/freeswitch > tcp 0 0 2001:1448:246:9f47:216:5060 :::* LISTEN 1167/freeswitch > udp 0 0 83.221.133.18:5080 0.0.0.0:* 1167/freeswitch > udp 0 0 83.221.133.18:5060 0.0.0.0:* 1167/freeswitch > udp 0 0 2001:1448:246:9f47:5060 :::* 1167/freeswitch > > In fs_cli, when I run sofia status profile external, I get: > Name external > Domain Name N/A > Auto-NAT false > DBName sofia_reg_external > Pres Hosts > Dialplan XML > Context public > Challenge Realm auto_to > RTP-IP 83.221.133.18 > SIP-IP 83.221.133.18 > URL sip:mod_sofia at 83.221.133.18:5080 > BIND-URL sip:mod_sofia at 83.221.133.18:5080 > HOLD-MUSIC local_stream://moh > OUTBOUND-PROXY N/A > CODECS IN G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM > CODECS OUT PCMU,PCMA,GSM > TEL-EVENT 101 > DTMF-MODE rfc2833 > CNG 13 > SESSION-TO 0 > MAX-DIALOG 0 > NOMEDIA false > LATE-NEG false > PROXY-MEDIA false > AGGRESSIVENAT false > STUN-ENABLED true > STUN-AUTO-DISABLE false > CALLS-IN 0 > FAILED-CALLS-IN 0 > CALLS-OUT 0 > FAILED-CALLS-OUT 0 > > When I try to connect and run tshark -f "port 5060" I get: > 0.000000 96.224.14.164 -> 83.221.133.18 SIP Request: REGISTER sip:83.221.133.18 > 3.997431 96.224.14.164 -> 83.221.133.18 SIP Request: REGISTER sip:83.221.133.18 > ... > > I've setup FreeSWITCH on other CentOS 6 distro's and Ubuntu, without any problem. This one I cannot figure out though. > > Can anyone help? > > > !DSPAM:4ee8da2532761595686402! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Wed Dec 14 21:02:39 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 14 Dec 2011 10:02:39 -0800 Subject: [Freeswitch-users] FreeSWITCH Logger In-Reply-To: <69951c54bd6dd7193c098d639496c4d6.squirrel@webmail1.web.com> References: <69951c54bd6dd7193c098d639496c4d6.squirrel@webmail1.web.com> Message-ID: On Wed, Dec 14, 2011 at 2:50 AM, Arnuld Uttre (Phonologies) < arnuld at phonologies.com> wrote: > > ESL? subscribe to events, or use zeromq or some other way of getting > them, > > then log them however you want? > > I see I can use command "log debug/info/etc." through telnet (connected > using ESL). I can get useful logs. Only trouble is I have 10 calls in > progress and there is no way of knowing which log belongs to which call: > > In logfile.conf.xml you can use and it > will print uuid of each call as first thing before every text line being > send to log file. Is there some kind of similar identification I can use > in logs coming through ESL ? No, there is not. If you absolutely must have the uuid on the log lines then your best bet is to write your own logger in the vein of mod_syslog. You can find out how Mathieu added uuid logging to mod_logfile.c by searching for "log_uuid" - there's not too much to it. -MC -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111214/011b299d/attachment.html From acrow at integrafin.co.uk Wed Dec 14 21:11:02 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Wed, 14 Dec 2011 18:11:02 +0000 Subject: [Freeswitch-users] Freeswitch core dump on ftmod_wanpipe In-Reply-To: <4EE7B2DE.6080906@tiendalinux.com> References: <4EE7B2DE.6080906@tiendalinux.com> Message-ID: <4EE8E6B6.2090102@integrafin.co.uk> On 13/12/11 20:17, Nestor A Diaz wrote: > Hi people, i am experiencing core dumps using freeswitch, they are > related to ftmod_wanpipe: > > Hi Nestor, There is no harm in getting in contact with Sangoma. They were incredibly helpful with my BRI card issue, and gave me a fixed package within a few days of diagnosing the problem. Absolutely a credit to the telephony hardware industry, and a few of the developers are Freeswitch contributors I think... Best to make sure you are on latest git of FS though, and have followed all the relevant instructions for building wanpipe from their site. The distro Wanpipe is probably no good for FS use, in case you are using that. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From peter.olsson at visionutveckling.se Wed Dec 14 21:11:39 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 14 Dec 2011 19:11:39 +0100 Subject: [Freeswitch-users] FreeSWITCH doesn't get SIP registers. Message-ID: CentOS enables it by default I think. Try 'service firewall stop' if it helps. Wireshark might detect packets even if they are filtered later on. /Peter ----- Reply message ----- Fr?n: "Henrik Aagaard S?rensen" Datum: ons, dec 14, 2011 18:56 Rubrik: [Freeswitch-users] Ang.: FreeSWITCH doesn't get SIP registers. Till: "FreeSWITCH Users Help" But I'm able to see incoming SIP on port 5060 with wireshark. And I haven't installed any firewall? On 14/12/2011, at 12.47, Peter Olsson wrote: > Check the firewall settings on the box, or deactivate the firewall, that's the most probable cause. > > /Peter > > ----- Reply message ----- > Fr?n: "Henrik Aagaard S?rensen" > Datum: ons, dec 14, 2011 18:26 > Rubrik: [Freeswitch-users] FreeSWITCH doesn't get SIP registers. > Till: "FreeSWITCH Users Help" > > I've installed FreeSWITCH on a clean CentOS 6 server. No errors etc. > It startup fine, again without any errors. > > But when I try to connect a SIP client to the sample extension 1000 (or any other) I get a timeout. > > When I run netstat -tnulp | grep "freeswitch" I get: > tcp 0 0 83.221.133.18:5080 0.0.0.0:* LISTEN 1167/freeswitch > tcp 0 0 83.221.133.18:5060 0.0.0.0:* LISTEN 1167/freeswitch > tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN 1167/freeswitch > tcp 0 0 2001:1448:246:9f47:216:5060 :::* LISTEN 1167/freeswitch > udp 0 0 83.221.133.18:5080 0.0.0.0:* 1167/freeswitch > udp 0 0 83.221.133.18:5060 0.0.0.0:* 1167/freeswitch > udp 0 0 2001:1448:246:9f47:5060 :::* 1167/freeswitch > > In fs_cli, when I run sofia status profile external, I get: > Name external > Domain Name N/A > Auto-NAT false > DBName sofia_reg_external > Pres Hosts > Dialplan XML > Context public > Challenge Realm auto_to > RTP-IP 83.221.133.18 > SIP-IP 83.221.133.18 > URL sip:mod_sofia at 83.221.133.18:5080 > BIND-URL sip:mod_sofia at 83.221.133.18:5080 > HOLD-MUSIC local_stream://moh > OUTBOUND-PROXY N/A > CODECS IN G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM > CODECS OUT PCMU,PCMA,GSM > TEL-EVENT 101 > DTMF-MODE rfc2833 > CNG 13 > SESSION-TO 0 > MAX-DIALOG 0 > NOMEDIA false > LATE-NEG false > PROXY-MEDIA false > AGGRESSIVENAT false > STUN-ENABLED true > STUN-AUTO-DISABLE false > CALLS-IN 0 > FAILED-CALLS-IN 0 > CALLS-OUT 0 > FAILED-CALLS-OUT 0 > > When I try to connect and run tshark -f "port 5060" I get: > 0.000000 96.224.14.164 -> 83.221.133.18 SIP Request: REGISTER sip:83.221.133.18 > 3.997431 96.224.14.164 -> 83.221.133.18 SIP Request: REGISTER sip:83.221.133.18 > ... > > I've setup FreeSWITCH on other CentOS 6 distro's and Ubuntu, without any problem. This one I cannot figure out though. > > Can anyone help? > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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:4ee8e13532768713110985! From henrikaagaardsorensen at gmail.com Wed Dec 14 21:18:55 2011 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Wed, 14 Dec 2011 19:18:55 +0100 Subject: [Freeswitch-users] FreeSWITCH doesn't get SIP registers. In-Reply-To: References: Message-ID: I just get this: service firewall stop firewall: unrecognized service On Wed, Dec 14, 2011 at 7:11 PM, Peter Olsson < peter.olsson at visionutveckling.se> wrote: > CentOS enables it by default I think. Try 'service firewall stop' if it > helps. Wireshark might detect packets even if they are filtered later on. > > /Peter > > ----- Reply message ----- > Fr?n: "Henrik Aagaard S?rensen" > Datum: ons, dec 14, 2011 18:56 > Rubrik: [Freeswitch-users] Ang.: FreeSWITCH doesn't get SIP registers. > Till: "FreeSWITCH Users Help" > > But I'm able to see incoming SIP on port 5060 with wireshark. And I > haven't installed any firewall? > > On 14/12/2011, at 12.47, Peter Olsson > wrote: > > > Check the firewall settings on the box, or deactivate the firewall, > that's the most probable cause. > > > > /Peter > > > > ----- Reply message ----- > > Fr?n: "Henrik Aagaard S?rensen" > > Datum: ons, dec 14, 2011 18:26 > > Rubrik: [Freeswitch-users] FreeSWITCH doesn't get SIP registers. > > Till: "FreeSWITCH Users Help" > > > > I've installed FreeSWITCH on a clean CentOS 6 server. No errors etc. > > It startup fine, again without any errors. > > > > But when I try to connect a SIP client to the sample extension 1000 (or > any other) I get a timeout. > > > > When I run netstat -tnulp | grep "freeswitch" I get: > > tcp 0 0 83.221.133.18:5080 > 0.0.0.0:* LISTEN 1167/freeswitch > > tcp 0 0 83.221.133.18:5060 > 0.0.0.0:* LISTEN 1167/freeswitch > > tcp 0 0 127.0.0.1:8021 0.0.0.0:* > LISTEN 1167/freeswitch > > tcp 0 0 2001:1448:246:9f47:216:5060 :::* > LISTEN 1167/freeswitch > > udp 0 0 83.221.133.18:5080 > 0.0.0.0:* 1167/freeswitch > > udp 0 0 83.221.133.18:5060 > 0.0.0.0:* 1167/freeswitch > > udp 0 0 2001:1448:246:9f47:5060 :::* > 1167/freeswitch > > > > In fs_cli, when I run sofia status profile external, I get: > > Name external > > Domain Name N/A > > Auto-NAT false > > DBName sofia_reg_external > > Pres Hosts > > Dialplan XML > > Context public > > Challenge Realm auto_to > > RTP-IP 83.221.133.18 > > SIP-IP 83.221.133.18 > > URL sip:mod_sofia at 83.221.133.18:5080< > http://sip:mod_sofia at 83.221.133.18:5080> > > BIND-URL sip:mod_sofia at 83.221.133.18:5080< > http://sip:mod_sofia at 83.221.133.18:5080> > > HOLD-MUSIC local_stream://moh > > OUTBOUND-PROXY N/A > > CODECS IN G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM > > CODECS OUT PCMU,PCMA,GSM > > TEL-EVENT 101 > > DTMF-MODE rfc2833 > > CNG 13 > > SESSION-TO 0 > > MAX-DIALOG 0 > > NOMEDIA false > > LATE-NEG false > > PROXY-MEDIA false > > AGGRESSIVENAT false > > STUN-ENABLED true > > STUN-AUTO-DISABLE false > > CALLS-IN 0 > > FAILED-CALLS-IN 0 > > CALLS-OUT 0 > > FAILED-CALLS-OUT 0 > > > > When I try to connect and run tshark -f "port 5060" I get: > > 0.000000 96.224.14.164 -> 83.221.133.18 SIP Request: REGISTER sip: > 83.221.133.18 > > 3.997431 96.224.14.164 -> 83.221.133.18 SIP Request: REGISTER sip: > 83.221.133.18 > > ... > > > > I've setup FreeSWITCH on other CentOS 6 distro's and Ubuntu, without any > problem. This one I cannot figure out though. > > > > Can anyone help? > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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:4ee8e13532768713110985! > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111214/ba6f1490/attachment-0001.html From Hector.Geraldino at ip-soft.net Wed Dec 14 21:31:58 2011 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Wed, 14 Dec 2011 13:31:58 -0500 Subject: [Freeswitch-users] FreeSWITCH doesn't get SIP registers. In-Reply-To: References: Message-ID: <6A6B4C284AD15042B429EB9D904544AD0225506DD2@NY1-EXMB-01.ip-soft.net> A few tips: First, is FS running? Can you check if the port is in use by doing a lsof -i ? Can you try to listen to this port, by using nc? (something like nc -v -l ipaddress portnumber). After doing that, try to connect to the machine/port using telnet, so we can discard a network issue. Good luck! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Henrik Aagaard S?rensen Sent: Wednesday, December 14, 2011 1:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH doesn't get SIP registers. I just get this: service firewall stop firewall: unrecognized service On Wed, Dec 14, 2011 at 7:11 PM, Peter Olsson > wrote: CentOS enables it by default I think. Try 'service firewall stop' if it helps. Wireshark might detect packets even if they are filtered later on. /Peter ----- Reply message ----- Fr?n: "Henrik Aagaard S?rensen" > Datum: ons, dec 14, 2011 18:56 Rubrik: [Freeswitch-users] Ang.: FreeSWITCH doesn't get SIP registers. Till: "FreeSWITCH Users Help" > But I'm able to see incoming SIP on port 5060 with wireshark. And I haven't installed any firewall? On 14/12/2011, at 12.47, Peter Olsson > wrote: > Check the firewall settings on the box, or deactivate the firewall, that's the most probable cause. > > /Peter > > ----- Reply message ----- > Fr?n: "Henrik Aagaard S?rensen" > > Datum: ons, dec 14, 2011 18:26 > Rubrik: [Freeswitch-users] FreeSWITCH doesn't get SIP registers. > Till: "FreeSWITCH Users Help" > > > I've installed FreeSWITCH on a clean CentOS 6 server. No errors etc. > It startup fine, again without any errors. > > But when I try to connect a SIP client to the sample extension 1000 (or any other) I get a timeout. > > When I run netstat -tnulp | grep "freeswitch" I get: > tcp 0 0 83.221.133.18:5080 0.0.0.0:* LISTEN 1167/freeswitch > tcp 0 0 83.221.133.18:5060 0.0.0.0:* LISTEN 1167/freeswitch > tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN 1167/freeswitch > tcp 0 0 2001:1448:246:9f47:216:5060 :::* LISTEN 1167/freeswitch > udp 0 0 83.221.133.18:5080 0.0.0.0:* 1167/freeswitch > udp 0 0 83.221.133.18:5060 0.0.0.0:* 1167/freeswitch > udp 0 0 2001:1448:246:9f47:5060 :::* 1167/freeswitch > > In fs_cli, when I run sofia status profile external, I get: > Name external > Domain Name N/A > Auto-NAT false > DBName sofia_reg_external > Pres Hosts > Dialplan XML > Context public > Challenge Realm auto_to > RTP-IP 83.221.133.18 > SIP-IP 83.221.133.18 > URL sip:mod_sofia at 83.221.133.18:5080 > BIND-URL sip:mod_sofia at 83.221.133.18:5080 > HOLD-MUSIC local_stream://moh > OUTBOUND-PROXY N/A > CODECS IN G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM > CODECS OUT PCMU,PCMA,GSM > TEL-EVENT 101 > DTMF-MODE rfc2833 > CNG 13 > SESSION-TO 0 > MAX-DIALOG 0 > NOMEDIA false > LATE-NEG false > PROXY-MEDIA false > AGGRESSIVENAT false > STUN-ENABLED true > STUN-AUTO-DISABLE false > CALLS-IN 0 > FAILED-CALLS-IN 0 > CALLS-OUT 0 > FAILED-CALLS-OUT 0 > > When I try to connect and run tshark -f "port 5060" I get: > 0.000000 96.224.14.164 -> 83.221.133.18 SIP Request: REGISTER sip:83.221.133.18 > 3.997431 96.224.14.164 -> 83.221.133.18 SIP Request: REGISTER sip:83.221.133.18 > ... > > I've setup FreeSWITCH on other CentOS 6 distro's and Ubuntu, without any problem. This one I cannot figure out though. > > Can anyone help? > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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:4ee8e13532768713110985! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111214/3359face/attachment.html From peter.olsson at visionutveckling.se Wed Dec 14 21:34:05 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 14 Dec 2011 19:34:05 +0100 Subject: [Freeswitch-users] FreeSWITCH doesn't get SIP registers. Message-ID: <30720BA7-FEE6-4A2E-AB88-306FBD97A35D@visionutveckling.se> Try service iptables stop. My mistake... /Peter ----- Reply message ----- Fr?n: "Henrik Aagaard S?rensen" Datum: ons, dec 14, 2011 19:28 Rubrik: [Freeswitch-users] FreeSWITCH doesn't get SIP registers. Till: "FreeSWITCH Users Help" I just get this: service firewall stop firewall: unrecognized service On Wed, Dec 14, 2011 at 7:11 PM, Peter Olsson > wrote: CentOS enables it by default I think. Try 'service firewall stop' if it helps. Wireshark might detect packets even if they are filtered later on. /Peter ----- Reply message ----- Fr?n: "Henrik Aagaard S?rensen" > Datum: ons, dec 14, 2011 18:56 Rubrik: [Freeswitch-users] Ang.: FreeSWITCH doesn't get SIP registers. Till: "FreeSWITCH Users Help" > But I'm able to see incoming SIP on port 5060 with wireshark. And I haven't installed any firewall? On 14/12/2011, at 12.47, Peter Olsson > wrote: > Check the firewall settings on the box, or deactivate the firewall, that's the most probable cause. > > /Peter > > ----- Reply message ----- > Fr?n: "Henrik Aagaard S?rensen" > > Datum: ons, dec 14, 2011 18:26 > Rubrik: [Freeswitch-users] FreeSWITCH doesn't get SIP registers. > Till: "FreeSWITCH Users Help" > > > I've installed FreeSWITCH on a clean CentOS 6 server. No errors etc. > It startup fine, again without any errors. > > But when I try to connect a SIP client to the sample extension 1000 (or any other) I get a timeout. > > When I run netstat -tnulp | grep "freeswitch" I get: > tcp 0 0 83.221.133.18:5080 0.0.0.0:* LISTEN 1167/freeswitch > tcp 0 0 83.221.133.18:5060 0.0.0.0:* LISTEN 1167/freeswitch > tcp 0 0 127.0.0.1:8021 0.0.0.0:* LISTEN 1167/freeswitch > tcp 0 0 2001:1448:246:9f47:216:5060 :::* LISTEN 1167/freeswitch > udp 0 0 83.221.133.18:5080 0.0.0.0:* 1167/freeswitch > udp 0 0 83.221.133.18:5060 0.0.0.0:* 1167/freeswitch > udp 0 0 2001:1448:246:9f47:5060 :::* 1167/freeswitch > > In fs_cli, when I run sofia status profile external, I get: > Name external > Domain Name N/A > Auto-NAT false > DBName sofia_reg_external > Pres Hosts > Dialplan XML > Context public > Challenge Realm auto_to > RTP-IP 83.221.133.18 > SIP-IP 83.221.133.18 > URL sip:mod_sofia at 83.221.133.18:5080 > BIND-URL sip:mod_sofia at 83.221.133.18:5080 > HOLD-MUSIC local_stream://moh > OUTBOUND-PROXY N/A > CODECS IN G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM > CODECS OUT PCMU,PCMA,GSM > TEL-EVENT 101 > DTMF-MODE rfc2833 > CNG 13 > SESSION-TO 0 > MAX-DIALOG 0 > NOMEDIA false > LATE-NEG false > PROXY-MEDIA false > AGGRESSIVENAT false > STUN-ENABLED true > STUN-AUTO-DISABLE false > CALLS-IN 0 > FAILED-CALLS-IN 0 > CALLS-OUT 0 > FAILED-CALLS-OUT 0 > > When I try to connect and run tshark -f "port 5060" I get: > 0.000000 96.224.14.164 -> 83.221.133.18 SIP Request: REGISTER sip:83.221.133.18 > 3.997431 96.224.14.164 -> 83.221.133.18 SIP Request: REGISTER sip:83.221.133.18 > ... > > I've setup FreeSWITCH on other CentOS 6 distro's and Ubuntu, without any problem. This one I cannot figure out though. > > Can anyone help? > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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:4ee8e8b132761321042113! From freeswitch-list at puzzled.xs4all.nl Wed Dec 14 21:37:41 2011 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 14 Dec 2011 19:37:41 +0100 Subject: [Freeswitch-users] FreeSWITCH doesn't get SIP registers. In-Reply-To: References: Message-ID: <4EE8ECF5.7070200@puzzled.xs4all.nl> On 14-12-11 19:18, Henrik Aagaard S?rensen wrote: > I just get this: > service firewall stop > firewall: unrecognized service Make that (as root): /sbin/service iptables stop And if you also use IPv6: /sbin/service ip6tables stop Regards, Patrick From benkokakao at gmail.com Wed Dec 14 21:38:02 2011 From: benkokakao at gmail.com (Christian Benke) Date: Wed, 14 Dec 2011 19:38:02 +0100 Subject: [Freeswitch-users] "302 Moved Temporarily" invokes ${hold_music} instead of ${transfer_ringback} In-Reply-To: References: Message-ID: > transfer_ringback is used on > answered calls unless its blank then ringback is used Looks like i've confused my traces and quoted the wrong one - it doesn't happen on "Moved Temporarily" but on "REFER" obviously. Sorry for this misleading information. However, i was actually talking about an already answered call where transfer_ringback is not used on a attended-transfer. But setting a channel-specific hold_music has solved this issue too. Best regards Christian From fsrichard at ghz.fr Wed Dec 14 21:16:27 2011 From: fsrichard at ghz.fr (Richard) Date: Wed, 14 Dec 2011 19:16:27 +0100 Subject: [Freeswitch-users] How to set voicemail default language (leaving message not listening to message)? Message-ID: <4EE8E7FB.9020206@ghz.fr> Hello, I'm trying to get the voice that talks on the voicemail before leaving a message (dial # or stop taking to end your message) to french language but cannot find out where to set this ! I havent been able to find any documentation about where to set application languages when called in a bridge loopback like in the following line : I added the default language line to the extension that contains this line (default.xml : Local_Extension) but it does not affect the loopback. Full extension : ** I have got my French language working when I call the 4000 extension : ** If I change the path of the audio files in conf/lang/en/en.xml everything works in French. However I want to have one extension with English language and another with French langauge so I need to be able to set the default language to fr when bridge/loopbacking to app=voicemail. It's the application="bridge" data="loopback/app= that baffles me ! I presume the loopback passes the call onto the voicemail application but I haven't found where to set the default language for this application when using a bridge and loopback ! Thanks in advance, Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111214/16c95c94/attachment.html From fsrichard at ghz.fr Wed Dec 14 21:50:26 2011 From: fsrichard at ghz.fr (fsrichard at ghz.fr) Date: Wed, 14 Dec 2011 19:50:26 +0100 Subject: [Freeswitch-users] Howto Set default language for an application ? Message-ID: <4EE8EFF2.6090808@ghz.fr> Hello, I'm trying to set up my first FreeSWITCH server but I don't know where to set the default language corresponding to the following line in conf/dialplan/default.xml : > > ... > > ... > I have tried this : > > ... > ** > ... > > ... > From what I have understood it is normal that it doesn't work as the bridge to a loopback would use it's own settings, but I don't have a clue where to set it of app=voicemail Is there a way to set the default language per SIP connection ? I have got an english and a french SIP line and would like to make calls going to the english line have english instructions for leaving a message on the voicemail and the calls to the french line to have french instructions for leaving a message on the voicemail. Thank you Richard -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111214/6eedd249/attachment.html From fredyg1965 at gmail.com Wed Dec 14 22:07:53 2011 From: fredyg1965 at gmail.com (Fredy Gonzales) Date: Wed, 14 Dec 2011 14:07:53 -0500 Subject: [Freeswitch-users] Lazarus - FreeSwitch References: <4EE8EFF2.6090808@ghz.fr> Message-ID: Hello, Can anybody help me and tell me how do install a module FreeSwitch with Lazarus-Pascal.. And there is any other way beside socket and it is have to be with socket what module activated Saludos. Fredy G. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111214/2c34caca/attachment.html From acrow at integrafin.co.uk Wed Dec 14 23:24:24 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Wed, 14 Dec 2011 20:24:24 +0000 Subject: [Freeswitch-users] Lua - origination to local endpoint then bridge to either local or remote destination: no audio Message-ID: <4EE905F8.6010908@integrafin.co.uk> Hi all, I have a Lua script that I am calling from XML RPC from a browser (also same behaviour from the console). The script originates a session to one extension, when that is picked up it originates another session to another endpoint and then bridges the two sessions. The issue is that if the first freeswitch.Session() has the endpoint local to the FS box, there is no audio at all after the bridge is executed. If the first session is a remote endpoint, it works OK apart from the fact that that endpoint does not hear ringing, but when the second session is answered all is well. If both endpoints are remote, everything is perfect. Here is my Lua script - the no-audio version: local calling_user = argv[1]; local called_num = argv[2]; local session1 = freeswitch.Session("user/"..calling_user); session1:sleep(500); session1:answer(); if (session1:ready() == true) then local session2 = freeswitch.Session("[origination_caller_id_number=02023493482]sofia/gateway/10.10.0.2/" .. called_num); session2:answer(); if (session2:ready() == true) then freeswitch.consoleLog("INFO","Bridging\n"); freeswitch.bridge(session1, session2); end end This one works fine: local calling_user = argv[1]; local called_num = argv[2]; local session1 = freeswitch.Session("[origination_caller_id_name="..called_num.."]sofia/gateway/10.10.0.2/" .. calling_user); session1:answer(); if (session1:ready() == true) then local session2 = freeswitch.Session("[origination_caller_id_number=02076084900]sofia/gateway/10.10.0.2/" .. called_num); --either the above or the below both work in this script, except the below does not present ringing to session1 --local session2 = freeswitch.Session("[ringback=%(400,200,400,450);%(400,2200,400,450)]user/" .. called_num); session2:answer(); if (session2:ready() == true) then freeswitch.consoleLog("INFO","Bridging\n"); freeswitch.bridge(session1, session2); end end Where gateway 10.10.0.2 is a Mitel 3300 connected to the PSTN via ISDN30. I can provide logs if required - please advise as this is driving me bonkers! Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) From henrikaagaardsorensen at gmail.com Thu Dec 15 01:04:39 2011 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Wed, 14 Dec 2011 23:04:39 +0100 Subject: [Freeswitch-users] FreeSWITCH doesn't get SIP registers. In-Reply-To: <4EE8ECF5.7070200@puzzled.xs4all.nl> References: <4EE8ECF5.7070200@puzzled.xs4all.nl> Message-ID: Dear Patrick. The iptables service was running and causing my problem. THANK YOU so much for all your help. I will read up on iptables and opening the ports I need. Once again, thank you. On Wed, Dec 14, 2011 at 7:37 PM, Patrick Lists < freeswitch-list at puzzled.xs4all.nl> wrote: > On 14-12-11 19:18, Henrik Aagaard S?rensen wrote: > > I just get this: > > service firewall stop > > firewall: unrecognized service > > Make that (as root): > > /sbin/service iptables stop > > And if you also use IPv6: > > /sbin/service ip6tables stop > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111214/31611bd5/attachment.html From stkn at freeswitch.org Thu Dec 15 01:43:20 2011 From: stkn at freeswitch.org (Stefan Knoblich) Date: Wed, 14 Dec 2011 23:43:20 +0100 Subject: [Freeswitch-users] FreeTDM & libisdn question In-Reply-To: <67F1171E7AB3444193F564B86EC7C988@hpelite> References: <67F1171E7AB3444193F564B86EC7C988@hpelite> Message-ID: <4EE92688.8020901@freeswitch.org> On 12/13/11 17:31, Stuart Mills wrote: > Hi, > I'm new to freeswitch as I'm trying to migrate a system from Asterisk. > The Asterisk system has a digium primary rate card in it, so I am trying to get this working, everything else is great it is literally just the card. > The model of this card is a 2nd generation Wildcard TE410P and supposedly freetdm does support this when compiled with libisdn, the problem I am having relates to libisdn not compiling properly when I > run "make && make install" libisdn is unsupported at the moment, please use digium's libpri and ftmod_libpri (./configure --with-libpri). From gabe at gundy.org Thu Dec 15 01:54:31 2011 From: gabe at gundy.org (Gabriel Gunderson) Date: Wed, 14 Dec 2011 15:54:31 -0700 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: References: Message-ID: On Wed, Dec 14, 2011 at 9:30 AM, Michael Collins wrote: > We are having Areski from the Newfies Dialer project come and talk to us! Sorry I missed this one... I was really hoping I'd be able to join in. Looking forward to the recording :) Gabe From msc at freeswitch.org Thu Dec 15 02:01:40 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 14 Dec 2011 15:01:40 -0800 Subject: [Freeswitch-users] Lua - origination to local endpoint then bridge to either local or remote destination: no audio In-Reply-To: <4EE905F8.6010908@integrafin.co.uk> References: <4EE905F8.6010908@integrafin.co.uk> Message-ID: Why are you using Lua at all? It looks like a simple originate that you should be able to do as an API call. -MC On Wed, Dec 14, 2011 at 12:24 PM, Alex Crow wrote: > Hi all, > > I have a Lua script that I am calling from XML RPC from a browser (also > same behaviour from the console). The script originates a session to one > extension, when that is picked up it originates another session to > another endpoint and then bridges the two sessions. > > The issue is that if the first freeswitch.Session() has the > endpoint local to the FS box, there is no audio at all after the bridge > is executed. If the first session is a remote endpoint, it works OK > apart from the fact that that endpoint does not hear ringing, but when > the second session is answered all is well. If both endpoints are > remote, everything is perfect. > > Here is my Lua script - the no-audio version: > > local calling_user = argv[1]; > local called_num = argv[2]; > > local session1 = freeswitch.Session("user/"..calling_user); > > session1:sleep(500); > > session1:answer(); > > if (session1:ready() == true) then > local session2 = > > freeswitch.Session("[origination_caller_id_number=02023493482]sofia/gateway/ > 10.10.0.2/" > .. called_num); > session2:answer(); > > if (session2:ready() == true) then > freeswitch.consoleLog("INFO","Bridging\n"); > freeswitch.bridge(session1, session2); > end > > end > > This one works fine: > > local calling_user = argv[1]; > local called_num = argv[2]; > > > local session1 = > > freeswitch.Session("[origination_caller_id_name="..called_num.."]sofia/gateway/ > 10.10.0.2/" > .. calling_user); > > session1:answer(); > > if (session1:ready() == true) then > local session2 = > > freeswitch.Session("[origination_caller_id_number=02076084900]sofia/gateway/ > 10.10.0.2/" > .. called_num); > --either the above or the below both work in this script, > except the below does not present ringing to session1 > --local session2 = > freeswitch.Session("[ringback=%(400,200,400,450);%(400,2200,400,450)]user/" > .. called_num); > session2:answer(); > > if (session2:ready() == true) then > freeswitch.consoleLog("INFO","Bridging\n"); > freeswitch.bridge(session1, session2); > end > > end > > Where gateway 10.10.0.2 is a Mitel 3300 connected to the PSTN via ISDN30. > > I can provide logs if required - please advise as this is driving me > bonkers! > > Alex > > -- > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > > "Transact" is operated by Integrated Financial Arrangements plc > Domain House, 5-7 Singer Street, London EC2A 4BQ > Tel: (020) 7608 4900 Fax: (020) 7608 5300 > (Registered office: as above; Registered in England and Wales under > number: 3727592) > Authorised and regulated by the Financial Services Authority (entered on > the FSA Register; number: 190856) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111214/bdfaea5a/attachment.html From msc at freeswitch.org Thu Dec 15 02:02:39 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 14 Dec 2011 15:02:39 -0800 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: References: Message-ID: Knock yerself out! -MC http://torrents.freeswitch.org/conf_call_2011-12-14.torrent On Wed, Dec 14, 2011 at 2:54 PM, Gabriel Gunderson wrote: > On Wed, Dec 14, 2011 at 9:30 AM, Michael Collins > wrote: > > We are having Areski from the Newfies Dialer project come and talk to us! > > Sorry I missed this one... I was really hoping I'd be able to join in. > Looking forward to the recording :) > > 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111214/5c2bc799/attachment.html From henrikaagaardsorensen at gmail.com Thu Dec 15 02:50:15 2011 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Thu, 15 Dec 2011 00:50:15 +0100 Subject: [Freeswitch-users] Reinstall FreeSWITCH after new kernel? Message-ID: I've read a lot about the kernel timer and how it should be on 1000hz. My clean install of Ubuntu has 100hz. So I guess I have to recompile the kernel to 1000hz? If so, does I have to reinstall FreeSWITCH after the new kernel compile? And does anyone knows of any tutorial about recompiling a Ubuntu 10.04 LTS kernel to 1000hz? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111215/b5d138b5/attachment.html From stuart.mills3 at btopenworld.com Thu Dec 15 04:11:41 2011 From: stuart.mills3 at btopenworld.com (Stuart Mills) Date: Thu, 15 Dec 2011 01:11:41 -0000 Subject: [Freeswitch-users] FreeTDM & libisdn question References: <67F1171E7AB3444193F564B86EC7C988@hpelite> <4EE92688.8020901@freeswitch.org> Message-ID: <2B272E37A1D44FD19BFF1E2FF0AB4928@hpelite> Thanks for the advise Stefan, it's working using the config you suggested, had a little fun with the freetdm config as there's not many examples using this card, apart from that its all good. ----- Original Message ----- From: "Stefan Knoblich" To: "FreeSWITCH Users Help" Cc: "Stuart Mills" Sent: Wednesday, December 14, 2011 10:43 PM Subject: Re: [Freeswitch-users] FreeTDM & libisdn question > On 12/13/11 17:31, Stuart Mills wrote: >> Hi, >> I'm new to freeswitch as I'm trying to migrate a system from Asterisk. >> The Asterisk system has a digium primary rate card in it, so I am trying >> to get this working, everything else is great it is literally just the >> card. >> The model of this card is a 2nd generation Wildcard TE410P and supposedly >> freetdm does support this when compiled with libisdn, the problem I am >> having relates to libisdn not compiling properly when I >> run "make && make install" > > libisdn is unsupported at the moment, please use digium's libpri and > ftmod_libpri (./configure --with-libpri). > > From william.suffill at gmail.com Thu Dec 15 08:46:44 2011 From: william.suffill at gmail.com (William Suffill) Date: Thu, 15 Dec 2011 00:46:44 -0500 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: References: Message-ID: No more direct download links for the recordings? -- W From acrow at integrafin.co.uk Thu Dec 15 10:37:41 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Thu, 15 Dec 2011 07:37:41 +0000 Subject: [Freeswitch-users] {Disarmed} Re: Lua - origination to local endpoint then bridge to either local or remote destination: no audio In-Reply-To: References: <4EE905F8.6010908@integrafin.co.uk> Message-ID: <4EE9A3C5.3030407@integrafin.co.uk> Michael, I'm using Lua as I want to extend this to do things such as pull a user's phone number from LDAP (with lualdap), wait a couple of seconds before placing the outbound call in case it hits voicemail on the Mitel (so we don't connect an internal extension's voicemail to an external party) etc. I also found a simple originate rings both endpoints at once. I don't want the second endpoint to be called until the first is answered. Regardless, the Lua should work but it doesn't for me, and I'd like to know if I'm doing something wrong. Cheers Alex On 14/12/11 23:01, Michael Collins wrote: > Why are you using Lua at all? It looks like a simple originate that > you should be able to do as an API call. > > -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/20111215/2a166049/attachment.html From fdelawarde at wirelessmundi.com Thu Dec 15 12:16:39 2011 From: fdelawarde at wirelessmundi.com (=?ISO-8859-1?Q?Fran=E7ois?= Delawarde) Date: Thu, 15 Dec 2011 10:16:39 +0100 Subject: [Freeswitch-users] Reinstall FreeSWITCH after new kernel? In-Reply-To: References: Message-ID: <1323940599.15906.281.camel@luna.madrid.commsmundi.com> Did you test with your 100hz kernel? It might work for your scenario. On Thu, 2011-12-15 at 00:50 +0100, Henrik Aagaard S?rensen wrote: > I've read a lot about the kernel timer and how it should be on 1000hz. > My clean install of Ubuntu has 100hz. So I guess I have to recompile > the kernel to 1000hz? > > > If so, does I have to reinstall FreeSWITCH after the new kernel > compile? > > > And does anyone knows of any tutorial about recompiling a Ubuntu 10.04 > LTS kernel to 1000hz? From fsrichard at ghz.fr Thu Dec 15 12:45:16 2011 From: fsrichard at ghz.fr (fsrichard at ghz.fr) Date: Thu, 15 Dec 2011 10:45:16 +0100 Subject: [Freeswitch-users] Howto Set default language for an application ? In-Reply-To: <4EE8EFF2.6090808@ghz.fr> References: <4EE8EFF2.6090808@ghz.fr> Message-ID: <4EE9C1AC.8000200@ghz.fr> Hello again, I'm sorry about the two e-mails asking the same question, I did not think the first one had been sent. I have found a work around but need your advice as I do not know why it was not done this way in the first place ! Old code (default in default.xml) : > Replacement code : > What is the difference between the two ? Why is the bridge application used to transfer to voicemail ? Is the replacement code the wrong approach ? Is it a question of how much ressources are used ? If so what is the real impact ? Even if my approach is correct, I'm still interested in knowing if there is a way to set the voicemail application default language. Thank you, Richard Le 14.12.11 19:50, fsrichard at ghz.fr a ?crit : > Hello, > > I'm trying to set up my first FreeSWITCH server but I don't know where > to set the default language corresponding to the following line in > conf/dialplan/default.xml : > >> >> ... >> >> ... >> > > I have tried this : > >> >> ... >> ** >> ... >> >> ... >> > > From what I have understood it is normal that it doesn't work as the > bridge to a loopback would use it's own settings, but I don't have a > clue where to set it of app=voicemail > > Is there a way to set the default language per SIP connection ? > > I have got an english and a french SIP line and would like to make > calls going to the english line have english instructions for leaving > a message on the voicemail and the calls to the french line to have > french instructions for leaving a message on the voicemail. > > Thank you > > Richard > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111215/59374026/attachment.html From freeswitch at earthspike.net Thu Dec 15 14:08:26 2011 From: freeswitch at earthspike.net (John) Date: Thu, 15 Dec 2011 11:08:26 +0000 Subject: [Freeswitch-users] Reinstall FreeSWITCH after new kernel? In-Reply-To: <1323940599.15906.281.camel@luna.madrid.commsmundi.com> References: <1323940599.15906.281.camel@luna.madrid.commsmundi.com> Message-ID: <4EE9D52A.1000805@earthspike.net> Henrik, I and some others have had problems with the 100Hz Ubuntu 10.04 LTS kernel. All appeared to work for some days but occasionally choppy calls occur, often only in one direction. We run a Sangoma BRI card (B700) and the problem manifests on the outgoing path. I have recently changed to the Ubuntu preemptive kernel (no kernel recompilation required!) which has the 1000Hz timer and low-latency interrupt response. After 10 days running so far, it appears to have cured the problem. Installing the pre-emptive kernel: sudo apt-get install linux-image-preempt linux-headers-preempt You will need the linux-headers-preempt if you need to rebuild kernel drivers for any hardware (which I do). You will be pleased to hear that you do not need to reinstall your FreeSWITCH installation; the whole job takes less than 5 minutes. John On 15/12/11 09:16, Fran?ois Delawarde wrote: > Did you test with your 100hz kernel? It might work for your scenario. > > > On Thu, 2011-12-15 at 00:50 +0100, Henrik Aagaard S?rensen wrote: >> I've read a lot about the kernel timer and how it should be on 1000hz. >> My clean install of Ubuntu has 100hz. So I guess I have to recompile >> the kernel to 1000hz? >> >> >> If so, does I have to reinstall FreeSWITCH after the new kernel >> compile? >> >> >> And does anyone knows of any tutorial about recompiling a Ubuntu 10.04 >> LTS kernel to 1000hz? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From miha at softnet.si Thu Dec 15 16:34:05 2011 From: miha at softnet.si (Miha Zoubek) Date: Thu, 15 Dec 2011 14:34:05 +0100 Subject: [Freeswitch-users] user directory variable Message-ID: <4EE9F74D.3000005@softnet.si> Hi, I would like to disable radius for incoming calls. This two variables disable_radius_start=true and disable_radius_stop=true are for disabling radius. I have put this in action application form in dialplan, I guess this is too late, as start packet is also send. I need to put this in user directory so that radius will be stoped in right time (and some logic that this would be only for incoming calls). Problem is that I have put this as variable (like this ) in user directory and the radius packet is still send. So why this variable is not set as packets are still send? Thank for your help! Regards, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. From henrikaagaardsorensen at gmail.com Thu Dec 15 17:09:57 2011 From: henrikaagaardsorensen at gmail.com (=?ISO-8859-1?Q?Henrik_Aagaard_S=F8rensen?=) Date: Thu, 15 Dec 2011 15:09:57 +0100 Subject: [Freeswitch-users] Reinstall FreeSWITCH after new kernel? In-Reply-To: <4EE9D52A.1000805@earthspike.net> References: <1323940599.15906.281.camel@luna.madrid.commsmundi.com> <4EE9D52A.1000805@earthspike.net> Message-ID: Thank you John. I'll try that. On Thu, Dec 15, 2011 at 12:08 PM, John wrote: > Henrik, > > I and some others have had problems with the 100Hz Ubuntu 10.04 LTS > kernel. All appeared to work for some days but occasionally choppy > calls occur, often only in one direction. We run a Sangoma BRI card > (B700) and the problem manifests on the outgoing path. I have recently > changed to the Ubuntu preemptive kernel (no kernel recompilation > required!) which has the 1000Hz timer and low-latency interrupt > response. After 10 days running so far, it appears to have cured the > problem. > > Installing the pre-emptive kernel: > > sudo apt-get install linux-image-preempt linux-headers-preempt > > You will need the linux-headers-preempt if you need to rebuild kernel > drivers for any hardware (which I do). > > You will be pleased to hear that you do not need to reinstall your > FreeSWITCH installation; the whole job takes less than 5 minutes. > > John > > On 15/12/11 09:16, Fran?ois Delawarde wrote: > > Did you test with your 100hz kernel? It might work for your scenario. > > > > > > On Thu, 2011-12-15 at 00:50 +0100, Henrik Aagaard S?rensen wrote: > >> I've read a lot about the kernel timer and how it should be on 1000hz. > >> My clean install of Ubuntu has 100hz. So I guess I have to recompile > >> the kernel to 1000hz? > >> > >> > >> If so, does I have to reinstall FreeSWITCH after the new kernel > >> compile? > >> > >> > >> And does anyone knows of any tutorial about recompiling a Ubuntu 10.04 > >> LTS kernel to 1000hz? > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111215/46732f86/attachment-0001.html From x.liu at hw.ac.uk Thu Dec 15 17:55:00 2011 From: x.liu at hw.ac.uk (x.liu) Date: Thu, 15 Dec 2011 14:55:00 +0000 Subject: [Freeswitch-users] How to escape double-quote in dialplan and use TTS SSML tags In-Reply-To: References: <1323940599.15906.281.camel@luna.madrid.commsmundi.com> <4EE9D52A.1000805@earthspike.net> Message-ID: <4EEA0A44.3020606@hw.ac.uk> Hi, I'd like to add SSML tag to my TTS string. I am sure my TTS server supports these tags. In this example, This is to test our text-to-speech system. "/> Whether I use or FS complains: unexpected closing tag If I use single quote like , FS starts ok but TTS will read out the string of And I notice that this string is still read out even if it is passed as double quoted ( ) through my ESL app. Any advices for using SSML in unimrcp TTS string? Many thanks! Xing -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 From orbit at klbank.ru Thu Dec 15 14:34:08 2011 From: orbit at klbank.ru (Zhuravlov Sergey) Date: Thu, 15 Dec 2011 15:34:08 +0400 Subject: [Freeswitch-users] from FS to * CALL_REJECTED Message-ID: <20111215113405.GA13615@klbank.ru> Good day! Can not connect sobstvennye FS and asterisk. I do, as described here http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk !!!With ACLs!!! call with asterisk to FS happened, but with FS to asterisk call fail: call is made so EXECUTE sofia/internal/272 at my.domain.com:5060 bridge(sofia/external/711 at xx.xxx.xx.xx) http://pastebin.freeswitch.org/18014 and packets are not with FS in the direction of asterisk is not going [root at fs0 freeswitch]# tcpdump -nS -i venet0:0 dst xx.xxx.xx.xx where xx.xxx.xx.xx -- asterisk what could be the reason? -- Zhuravlov Sergey GTALK/JABBER:4orbit at gmail.com From Hector.Geraldino at ip-soft.net Thu Dec 15 19:22:50 2011 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Thu, 15 Dec 2011 11:22:50 -0500 Subject: [Freeswitch-users] Lazarus - FreeSwitch In-Reply-To: References: <4EE8EFF2.6090808@ghz.fr> Message-ID: <6A6B4C284AD15042B429EB9D904544AD0225506EA1@NY1-EXMB-01.ip-soft.net> Hello Freddy, I don't think there's an ESL client library for lazarus/pascal. My best guess is that you can reuse the one written in C if lazarus can P/Invoke system libraries (I'm not familiar with Lazarus at all). Second choice (and a harder one) would be to create your own ESL client. This involves listening on a socket port, parse the events sent by FS and send commands back to freeswitch using sockets. You can take a look at the Java ESL client library source code (org.freeswitch.esl.client) as a reference in case you decide to embark yourself in this titanic task. http://wiki.freeswitch.org/wiki/Java_ESL Good luck! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Fredy Gonzales Sent: Wednesday, December 14, 2011 2:08 PM To: FreeSWITCH Users Help Subject: [Freeswitch-users] Lazarus - FreeSwitch Hello, Can anybody help me and tell me how do install a module FreeSwitch with Lazarus-Pascal.... And there is any other way beside socket and it is have to be with socket what module activated Saludos. Fredy G. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111215/2247284f/attachment.html From msc at freeswitch.org Thu Dec 15 19:33:31 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 15 Dec 2011 08:33:31 -0800 Subject: [Freeswitch-users] How to escape double-quote in dialplan and use TTS SSML tags In-Reply-To: <4EEA0A44.3020606@hw.ac.uk> References: <1323940599.15906.281.camel@luna.madrid.commsmundi.com> <4EE9D52A.1000805@earthspike.net> <4EEA0A44.3020606@hw.ac.uk> Message-ID: How about using CDATA? Like this: This is to test our text-to-speech system.]] That's just off the top of my head so please tinker with it if it doesn't work on the first try. -MC On Thu, Dec 15, 2011 at 6:55 AM, x.liu wrote: > Hi, > > I'd like to add SSML tag to my TTS string. I am sure my TTS server > supports these tags. > > In this example, > > This > is to test our text-to-speech system. "/> > > Whether I use or > FS complains: unexpected closing tag > > If I use single quote like , FS starts ok but TTS > will read out the string of > > > And I notice that this string is still read out even if it is passed as > double quoted ( ) through my ESL app. > > Any advices for using SSML in unimrcp TTS string? > > Many thanks! > > Xing > > > > -- > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > Heriot-Watt University is the Sunday Times > Scottish University of the Year 2011-2012 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111215/cafe4912/attachment.html From msc at freeswitch.org Thu Dec 15 19:34:37 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 15 Dec 2011 08:34:37 -0800 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: References: Message-ID: Correct. We decided to use torrents to consolidate our media files and use our CDN for stuff like this. -MC On Wed, Dec 14, 2011 at 9:46 PM, William Suffill wrote: > No more direct download links for the recordings? > > -- W > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111215/5612ba02/attachment-0001.html From msc at freeswitch.org Thu Dec 15 19:37:40 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 15 Dec 2011 08:37:40 -0800 Subject: [Freeswitch-users] from FS to * CALL_REJECTED In-Reply-To: <20111215113405.GA13615@klbank.ru> References: <20111215113405.GA13615@klbank.ru> Message-ID: Most likely this is a configuration issue on the Asterisk side. Have you set up the Asterisk config according to this section of that page? http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk#Asterisk_Side_2 If Asterisk is rejecting the call then you should probably look at the Asterisk console for clues. -MC On Thu, Dec 15, 2011 at 3:34 AM, Zhuravlov Sergey wrote: > Good day! > Can not connect sobstvennye FS and asterisk. > I do, as described here > > http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk > > !!!With ACLs!!! > > call with asterisk to FS happened, > but with FS to asterisk call fail: > > > > call is made so > > EXECUTE sofia/internal/272 at my.domain.com:5060bridge(sofia/external/711 at xx.xxx.xx.xx > ) > > > http://pastebin.freeswitch.org/18014 > > > and packets are not with FS in the direction of asterisk is not going > > > [root at fs0 freeswitch]# tcpdump -nS -i venet0:0 dst xx.xxx.xx.xx > > > > where xx.xxx.xx.xx -- asterisk > > > > what could be the reason? > > -- > Zhuravlov Sergey > > GTALK/JABBER:4orbit at gmail.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/20111215/34d2c69c/attachment.html From anthony.minessale at gmail.com Thu Dec 15 19:48:26 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 15 Dec 2011 10:48:26 -0600 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today In-Reply-To: References: Message-ID: I suggest we make these announcements earlier and get speakers lined up a few weeks in advance to give people a chance to prepare. On Thu, Dec 15, 2011 at 10:34 AM, Michael Collins wrote: > Correct. We decided to use torrents to consolidate our media files and use > our CDN for stuff like this. > > -MC > > > On Wed, Dec 14, 2011 at 9:46 PM, William Suffill < > william.suffill at gmail.com> wrote: > >> No more direct download links for the recordings? >> >> -- W >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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/20111215/8e3f63e5/attachment.html From avi at avimarcus.net Thu Dec 15 20:00:52 2011 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 15 Dec 2011 19:00:52 +0200 Subject: [Freeswitch-users] Lazarus - FreeSwitch In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD0225506EA1@NY1-EXMB-01.ip-soft.net> References: <4EE8EFF2.6090808@ghz.fr> <6A6B4C284AD15042B429EB9D904544AD0225506EA1@NY1-EXMB-01.ip-soft.net> Message-ID: Might 0mq help, they might have bindings. On Dec 15, 2011 6:24 PM, "Hector Geraldino" wrote: Hello Freddy,**** ** ** I don?t think there?s an ESL client library for lazarus/pascal. My best guess is that you can reuse the one written in C if lazarus can P/Invoke system libraries (I?m not familiar with Lazarus at all). **** ** ** Second choice (and a harder one) would be to create your own ESL client. This involves listening on a socket port, parse the events sent by FS and send commands back to freeswitch using sockets. You can take a look at the Java ESL client library source code (org.freeswitch.esl.client) as a reference in case you decide to embark yourself in this titanic task.**** ** ** http://wiki.freeswitch.org/wiki/Java_ESL**** ** ** Good luck!**** ** ** *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Fredy Gonzales *Sent:* Wednesday, December 14, 2011 2:08 PM *To:* FreeSWITCH Users Help *Subject:* [Freeswitch-users] Lazarus - FreeSwitch**** Hello, Can anybody help me and tell me how do install a module FreeSwitch with Lazarus-Pasc... _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111215/e3774cbd/attachment.html From fraserredmond at gmail.com Thu Dec 15 20:11:11 2011 From: fraserredmond at gmail.com (Fraser Redmond) Date: Thu, 15 Dec 2011 12:11:11 -0500 Subject: [Freeswitch-users] Logging/monitor call-quality indicators Message-ID: How do you monitor the health of your servers/calls? I come from a web-development background where you can track things like how long it takes to generate a page, run a sql query, how many php or sql errors you get, and so on, then go back through the logs to solve problems that hadn't been reported - it'd be nice to do similar for voip, especially as there are so many more variables beyond my control. I was wondering if there are any ways to go about logging things related to call-quality like audio-latency, jitter, packet-loss, sip-errors, etc. I know about sip-traces, and the FS log, but that's too granular - I first would need to know a certain type of problem is occurring before I can investigate individual calls related to it. The only thing I can find in the wiki is PBXMate - but there isn't any mention of it in the maililng-list. Is anyone using it? Any feedback? Cheers, Fraser -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111215/a321078a/attachment-0001.html From x.liu at hw.ac.uk Thu Dec 15 20:20:55 2011 From: x.liu at hw.ac.uk (x.liu) Date: Thu, 15 Dec 2011 17:20:55 +0000 Subject: [Freeswitch-users] How to escape double-quote in dialplan and use TTS SSML tags In-Reply-To: References: <1323940599.15906.281.camel@luna.madrid.commsmundi.com> <4EE9D52A.1000805@earthspike.net> <4EEA0A44.3020606@hw.ac.uk> Message-ID: <4EEA2C77.9030201@hw.ac.uk> Thanks, Michael! Your suggestion works.(there is a typo: !CDATA should be ![CDATA ) And following two ways also work: This is to test our text-to-speech system.]]> this one uses single quote for data= and escape double quote: But it looks like the apostrophe would be removed by FS, e.g. From the FS terminal and log I see if Im wrong, how about Miros Cantina Mexicana? Don't know yet how to keep the apostrophe. On 12/15/2011 04:33 PM, Michael Collins wrote: > How about using CDATA? Like this: > > emotion="calm">This > is to test our text-to-speech system.]] > > > That's just off the top of my head so please tinker with it if it > doesn't work on the first try. > > -MC > > On Thu, Dec 15, 2011 at 6:55 AM, x.liu > wrote: > > Hi, > > I'd like to add SSML tag to my TTS string. I am sure my TTS server > supports these tags. > > In this example, > > This > is to test our text-to-speech system. "/> > > Whether I use or > FS complains: unexpected closing tag > > If I use single quote like , FS starts ok > but TTS > will read out the string of > > > And I notice that this string is still read out even if it is > passed as > double quoted ( ) through my ESL app. > > Any advices for using SSML in unimrcp TTS string? > > Many thanks! > > Xing > > > > -- > Heriot-Watt University is a Scottish charity > registered under charity number SC000278. > > Heriot-Watt University is the Sunday Times > Scottish University of the Year 2011-2012 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Heriot-Watt University is a Scottish charity registered under charity number SC000278. Heriot-Watt University is the Sunday Times Scottish University of the Year 2011-2012 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111215/15b1c56b/attachment.html From acrow at integrafin.co.uk Thu Dec 15 20:45:22 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Thu, 15 Dec 2011 17:45:22 +0000 Subject: [Freeswitch-users] {Disarmed} Re: Lua - origination to local endpoint then bridge to either local or remote destination: no audio In-Reply-To: <4EE9A3C5.3030407@integrafin.co.uk> References: <4EE905F8.6010908@integrafin.co.uk> <4EE9A3C5.3030407@integrafin.co.uk> Message-ID: <4EEA3232.90807@integrafin.co.uk> BTW, It does work OK when bridging to the Mitel even for both legs. We have a plan to move away from the Mitel to a pure FS environment by stages, so we'd like to have something like this working beforehand. Cheers Alex On 15/12/11 07:37, Alex Crow wrote: > Michael, > > I'm using Lua as I want to extend this to do things such as pull a > user's phone number from LDAP (with lualdap), wait a couple of seconds > before placing the outbound call in case it hits voicemail on the > Mitel (so we don't connect an internal extension's voicemail to an > external party) etc. > > I also found a simple originate rings both endpoints at once. I don't > want the second endpoint to be called until the first is answered. > > Regardless, the Lua should work but it doesn't for me, and I'd like to > know if I'm doing something wrong. > > Cheers > > Alex > > On 14/12/11 23:01, Michael Collins wrote: >> Why are you using Lua at all? It looks like a simple originate that >> you should be able to do as an API call. >> >> -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 > > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111215/b71799ce/attachment.html From anthony.minessale at gmail.com Thu Dec 15 21:15:45 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 15 Dec 2011 12:15:45 -0600 Subject: [Freeswitch-users] Reinstall FreeSWITCH after new kernel? In-Reply-To: References: <1323940599.15906.281.camel@luna.madrid.commsmundi.com> <4EE9D52A.1000805@earthspike.net> Message-ID: If you use latest GIT, it uses timerfd which should work regardless of the kernel speed, ticklessness etc. 2011/12/15 Henrik Aagaard S?rensen > Thank you John. I'll try that. > > > On Thu, Dec 15, 2011 at 12:08 PM, John wrote: > >> Henrik, >> >> I and some others have had problems with the 100Hz Ubuntu 10.04 LTS >> kernel. All appeared to work for some days but occasionally choppy >> calls occur, often only in one direction. We run a Sangoma BRI card >> (B700) and the problem manifests on the outgoing path. I have recently >> changed to the Ubuntu preemptive kernel (no kernel recompilation >> required!) which has the 1000Hz timer and low-latency interrupt >> response. After 10 days running so far, it appears to have cured the >> problem. >> >> Installing the pre-emptive kernel: >> >> sudo apt-get install linux-image-preempt linux-headers-preempt >> >> You will need the linux-headers-preempt if you need to rebuild kernel >> drivers for any hardware (which I do). >> >> You will be pleased to hear that you do not need to reinstall your >> FreeSWITCH installation; the whole job takes less than 5 minutes. >> >> John >> >> On 15/12/11 09:16, Fran?ois Delawarde wrote: >> > Did you test with your 100hz kernel? It might work for your scenario. >> > >> > >> > On Thu, 2011-12-15 at 00:50 +0100, Henrik Aagaard S?rensen wrote: >> >> I've read a lot about the kernel timer and how it should be on 1000hz. >> >> My clean install of Ubuntu has 100hz. So I guess I have to recompile >> >> the kernel to 1000hz? >> >> >> >> >> >> If so, does I have to reinstall FreeSWITCH after the new kernel >> >> compile? >> >> >> >> >> >> And does anyone knows of any tutorial about recompiling a Ubuntu 10.04 >> >> LTS kernel to 1000hz? >> > >> > >> > >> _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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/20111215/e4c1b5d0/attachment-0001.html From freeswitch at earthspike.net Thu Dec 15 21:24:39 2011 From: freeswitch at earthspike.net (John) Date: Thu, 15 Dec 2011 18:24:39 +0000 Subject: [Freeswitch-users] Reinstall FreeSWITCH after new kernel? In-Reply-To: References: <1323940599.15906.281.camel@luna.madrid.commsmundi.com> <4EE9D52A.1000805@earthspike.net> Message-ID: <4EEA3B67.8090903@earthspike.net> When was that introduced? On 15/12/11 18:15, Anthony Minessale wrote: > If you use latest GIT, it uses timerfd which should work regardless of > the kernel speed, ticklessness etc. > > > > 2011/12/15 Henrik Aagaard S?rensen > > > Thank you John. I'll try that. > > > On Thu, Dec 15, 2011 at 12:08 PM, John > wrote: > > Henrik, > > I and some others have had problems with the 100Hz Ubuntu > 10.04 LTS > kernel. All appeared to work for some days but occasionally > choppy > calls occur, often only in one direction. We run a Sangoma > BRI card > (B700) and the problem manifests on the outgoing path. I have > recently > changed to the Ubuntu preemptive kernel (no kernel recompilation > required!) which has the 1000Hz timer and low-latency interrupt > response. After 10 days running so far, it appears to have > cured the > problem. > > Installing the pre-emptive kernel: > > sudo apt-get install linux-image-preempt linux-headers-preempt > > You will need the linux-headers-preempt if you need to rebuild > kernel > drivers for any hardware (which I do). > > You will be pleased to hear that you do not need to reinstall your > FreeSWITCH installation; the whole job takes less than 5 minutes. > > John > > On 15/12/11 09:16, Fran?ois Delawarde wrote: > > Did you test with your 100hz kernel? It might work for your > scenario. > > > > > > On Thu, 2011-12-15 at 00:50 +0100, Henrik Aagaard S?rensen > wrote: > >> I've read a lot about the kernel timer and how it should be > on 1000hz. > >> My clean install of Ubuntu has 100hz. So I guess I have to > recompile > >> the kernel to 1000hz? > >> > >> > >> If so, does I have to reinstall FreeSWITCH after the new kernel > >> compile? > >> > >> > >> And does anyone knows of any tutorial about recompiling a > Ubuntu 10.04 > >> LTS kernel to 1000hz? > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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/20111215/a42b53a6/attachment.html From wstephen80 at gmail.com Thu Dec 15 22:03:54 2011 From: wstephen80 at gmail.com (Stephen Wilde) Date: Thu, 15 Dec 2011 20:03:54 +0100 Subject: [Freeswitch-users] The system cannot create any sessions at this time Message-ID: Hi, what are the possible cause of the critical errors: 2011-12-15 19:14:37.538948 [CRIT] mod_sofia.c:4189 Error Creating Session 2011-12-15 19:14:37.538948 [CRIT] switch_core_session.c:1775 The system cannot create any sessions at this time. 2011-12-15 19:14:37.538948 [CRIT] mod_sofia.c:4189 Error Creating Session 2011-12-15 19:14:37.548945 [CRIT] switch_core_session.c:1775 The system cannot create any sessions at this time. 2011-12-15 19:14:37.548945 [CRIT] mod_sofia.c:4189 Error Creating Session 2011-12-15 19:14:37.548945 [CRIT] switch_core_session.c:1775 The system cannot create any sessions at this time. 2011-12-15 19:14:37.548945 [CRIT] mod_sofia.c:4189 Error Creating Session and so on. This kind of errors appear in a specific time interval (for example from 19:14:31 until 19:14:39) and after this time interval all continue to work normally. This is not a "max_sessions" related problem because in this case appear a "Over Session Limit!" message. I think also that this is not a "min idle" problem because I have not set this parameter, my status is: UP 0 years, 0 days, 15 hours, 26 minutes, 32 seconds, 182 milliseconds, 131 microseconds FreeSWITCH is ready 4279392 session(s) since startup 3463 session(s) 30/100 6000 session(s) max min idle cpu 0.00/77.00 What are the possible causes of this message? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111215/6e066b4b/attachment.html From fsrichard at ghz.fr Fri Dec 16 00:53:55 2011 From: fsrichard at ghz.fr (fsrichard at ghz.fr) Date: Thu, 15 Dec 2011 22:53:55 +0100 Subject: [Freeswitch-users] Howto Set default language for an application ? In-Reply-To: <4EE9C1AC.8000200@ghz.fr> References: <4EE8EFF2.6090808@ghz.fr> <4EE9C1AC.8000200@ghz.fr> Message-ID: <4EEA6C73.9080600@ghz.fr> Hello, Just in cas any one is interested I got an answer from the IRC. > data="{default_language=fr}loopback/app=voicemail:default > ${domain_name} ${dialed_extension}"/> :) Le 15.12.11 10:45, fsrichard at ghz.fr a ?crit : > Hello again, I'm sorry about the two e-mails asking the same question, > I did not think the first one had been sent. > > I have found a work around but need your advice as I do not know why > it was not done this way in the first place ! > > Old code (default in default.xml) : >> > > Replacement code : > >> > > What is the difference between the two ? Why is the bridge application > used to transfer to voicemail ? Is the replacement code the wrong > approach ? Is it a question of how much ressources are used ? If so > what is the real impact ? > > Even if my approach is correct, I'm still interested in knowing if > there is a way to set the voicemail application default language. > > Thank you, > > Richard > > > Le 14.12.11 19:50, fsrichard at ghz.fr a ?crit : >> Hello, >> >> I'm trying to set up my first FreeSWITCH server but I don't know >> where to set the default language corresponding to the following line >> in conf/dialplan/default.xml : >> >>> >>> ... >>> >>> ... >>> >> >> I have tried this : >> >>> >>> ... >>> ** >>> ... >>> >>> ... >>> >> >> From what I have understood it is normal that it doesn't work as the >> bridge to a loopback would use it's own settings, but I don't have a >> clue where to set it of app=voicemail >> >> Is there a way to set the default language per SIP connection ? >> >> I have got an english and a french SIP line and would like to make >> calls going to the english line have english instructions for leaving >> a message on the voicemail and the calls to the french line to have >> french instructions for leaving a message on the voicemail. >> >> Thank you >> >> Richard >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111215/e7cd0e91/attachment-0001.html From Ryan at ocens.com Fri Dec 16 03:25:38 2011 From: Ryan at ocens.com (Ryan Watkins) Date: Fri, 16 Dec 2011 00:25:38 +0000 Subject: [Freeswitch-users] Issue adding a SIP Gateway Message-ID: <44E5C0A9D48A3246966A4AE04692014D102A6FB5@CH1PRD0604MB109.namprd06.prod.outlook.com> Hello all, Having an issue adding a gateway, and hoping someone can help. Please forgive me... still new to Freeswitch and XML, but I've exhausted all that I know, so I need to reach out. Running Ubuntu, and I installed Freeswitch following the wiki guide. Didn't appear to have any issues installing, didn't get any noticeable errors during the setup anyways. Been following the Freeswitch 1.0.6 book to add a gateway (using iptel) I've created the iptel.org.xml in both /opt/freeswitch/conf/sip_profiles/external and /usr/src/freeswitch/conf/sip_profiles/external However I am unable to get the gateway to load in fs_cli Looking at all the log data, it only gives notice regarding the example.com gateway... does not appear to mention iptel at all, and no err log shows regarding iptel I attempted to manually unregister/register iptel, but I get the following error when I try sofia profile external unregister iptel (same error with register iptel) Invalid gateway! Any ideas? Thanks Ryan Watkins Ryan Watkins Networking & Customer Support OCENS 19655 1st AVE S. #203 Seattle, WA 98148 Satellite Systems and Services: Iridium, Inmarsat, Globalstar, KVH ________________________________________________________ www.ocens.com | support.ocens.com Office: (206) 878-8270 | Cell: (360) 521-7334 | Fax: (206) 878-8314 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111216/0aa88727/attachment.html From ash at archerdrive.com Fri Dec 16 00:50:25 2011 From: ash at archerdrive.com (Ash) Date: Fri, 16 Dec 2011 08:50:25 +1100 Subject: [Freeswitch-users] Logging/monitor call-quality indicators In-Reply-To: References: Message-ID: <6D44E9E8-ABB7-423D-A78C-D645615D659B@archerdrive.com> Hi Fraser, Personally I use voipmonitor. It will monitor your server looking at the raw packets to the server. All data like jitter etc is stored in a mysql database. It also outputs the rrd call information to a directory so that you can graph latency etc. http://sourceforge.net/projects/voipmonitor/ If you don't want to write something there is already a monitoring gui, but you will have to pay for it - http://www.voipmonitor.org/ Ash. On 16/12/2011, at 4:11 AM, Fraser Redmond wrote: > How do you monitor the health of your servers/calls? > > I come from a web-development background where you can track things like how long it takes to generate a page, run a sql query, how many php or sql errors you get, and so on, then go back through the logs to solve problems that hadn't been reported - it'd be nice to do similar for voip, especially as there are so many more variables beyond my control. > > I was wondering if there are any ways to go about logging things related to call-quality like audio-latency, jitter, packet-loss, sip-errors, etc. > > I know about sip-traces, and the FS log, but that's too granular - I first would need to know a certain type of problem is occurring before I can investigate individual calls related to it. > > The only thing I can find in the wiki is PBXMate - but there isn't any mention of it in the maililng-list. Is anyone using it? Any feedback? > > Cheers, > Fraser > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111216/b1daa9db/attachment.html From justlikeef at gmail.com Fri Dec 16 03:54:01 2011 From: justlikeef at gmail.com (Rob Hutton) Date: Thu, 15 Dec 2011 19:54:01 -0500 Subject: [Freeswitch-users] Issue adding a SIP Gateway In-Reply-To: <44E5C0A9D48A3246966A4AE04692014D102A6FB5@CH1PRD0604MB109.namprd06.prod.outlook.com> References: <44E5C0A9D48A3246966A4AE04692014D102A6FB5@CH1PRD0604MB109.namprd06.prod.outlook.com> Message-ID: <201112151954.01880.justlikeef@gmail.com> Did you issue reloadxml? On Thursday 15 December 2011 19:25:38 Ryan Watkins wrote: > Hello all, > > Having an issue adding a gateway, and hoping someone can help. Please forgive me... still new to Freeswitch and XML, but I've exhausted all that I know, so I need to reach out. > > Running Ubuntu, and I installed Freeswitch following the wiki guide. Didn't appear to have any issues installing, didn't get any noticeable errors during the setup anyways. Been following the Freeswitch 1.0.6 book to add a gateway (using iptel) > > I've created the iptel.org.xml in both /opt/freeswitch/conf/sip_profiles/external and /usr/src/freeswitch/conf/sip_profiles/external > > However I am unable to get the gateway to load in fs_cli > Looking at all the log data, it only gives notice regarding the example.com gateway... does not appear to mention iptel at all, and no err log shows regarding iptel > > I attempted to manually unregister/register iptel, but I get the following error when I try sofia profile external unregister iptel (same error with register iptel) > > Invalid gateway! > > Any ideas? > > Thanks > Ryan Watkins > > > > Ryan Watkins > Networking & Customer Support > > OCENS > 19655 1st AVE S. #203 > Seattle, WA 98148 > > Satellite Systems and Services: Iridium, Inmarsat, Globalstar, KVH > ________________________________________________________ > www.ocens.com | support.ocens.com > Office: (206) 878-8270 | Cell: (360) 521-7334 | Fax: (206) 878-8314 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111215/3cfdf100/attachment.html From drew_terenzini at wesleycloversolutions.com Fri Dec 16 03:57:48 2011 From: drew_terenzini at wesleycloversolutions.com (Drew Terenzini) Date: Thu, 15 Dec 2011 16:57:48 -0800 Subject: [Freeswitch-users] freeswitch.email () failing Message-ID: <1014B6A210DF46F3B00BF7F6A51B31C6@DREWPC> Good evening, I'm attempting to use the freeswitch.email from http://wiki.freeswitch.org/wiki/Lua#freeswitch.email. I am running FS 1.04 on Ubuntu 10.04 LTS, and for various reasons having to do with programs running on top of FS, I cannot just upgrade FS without breaking other things. I have installed postfix successfully and tested sendmail from the Linux CLI without a problem. Emails are delivered to the appropriate external addresses. My switch.conf.xml has: -> . and I have a stripped down test script called "email.lua" in my scripts directory as follows: freeswitch.email("","","subject: Test message from VTAPENG\n","Hello you have a test message.") The destination and source email addresses are valid and receiving email when sent from other programs/sites. In the FS_CLI, I attempt to test-run the script by entering "lua email.lua", only to get the following error each time: -ERR encounterd freeswitch at vtap> 2011-12-15 16:36:19.246622 [ERR] mod_lua.cpp:182 /opt/freeswitch/scripts/email.lua:1: attempt to call field 'email' (a nil value) stack traceback: /opt/freeswitch/scripts/email.lua:1: in main chunk This error appears to terminate the script, and no response is received from postfix, and no error is logged to mail.log. I've tried a number of ways to modify the outcome, without real result. Searching on the Wiki and the mailing lists hasn't resulted in anything that has worked either. Is freeswitch.email() not supported in FS 1.04? Is there some context that I'm missing for this to work? Any help would be appreciated, thanks! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111215/b641e89e/attachment-0001.html From Ryan at ocens.com Fri Dec 16 04:05:54 2011 From: Ryan at ocens.com (Ryan Watkins) Date: Fri, 16 Dec 2011 01:05:54 +0000 Subject: [Freeswitch-users] Issue adding a SIP Gateway In-Reply-To: <201112151954.01880.justlikeef@gmail.com> References: <44E5C0A9D48A3246966A4AE04692014D102A6FB5@CH1PRD0604MB109.namprd06.prod.outlook.com> <201112151954.01880.justlikeef@gmail.com> Message-ID: <44E5C0A9D48A3246966A4AE04692014D102A803C@CH1PRD0604MB109.namprd06.prod.outlook.com> I did run the" sofia profile external restart reloadxml" command.... It didn't load the new gateway, so that's why I tried registering the gateway specifically. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Rob Hutton Sent: Thursday, December 15, 2011 4:54 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Issue adding a SIP Gateway Did you issue reloadxml? On Thursday 15 December 2011 19:25:38 Ryan Watkins wrote: > Hello all, > > Having an issue adding a gateway, and hoping someone can help. Please forgive me... still new to Freeswitch and XML, but I've exhausted all that I know, so I need to reach out. > > Running Ubuntu, and I installed Freeswitch following the wiki guide. Didn't appear to have any issues installing, didn't get any noticeable errors during the setup anyways. Been following the Freeswitch 1.0.6 book to add a gateway (using iptel) > > I've created the iptel.org.xml in both /opt/freeswitch/conf/sip_profiles/external and /usr/src/freeswitch/conf/sip_profiles/external > > However I am unable to get the gateway to load in fs_cli > Looking at all the log data, it only gives notice regarding the example.com gateway... does not appear to mention iptel at all, and no err log shows regarding iptel > > I attempted to manually unregister/register iptel, but I get the following error when I try sofia profile external unregister iptel (same error with register iptel) > > Invalid gateway! > > Any ideas? > > Thanks > Ryan Watkins > > > > Ryan Watkins > Networking & Customer Support > > OCENS > 19655 1st AVE S. #203 > Seattle, WA 98148 > > Satellite Systems and Services: Iridium, Inmarsat, Globalstar, KVH > ________________________________________________________ > www.ocens.com> | support.ocens.com > Office: (206) 878-8270 | Cell: (360) 521-7334 | Fax: (206) 878-8314 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111216/9871ee37/attachment.html From xing2kin at yahoo.com Fri Dec 16 07:14:02 2011 From: xing2kin at yahoo.com (king2kin) Date: Thu, 15 Dec 2011 20:14:02 -0800 (PST) Subject: [Freeswitch-users] Question Answertime variable? billsec is not available In-Reply-To: References: <1323497014.35405.YahooMailNeo@web39705.mail.mud.yahoo.com> Message-ID: <1324008842.58067.YahooMailNeo@web39706.mail.mud.yahoo.com> MC, ? Thank you for your reply. The problem still exists even though I follow your advice to access?the variable 'billsec'?after the channel hangup is complete. ? lua script: { session:answer() ......... session:hangup() session:sleep(1250) ? ctsec = session:getVariable('billsec') freeswitch.consoleLog("INFO","***** Call-Time: sec=" .. ctsec .. "\n") } ? freeswitch log info from console are copied as follows: { freeswitch at W2k3T602> 2011-12-16 12:06:01.187500 [INFO] switch_rtp.c:3152 Auto Ch anging port from 127.0.0.1:40036 to 192.168.0.119:40036 2011-12-16 12:06:03.234375 [INFO] switch_cpp.cpp:1199 Prompt file is 'ivr\ivr-we lcome_to_freeswitch.wav' 2011-12-16 12:06:19.156250 [NOTICE] switch_cpp.cpp:620 Hangup sofia/internal/sip :1006 at 127.0.0.1:1312 [CS_EXECUTE] [NORMAL_CLEARING] 2011-12-16 12:06:24.515625 [ERR] mod_lua.cpp:191 C:\c4dev\freeswitch\Release\scr ipts/test1.lua:61: attempt to concatenate global 'ctsec' (a nil value) stack traceback: ??????? C:\c4dev\freeswitch\Release\scripts/test1.lua:61: in main chunk 2011-12-16 12:06:24.515625 [NOTICE] switch_core_session.c:1354 Session 6 (sofia/ internal/sip:1006 at 127.0.0.1:1312) Ended 2011-12-16 12:06:24.515625 [NOTICE] switch_core_session.c:1356 Close Channel sof ia/internal/sip:1006 at 127.0.0.1:1312 [CS_DESTROY] } ? please continue to explain why this happens. ? x.k. ? From: Michael Collins To: FreeSWITCH Users Help Sent: Wednesday, December 14, 2011 8:38 AM Subject: Re: [Freeswitch-users] Question Answertime variable? billsec is not available Are you trying to access these before the channel hangup is complete? If so then yes, they will always be empty. They are not populated until the call actually ends. Are you doing a hangup hook in Lua or something else? -MC On Fri, Dec 9, 2011 at 10:03 PM, king2kin wrote: The channel variables for this purpose seems to be >{duration, billsec, answersec, flow_billsec, mduration, billmsec, answermsec, uduration, billusec, answerusec} >see: source code "switch_channel.c"; >and wiki document on categories: channel variables >see: http://wiki.freeswitch.com/index.php?title=Category:Variable&from=Variable+billmsec? >? >However, when I tried to access these channel variables from Lua script or dialplan, their values are always "nil" or empty. >? >-- dialplan: >{ >?? >????........ >???? >????? >????? >} >? >-- Lua script: >2011-12-10 13:39:55.609375 [ERR] mod_lua.cpp:191 C:\c4dev\freeswitch\Release\scr >ipts/test1.lua:60: attempt to concatenate global 'ctsec' (a nil value) >stack traceback: >??????? C:\c4dev\freeswitch\Release\scripts/test1.lua:60: in main chunk >? >see: >{ >session:answer() >......... >ctsec = session:getVariable('billsec') >ctmsec = session:getVariable('billmsec') >freeswitch.consoleLog("INFO","***** Call-Time: sec=" .. ctsec .. "\n") >freeswitch.consoleLog("INFO","***** Call-Time: msec=" .. ctmsec .. "\n") >session:hangup() >} >? > > >From: curriegrad2004 >To: FreeSWITCH Users Help >Sent: Wednesday, November 30, 2011 11:15 PM >Subject: Re: [Freeswitch-users] Question Answertime variable? > >billseconds or bill_msec would be the one to turn to. > >On Wed, Nov 30, 2011 at 4:22 AM, Thomas Hoellriegel wrote: >> Hi all, >> Is there a possibility to determine the caller time for a channel at >> the end of a call? >> >> For example: in asterisk exists a variable: >> ${ANSWEREDTIME} >> I like to store the Answertime in a Database for example: >> Dailly minutes to call a Cellphone. >> Can your help plese? >> 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 >> > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111215/4ffffa7b/attachment-0001.html From msc at freeswitch.org Fri Dec 16 08:30:38 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 15 Dec 2011 21:30:38 -0800 Subject: [Freeswitch-users] {Disarmed} Re: Lua - origination to local endpoint then bridge to either local or remote destination: no audio In-Reply-To: <4EEA3232.90807@integrafin.co.uk> References: <4EE905F8.6010908@integrafin.co.uk> <4EE9A3C5.3030407@integrafin.co.uk> <4EEA3232.90807@integrafin.co.uk> Message-ID: Well, I got this script working with non-Snom phones: -- test_call -- create first leg, wait, then connect 2nd leg -- local calling_user = argv[1]; local called_num = argv[2]; freeswitch.consoleLog("INFO","Attempting to contact user " .. calling_user .. "\n"); local session1 = freeswitch.Session("{origination_caller_id_number=9876}user/"..calling_user); leg1_dispo = 'None'; while (session1:ready() == true and leg1_dispo ~= 'ANSWER') do leg1_dispo = session1:getVariable("endpoint_disposition"); freeswitch.consoleLog("INFO","Leg 1 disposition:" .. leg1_dispo .. "\n"); os.execute("sleep 1"); end if ( not session1:ready() ) then -- Oops, leg 1 hung up. Bummer. freeswitch.consoleLog("INFO","It appears that " .. calling_user .. " disconnected...\n") else freeswitch.consoleLog("INFO","Playing a prompt to " .. calling_user .. "\n"); session1:streamFile('ivr/ivr-hold_connect_call.wav'); session2 = freeswitch.Session("{origination_caller_id_number=" .. calling_user .."}user/" .. called_num); leg2_dispo = 'None'; while(session1:ready() and session2:ready() and leg2_dispo ~= "ANSWER") do if ( not session1:ready() ) then -- oops, leg 1 hung up freeswitch.consoleLog("INFO","Well, it appears that " .. calling_user .. " has disconnected.\n"); else os.execute("sleep 1"); leg2_dispo = session2:getVariable("endpoint_disposition"); freeswitch.consoleLog("INFO","Leg 2 disposition: " .. leg2_dispo .. "\n"); end end -- While if ( session1:ready() and session2:ready() ) then -- Looks good, bridge 'em freeswitch.bridge(session1,session2); else -- Uh oh, someone went away freeswitch.consoleLog("INFO","Somebody hung up :(\n"); end end FWIW, I drew inspiration from this: http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control I'm still not a fan of doing it this way, but at least you know it's possible. (Personally I prefer an ESL-ish approach.) -MC On Thu, Dec 15, 2011 at 9:45 AM, Alex Crow wrote: > ** > BTW, > > It does work OK when bridging to the Mitel even for both legs. We have a > plan to move away from the Mitel to a pure FS environment by stages, so > we'd like to have something like this working beforehand. > > Cheers > > Alex > > On 15/12/11 07:37, Alex Crow wrote: > > Michael, > > I'm using Lua as I want to extend this to do things such as pull a user's > phone number from LDAP (with lualdap), wait a couple of seconds before > placing the outbound call in case it hits voicemail on the Mitel (so we > don't connect an internal extension's voicemail to an external party) etc. > > I also found a simple originate rings both endpoints at once. I don't want > the second endpoint to be called until the first is answered. > > Regardless, the Lua should work but it doesn't for me, and I'd like to > know if I'm doing something wrong. > > Cheers > > Alex > > On 14/12/11 23:01, Michael Collins wrote: > > Why are you using Lua at all? It looks like a simple originate that you > should be able to do as an API call. > > -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 > > > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services:consulting at freeswitch.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 > > > > -- > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > > "Transact" is operated by Integrated Financial Arrangements plc > Domain House, 5-7 Singer Street, London EC2A 4BQ > Tel: (020) 7608 4900 Fax: (020) 7608 5300 > (Registered office: as above; Registered in England and Wales under number: 3727592) > Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111215/0ac5a7af/attachment.html From orbit at klbank.ru Fri Dec 16 10:18:51 2011 From: orbit at klbank.ru (Zhuravlov Sergey) Date: Fri, 16 Dec 2011 11:18:51 +0400 Subject: [Freeswitch-users] from FS to * CALL_REJECTED In-Reply-To: References: <20111215113405.GA13615@klbank.ru> Message-ID: <20111216071850.GA8161@klbank.ru> Thank you! Problem solved after replacing in file conf/sip_profiles/external.xml from to I use a multi-domain configuration apparently fail outgoing call on ACL on FS. On Thu, Dec 15, 2011 at 08:37:40AM -0800, Michael Collins wrote: > Most likely this is a configuration issue on the Asterisk side. Have you > set up the Asterisk config according to this section of that page? > http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk#Asterisk_Side_2 > > If Asterisk is rejecting the call then you should probably look at the > Asterisk console for clues. > > -MC > > On Thu, Dec 15, 2011 at 3:34 AM, Zhuravlov Sergey wrote: > > > Good day! > > Can not connect sobstvennye FS and asterisk. > > I do, as described here > > > > http://wiki.freeswitch.org/wiki/Connecting_Freeswitch_And_Asterisk > > > > !!!With ACLs!!! > > > > call with asterisk to FS happened, > > but with FS to asterisk call fail: > > > > > > > > call is made so > > > > EXECUTE sofia/internal/272 at my.domain.com:5060bridge(sofia/external/711 at xx.xxx.xx.xx > > ) > > > > > > http://pastebin.freeswitch.org/18014 > > > > > > and packets are not with FS in the direction of asterisk is not going > > > > > > [root at fs0 freeswitch]# tcpdump -nS -i venet0:0 dst xx.xxx.xx.xx > > > > > > > > where xx.xxx.xx.xx -- asterisk > > > > > > > > what could be the reason? > > > > -- > > Zhuravlov Sergey > > > > GTALK/JABBER:4orbit at gmail.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 > > -- Zhuravlov Sergey GTALK/JABBER:4orbit at gmail.com From B.Tietz at pinguin.ag Fri Dec 16 10:29:50 2011 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Fri, 16 Dec 2011 08:29:50 +0100 Subject: [Freeswitch-users] Issue adding a SIP Gateway In-Reply-To: <44E5C0A9D48A3246966A4AE04692014D102A803C@CH1PRD0604MB109.namprd06.prod.outlook.com> References: <44E5C0A9D48A3246966A4AE04692014D102A6FB5@CH1PRD0604MB109.namprd06.prod.outlook.com> <201112151954.01880.justlikeef@gmail.com> <44E5C0A9D48A3246966A4AE04692014D102A803C@CH1PRD0604MB109.namprd06.prod.outlook.com> Message-ID: <07BF4904977CC645B485E970424193AD0E68FD5299@localhost> Hi, try 'sofia profile external rescan restart reloadxml' But I'm not sure if you have to restart Freeswitch or 'reload mod_sofia' (dropping calls!) because you made changes in 'switch.conf.xml' with adding a gateway like mentioned here: http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files#Reloading greets. I did run the" sofia profile external restart reloadxml" command.... It didn't load the new gateway, so that's why I tried registering the gateway specifically. Did you issue reloadxml? On Thursday 15 December 2011 19:25:38 Ryan Watkins wrote: > Hello all, > > Having an issue adding a gateway, and hoping someone can help. Please forgive me... still new to Freeswitch and XML, but I've exhausted all that I know, so I need to reach out. > > Running Ubuntu, and I installed Freeswitch following the wiki guide. Didn't appear to have any issues installing, didn't get any noticeable errors during the setup anyways. Been following the Freeswitch 1.0.6 book to add a gateway (using iptel) > > I've created the iptel.org.xml in both /opt/freeswitch/conf/sip_profiles/external and /usr/src/freeswitch/conf/sip_profiles/external > > However I am unable to get the gateway to load in fs_cli > Looking at all the log data, it only gives notice regarding the example.com gateway... does not appear to mention iptel at all, and no err log shows regarding iptel > > I attempted to manually unregister/register iptel, but I get the following error when I try sofia profile external unregister iptel (same error with register iptel) > > Invalid gateway! > > Any ideas? > > Thanks > Ryan Watkins > > > > Ryan Watkins > Networking & Customer Support > > OCENS > 19655 1st AVE S. #203 > Seattle, WA 98148 > > Satellite Systems and Services: Iridium, Inmarsat, Globalstar, KVH > ________________________________________________________ > www.ocens.com> | support.ocens.com > Office: (206) 878-8270 | Cell: (360) 521-7334 | Fax: (206) 878-8314 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111216/715a09bc/attachment-0001.html From acrow at integrafin.co.uk Fri Dec 16 10:54:18 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Fri, 16 Dec 2011 07:54:18 +0000 Subject: [Freeswitch-users] {Disarmed} Re: Lua - origination to local endpoint then bridge to either local or remote destination: no audio In-Reply-To: References: <4EE905F8.6010908@integrafin.co.uk> <4EE9A3C5.3030407@integrafin.co.uk> <4EEA3232.90807@integrafin.co.uk> Message-ID: <4EEAF92A.504@integrafin.co.uk> Michael, I did find out some more last night while trying to reproduce the issues using a simple "originate" (which I could), and discovered it was down to FS detecting some of my phones (on other internal network segments) as "UDP_NAT" even when they were not NATed. I changed my internal profile to comment out "apply-nat-acl" and then all my local LAN phones now were registered simply as "UDP". Now more situations work, however there is still at least one weird one. I have my mobile registered to a created "doublenat" profile (my FS box is behind NAT) following the doublenat example on the wiki. I also use the "external" profile to register to a provider. I can originate from this extension to a LAN extension fine, so NAT in this situation is working well. However originating from a LAN extension to a number via the provider above still gives silence at both ends. Strangely if I place the same call from the phone itself it works fine. I will keep working on this but do you have any ideas? Thanks Alex On 16/12/11 05:30, Michael Collins wrote: > Well, I got this script working with non-Snom phones: > > -- test_call > -- create first leg, wait, then connect 2nd leg > -- > local calling_user = argv[1]; > local called_num = argv[2]; > > freeswitch.consoleLog("INFO","Attempting to contact user " .. > calling_user .. "\n"); > local session1 = > freeswitch.Session("{origination_caller_id_number=9876}user/"..calling_user); > leg1_dispo = 'None'; > > while (session1:ready() == true and leg1_dispo ~= 'ANSWER') do > leg1_dispo = session1:getVariable("endpoint_disposition"); > freeswitch.consoleLog("INFO","Leg 1 disposition:" .. leg1_dispo .. > "\n"); > os.execute("sleep 1"); > end > > if ( not session1:ready() ) then > -- Oops, leg 1 hung up. Bummer. > freeswitch.consoleLog("INFO","It appears that " .. calling_user .. > " disconnected...\n") > else > freeswitch.consoleLog("INFO","Playing a prompt to " .. > calling_user .. "\n"); > session1:streamFile('ivr/ivr-hold_connect_call.wav'); > session2 = freeswitch.Session("{origination_caller_id_number=" .. > calling_user .."}user/" .. called_num); > leg2_dispo = 'None'; > while(session1:ready() and session2:ready() and leg2_dispo ~= > "ANSWER") do > if ( not session1:ready() ) then > -- oops, leg 1 hung up > freeswitch.consoleLog("INFO","Well, it appears that " .. > calling_user .. " has disconnected.\n"); > else > os.execute("sleep 1"); > leg2_dispo = session2:getVariable("endpoint_disposition"); > freeswitch.consoleLog("INFO","Leg 2 disposition: " .. > leg2_dispo .. "\n"); > end > end -- While > if ( session1:ready() and session2:ready() ) then > -- Looks good, bridge 'em > freeswitch.bridge(session1,session2); > else > -- Uh oh, someone went away > freeswitch.consoleLog("INFO","Somebody hung up :(\n"); > end > end > > FWIW, I drew inspiration from this: > http://wiki.freeswitch.org/wiki/Mod_lua#Example:_Call_Control > > I'm still not a fan of doing it this way, but at least you know it's > possible. (Personally I prefer an ESL-ish approach.) > > -MC > > On Thu, Dec 15, 2011 at 9:45 AM, Alex Crow > wrote: > > BTW, > > It does work OK when bridging to the Mitel even for both legs. We > have a plan to move away from the Mitel to a pure FS environment > by stages, so we'd like to have something like this working > beforehand. > > Cheers > > Alex > > On 15/12/11 07:37, Alex Crow wrote: >> Michael, >> >> I'm using Lua as I want to extend this to do things such as pull >> a user's phone number from LDAP (with lualdap), wait a couple of >> seconds before placing the outbound call in case it hits >> voicemail on the Mitel (so we don't connect an internal >> extension's voicemail to an external party) etc. >> >> I also found a simple originate rings both endpoints at once. I >> don't want the second endpoint to be called until the first is >> answered. >> >> Regardless, the Lua should work but it doesn't for me, and I'd >> like to know if I'm doing something wrong. >> >> Cheers >> >> Alex >> >> On 14/12/11 23:01, Michael Collins wrote: >>> Why are you using Lua at all? It looks like a simple originate >>> that you should be able to do as an API call. >>> >>> -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 >> >> >> -- >> This message has been scanned for viruses and >> dangerous content by *MailScanner* >> , and is >> believed to be clean. >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > -- > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > > "Transact" is operated by Integrated Financial Arrangements plc > Domain House, 5-7 Singer Street, London EC2A 4BQ > Tel: (020) 7608 4900 Fax: (020) 7608 5300 > (Registered office: as above; Registered in England and Wales under number: 3727592) > Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111216/8ea47253/attachment.html From miha at softnet.si Fri Dec 16 11:47:58 2011 From: miha at softnet.si (Miha Zoubek) Date: Fri, 16 Dec 2011 09:47:58 +0100 Subject: [Freeswitch-users] freeswitch logic before dialplan Message-ID: <4EEB05BE.6080008@softnet.si> Hi, how to disable one variable before it hit dialplan for incoming calls? regards, Miha -- Best regards / Lep Pozdrav Miha Zoubek Softnet d.o.o. From sharad at coraltele.com Fri Dec 16 14:47:55 2011 From: sharad at coraltele.com (sharad) Date: Fri, 16 Dec 2011 17:17:55 +0530 Subject: [Freeswitch-users] Freeswitch RxFax with T38 & Javascript References: Message-ID: <80DFFA470C59492DB12715AC051F9893@sharad> Hi All With xml dialplan, Rxfax with T38 is working very well. I am trying to do it differently - Means I want to take the call from xml to Javascript & from Javascript I want to answer the call & switch to T38. Plz let me know if someone has done it Thanks in advance. Sharad From msc at freeswitch.org Fri Dec 16 19:26:48 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 16 Dec 2011 08:26:48 -0800 Subject: [Freeswitch-users] Question Answertime variable? billsec is not available In-Reply-To: <1324008842.58067.YahooMailNeo@web39706.mail.mud.yahoo.com> References: <1323497014.35405.YahooMailNeo@web39705.mail.mud.yahoo.com> <1324008842.58067.YahooMailNeo@web39706.mail.mud.yahoo.com> Message-ID: pastebin the extension that calls this script and also the console log of the call from start to finish. -MC On Thu, Dec 15, 2011 at 8:14 PM, king2kin wrote: > MC, > > Thank you for your reply. The problem still exists even though I follow > your advice to access the variable 'billsec' after the channel hangup is > complete. > > lua script: > { > session:answer() > ......... > session:hangup() > session:sleep(1250) > > ctsec = session:getVariable('billsec') > freeswitch.consoleLog("INFO","***** Call-Time: sec=" .. ctsec .. "\n") > } > > freeswitch log info from console are copied as follows: > { > freeswitch at W2k3T602> 2011-12-16 12:06:01.187500 [INFO] switch_rtp.c:3152 > Auto Ch > anging port from 127.0.0.1:40036 to 192.168.0.119:40036 > 2011-12-16 12:06:03.234375 [INFO] switch_cpp.cpp:1199 Prompt file is > 'ivr\ivr-we > lcome_to_freeswitch.wav' > 2011-12-16 12:06:19.156250 [NOTICE] switch_cpp.cpp:620 Hangup > sofia/internal/sip > :1006 at 127.0.0.1:1312 [CS_EXECUTE] [NORMAL_CLEARING] > *2011-12-16 12:06:24.515625 [ERR] mod_lua.cpp:191 > C:\c4dev\freeswitch\Release\scr > ipts/test1.lua:61: attempt to concatenate global 'ctsec' (a nil value) > stack traceback: > C:\c4dev\freeswitch\Release\scripts/test1.lua:61: in main chunk > *2011-12-16 12:06:24.515625 [NOTICE] switch_core_session.c:1354 Session 6 > (sofia/ > internal/sip:1006 at 127.0.0.1:1312) Ended > 2011-12-16 12:06:24.515625 [NOTICE] switch_core_session.c:1356 Close > Channel sof > ia/internal/sip:1006 at 127.0.0.1:1312 [CS_DESTROY] > } > > please continue to explain why this happens. > > x.k. > > *From:* Michael Collins > *To:* FreeSWITCH Users Help > *Sent:* Wednesday, December 14, 2011 8:38 AM > *Subject:* Re: [Freeswitch-users] Question Answertime variable? billsec > is not available > > Are you trying to access these before the channel hangup is complete? If > so then yes, they will always be empty. They are not populated until the > call actually ends. Are you doing a hangup hook in Lua or something else? > > -MC > > On Fri, Dec 9, 2011 at 10:03 PM, king2kin wrote: > > The channel variables for this purpose seems to be > {duration, billsec, answersec, flow_billsec, mduration, billmsec, > answermsec, uduration, billusec, answerusec} > see: source code "switch_channel.c"; > and wiki document on categories: channel variables > see: > http://wiki.freeswitch.com/index.php?title=Category:Variable&from=Variable+billmsec > > > However, when I tried to access these channel variables from Lua script or > dialplan, their values are always "nil" or empty. > > -- dialplan: > { > > ........ > > > > } > > -- Lua script: > 2011-12-10 13:39:55.609375 [ERR] mod_lua.cpp:191 > C:\c4dev\freeswitch\Release\scr > ipts/test1.lua:60: attempt to concatenate global 'ctsec' (a nil value) > stack traceback: > C:\c4dev\freeswitch\Release\scripts/test1.lua:60: in main chunk > > see: > { > session:answer() > ......... > ctsec = session:getVariable('billsec') > ctmsec = session:getVariable('billmsec') > freeswitch.consoleLog("INFO","***** Call-Time: sec=" .. ctsec .. "\n") > freeswitch.consoleLog("INFO","***** Call-Time: msec=" .. ctmsec .. "\n") > session:hangup() > } > > > *From:* curriegrad2004 > *To:* FreeSWITCH Users Help > *Sent:* Wednesday, November 30, 2011 11:15 PM > *Subject:* Re: [Freeswitch-users] Question Answertime variable? > > billseconds or bill_msec would be the one to turn to. > > On Wed, Nov 30, 2011 at 4:22 AM, Thomas Hoellriegel > wrote: > > Hi all, > > Is there a possibility to determine the caller time for a channel at > > the end of a call? > > > > For example: in asterisk exists a variable: > > ${ANSWEREDTIME} > > I like to store the Answertime in a Database for example: > > Dailly minutes to call a Cellphone. > > Can your help plese? > > 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111216/69e01f59/attachment.html From msc at freeswitch.org Fri Dec 16 19:28:24 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 16 Dec 2011 08:28:24 -0800 Subject: [Freeswitch-users] freeswitch logic before dialplan In-Reply-To: <4EEB05BE.6080008@softnet.si> References: <4EEB05BE.6080008@softnet.si> Message-ID: Could you explain a little more about what you are trying to do? -MC On Fri, Dec 16, 2011 at 12:47 AM, Miha Zoubek wrote: > Hi, > > how to disable one variable before it hit dialplan for incoming calls? > > 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/20111216/fc025a2a/attachment-0001.html From msc at freeswitch.org Fri Dec 16 19:32:08 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 16 Dec 2011 08:32:08 -0800 Subject: [Freeswitch-users] The system cannot create any sessions at this time In-Reply-To: References: Message-ID: You may have accidentally turned on the 'fsctl pause' function. (I forget which f-key used to be bound to that...) Try 'fsctl resume' and see if that clears it up. -MC On Thu, Dec 15, 2011 at 11:03 AM, Stephen Wilde wrote: > Hi, > what are the possible cause of the critical errors: > > 2011-12-15 19:14:37.538948 [CRIT] mod_sofia.c:4189 Error Creating Session > 2011-12-15 19:14:37.538948 [CRIT] switch_core_session.c:1775 The system > cannot create any sessions at this time. > 2011-12-15 19:14:37.538948 [CRIT] mod_sofia.c:4189 Error Creating Session > 2011-12-15 19:14:37.548945 [CRIT] switch_core_session.c:1775 The system > cannot create any sessions at this time. > 2011-12-15 19:14:37.548945 [CRIT] mod_sofia.c:4189 Error Creating Session > 2011-12-15 19:14:37.548945 [CRIT] switch_core_session.c:1775 The system > cannot create any sessions at this time. > 2011-12-15 19:14:37.548945 [CRIT] mod_sofia.c:4189 Error Creating Session > > and so on. > > This kind of errors appear in a specific time interval (for example from > 19:14:31 until 19:14:39) and after this time interval all continue to work > normally. > > This is not a "max_sessions" related problem because in this case appear a > "Over Session Limit!" message. > > I think also that this is not a "min idle" problem because I have not set > this parameter, my status is: > > UP 0 years, 0 days, 15 hours, 26 minutes, 32 seconds, 182 milliseconds, > 131 microseconds > FreeSWITCH is ready > 4279392 session(s) since startup > 3463 session(s) 30/100 > 6000 session(s) max > min idle cpu 0.00/77.00 > > What are the possible causes of this message? > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111216/4bf2e37e/attachment.html From msc at freeswitch.org Fri Dec 16 19:37:07 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 16 Dec 2011 08:37:07 -0800 Subject: [Freeswitch-users] freeswitch.email () failing In-Reply-To: <1014B6A210DF46F3B00BF7F6A51B31C6@DREWPC> References: <1014B6A210DF46F3B00BF7F6A51B31C6@DREWPC> Message-ID: Wow. 1.0.4 is *really* old. It looks like the email method was not added until later. My guess is that it would be less work to get your "various programs running on top of FS" to work with the latest git than it would be to try to work around this issue. I'd say your best and fastest path to a blissful FreeSWITCH install is to get professional assistance from consulting at freeswitch.org. -MC On Thu, Dec 15, 2011 at 4:57 PM, Drew Terenzini < drew_terenzini at wesleycloversolutions.com> wrote: > ** > > ** ** > > Good evening, I?m attempting to use the freeswitch.email from > http://wiki.freeswitch.org/wiki/Lua#freeswitch.email. I am running FS > 1.04 on Ubuntu 10.04 LTS, and for various reasons having to do with > programs running on top of FS, I cannot just upgrade FS without breaking > other things.**** > > ** ** > > I have installed postfix successfully and tested sendmail from the Linux > CLI without a problem. Emails are delivered to the appropriate external > addresses. My switch.conf.xml has:**** > > ->**** > > **** > > **** > > ** ** > > ? and I have a stripped down test script called ?email.lua? in > my scripts directory as follows:**** > > ** ** > > freeswitch.email("","< > source at freeswitch.source.com>","subject: Test message from > VTAPENG\n","Hello you have a test message.?)**** > > ** ** > > The destination and source email addresses are valid and > receiving email when sent from other programs/sites.**** > > ** ** > > In the FS_CLI, I attempt to test-run the script by entering > ?lua email.lua?, only to get the following error each time:**** > > ** ** > > -ERR encounterd**** > > ** ** > > freeswitch at vtap> **2011-12-15 16**:36:**19.246622** [ERR] mod_lua.cpp:182 > /opt/freeswitch/scripts/email.lua:1: attempt to call field 'email' (a nil > value)**** > > stack traceback:**** > > /opt/freeswitch/scripts/email.lua:1: in main chunk**** > > ** ** > > This error appears to terminate the script, and no response > is received from postfix, and no error is logged to mail.log. I?ve tried a > number of ways to modify the outcome, without real result. Searching on > the Wiki and the mailing lists hasn?t resulted in anything that has worked > either.**** > > ** ** > > Is freeswitch.email() not supported in FS 1.04? Is there some > context that I?m missing for this to work? Any help would be appreciated, > 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/20111216/023f1e09/attachment.html From wstephen80 at gmail.com Fri Dec 16 19:47:26 2011 From: wstephen80 at gmail.com (Stephen Wilde) Date: Fri, 16 Dec 2011 17:47:26 +0100 Subject: [Freeswitch-users] The system cannot create any sessions at this time In-Reply-To: References: Message-ID: Thank you for your reply. I know the effect of pause/resume function but I have seen these messages in the log and not when I have connected to Freeswitch with fs_cli so I'm sure that the problem is not this. On Fri, Dec 16, 2011 at 5:32 PM, Michael Collins wrote: > You may have accidentally turned on the 'fsctl pause' function. (I forget > which f-key used to be bound to that...) > > Try 'fsctl resume' and see if that clears it up. > -MC > > On Thu, Dec 15, 2011 at 11:03 AM, Stephen Wilde wrote: > >> Hi, >> what are the possible cause of the critical errors: >> >> 2011-12-15 19:14:37.538948 [CRIT] mod_sofia.c:4189 Error Creating Session >> 2011-12-15 19:14:37.538948 [CRIT] switch_core_session.c:1775 The system >> cannot create any sessions at this time. >> 2011-12-15 19:14:37.538948 [CRIT] mod_sofia.c:4189 Error Creating Session >> 2011-12-15 19:14:37.548945 [CRIT] switch_core_session.c:1775 The system >> cannot create any sessions at this time. >> 2011-12-15 19:14:37.548945 [CRIT] mod_sofia.c:4189 Error Creating Session >> 2011-12-15 19:14:37.548945 [CRIT] switch_core_session.c:1775 The system >> cannot create any sessions at this time. >> 2011-12-15 19:14:37.548945 [CRIT] mod_sofia.c:4189 Error Creating Session >> >> and so on. >> >> This kind of errors appear in a specific time interval (for example from >> 19:14:31 until 19:14:39) and after this time interval all continue to work >> normally. >> >> This is not a "max_sessions" related problem because in this case appear >> a "Over Session Limit!" message. >> >> I think also that this is not a "min idle" problem because I have not set >> this parameter, my status is: >> >> UP 0 years, 0 days, 15 hours, 26 minutes, 32 seconds, 182 milliseconds, >> 131 microseconds >> FreeSWITCH is ready >> 4279392 session(s) since startup >> 3463 session(s) 30/100 >> 6000 session(s) max >> min idle cpu 0.00/77.00 >> >> What are the possible causes of this message? >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111216/04604747/attachment-0001.html From msc at freeswitch.org Fri Dec 16 22:11:25 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 16 Dec 2011 11:11:25 -0800 Subject: [Freeswitch-users] The system cannot create any sessions at this time In-Reply-To: References: Message-ID: How about sps (sessions per second)? Perhaps the max sessions isn't being hit but your rate of adding new sessions is a little too fast? Check sessions-per-second param in switch.conf.xml. -MC On Fri, Dec 16, 2011 at 8:47 AM, Stephen Wilde wrote: > Thank you for your reply. > I know the effect of pause/resume function but I have seen these messages > in the log and not when I have connected to Freeswitch with fs_cli so I'm > sure that the problem is not this. > > > On Fri, Dec 16, 2011 at 5:32 PM, Michael Collins wrote: > >> You may have accidentally turned on the 'fsctl pause' function. (I forget >> which f-key used to be bound to that...) >> >> Try 'fsctl resume' and see if that clears it up. >> -MC >> >> On Thu, Dec 15, 2011 at 11:03 AM, Stephen Wilde wrote: >> >>> Hi, >>> what are the possible cause of the critical errors: >>> >>> 2011-12-15 19:14:37.538948 [CRIT] mod_sofia.c:4189 Error Creating Session >>> 2011-12-15 19:14:37.538948 [CRIT] switch_core_session.c:1775 The system >>> cannot create any sessions at this time. >>> 2011-12-15 19:14:37.538948 [CRIT] mod_sofia.c:4189 Error Creating Session >>> 2011-12-15 19:14:37.548945 [CRIT] switch_core_session.c:1775 The system >>> cannot create any sessions at this time. >>> 2011-12-15 19:14:37.548945 [CRIT] mod_sofia.c:4189 Error Creating Session >>> 2011-12-15 19:14:37.548945 [CRIT] switch_core_session.c:1775 The system >>> cannot create any sessions at this time. >>> 2011-12-15 19:14:37.548945 [CRIT] mod_sofia.c:4189 Error Creating Session >>> >>> and so on. >>> >>> This kind of errors appear in a specific time interval (for example from >>> 19:14:31 until 19:14:39) and after this time interval all continue to work >>> normally. >>> >>> This is not a "max_sessions" related problem because in this case appear >>> a "Over Session Limit!" message. >>> >>> I think also that this is not a "min idle" problem because I have not >>> set this parameter, my status is: >>> >>> UP 0 years, 0 days, 15 hours, 26 minutes, 32 seconds, 182 milliseconds, >>> 131 microseconds >>> FreeSWITCH is ready >>> 4279392 session(s) since startup >>> 3463 session(s) 30/100 >>> 6000 session(s) max >>> min idle cpu 0.00/77.00 >>> >>> What are the possible causes of this message? >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111216/079d4fea/attachment.html From notlikeme75 at yahoo.com Fri Dec 16 22:51:30 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Fri, 16 Dec 2011 11:51:30 -0800 (PST) Subject: [Freeswitch-users] pagd and dynamic conferences Message-ID: <1324065090.5431.YahooMailNeo@web65310.mail.ac2.yahoo.com> I finally got pagd working for a dynamic conference using transfer from IVR but i want to know if there is a way to do this from within another static conference.: I use? ?? from within a static conference, it transfers me to the proper conference but the caller looses control and has to hangup to call back to get another option.? I want them to have the same caller-controls and be able to hangup back to ivr just like the static conferences work. if i have to use a separate caller control group that is fine, but as it stands this only works from main ivr, it is very important for me to have a direct transfer to a dynamic conference pagd from within a static. does this have something do with clearing the bindings to get new ones? i tried clear digit action but don't think it works for the hard coded caller controls. maybe i am using the wrong method to transfer to the extension 759 from within conference. all help is appreciated. ?condition??? ? destination_number??? ? ^759$??? ? 1??? ? ? action??? ? answer??? ? ??? ? 2??? ? ? action??? ? clear_digit_action??? ? conf??? ? 3??? ? ? action??? ? set??? ? conference_user_list=|??? ? 11??? ? ? action??? ? play_and_get_digits??? ? 4 4 3 5000 # conf-pin.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+??? ? 15??? ? ? action??? ? phrase??? ? spell,${target_num}??? ? 16??? ? ? action??? ? conference??? ? ${target_num}-127.0.0.1 at default -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111216/9fd39189/attachment.html From acrow at integrafin.co.uk Fri Dec 16 23:24:00 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Fri, 16 Dec 2011 20:24:00 +0000 Subject: [Freeswitch-users] {Disarmed} Re: Lua - origination to local endpoint then bridge to either local or remote destination: no audio In-Reply-To: <4EEAF92A.504@integrafin.co.uk> References: <4EE905F8.6010908@integrafin.co.uk> <4EE9A3C5.3030407@integrafin.co.uk> <4EEA3232.90807@integrafin.co.uk> <4EEAF92A.504@integrafin.co.uk> Message-ID: <4EEBA8E0.7020703@integrafin.co.uk> On 16/12/11 07:54, Alex Crow wrote: > Michael, > > I did find out some more last night while trying to reproduce the > issues using a simple "originate" (which I could), and discovered it > was down to FS detecting some of my phones (on other internal network > segments) as "UDP_NAT" even when they were not NATed. I changed my > internal profile to comment out "apply-nat-acl" and then all my local > LAN phones now were registered simply as "UDP". > > Now more situations work, however there is still at least one weird > one. I have my mobile registered to a created "doublenat" profile (my > FS box is behind NAT) following the doublenat example on the wiki. > > I also use the "external" profile to register to a provider. > > I can originate from this extension to a LAN extension fine, so NAT in > this situation is working well. However originating from a LAN > extension to a number via the provider above still gives silence at > both ends. Strangely if I place the same call from the phone itself it > works fine. > > I will keep working on this but do you have any ideas? > > Thanks > > Alex I think I've found some more out - the remaining problems only happen on SNOM phones where G726-32 is enabled. It appears that FS cannot find a match for that codec when AAL is set on the phones: the Snoms ask for dynamic codec G726-32:99:8000:20:0, FS wants to see AAL2-G726-32:122:8000:20:32000, FS does not match that and requests G722, and G722 is negotiated fine, but the audio breaks. If AAL isn't set, and FS has G726-32 enabled, FS sends back dynamic codec 100 and that doesn't match anything in the vars.xml description. Again no audio. If I remove G276-32 from the list on the Snoms all works perfectly again. I find it interesting that the default vars.xml states that AAL2-G726-32 is dynamic number 122 but there is nothing listed for just g726-32, and even if I set the snoms to use AAL then they still request G726-32, not AAL2-G726-32. Could that be part of the issue with this codec? I didn't have to worry with the Polycoms as they dont offer any G726 codec at all. In the interim I will just disable G726 on the Snom endpoints. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111216/aba2f4be/attachment.html From admin at blindi.net Sat Dec 17 00:10:50 2011 From: admin at blindi.net (Thomas Hoellriegel) Date: Fri, 16 Dec 2011 22:10:50 +0100 (CET) Subject: [Freeswitch-users] Answerconfirmation with ivr not working In-Reply-To: References: Message-ID: Hi Michael Am 08.12.11 um 12:08 schrieb Michael Collins: > So you're trying to have your group confirm file be an ivr? Is that even > possible? If so, it's news to me. Where did you hear that this was possible? Exactly. This is my callscreening extension: My Ivrmenu: The problem is the same: A person call my phonenumber in germany: 08925007676 The phone is rining. I answer the call, and heare the prompt from the ivrmenu. I press 1 from my phone to transfer to the conference (aleg and bleg). I heare the voiceprompt in the conference, but the caller does not transferd. The line rinings anytime. for example: i press 2, i heare the demo_ivr prompt, but the caller is no transfer to the ivr, and the phone rings. can you look what is wrong please? you can test my dialplan. 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 anthony.minessale at gmail.com Sat Dec 17 01:10:24 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 16 Dec 2011 16:10:24 -0600 Subject: [Freeswitch-users] Stable Branch Maintainer Wanted [Employment Opportunity] Message-ID: Hello, We have been fortunate enough to be in a circumstance where we have the proper funding from a generous supporter to advance an energetic and logistically gifted individual from the ranks of our community to become our very own Stable Branch Maintainer. This is a challenging job helping to put together a stable release and other community maintenance tasks. Anyone who is interested, please send references and other relevant info to jobs at freeswitch.org Thanks, -- 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/20111216/136756d9/attachment.html From anthony.minessale at gmail.com Sat Dec 17 01:16:34 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Fri, 16 Dec 2011 16:16:34 -0600 Subject: [Freeswitch-users] Coders Wanted Perl/C/JS [Employment Opportunity] Message-ID: Barracuda Networks is hiring developers right now for the CudaTEL product. As you may already know, CudaTEL is the FreeSWITCH-powered PBX appliance designed by the core FreeSWITCH team. We are looking for developers for a few positions with some combo of the following skills: Programmer in the following languages: C Perl Javascript/Jquery Database skills: Postgres We need someone who lives in the United States and preferably someone already in or willing to be located in MI CA WI OK TX Please respond with CV/Resume to jobs at 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/20111216/23b41208/attachment.html From jmesquita at freeswitch.org Sat Dec 17 01:40:50 2011 From: jmesquita at freeswitch.org (=?ISO-8859-1?Q?Jo=E3o_Mesquita?=) Date: Fri, 16 Dec 2011 19:40:50 -0300 Subject: [Freeswitch-users] Stable Branch Maintainer Wanted [Employment Opportunity] In-Reply-To: References: Message-ID: Congratulations Tony and team. Another battle won! And thanks for the fellow contributor. Regards, Jo?o Mesquita On Fri, Dec 16, 2011 at 7:10 PM, Anthony Minessale < anthony.minessale at gmail.com> wrote: > Hello, > > We have been fortunate enough to be in a circumstance where we have the > proper funding from a generous supporter to advance an energetic > and logistically gifted individual from the ranks of our community to > become our very own Stable Branch Maintainer. This is a challenging job > helping to put together a stable release and other > community maintenance tasks. > > Anyone who is interested, please send references and other relevant info > to jobs at freeswitch.org > > Thanks, > > -- > 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/20111216/f51dd084/attachment.html From peter at spinato.ca Fri Dec 16 23:46:16 2011 From: peter at spinato.ca (Peter Spinato) Date: Fri, 16 Dec 2011 15:46:16 -0500 Subject: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue Message-ID: <01d301ccbc33$c2d7d360$48877a20$@ca> All, Hopefully someone can assist me - I'll gladly give $50 to the person who helps me fix the issue - I have an OpenSIPs server configured as a load balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) that receives the call and forwards it to my Freeswitch Server for an IVR. When I had the call just routing to the Freeswitch server I got the audio working by setting the ext-rtp-ip to the public IP. Now that I route the SIP call through the OpenSIPs server there is no audio - I'm guessing it a NAT issue as always. Both the OpenSIPS and FreeSwitch server have an internal private IP - OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2 Both servers also have a public IP that routes to the internal IP OpensIPS = 47.1.1.1 and Freeswitch = 47.1.1.2 Call gets received by the OpenSIPs via the external IP - routes the call to the internal IP on the Freeswitch server which answers the call - but no audio - I think the FreeSwitch is trying to route the RTP Audio via its internal private IP instead of the public IP of 47.1.1.2. Not sure if this is the real issue or how to configure it route RTP properly ... all help is appreciated. $50 Paypal to whoever fixes this for me! Thanks -Peter -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111216/cb42f9d3/attachment.html From gabe at gundy.org Sat Dec 17 05:51:26 2011 From: gabe at gundy.org (Gabriel Gunderson) Date: Fri, 16 Dec 2011 19:51:26 -0700 Subject: [Freeswitch-users] Stable Branch Maintainer Wanted [Employment Opportunity] In-Reply-To: References: Message-ID: On Fri, Dec 16, 2011 at 3:10 PM, Anthony Minessale wrote: > We have been fortunate enough to be in a circumstance where we have the > proper funding from a generous supporter to advance an energetic > and?logistically?gifted individual from the ranks of our community to become > our very own Stable Branch Maintainer. ?This is a challenging job helping to > put together a stable release and other community?maintenance?tasks. This is very exciting news. I see it as a sign of FreeSWITCH's continued growth and increasing importance in the marketplace. Props to the "generous supporter" and best of luck filling the position! Gabe From ndavis at inetwork.com Sat Dec 17 03:08:37 2011 From: ndavis at inetwork.com (Neil Davis) Date: Fri, 16 Dec 2011 17:08:37 -0700 Subject: [Freeswitch-users] Threads remain after calling close on Java client Message-ID: <8b41de351c0d1365e3786e7a60645275@mail.gmail.com> Hi, I built a web application that connects to Freeswitch using the org.freeswitch.esl.client.Client. I connect the Client object from a Spring annotated service that I call from a Spring controller. I put the connected client in my ServletContext, so I can access it later to call client.cancelEventSubscriptions() and client.close() from my ServletContextListener contextDestroyed method when Tomcat is shutting down. The problem I'm having is that even after I call close on the client, there are still a bunch of active threads that the client has spawned in the background. These threads are causing Tomcat to hang when I'm shutting down. Can anyone suggest an approach that would enable my application to disconnect the Freeswitch client when Tomcat is shutting down that would allow Tomcat to shutdown gracefully? Below are errors from my Tomcat log for the threads that I have identified as being related to the Freeswitch client. I don't know how I can get to these threads to interrupt them and Client.close() seems to leave them hanging. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-3-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-4-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-2] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-3] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-4] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-5] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-6] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-7] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-8] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-9] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-10] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-11] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-12] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-13] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-14] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-15] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-16] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.internal.ThreadLocalBoolean] (value [org.jboss.netty.util.internal.ThreadLocalBoolean at 186e192]) and a value of type [java.lang.Boolean] (value [false]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.CharsetUtil$1] (value [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type [java.util.IdentityHashMap] (value [{windows-1252=sun.nio.cs.MS1252$Encoder at 373f86}]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.internal.ThreadLocalRandom$1] (value [org.jboss.netty.util.internal.ThreadLocalRandom$1 at 12bb519]) and a value of type [org.jboss.netty.util.internal.ThreadLocalRandom] (value [org.jboss.netty.util.internal.ThreadLocalRandom at 7e9dbc]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.CharsetUtil$1] (value [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type [java.util.IdentityHashMap] (value [{windows-1252=sun.nio.cs.MS1252$Encoder at a5b041}]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Thanks, Neil Davis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111216/97f9a00c/attachment-0001.html From avi at avimarcus.net Sat Dec 17 19:06:55 2011 From: avi at avimarcus.net (Avi Marcus) Date: Sat, 17 Dec 2011 18:06:55 +0200 Subject: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue In-Reply-To: <01d301ccbc33$c2d7d360$48877a20$@ca> References: <01d301ccbc33$c2d7d360$48877a20$@ca> Message-ID: Someone familiar with this might be able to answer right off the bat, but if you pastebin a SIP trace (and an fs_cli log for completeness) , the problem should become apparent. Are you using the internal profile for all the calls? If you use the external.xml profile, you might need to set your ext-rtp-ip in that file, too. Seeing a trace will tell (almost) the whole story. -Avi On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato wrote: > All,**** > > Hopefully someone can assist me - I'll gladly give $50 to the person who > helps me fix the issue - I have an OpenSIPs server configured as a load > balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) > that receives the call and forwards it to my Freeswitch Server for an IVR. > When I had the call just routing to the Freeswitch server I got the audio > working by setting the ext-rtp-ip to the public IP. Now that I route the > SIP call through the OpenSIPs server there is no audio - I'm guessing it a > NAT issue as always.**** > > ** ** > > Both the OpenSIPS and FreeSwitch server have an internal private IP - > OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2**** > > Both servers also have a public IP that routes to the internal IP OpensIPS > = 47.1.1.1 and Freeswitch = 47.1.1.2**** > > ** ** > > Call gets received by the OpenSIPs via the external IP - routes the call > to the internal IP on the Freeswitch server which answers the call - but no > audio - I think the FreeSwitch is trying to route the RTP Audio via its > internal private IP instead of the public IP of 47.1.1.2. Not sure if this > is the real issue or how to configure it route RTP properly ... all help > is appreciated. $50 Paypal to whoever fixes this for me! Thanks**** > > ** ** > > -Peter**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111217/d55b7836/attachment.html From manieq at wp.eu Sat Dec 17 22:55:04 2011 From: manieq at wp.eu (Mariusz Czulada) Date: Sat, 17 Dec 2011 20:55:04 +0100 Subject: [Freeswitch-users] Transfer a call to other node Message-ID: <4eecf3984e0931.53893715@wp.pl> Hi all, Looking at Tech-Invite site I found such flow example: http://www.tech-invite.com/Ti-sip-service-06.html I wonder if it's (currently) possible to implement a replacing INVITE using FreeSWITCH (for Bob and Carol sides). Regards, Mariusz From wstephen80 at gmail.com Sat Dec 17 23:25:18 2011 From: wstephen80 at gmail.com (Stephen Wilde) Date: Sat, 17 Dec 2011 21:25:18 +0100 Subject: [Freeswitch-users] The system cannot create any sessions at this time In-Reply-To: References: Message-ID: But in this case I think that should appear a "Throttle Error" in the log (in the past I saw this message and I changed my limits). Or this message has a ERR level and I don't see it because I have the log filter at CRIT level? On Fri, Dec 16, 2011 at 8:11 PM, Michael Collins wrote: > How about sps (sessions per second)? Perhaps the max sessions isn't being > hit but your rate of adding new sessions is a little too fast? Check > sessions-per-second param in switch.conf.xml. > > -MC > > > On Fri, Dec 16, 2011 at 8:47 AM, Stephen Wilde wrote: > >> Thank you for your reply. >> I know the effect of pause/resume function but I have seen these messages >> in the log and not when I have connected to Freeswitch with fs_cli so I'm >> sure that the problem is not this. >> >> >> On Fri, Dec 16, 2011 at 5:32 PM, Michael Collins wrote: >> >>> You may have accidentally turned on the 'fsctl pause' function. (I >>> forget which f-key used to be bound to that...) >>> >>> Try 'fsctl resume' and see if that clears it up. >>> -MC >>> >>> On Thu, Dec 15, 2011 at 11:03 AM, Stephen Wilde wrote: >>> >>>> Hi, >>>> what are the possible cause of the critical errors: >>>> >>>> 2011-12-15 19:14:37.538948 [CRIT] mod_sofia.c:4189 Error Creating >>>> Session >>>> 2011-12-15 19:14:37.538948 [CRIT] switch_core_session.c:1775 The system >>>> cannot create any sessions at this time. >>>> 2011-12-15 19:14:37.538948 [CRIT] mod_sofia.c:4189 Error Creating >>>> Session >>>> 2011-12-15 19:14:37.548945 [CRIT] switch_core_session.c:1775 The system >>>> cannot create any sessions at this time. >>>> 2011-12-15 19:14:37.548945 [CRIT] mod_sofia.c:4189 Error Creating >>>> Session >>>> 2011-12-15 19:14:37.548945 [CRIT] switch_core_session.c:1775 The system >>>> cannot create any sessions at this time. >>>> 2011-12-15 19:14:37.548945 [CRIT] mod_sofia.c:4189 Error Creating >>>> Session >>>> >>>> and so on. >>>> >>>> This kind of errors appear in a specific time interval (for example >>>> from 19:14:31 until 19:14:39) and after this time interval all continue to >>>> work normally. >>>> >>>> This is not a "max_sessions" related problem because in this case >>>> appear a "Over Session Limit!" message. >>>> >>>> I think also that this is not a "min idle" problem because I have not >>>> set this parameter, my status is: >>>> >>>> UP 0 years, 0 days, 15 hours, 26 minutes, 32 seconds, 182 milliseconds, >>>> 131 microseconds >>>> FreeSWITCH is ready >>>> 4279392 session(s) since startup >>>> 3463 session(s) 30/100 >>>> 6000 session(s) max >>>> min idle cpu 0.00/77.00 >>>> >>>> What are the possible causes of this message? >>>> >>>> >>>> >>>> _________________________________________________________________________ >>>> Professional FreeSWITCH Consulting Services: >>>> consulting at freeswitch.org >>>> http://www.freeswitchsolutions.com >>>> >>>> >>>> >>>> >>>> Official FreeSWITCH Sites >>>> http://www.freeswitch.org >>>> http://wiki.freeswitch.org >>>> http://www.cluecon.com >>>> >>>> FreeSWITCH-users mailing list >>>> FreeSWITCH-users at lists.freeswitch.org >>>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>>> UNSUBSCRIBE: >>>> http://lists.freeswitch.org/mailman/options/freeswitch-users >>>> http://www.freeswitch.org >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111217/c8347d30/attachment.html From sdame at 207me.com Sun Dec 18 01:50:50 2011 From: sdame at 207me.com (Stephen Dame) Date: Sat, 17 Dec 2011 17:50:50 -0500 Subject: [Freeswitch-users] Weekly Conference Torrents. In-Reply-To: References: Message-ID: <01ac01ccbd0e$53a8e070$fafaa150$@com> Question. I have downloaded the last 4 weekly confrerecne calls from torrents, but all the files play in slow motion sound like lion roaring Tried all the different formats, in different players and they all are not understandable. Has anyone successfully listened to them. My normal freeswitch recordings in mp3, wav play fine in vlc, itunes, media player on win7. All three formats are sounding the same ogg, wav, mp3 for all 4 weeks? Thanks Stephen From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Michael Collins Sent: Thursday, December 15, 2011 11:35 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] FreeSWITCH Conference Call Today Correct. We decided to use torrents to consolidate our media files and use our CDN for stuff like this. -MC On Wed, Dec 14, 2011 at 9:46 PM, William Suffill wrote: No more direct download links for the recordings? -- W _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111217/d9541fa8/attachment-0001.html From peter at spinato.ca Sun Dec 18 06:07:42 2011 From: peter at spinato.ca (Peter Spinato) Date: Sat, 17 Dec 2011 22:07:42 -0500 Subject: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue In-Reply-To: References: <01d301ccbc33$c2d7d360$48877a20$@ca> Message-ID: <02ac01ccbd32$36b4ba10$a41e2e30$@ca> The calls go to internal profile - I guess case the OpenSIPs server is local lan - is there a way to force that calls to external profile as I find calls that hit that profile load the ext-rtp-ip ip and work. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Saturday, December 17, 2011 11:07 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue Someone familiar with this might be able to answer right off the bat, but if you pastebin a SIP trace (and an fs_cli log for completeness) , the problem should become apparent. Are you using the internal profile for all the calls? If you use the external.xml profile, you might need to set your ext-rtp-ip in that file, too. Seeing a trace will tell (almost) the whole story. -Avi On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato wrote: All, Hopefully someone can assist me - I'll gladly give $50 to the person who helps me fix the issue - I have an OpenSIPs server configured as a load balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) that receives the call and forwards it to my Freeswitch Server for an IVR. When I had the call just routing to the Freeswitch server I got the audio working by setting the ext-rtp-ip to the public IP. Now that I route the SIP call through the OpenSIPs server there is no audio - I'm guessing it a NAT issue as always. Both the OpenSIPS and FreeSwitch server have an internal private IP - OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2 Both servers also have a public IP that routes to the internal IP OpensIPS = 47.1.1.1 and Freeswitch = 47.1.1.2 Call gets received by the OpenSIPs via the external IP - routes the call to the internal IP on the Freeswitch server which answers the call - but no audio - I think the FreeSwitch is trying to route the RTP Audio via its internal private IP instead of the public IP of 47.1.1.2. Not sure if this is the real issue or how to configure it route RTP properly ... all help is appreciated. $50 Paypal to whoever fixes this for me! Thanks -Peter _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111217/aa26d3d0/attachment.html From avi at avimarcus.net Sun Dec 18 06:31:49 2011 From: avi at avimarcus.net (Avi Marcus) Date: Sun, 18 Dec 2011 05:31:49 +0200 Subject: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue In-Reply-To: <02ac01ccbd32$36b4ba10$a41e2e30$@ca> References: <01d301ccbc33$c2d7d360$48877a20$@ca> <02ac01ccbd32$36b4ba10$a41e2e30$@ca> Message-ID: Yes, send it to the freeswitch server port 5080 Or if you don't need the internal profile, remove it and set external to use port 5060. -Avi On Sun, Dec 18, 2011 at 5:07 AM, Peter Spinato wrote: > The calls go to internal profile ? I guess case the OpenSIPs server is > local lan ? is there a way to force that calls to external profile as I > find calls that hit that profile load the ext-rtp-ip ip and work.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Saturday, December 17, 2011 11:07 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch > issue**** > > ** ** > > Someone familiar with this might be able to answer right off the bat, but > if you pastebin a SIP trace (and an fs_cli log for completeness) , the > problem should become apparent.**** > > ** ** > > Are you using the internal profile for all the calls? If you use the > external.xml profile, you might need to set your ext-rtp-ip in that file, > too. Seeing a trace will tell (almost) the whole story.**** > > > **** > > -Avi**** > > ** ** > > On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato wrote:* > *** > > All,**** > > Hopefully someone can assist me - I'll gladly give $50 to the person who > helps me fix the issue - I have an OpenSIPs server configured as a load > balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) > that receives the call and forwards it to my Freeswitch Server for an IVR.. > When I had the call just routing to the Freeswitch server I got the audio > working by setting the ext-rtp-ip to the public IP. Now that I route the > SIP call through the OpenSIPs server there is no audio - I'm guessing it a > NAT issue as always.**** > > **** > > Both the OpenSIPS and FreeSwitch server have an internal private IP - > OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2**** > > Both servers also have a public IP that routes to the internal IP OpensIPS > = 47.1.1.1 and Freeswitch = 47.1.1.2**** > > **** > > Call gets received by the OpenSIPs via the external IP - routes the call > to the internal IP on the Freeswitch server which answers the call - but no > audio - I think the FreeSwitch is trying to route the RTP Audio via its > internal private IP instead of the public IP of 47.1.1.2. Not sure if this > is the real issue or how to configure it route RTP properly ... all help > is appreciated. $50 Paypal to whoever fixes this for me! Thanks**** > > **** > > -Peter**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11* > *** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111218/eb095376/attachment.html From jeroeneeuwes at gmail.com Sun Dec 18 09:35:51 2011 From: jeroeneeuwes at gmail.com (Jeroen Eeuwes) Date: Sun, 18 Dec 2011 07:35:51 +0100 Subject: [Freeswitch-users] Weekly Conference Torrents. In-Reply-To: <01ac01ccbd0e$53a8e070$fafaa150$@com> References: <01ac01ccbd0e$53a8e070$fafaa150$@com> Message-ID: Hi Stephen, > Question? I have downloaded the last 4 weekly confrerecne calls from > torrents, but all the files play in slow motion sound like lion roaring I never downloaded it before, but it sounds the same here. If you speed up the file to 300% it sounds OK. Under linux I used this to play the file. For example: mplayer -speed 3 conf_call_2011-12-14.mp3 You can also use sox to get a file which sounds better. For example: sox conf_call_2011-12-14.ogg soundsbetter.ogg speed 3 Beste regards, Jeroen Eeuwes From gmaruzz at gmail.com Sun Dec 18 12:35:35 2011 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Sun, 18 Dec 2011 10:35:35 +0100 Subject: [Freeswitch-users] Weekly Conference Torrents. In-Reply-To: References: <01ac01ccbd0e$53a8e070$fafaa150$@com> Message-ID: maybe the recordings are at 48khz and you're playing at 16khz? -giovanni On 12/18/11, Jeroen Eeuwes wrote: > Hi Stephen, > >> Question? I have downloaded the last 4 weekly confrerecne calls from >> torrents, but all the files play in slow motion sound like lion roaring > > I never downloaded it before, but it sounds the same here. > > If you speed up the file to 300% it sounds OK. Under linux I used this > to play the file. For example: > > mplayer -speed 3 conf_call_2011-12-14.mp3 > > You can also use sox to get a file which sounds better. For example: > > sox conf_call_2011-12-14.ogg soundsbetter.ogg speed 3 > > Beste regards, > Jeroen Eeuwes > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 From acrow at integrafin.co.uk Sun Dec 18 13:08:50 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Sun, 18 Dec 2011 10:08:50 +0000 Subject: [Freeswitch-users] Issue adding a SIP Gateway In-Reply-To: <44E5C0A9D48A3246966A4AE04692014D102A803C@CH1PRD0604MB109.namprd06.prod.outlook.com> References: <44E5C0A9D48A3246966A4AE04692014D102A6FB5@CH1PRD0604MB109.namprd06.prod.outlook.com> <201112151954.01880.justlikeef@gmail.com> <44E5C0A9D48A3246966A4AE04692014D102A803C@CH1PRD0604MB109.namprd06.prod.outlook.com> Message-ID: <4EEDBBB2.50302@integrafin.co.uk> On 16/12/11 01:05, Ryan Watkins wrote: > > I did run the" sofia profile external restart reloadxml" command.... > It didn't load the new gateway, so that's why I tried registering the > gateway specifically. > > * *Ryan,* *Did you follow this example: http://wiki.freeswitch.org/wiki/Provider_Configuration:_iptel and replace the usename and password with your own? Check that the permissions on the new XML file allow it to be read by the user freeswitch is running as. Also double-check your closing tags on the file. This can cause your gateway to be skipped, hence the "invalid gateway" when you try to use it. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111218/1f7018cb/attachment.html From xing2kin at yahoo.com Sun Dec 18 14:23:34 2011 From: xing2kin at yahoo.com (king2kin) Date: Sun, 18 Dec 2011 03:23:34 -0800 (PST) Subject: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) Message-ID: <1324207414.49834.YahooMailNeo@web39703.mail.mud.yahoo.com> Hi there, ? On Dec. 18, 2011, I dowloaded the latest GIT version of freeswitch source codes by ? git clone git://git.freeswitch.org/freeswitch.git ? Then, I compiled the project solution with vs2008, and run FreeSwitch.exe from command-line on windows 2003. I got a bunch of errors complaining about SQL errors, for example, { switch_core_sqldb.c:890 SQL ERR [no such table: sip_registrations] [delete from sip_registrations where (contact like '%TCP%' or status like '%TCP%' or status like '%TLS%') and hostname='W2k3T602' and network_ip like '%' and network_port like '%' and sip_username like '%' and mwi_user? like '%' and mwi_host like '%' and orig_server_host like '%' and orig_hostname like '%'] } ? Unlike old version, the new version on windows system seems to require setting up extra database tables according to log file [freeswitch.log].The log file?gave basic definitions (see below) of these SQL tables, however, primary key and/or constraints are NOT provided. ? Could anyone tell me where to find the full definition of these SQL tables that are required by the latest GIT version of freeswitch?? Thanks in adavance! ? ==== table definitions gave in freeswitch.log ========================= ? -- -- table [sip_registrations] -- CREATE TABLE sip_registrations ( ?? call_id????????? VARCHAR(255), ?? sip_user???????? VARCHAR(255), ?? sip_host???????? VARCHAR(255), ?? presence_hosts?? VARCHAR(255), ?? contact????????? VARCHAR(1024), ?? status?????????? VARCHAR(255), ?? rpid???????????? VARCHAR(255), ?? expires????????? INTEGER, ?? user_agent?????? VARCHAR(255), ?? server_user????? VARCHAR(255), ?? server_host????? VARCHAR(255), ?? profile_name???? VARCHAR(255), ?? hostname???????? VARCHAR(255), ?? network_ip?????? VARCHAR(255), ?? network_port???? VARCHAR(6), ?? sip_username???? VARCHAR(255), ?? sip_realm??????? VARCHAR(255), ?? mwi_user???????? VARCHAR(255), ?? mwi_host???????? VARCHAR(255), ?? orig_server_host VARCHAR(255), ?? orig_hostname??? VARCHAR(255) ); -- -- table [sip_subscriptions] -- CREATE TABLE sip_subscriptions ( ?? proto?????????? VARCHAR(255), ?? sip_user??????? VARCHAR(255), ?? sip_host??????? VARCHAR(255), ?? sub_to_user???? VARCHAR(255), ?? sub_to_host???? VARCHAR(255), ?? presence_hosts? VARCHAR(255), ?? event?????????? VARCHAR(255), ?? contact???????? VARCHAR(1024), ?? call_id???????? VARCHAR(255), ?? full_from?????? VARCHAR(255), ?? full_via??????? VARCHAR(255), ?? expires???????? INTEGER, ?? user_agent????? VARCHAR(255), ?? accept????????? VARCHAR(255), ?? profile_name??? VARCHAR(255), ?? hostname??????? VARCHAR(255), ?? network_port??? VARCHAR(6), ?? network_ip????? VARCHAR(255), ?? version???????? INTEGER DEFAULT 0 NOT NULL, ?? orig_proto????? VARCHAR(255), ?? full_to???????? VARCHAR(255) ); -- -- table [sip_dialogs] -- CREATE TABLE sip_dialogs ( ?? call_id???????? VARCHAR(255), ?? uuid??????????? VARCHAR(255), ?? sip_to_user???? VARCHAR(255), ?? sip_to_host???? VARCHAR(255), ?? sip_from_user?? VARCHAR(255), ?? sip_from_host?? VARCHAR(255), ?? contact_user??? VARCHAR(255), ?? contact_host??? VARCHAR(255), ?? state?????????? VARCHAR(255), ?? direction?????? VARCHAR(255), ?? user_agent????? VARCHAR(255), ?? profile_name??? VARCHAR(255), ?? hostname??????? VARCHAR(255), ?? contact???????? VARCHAR(255), ?? presence_id???? VARCHAR(255), ?? presence_data?? VARCHAR(255), ?? call_info?????? VARCHAR(255), ?? call_info_state VARCHAR(255), ?? expires???????? INTEGER default 0, ?? status????????? VARCHAR(255), ?? rpid??????????? VARCHAR(255), ?? sip_to_tag????? VARCHAR(255), ?? sip_from_tag??? VARCHAR(255), ?? rcd???????????? INTEGER not null default 0 ); -- -- table [sip_presence] -- CREATE TABLE sip_presence ( ?? sip_user??????? VARCHAR(255), ?? sip_host??????? VARCHAR(255), ?? status????????? VARCHAR(255), ?? rpid??????????? VARCHAR(255), ?? expires???????? INTEGER, ?? user_agent????? VARCHAR(255), ?? profile_name??? VARCHAR(255), ?? hostname??????? VARCHAR(255), ?? network_ip????? VARCHAR(255), ?? network_port??? VARCHAR(6), ?? open_closed???? VARCHAR(255) ); -- -- table [sip_authentication] -- CREATE TABLE sip_authentication ( ?? nonce?????????? VARCHAR(255), ?? expires???????? INTEGER,?? profile_name??? VARCHAR(255), ?? hostname??????? VARCHAR(255), ?? last_nc???????? INTEGER ); -- -- table [sip_shared_appearance_subscriptions] -- CREATE TABLE sip_shared_appearance_subscriptions ( ?? subscriber??????? VARCHAR(255), ?? call_id?????????? VARCHAR(255), ?? aor?????????????? VARCHAR(255), ?? profile_name????? VARCHAR(255), ?? hostname????????? VARCHAR(255), ?? contact_str?????? VARCHAR(255), ?? network_ip??????? VARCHAR(255) ); -- -- table [sip_shared_appearance_dialogs] -- CREATE TABLE sip_shared_appearance_dialogs ( ?? profile_name????? VARCHAR(255), ?? hostname????????? VARCHAR(255), ?? contact_str?????? VARCHAR(255), ?? call_id?????????? VARCHAR(255), ?? network_ip??????? VARCHAR(255), ?? expires?????????? INTEGER ); -- -- table [sip_recovery] -- CREATE TABLE sip_recovery ( ?? runtime_uuid??? VARCHAR(255), ?? profile_name??? VARCHAR(255), ?? hostname??????? VARCHAR(255), ?? uuid??????????? VARCHAR(255), ?? metadata??????? text ); -- -- table [fifo_outbound] -- create table fifo_outbound ( ?uuid varchar(255), ?fifo_name varchar(255), ?originate_string varchar(255), ?simo_count integer, ?use_count integer, ?timeout integer, ?lag integer, ?next_avail integer not null default 0, ?expires integer not null default 0, ?static integer not null default 0, ?outbound_call_count integer not null default 0, ?outbound_fail_count integer not null default 0, ?hostname varchar(255), ?taking_calls integer not null default 1, ?status varchar(255), ?outbound_call_total_count integer not null default 0, ?outbound_fail_total_count integer not null default 0, ?active_time integer not null default 0, ?inactive_time integer not null default 0, ?manual_calls_out_count integer not null default 0, ?manual_calls_in_count integer not null default 0, ?manual_calls_out_total_count integer not null default 0, ?manual_calls_in_total_count integer not null default 0, ?ring_count integer not null default 0, ?start_time integer not null default 0, ?stop_time integer not null default 0 ); -- -- table [fifo_bridge] -- create table fifo_bridge ( ?fifo_name varchar(1024) not null, ?caller_uuid varchar(255) not null, ?caller_caller_id_name varchar(255), ?caller_caller_id_number varchar(255), ?consumer_uuid varchar(255) not null, ?consumer_outgoing_uuid varchar(255), ?bridge_start integer ); -- -- table [fifo_callers] -- create table fifo_callers ( ?fifo_name varchar(255) not null, ?uuid varchar(255) not null, ?caller_caller_id_name varchar(255), ?caller_caller_id_number varchar(255), ?timestamp integer ); -- -- table [voicemail_msgs] -- CREATE TABLE voicemail_msgs ( ?? created_epoch INTEGER, ?? read_epoch??? INTEGER, ?? username????? VARCHAR(255), ?? domain??????? VARCHAR(255), ?? uuid????????? VARCHAR(255), ?? cid_name????? VARCHAR(255), ?? cid_number??? VARCHAR(255), ?? in_folder???? VARCHAR(255), ?? file_path???? VARCHAR(255), ?? message_len?? INTEGER, ?? flags???????? VARCHAR(255), ?? read_flags??? VARCHAR(255), ?? forwarded_by? VARCHAR(255) ); -- -- table [voicemail_prefs] -- CREATE TABLE voicemail_prefs ( ?? username??????? VARCHAR(255), ?? domain????????? VARCHAR(255), ?? name_path?????? VARCHAR(255), ?? greeting_path?? VARCHAR(255), ?? password??????? VARCHAR(255) ); -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111218/e6b4f735/attachment-0001.html From sdame at 207me.com Sun Dec 18 16:28:05 2011 From: sdame at 207me.com (Stephen Dame) Date: Sun, 18 Dec 2011 08:28:05 -0500 Subject: [Freeswitch-users] Weekly Conference Torrents. In-Reply-To: References: <01ac01ccbd0e$53a8e070$fafaa150$@com> Message-ID: <008f01ccbd88$e0683770$a138a650$@com> Jeroen,Giovanni Thanks, I tried playing all three wav,ogg,mp3 at 3x in vlc still not understandable. This is on Windows 7, where itunes, media player do the same thing. I'll try playing direct from within freeswitch for now. Stephen -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: Sunday, December 18, 2011 4:36 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Weekly Conference Torrents. maybe the recordings are at 48khz and you're playing at 16khz? -giovanni On 12/18/11, Jeroen Eeuwes wrote: > Hi Stephen, > >> Question. I have downloaded the last 4 weekly confrerecne calls from >> torrents, but all the files play in slow motion sound like lion roaring > > I never downloaded it before, but it sounds the same here. > > If you speed up the file to 300% it sounds OK. Under linux I used this > to play the file. For example: > > mplayer -speed 3 conf_call_2011-12-14.mp3 > > You can also use sox to get a file which sounds better. For example: > > sox conf_call_2011-12-14.ogg soundsbetter.ogg speed 3 > > Beste regards, > Jeroen Eeuwes > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From sdame at 207me.com Sun Dec 18 18:04:30 2011 From: sdame at 207me.com (Stephen Dame) Date: Sun, 18 Dec 2011 10:04:30 -0500 Subject: [Freeswitch-users] Weekly Conference Torrents. In-Reply-To: References: <01ac01ccbd0e$53a8e070$fafaa150$@com> Message-ID: <00a101ccbd96$5876b360$09641a20$@com> Just tried playing them direct in freeswitch use latest git... they sound the same not understandable by default I tried in normal, and cd quality 48,000 conf using conference 3300 play file.mp3 stephen -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeroen Eeuwes Sent: Sunday, December 18, 2011 1:36 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Weekly Conference Torrents. Hi Stephen, > Question. I have downloaded the last 4 weekly confrerecne calls from > torrents, but all the files play in slow motion sound like lion roaring I never downloaded it before, but it sounds the same here. If you speed up the file to 300% it sounds OK. Under linux I used this to play the file. For example: mplayer -speed 3 conf_call_2011-12-14.mp3 You can also use sox to get a file which sounds better. For example: sox conf_call_2011-12-14.ogg soundsbetter.ogg speed 3 Beste regards, Jeroen Eeuwes _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From jeff at jefflenk.com Sun Dec 18 20:30:45 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Sun, 18 Dec 2011 09:30:45 -0800 (PST) Subject: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) In-Reply-To: <1324207414.49834.YahooMailNeo@web39703.mail.mud.yahoo.com> References: <1324207414.49834.YahooMailNeo@web39703.mail.mud.yahoo.com> Message-ID: <1324229445946-7106232.post@n2.nabble.com> All the required tables should be created on first run. Same as always. Is that not happening? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Where-are-table-definitions-of-database-for-freewsitch-latest-GIT-version-tp7105666p7106232.html Sent from the freeswitch-users mailing list archive at Nabble.com. From notlikeme75 at yahoo.com Sun Dec 18 21:22:47 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Sun, 18 Dec 2011 10:22:47 -0800 (PST) Subject: [Freeswitch-users] announce conf count (total callers minus 1)? Message-ID: <1324232567.1676.YahooMailNeo@web65301.mail.ac2.yahoo.com> this works great, thank you to whomever created it. is there a way to make the conf count = list count -1 so the playback will only speak total "other" callers instead of including the person who press the option? conf/dialplan/default/01_Announce_Conf_Count.xml: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111218/a437d869/attachment.html From elliott at zoogmedia.com Mon Dec 19 00:26:32 2011 From: elliott at zoogmedia.com (Elliott Vogel) Date: Sun, 18 Dec 2011 21:26:32 +0000 Subject: [Freeswitch-users] destination_number cleanup Message-ID: I was wondering if anyone has a regex expression that works to return just digest? I have some clients sending requests formatted to +1 (555) 555-5555, 555-555-5555, 555.555.5555 which aren't be processed by our dial plan because we are expecting all numbers (5555555555) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111218/c34d322b/attachment.html From ted at schober.us Sun Dec 18 04:39:37 2011 From: ted at schober.us (Ted Schober) Date: Sun, 18 Dec 2011 01:39:37 +0000 (UTC) Subject: [Freeswitch-users] Grandstream GXP 2000 References: Message-ID: Jonas Gauffin writes: > > Hello > > I have problems with TCP and GXP2000 phones. > They work fine using UDP, but no calls arrive if I switch to TCP. > I've upgraded to the latest GS firmware today to see it that helped, > but it didn't. > > Have anyone else had problems with TCP? > > Regards, > Jonas > > If you touch the account page after setting it up and registering with FS, FS will then deny registration. I suspect the password gets messed up. If you need to change the account settings you need to factory reset and get it all right with one pass. From manieq at wp.eu Mon Dec 19 03:17:29 2011 From: manieq at wp.eu (Mariusz Czulada) Date: Mon, 19 Dec 2011 01:17:29 +0100 Subject: [Freeswitch-users] Odp: announce conf count (total callers minus 1)? In-Reply-To: <1324232567.1676.YahooMailNeo@web65301.mail.ac2.yahoo.com> References: <1324232567.1676.YahooMailNeo@web65301.mail.ac2.yahoo.com> Message-ID: <4eee829942e6c7.90667932@wp.pl> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/008b0655/attachment-0001.html From manieq at wp.eu Mon Dec 19 03:17:29 2011 From: manieq at wp.eu (Mariusz Czulada) Date: Mon, 19 Dec 2011 01:17:29 +0100 Subject: [Freeswitch-users] Odp: announce conf count (total callers minus 1)? In-Reply-To: <1324232567.1676.YahooMailNeo@web65301.mail.ac2.yahoo.com> References: <1324232567.1676.YahooMailNeo@web65301.mail.ac2.yahoo.com> Message-ID: <4eee829942e6c7.90667932@wp.pl> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/008b0655/attachment-0002.html From notlikeme75 at yahoo.com Mon Dec 19 03:25:49 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Sun, 18 Dec 2011 16:25:49 -0800 (PST) Subject: [Freeswitch-users] conf call count In-Reply-To: References: Message-ID: <1324254349.59671.YahooMailNeo@web65301.mail.ac2.yahoo.com> Mariusz, thankyou, that expression worked great! ? ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Sunday, December 18, 2011 7:17 PM Subject: FreeSWITCH-users Digest, Vol 66, Issue 88 ----- Forwarded Message ----- 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: Weekly Conference Torrents. (Stephen Dame) ? 2. Re: Weekly Conference Torrents. (Stephen Dame) ? 3. Re: Where are table definitions of database for freewsitch ? ? ? (latest GIT version) (Jeff Lenk) ? 4. announce conf count (total callers minus 1)? (Rodney) ? 5. destination_number cleanup (Elliott Vogel) ? 6. Re: Grandstream GXP 2000 (Ted Schober) ? 7. Odp: announce conf count (total callers minus 1)? ? ? ? (Mariusz Czulada) Jeroen,Giovanni Thanks, I tried playing all three wav,ogg,mp3? at 3x in vlc still not understandable. This is on Windows 7, where itunes,? media player do the same thing. I'll try playing direct from within freeswitch for now. Stephen -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Giovanni Maruzzelli Sent: Sunday, December 18, 2011 4:36 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Weekly Conference Torrents. maybe the recordings are at 48khz and you're playing at 16khz? -giovanni On 12/18/11, Jeroen Eeuwes wrote: > Hi Stephen, > >> Question. I have downloaded the last 4 weekly confrerecne calls from >> torrents,? but all the files play in slow motion sound like lion roaring > > I never downloaded it before, but it sounds the same here. > > If you speed up the file to 300% it sounds OK. Under linux I used this > to play the file. For example: > > mplayer -speed 3 conf_call_2011-12-14.mp3 > > You can also use sox to get a file which sounds better. For example: > > sox conf_call_2011-12-14.ogg soundsbetter.ogg speed 3 > > Beste regards, > Jeroen Eeuwes > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Just tried playing them direct in freeswitch use latest git... they sound the same not understandable by default I tried in normal, and cd quality 48,000 conf using conference 3300 play file.mp3 stephen -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeroen Eeuwes Sent: Sunday, December 18, 2011 1:36 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Weekly Conference Torrents. Hi Stephen, > Question. I have downloaded the last 4 weekly confrerecne calls from > torrents,? but all the files play in slow motion sound like lion roaring I never downloaded it before, but it sounds the same here. If you speed up the file to 300% it sounds OK. Under linux I used this to play the file. For example: mplayer -speed 3 conf_call_2011-12-14.mp3 You can also use sox to get a file which sounds better. For example: sox conf_call_2011-12-14.ogg soundsbetter.ogg speed 3 Beste regards, Jeroen Eeuwes _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org All the required tables should be created on first run. Same as always. Is that not happening? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Where-are-table-definitions-of-database-for-freewsitch-latest-GIT-version-tp7105666p7106232.html Sent from the freeswitch-users mailing list archive at Nabble.com. this works great, thank you to whomever created it. is there a way to make the conf count = list count -1 so the playback will only speak total "other" callers instead of including the person who press the option? conf/dialplan/default/01_Announce_Conf_Count.xml: I was wondering if anyone has a regex expression that works to return just digest? I have some clients sending requests formatted to +1 (555) 555-5555, 555-555-5555, 555.555.5555 which aren?t be processed ?by our dial plan because we are expecting all numbers (5555555555) Jonas Gauffin writes: > > Hello > > I have problems with TCP and GXP2000 phones. > They work fine using UDP, but no calls arrive if I switch to TCP. > I've upgraded to the latest GS firmware today to see it that helped, > but it didn't. > > Have anyone else had problems with TCP? > > Regards, >? Jonas > > If you touch the account page after setting it up and registering with FS, FS will then deny registration.? I suspect the password gets messed up.? If you need to change the account settings you need to factory reset and get it all right with one pass. What about ? ? Regards, Mariusz ? Dnia 18-12-2011 o godz. 19:22 Rodney napisa?(a): this works great, thank you to whomever created it. is there a way to make the conf count = list count -1 so the playback will only speak total "other" callers instead of including the person who press the option? > > > > >conf/dialplan/default/01_Announce_Conf_Count.xml: > > > > > > > > > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111218/a7abd9c2/attachment-0001.html From peter at spinato.ca Mon Dec 19 03:28:14 2011 From: peter at spinato.ca (Peter Spinato) Date: Sun, 18 Dec 2011 19:28:14 -0500 Subject: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue In-Reply-To: References: <01d301ccbc33$c2d7d360$48877a20$@ca> <02ac01ccbd32$36b4ba10$a41e2e30$@ca> Message-ID: <032901ccbde5$1a01d570$4e058050$@ca> I changed the external profile to port 5060 - I can see it using that profile and it still sends out the local IP at the RTP IP . even though ext-rtp-ip is set to the public one .? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Saturday, December 17, 2011 10:32 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue Yes, send it to the freeswitch server port 5080 Or if you don't need the internal profile, remove it and set external to use port 5060. -Avi On Sun, Dec 18, 2011 at 5:07 AM, Peter Spinato wrote: The calls go to internal profile - I guess case the OpenSIPs server is local lan - is there a way to force that calls to external profile as I find calls that hit that profile load the ext-rtp-ip ip and work. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Saturday, December 17, 2011 11:07 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue Someone familiar with this might be able to answer right off the bat, but if you pastebin a SIP trace (and an fs_cli log for completeness) , the problem should become apparent. Are you using the internal profile for all the calls? If you use the external.xml profile, you might need to set your ext-rtp-ip in that file, too. Seeing a trace will tell (almost) the whole story. -Avi On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato wrote: All, Hopefully someone can assist me - I'll gladly give $50 to the person who helps me fix the issue - I have an OpenSIPs server configured as a load balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) that receives the call and forwards it to my Freeswitch Server for an IVR. When I had the call just routing to the Freeswitch server I got the audio working by setting the ext-rtp-ip to the public IP. Now that I route the SIP call through the OpenSIPs server there is no audio - I'm guessing it a NAT issue as always. Both the OpenSIPS and FreeSwitch server have an internal private IP - OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2 Both servers also have a public IP that routes to the internal IP OpensIPS = 47.1.1.1 and Freeswitch = 47.1.1.2 Call gets received by the OpenSIPs via the external IP - routes the call to the internal IP on the Freeswitch server which answers the call - but no audio - I think the FreeSwitch is trying to route the RTP Audio via its internal private IP instead of the public IP of 47.1.1.2. Not sure if this is the real issue or how to configure it route RTP properly ... all help is appreciated. $50 Paypal to whoever fixes this for me! Thanks -Peter _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111218/474dc775/attachment.html From xing2kin at yahoo.com Mon Dec 19 04:11:10 2011 From: xing2kin at yahoo.com (king2kin) Date: Sun, 18 Dec 2011 17:11:10 -0800 (PST) Subject: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) In-Reply-To: <1324229445946-7106232.post@n2.nabble.com> References: <1324207414.49834.YahooMailNeo@web39703.mail.mud.yahoo.com> <1324229445946-7106232.post@n2.nabble.com> Message-ID: <1324257070.51489.YahooMailNeo@web39703.mail.mud.yahoo.com> No, these tables?were not created automatically by freeswitch on windows 2003 while MySQL is running. During the past three months, I had been playing around the GIT version of 2011-08-31, whose log file [freeswitch.log] didn't complain those database tables that I listed in my previous email. ? It seems to me that these db tables?come up only?in more recent GIT version. From: Jeff Lenk To: freeswitch-users at lists.freeswitch.org Sent: Monday, December 19, 2011 1:30 AM Subject: Re: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) All the required tables should be created on first run. Same as always. Is that not happening? -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Where-are-table-definitions-of-database-for-freewsitch-latest-GIT-version-tp7105666p7106232.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/20111218/61cc0420/attachment.html From brian at freeswitch.org Mon Dec 19 04:24:07 2011 From: brian at freeswitch.org (Brian West) Date: Sun, 18 Dec 2011 19:24:07 -0600 Subject: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) In-Reply-To: <1324257070.51489.YahooMailNeo@web39703.mail.mud.yahoo.com> References: <1324207414.49834.YahooMailNeo@web39703.mail.mud.yahoo.com> <1324229445946-7106232.post@n2.nabble.com> <1324257070.51489.YahooMailNeo@web39703.mail.mud.yahoo.com> Message-ID: <7AE44DCF-05C7-42CB-A24C-7AD299F5594E@freeswitch.org> You're using a very old version of FreeSWITCH... why ? /b On Dec 18, 2011, at 7:11 PM, king2kin wrote: > No, these tables were not created automatically by freeswitch on windows 2003 while MySQL is running. > During the past three months, I had been playing around the GIT version of 2011-08-31, whose log file [freeswitch.log] didn't complain those database tables that I listed in my previous email. > > It seems to me that these db tables come up only in more recent GIT version. > From: Jeff Lenk > To: freeswitch-users at lists.freeswitch.org > Sent: Monday, December 19, 2011 1:30 AM > Subject: Re: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) > > All the required tables should be created on first run. Same as always. Is > that not happening? -- 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/20111218/2d6b3b21/attachment-0001.html From ryan at kaevee.com Mon Dec 19 05:08:46 2011 From: ryan at kaevee.com (Ryan V) Date: Mon, 19 Dec 2011 07:38:46 +0530 Subject: [Freeswitch-users] destination_number cleanup In-Reply-To: References: Message-ID: On Mon, Dec 19, 2011 at 2:56 AM, Elliott Vogel wrote: > I was wondering if anyone has a regex expression that works to return > just digest? I have some clients sending requests formatted to +1 (555) > 555-5555, 555-555-5555, 555.555.5555 which aren?t be processed by our dial > plan because we are expecting all numbers (5555555555) **** > > How about this? ^\+1\s+\((\d{3})\)\s(\d{3})-(\d{4})$|^(\d{3})-(\d{3})-(\d{4})$|^(\d{3})\.(\d{3})\.(\d{4})$ $1, $2 and $3 should give you the full number. Regards, Ryan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/c0399d23/attachment.html From brian at freeswitch.org Mon Dec 19 05:12:32 2011 From: brian at freeswitch.org (Brian West) Date: Sun, 18 Dec 2011 20:12:32 -0600 Subject: [Freeswitch-users] destination_number cleanup In-Reply-To: References: Message-ID: <9714BFE2-4E7E-47B2-9994-794CE037F285@freeswitch.org> Why do you have broken devices sending you numbers in that screwed up format in the request URI in the first place? At this point you should be using XML_CURL then cleaning that up if this is what you're hitting. Or you tell your clients to fix their badly broken behavior. /b On Dec 18, 2011, at 8:08 PM, Ryan V wrote: > On Mon, Dec 19, 2011 at 2:56 AM, Elliott Vogel wrote: > >> I was wondering if anyone has a regex expression that works to return >> just digest? I have some clients sending requests formatted to +1 (555) >> 555-5555, 555-555-5555, 555.555.5555 which aren?t be processed by our dial >> plan because we are expecting all numbers (5555555555) **** >> >> > How about this? > > ^\+1\s+\((\d{3})\)\s(\d{3})-(\d{4})$|^(\d{3})-(\d{3})-(\d{4})$|^(\d{3})\.(\d{3})\.(\d{4})$ > > $1, $2 and $3 should give you the full number. > > Regards, > > Ryan -- 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/20111218/ebb8d7ca/attachment.html From govoiper at gmail.com Mon Dec 19 07:41:54 2011 From: govoiper at gmail.com (Sammy Govind) Date: Mon, 19 Dec 2011 09:41:54 +0500 Subject: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue In-Reply-To: <032901ccbde5$1a01d570$4e058050$@ca> References: <01d301ccbc33$c2d7d360$48877a20$@ca> <02ac01ccbd32$36b4ba10$a41e2e30$@ca> <032901ccbde5$1a01d570$4e058050$@ca> Message-ID: Hey, I think you need to use RTP proxy in bridged mode on OpenSIPS and use the force_rtp_proxy() function with IE and EI flags so that the SDPs IPs change as follows End-USER<===========>[*Public-IP*]OPENSIPS/RTPProxy[*Private-IP* ]<===========>[*Private-IP*]FreeSWITCH Use FreeSWITCH internal or external domain, make sure you've correct properties and contexts set. Regards, Sammy. On Mon, Dec 19, 2011 at 5:28 AM, Peter Spinato wrote: > I changed the external profile to port 5060 ? I can see it using that > profile and it still sends out the local IP at the RTP IP ? even though > ext-rtp-ip is set to the public one ??**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Saturday, December 17, 2011 10:32 PM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch > issue**** > > ** ** > > Yes, send it to the freeswitch server port 5080**** > > Or if you don't need the internal profile, remove it and set external to > use port 5060. > **** > > -Avi**** > > ** ** > > On Sun, Dec 18, 2011 at 5:07 AM, Peter Spinato wrote:** > ** > > The calls go to internal profile ? I guess case the OpenSIPs server is > local lan ? is there a way to force that calls to external profile as I > find calls that hit that profile load the ext-rtp-ip ip and work.**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Saturday, December 17, 2011 11:07 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch > issue**** > > **** > > Someone familiar with this might be able to answer right off the bat, but > if you pastebin a SIP trace (and an fs_cli log for completeness) , the > problem should become apparent.**** > > **** > > Are you using the internal profile for all the calls? If you use the > external.xml profile, you might need to set your ext-rtp-ip in that file, > too. Seeing a trace will tell (almost) the whole story.**** > > > **** > > -Avi**** > > **** > > On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato wrote:* > *** > > All,**** > > Hopefully someone can assist me - I'll gladly give $50 to the person who > helps me fix the issue - I have an OpenSIPs server configured as a load > balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) > that receives the call and forwards it to my Freeswitch Server for an IVR. > When I had the call just routing to the Freeswitch server I got the audio > working by setting the ext-rtp-ip to the public IP. Now that I route the > SIP call through the OpenSIPs server there is no audio - I'm guessing it a > NAT issue as always.**** > > **** > > Both the OpenSIPS and FreeSwitch server have an internal private IP - > OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2**** > > Both servers also have a public IP that routes to the internal IP OpensIPS > = 47.1.1.1 and Freeswitch = 47.1.1.2**** > > **** > > Call gets received by the OpenSIPs via the external IP - routes the call > to the internal IP on the Freeswitch server which answers the call - but no > audio - I think the FreeSwitch is trying to route the RTP Audio via its > internal private IP instead of the public IP of 47.1.1.2. Not sure if this > is the real issue or how to configure it route RTP properly ... all help > is appreciated. $50 Paypal to whoever fixes this for me! Thanks**** > > **** > > -Peter**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11* > *** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11* > *** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/01608604/attachment-0001.html From xing2kin at yahoo.com Mon Dec 19 09:04:41 2011 From: xing2kin at yahoo.com (king2kin) Date: Sun, 18 Dec 2011 22:04:41 -0800 (PST) Subject: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) In-Reply-To: <7AE44DCF-05C7-42CB-A24C-7AD299F5594E@freeswitch.org> References: <1324207414.49834.YahooMailNeo@web39703.mail.mud.yahoo.com> <1324229445946-7106232.post@n2.nabble.com> <1324257070.51489.YahooMailNeo@web39703.mail.mud.yahoo.com> <7AE44DCF-05C7-42CB-A24C-7AD299F5594E@freeswitch.org> Message-ID: <1324274681.14128.YahooMailNeo@web39704.mail.mud.yahoo.com> No, this occurs on the latest GIT version of freeswitch that I downloaded on Dec. 18, 2011 by command: { ?git clone git://git.freeswitch.org/freeswitch.git } ?The early GIT version (e.g. 2011-08-31) doesn't have such problems. ? The latest version can run on windows 2003, but keep printing out error messages in freeswitch.log and console, for example: ? [DEBUG] switch_core_sqldb.c:890 SQL ERR [no such table: sip_registrations] [delete from sip_registrations where (contact like '%TCP%' or status like '%TCP%' or status like '%TLS%') and hostname='W2k3T602' and network_ip like '%' and network_port like '%' and sip_username like '%' and mwi_user? like '%' and mwi_host like '%' and orig_server_host like '%' and orig_hostname like '%'] Auto Generating Table! ? [DEBUG] switch_core_sqldb.c:897 SQL ERR [no such table: sip_subscriptions] [DEBUG] switch_core_sqldb.c:897 SQL ERR [no such table: sip_presence] [DEBUG] switch_core_sqldb.c:890 SQL ERR [no such table: sip_dialogs] ...... ? also, while running ?the new version of freeswitch, the following error messages keep coming up in freeswitch.log: { 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it } ? From: Brian West To: FreeSWITCH Users Help Sent: Monday, December 19, 2011 9:24 AM Subject: Re: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) You're using a very old version of FreeSWITCH... why ? /b On Dec 18, 2011, at 7:11 PM, king2kin wrote: No, these tables?were not created automatically by freeswitch on windows 2003 while MySQL is running. >During the past three months, I had been playing around the GIT version of 2011-08-31, whose log file [freeswitch.log] didn't complain those database tables that I listed in my previous email. >? >It seems to me that these db tables?come up only?in more recent GIT version. >From: Jeff Lenk >To:?freeswitch-users at lists.freeswitch.org? >Sent: Monday, December 19, 2011 1:30 AM >Subject: Re: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) > >All the required tables should be created on first run. Same as always. Is >that not happening? --? 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/20111218/5177a529/attachment.html From peter.olsson at visionutveckling.se Mon Dec 19 09:50:05 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Mon, 19 Dec 2011 07:50:05 +0100 Subject: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) Message-ID: Did you try to remove all tables, so FS can create everything from scratch? /Peter ----- Reply message ----- Fr?n: "king2kin" Datum: m?n, dec 19, 2011 07:11 Rubrik: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) Till: "FreeSWITCH Users Help" No, this occurs on the latest GIT version of freeswitch that I downloaded on Dec. 18, 2011 by command: { git clone git://git.freeswitch.org/freeswitch.git } The early GIT version (e.g. 2011-08-31) doesn't have such problems. The latest version can run on windows 2003, but keep printing out error messages in freeswitch.log and console, for example: [DEBUG] switch_core_sqldb.c:890 SQL ERR [no such table: sip_registrations] [delete from sip_registrations where (contact like '%TCP%' or status like '%TCP%' or status like '%TLS%') and hostname='W2k3T602' and network_ip like '%' and network_port like '%' and sip_username like '%' and mwi_user like '%' and mwi_host like '%' and orig_server_host like '%' and orig_hostname like '%'] Auto Generating Table! [DEBUG] switch_core_sqldb.c:897 SQL ERR [no such table: sip_subscriptions] [DEBUG] switch_core_sqldb.c:897 SQL ERR [no such table: sip_presence] [DEBUG] switch_core_sqldb.c:890 SQL ERR [no such table: sip_dialogs] ...... also, while running the new version of freeswitch, the following error messages keep coming up in freeswitch.log: { 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it } From: Brian West To: FreeSWITCH Users Help Sent: Monday, December 19, 2011 9:24 AM Subject: Re: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) You're using a very old version of FreeSWITCH... why ? /b On Dec 18, 2011, at 7:11 PM, king2kin wrote: No, these tables were not created automatically by freeswitch on windows 2003 while MySQL is running. During the past three months, I had been playing around the GIT version of 2011-08-31, whose log file [freeswitch.log] didn't complain those database tables that I listed in my previous email. It seems to me that these db tables come up only in more recent GIT version. From: Jeff Lenk > To: freeswitch-users at lists.freeswitch.org Sent: Monday, December 19, 2011 1:30 AM Subject: Re: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) All the required tables should be created on first run. Same as always. Is that not happening? -- 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 !DSPAM:4eeed36a32763536013952! From avi at avimarcus.net Mon Dec 19 09:56:58 2011 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 19 Dec 2011 08:56:58 +0200 Subject: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue In-Reply-To: References: <01d301ccbc33$c2d7d360$48877a20$@ca> <02ac01ccbd32$36b4ba10$a41e2e30$@ca> <032901ccbde5$1a01d570$4e058050$@ca> Message-ID: If that doesn't help, can you provide an actual siptrace of the call so we can SEE what's going on? ngrep can be quite helpful.. -Avi On Mon, Dec 19, 2011 at 6:41 AM, Sammy Govind wrote: > Hey, > > I think you need to use RTP proxy in bridged mode on OpenSIPS and use the > force_rtp_proxy() function with IE and EI flags so that the SDPs IPs change > as follows > > End-USER<===========>[*Public-IP*]OPENSIPS/RTPProxy[*Private-IP* > ]<===========>[*Private-IP*]FreeSWITCH > > Use FreeSWITCH internal or external domain, make sure you've correct > properties and contexts set. > > Regards, > Sammy. > > > On Mon, Dec 19, 2011 at 5:28 AM, Peter Spinato wrote: > >> I changed the external profile to port 5060 ? I can see it using that >> profile and it still sends out the local IP at the RTP IP ? even though >> ext-rtp-ip is set to the public one ??**** >> >> ** ** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus >> *Sent:* Saturday, December 17, 2011 10:32 PM >> >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch >> issue**** >> >> ** ** >> >> Yes, send it to the freeswitch server port 5080**** >> >> Or if you don't need the internal profile, remove it and set external to >> use port 5060. >> **** >> >> -Avi**** >> >> ** ** >> >> On Sun, Dec 18, 2011 at 5:07 AM, Peter Spinato wrote:* >> *** >> >> The calls go to internal profile ? I guess case the OpenSIPs server is >> local lan ? is there a way to force that calls to external profile as I >> find calls that hit that profile load the ext-rtp-ip ip and work.**** >> >> **** >> >> *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus >> *Sent:* Saturday, December 17, 2011 11:07 AM >> *To:* FreeSWITCH Users Help >> *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch >> issue**** >> >> **** >> >> Someone familiar with this might be able to answer right off the bat, but >> if you pastebin a SIP trace (and an fs_cli log for completeness) , the >> problem should become apparent.**** >> >> **** >> >> Are you using the internal profile for all the calls? If you use the >> external.xml profile, you might need to set your ext-rtp-ip in that file, >> too. Seeing a trace will tell (almost) the whole story.**** >> >> >> **** >> >> -Avi**** >> >> **** >> >> On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato wrote: >> **** >> >> All,**** >> >> Hopefully someone can assist me - I'll gladly give $50 to the person >> who helps me fix the issue - I have an OpenSIPs server configured as a load >> balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) >> that receives the call and forwards it to my Freeswitch Server for an IVR. >> When I had the call just routing to the Freeswitch server I got the audio >> working by setting the ext-rtp-ip to the public IP. Now that I route the >> SIP call through the OpenSIPs server there is no audio - I'm guessing it a >> NAT issue as always.**** >> >> **** >> >> Both the OpenSIPS and FreeSwitch server have an internal private IP - >> OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2**** >> >> Both servers also have a public IP that routes to the internal IP >> OpensIPS = 47.1.1.1 and Freeswitch = 47.1.1.2**** >> >> **** >> >> Call gets received by the OpenSIPs via the external IP - routes the call >> to the internal IP on the Freeswitch server which answers the call - but no >> audio - I think the FreeSwitch is trying to route the RTP Audio via its >> internal private IP instead of the public IP of 47.1.1.2. Not sure if this >> is the real issue or how to configure it route RTP properly ... all help >> is appreciated. $50 Paypal to whoever fixes this for me! Thanks**** >> >> **** >> >> -Peter**** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11 >> **** >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at 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: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11 >> **** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/c431a73d/attachment-0001.html From avi at avimarcus.net Mon Dec 19 10:07:33 2011 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 19 Dec 2011 09:07:33 +0200 Subject: [Freeswitch-users] destination_number cleanup In-Reply-To: <9714BFE2-4E7E-47B2-9994-794CE037F285@freeswitch.org> References: <9714BFE2-4E7E-47B2-9994-794CE037F285@freeswitch.org> Message-ID: Yes, this is really broken input! I thought it was bad some do +1 or 001 1 or no prefix to dial America! To modify Ryan's code to make it more flexible... ^(?:\+1|001|1)?\D*(\d{3})\D*(\d{3})\D*(\d{4})$ $1, $2 and $3 should give you the full number, so bridge to 1$1$2$3. it will match anything that has xxx-xxx-xxxx with *whatever* separating non-digits is in between. But it can only handle (by discarding) a prefix of +1, 1, or 001. I'd suggest using xml_curl or a lua script and doing a regex replace of any non-digit characters, if you really need. But how in the world is this your input? What devices are sending this?? -Avi On Mon, Dec 19, 2011 at 4:12 AM, Brian West wrote: > Why do you have broken devices sending you numbers in that screwed up > format in the request URI in the first place? At this point you should be > using XML_CURL then cleaning that up if this is what you're hitting. Or > you tell your clients to fix their badly broken behavior. > > /b > > On Dec 18, 2011, at 8:08 PM, Ryan V wrote: > > On Mon, Dec 19, 2011 at 2:56 AM, Elliott Vogel >wrote: > > I was wondering if anyone has a regex expression that works to return > > just digest? I have some clients sending requests formatted to +1 (555) > > 555-5555, 555-555-5555, 555.555.5555 which aren?t be processed by our dial > > plan because we are expecting all numbers (5555555555) **** > > > > How about this? > > > ^\+1\s+\((\d{3})\)\s(\d{3})-(\d{4})$|^(\d{3})-(\d{3})-(\d{4})$|^(\d{3})\.(\d{3})\.(\d{4})$ > > $1, $2 and $3 should give you the full number. > > Regards, > > Ryan > > > -- > 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/20111219/1ca38d79/attachment.html From justlikeef at gmail.com Mon Dec 19 10:17:11 2011 From: justlikeef at gmail.com (Rob Hutton) Date: Mon, 19 Dec 2011 02:17:11 -0500 Subject: [Freeswitch-users] Excluding busy extensions from intercom Message-ID: <201112190217.11783.justlikeef@gmail.com> I am trying to get a basic "All Page" setup working and have used the wiki and example configs to put together the following dialplan: Everything works as expected, except that intercom forces any active calls to hold, and likewise forces the originating call on hold. Is there a way of excluding any active users? I have tried both ${network_addr} and ${presence_id} in the sip_exclude_contact variable, and neither causes the orginator to be excluded. Console log is here: http://pastebin.freeswitch.org/18022 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/28aef67c/attachment.html From avi at avimarcus.net Mon Dec 19 14:52:29 2011 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 19 Dec 2011 13:52:29 +0200 Subject: [Freeswitch-users] PCI Compliance Over Telephone for Credit Cards- how? Message-ID: I'm planning on an IVR to accept credit card information for signing up and renewal of my services. Regarding fraud, I'm going to require at minimum a recording of name, who they are, or something or an actual live call. But for PCI compliance.. this says https://www.pcisecuritystandards.org/documents/protecting_telephone-based_payment_card_data.pdf on page 9: Call centers will need to ensure that transmission of cardholder data > across public networks is encrypted. > This is part of PCI DSS Requirement 4 and includes: > > - ... > > > - *Voice or data streams over Voice over IP (VoIP) telephone > systems, whenever sent over an open or public network. Note that only > those consumer or enterprise VoIP systems that provide strong > cryptography should be used. * > > > - Requiring agents to use analog telephone lines when a VoIP > telephone system does not provide strong cryptography. > > I'm doing dtmf, not voice, but I can't imagine that's LESS strict. I haven't really heard of any end-to-end encrypted origination lines. Is this guideline ignored? How do people deal with this? Does someone have T1 lines and offers encryption for origination...? -Avi Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/59878ef3/attachment-0001.html From peter at spinato.ca Mon Dec 19 15:12:31 2011 From: peter at spinato.ca (Peter Spinato) Date: Mon, 19 Dec 2011 07:12:31 -0500 Subject: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue In-Reply-To: References: <01d301ccbc33$c2d7d360$48877a20$@ca> <02ac01ccbd32$36b4ba10$a41e2e30$@ca> <032901ccbde5$1a01d570$4e058050$@ca> Message-ID: <03b301ccbe47$7d50e230$77f2a690$@ca> From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Monday, December 19, 2011 1:57 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue If that doesn't help, can you provide an actual siptrace of the call so we can SEE what's going on? ngrep can be quite helpful.. -Avi On Mon, Dec 19, 2011 at 6:41 AM, Sammy Govind wrote: Hey, I think you need to use RTP proxy in bridged mode on OpenSIPS and use the force_rtp_proxy() function with IE and EI flags so that the SDPs IPs change as follows End-USER<===========>[Public-IP]OPENSIPS/RTPProxy[Private-IP]<===========>[P rivate-IP]FreeSWITCH Use FreeSWITCH internal or external domain, make sure you've correct properties and contexts set. Regards, Sammy. On Mon, Dec 19, 2011 at 5:28 AM, Peter Spinato wrote: I changed the external profile to port 5060 - I can see it using that profile and it still sends out the local IP at the RTP IP . even though ext-rtp-ip is set to the public one .? From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Saturday, December 17, 2011 10:32 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue Yes, send it to the freeswitch server port 5080 Or if you don't need the internal profile, remove it and set external to use port 5060. -Avi On Sun, Dec 18, 2011 at 5:07 AM, Peter Spinato wrote: The calls go to internal profile - I guess case the OpenSIPs server is local lan - is there a way to force that calls to external profile as I find calls that hit that profile load the ext-rtp-ip ip and work. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Saturday, December 17, 2011 11:07 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue Someone familiar with this might be able to answer right off the bat, but if you pastebin a SIP trace (and an fs_cli log for completeness) , the problem should become apparent. Are you using the internal profile for all the calls? If you use the external.xml profile, you might need to set your ext-rtp-ip in that file, too. Seeing a trace will tell (almost) the whole story. -Avi On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato wrote: All, Hopefully someone can assist me - I'll gladly give $50 to the person who helps me fix the issue - I have an OpenSIPs server configured as a load balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) that receives the call and forwards it to my Freeswitch Server for an IVR. When I had the call just routing to the Freeswitch server I got the audio working by setting the ext-rtp-ip to the public IP. Now that I route the SIP call through the OpenSIPs server there is no audio - I'm guessing it a NAT issue as always. Both the OpenSIPS and FreeSwitch server have an internal private IP - OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2 Both servers also have a public IP that routes to the internal IP OpensIPS = 47.1.1.1 and Freeswitch = 47.1.1.2 Call gets received by the OpenSIPs via the external IP - routes the call to the internal IP on the Freeswitch server which answers the call - but no audio - I think the FreeSwitch is trying to route the RTP Audio via its internal private IP instead of the public IP of 47.1.1.2. Not sure if this is the real issue or how to configure it route RTP properly ... all help is appreciated. $50 Paypal to whoever fixes this for me! Thanks -Peter _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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: 2012.0.1890 / Virus Database: 2108/4689 - Release Date: 12/18/11 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/d00ddec4/attachment-0001.html -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: opensips_call.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/d00ddec4/attachment-0002.txt -------------- next part -------------- An embedded and charset-unspecified text was scrubbed... Name: freeswitch_call.txt Url: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/d00ddec4/attachment-0003.txt From avi at avimarcus.net Mon Dec 19 15:23:49 2011 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 19 Dec 2011 14:23:49 +0200 Subject: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue In-Reply-To: <03b301ccbe47$7d50e230$77f2a690$@ca> References: <01d301ccbc33$c2d7d360$48877a20$@ca> <02ac01ccbd32$36b4ba10$a41e2e30$@ca> <032901ccbde5$1a01d570$4e058050$@ca> <03b301ccbe47$7d50e230$77f2a690$@ca> Message-ID: 192.168.23.33 is your FS box? So did you edit your new external.xml and set rtp-ip (and ext-rtp-ip?) to the public IP, and then reload FS or just mod_sofia, or just "sofia profile $profile_name restart" ? -Avi On Mon, Dec 19, 2011 at 2:12 PM, Peter Spinato wrote: > ** ** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Monday, December 19, 2011 1:57 AM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch > issue**** > > ** ** > > If that doesn't help, can you provide an actual siptrace of the call so we > can SEE what's going on?**** > > ngrep can be quite helpful..**** > > > **** > > -Avi**** > > ** ** > > On Mon, Dec 19, 2011 at 6:41 AM, Sammy Govind wrote:* > *** > > Hey, **** > > ** ** > > I think you need to use RTP proxy in bridged mode on OpenSIPS and use the > force_rtp_proxy() function with IE and EI flags so that the SDPs IPs change > as follows**** > > ** ** > > End-USER<===========>[*Public-IP*]OPENSIPS/RTPProxy[*Private-IP* > ]<===========>[*Private-IP*]FreeSWITCH**** > > ** ** > > Use FreeSWITCH internal or external domain, make sure you've correct > properties and contexts set.**** > > ** ** > > Regards,**** > > Sammy.**** > > ** ** > > On Mon, Dec 19, 2011 at 5:28 AM, Peter Spinato wrote:** > ** > > I changed the external profile to port 5060 ? I can see it using that > profile and it still sends out the local IP at the RTP IP ? even though > ext-rtp-ip is set to the public one ??**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Saturday, December 17, 2011 10:32 PM**** > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch > issue**** > > **** > > Yes, send it to the freeswitch server port 5080**** > > Or if you don't need the internal profile, remove it and set external to > use port 5060. > **** > > -Avi**** > > **** > > On Sun, Dec 18, 2011 at 5:07 AM, Peter Spinato wrote:** > ** > > The calls go to internal profile ? I guess case the OpenSIPs server is > local lan ? is there a way to force that calls to external profile as I > find calls that hit that profile load the ext-rtp-ip ip and work.**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Saturday, December 17, 2011 11:07 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch > issue**** > > **** > > Someone familiar with this might be able to answer right off the bat, but > if you pastebin a SIP trace (and an fs_cli log for completeness) , the > problem should become apparent.**** > > **** > > Are you using the internal profile for all the calls? If you use the > external.xml profile, you might need to set your ext-rtp-ip in that file, > too. Seeing a trace will tell (almost) the whole story.**** > > > **** > > -Avi**** > > **** > > On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato wrote:* > *** > > All,**** > > Hopefully someone can assist me - I'll gladly give $50 to the person who > helps me fix the issue - I have an OpenSIPs server configured as a load > balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) > that receives the call and forwards it to my Freeswitch Server for an IVR.. > When I had the call just routing to the Freeswitch server I got the audio > working by setting the ext-rtp-ip to the public IP. Now that I route the > SIP call through the OpenSIPs server there is no audio - I'm guessing it a > NAT issue as always.**** > > **** > > Both the OpenSIPS and FreeSwitch server have an internal private IP - > OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2**** > > Both servers also have a public IP that routes to the internal IP OpensIPS > = 47.1.1.1 and Freeswitch = 47.1.1.2**** > > **** > > Call gets received by the OpenSIPs via the external IP - routes the call > to the internal IP on the Freeswitch server which answers the call - but no > audio - I think the FreeSwitch is trying to route the RTP Audio via its > internal private IP instead of the public IP of 47.1.1.2. Not sure if this > is the real issue or how to configure it route RTP properly ... all help > is appreciated. $50 Paypal to whoever fixes this for me! Thanks**** > > **** > > -Peter**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11* > *** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11* > *** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: 2012.0.1890 / Virus Database: 2108/4689 - Release Date: 12/18/11* > *** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/d081d651/attachment-0001.html From govoiper at gmail.com Mon Dec 19 15:29:44 2011 From: govoiper at gmail.com (Sammy Govind) Date: Mon, 19 Dec 2011 17:29:44 +0500 Subject: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue In-Reply-To: <03b301ccbe47$7d50e230$77f2a690$@ca> References: <01d301ccbc33$c2d7d360$48877a20$@ca> <02ac01ccbd32$36b4ba10$a41e2e30$@ca> <032901ccbde5$1a01d570$4e058050$@ca> <03b301ccbe47$7d50e230$77f2a690$@ca> Message-ID: Hi, I'm having hard time understanding which one is FreeSWICTH and which on is openSIPS. Looking at the sip tarces however, I must say you should at-max have a one-way audio because the remote end-point is sending its Public Address for Media-connectivity, you freeswitch in return is sending its private IP to OpenSIPS and openSIPs is relaying the same Private-IP to the remote-end point... U 192.168.23.34:5060 -> 184.150.225.230:5062 SIP/2.0 200 OK. Via: SIP/2.0/UDP 184.150.225.230:5062;received=184.150.225.230;rport=5062;branch=z9hG4bK-4008200999-3775991592-3239888547-3704563422. Record-Route: . From: ;tag=917850919-3775991592-3239888547-3704563422. To: ;tag=9mr8cyc2pgjmS. Call-ID: 274bb6c6280f11e1a3c61cc1de26cfdc at 184.150.225.230. CSeq: 1 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f4320b5 2011-11-28 08-27-46 -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-Type: application/sdp. Content-Disposition: session. Content-Length: 249. Remote-Party-ID: "16474272135" ;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1324008356 1324008357 IN IP4 *192.168.23.33*. s=FreeSWITCH. c=IN IP4 *192.168.23.33*. t=0 0. m=audio 49774 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. And you know the remote end-point is sending the media to provate ip of FS within its LAN...!! So, Like I said use RTPproxy module such that newer INVITE look Like below.. U 192.168.23.34:5060 -> 184.150.225.230:5062 SIP/2.0 200 OK. Via: SIP/2.0/UDP 184.150.225.230:5062;received=184.150.225.230;rport=5062;branch=z9hG4bK-4008200999-3775991592-3239888547-3704563422. Record-Route: . From: ;tag=917850919-3775991592-3239888547-3704563422. To: ;tag=9mr8cyc2pgjmS. Call-ID: 274bb6c6280f11e1a3c61cc1de26cfdc at 184.150.225.230. CSeq: 1 INVITE. Contact: . User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-f4320b5 2011-11-28 08-27-46 -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-Type: application/sdp. Content-Disposition: session. Content-Length: 249. Remote-Party-ID: "16474272135" ;party=calling;privacy=off;screen=no. . v=0. o=FreeSWITCH 1324008356 1324008357 IN IP4 *PU.BL.IC.IP of OpenSIPS*. s=FreeSWITCH. c=IN IP4 *PU.BL.IC.IP of OpenSIPS* . t=0 0. m=audio 49774 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. Wish you best of Luck.! Regards, Sammy On Mon, Dec 19, 2011 at 5:12 PM, Peter Spinato wrote: > ** ** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Monday, December 19, 2011 1:57 AM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch > issue**** > > ** ** > > If that doesn't help, can you provide an actual siptrace of the call so we > can SEE what's going on?**** > > ngrep can be quite helpful..**** > > > **** > > -Avi**** > > ** ** > > On Mon, Dec 19, 2011 at 6:41 AM, Sammy Govind wrote:* > *** > > Hey, **** > > ** ** > > I think you need to use RTP proxy in bridged mode on OpenSIPS and use the > force_rtp_proxy() function with IE and EI flags so that the SDPs IPs change > as follows**** > > ** ** > > End-USER<===========>[*Public-IP*]OPENSIPS/RTPProxy[*Private-IP* > ]<===========>[*Private-IP*]FreeSWITCH**** > > ** ** > > Use FreeSWITCH internal or external domain, make sure you've correct > properties and contexts set.**** > > ** ** > > Regards,**** > > Sammy.**** > > ** ** > > On Mon, Dec 19, 2011 at 5:28 AM, Peter Spinato wrote:** > ** > > I changed the external profile to port 5060 ? I can see it using that > profile and it still sends out the local IP at the RTP IP ? even though > ext-rtp-ip is set to the public one ??**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Saturday, December 17, 2011 10:32 PM**** > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch > issue**** > > **** > > Yes, send it to the freeswitch server port 5080**** > > Or if you don't need the internal profile, remove it and set external to > use port 5060. > **** > > -Avi**** > > **** > > On Sun, Dec 18, 2011 at 5:07 AM, Peter Spinato wrote:** > ** > > The calls go to internal profile ? I guess case the OpenSIPs server is > local lan ? is there a way to force that calls to external profile as I > find calls that hit that profile load the ext-rtp-ip ip and work.**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Saturday, December 17, 2011 11:07 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch > issue**** > > **** > > Someone familiar with this might be able to answer right off the bat, but > if you pastebin a SIP trace (and an fs_cli log for completeness) , the > problem should become apparent.**** > > **** > > Are you using the internal profile for all the calls? If you use the > external.xml profile, you might need to set your ext-rtp-ip in that file, > too. Seeing a trace will tell (almost) the whole story.**** > > > **** > > -Avi**** > > **** > > On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato wrote:* > *** > > All,**** > > Hopefully someone can assist me - I'll gladly give $50 to the person who > helps me fix the issue - I have an OpenSIPs server configured as a load > balancer (http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS) > that receives the call and forwards it to my Freeswitch Server for an IVR. > When I had the call just routing to the Freeswitch server I got the audio > working by setting the ext-rtp-ip to the public IP. Now that I route the > SIP call through the OpenSIPs server there is no audio - I'm guessing it a > NAT issue as always.**** > > **** > > Both the OpenSIPS and FreeSwitch server have an internal private IP - > OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2**** > > Both servers also have a public IP that routes to the internal IP OpensIPS > = 47.1.1.1 and Freeswitch = 47.1.1.2**** > > **** > > Call gets received by the OpenSIPs via the external IP - routes the call > to the internal IP on the Freeswitch server which answers the call - but no > audio - I think the FreeSwitch is trying to route the RTP Audio via its > internal private IP instead of the public IP of 47.1.1.2. Not sure if this > is the real issue or how to configure it route RTP properly ... all help > is appreciated. $50 Paypal to whoever fixes this for me! Thanks**** > > **** > > -Peter**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11* > *** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11* > *** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: 2012.0.1890 / Virus Database: 2108/4689 - Release Date: 12/18/11* > *** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/20e43033/attachment-0001.html From dgarcia at anew.com.ve Mon Dec 19 15:54:24 2011 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Mon, 19 Dec 2011 08:24:24 -0430 Subject: [Freeswitch-users] PCI Compliance Over Telephone for Credit Cards- how? In-Reply-To: References: Message-ID: <4EEF3400.4080204@anew.com.ve> Hi, Your questin is quite difficult to answer because depend on your country laws. About public network, I think you are concern about TDM service (PSTN). Well, as always, some stuff are not considered by the people who made some laws. Well, in my country customer are encouraged to deploy cross-systems. For example, PIN number is set by internet (where you can implement all crypting available). By phone in TDM no crypting, ( and by VoIP you have to implement SIPS and SRTP, TLS at least) then very strong set of questions/answer to do a positive verification of the caller. Of course, where you have to put encryption and security is inside of your systems, and deploy a good system control to control and manage your sensible data. Take a look this links: http://wiki.linuxwall.info/doku.php/en:ressources:dossiers:voip:tls_sips_rtps and http://www.vadese.org/files/upload/Best_practices_VoIP_en_v20.pdf I hope this helps On 12/19/2011 7:22 AM, Avi Marcus wrote: > I'm planning on an IVR to accept credit card information for signing > up and renewal of my services. > Regarding fraud, I'm going to require at minimum a recording of name, > who they are, or something or an actual live call. > > But for PCI compliance.. this says > https://www.pcisecuritystandards.org/documents/protecting_telephone-based_payment_card_data.pdf on > page 9: > > Call centers will need to ensure that transmission of cardholder > data across public networks is encrypted. > This is part of PCI DSS Requirement 4 and includes: > > * ... > > * *Voice or data streams over Voice over IP (VoIP) telephone > systems, whenever sent over an open or public network. Note > that only those consumer or enterprise VoIP systems that > provide strong cryptography should be used. * > > * Requiring agents to use analog telephone lines when a VoIP > telephone system does not provide strong cryptography. > > I'm doing dtmf, not voice, but I can't imagine that's LESS strict. > > I haven't really heard of any end-to-end encrypted origination lines. > Is this guideline ignored? How do people deal with this? Does someone > have T1 lines and offers encryption for origination...? > > -Avi Marcus > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: 2012.0.1890 / Virus Database: 2108/4684 - Release Date: 12/16/11 > -- 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/20111219/fb4634fb/attachment.html From notlikeme75 at yahoo.com Mon Dec 19 18:04:31 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Mon, 19 Dec 2011 07:04:31 -0800 (PST) Subject: [Freeswitch-users] leave voicemail then drop to IVR Message-ID: <1324307071.13580.YahooMailNeo@web65308.mail.ac2.yahoo.com> I am trying to send callers to a voicemail box using pagd and then after they leave the message get dropped back into IVR main_menu. what is the method for this? using the following dialplan. the caller gets hung up from the whole system. thank you ?condition??? ? destination_number??? ? ^757$??? ? 1??? ? ? action??? ? answer??? ? ??? ? 3??? ? ? action??? ? sleep??? ? 1000??? ? 4??? ? ? action??? ? play_and_get_digits??? ? 4 4 3 5000 # checkvoicemail.wav ivr/ivr-that_was_an_invalid_entry.wav vmb \d+??? ? 12??? ? ? action??? ? phrase??? ? spell,${vmb}??? ? 15??? ? ? action??? ? voicemail??? ? default ${domain_name} ${vmb}??? ? 20??? ? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/513e1b7c/attachment.html From Hector.Geraldino at ip-soft.net Mon Dec 19 18:29:15 2011 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Mon, 19 Dec 2011 10:29:15 -0500 Subject: [Freeswitch-users] Threads remain after calling close on Java client In-Reply-To: <8b41de351c0d1365e3786e7a60645275@mail.gmail.com> References: <8b41de351c0d1365e3786e7a60645275@mail.gmail.com> Message-ID: <6A6B4C284AD15042B429EB9D904544AD0225507041@NY1-EXMB-01.ip-soft.net> Hi Neil, Can you get a thread dump of the tomcat process to try to figure out what this problem is about? Or at least, try to connect the jconsole to the tomcat process and get the StackTrace of one of these threads to have a better idea of what is going on. IIRC I've fixed a couple of bugs for this library, but the patches haven't been tested by the main developer (dvarnes) nor integrated on the repository (freeswitch-contrib). If this problem can be fixed with my patched code, I would be happy to share it with you. Good luck! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Neil Davis Sent: Friday, December 16, 2011 7:09 PM To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] Threads remain after calling close on Java client Hi, I built a web application that connects to Freeswitch using the org.freeswitch.esl.client.Client. I connect the Client object from a Spring annotated service that I call from a Spring controller. I put the connected client in my ServletContext, so I can access it later to call client.cancelEventSubscriptions() and client.close() from my ServletContextListener contextDestroyed method when Tomcat is shutting down. The problem I'm having is that even after I call close on the client, there are still a bunch of active threads that the client has spawned in the background. These threads are causing Tomcat to hang when I'm shutting down. Can anyone suggest an approach that would enable my application to disconnect the Freeswitch client when Tomcat is shutting down that would allow Tomcat to shutdown gracefully? Below are errors from my Tomcat log for the threads that I have identified as being related to the Freeswitch client. I don't know how I can get to these threads to interrupt them and Client.close() seems to leave them hanging. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-3-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-4-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-2] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-3] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-4] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-5] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-6] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-7] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-8] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-9] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-10] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-11] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-12] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-13] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-14] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-15] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-16] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.internal.ThreadLocalBoolean] (value [org.jboss.netty.util.internal.ThreadLocalBoolean at 186e192]) and a value of type [java.lang.Boolean] (value [false]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.CharsetUtil$1] (value [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type [java.util.IdentityHashMap] (value [{windows-1252=sun.nio.cs.MS1252$Encoder at 373f86}]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.internal.ThreadLocalRandom$1] (value [org.jboss.netty.util.internal.ThreadLocalRandom$1 at 12bb519]) and a value of type [org.jboss.netty.util.internal.ThreadLocalRandom] (value [org.jboss.netty.util.internal.ThreadLocalRandom at 7e9dbc]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.CharsetUtil$1] (value [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type [java.util.IdentityHashMap] (value [{windows-1252=sun.nio.cs.MS1252$Encoder at a5b041}]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Thanks, Neil Davis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/bc87ccf6/attachment-0001.html From rakib0000 at hotmail.com Sun Dec 18 14:28:42 2011 From: rakib0000 at hotmail.com (rakib 0000) Date: Sun, 18 Dec 2011 22:28:42 +1100 Subject: [Freeswitch-users] Problem with using mod_xml_curl at user authentication Message-ID: Hello list, I'm a newbie FreeSwitch user, trying to learn how mod_xml_curl works. I've done the basics things mentioned in FreeSwitch wiki pages about using mod_xml_curl. Now my plan to use mod_xml_curl for user registration. So, what I've done is - I wrote a server end application which reads POST request from mod_xml_curl and tries to find whether it's a valid post request (REGISTER) or not. If the request is a REGISTER request - then the application extracts the user id, domain. Then, it queries the database and extracts password and creates xml response. One of my sample response are like below (the following response has been taken from /tmp):
With above response, I didn't had any luck. It fails to register, with following response: 2011-12-18 14:51:29.253474 [WARNING] sofia_reg.c:2446 Can't find user [1100 at 192.168.0.108] You must define a domain called '192.168.0.108' in your directory and add a user with the id="1100" attribute and you must configure your device to use the proper domain in it's authentication credentials. To figure it out, I've tried in different ways too. Like, I've used the above response patterns to create a user under conf/directory/default/ directory, but didn't work. Then I tried the following xml response pattern : with the above xml I manage to create a user and authenticate perfectly. But, I've modified my server end application to give a that kind of response, but didn't work though. I've copied the server response (from the content dumped under /tmp) to create an user under conf/directory/default/ directory - but it works. So, my question is what am I missing here? FreeSwitch tries to get the response from the server, it gets the response too, but somehow not interpreting the response (or something else?) and gives "can't find user" error. Any hint's , suggestion will be very helpful. (note, I'm not using any webserver here.) Thanks in advance, Rakib -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111218/07882dec/attachment-0001.html From Dave.May at patlive.com Sat Dec 17 15:26:44 2011 From: Dave.May at patlive.com (Dave May) Date: Sat, 17 Dec 2011 07:26:44 -0500 Subject: [Freeswitch-users] Threads remain after calling close on Java client In-Reply-To: <8b41de351c0d1365e3786e7a60645275@mail.gmail.com> References: <8b41de351c0d1365e3786e7a60645275@mail.gmail.com> Message-ID: <009DEE08474F5246848F1B355FA1C380010A7F5A@mail2.patlive.local> I experienced similar problems when load testing Plivo on the latest Git, but didn't feel like I had enough data gathered for a "proper" report. After each call, the remote ESL socket would be left in a CLOSE_WAIT state. I think the problem started on December 8th, with the resolution of this Jira: http://jira.freeswitch.org/browse/FS-3750 http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/event_handle rs/mod_event_socket/mod_event_socket.c?r2=19dad4a527e4e87bdbecf7b97e3d07 fd11e2a04c&r1=6bd2798ea1df47e2a5b9de99defbd79e33f5726f When I rolled the code back to the revision just prior to this change, my problems went away. git checkout 1868e145201cc6ba5a14d6929695977780917a38 Dave. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Neil Davis Sent: Friday, December 16, 2011 7:09 PM To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] Threads remain after calling close on Java client Hi, I built a web application that connects to Freeswitch using the org.freeswitch.esl.client.Client. I connect the Client object from a Spring annotated service that I call from a Spring controller. I put the connected client in my ServletContext, so I can access it later to call client.cancelEventSubscriptions() and client.close() from my ServletContextListener contextDestroyed method when Tomcat is shutting down. The problem I'm having is that even after I call close on the client, there are still a bunch of active threads that the client has spawned in the background. These threads are causing Tomcat to hang when I'm shutting down. Can anyone suggest an approach that would enable my application to disconnect the Freeswitch client when Tomcat is shutting down that would allow Tomcat to shutdown gracefully? Below are errors from my Tomcat log for the threads that I have identified as being related to the Freeswitch client. I don't know how I can get to these threads to interrupt them and Client.close() seems to leave them hanging. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-3-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-4-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-2] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-3] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-4] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-5] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-6] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-7] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-8] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-9] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-10] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-11] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-12] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-13] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-14] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-15] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-16] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.internal.ThreadLocalBoolean] (value [org.jboss.netty.util.internal.ThreadLocalBoolean at 186e192]) and a value of type [java.lang.Boolean] (value [false]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.CharsetUtil$1] (value [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type [java.util.IdentityHashMap] (value [{windows-1252=sun.nio.cs.MS1252$Encoder at 373f86}]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.internal.ThreadLocalRandom$1] (value [org.jboss.netty.util.internal.ThreadLocalRandom$1 at 12bb519]) and a value of type [org.jboss.netty.util.internal.ThreadLocalRandom] (value [org.jboss.netty.util.internal.ThreadLocalRandom at 7e9dbc]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.CharsetUtil$1] (value [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type [java.util.IdentityHashMap] (value [{windows-1252=sun.nio.cs.MS1252$Encoder at a5b041}]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Thanks, Neil Davis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111217/d59c8771/attachment-0001.html From gb10hkzo-freeswitch at yahoo.co.uk Sat Dec 17 16:54:17 2011 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob) Date: Sat, 17 Dec 2011 13:54:17 +0000 (GMT) Subject: [Freeswitch-users] Recommended ULIMIT settings problem Message-ID: <1324130057.53868.YahooMailNeo@web29405.mail.ird.yahoo.com> Hi, As per the performance docs, I am trying to set the recommended ULIMIT of 8192 stacksize on a 64 bit machine. I am using the current version of Freeswitch (as pulled off GIT yesterday, Friday). I am running Scientific Linux 6.1 : Linux 2.6.32-131.21.1.el6.x86_64 #1 SMP Tue Nov 22 14:15:09 CST 2011 x86_64 x86_64 x86_64 GNU/Linux My present ulimit settings are as follows : $ ulimit -a core file size ? ? ? ? ?(blocks, -c) unlimited data seg size ? ? ? ? ? (kbytes, -d) unlimited scheduling priority ? ? ? ? ? ? (-e) 0 file size ? ? ? ? ? ? ? (blocks, -f) unlimited pending signals ? ? ? ? ? ? ? ? (-i) unlimited max locked memory ? ? ? (kbytes, -l) unlimited max memory size ? ? ? ? (kbytes, -m) unlimited open files ? ? ? ? ? ? ? ? ? ? ?(-n) 999999 pipe size ? ? ? ? ? ?(512 bytes, -p) 8 POSIX message queues ? ? (bytes, -q) unlimited real-time priority ? ? ? ? ? ? ?(-r) 0 stack size ? ? ? ? ? ? ?(kbytes, -s) 8192 cpu time ? ? ? ? ? ? ? (seconds, -t) unlimited max user processes ? ? ? ? ? ? ?(-u) unlimited virtual memory ? ? ? ? ?(kbytes, -v) unlimited file locks ? ? ? ? ? ? ? ? ? ? ?(-x) unlimited However when I start freeswitch "bin/freeswitch -nc -nonat -nonatmap", I get the following : Error: stacksize 8192 is too large: run ulimit -s 240 from your shell before starting the application. auto-adjusting stack size for optimal performance... Surely this is incorrect, because I am running a 64 bit kernel, 240 is the recommended value for 32 bit ? From ib-freeswitch at bzsolutions.it Sat Dec 17 23:48:30 2011 From: ib-freeswitch at bzsolutions.it (ib-freeswitch at bzsolutions.it) Date: Sat, 17 Dec 2011 21:48:30 +0100 (CET) Subject: [Freeswitch-users] Some questions about FS-2828 In-Reply-To: <16904813.15571324154846388.JavaMail.javamailuser@localhost> Message-ID: <23153745.15591324154910267.JavaMail.javamailuser@localhost> Hi, I'm studing FS-2828, why freeswitch statemachine always 'remap' 180 with SDP in a 183 ? I already ported FS-2828 patch to current GIT but my patch didn't forward new SDP when 180 is received after first 183 http://pastebin.com/dqiBcYC8 My idea is to create a patch to forward all 1XX messages to calling party. Thanks Igor- From gael at kelta.net Sun Dec 18 19:01:50 2011 From: gael at kelta.net (Gael Martin) Date: Sun, 18 Dec 2011 17:01:50 +0100 Subject: [Freeswitch-users] N-tier hierarchy real time billing using FS and NibbleBill Message-ID: Hi, I need to develop a project with FreeSwitch that can bill calls in realtime not just for the end user but for all its parent account at different rate too and if one of them run out of credit the call stops. The NibbleBill module only allow to rate 2 account and 2 rates at the same time if we bill the A and B leg, this would be fine but I sometime need to bill 3 or more tier but realistically this is unlikely to go beyond 5 tiers. I tried a proof of concept by looping the call to FreeSwitch and NibbleBill each loop using it's own account and rate until there's no parent account for a user and then go out to the PSTN gateways, I cannot use the loopback system because it would remove all the middle calls once the 2 ends are connected together, so I use a sofia bridge to the local IP, (I just used a different sofia interface than the one the end user normally register into and this looping interface is not public) This is working fine and does exactly what I want, so far I've tested it with 5 tiers and as soon as one tier run out of credit it will disconnect the entire chain of call and adjust each account the correct amount. I have not tested with multiple end user calls at the same time but I expect it would work the same way. My only concern is performance and any potential issue doing some many loopback in terms of quality or timing. Looping calls in telephony system are normally reserved only as a last chance attempt to sort a problem. So I was wondering if someone had done something similar (i.e. looping calls through the same FreeSwitch instance) with no issues, and if there were anything to watch for or any tweaks to the sofia interface to make sure this will not cause an issue when scaling this system. I'm fairly new to FreeSwitch so I dont know by heart all the different options yet. Thanks. PS: I'm happy to share my proof of concept if anyone's interested doing the same. From rakib0000 at hotmail.com Mon Dec 19 08:02:23 2011 From: rakib0000 at hotmail.com (rakib 0000) Date: Mon, 19 Dec 2011 16:02:23 +1100 Subject: [Freeswitch-users] Problems in using mod_xml_curl at user authentication. Message-ID: Hello list, I'm a newbie FreeSwitch user, trying to learn how mod_xml_curl works. I've done the basics things mentioned in FreeSwitch wiki pages about using mod_xml_curl. Now my plan to use mod_xml_curl for user registration. So, what I've done is - I wrote a server end application which reads POST request from mod_xml_curl and tries to find whether it's a valid post request (REGISTER) or not. If the request is a REGISTER request - then the application extracts the user id, domain. Then, it queries the database and extracts password and creates xml response. One of my sample response are like below (the following response has been taken from /tmp):
With above response, I didn't had any luck. It fails to register, with following response: 2011-12-18 14:51:29.253474 [WARNING] sofia_reg.c:2446 Can't find user [1100 at 192.168.0.108] You must define a domain called '192.168.0.108' in your directory and add a user with the id="1100" attribute and you must configure your device to use the proper domain in it's authentication credentials. To figure it out, I've tried in different ways too. Like, I've used the above response patterns to create a user under conf/directory/default/ directory, but didn't work. Then I tried the following xml response pattern : with the above xml I manage to create a user and authenticate perfectly. But, I've modified my server end application to give a that kind of response, but didn't work though. I've copied the server response (from the content dumped under /tmp) to create an user under conf/directory/default/ directory - but it works. So, my question is what am I missing here? FreeSwitch tries to get the response from the server, it gets the response too, but somehow not interpreting the response (or something else?) and gives "can't find user" error. Any hint's , suggestion will be very helpful. (note, I'm not using any webserver here.) Thanks in advance, Rakib -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/34813b6d/attachment-0001.html From gb10hkzo-freeswitch at yahoo.co.uk Mon Dec 19 11:31:47 2011 From: gb10hkzo-freeswitch at yahoo.co.uk (Bob) Date: Mon, 19 Dec 2011 08:31:47 +0000 (GMT) Subject: [Freeswitch-users] Freeswitch ULIMIT on 64 bit Message-ID: <1324283507.61081.YahooMailNeo@web29402.mail.ird.yahoo.com> Hi, Apologies if I'm double posting this, but I have not seen my original post to list appear in the archives, so I'm assuming it got lost in cyberspace ! Hi, As per the performance docs, I am trying to set the recommended ULIMIT of 8192 stacksize on a 64 bit machine. I am using the current version of Freeswitch (as pulled off GIT yesterday, Friday). I am running Scientific Linux 6.1 : Linux 2.6.32-131.21.1.el6.x86_64 #1 SMP Tue Nov 22 14:15:09 CST 2011 x86_64 x86_64 x86_64 GNU/Linux My present ulimit settings are as follows : $ ulimit -a core file size ? ? ? ? ?(blocks, -c) unlimited data seg size ? ? ? ? ? (kbytes, -d) unlimited scheduling priority ? ? ? ? ? ? (-e) 0 file size ? ? ? ? ? ? ? (blocks, -f) unlimited pending signals ? ? ? ? ? ? ? ? (-i) unlimited max locked memory ? ? ? (kbytes, -l) unlimited max memory size ? ? ? ? (kbytes, -m) unlimited open files ? ? ? ? ? ? ? ? ? ? ?(-n) 999999 pipe size ? ? ? ? ? ?(512 bytes, -p) 8 POSIX message queues ? ? (bytes, -q) unlimited real-time priority ? ? ? ? ? ? ?(-r) 0 stack size ? ? ? ? ? ? ?(kbytes, -s) 8192 cpu time ? ? ? ? ? ? ? (seconds, -t) unlimited max user processes ? ? ? ? ? ? ?(-u) unlimited virtual memory ? ? ? ? ?(kbytes, -v) unlimited file locks ? ? ? ? ? ? ? ? ? ? ?(-x) unlimited However when I start freeswitch "bin/freeswitch -nc -nonat -nonatmap", I get the following : Error: stacksize 8192 is too large: run ulimit -s 240 from your shell before starting the application. auto-adjusting stack size for optimal performance... Surely this is incorrect, because I am running a 64 bit kernel, 240 is the recommended value for 32 bit ? From gael at kelta.net Mon Dec 19 16:10:36 2011 From: gael at kelta.net (gaelmartin) Date: Mon, 19 Dec 2011 05:10:36 -0800 (PST) Subject: [Freeswitch-users] N-tier hierarchy real time billing using FS and NibbleBill Message-ID: <1324300236422-7108105.post@n2.nabble.com> Hi, I need to develop a project with FreeSwitch that can bill calls in realtime not just for the end user but for all its parent account at different rate too and if one of them run out of credit the call stops. The NibbleBill module only allow to rate 2 account and 2 rates at the same time if we bill the A and B leg, this would be fine but I sometime need to bill 3 or more tier but realistically this is unlikely to go beyond 5 tiers. I tried a proof of concept by looping the call to FreeSwitch and NibbleBill each loop using it's own account and rate until there's no parent account for a user and then go out to the PSTN gateways, I cannot use the loopback system because it would remove all the middle calls once the 2 ends are connected together, so I use a sofia bridge to the local IP, (I just used a different sofia interface than the one the end user normally register into and this looping interface is not public) This is working fine and does exactly what I want, so far I've tested it with 5 tiers and as soon as one tier run out of credit it will disconnect the entire chain of call and adjust each account the correct amount. I have not tested with multiple end user calls at the same time but I expect it would work the same way. My only concern is performance and any potential issue doing some many loopback in terms of quality or timing. Looping calls in telephony system are normally reserved only as a last chance attempt to sort a problem. So I was wondering if someone had done something similar (i.e. looping calls through the same FreeSwitch instance) with no issues, and if there were anything to watch for or any tweaks to the sofia interface to make sure this will not cause an issue when scaling this system. I'm fairly new to FreeSwitch so I dont know by heart all the different options yet. Thanks. PS: I'm happy to share my proof of concept if anyone's interested doing the same. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/N-tier-hierarchy-real-time-billing-using-FS-and-NibbleBill-tp7108105p7108105.html Sent from the freeswitch-users mailing list archive at Nabble.com. From msc at freeswitch.org Mon Dec 19 19:27:26 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 19 Dec 2011 08:27:26 -0800 Subject: [Freeswitch-users] Weekly Conference Torrents. In-Reply-To: <01ac01ccbd0e$53a8e070$fafaa150$@com> References: <01ac01ccbd0e$53a8e070$fafaa150$@com> Message-ID: I'll get with Ray and confirm that our script is using the correct sox params. -MC On Sat, Dec 17, 2011 at 2:50 PM, Stephen Dame wrote: > Question? I have downloaded the last 4 weekly confrerecne calls from > torrents, but all the files play in slow motion sound like lion roaring > Tried all the different formats, in different players and they all are not > understandable.**** > > ** ** > > Has anyone successfully listened to them? My normal freeswitch recordings > in mp3, wav play fine in vlc, itunes, media player on win7? **** > > ** ** > > All three formats are sounding the same ogg, wav, mp3 for all 4 weeks?* > *** > > ** ** > > Thanks**** > > Stephen**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Michael > Collins > *Sent:* Thursday, December 15, 2011 11:35 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] FreeSWITCH Conference Call Today**** > > ** ** > > Correct. We decided to use torrents to consolidate our media files and use > our CDN for stuff like this.**** > > ** ** > > -MC**** > > On Wed, Dec 14, 2011 at 9:46 PM, William Suffill < > william.suffill at gmail.com> wrote:**** > > No more direct download links for the recordings? > > -- W > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/b58bd7ff/attachment.html From orbit at klbank.ru Mon Dec 19 19:54:12 2011 From: orbit at klbank.ru (Zhuravlov Sergey) Date: Mon, 19 Dec 2011 20:54:12 +0400 Subject: [Freeswitch-users] unexpectedly dial through gateway Message-ID: <20111219165411.GA9141@klbank.ru> Hi, in my dialplan I try to call directly to the asterisk server follows: But it does not work for me! If the profile external has gateways then a call goes through a gateway, and receives forbidden. If in the profile external NOT have gateways - all is good and asterisk recieve call. I do not understand why this is happening? Where to look? -- Zhuravlov Sergey GTALK/JABBER:4orbit at gmail.com From justlikeef at gmail.com Mon Dec 19 20:03:04 2011 From: justlikeef at gmail.com (Rob Hutton) Date: Mon, 19 Dec 2011 12:03:04 -0500 Subject: [Freeswitch-users] unexpectedly dial through gateway In-Reply-To: <20111219165411.GA9141@klbank.ru> References: <20111219165411.GA9141@klbank.ru> Message-ID: <201112191203.05026.justlikeef@gmail.com> Without seeing your dialplan, this is just a guess, but, I would bet that your gateway appears before this trunk in the external dialplan and the call matches the regex on the gateway. So, tighten your regex on the gateway and/or move this entry above the gateway entry. On Monday 19 December 2011 11:54:12 Zhuravlov Sergey wrote: > Hi, > > in my dialplan I try to call directly to the asterisk server follows: > > > > > > > > > > But it does not work for me! If the profile external has gateways then a call goes > through a gateway, and receives forbidden. > > If in the profile external NOT have gateways - all is good and asterisk > recieve call. > > I do not understand why this is happening? Where to look? > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/e6a6c406/attachment-0001.html From elliott at zoogmedia.com Mon Dec 19 20:01:33 2011 From: elliott at zoogmedia.com (Elliott Vogel) Date: Mon, 19 Dec 2011 17:01:33 +0000 Subject: [Freeswitch-users] destination_number cleanup Message-ID: Some custom softphone a call center we use has; they auto dial with it and I was able to make brea, x-lite and polycom phone (using uri dialing) send some messed up strings too. I was trying to see if breaks the sip specification so I could push back but I couldn't find anything what a uri couldn't use. It looks to me it can be almost anything - personally they are being lazy and not scrubbing the list first Thanks for the regex string and I will look into writing a script however I would prefer not too... From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Monday, December 19, 2011 1:08 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] destination_number cleanup Yes, this is really broken input! I thought it was bad some do +1 or 001 1 or no prefix to dial America! To modify Ryan's code to make it more flexible... ^(?:\+1|001|1)?\D*(\d{3})\D*(\d{3})\D*(\d{4})$ $1, $2 and $3 should give you the full number, so bridge to 1$1$2$3. it will match anything that has xxx-xxx-xxxx with whatever separating non-digits is in between. But it can only handle (by discarding) a prefix of +1, 1, or 001. I'd suggest using xml_curl or a lua script and doing a regex replace of any non-digit characters, if you really need. But how in the world is this your input? What devices are sending this?? -Avi On Mon, Dec 19, 2011 at 4:12 AM, Brian West > wrote: Why do you have broken devices sending you numbers in that screwed up format in the request URI in the first place? At this point you should be using XML_CURL then cleaning that up if this is what you're hitting. Or you tell your clients to fix their badly broken behavior. /b On Dec 18, 2011, at 8:08 PM, Ryan V wrote: On Mon, Dec 19, 2011 at 2:56 AM, Elliott Vogel >wrote: I was wondering if anyone has a regex expression that works to return just digest? I have some clients sending requests formatted to +1 (555) 555-5555, 555-555-5555, 555.555.5555 which aren't be processed by our dial plan because we are expecting all numbers (5555555555) **** How about this? ^\+1\s+\((\d{3})\)\s(\d{3})-(\d{4})$|^(\d{3})-(\d{3})-(\d{4})$|^(\d{3})\.(\d{3})\.(\d{4})$ $1, $2 and $3 should give you the full number. Regards, Ryan -- 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/20111219/346cb774/attachment.html From elliott at zoogmedia.com Mon Dec 19 20:39:51 2011 From: elliott at zoogmedia.com (Elliott Vogel) Date: Mon, 19 Dec 2011 17:39:51 +0000 Subject: [Freeswitch-users] PCI Compliance Over Telephone for Credit Cards- how? In-Reply-To: References: Message-ID: Well, I have worked a lot with PCI compliance in the past and I don't think you can meet the requirements of encryption if you're not doing encoding yourself because most voip service providers aren't encrypting the calls. Also dtmf has the same for requirements and for T1 not being encrypted this is true but because the network is considered secured(funny)/private it's doesn't need to be - now if you would encapsulate t1 traffic to send it over the internet without encrypting it this would be unsecured. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Monday, December 19, 2011 5:52 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] PCI Compliance Over Telephone for Credit Cards- how? I'm planning on an IVR to accept credit card information for signing up and renewal of my services. Regarding fraud, I'm going to require at minimum a recording of name, who they are, or something or an actual live call. But for PCI compliance.. this says https://www.pcisecuritystandards.org/documents/protecting_telephone-based_payment_card_data.pdf on page 9: Call centers will need to ensure that transmission of cardholder data across public networks is encrypted. This is part of PCI DSS Requirement 4 and includes: * ... * Voice or data streams over Voice over IP (VoIP) telephone systems, whenever sent over an open or public network. Note that only those consumer or enterprise VoIP systems that provide strong cryptography should be used. * Requiring agents to use analog telephone lines when a VoIP telephone system does not provide strong cryptography. I'm doing dtmf, not voice, but I can't imagine that's LESS strict. I haven't really heard of any end-to-end encrypted origination lines. Is this guideline ignored? How do people deal with this? Does someone have T1 lines and offers encryption for origination...? -Avi Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/0d4f7b4a/attachment-0001.html From anthony.minessale at gmail.com Mon Dec 19 20:41:38 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 19 Dec 2011 11:41:38 -0600 Subject: [Freeswitch-users] Freeswitch ULIMIT on 64 bit In-Reply-To: <1324283507.61081.YahooMailNeo@web29402.mail.ird.yahoo.com> References: <1324283507.61081.YahooMailNeo@web29402.mail.ird.yahoo.com> Message-ID: 240 is the recommended size in all cases. On Mon, Dec 19, 2011 at 2:31 AM, Bob wrote: > Hi, > > Apologies if I'm double posting this, but I have not seen my original post > to list appear in the archives, so I'm assuming it got lost in cyberspace ! > > Hi, > > As per the performance docs, I am trying to set the recommended ULIMIT of > 8192 stacksize on a 64 bit machine. > > I am using the current version of Freeswitch (as pulled off GIT yesterday, > Friday). > > I am running Scientific Linux 6.1 : > Linux 2.6.32-131.21.1.el6.x86_64 #1 SMP Tue Nov 22 14:15:09 CST 2011 > x86_64 x86_64 x86_64 GNU/Linux > > > My present ulimit settings are as follows : > $ ulimit -a > > core file size (blocks, -c) unlimited > data seg size (kbytes, -d) unlimited > scheduling priority (-e) 0 > file size (blocks, -f) unlimited > pending signals (-i) unlimited > max locked memory (kbytes, -l) unlimited > max memory size (kbytes, -m) unlimited > open files (-n) 999999 > pipe size (512 bytes, -p) 8 > POSIX message queues (bytes, -q) unlimited > real-time priority (-r) 0 > stack size (kbytes, -s) 8192 > cpu time (seconds, -t) unlimited > max user processes (-u) unlimited > virtual memory (kbytes, -v) unlimited > file locks (-x) unlimited > > > However when I start freeswitch "bin/freeswitch -nc -nonat -nonatmap", I > get the following : > > Error: stacksize 8192 is too large: run ulimit -s 240 from your shell > before starting the application. > auto-adjusting stack size for optimal performance... > > Surely this is incorrect, because I am running a 64 bit kernel, 240 is the > recommended value for 32 bit ? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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/20111219/6cb73d54/attachment.html From Claudio.Cavalera at italtel.it Mon Dec 19 20:53:01 2011 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Mon, 19 Dec 2011 18:53:01 +0100 Subject: [Freeswitch-users] error compiling Freeswitchin mod_spandsp Message-ID: Hello, I'm facing a compilation issue regarding mod_spandsp which could be similar to this one: http://jira.freeswitch.org/browse/FS-3473?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel#issue-tabs or this one: http://lists.freeswitch.org/pipermail/freeswitch-dev/2011-August/005159.html Here is the error on the pastebin: http://pastebin.freeswitch.org/18028 uname -ar output is Linux DBS_A6_A 2.6.9-89.ELsmp #1 SMP Mon Apr 20 10:33:05 EDT 2009 x86_64 x86_64 x86_64 GNU/Linux and I have ./configure --enable-64 has someone any hints to narrow down the problem? Thanks, Claudio From: Michael Collins Post your question to the -users list and give us the details. -MC On Mon, Dec 19, 2011 at 1:47 AM, Cavalera Claudio Luigi wrote: Hello guys, I've read about your trouble on the mailing list http://lists.freeswitch.org/pipermail/freeswitch-dev/2011-August/005159. html Unfortunately it does not come to a happy end :-\ Did you find a solution for this problem I'm also facing? :-) Kind Regards, Claudio PS: Please forgive the automatic disclaimer in the email. Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From avi at avimarcus.net Mon Dec 19 21:03:27 2011 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 19 Dec 2011 20:03:27 +0200 Subject: [Freeswitch-users] PCI Compliance Over Telephone for Credit Cards- how? In-Reply-To: References: Message-ID: So is there a provider for USA who takes T1 and encrypts it, so I can buy origination from them? -Avi On Mon, Dec 19, 2011 at 7:39 PM, Elliott Vogel wrote: > Well, I have worked a lot with PCI compliance in the past and I don?t > think you can meet the requirements of encryption if you?re not doing > encoding yourself because most voip service providers aren?t encrypting the > calls. Also dtmf has the same for requirements and for T1 not being > encrypted this is true but because the network is considered > secured(funny)/private it?s doesn?t need to be ? now if you would > encapsulate t1 traffic to send it over the internet without encrypting it > this would be unsecured.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Monday, December 19, 2011 5:52 AM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] PCI Compliance Over Telephone for Credit > Cards- how?**** > > ** ** > > I'm planning on an IVR to accept credit card information for signing up > and renewal of my services.**** > > Regarding fraud, I'm going to require at minimum a recording of name, who > they are, or something or an actual live call.**** > > ** ** > > But for PCI compliance.. this says > https://www.pcisecuritystandards.org/documents/protecting_telephone-based_payment_card_data.pdf on > page 9:**** > > Call centers will need to ensure that transmission of cardholder data > across public networks is encrypted. > This is part of PCI DSS Requirement 4 and includes:**** > > - ...**** > > > - *Voice or data streams over Voice over IP (VoIP) telephone > systems, whenever sent over an open or public network. Note that only > those consumer or enterprise VoIP systems that provide strong > cryptography should be used. ***** > > > - Requiring agents to use analog telephone lines when a VoIP > telephone system does not provide strong cryptography.**** > > I'm doing dtmf, not voice, but I can't imagine that's LESS strict.**** > > ** ** > > I haven't really heard of any end-to-end encrypted origination lines. Is > this guideline ignored? How do people deal with this? Does someone have T1 > lines and offers encryption for origination...?**** > > > **** > > -Avi Marcus**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/1f6003f2/attachment.html From Ryan at ocens.com Mon Dec 19 21:36:57 2011 From: Ryan at ocens.com (Ryan Watkins) Date: Mon, 19 Dec 2011 18:36:57 +0000 Subject: [Freeswitch-users] Issue adding a SIP Gateway In-Reply-To: <4EEDBBB2.50302@integrafin.co.uk> References: <44E5C0A9D48A3246966A4AE04692014D102A6FB5@CH1PRD0604MB109.namprd06.prod.outlook.com> <201112151954.01880.justlikeef@gmail.com> <44E5C0A9D48A3246966A4AE04692014D102A803C@CH1PRD0604MB109.namprd06.prod.outlook.com> <4EEDBBB2.50302@integrafin.co.uk> Message-ID: <44E5C0A9D48A3246966A4AE04692014D102AA3C6@CH1PRD0604MB109.namprd06.prod.outlook.com> Thanks for the reply Alex, I followed the example in the FreeSWITCH 1.0.6 book for iptel, which is as follows: (yes, I supplied my iptel username in this line) (again, I supplied my iptel password on this line) I've checked the file permissions and changed them so that all users have rwx for the iptel.org.xml file; as well as every folder up to /external for both /usr/src/freeswitch and the /opt/freeswitch paths I've also changed the iptel.org.xml files in those paths to the example that you linked from the wiki However, I'm still getting the same result.... Any other suggestions? Thanks again! From: Alex Crow [mailto:acrow at integrafin.co.uk] Sent: Sunday, December 18, 2011 2:09 AM To: FreeSWITCH Users Help Cc: Ryan Watkins Subject: Re: [Freeswitch-users] Issue adding a SIP Gateway On 16/12/11 01:05, Ryan Watkins wrote: I did run the" sofia profile external restart reloadxml" command.... It didn't load the new gateway, so that's why I tried registering the gateway specifically. Ryan, Did you follow this example: http://wiki.freeswitch.org/wiki/Provider_Configuration:_iptel and replace the usename and password with your own? Check that the permissions on the new XML file allow it to be read by the user freeswitch is running as. Also double-check your closing tags on the file. This can cause your gateway to be skipped, hence the "invalid gateway" when you try to use it. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/393e1f88/attachment-0001.html From acrow at integrafin.co.uk Mon Dec 19 22:22:24 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Mon, 19 Dec 2011 19:22:24 +0000 Subject: [Freeswitch-users] Issue adding a SIP Gateway In-Reply-To: <44E5C0A9D48A3246966A4AE04692014D102AA3C6@CH1PRD0604MB109.namprd06.prod.outlook.com> References: <44E5C0A9D48A3246966A4AE04692014D102A6FB5@CH1PRD0604MB109.namprd06.prod.outlook.com> <201112151954.01880.justlikeef@gmail.com> <44E5C0A9D48A3246966A4AE04692014D102A803C@CH1PRD0604MB109.namprd06.prod.outlook.com> <4EEDBBB2.50302@integrafin.co.uk> <44E5C0A9D48A3246966A4AE04692014D102AA3C6@CH1PRD0604MB109.namprd06.prod.outlook.com> Message-ID: <4EEF8EF0.6010207@integrafin.co.uk> OK, As long as those "funny quotes" in your last post aren't present in your XML file I can't see what is the problem. I had an issue copying dialplans from the web with these, had to go through and change them all by hand. If you shut down freeswitch, clear your logfile, and start it again, can you post the contents of /opt/freeswitch/log/freeswitch.log to the list (or to pastebin etc and link here). There must be something in the logs. BTW you should not need to change things in /usr/src/*, once you've installed that doesn't make any difference. Cheers Alex On 19/12/11 18:36, Ryan Watkins wrote: > > Thanks for the reply Alex, > > I followed the example in the FreeSWITCH 1.0.6 book for iptel, which > is as follows: > > > > > > (yes, I supplied my iptel > username in this line) > > (again, I supplied my iptel > password on this line) > > > > > > > > > > > > I've checked the file permissions and changed them so that all users > have rwx for the iptel.org.xml file; as well as every folder up to > /external for both /usr/src/freeswitch and the /opt/freeswitch paths > > I've also changed the iptel.org.xml files in those paths to the > example that you linked from the wiki > > However, I'm still getting the same result.... Any other suggestions? > > Thanks again! > > *From:*Alex Crow [mailto:acrow at integrafin.co.uk] > *Sent:* Sunday, December 18, 2011 2:09 AM > *To:* FreeSWITCH Users Help > *Cc:* Ryan Watkins > *Subject:* Re: [Freeswitch-users] Issue adding a SIP Gateway > > On 16/12/11 01:05, Ryan Watkins wrote: > > I did run the" sofia profile external restart reloadxml" command.... > It didn't load the new gateway, so that's why I tried registering the > gateway specifically. > > * > *Ryan,* > > *Did you follow this example: > http://wiki.freeswitch.org/wiki/Provider_Configuration:_iptel and > replace the usename and password with your own? > > Check that the permissions on the new XML file allow it to be read by > the user freeswitch is running as. > > Also double-check your closing tags on the file. This can cause your > gateway to be skipped, hence the "invalid gateway" when you try to use it. > > Cheers > > Alex > > -- > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > > "Transact" is operated by Integrated Financial Arrangements plc > Domain House, 5-7 Singer Street, London EC2A 4BQ > Tel: (020) 7608 4900 Fax: (020) 7608 5300 > (Registered office: as above; Registered in England and Wales under number: 3727592) > Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/cc298790/attachment.html From elliott at zoogmedia.com Mon Dec 19 22:28:20 2011 From: elliott at zoogmedia.com (Elliott Vogel) Date: Mon, 19 Dec 2011 19:28:20 +0000 Subject: [Freeswitch-users] PCI Compliance Over Telephone for Credit Cards- how? In-Reply-To: References: Message-ID: I haven't seen a company yet and I have searched - none of the big origination providers do and many of the smaller ones use the big providers - we are force to do our own encoding From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Monday, December 19, 2011 12:03 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] PCI Compliance Over Telephone for Credit Cards- how? So is there a provider for USA who takes T1 and encrypts it, so I can buy origination from them? -Avi On Mon, Dec 19, 2011 at 7:39 PM, Elliott Vogel > wrote: Well, I have worked a lot with PCI compliance in the past and I don't think you can meet the requirements of encryption if you're not doing encoding yourself because most voip service providers aren't encrypting the calls. Also dtmf has the same for requirements and for T1 not being encrypted this is true but because the network is considered secured(funny)/private it's doesn't need to be - now if you would encapsulate t1 traffic to send it over the internet without encrypting it this would be unsecured. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Monday, December 19, 2011 5:52 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] PCI Compliance Over Telephone for Credit Cards- how? I'm planning on an IVR to accept credit card information for signing up and renewal of my services. Regarding fraud, I'm going to require at minimum a recording of name, who they are, or something or an actual live call. But for PCI compliance.. this says https://www.pcisecuritystandards.org/documents/protecting_telephone-based_payment_card_data.pdf on page 9: Call centers will need to ensure that transmission of cardholder data across public networks is encrypted. This is part of PCI DSS Requirement 4 and includes: * ... * Voice or data streams over Voice over IP (VoIP) telephone systems, whenever sent over an open or public network. Note that only those consumer or enterprise VoIP systems that provide strong cryptography should be used. * Requiring agents to use analog telephone lines when a VoIP telephone system does not provide strong cryptography. I'm doing dtmf, not voice, but I can't imagine that's LESS strict. I haven't really heard of any end-to-end encrypted origination lines. Is this guideline ignored? How do people deal with this? Does someone have T1 lines and offers encryption for origination...? -Avi Marcus _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/7022bb65/attachment-0001.html From avi at avimarcus.net Mon Dec 19 22:34:08 2011 From: avi at avimarcus.net (Avi Marcus) Date: Mon, 19 Dec 2011 21:34:08 +0200 Subject: [Freeswitch-users] PCI Compliance Over Telephone for Credit Cards- how? In-Reply-To: References: Message-ID: Encrypting yourself only helps if you have a T1/BRI whatever private link to the telco. I don't.. what are my options? -Avi On Mon, Dec 19, 2011 at 9:28 PM, Elliott Vogel wrote: > I haven?t seen a company yet and I have searched ? none of the big > origination providers do and many of the smaller ones use the big providers > ? we are force to do our own encoding**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Monday, December 19, 2011 12:03 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] PCI Compliance Over Telephone for > Credit Cards- how?**** > > ** ** > > So is there a provider for USA who takes T1 and encrypts it, so I can buy > origination from them?**** > > > **** > > -Avi**** > > ** ** > > On Mon, Dec 19, 2011 at 7:39 PM, Elliott Vogel > wrote:**** > > Well, I have worked a lot with PCI compliance in the past and I don?t > think you can meet the requirements of encryption if you?re not doing > encoding yourself because most voip service providers aren?t encrypting the > calls. Also dtmf has the same for requirements and for T1 not being > encrypted this is true but because the network is considered > secured(funny)/private it?s doesn?t need to be ? now if you would > encapsulate t1 traffic to send it over the internet without encrypting it > this would be unsecured.**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > *Sent:* Monday, December 19, 2011 5:52 AM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] PCI Compliance Over Telephone for Credit > Cards- how?**** > > **** > > I'm planning on an IVR to accept credit card information for signing up > and renewal of my services.**** > > Regarding fraud, I'm going to require at minimum a recording of name, who > they are, or something or an actual live call.**** > > **** > > But for PCI compliance.. this says > https://www.pcisecuritystandards.org/documents/protecting_telephone-based_payment_card_data.pdf on > page 9:**** > > Call centers will need to ensure that transmission of cardholder data > across public networks is encrypted. > This is part of PCI DSS Requirement 4 and includes:**** > > - ...**** > > > - *Voice or data streams over Voice over IP (VoIP) telephone > systems, whenever sent over an open or public network. Note that only > those consumer or enterprise VoIP systems that provide strong > cryptography should be used. ***** > > > - Requiring agents to use analog telephone lines when a VoIP > telephone system does not provide strong cryptography.**** > > I'm doing dtmf, not voice, but I can't imagine that's LESS strict.**** > > **** > > I haven't really heard of any end-to-end encrypted origination lines. Is > this guideline ignored? How do people deal with this? Does someone have T1 > lines and offers encryption for origination...?**** > > > **** > > -Avi Marcus**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org**** > > ** ** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/432c7afe/attachment.html From msc at freeswitch.org Mon Dec 19 22:53:18 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 19 Dec 2011 11:53:18 -0800 Subject: [Freeswitch-users] announce conf count (total callers minus 1)? In-Reply-To: <1324232567.1676.YahooMailNeo@web65301.mail.ac2.yahoo.com> References: <1324232567.1676.YahooMailNeo@web65301.mail.ac2.yahoo.com> Message-ID: Try this: That's just off the top of my head, so if it doesn't work then tinker with it until you get it working. -MC On Sun, Dec 18, 2011 at 10:22 AM, Rodney wrote: > this works great, thank you to whomever created it. is there a way to make > the conf count = list count -1 so the playback will only speak total > "other" callers instead of including the person who press the option? > > > conf/dialplan/default/01_Announce_Conf_Count.xml: > > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/3cd54e67/attachment.html From vaerge at gmail.com Mon Dec 19 22:14:19 2011 From: vaerge at gmail.com (Anders) Date: Mon, 19 Dec 2011 13:14:19 -0600 Subject: [Freeswitch-users] originate via rpc Message-ID: Hi, I am trying to originate a call, using a mod command through the xml_rpc, but I am doing something wrong - not sure what it is. I post this url: http://1.2.3.4/webapi/originate?{origination_caller_id_number=14007654321}sofia/carriers/111#18001234567 at 4.5.6.7%20&playback(/vr/migstory.wav) (numbers and IPs modified here) The reponse I get is this: -USAGE: |&() [] [] [] [] [] I can do other commands directly in the browser like this (such as http://1.2.3.4/webapi/show?calls) without any problems. Therefore my conclusion is that I am doing something wrong in this syntax for http. Any ideas? Thanks in advance! //Anders From msc at freeswitch.org Mon Dec 19 22:56:50 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 19 Dec 2011 11:56:50 -0800 Subject: [Freeswitch-users] originate via rpc In-Reply-To: References: Message-ID: Don't you have to URL encode the & character? -MC On Mon, Dec 19, 2011 at 11:14 AM, Anders wrote: > Hi, > > I am trying to originate a call, using a mod command through the > xml_rpc, but I am doing something wrong - not sure what it is. I post > this url: > > > http://1.2.3.4/webapi/originate?{origination_caller_id_number=14007654321}sofia/carriers/111#18001234567 at 4.5.6.7%20&playback(/vr/migstory.wav) > > (numbers and IPs modified here) > > The reponse I get is this: -USAGE: > |&() [] [] > [] [] [] > > I can do other commands directly in the browser like this (such as > http://1.2.3.4/webapi/show?calls) without any problems. Therefore my > conclusion is that I am doing something wrong in this syntax for http. > > Any ideas? > > Thanks in advance! > > //Anders > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/a349147a/attachment-0001.html From justlikeef at gmail.com Mon Dec 19 23:03:26 2011 From: justlikeef at gmail.com (Rob Hutton) Date: Mon, 19 Dec 2011 15:03:26 -0500 Subject: [Freeswitch-users] PCI Compliance Over Telephone for Credit Cards- how? In-Reply-To: References: Message-ID: <201112191503.26786.justlikeef@gmail.com> None of the major PSTN gateway providers will do SRTP or whatever. We ran into this several months ago. Funny part about all of this is that the LECs send calls across data networks mixed with other traffic regularly, so the compliance issue involves a man behind the curtain in the first place. The only option that I know of that is PCI compliant is to have your own line from Ma Bell. On Monday 19 December 2011 14:34:08 Avi Marcus wrote: > Encrypting yourself only helps if you have a T1/BRI whatever private link > to the telco. I don't.. what are my options? > -Avi > > On Mon, Dec 19, 2011 at 9:28 PM, Elliott Vogel wrote: > > > I haven?t seen a company yet and I have searched ? none of the big > > origination providers do and many of the smaller ones use the big providers > > ? we are force to do our own encoding**** > > > > ** ** > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > > *Sent:* Monday, December 19, 2011 12:03 PM > > *To:* FreeSWITCH Users Help > > *Subject:* Re: [Freeswitch-users] PCI Compliance Over Telephone for > > Credit Cards- how?**** > > > > ** ** > > > > So is there a provider for USA who takes T1 and encrypts it, so I can buy > > origination from them?**** > > > > > > **** > > > > -Avi**** > > > > ** ** > > > > On Mon, Dec 19, 2011 at 7:39 PM, Elliott Vogel > > wrote:**** > > > > Well, I have worked a lot with PCI compliance in the past and I don?t > > think you can meet the requirements of encryption if you?re not doing > > encoding yourself because most voip service providers aren?t encrypting the > > calls. Also dtmf has the same for requirements and for T1 not being > > encrypted this is true but because the network is considered > > secured(funny)/private it?s doesn?t need to be ? now if you would > > encapsulate t1 traffic to send it over the internet without encrypting it > > this would be unsecured.**** > > > > **** > > > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Avi Marcus > > *Sent:* Monday, December 19, 2011 5:52 AM > > *To:* FreeSWITCH Users Help > > *Subject:* [Freeswitch-users] PCI Compliance Over Telephone for Credit > > Cards- how?**** > > > > **** > > > > I'm planning on an IVR to accept credit card information for signing up > > and renewal of my services.**** > > > > Regarding fraud, I'm going to require at minimum a recording of name, who > > they are, or something or an actual live call.**** > > > > **** > > > > But for PCI compliance.. this says > > https://www.pcisecuritystandards.org/documents/protecting_telephone-based_payment_card_data.pdf on > > page 9:**** > > > > Call centers will need to ensure that transmission of cardholder data > > across public networks is encrypted. > > This is part of PCI DSS Requirement 4 and includes:**** > > > > - ...**** > > > > > > - *Voice or data streams over Voice over IP (VoIP) telephone > > systems, whenever sent over an open or public network. Note that only > > those consumer or enterprise VoIP systems that provide strong > > cryptography should be used. ***** > > > > > > - Requiring agents to use analog telephone lines when a VoIP > > telephone system does not provide strong cryptography.**** > > > > I'm doing dtmf, not voice, but I can't imagine that's LESS strict.**** > > > > **** > > > > I haven't really heard of any end-to-end encrypted origination lines. Is > > this guideline ignored? How do people deal with this? Does someone have T1 > > lines and offers encryption for origination...?**** > > > > > > **** > > > > -Avi Marcus**** > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org**** > > > > ** ** > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/2adbdbfa/attachment-0001.html From dgarcia at anew.com.ve Mon Dec 19 23:04:23 2011 From: dgarcia at anew.com.ve (Saugort Dario Garcia Tovar) Date: Mon, 19 Dec 2011 15:34:23 -0430 Subject: [Freeswitch-users] PCI Compliance Over Telephone for Credit Cards- how? In-Reply-To: References: Message-ID: <4EEF98C7.5040800@anew.com.ve> Avi, You have not many options. Firtst, tell us about your architecture. Second, about TDM, there is some options but it when you use T1/E1 to transmit data; but for voice, perhaps, the only option is: no encryption. For voice, like mobile and fix phones, the technology used, it does not offer a way to do it. Exist mechanism to use tuneling and some security between sites when a private link between premises are used but it is basically use a T1/E1 data to transport voice, and it depend on equipments and providers. Third, if you have a VoIP provider, there is some options like as mention before: TLS, SRTP and SIPS. Fourth, You have to worried when you have the call in your control, "surfing" in your IVR and start to manage sensible data (PIN, account numbers, login, passwords, etc). How to encrypt/decrypt them as long the call exist: you need to use sensible data with other systems inside and outside of your organization. On 12/19/2011 3:04 PM, Avi Marcus wrote: > Encrypting yourself only helps if you have a T1/BRI whatever private > link to the telco. I don't.. what are my options? > -Avi > > On Mon, Dec 19, 2011 at 9:28 PM, Elliott Vogel > wrote: > > I haven?t seen a company yet and I have searched ? none of the big > origination providers do and many of the smaller ones use the big > providers ? we are force to do our own encoding > > *From:*freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf > Of *Avi Marcus > *Sent:* Monday, December 19, 2011 12:03 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] PCI Compliance Over Telephone > for Credit Cards- how? > > So is there a provider for USA who takes T1 and encrypts it, so I > can buy origination from them? > > > -Avi > > On Mon, Dec 19, 2011 at 7:39 PM, Elliott Vogel > > wrote: > > Well, I have worked a lot with PCI compliance in the past and I > don?t think you can meet the requirements of encryption if you?re > not doing encoding yourself because most voip service providers > aren?t encrypting the calls. Also dtmf has the same for > requirements and for T1 not being encrypted this is true but > because the network is considered secured(funny)/private it?s > doesn?t need to be ? now if you would encapsulate t1 traffic to > send it over the internet without encrypting it this would be > unsecured. > > *From:*freeswitch-users-bounces at lists.freeswitch.org > > [mailto:freeswitch-users-bounces at lists.freeswitch.org > ] *On Behalf > Of *Avi Marcus > *Sent:* Monday, December 19, 2011 5:52 AM > *To:* FreeSWITCH Users Help > *Subject:* [Freeswitch-users] PCI Compliance Over Telephone for > Credit Cards- how? > > I'm planning on an IVR to accept credit card information for > signing up and renewal of my services. > > Regarding fraud, I'm going to require at minimum a recording of > name, who they are, or something or an actual live call. > > But for PCI compliance.. this says > https://www.pcisecuritystandards.org/documents/protecting_telephone-based_payment_card_data.pdf on > page 9: > > Call centers will need to ensure that transmission of > cardholder data across public networks is encrypted. > This is part of PCI DSS Requirement 4 and includes: > > * ... > > * *Voice or data streams over Voice over IP (VoIP) telephone > systems, whenever sent over an open or public network. > Note that only those consumer or enterprise VoIP systems > that provide strong cryptography should be used. * > > * Requiring agents to use analog telephone lines when a VoIP > telephone system does not provide strong cryptography. > > I'm doing dtmf, not voice, but I can't imagine that's LESS strict. > > I haven't really heard of any end-to-end encrypted origination > lines. Is this guideline ignored? How do people deal with this? > Does someone have T1 lines and offers encryption for origination...? > > > -Avi Marcus > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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: 2012.0.1890 / Virus Database: 2108/4690 - Release Date: 12/19/11 > -- 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/20111219/f5c6ad44/attachment-0001.html From elliott at zoogmedia.com Mon Dec 19 23:14:16 2011 From: elliott at zoogmedia.com (Elliott Vogel) Date: Mon, 19 Dec 2011 20:14:16 +0000 Subject: [Freeswitch-users] PCI Compliance Over Telephone for Credit Cards- how? In-Reply-To: References: Message-ID: I think if you could get a mpls line from one of the big three and have them termite voice over it that would be PCI compliant but I bet this will cost more than getting a PRI. Another option would be seeing if someone would be will to encoding traffic and would sell you a few channels but I bet the cost per channel could be high - we estimate our cost to be around 39.00 per channel per month plus the per minute rate of .007 local and .035 for 800. I may be able to help you, email me off line. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Monday, December 19, 2011 1:34 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] PCI Compliance Over Telephone for Credit Cards- how? Encrypting yourself only helps if you have a T1/BRI whatever private link to the telco. I don't.. what are my options? -Avi On Mon, Dec 19, 2011 at 9:28 PM, Elliott Vogel > wrote: I haven't seen a company yet and I have searched - none of the big origination providers do and many of the smaller ones use the big providers - we are force to do our own encoding From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Monday, December 19, 2011 12:03 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] PCI Compliance Over Telephone for Credit Cards- how? So is there a provider for USA who takes T1 and encrypts it, so I can buy origination from them? -Avi On Mon, Dec 19, 2011 at 7:39 PM, Elliott Vogel > wrote: Well, I have worked a lot with PCI compliance in the past and I don't think you can meet the requirements of encryption if you're not doing encoding yourself because most voip service providers aren't encrypting the calls. Also dtmf has the same for requirements and for T1 not being encrypted this is true but because the network is considered secured(funny)/private it's doesn't need to be - now if you would encapsulate t1 traffic to send it over the internet without encrypting it this would be unsecured. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus Sent: Monday, December 19, 2011 5:52 AM To: FreeSWITCH Users Help Subject: [Freeswitch-users] PCI Compliance Over Telephone for Credit Cards- how? I'm planning on an IVR to accept credit card information for signing up and renewal of my services. Regarding fraud, I'm going to require at minimum a recording of name, who they are, or something or an actual live call. But for PCI compliance.. this says https://www.pcisecuritystandards.org/documents/protecting_telephone-based_payment_card_data.pdf on page 9: Call centers will need to ensure that transmission of cardholder data across public networks is encrypted. This is part of PCI DSS Requirement 4 and includes: * ... * Voice or data streams over Voice over IP (VoIP) telephone systems, whenever sent over an open or public network. Note that only those consumer or enterprise VoIP systems that provide strong cryptography should be used. * Requiring agents to use analog telephone lines when a VoIP telephone system does not provide strong cryptography. I'm doing dtmf, not voice, but I can't imagine that's LESS strict. I haven't really heard of any end-to-end encrypted origination lines. Is this guideline ignored? How do people deal with this? Does someone have T1 lines and offers encryption for origination...? -Avi Marcus _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/33f245c8/attachment.html From notlikeme75 at yahoo.com Mon Dec 19 23:15:40 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Mon, 19 Dec 2011 12:15:40 -0800 (PST) Subject: [Freeswitch-users] conf count minus 1 /invalid UTF-8 character In-Reply-To: References: Message-ID: <1324325740.69360.YahooMailNeo@web65311.mail.ac2.yahoo.com> both the following expressions work. but since i went to the new version msi this week i get the following error echo'd in console about 20 times at the time of pressing digit to initiate the call count extension: {warning} switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it marius mc ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Monday, December 19, 2011 2:57 PM Subject: FreeSWITCH-users Digest, Vol 66, Issue 103 ----- Forwarded Message ----- 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: PCI Compliance Over Telephone for Credit Cards- how? ? ? ? (Avi Marcus) ? 2. Re: announce conf count (total callers minus 1)? (Michael Collins) ? 3. originate via rpc (Anders) ? 4. Re: originate via rpc (Michael Collins) Encrypting yourself only helps if you have a T1/BRI whatever private link to the telco. I don't.. what are my options? -Avi On Mon, Dec 19, 2011 at 9:28 PM, Elliott Vogel wrote: I haven?t seen a company yet and I have searched ? none of the big origination providers do and many of the smaller ones use the big providers ? we are force to do our own encoding >? >From:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus >Sent: Monday, December 19, 2011 12:03 PM >To: FreeSWITCH Users Help >Subject: Re: [Freeswitch-users] PCI Compliance Over Telephone for Credit Cards- how? >? >So is there a provider?for USA?who takes T1 and encrypts it, so I can buy origination from them? > > >-Avi >? >On Mon, Dec 19, 2011 at 7:39 PM, Elliott Vogel wrote: >Well, I have worked a lot with PCI compliance in the past and I don?t think you can meet the requirements of encryption if you?re not doing encoding yourself because most voip service providers aren?t encrypting the calls. ?Also dtmf has the same for requirements and for T1 not being encrypted this is true but because the network is considered secured(funny)/private it?s doesn?t need to be ? now if you would encapsulate t1 traffic to send it over the internet without encrypting it this would be unsecured. >? >From:freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Avi Marcus >Sent: Monday, December 19, 2011 5:52 AM >To: FreeSWITCH Users Help >Subject: [Freeswitch-users] PCI Compliance Over Telephone for Credit Cards- how? >? >I'm planning on an IVR to accept credit card information for signing up and renewal of my services. >Regarding fraud, I'm going to require at minimum a recording of name, who they are, or something or an actual live call. >? >But for PCI compliance.. this says https://www.pcisecuritystandards.org/documents/protecting_telephone-based_payment_card_data.pdf?on page 9: >Call centers will need to ensure that transmission of cardholder data across?public networks is encrypted. >>This is part of PCI DSS Requirement 4 and includes: >> * ... >> * Voice or data streams over Voice over IP (VoIP) telephone systems,?whenever sent over an open or public network. Note that only those?consumer or enterprise VoIP systems that provide strong cryptography?should be used.? >> * Requiring agents to use analog telephone lines when a VoIP telephone?system does not provide strong cryptography. >I'm doing dtmf, not voice, but I can't imagine that's LESS strict. >? >I haven't really heard of any end-to-end encrypted origination lines. Is this guideline ignored? How do people deal with this? Does someone have T1 lines and offers encryption for origination...? > > >-Avi Marcus > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org >? >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > Try this: That's just off the top of my head, so if it doesn't work then tinker with it until you get it working. -MC On Sun, Dec 18, 2011 at 10:22 AM, Rodney wrote: this works great, thank you to whomever created it. is there a way to make the conf count = list count -1 so the playback will only speak total "other" callers instead of including the person who press the option? > > > > >conf/dialplan/default/01_Announce_Conf_Count.xml: > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > Hi, I am trying to originate a call, using a mod command through the xml_rpc, but I am doing something wrong - not sure what it is. I post this url: http://1.2.3.4/webapi/originate?{origination_caller_id_number=14007654321}sofia/carriers/111#18001234567 at 4.5.6.7%20&playback(/vr/migstory.wav) (numbers and IPs modified here) The reponse I get is this: -USAGE: |&() [] [] [] [] [] I can do other commands directly in the browser like this (such as http://1.2.3.4/webapi/show?calls) without any problems. Therefore my conclusion is that I am doing something wrong in this syntax for http. Any ideas? Thanks in advance! //Anders Don't you have to URL encode the & character? -MC On Mon, Dec 19, 2011 at 11:14 AM, Anders wrote: Hi, > >I am trying to originate a call, using a mod command through the >xml_rpc, but I am doing something wrong - not sure what it is. I post >this url: > >http://1.2.3.4/webapi/originate?{origination_caller_id_number=14007654321}sofia/carriers/111#18001234567 at 4.5.6.7%20&playback(/vr/migstory.wav) > >(numbers and IPs modified here) > >The reponse I get is this: -USAGE: >|&() [] [] >[] [] [] > >I can do other commands directly in the browser like this (such as >http://1.2.3.4/webapi/show?calls) without any problems. Therefore my >conclusion is that I am doing something wrong in this syntax for http. > >Any ideas? > >Thanks in advance! > >//Anders > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/96679314/attachment-0001.html From msc at freeswitch.org Mon Dec 19 23:23:24 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 19 Dec 2011 12:23:24 -0800 Subject: [Freeswitch-users] announce conf count (total callers minus 1)? In-Reply-To: References: <1324232567.1676.YahooMailNeo@web65301.mail.ac2.yahoo.com> Message-ID: Oh, my apologies to Mariusz - I accidentally scrolled right past his post in this thread. :) -MC On Mon, Dec 19, 2011 at 11:53 AM, Michael Collins wrote: > Try this: > > > > That's just off the top of my head, so if it doesn't work then tinker with > it until you get it working. > > -MC > > On Sun, Dec 18, 2011 at 10:22 AM, Rodney wrote: > >> this works great, thank you to whomever created it. is there a way to >> make the conf count = list count -1 so the playback will only speak total >> "other" callers instead of including the person who press the option? >> >> >> conf/dialplan/default/01_Announce_Conf_Count.xml: >> >> >> >> >> >> >> >> >> >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/a78c626a/attachment.html From wstephen80 at gmail.com Mon Dec 19 23:30:02 2011 From: wstephen80 at gmail.com (Stephen Wilde) Date: Mon, 19 Dec 2011 21:30:02 +0100 Subject: [Freeswitch-users] Originate Resulted in Error Cause: 100 [INVALID_IE_CONTENTS] Message-ID: I have a problem with a provider where the call are dropped by Freeswitch after receiving a 183 + SDP. I think that the problem is in SDP but I don't know where. The SDP that Freeswitch sends in INVITE is: v=0 o=FreeSWITCH 1324273767 1324273768 IN IP4 62.196.58.2 s=FreeSWITCH c=IN IP4 62.196.58.2 t=0 0 m=audio 52062 RTP/AVP 18 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 where the SDP that Freeswitch receives from the provider is: v=0 o=sansay-VSX 10 10 IN IP4 206.172.44.229 s=session controller c=IN IP4 206.172.44.229 t=0 0 m=audio 28364 RTP/AVP 18 150 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:150 telephone-event/8000 a=fmtp:150 0-15 a=ptime:20 After this SDP I see in the log: 2011-12-19 21:17:11.567017 [DEBUG] switch_ivr_originate.c:3364 Originate Resulted in Error Cause: 100 [INVALID_IE_CONTENTS] Can anyone give me an advice to solve this issue? Stephen -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/32800e7a/attachment.html From Ryan at ocens.com Mon Dec 19 23:38:13 2011 From: Ryan at ocens.com (Ryan Watkins) Date: Mon, 19 Dec 2011 20:38:13 +0000 Subject: [Freeswitch-users] Issue adding a SIP Gateway In-Reply-To: <4EEF8EF0.6010207@integrafin.co.uk> References: <44E5C0A9D48A3246966A4AE04692014D102A6FB5@CH1PRD0604MB109.namprd06.prod.outlook.com> <201112151954.01880.justlikeef@gmail.com> <44E5C0A9D48A3246966A4AE04692014D102A803C@CH1PRD0604MB109.namprd06.prod.outlook.com> <4EEDBBB2.50302@integrafin.co.uk> <44E5C0A9D48A3246966A4AE04692014D102AA3C6@CH1PRD0604MB109.namprd06.prod.outlook.com> <4EEF8EF0.6010207@integrafin.co.uk> Message-ID: <44E5C0A9D48A3246966A4AE04692014D102AA5E4@CH1PRD0604MB109.namprd06.prod.outlook.com> Here's the log file... thanks again Alex and all From: Alex Crow [mailto:acrow at integrafin.co.uk] Sent: Monday, December 19, 2011 11:22 AM To: FreeSWITCH Users Help Cc: Ryan Watkins Subject: Re: [Freeswitch-users] Issue adding a SIP Gateway OK, As long as those "funny quotes" in your last post aren't present in your XML file I can't see what is the problem. I had an issue copying dialplans from the web with these, had to go through and change them all by hand. If you shut down freeswitch, clear your logfile, and start it again, can you post the contents of /opt/freeswitch/log/freeswitch.log to the list (or to pastebin etc and link here). There must be something in the logs. BTW you should not need to change things in /usr/src/*, once you've installed that doesn't make any difference. Cheers Alex On 19/12/11 18:36, Ryan Watkins wrote: Thanks for the reply Alex, I followed the example in the FreeSWITCH 1.0.6 book for iptel, which is as follows: (yes, I supplied my iptel username in this line) (again, I supplied my iptel password on this line) I've checked the file permissions and changed them so that all users have rwx for the iptel.org.xml file; as well as every folder up to /external for both /usr/src/freeswitch and the /opt/freeswitch paths I've also changed the iptel.org.xml files in those paths to the example that you linked from the wiki However, I'm still getting the same result.... Any other suggestions? Thanks again! From: Alex Crow [mailto:acrow at integrafin.co.uk] Sent: Sunday, December 18, 2011 2:09 AM To: FreeSWITCH Users Help Cc: Ryan Watkins Subject: Re: [Freeswitch-users] Issue adding a SIP Gateway On 16/12/11 01:05, Ryan Watkins wrote: I did run the" sofia profile external restart reloadxml" command.... It didn't load the new gateway, so that's why I tried registering the gateway specifically. Ryan, Did you follow this example: http://wiki.freeswitch.org/wiki/Provider_Configuration:_iptel and replace the usename and password with your own? Check that the permissions on the new XML file allow it to be read by the user freeswitch is running as. Also double-check your closing tags on the file. This can cause your gateway to be skipped, hence the "invalid gateway" when you try to use it. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/5f2e8e6d/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: freeswitch.log Type: application/octet-stream Size: 78725 bytes Desc: freeswitch.log Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/5f2e8e6d/attachment-0001.obj From acrow at integrafin.co.uk Mon Dec 19 23:51:09 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Mon, 19 Dec 2011 20:51:09 +0000 Subject: [Freeswitch-users] Issue adding a SIP Gateway In-Reply-To: <44E5C0A9D48A3246966A4AE04692014D102AA5E4@CH1PRD0604MB109.namprd06.prod.outlook.com> References: <44E5C0A9D48A3246966A4AE04692014D102A6FB5@CH1PRD0604MB109.namprd06.prod.outlook.com> <201112151954.01880.justlikeef@gmail.com> <44E5C0A9D48A3246966A4AE04692014D102A803C@CH1PRD0604MB109.namprd06.prod.outlook.com> <4EEDBBB2.50302@integrafin.co.uk> <44E5C0A9D48A3246966A4AE04692014D102AA3C6@CH1PRD0604MB109.namprd06.prod.outlook.com> <4EEF8EF0.6010207@integrafin.co.uk> <44E5C0A9D48A3246966A4AE04692014D102AA5E4@CH1PRD0604MB109.namprd06.prod.outlook.com> Message-ID: <4EEFA3BD.5010201@integrafin.co.uk> Ryan, I see this: 2011-12-19 12:28:30.388149 [DEBUG] sofia.c:1736 Launching worker thread for external 2011-12-19 12:28:30.388202 [ERR] sofia.c:2490 ERROR: password param is REQUIRED! This could actually be from your "example.com" gateway, it is indeed an error relating to gateway defs. I suggest you delete that gateway or move it to example.com.noload (or the like). It could be stopping your proper gateway from loading. I'm sure you don't need it anyway. Also check, if you've edited your external.xml file that you haven't got messed up tags before the include of external/*.xml. Cheers Alex On 19/12/11 20:38, Ryan Watkins wrote: > > Here's the log file... thanks again Alex and all > > *From:*Alex Crow [mailto:acrow at integrafin.co.uk] > *Sent:* Monday, December 19, 2011 11:22 AM > *To:* FreeSWITCH Users Help > *Cc:* Ryan Watkins > *Subject:* Re: [Freeswitch-users] Issue adding a SIP Gateway > > OK, > > As long as those "funny quotes" in your last post aren't present in > your XML file I can't see what is the problem. I had an issue copying > dialplans from the web with these, had to go through and change them > all by hand. > > If you shut down freeswitch, clear your logfile, and start it again, > can you post the contents of /opt/freeswitch/log/freeswitch.log to the > list (or to pastebin etc and link here). > > There must be something in the logs. > > BTW you should not need to change things in /usr/src/*, once you've > installed that doesn't make any difference. > > Cheers > > Alex > > On 19/12/11 18:36, Ryan Watkins wrote: > > Thanks for the reply Alex, > > I followed the example in the FreeSWITCH 1.0.6 book for iptel, which > is as follows: > > > > > > (yes, I supplied my iptel > username in this line) > > (again, I supplied my iptel > password on this line) > > > > > > > > > > > > I've checked the file permissions and changed them so that all users > have rwx for the iptel.org.xml file; as well as every folder up to > /external for both /usr/src/freeswitch and the /opt/freeswitch paths > > I've also changed the iptel.org.xml files in those paths to the > example that you linked from the wiki > > However, I'm still getting the same result.... Any other suggestions? > > Thanks again! > > *From:*Alex Crow [mailto:acrow at integrafin.co.uk] > *Sent:* Sunday, December 18, 2011 2:09 AM > *To:* FreeSWITCH Users Help > *Cc:* Ryan Watkins > *Subject:* Re: [Freeswitch-users] Issue adding a SIP Gateway > > On 16/12/11 01:05, Ryan Watkins wrote: > > I did run the" sofia profile external restart reloadxml" command.... > It didn't load the new gateway, so that's why I tried registering the > gateway specifically. > > * > *Ryan,* > > *Did you follow this example: > http://wiki.freeswitch.org/wiki/Provider_Configuration:_iptel and > replace the usename and password with your own? > > Check that the permissions on the new XML file allow it to be read by > the user freeswitch is running as. > > Also double-check your closing tags on the file. This can cause your > gateway to be skipped, hence the "invalid gateway" when you try to use it. > > Cheers > > Alex > > > -- > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > > "Transact" is operated by Integrated Financial Arrangements plc > Domain House, 5-7 Singer Street, London EC2A 4BQ > Tel: (020) 7608 4900 Fax: (020) 7608 5300 > (Registered office: as above; Registered in England and Wales under number: 3727592) > Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) > > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/c38aec54/attachment.html From Ryan at ocens.com Tue Dec 20 00:05:43 2011 From: Ryan at ocens.com (Ryan Watkins) Date: Mon, 19 Dec 2011 21:05:43 +0000 Subject: [Freeswitch-users] Issue adding a SIP Gateway In-Reply-To: <4EEFA3BD.5010201@integrafin.co.uk> References: <44E5C0A9D48A3246966A4AE04692014D102A6FB5@CH1PRD0604MB109.namprd06.prod.outlook.com> <201112151954.01880.justlikeef@gmail.com> <44E5C0A9D48A3246966A4AE04692014D102A803C@CH1PRD0604MB109.namprd06.prod.outlook.com> <4EEDBBB2.50302@integrafin.co.uk> <44E5C0A9D48A3246966A4AE04692014D102AA3C6@CH1PRD0604MB109.namprd06.prod.outlook.com> <4EEF8EF0.6010207@integrafin.co.uk> <44E5C0A9D48A3246966A4AE04692014D102AA5E4@CH1PRD0604MB109.namprd06.prod.outlook.com> <4EEFA3BD.5010201@integrafin.co.uk> Message-ID: <44E5C0A9D48A3246966A4AE04692014D102AA623@CH1PRD0604MB109.namprd06.prod.outlook.com> Alex, I've seen that as well... but where I figured it was part of the example.xml gateway I didn't pay much mind. I have, however, renamed the example.xml to example.xml.noload, stopped and started the daemon, and attempted to reload the external profile within freeswitch... and it's STILL loading the example gateway... and will not load iptel, I haven't ever touched the external.xml, but I'm wondering if maybe it needed to be edited? Or perhaps the freeswitch install put yet another location in that this setup that it might be looking to?? Really lost at this point... don't know how it's loading the example gateway when it's been renamed (even renamed the example.xml in /usr/src/freeswitch* for S&Gs) From: Alex Crow [mailto:acrow at integrafin.co.uk] Sent: Monday, December 19, 2011 12:51 PM To: Ryan Watkins Cc: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Issue adding a SIP Gateway Ryan, I see this: 2011-12-19 12:28:30.388149 [DEBUG] sofia.c:1736 Launching worker thread for external 2011-12-19 12:28:30.388202 [ERR] sofia.c:2490 ERROR: password param is REQUIRED! This could actually be from your "example.com" gateway, it is indeed an error relating to gateway defs. I suggest you delete that gateway or move it to example.com.noload (or the like). It could be stopping your proper gateway from loading. I'm sure you don't need it anyway. Also check, if you've edited your external.xml file that you haven't got messed up tags before the include of external/*.xml. Cheers Alex On 19/12/11 20:38, Ryan Watkins wrote: Here's the log file... thanks again Alex and all From: Alex Crow [mailto:acrow at integrafin.co.uk] Sent: Monday, December 19, 2011 11:22 AM To: FreeSWITCH Users Help Cc: Ryan Watkins Subject: Re: [Freeswitch-users] Issue adding a SIP Gateway OK, As long as those "funny quotes" in your last post aren't present in your XML file I can't see what is the problem. I had an issue copying dialplans from the web with these, had to go through and change them all by hand. If you shut down freeswitch, clear your logfile, and start it again, can you post the contents of /opt/freeswitch/log/freeswitch.log to the list (or to pastebin etc and link here). There must be something in the logs. BTW you should not need to change things in /usr/src/*, once you've installed that doesn't make any difference. Cheers Alex On 19/12/11 18:36, Ryan Watkins wrote: Thanks for the reply Alex, I followed the example in the FreeSWITCH 1.0.6 book for iptel, which is as follows: (yes, I supplied my iptel username in this line) (again, I supplied my iptel password on this line) I've checked the file permissions and changed them so that all users have rwx for the iptel.org.xml file; as well as every folder up to /external for both /usr/src/freeswitch and the /opt/freeswitch paths I've also changed the iptel.org.xml files in those paths to the example that you linked from the wiki However, I'm still getting the same result.... Any other suggestions? Thanks again! From: Alex Crow [mailto:acrow at integrafin.co.uk] Sent: Sunday, December 18, 2011 2:09 AM To: FreeSWITCH Users Help Cc: Ryan Watkins Subject: Re: [Freeswitch-users] Issue adding a SIP Gateway On 16/12/11 01:05, Ryan Watkins wrote: I did run the" sofia profile external restart reloadxml" command.... It didn't load the new gateway, so that's why I tried registering the gateway specifically. Ryan, Did you follow this example: http://wiki.freeswitch.org/wiki/Provider_Configuration:_iptel and replace the usename and password with your own? Check that the permissions on the new XML file allow it to be read by the user freeswitch is running as. Also double-check your closing tags on the file. This can cause your gateway to be skipped, hence the "invalid gateway" when you try to use it. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/d14259c7/attachment-0001.html From ira at connectmevoice.com Tue Dec 20 00:09:47 2011 From: ira at connectmevoice.com (Ira Tessler) Date: Mon, 19 Dec 2011 16:09:47 -0500 Subject: [Freeswitch-users] Compact Sip Headers Message-ID: <1f358b8fdef7b855689d20f6bb7b1758@mail.gmail.com> Is Freeswitch compatible with compact sip headers? If so, is there a parameter I have to set on the external profile to get them to work? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/1c179921/attachment.html From vaerge at gmail.com Tue Dec 20 00:22:28 2011 From: vaerge at gmail.com (Anders) Date: Mon, 19 Dec 2011 15:22:28 -0600 Subject: [Freeswitch-users] originate via rpc In-Reply-To: References: Message-ID: You were right - thanks a lot. I did URL encoding on the whole thing and it (almost) worked. For some reason it decided that the space before '&' should be encoded to + but I changed that manually to %20, and it worked. Now, I just need to understand how I can originate a call to use the lcr and not a specific carrier...pointers? Thanks! On Mon, Dec 19, 2011 at 1:56 PM, Michael Collins wrote: > Don't you have to URL encode the & character? > -MC > > On Mon, Dec 19, 2011 at 11:14 AM, Anders wrote: >> >> Hi, >> >> I am trying to originate a call, using a mod command through the >> xml_rpc, but I am doing something wrong - not sure what it is. I post >> this url: >> >> >> http://1.2.3.4/webapi/originate?{origination_caller_id_number=14007654321}sofia/carriers/111#18001234567 at 4.5.6.7%20&playback(/vr/migstory.wav) >> >> (numbers and IPs modified here) >> >> The reponse I get is this: -USAGE: >> |&() [] [] >> [] [] [] >> >> I can do other commands directly in the browser like this (such as >> http://1.2.3.4/webapi/show?calls) without any problems. Therefore my >> conclusion is that I am doing something wrong in this syntax for http. >> >> Any ideas? >> >> Thanks in advance! >> >> //Anders >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From kris at kriskinc.com Tue Dec 20 01:26:45 2011 From: kris at kriskinc.com (Kristian Kielhofner) Date: Mon, 19 Dec 2011 17:26:45 -0500 Subject: [Freeswitch-users] Originate Resulted in Error Cause: 100 [INVALID_IE_CONTENTS] In-Reply-To: References: Message-ID: Export verbose_sdp=true before you bridge: http://wiki.freeswitch.org/wiki/Variable_verbose_sdp ...and set supress-cng to true on the profile. On Mon, Dec 19, 2011 at 3:30 PM, Stephen Wilde wrote: > I have a problem with a provider where the call are dropped by Freeswitch > after receiving a 183 + SDP. > I think that the problem is in SDP but I don't know where. > The SDP that Freeswitch sends in INVITE is: > > ? ?v=0 > ? ?o=FreeSWITCH 1324273767 1324273768 IN IP4 62.196.58.2 > ? ?s=FreeSWITCH > ? ?c=IN IP4 62.196.58.2 > ? ?t=0 0 > ? ?m=audio 52062 RTP/AVP 18 101 13 > ? ?a=rtpmap:101 telephone-event/8000 > ? ?a=fmtp:101 0-16 > ? ?a=ptime:20 > > > where the SDP that Freeswitch receives from the provider is: > > ? ?v=0 > ? ?o=sansay-VSX 10 10 IN IP4 206.172.44.229 > ? ?s=session controller > ? ?c=IN IP4 206.172.44.229 > ? ?t=0 0 > ? ?m=audio 28364 RTP/AVP 18 150 > ? ?a=rtpmap:18 G729/8000 > ? ?a=fmtp:18 annexb=no > ? ?a=rtpmap:150 telephone-event/8000 > ? ?a=fmtp:150 0-15 > ? ?a=ptime:20 > > > After this SDP I see in the log: > > 2011-12-19 21:17:11.567017 [DEBUG] switch_ivr_originate.c:3364 Originate > Resulted in Error Cause: 100 [INVALID_IE_CONTENTS] > > Can anyone give me an advice to solve this issue? > > Stephen > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Kristian Kielhofner From anthony.minessale at gmail.com Tue Dec 20 02:01:43 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 19 Dec 2011 17:01:43 -0600 Subject: [Freeswitch-users] Threads remain after calling close on Java client In-Reply-To: <009DEE08474F5246848F1B355FA1C380010A7F5A@mail2.patlive.local> References: <8b41de351c0d1365e3786e7a60645275@mail.gmail.com> <009DEE08474F5246848F1B355FA1C380010A7F5A@mail2.patlive.local> Message-ID: Dave, This patch only would come into play if you were sending the linger command over the socket once connected. Before the patch, if you executed "linger" it would never disconnect the socket until you did it yourself from the client side. After the patch it should apply a default timeout of 600 seconds or allow you to specify a smaller one such as "linger 20" In any case this would be moot if you manually call the disconnect yourself before you close your script. Are you sure this patch is causing you problems? This really should go in a JIRA if so, at least re-open the existing one that added the patch and post your findings. Also the guy with the local mods to the java stuff, please contribute them and we can add them to tree. On Sat, Dec 17, 2011 at 6:26 AM, Dave May wrote: > I experienced similar problems when load testing Plivo on the latest Git, > but didn't feel like I had enough data gathered for a "proper" report. > After each call, the remote ESL socket would be left in a CLOSE_WAIT state. > **** > > ** ** > > I think the problem started on December 8th, with the resolution of this > Jira:**** > > ** ** > > http://jira.freeswitch.org/browse/FS-3750**** > > ** ** > > > http://fisheye.freeswitch.org/browse/freeswitch.git/src/mod/event_handlers/mod_event_socket/mod_event_socket.c?r2=19dad4a527e4e87bdbecf7b97e3d07fd11e2a04c&r1=6bd2798ea1df47e2a5b9de99defbd79e33f5726f > **** > > ** ** > > When I rolled the code back to the revision just prior to this change, my > problems went away.**** > > ** ** > > git checkout 1868e145201cc6ba5a14d6929695977780917a38**** > > ** ** > > Dave.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Neil Davis > *Sent:* Friday, December 16, 2011 7:09 PM > *To:* FreeSWITCH-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Threads remain after calling close on Java > client**** > > ** ** > > Hi,**** > > **** > > I built a web application that connects to Freeswitch using the > org.freeswitch.esl.client.Client. I connect the Client object from a > Spring annotated service that I call from a Spring controller. I put the > connected client in my ServletContext, so I can access it later to call > client.cancelEventSubscriptions() and client.close() from my > ServletContextListener contextDestroyed method when Tomcat is shutting down. > **** > > **** > > The problem I'm having is that even after I call close on the client, > there are still a bunch of active threads that the client has spawned in > the background. These threads are causing Tomcat to hang when I'm shutting > down. Can anyone suggest an approach that would enable my application to > disconnect the Freeswitch client when Tomcat is shutting down that would > allow Tomcat to shutdown gracefully?**** > > **** > > Below are errors from my Tomcat log for the threads that I have identified > as being related to the Freeswitch client. I don't know how I can get to > these threads to interrupt them and Client.close() seems to leave them > hanging.**** > > **** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-1] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-3-thread-1] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-4-thread-1] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-2] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-3] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-4] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-5] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-6] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-7] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-8] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-9] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-10] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-11] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-12] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-13] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-14] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-15] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-16] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > checkThreadLocalMapForLeaks**** > > SEVERE: The web application [/socketspy] created a ThreadLocal with key of > type [org.jboss.netty.util.internal.ThreadLocalBoolean] (value > [org.jboss.netty.util.internal.ThreadLocalBoolean at 186e192]) and a value > of type [java.lang.Boolean] (value [false]) but failed to remove it when > the web application was stopped. Threads are going to be renewed over time > to try and avoid a probable memory leak. **** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > checkThreadLocalMapForLeaks**** > > SEVERE: The web application [/socketspy] created a ThreadLocal with key of > type [org.jboss.netty.util.CharsetUtil$1] (value > [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type > [java.util.IdentityHashMap] (value > [{windows-1252=sun.nio.cs.MS1252$Encoder at 373f86}]) but failed to remove > it when the web application was stopped. Threads are going to be renewed > over time to try and avoid a probable memory leak. **** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > checkThreadLocalMapForLeaks**** > > SEVERE: The web application [/socketspy] created a ThreadLocal with key of > type [org.jboss.netty.util.internal.ThreadLocalRandom$1] (value > [org.jboss.netty.util.internal.ThreadLocalRandom$1 at 12bb519]) and a value > of type [org.jboss.netty.util.internal.ThreadLocalRandom] (value > [org.jboss.netty.util.internal.ThreadLocalRandom at 7e9dbc]) but failed to > remove it when the web application was stopped. Threads are going to be > renewed over time to try and avoid a probable memory leak. **** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > checkThreadLocalMapForLeaks**** > > SEVERE: The web application [/socketspy] created a ThreadLocal with key of > type [org.jboss.netty.util.CharsetUtil$1] (value > [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type > [java.util.IdentityHashMap] (value > [{windows-1252=sun.nio.cs.MS1252$Encoder at a5b041}]) but failed to remove > it when the web application was stopped. Threads are going to be renewed > over time to try and avoid a probable memory leak. **** > > **** > > **** > > Thanks,**** > > **** > > Neil Davis**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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/20111219/dd834df8/attachment-0001.html From jeff at jefflenk.com Tue Dec 20 02:09:11 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 19 Dec 2011 15:09:11 -0800 (PST) Subject: [Freeswitch-users] conf count minus 1 /invalid UTF-8 character In-Reply-To: <1324325740.69360.YahooMailNeo@web65311.mail.ac2.yahoo.com> References: <1324325740.69360.YahooMailNeo@web65311.mail.ac2.yahoo.com> Message-ID: <1324336151927-7110155.post@n2.nabble.com> moc do you know what the problem is here? this is your submission for FS-2081 -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/conf-count-minus-1-invalid-UTF-8-character-tp7109625p7110155.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ib-freeswitch at bzsolutions.it Tue Dec 20 02:46:17 2011 From: ib-freeswitch at bzsolutions.it (ib-freeswitch at bzsolutions.it) Date: Tue, 20 Dec 2011 00:46:17 +0100 (CET) Subject: [Freeswitch-users] FS-2828 In-Reply-To: <30279706.16221324338322602.JavaMail.javamailuser@localhost> Message-ID: <17436676.16241324338377799.JavaMail.javamailuser@localhost> Hi, I'm studing FS-2828, why freeswitch statemachine always rewrite a 180 with SDP in a 183 ? I already ported (it's a test and it must be cleaned from my debug logs) FS-2828 patch to current GIT but my patch didn't forward new SDP when 180 is received after first 183 http://pastebin.com/dqiBcYC8 Now I'm working also to forward same 18X to ALEG, i don't want to touch FS state machine i'm working exporting channels variables from BLEG to ALEG, i create a function: void sofia_glue_pass_18X(private_object_t *tech_pvt, int forward_18X, int forward_180_sdp); if (sofia_use_soa(tech_pvt)) { nua_respond(tech_pvt->nh, switch_true(switch_channel_get_variable(channel, "forward_180_sdp")) ? 180 : 183, switch_true(switch_channel_get_variable(channel, "forward_180_sdp")) ? "RINGING" : "PROGRESS", //SIP_183_PROGRESS Is it the right way? Thanks Igor- From steveu at coppice.org Tue Dec 20 02:46:52 2011 From: steveu at coppice.org (Steve Underwood) Date: Tue, 20 Dec 2011 07:46:52 +0800 Subject: [Freeswitch-users] error compiling Freeswitchin mod_spandsp In-Reply-To: References: Message-ID: <4EEFCCEC.10507@coppice.org> On 12/20/2011 01:53 AM, Cavalera Claudio Luigi wrote: > Hello, > I'm facing a compilation issue regarding mod_spandsp which could be similar to this one: > > http://jira.freeswitch.org/browse/FS-3473?page=com.atlassian.jira.plugin.system.issuetabpanels%3Aall-tabpanel#issue-tabs > > or this one: > http://lists.freeswitch.org/pipermail/freeswitch-dev/2011-August/005159.html > > Here is the error on the pastebin: > http://pastebin.freeswitch.org/18028 > > uname -ar output is > Linux DBS_A6_A 2.6.9-89.ELsmp #1 SMP Mon Apr 20 10:33:05 EDT 2009 x86_64 x86_64 x86_64 GNU/Linux > and I have ./configure --enable-64 > > has someone any hints to narrow down the problem? > > Thanks, > Claudio That's a pretty old kernel, so I guess you are using a pretty old version of the compiler. There have been a couple of issues over the years where the compiler ran out of registers to allocate in the embedded assembly language, and compilation failed. I haven't seen that for quite a while, though. Because the error refers to a line in a temporary file, at the time the output of the compiler is assembled, its hard to relate to the line number specified. Finding the cause of your error will probably require inspecting that temporary file in more detail. Steve From anthony.minessale at gmail.com Tue Dec 20 06:00:15 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 19 Dec 2011 21:00:15 -0600 Subject: [Freeswitch-users] Compact Sip Headers In-Reply-To: <1f358b8fdef7b855689d20f6bb7b1758@mail.gmail.com> References: <1f358b8fdef7b855689d20f6bb7b1758@mail.gmail.com> Message-ID: yes, Profile param: enable-compact-headers=true On Mon, Dec 19, 2011 at 3:09 PM, Ira Tessler wrote: > Is Freeswitch compatible with compact sip headers? If so, is there a > parameter I have to set on the external profile to get them to work? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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/20111219/3dbf7f5b/attachment.html From justlikeef at gmail.com Tue Dec 20 07:35:04 2011 From: justlikeef at gmail.com (Rob Hutton) Date: Mon, 19 Dec 2011 23:35:04 -0500 Subject: [Freeswitch-users] Paging and Intercom Message-ID: <201112192335.04932.justlikeef@gmail.com> How is everyone doing intercom and paging? We are having problems with Grandstream phones where they put an active call on hold to answer the second call if they see the "auto answer" headers. Do all brands behave this way or is this something unique to Grandstream? I was thinking earlyier about handling this from the switch, but since current phones allow you to log in to multiple accounts even across multiple switches, it would be impossible to do this from the switch level... -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111219/41cb2de2/attachment.html From yehavi.bourvine at gmail.com Tue Dec 20 09:35:08 2011 From: yehavi.bourvine at gmail.com (Yehavi Bourvine) Date: Tue, 20 Dec 2011 08:35:08 +0200 Subject: [Freeswitch-users] Extending the field size of core's db_data table? Message-ID: Hello, We use the DB api for storing and caching various data. We need to store data that is longer than 255 characters, which is the current limit on the field size there. The backend for this API is MySQL (via ODBC). Can I just increase the field size in MySQL, or is there some dependency in FreeSwitch on this size? Thanks, __Yehavi: -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111220/0dd1be42/attachment.html From Claudio.Cavalera at italtel.it Tue Dec 20 10:56:47 2011 From: Claudio.Cavalera at italtel.it (Cavalera Claudio Luigi) Date: Tue, 20 Dec 2011 08:56:47 +0100 Subject: [Freeswitch-users] error compiling Freeswitchin mod_spandsp In-Reply-To: <4EEFCCEC.10507@coppice.org> References: <4EEFCCEC.10507@coppice.org> Message-ID: > -----Original Message----- > That's a pretty old kernel, so I guess you are using a pretty old > version of the compiler. There have been a couple of issues over the > years where the compiler ran out of registers to allocate in the > embedded assembly language, and compilation failed. I haven't seen that > for quite a while, though. Because the error refers to a line in a > temporary file, at the time the output of the compiler is assembled, > its > hard to relate to the line number specified. Finding the cause of your > error will probably require inspecting that temporary file in more > detail. > > Steve Thank you for the hints Steve. For the records this is my environment at the moment: gcc --version gcc (GCC) 3.4.6 20060404 (Red Hat 3.4.6-11) Copyright (C) 2006 Free Software Foundation, Inc. This is free software; see the source for copying conditions. There is NO warranty; not even for MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. cat /etc/redhat-release Red Hat Enterprise Linux ES release 4 (Nahant Update 8) Regards, Claudio Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- From orbit at klbank.ru Tue Dec 20 12:03:17 2011 From: orbit at klbank.ru (Zhuravlov Sergey) Date: Tue, 20 Dec 2011 13:03:17 +0400 Subject: [Freeswitch-users] unexpectedly dial through gateway In-Reply-To: <201112191203.05026.justlikeef@gmail.com> References: <20111219165411.GA9141@klbank.ru> <201112191203.05026.justlikeef@gmail.com> Message-ID: <20111220090317.GB4367@klbank.ru> Thank you! I was delighted and thought, as I did not think to myself! Indeed found earlier definition of gateways, fixed regexp. However, I realized that the call is routed through the gateway, which I only use for incoming calls, outgoing is no any rules or regexps, which would allow the call to go through it. Mystery! And most importantly, that when turned off via the CLI this particular gateway - everything works as expected. Just turn on again the call goes through it for some reason! Maybe you can somehow increase the debug level for routing? yy.yyy.yy.yy -- FS xx.xxx.xx.xx -- Asterisk Example call 1 - gateway with problem is off - the call goes as it should EXECUTE sofia/internal/271 at intra.mydom.com hash(insert/intra.mydom.com-last_dial/271/711) EXECUTE sofia/internal/271 at intra.mydom.com hash(insert/intra.mydom.com-last_dial/global/042c1036-2a81-11e1-9963-ab2ff8b5cdf8) EXECUTE sofia/internal/271 at intra.mydom.com set(RFC2822_DATE=Mon, 19 Dec 2011 22:35:40 +0200) 2011-12-19 22:35:40.694311 [DEBUG] mod_dptools.c:1063 sofia/internal/271 at intra.mydom.com SET [RFC2822_DATE]=[Mon, 19 Dec 2011 2 2:35:40 +0200] EXECUTE sofia/internal/271 at intra.mydom.com set(hangup_after_bridge=true) 2011-12-19 22:35:40.694311 [DEBUG] mod_dptools.c:1063 sofia/internal/271 at intra.mydom.com SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/271 at intra.mydom.com bridge(sofia/external/711 at xx.xxx.xx.xx) 2011-12-19 22:35:40.694311 [DEBUG] switch_ivr_originate.c:1869 Parsing global variables 2011-12-19 22:35:40.694311 [NOTICE] switch_channel.c:816 New Channel sofia/external/711 at xx.xxx.xx.xx [042d9b5e-2a81-11e1-9964-a b2ff8b5cdf8] 2011-12-19 22:35:40.694311 [DEBUG] mod_sofia.c:4311 (sofia/external/711 at xx.xxx.xx.xx) State Change CS_NEW -> CS_INIT 2011-12-19 22:35:40.694311 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/711 at xx.xxx.xx.xx [BREAK] 2011-12-19 22:35:40.694311 [DEBUG] switch_core_state_machine.c:325 (sofia/external/711 at xx.xxx.xx.xx) Running State Change CS_IN IT 2011-12-19 22:35:40.694311 [DEBUG] switch_core_state_machine.c:361 (sofia/external/711 at xx.xxx.xx.xx) State INIT 2011-12-19 22:35:40.694311 [DEBUG] mod_sofia.c:85 sofia/external/711 at xx.xxx.xx.xx SOFIA INIT send 983 bytes to udp/[xx.xxx.xx.xx]:5060 at 20:35:40.708576: ------------------------------------------------------------------------ INVITE sip:711 at xx.xxx.xx.xx SIP/2.0 Via: SIP/2.0/UDP yy.yyy.yy.yy:5080;rport;branch=z9hG4bKF00gejSj294yr Max-Forwards: 69 From: "NTCOM" ;tag=tt3ev7SFNK53e To: Call-ID: db8c0ed3-a523-122f-b690-bf4d9e35213b CSeq: 21827406 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4480abc 2011-05-27 21-46-28 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Example Call 2 - Gateway is turned on and the call goes through it -- it's WRONG EXECUTE sofia/internal/271 at intra.mydom.com hash(insert/intra.mydom.com-last_dial/271/711) EXECUTE sofia/internal/271 at intra.mydom.com hash(insert/intra.mydom.com-last_dial/global/b27a7a7e-2a81-11e1-99a5-ab2ff8b5cdf8) EXECUTE sofia/internal/271 at intra.mydom.com set(RFC2822_DATE=Mon, 19 Dec 2011 22:40:33 +0200) 2011-12-19 22:40:33.136251 [DEBUG] mod_dptools.c:1063 sofia/internal/271 at intra.mydom.com SET [RFC2822_DATE]=[Mon, 19 Dec 2011 2 2:40:33 +0200] EXECUTE sofia/internal/271 at intra.mydom.com set(hangup_after_bridge=true) 2011-12-19 22:40:33.136251 [DEBUG] mod_dptools.c:1063 sofia/internal/271 at intra.mydom.com SET [hangup_after_bridge]=[true] EXECUTE sofia/internal/271 at intra.mydom.com bridge(sofia/external/711 at xx.xxx.xx.xx) 2011-12-19 22:40:33.136251 [DEBUG] switch_ivr_originate.c:1869 Parsing global variables 2011-12-19 22:40:33.136251 [NOTICE] switch_channel.c:816 New Channel sofia/external/711 at xx.xxx.xx.xx [b27bc0dc-2a81-11e1-99a6-a b2ff8b5cdf8] 2011-12-19 22:40:33.136251 [DEBUG] mod_sofia.c:4311 (sofia/external/711 at xx.xxx.xx.xx) State Change CS_NEW -> CS_INIT 2011-12-19 22:40:33.136251 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/711 at xx.xxx.xx.xx [BREAK] 2011-12-19 22:40:33.136251 [DEBUG] switch_core_state_machine.c:325 (sofia/external/711 at xx.xxx.xx.xx) Running State Change CS_IN IT 2011-12-19 22:40:33.136251 [DEBUG] switch_core_state_machine.c:361 (sofia/external/711 at xx.xxx.xx.xx) State INIT 2011-12-19 22:40:33.136251 [DEBUG] mod_sofia.c:85 sofia/external/711 at xx.xxx.xx.xx SOFIA INIT send 1072 bytes to udp/[193.201.229.35]:5060 at 20:40:33.144177: ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ ------------------------------------------------------------------------ INVITE sip:711 at xx.xxx.xx.xx SIP/2.0 Via: SIP/2.0/UDP yy.yyy.yy.yy:5080;rport;branch=z9hG4bK0SepcyQeH1ZvD Route: ;gw=multifon-4orbit ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ Max-Forwards: 69 From: "NTCOM" ;tag=3B7DcUHFU3DcQ To: Call-ID: 89da3022-a524-122f-b690-bf4d9e35213b CSeq: 21827552 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4480abc 2011-05-27 21-46-28 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Allow-Events: talk, hold, refer Content-Type: application/sdp Content-Disposition: session On Mon, Dec 19, 2011 at 12:03:04PM -0500, Rob Hutton wrote: > Without seeing your dialplan, this is just a guess, but, I would bet that your gateway appears before this trunk in the external dialplan and the call matches the regex on the gateway. So, tighten your regex on the gateway and/or move this entry above the gateway entry. > > > On Monday 19 December 2011 11:54:12 Zhuravlov Sergey wrote: > > Hi, > > > > in my dialplan I try to call directly to the asterisk server follows: > > > > > > > > > > > > > > > > > > > > But it does not work for me! If the profile external has gateways then a call goes > > through a gateway, and receives forbidden. > > > > If in the profile external NOT have gateways - all is good and asterisk > > recieve call. > > > > I do not understand why this is happening? Where to look? > > > > > > > > -- Zhuravlov Sergey GTALK/JABBER:4orbit at gmail.com From acrow at integrafin.co.uk Tue Dec 20 12:40:43 2011 From: acrow at integrafin.co.uk (Alex Crow) Date: Tue, 20 Dec 2011 09:40:43 +0000 Subject: [Freeswitch-users] Issue adding a SIP Gateway In-Reply-To: <44E5C0A9D48A3246966A4AE04692014D102AA623@CH1PRD0604MB109.namprd06.prod.outlook.com> References: <44E5C0A9D48A3246966A4AE04692014D102A6FB5@CH1PRD0604MB109.namprd06.prod.outlook.com> <201112151954.01880.justlikeef@gmail.com> <44E5C0A9D48A3246966A4AE04692014D102A803C@CH1PRD0604MB109.namprd06.prod.outlook.com> <4EEDBBB2.50302@integrafin.co.uk> <44E5C0A9D48A3246966A4AE04692014D102AA3C6@CH1PRD0604MB109.namprd06.prod.outlook.com> <4EEF8EF0.6010207@integrafin.co.uk> <44E5C0A9D48A3246966A4AE04692014D102AA5E4@CH1PRD0604MB109.namprd06.prod.outlook.com> <4EEFA3BD.5010201@integrafin.co.uk> <44E5C0A9D48A3246966A4AE04692014D102AA623@CH1PRD0604MB109.namprd06.prod.outlook.com> Message-ID: <4EF0581B.30701@integrafin.co.uk> On 19/12/11 21:05, Ryan Watkins wrote: > > Alex, > > I've seen that as well... but where I figured it was part of the > example.xml gateway I didn't pay much mind. I have, however, renamed > the example.xml to example.xml.noload, stopped and started the daemon, > and attempted to reload the external profile within freeswitch... and > it's STILL loading the example gateway... and will not load iptel, > > I haven't ever touched the external.xml, but I'm wondering if maybe it > needed to be edited? Or perhaps the freeswitch install put yet another > location in that this setup that it might be looking to?? Really lost > at this point... don't know how it's loading the example gateway when > it's been renamed (even renamed the example.xml in > /usr/src/freeswitch* for S&Gs) > Ryan, Do you have exactly this at the top of /opt/freeswitch/conf/sip_profiles/external.xml? If you do, and it's still not loading, I am stumped too. Can you post your entire external.xml? Cheers Alex > *From:*Alex Crow [mailto:acrow at integrafin.co.uk] > *Sent:* Monday, December 19, 2011 12:51 PM > *To:* Ryan Watkins > *Cc:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Issue adding a SIP Gateway > > Ryan, > > I see this: > > 2011-12-19 12:28:30.388149 [DEBUG] sofia.c:1736 Launching worker > thread for external > 2011-12-19 12:28:30.388202 [ERR] sofia.c:2490 ERROR: password param is > REQUIRED! > > This could actually be from your "example.com" gateway, it is indeed > an error relating to gateway defs. > > I suggest you delete that gateway or move it to example.com.noload (or > the like). It could be stopping your proper gateway from loading. I'm > sure you don't need it anyway. > > Also check, if you've edited your external.xml file that you haven't > got messed up tags before the include of external/*.xml. > > Cheers > > Alex > > On 19/12/11 20:38, Ryan Watkins wrote: > > Here's the log file... thanks again Alex and all > > *From:*Alex Crow [mailto:acrow at integrafin.co.uk] > *Sent:* Monday, December 19, 2011 11:22 AM > *To:* FreeSWITCH Users Help > *Cc:* Ryan Watkins > *Subject:* Re: [Freeswitch-users] Issue adding a SIP Gateway > > OK, > > As long as those "funny quotes" in your last post aren't present in > your XML file I can't see what is the problem. I had an issue copying > dialplans from the web with these, had to go through and change them > all by hand. > > If you shut down freeswitch, clear your logfile, and start it again, > can you post the contents of /opt/freeswitch/log/freeswitch.log to the > list (or to pastebin etc and link here). > > There must be something in the logs. > > BTW you should not need to change things in /usr/src/*, once you've > installed that doesn't make any difference. > > Cheers > > Alex > > On 19/12/11 18:36, Ryan Watkins wrote: > > Thanks for the reply Alex, > > I followed the example in the FreeSWITCH 1.0.6 book for iptel, which > is as follows: > > > > > > (yes, I supplied my iptel > username in this line) > > (again, I supplied my iptel > password on this line) > > > > > > > > > > > > I've checked the file permissions and changed them so that all users > have rwx for the iptel.org.xml file; as well as every folder up to > /external for both /usr/src/freeswitch and the /opt/freeswitch paths > > I've also changed the iptel.org.xml files in those paths to the > example that you linked from the wiki > > However, I'm still getting the same result.... Any other suggestions? > > Thanks again! > > *From:*Alex Crow [mailto:acrow at integrafin.co.uk] > *Sent:* Sunday, December 18, 2011 2:09 AM > *To:* FreeSWITCH Users Help > *Cc:* Ryan Watkins > *Subject:* Re: [Freeswitch-users] Issue adding a SIP Gateway > > On 16/12/11 01:05, Ryan Watkins wrote: > > I did run the" sofia profile external restart reloadxml" command.... > It didn't load the new gateway, so that's why I tried registering the > gateway specifically. > > * > *Ryan,* > > *Did you follow this example: > http://wiki.freeswitch.org/wiki/Provider_Configuration:_iptel and > replace the usename and password with your own? > > Check that the permissions on the new XML file allow it to be read by > the user freeswitch is running as. > > Also double-check your closing tags on the file. This can cause your > gateway to be skipped, hence the "invalid gateway" when you try to use it. > > Cheers > > Alex > > > > -- > This message is intended only for the addressee and may contain > confidential information. Unless you are that person, you may not > disclose its contents or use it in any way and are requested to delete > the message along with any attachments and notify us immediately. > > "Transact" is operated by Integrated Financial Arrangements plc > Domain House, 5-7 Singer Street, London EC2A 4BQ > Tel: (020) 7608 4900 Fax: (020) 7608 5300 > (Registered office: as above; Registered in England and Wales under number: 3727592) > Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) > > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. > > > -- > This message has been scanned for viruses and > dangerous content by *MailScanner* , and is > believed to be clean. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111220/68500b55/attachment.html From ira at connectmevoice.com Tue Dec 20 16:58:59 2011 From: ira at connectmevoice.com (Ira Tessler) Date: Tue, 20 Dec 2011 08:58:59 -0500 Subject: [Freeswitch-users] Compact Sip Headers In-Reply-To: References: <1f358b8fdef7b855689d20f6bb7b1758@mail.gmail.com> Message-ID: Thank you! *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony Minessale *Sent:* Monday, December 19, 2011 10:00 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Compact Sip Headers yes, Profile param: enable-compact-headers=true On Mon, Dec 19, 2011 at 3:09 PM, Ira Tessler wrote: Is Freeswitch compatible with compact sip headers? If so, is there a parameter I have to set on the external profile to get them to work? _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://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 ------------------------------ No virus found in this message. Checked by AVG - www.avg.com Version: 2012.0.1890 / Virus Database: 2108/4691 - Release Date: 12/19/11 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111220/910fc05c/attachment-0001.html From miha at softnet.si Tue Dec 20 17:21:44 2011 From: miha at softnet.si (Miha Zoubek) Date: Tue, 20 Dec 2011 15:21:44 +0100 Subject: [Freeswitch-users] xml dialplan explenation Message-ID: <4EF099F8.9080305@softnet.si> Hi, I need a help about understanding freeswitch. From behaviour I can see that dialplan is only for B-leg. If I make call and disable radius cdr from dialplan, this only disable b lag (a leg start and stop freeswitch still send). If I disable radius cdr in vars.xml, and than make disable_radius_stop=false, it enable packets for B leg, (a is missing, which is ok for me). My quastione is if dialplan is only for b leg? Other one, I have put disable_radius_start=false and disable_radius_stop=false in dial plan. Only stop packet is send. Does anyone know why? From my log I can see this :Action set(disable_radius_start=false) Dialplan: sofia/internal/018108500 at xxx.xxx.xxx.xxx Action set(disable_radius_stop=false). But further than 2011-12-20 12:19:54.357167 [DEBUG] mod_radius_cdr.c:156 [mod_radius_cdr] Entering my_on_routing 2011-12-20 12:19:54.357167 [DEBUG] mod_radius_cdr.c:165 [mod_radius_cdr] Not Sending RADIUS Start Can anyone know why? 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/20111220/497707e4/attachment.html From ryan at kaevee.com Tue Dec 20 17:33:39 2011 From: ryan at kaevee.com (Ryan V) Date: Tue, 20 Dec 2011 20:03:39 +0530 Subject: [Freeswitch-users] One way voice on incoming calls Message-ID: Hi, We have configured a Sangoma A101DE. We are using Grandstream GXP280 phones and no NAT involved. We are using example configuration files except for one change in 00_inbound_did.xml. Calls between extensions works fine and outbound calls also work fine. Though we are able to receive inbound calls, incoming audio is missing. Calling party hears us loud and clear. Any suggestions? Thanks, Ryan. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111220/7932c3db/attachment.html From wstephen80 at gmail.com Tue Dec 20 17:33:50 2011 From: wstephen80 at gmail.com (Stephen Wilde) Date: Tue, 20 Dec 2011 15:33:50 +0100 Subject: [Freeswitch-users] Originate Resulted in Error Cause: 100 [INVALID_IE_CONTENTS] In-Reply-To: References: Message-ID: Ok, I solved this issue: the problem was the payload used by my provider for telephone-event that was 150 instead of 101. When the provider fixed this payload number all calls works fine. Stephen On Mon, Dec 19, 2011 at 9:30 PM, Stephen Wilde wrote: > I have a problem with a provider where the call are dropped by Freeswitch > after receiving a 183 + SDP. > I think that the problem is in SDP but I don't know where. > The SDP that Freeswitch sends in INVITE is: > > v=0 > o=FreeSWITCH 1324273767 1324273768 IN IP4 62.196.58.2 > s=FreeSWITCH > c=IN IP4 62.196.58.2 > t=0 0 > m=audio 52062 RTP/AVP 18 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > > > where the SDP that Freeswitch receives from the provider is: > > v=0 > o=sansay-VSX 10 10 IN IP4 206.172.44.229 > s=session controller > c=IN IP4 206.172.44.229 > t=0 0 > m=audio 28364 RTP/AVP 18 150 > a=rtpmap:18 G729/8000 > a=fmtp:18 annexb=no > a=rtpmap:150 telephone-event/8000 > a=fmtp:150 0-15 > a=ptime:20 > > > After this SDP I see in the log: > > 2011-12-19 21:17:11.567017 [DEBUG] switch_ivr_originate.c:3364 Originate > Resulted in Error Cause: 100 [INVALID_IE_CONTENTS] > > Can anyone give me an advice to solve this issue? > > Stephen > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111220/daeae597/attachment.html From mays.david at gmail.com Tue Dec 20 17:58:16 2011 From: mays.david at gmail.com (dma) Date: Tue, 20 Dec 2011 06:58:16 -0800 (PST) Subject: [Freeswitch-users] Error "Channels not ready" when bridge call in Lua Message-ID: <1324393096164-7112050.post@n2.nabble.com> Dear support, I have a Lua script that accepts inbound call, creates outbound call, and then bridges the 2 sessions. Its code is simply like this: -------------------- session:answer() session2 = freeswitch.Session(callstring) if (session2:ready()) then freeswitch.bridge(session, session2) end -------------------- However, when bridge() is called, the call immediately hangs up. The following is reported: 2011-12-13 18:08:00.674769 [ERR] switch_cpp.cpp:1247 Channels not ready 2011-12-13 18:08:00.674769 [DEBUG] switch_cpp.cpp:618 CoreSession::hangup By searching through the forum, I find a previous report of the same error with Lua script. However no solution is provided for that ticket. Please kind advise how to fix this problem. Thanks, D.Ma -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-Channels-not-ready-when-bridge-call-in-Lua-tp7112050p7112050.html Sent from the freeswitch-users mailing list archive at Nabble.com. From boris at tagnet.ru Tue Dec 20 18:18:47 2011 From: boris at tagnet.ru (Boris Kovalenko) Date: Tue, 20 Dec 2011 21:18:47 +0600 Subject: [Freeswitch-users] Error "Channels not ready" when bridge call in Lua In-Reply-To: <1324393096164-7112050.post@n2.nabble.com> References: <1324393096164-7112050.post@n2.nabble.com> Message-ID: <4EF0A757.8070404@tagnet.ru> Hello! A have same problem about a three months ago. Fixed with upgrade to the latest git. > Dear support, > > I have a Lua script that accepts inbound call, creates outbound call, and > then bridges the 2 sessions. Its code is simply like this: > > -------------------- > session:answer() > session2 = freeswitch.Session(callstring) > > if (session2:ready()) then > freeswitch.bridge(session, session2) > end > -------------------- > > However, when bridge() is called, the call immediately hangs up. The > following is reported: > > 2011-12-13 18:08:00.674769 [ERR] switch_cpp.cpp:1247 Channels not ready > 2011-12-13 18:08:00.674769 [DEBUG] switch_cpp.cpp:618 CoreSession::hangup > > By searching through the forum, I find a previous report of the same error > with Lua script. However no solution is provided for that ticket. > > Please kind advise how to fix this problem. > > Thanks, > D.Ma > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-Channels-not-ready-when-bridge-call-in-Lua-tp7112050p7112050.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 -- Regards, Boris From curriegrad2004 at gmail.com Tue Dec 20 18:22:09 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 20 Dec 2011 07:22:09 -0800 Subject: [Freeswitch-users] One way voice on incoming calls In-Reply-To: References: Message-ID: I'm suggesting there may be mismatched rtp port ranges on the grandstream phone and on the FreeSWITCH server. Use wireshark or run a tcpdump and see what's going on. It's way easier to see problems from the inside than outside. On 2011-12-20 6:34 AM, "Ryan V" wrote: > Hi, > > We have configured a Sangoma A101DE. We are using Grandstream GXP280 > phones and no NAT involved. We are using example configuration files except > for one change in 00_inbound_did.xml. > > > > Calls between extensions works fine and outbound calls also work fine. > > Though we are able to receive inbound calls, incoming audio is missing. > Calling party hears us loud and clear. > > Any suggestions? > > Thanks, > > 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/20111220/eab0380e/attachment.html From beppe.grillo at gmail.com Tue Dec 20 18:54:00 2011 From: beppe.grillo at gmail.com (Beppe Grillo) Date: Tue, 20 Dec 2011 16:54:00 +0100 Subject: [Freeswitch-users] RFC-3841 : Insert Accept-contact in sip INVITE Message-ID: Hi, I need insert Accept-Contact in sip INVITE . I saw that RFC-3841 is supported . Do I configure something in the profile of gw to enter the Accept-contact in Sip Invite ? Regards, Giuseppe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111220/80c82a50/attachment.html From jeff at jefflenk.com Tue Dec 20 19:27:01 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Tue, 20 Dec 2011 08:27:01 -0800 (PST) Subject: [Freeswitch-users] conf count minus 1 /invalid UTF-8 character In-Reply-To: <1324336151927-7110155.post@n2.nabble.com> References: <1324325740.69360.YahooMailNeo@web65311.mail.ac2.yahoo.com> <1324336151927-7110155.post@n2.nabble.com> Message-ID: <1324398421739-7112372.post@n2.nabble.com> Rodney, Do you have any unusual chars embedded in that string? that are not visible? Jeff -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/conf-count-minus-1-invalid-UTF-8-character-tp7109625p7112372.html Sent from the freeswitch-users mailing list archive at Nabble.com. From kris at kriskinc.com Tue Dec 20 19:28:01 2011 From: kris at kriskinc.com (Kristian Kielhofner) Date: Tue, 20 Dec 2011 11:28:01 -0500 Subject: [Freeswitch-users] Originate Resulted in Error Cause: 100 [INVALID_IE_CONTENTS] In-Reply-To: References: Message-ID: Wow, I'm sorry I missed that first time around. I hope your provider otherwise pays more attention to standards... RTP dynamic payload type 150 is completely invalid and out of range: http://www.iana.org/assignments/rtp-parameters The topend for dynamic payload types is 127. On Tue, Dec 20, 2011 at 9:33 AM, Stephen Wilde wrote: > Ok, I solved this issue: the problem was the payload used by my provider for > telephone-event that was 150 instead of 101. > When the provider fixed this payload number all calls works fine. > > Stephen -- Kristian Kielhofner From justlikeef at gmail.com Tue Dec 20 19:48:08 2011 From: justlikeef at gmail.com (Rob Hutton) Date: Tue, 20 Dec 2011 11:48:08 -0500 Subject: [Freeswitch-users] unexpectedly dial through gateway In-Reply-To: <20111220090317.GB4367@klbank.ru> References: <20111219165411.GA9141@klbank.ru> <201112191203.05026.justlikeef@gmail.com> <20111220090317.GB4367@klbank.ru> Message-ID: <201112201148.09010.justlikeef@gmail.com> Can you post your complete dialplan and the call logs of the two situations that you describe to pastebin? It is hard to follow what you are asking, but I can explain what is going on that way. On Tuesday 20 December 2011 04:03:17 Zhuravlov Sergey wrote: > Thank you! > I was delighted and thought, as I did not think to myself! Indeed > found earlier definition of gateways, fixed regexp. > > However, I realized that the call is routed through the gateway, which I only > use for incoming calls, outgoing is no any rules or regexps, which would allow > the call to go through it. > > Mystery! > > And most importantly, that when turned off via the CLI this particular gateway > - everything works as expected. Just turn on again the call goes through it for > some reason! > > Maybe you can somehow increase the debug level for routing? > > yy.yyy.yy.yy -- FS > xx.xxx.xx.xx -- Asterisk > > Example call 1 - gateway with problem is off - the call goes as it should > > > EXECUTE sofia/internal/271 at intra.mydom.com hash(insert/intra.mydom.com-last_dial/271/711) > EXECUTE sofia/internal/271 at intra.mydom.com hash(insert/intra.mydom.com-last_dial/global/042c1036-2a81-11e1-9963-ab2ff8b5cdf8) > EXECUTE sofia/internal/271 at intra.mydom.com set(RFC2822_DATE=Mon, 19 Dec 2011 22:35:40 +0200) > 2011-12-19 22:35:40.694311 [DEBUG] mod_dptools.c:1063 sofia/internal/271 at intra.mydom.com SET [RFC2822_DATE]=[Mon, 19 Dec 2011 2 > 2:35:40 +0200] > EXECUTE sofia/internal/271 at intra.mydom.com set(hangup_after_bridge=true) > 2011-12-19 22:35:40.694311 [DEBUG] mod_dptools.c:1063 sofia/internal/271 at intra.mydom.com SET [hangup_after_bridge]=[true] > EXECUTE sofia/internal/271 at intra.mydom.com bridge(sofia/external/711 at xx.xxx.xx.xx) > 2011-12-19 22:35:40.694311 [DEBUG] switch_ivr_originate.c:1869 Parsing global variables > 2011-12-19 22:35:40.694311 [NOTICE] switch_channel.c:816 New Channel sofia/external/711 at xx.xxx.xx.xx [042d9b5e-2a81-11e1-9964-a > b2ff8b5cdf8] > 2011-12-19 22:35:40.694311 [DEBUG] mod_sofia.c:4311 (sofia/external/711 at xx.xxx.xx.xx) State Change CS_NEW -> CS_INIT > 2011-12-19 22:35:40.694311 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/711 at xx.xxx.xx.xx [BREAK] > 2011-12-19 22:35:40.694311 [DEBUG] switch_core_state_machine.c:325 (sofia/external/711 at xx.xxx.xx.xx) Running State Change CS_IN > IT > 2011-12-19 22:35:40.694311 [DEBUG] switch_core_state_machine.c:361 (sofia/external/711 at xx.xxx.xx.xx) State INIT > 2011-12-19 22:35:40.694311 [DEBUG] mod_sofia.c:85 sofia/external/711 at xx.xxx.xx.xx SOFIA INIT > send 983 bytes to udp/[xx.xxx.xx.xx]:5060 at 20:35:40.708576: > ------------------------------------------------------------------------ > INVITE sip:711 at xx.xxx.xx.xx SIP/2.0 > Via: SIP/2.0/UDP yy.yyy.yy.yy:5080;rport;branch=z9hG4bKF00gejSj294yr > Max-Forwards: 69 > From: "NTCOM" ;tag=tt3ev7SFNK53e > To: > Call-ID: db8c0ed3-a523-122f-b690-bf4d9e35213b > CSeq: 21827406 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4480abc 2011-05-27 21-46-28 -0500 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, refer > Content-Type: application/sdp > > > > > Example Call 2 - Gateway is turned on and the call goes through it -- it's > WRONG > > > EXECUTE sofia/internal/271 at intra.mydom.com hash(insert/intra.mydom.com-last_dial/271/711) > EXECUTE sofia/internal/271 at intra.mydom.com hash(insert/intra.mydom.com-last_dial/global/b27a7a7e-2a81-11e1-99a5-ab2ff8b5cdf8) > EXECUTE sofia/internal/271 at intra.mydom.com set(RFC2822_DATE=Mon, 19 Dec 2011 22:40:33 +0200) > 2011-12-19 22:40:33.136251 [DEBUG] mod_dptools.c:1063 sofia/internal/271 at intra.mydom.com SET [RFC2822_DATE]=[Mon, 19 Dec 2011 2 > 2:40:33 +0200] > EXECUTE sofia/internal/271 at intra.mydom.com set(hangup_after_bridge=true) > 2011-12-19 22:40:33.136251 [DEBUG] mod_dptools.c:1063 sofia/internal/271 at intra.mydom.com SET [hangup_after_bridge]=[true] > EXECUTE sofia/internal/271 at intra.mydom.com bridge(sofia/external/711 at xx.xxx.xx.xx) > 2011-12-19 22:40:33.136251 [DEBUG] switch_ivr_originate.c:1869 Parsing global variables > 2011-12-19 22:40:33.136251 [NOTICE] switch_channel.c:816 New Channel sofia/external/711 at xx.xxx.xx.xx [b27bc0dc-2a81-11e1-99a6-a > b2ff8b5cdf8] > 2011-12-19 22:40:33.136251 [DEBUG] mod_sofia.c:4311 (sofia/external/711 at xx.xxx.xx.xx) State Change CS_NEW -> CS_INIT > 2011-12-19 22:40:33.136251 [DEBUG] switch_core_session.c:1114 Send signal sofia/external/711 at xx.xxx.xx.xx [BREAK] > 2011-12-19 22:40:33.136251 [DEBUG] switch_core_state_machine.c:325 (sofia/external/711 at xx.xxx.xx.xx) Running State Change CS_IN > IT > 2011-12-19 22:40:33.136251 [DEBUG] switch_core_state_machine.c:361 (sofia/external/711 at xx.xxx.xx.xx) State INIT > 2011-12-19 22:40:33.136251 [DEBUG] mod_sofia.c:85 sofia/external/711 at xx.xxx.xx.xx SOFIA INIT > send 1072 bytes to udp/[193.201.229.35]:5060 at 20:40:33.144177: > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ > ------------------------------------------------------------------------ > INVITE sip:711 at xx.xxx.xx.xx SIP/2.0 > Via: SIP/2.0/UDP yy.yyy.yy.yy:5080;rport;branch=z9hG4bK0SepcyQeH1ZvD > Route: ;gw=multifon-4orbit > ^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^^ > Max-Forwards: 69 > From: "NTCOM" ;tag=3B7DcUHFU3DcQ > To: > Call-ID: 89da3022-a524-122f-b690-bf4d9e35213b > CSeq: 21827552 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-4480abc 2011-05-27 21-46-28 -0500 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Allow-Events: talk, hold, refer > Content-Type: application/sdp > Content-Disposition: session > > > > > > > > On Mon, Dec 19, 2011 at 12:03:04PM -0500, Rob Hutton wrote: > > Without seeing your dialplan, this is just a guess, but, I would bet that your gateway appears before this trunk in the external dialplan and the call matches the regex on the gateway. So, tighten your regex on the gateway and/or move this entry above the gateway entry. > > > > > > On Monday 19 December 2011 11:54:12 Zhuravlov Sergey wrote: > > > Hi, > > > > > > in my dialplan I try to call directly to the asterisk server follows: > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > But it does not work for me! If the profile external has gateways then a call goes > > > through a gateway, and receives forbidden. > > > > > > If in the profile external NOT have gateways - all is good and asterisk > > > recieve call. > > > > > > I do not understand why this is happening? Where to look? > > > > > > > > > > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111220/17e0bf42/attachment-0001.html From notlikeme75 at yahoo.com Tue Dec 20 20:11:04 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 20 Dec 2011 09:11:04 -0800 (PST) Subject: [Freeswitch-users] conference count minus 1 In-Reply-To: References: Message-ID: <1324401064.59499.YahooMailNeo@web65304.mail.ac2.yahoo.com> jeff, i have tried both expressions. even re tried original file and its same errors. just happen with new version update this week. also i get a bunch of "adding tone descripters" for busy, etc. that weren't there before on reloadxml rodney -Rodney, Do you have any unusual chars embedded in that string? that are not visible? Jeff - -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111220/8ca35652/attachment.html From notlikeme75 at yahoo.com Tue Dec 20 20:21:14 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 20 Dec 2011 09:21:14 -0800 (PST) Subject: [Freeswitch-users] transfer to conference from conference / caller controls Message-ID: <1324401674.88309.YahooMailNeo@web65310.mail.ac2.yahoo.com> I am trying to transfer to a dynamic conference from within a static conference but when I do my callers loose control and can not use any dtmf caller control options and must hang up to get back. I tried using clear digit action but do not think this works on the "conf" realm. the dynamic conference works fine with all controls if i send them directly from the ivr but for what i am trying it is important to do this transfer from another conference. is there a method transfer and release all previous controls? i even tried using a difference conference profile so i transferred from static 501 at default to @dynamic to see if giving new controls would work, to no avail. is there a log or something i can post that will help you in understanding how this is happening? or is it even fixable? thanks rodney ?? ????? ????? ????? ????? ????? -???? ??????? ??????? ? (main ivr menu extension) ??????? condition??? ? destination_number??? ? ^759$??? ? 1??? ? ? action??? ? answer??? ? ??? ? 2??? ? ? action??? ? set??? ? conference_user_list=|??? ? 11??? ? ? action??? ? play_and_get_digits??? ? 4 4 3 5000 # askprivateroom.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+??? ? 15??? ? ? action??? ? phrase??? ? spell,${target_num}??? ? 16??? ? ? action??? ? conference??? ? ${target_num}-127.0.0.1 at default -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111220/bac175f8/attachment.html From ryan at kaevee.com Tue Dec 20 20:26:40 2011 From: ryan at kaevee.com (Ryan V) Date: Tue, 20 Dec 2011 22:56:40 +0530 Subject: [Freeswitch-users] One way voice on incoming calls In-Reply-To: References: Message-ID: Thanks for the reply. Let me check first thing in morning. Ryan. On Tue, Dec 20, 2011 at 8:52 PM, curriegrad2004 wrote: > I'm suggesting there may be mismatched rtp port ranges on the grandstream > phone and on the FreeSWITCH server. Use wireshark or run a tcpdump and see > what's going on. It's way easier to see problems from the inside than > outside. > On 2011-12-20 6:34 AM, "Ryan V" wrote: > >> Hi, >> >> We have configured a Sangoma A101DE. We are using Grandstream GXP280 >> phones and no NAT involved. We are using example configuration files except >> for one change in 00_inbound_did.xml. >> >> >> >> Calls between extensions works fine and outbound calls also work fine. >> >> Though we are able to receive inbound calls, incoming audio is missing. >> Calling party hears us loud and clear. >> >> Any suggestions? >> >> Thanks, >> >> 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/20111220/0fd7552a/attachment.html From jeff at jefflenk.com Tue Dec 20 20:55:32 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Tue, 20 Dec 2011 09:55:32 -0800 (PST) Subject: [Freeswitch-users] conference count minus 1 In-Reply-To: <1324401064.59499.YahooMailNeo@web65304.mail.ac2.yahoo.com> References: <1324401064.59499.YahooMailNeo@web65304.mail.ac2.yahoo.com> Message-ID: <1324403732987-7112705.post@n2.nabble.com> I'm not seeing the problem. Can you attach a complete sample of the problem(simplified). -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/conference-count-minus-1-tp7112563p7112705.html Sent from the freeswitch-users mailing list archive at Nabble.com. From lloydie.t at gmail.com Tue Dec 20 21:03:53 2011 From: lloydie.t at gmail.com (lloyd thomas) Date: Tue, 20 Dec 2011 18:03:53 +0000 Subject: [Freeswitch-users] Problems with polycom registering Message-ID: I had what I thought was the perfect set up on my FS box. Until I changed internet provider (BT) All was fine for a week but all of a sudden some of my Polycom phones won't register. The polycom phones are behind a NAT and my FS box is on a static IP address. I am using a multi company setup One of my polycom phones seems to register OK Using 'sofia profile xxxxxxxx siptrace on' I can see that one of my polycom phones will not register because FS is trying to send replies to the private IP address (192.168.101.16) Any Ideas? Am I missing something in one of the profiles Good one ------------------------------------------------------------------------ send 702 bytes to udp/[87.194.242.110]:65004 at 17:49:32.226606: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 87.194.242.110:65004;branch=z9hG4bK15c264eE5B5988B From: "Suze" ;tag=CA76AF43-ED2A5944 To: ;tag=7Uy88X4pc8yHc Call-ID: 918d5430-ffb07455-eb2121fe at 87.194.242.110 CSeq: 888 REGISTER Contact: ;methods="INVITE, ACK, BYE, CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER";expires=30 Date: Tue, 20 Dec 2011 17:49:32 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2c009dd 2011-03-15 14-29-04 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces Content-Length: 0 ------------------------------------------------------------------------- Bad one ------------------------------------------------------------------------ send 680 bytes to udp/[81.137.114.169]:5060 at 17:50:17.659167: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.101.16:5060 ;branch=z9hG4bK2b8f42dA63D9E42;received=81.137.154.169 From: "Lloyd" ;tag=621EA6CE-AF1D573 To: ;tag=aQaKeFQ13219e Call-ID: 52e98627-d24590dc-528d7361 at 192.168.101.16 CSeq: 1 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2c009dd 2011-03-15 14-29-04 -0500 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="phisys.tele.phi.co.uk", nonce="1319a358-2b33-11e1-b5e4-dd1099e4b2d2", algorithm=MD5, qop="auth" Content-Length: 0 ------------------------------------------------------------------------ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111220/b7058ef4/attachment-0001.html From msc at freeswitch.org Tue Dec 20 21:48:22 2011 From: msc at freeswitch.org (Michael Collins) Date: Tue, 20 Dec 2011 10:48:22 -0800 Subject: [Freeswitch-users] originate via rpc In-Reply-To: References: Message-ID: You could use loopback and send it to an extension that calls lcr, or you could use the inline dialplan and create an epic URL that does it all in one go. :) -MC On Mon, Dec 19, 2011 at 1:22 PM, Anders wrote: > You were right - thanks a lot. I did URL encoding on the whole thing > and it (almost) worked. For some reason it decided that the space > before '&' should be encoded to + but I changed that manually to %20, > and it worked. > Now, I just need to understand how I can originate a call to use the > lcr and not a specific carrier...pointers? > Thanks! > On Mon, Dec 19, 2011 at 1:56 PM, Michael Collins > wrote: > > Don't you have to URL encode the & character? > > -MC > > > > On Mon, Dec 19, 2011 at 11:14 AM, Anders wrote: > >> > >> Hi, > >> > >> I am trying to originate a call, using a mod command through the > >> xml_rpc, but I am doing something wrong - not sure what it is. I post > >> this url: > >> > >> > >> > http://1.2.3.4/webapi/originate?{origination_caller_id_number=14007654321}sofia/carriers/111#18001234567 at 4.5.6.7%20&playback(/vr/migstory.wav) > >> > >> (numbers and IPs modified here) > >> > >> The reponse I get is this: -USAGE: > >> |&() [] [] > >> [] [] [] > >> > >> I can do other commands directly in the browser like this (such as > >> http://1.2.3.4/webapi/show?calls) without any problems. Therefore my > >> conclusion is that I am doing something wrong in this syntax for http. > >> > >> Any ideas? > >> > >> Thanks in advance! > >> > >> //Anders > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111220/29717f24/attachment.html From nestor at tiendalinux.com Wed Dec 21 01:05:55 2011 From: nestor at tiendalinux.com (Nestor A Diaz) Date: Tue, 20 Dec 2011 17:05:55 -0500 Subject: [Freeswitch-users] Freeswitch core dump on ftmod_wanpipe In-Reply-To: <4EE8E6B6.2090102@integrafin.co.uk> References: <4EE7B2DE.6080906@tiendalinux.com> <4EE8E6B6.2090102@integrafin.co.uk> Message-ID: <4EF106C3.2060806@tiendalinux.com> Hi, i did an upgrade to the latest freeswitch sources (2011-12-16) and it haven't core dumped for now, hoewever i haven't put the server in production, so still crossing fingers :) but right now i have a new set of warnings : 2011-12-19 23:12:14.732393 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it while i don't care about them since the system is running i will take care ot them later. Slds. -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-485-3020 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:211 at tiendalinux.com Email/MSN: nestor at tiendalinux.com http://www.tiendalinux.com/ Bogota, Colombia On 12/14/2011 01:11 PM, Alex Crow wrote: > > Hi Nestor, > > There is no harm in getting in contact with Sangoma. They were > incredibly helpful with my BRI card issue, and gave me a fixed package > within a few days of diagnosing the problem. Absolutely a credit to > the telephony hardware industry, and a few of the developers are > Freeswitch contributors I think... > > Best to make sure you are on latest git of FS though, and have > followed all the relevant instructions for building wanpipe from their > site. The distro Wanpipe is probably no good for FS use, in case you > are using that. > > Cheers > > Alex > > > > > > From brad at tech21.com Wed Dec 21 01:23:39 2011 From: brad at tech21.com (Brad Mina) Date: Tue, 20 Dec 2011 14:23:39 -0800 Subject: [Freeswitch-users] Freeswitch core dump on ftmod_wanpipe In-Reply-To: <4EF106C3.2060806@tiendalinux.com> References: <4EE7B2DE.6080906@tiendalinux.com> <4EE8E6B6.2090102@integrafin.co.uk> <4EF106C3.2060806@tiendalinux.com> Message-ID: Another user has reported this error in a completely different scenario. See the thread "conf count minus 1" I doubt this issue is related to the one you were having. On Tue, Dec 20, 2011 at 2:05 PM, Nestor A Diaz wrote: > Hi, i did an upgrade to the latest freeswitch sources (2011-12-16) and > it haven't core dumped for now, hoewever i haven't put the server in > production, so still crossing fingers :) but right now i have a new set > of warnings : > > 2011-12-19 23:12:14.732393 [WARNING] switch_xml.c:2328 Invalid UTF-8 > character to ampersand, skip it > > while i don't care about them since the system is running i will take > care ot them later. > > Slds. > > -- > Nestor A. Diaz > Ingeniero de Sistemas > Tel. +57 1-485-3020 x 211 > Cel. +57 316-227-3593 > Tel. SIP: sip:211 at tiendalinux.com > Email/MSN: nestor at tiendalinux.com > http://www.tiendalinux.com/ > Bogota, Colombia > > > On 12/14/2011 01:11 PM, Alex Crow wrote: > > > > Hi Nestor, > > > > There is no harm in getting in contact with Sangoma. They were > > incredibly helpful with my BRI card issue, and gave me a fixed package > > within a few days of diagnosing the problem. Absolutely a credit to > > the telephony hardware industry, and a few of the developers are > > Freeswitch contributors I think... > > > > Best to make sure you are on latest git of FS though, and have > > followed all the relevant instructions for building wanpipe from their > > site. The distro Wanpipe is probably no good for FS use, in case you > > are using that. > > > > Cheers > > > > Alex > > > > > > > > > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111220/06b0a870/attachment.html From manieq at wp.eu Wed Dec 21 01:30:35 2011 From: manieq at wp.eu (Mariusz Czulada) Date: Tue, 20 Dec 2011 23:30:35 +0100 Subject: [Freeswitch-users] Odp: transfer to conference from conference / caller controls In-Reply-To: <1324401674.88309.YahooMailNeo@web65310.mail.ac2.yahoo.com> References: <1324401674.88309.YahooMailNeo@web65310.mail.ac2.yahoo.com> Message-ID: <4ef10c8b296f48.49122800@wp.pl> An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111220/ec0c280a/attachment-0001.html From justlikeef at gmail.com Wed Dec 21 02:28:33 2011 From: justlikeef at gmail.com (Rob Hutton) Date: Tue, 20 Dec 2011 18:28:33 -0500 Subject: [Freeswitch-users] Freeswitch core dump on ftmod_wanpipe In-Reply-To: References: <4EE7B2DE.6080906@tiendalinux.com> <4EF106C3.2060806@tiendalinux.com> Message-ID: <201112201828.33818.justlikeef@gmail.com> I am seeing it on a test server making normal station to station calls, also. On Tuesday 20 December 2011 17:23:39 Brad Mina wrote: > Another user has reported this error in a completely different scenario. > See the thread "conf count minus 1" > > I doubt this issue is related to the one you were having. > > On Tue, Dec 20, 2011 at 2:05 PM, Nestor A Diaz wrote: > > > Hi, i did an upgrade to the latest freeswitch sources (2011-12-16) and > > it haven't core dumped for now, hoewever i haven't put the server in > > production, so still crossing fingers :) but right now i have a new set > > of warnings : > > > > 2011-12-19 23:12:14.732393 [WARNING] switch_xml.c:2328 Invalid UTF-8 > > character to ampersand, skip it > > > > while i don't care about them since the system is running i will take > > care ot them later. > > > > Slds. > > > > -- > > Nestor A. Diaz > > Ingeniero de Sistemas > > Tel. +57 1-485-3020 x 211 > > Cel. +57 316-227-3593 > > Tel. SIP: sip:211 at tiendalinux.com > > Email/MSN: nestor at tiendalinux.com > > http://www.tiendalinux.com/ > > Bogota, Colombia > > > > > > On 12/14/2011 01:11 PM, Alex Crow wrote: > > > > > > Hi Nestor, > > > > > > There is no harm in getting in contact with Sangoma. They were > > > incredibly helpful with my BRI card issue, and gave me a fixed package > > > within a few days of diagnosing the problem. Absolutely a credit to > > > the telephony hardware industry, and a few of the developers are > > > Freeswitch contributors I think... > > > > > > Best to make sure you are on latest git of FS though, and have > > > followed all the relevant instructions for building wanpipe from their > > > site. The distro Wanpipe is probably no good for FS use, in case you > > > are using that. > > > > > > Cheers > > > > > > Alex > > > > > > > > > > > > > > > > > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111220/bf2f8a14/attachment.html From mays.david at gmail.com Wed Dec 21 04:31:31 2011 From: mays.david at gmail.com (dma) Date: Tue, 20 Dec 2011 17:31:31 -0800 (PST) Subject: [Freeswitch-users] Error "Channels not ready" when bridge call in Lua In-Reply-To: <4EF0A757.8070404@tagnet.ru> References: <1324393096164-7112050.post@n2.nabble.com> <4EF0A757.8070404@tagnet.ru> Message-ID: <1324431091351-7113903.post@n2.nabble.com> Hi Boris, Thanks a lot for the information! I shall try to upgrade! Best regards, D.Ma -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-Channels-not-ready-when-bridge-call-in-Lua-tp7112050p7113903.html Sent from the freeswitch-users mailing list archive at Nabble.com. From lloydie.t at gmail.com Wed Dec 21 04:58:18 2011 From: lloydie.t at gmail.com (lloyd thomas) Date: Wed, 21 Dec 2011 01:58:18 +0000 Subject: [Freeswitch-users] Problem with make current Message-ID: Since I am not getting anywhere with my sip registration problem I decided to update FS from git using make current. Unfortunately I am getting the following errors. What can I do to resolve? In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:101, from ./src/include/private/switch_core_pvt.h:35, from src/switch_apr.c:37: /usr/src/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:145: error: expected declaration specifiers or ?...? before ?tz_t? make[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1 make[2]: Leaving directory `/usr/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/8bd4c1df/attachment.html From justlikeef at gmail.com Wed Dec 21 06:45:56 2011 From: justlikeef at gmail.com (Rob Hutton) Date: Tue, 20 Dec 2011 22:45:56 -0500 Subject: [Freeswitch-users] Need help with PortAudio Message-ID: <201112202245.57010.justlikeef@gmail.com> I am trying to get portaudio working, and have run into a couple of problems 1) If portaudio cannot access the device, it causes freeswitch to segfault 2) I have set up a dialplan following the intercom example which seems to be working, but I get no audio. I have tried setting every device shown in devlist as the output device. 3) When I issue a "pa play". the cli hangs, even if I give it a timeout. Any thoughts? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111220/7d90e663/attachment.html From nestor at tiendalinux.com Wed Dec 21 07:28:28 2011 From: nestor at tiendalinux.com (Nestor A Diaz) Date: Tue, 20 Dec 2011 23:28:28 -0500 Subject: [Freeswitch-users] Guidelines for setting up contexts per user / gateway Message-ID: <4EF1606C.9040006@tiendalinux.com> Hello Freeswitch Users List, I am trying to emulate asterisk behavior for user / gateway context under FreeSWITCH. For users inside the 'directory' directory I can set up different contexts using the 'user_context' variable, solved. But when i have to deal with external gateways sip_profiles seems to be the best place to put the configuration, and if i want to use just one sip profile due to firewall restricctions i am limited to just one context. According to FreeSWITCH FAQ: Q: How do I assign endpoints to contexts with different sets of extensions Here are the different possible approaches: 1. use 1 profile per context you want to route to (each one needs a distinct ip:port) 2. use different domains in the registration data and use the auto context thing 3. send them all to a common context and execute_exten or transfer to somewhere else 4. send them to an IVR to decide where they go 5. use xml_curl to make it dish out a different dialplan based on who they are in the list of data you are fed I don't want to use 1 profile per context, so option 1 discarded. I don't want to use different domains in the registration data, since it have no semantic for a one company, so option 2 discarded, but for gateways it makes sense. I don't want to use an IVR for every user call, so option 4 discarded. So that gives me two choices, use xml_curl, which means i have to set up a webserver but that seems overkill, it seems to be an superset of option 3, so that left me with option 3 using just static XML files. I found this example on the dialplan XML: It seems to be the first place to start adapting to internal extension and transfering to the correct dialplan, anybody have a more refined example ? which variables can i trust / use in order to validate a call from a particular gateway and then set up a context transfer according to them. p.d. anybody knows about a good cdr web stats software for freeswitch ? i have used cdr-stats for asterisk in the past, newer version seems to have freeswitch support on the way, but stable ? any suggestions will be apreciated . Thank you !! -- Nestor A. Diaz Ingeniero de Sistemas Tel. +57 1-485-3020 x 211 Cel. +57 316-227-3593 Tel. SIP: sip:211 at tiendalinux.com Email/MSN: nestor at tiendalinux.com http://www.tiendalinux.com/ Bogota, Colombia -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111220/84655ee8/attachment-0001.html From sunwood360 at gmail.com Wed Dec 21 08:25:45 2011 From: sunwood360 at gmail.com (envelopes envelopes) Date: Tue, 20 Dec 2011 21:25:45 -0800 Subject: [Freeswitch-users] How to invite an endpoint sip phone directly? Message-ID: The phone is not registered in FS. Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111220/dd94903c/attachment.html From dujinfang at gmail.com Wed Dec 21 09:09:49 2011 From: dujinfang at gmail.com (Seven Du) Date: Wed, 21 Dec 2011 14:09:49 +0800 Subject: [Freeswitch-users] quick question about eavesdrop Message-ID: Hi, first sorry I'm using an old version of FS FreeSWITCH Version 1.0.head (hacked-20110626T115851Z). Just a quick question I found that when I eavesdrop a bridged call between a sip channel and a skinny channel, if the eavesdrop applied to the sip uuid then everything is ok, but if applied to the skynny uuid, voice is stretched (speaks slower and slower), any hint on this? here are what I see in show channels. eavesdrop sip uuid uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid dc11d8a7-3a1c-42d6-a936-577438b44a8c,inbound,2011-12-21 22:14:41,1324476881,sofia/internal/6802 at 6802:5060,CS_EXECUTE,6802,6802,192.168.1.118,6807,bridge,skinny/internal/6807,XML,default,PCMU,8000,64000,PCMU,8000,64000,,ola,6802 at 6802,,ACTIVE,Outbound Call,internal/6807,SEND,dc11d8a7-3a1c-42d6-a936-577438b44a8c b07e40f4-1978-4142-b4fe-35e3ee00c8bd,outbound,2011-12-21 22:14:41,1324476881,SKINNY/internal/6807,CS_EXCHANGE_MEDIA,6802,6802,192.168.1.118,internal/6807,,,XML,default,PCMU,8000,64000,PCMU,8000,64000,,ola,,,ACTIVE,,,,dc11d8a7-3a1c-42d6-a936-577438b44a8c ef2aa92a-dad1-4738-99a2-0f3266b1f79f,outbound,2011-12-21 22:14:49,1324476889,sofia/internal/sip:6800 at 192.168.1.210:5062,CS_EXECUTE,OLA-QUEUE,00000000,,6800,eavesdrop,dc11d8a7-3a1c-42d6-a936-577438b44a8c,,default,L16,8000,128000,PCMU,8000,64000,,ola,6800 at 192.168.1.200,,ACTIVE,Outbound Call,6800,RECV,6a208fec-2297-407f-ba74-446122a91974 eavesdrop sccp uuid uuid,direction,created,created_epoch,name,state,cid_name,cid_num,ip_addr,dest,application,application_data,dialplan,context,read_codec,read_rate,read_bit_rate,write_codec,write_rate,write_bit_rate,secure,hostname,presence_id,presence_data,callstate,callee_name,callee_num,callee_direction,call_uuid 1f5118b4-2a9f-4682-adf6-1a378b381ee3,inbound,2011-12-21 22:13:19,1324476799,sofia/internal/6802 at 6802:5060,CS_EXECUTE,6802,6802,192.168.1.118,6807,bridge,skinny/internal/6807,XML,default,PCMU,8000,64000,PCMU,8000,64000,,ola,6802 at 6802,,ACTIVE,Outbound Call,internal/6807,SEND,1f5118b4-2a9f-4682-adf6-1a378b381ee3 7dfb8790-3d0d-4fbf-a716-7bf31285b5b0,outbound,2011-12-21 22:13:19,1324476799,SKINNY/internal/6807,CS_EXCHANGE_MEDIA,6802,6802,192.168.1.118,internal/6807,,,XML,default,PCMU,8000,64000,PCMU,8000,64000,,ola,,,ACTIVE,,,,1f5118b4-2a9f-4682-adf6-1a378b381ee3 df975188-fe2a-4817-b046-f911bf158447,outbound,2011-12-21 22:13:32,1324476812,sofia/internal/sip:6800 at 192.168.1.210:5062,CS_EXECUTE,OLA-QUEUE,00000000,,6800,eavesdrop,7dfb8790-3d0d-4fbf-a716-7bf31285b5b0,,default,L16,8000,128000,PCMU,8000,64000,,ola,6800 at 192.168.1.200,,ACTIVE,Outbound Call,6800,RECV,14fe5a58-75ea-416e-9f3d-30c57b7fa626 thanks. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj:? http://www.freeswitch.org.cn From peter.olsson at visionutveckling.se Wed Dec 21 10:00:01 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 21 Dec 2011 08:00:01 +0100 Subject: [Freeswitch-users] Problem with make current Message-ID: Try "make spandsp-reconf" Or just do a ./bootstrap.sh and ./configure. /Peter ----- Reply message ----- Fr?n: "lloyd thomas" Datum: ons, dec 21, 2011 03:05 Rubrik: [Freeswitch-users] Problem with make current Till: "freeswitch-users" Since I am not getting anywhere with my sip registration problem I decided to update FS from git using make current. Unfortunately I am getting the following errors. What can I do to resolve? In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:101, from ./src/include/private/switch_core_pvt.h:35, from src/switch_apr.c:37: /usr/src/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:145: error: expected declaration specifiers or ?...? before ?tz_t? make[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1 make[2]: Leaving directory `/usr/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2 !DSPAM:4ef13c9932768811635433! From joe.jflemmings at gmail.com Wed Dec 21 11:08:24 2011 From: joe.jflemmings at gmail.com (Joe Flemmings) Date: Wed, 21 Dec 2011 00:08:24 -0800 Subject: [Freeswitch-users] Call/Transaction Does Not Exist In-Reply-To: References: Message-ID: Sorry was late in testing this but i'm still having the same problem on all kinds of phones and after updating FreeSwitch. Please see FreeSwitch logs here http://pastebin.freeswitch.org/18052 The database trasaction log is http://pastebin.freeswitch.org/18053 On Thu, Oct 13, 2011 at 2:39 PM, Joe Flemmings wrote: > Actually just updated to the latest Freeswitch and its working. > > If i don't write back then consider this to be working. > > Thank you > > > On Thu, Oct 13, 2011 at 2:21 PM, Anthony Minessale < > anthony.minessale at gmail.com> wrote: > >> 4 things to try in order of likelihood to help but you should do all 4 >> regardless. >> >> 1) edit your sofia profile and add > value="true"/> >> 2) Update your polycom to latest firmware >> 3) Update to latest FreeSWITCH >> 4) make sure your network router does not have any SIP ALG enabled. >> >> >> On Wed, Oct 12, 2011 at 8:34 PM, Joe Flemmings >> wrote: >> > Thanks for the help Anthony, >> > >> > Bellow is the pastebin URL >> > >> > http://pastebin.freeswitch.org/17518 >> > >> > Joe >> > >> > On Wed, Oct 12, 2011 at 8:51 AM, Anthony Minessale >> > wrote: >> >> >> >> collect a full trace and put it on pastebin.freeswitch.org >> >> >> >> console loglevel debug (/log debug on fs_cli) >> >> sofia global siptrace on >> >> >> >> >> >> >> >> >> >> On Tue, Oct 11, 2011 at 4:29 PM, Joe Flemmings < >> joe.jflemmings at gmail.com> >> >> wrote: >> >> > When i call from one extension and hangup, the other call does not >> >> > hangup >> >> > and continues ringing. Please see log bellow. >> >> > >> >> > Any ideas? >> >> > >> >> > >> >> > >> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060 >> >> > INVITE sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: >> SIP/2.0/UDP >> >> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom >> One" >> >> > ;tag=F5263502-30CB1633..To: >> >> > ..CSeq: 1 INVITE..Call-ID: >> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact: >> >> > ..Allow: INVITE, ACK, BYE, >> CANCEL, >> >> > OPTIONS, INFO, MES >> >> > SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: >> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language: >> >> > en..Supported: >> >> > 100rel,replaces..Allow-Events: talk,hold,conference..Max-Forwards: >> >> > 70..Content-Type: application/sdp..Content-Length: 294....v=0..o=- >> >> > 1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone..c=IN >> IP4 >> >> > 10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 0 8 18 >> >> > 101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 >> >> > PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 >> >> > telephone-event/8000.. >> >> > # >> >> > U 10.10.78.36:5060 -> 10.10.78.24:52732 >> >> > SIP/2.0 100 Trying..Via: SIP/2.0/UDP >> >> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom >> One" >> >> > ;tag=F5263502-30CB1633..To: >> >> > >> >> > ..Call-ID: 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 1 >> >> > INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba >> 2011-05-23 >> >> > 22-51-43 -0500..Content-Length: 0.... >> >> > # >> >> > U 10.10.78.36:5060 -> 10.10.78.24:52732 >> >> > SIP/2.0 407 Proxy Authentication Required..Via: SIP/2.0/UDP >> >> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom >> One" >> >> > ;tag=F5263502-30CB1633..To: > >> > 6.113.78.36;user=phone>;tag=71XvH76tyavaF..Call-ID: >> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 1 INVITE..User-Agent: >> >> > FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-23 22-51-43 >> >> > -0500..Accept: >> >> > appl >> >> > ication/sdp..Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, >> >> > UPDATE, >> >> > INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE..Supported: timer, >> >> > precondition, path, replaces..Allow-Events: talk, hold, prese >> >> > nce, dialog, line-seize, call-info, sla, >> include-session-description, >> >> > presence.winfo, message-summary, refer..Proxy-Authenticate: Digest >> >> > realm="10.10.78.36", nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac >> >> > ", algorithm=MD5, qop="auth"..Content-Length: 0.... >> >> > # >> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060 >> >> > ACK sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: SIP/2.0/UDP >> >> > 10.10.78.24:52732;branch=z9hG4bK5c8ef878DBE4BA61..From: "Polycom >> One" >> >> > ;tag=F5263502-30CB1633..To: > >> > p:213 at 10.10.78.36;user=phone>;tag=71XvH76tyavaF..CSeq: 1 >> ACK..Call-ID: >> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact: >> >> > ..Allow: INVITE, ACK, BYE, >> CANCEL, >> >> > OPTION >> >> > S, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, >> REFER..User-Agent: >> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language: >> >> > en..Max-Forwards: 70..Content-Length: 0.... >> >> > # >> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060 >> >> > INVITE sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: >> SIP/2.0/UDP >> >> > 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom >> One" >> >> > ;tag=F5263502-30CB1633..To: >> >> > ..CSeq: 2 INVITE..Call-ID: >> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact: >> >> > ..Allow: INVITE, ACK, BYE, >> CANCEL, >> >> > OPTIONS, INFO, MES >> >> > SAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: >> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Accept-Language: >> >> > en..Supported: >> >> > 100rel,replaces..Allow-Events: talk,hold,conference..Proxy-Authoriz >> >> > ation: Digest username="pbxjker_2564", realm="10.10.78.36", >> >> > nonce="ab4a9421-fbb9-41fd-943d-46289eee03ac", qop=auth, >> >> > cnonce="KnfhF00Oi0GfJCi", nc=00000001, >> >> > uri="sip:213 at 10.10.78.36:5060;user=phone", >> >> > response="82a75edb7ba97bfe7dea585684701d0c", >> >> > algorithm=MD5..Max-Forwards: >> >> > 70..Content-Type: application/sdp..Content-Length: 294....v=0..o=- >> >> > 1318334822 1318334822 IN IP4 10.10.78.24..s=Polycom IP Phone >> >> > ..c=IN IP4 10.10.78.24..t=0 0..a=sendrecv..m=audio 2252 RTP/AVP 9 >> 0 8 >> >> > 18 >> >> > 101..a=rtpmap:9 G722/8000..a=rtpmap:0 PCMU/8000..a=rtpmap:8 >> >> > PCMA/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:101 >> >> > telephone-event/8000.. >> >> > # >> >> > U 10.10.78.36:5060 -> 10.10.78.24:52732 >> >> > SIP/2.0 100 Trying..Via: SIP/2.0/UDP >> >> > 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom >> One" >> >> > ;tag=F5263502-30CB1633..To: >> >> > >> >> > ..Call-ID: 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 >> >> > INVITE..User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-1086cba >> 2011-05-23 >> >> > 22-51-43 -0500..Content-Length: 0.... >> >> > # >> >> > U 10.10.78.36:5060 -> 10.10.78.24:52783 >> >> > INVITE sip:pbxjker_213 at 10.10.78.24:52783 SIP/2.0..Via: SIP/2.0/UDP >> >> > 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..Max-Forwards: >> 69..From: >> >> > "Polycom One" ;tag=9KgeNX81 >> >> > rv8Fp..To: ..Call-ID: >> >> > 4cedf474-6ee4-122f-0f99-00093d12d5dd..CSeq: 18845076 INVITE..Contact: >> >> > ..User-Agent: >> FreeSWITCH-mod_sofia/1.0. >> >> > head-git-1086cba 2011-05-23 22-51-43 -0500..Allow: INVITE, ACK, >> BYE, >> >> > CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, >> >> > PUBLISH, >> >> > SUBSCRIBE..Supported: timer, precondition, path, replaces.. >> >> > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, >> >> > sla, >> >> > include-session-description, presence.winfo, message-summary, >> >> > refer..Content-Type: application/sdp..Content-Disposition: session.. >> >> > Content-Length: 337..X-FS-Support: update_display..Remote-Party-ID: >> >> > "Polycom One" >> >> > >> >> > > >;party=calling;screen=yes;privacy=off....v=0..o=FreeSWITCH >> >> > 1318333156 1318333157 IN IP4 >> >> > 10.10.78.36..s=FreeSWITCH..c=IN IP4 10.10.78.36..t=0 0..m=audio >> 29124 >> >> > RTP/AVP 9 98 99 0 8 3 18 101 13..a=rtpmap:98 G7221/32000..a=fmtp:98 >> >> > bitrate=48000..a=rtpmap:99 G7221/16000..a=fmtp:99 bitrate=320 >> >> > 00..a=fmtp:18 annexb=no..a=rtpmap:101 >> telephone-event/8000..a=fmtp:101 >> >> > 0-16..a=ptime:20.. >> >> > # >> >> > U 10.10.78.24:55516 -> 10.10.78.36:5060 >> >> > SIP/2.0 100 Trying..Via: SIP/2.0/UDP >> >> > 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One" >> >> > ;tag=9KgeNX81rv8Fp..To: "Polycom Two" >> >> > > >> > 17.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076 INVITE..Call-ID: >> >> > 4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact: >> >> > ..User-Agent: >> >> > PolycomSoundPointIP-SPIP_320-UA/3.2 >> >> > .2.0477..Accept-Language: en..Content-Length: 0.... >> >> > # >> >> > U 10.10.78.24:55516 -> 10.10.78.36:5060 >> >> > SIP/2.0 180 Ringing..Via: SIP/2.0/UDP >> >> > 10.10.78.36;rport;branch=z9hG4bKecZ6j164t7j0B..From: "Polycom One" >> >> > ;tag=9KgeNX81rv8Fp..To: "Polycom Two" >> >> > > >> > 217.24:52783>;tag=1EB231C2-C4A8A531..CSeq: 18845076 >> INVITE..Call-ID: >> >> > 4cedf474-6ee4-122f-0f99-00093d12d5dd..Contact: >> >> > ..User-Agent: >> >> > PolycomSoundPointIP-SPIP_320-UA/3. >> >> > 2.2.0477..Allow-Events: talk,hold,conference..Accept-Language: >> >> > en..Content-Length: 0.... >> >> > # >> >> > U 10.10.78.36:5060 -> 10.10.78.24:52732 >> >> > SIP/2.0 180 Ringing..Via: SIP/2.0/UDP >> >> > 10.10.78.24:52732;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom >> One" >> >> > ;tag=F5263502-30CB1633..To: >> >> > > >> > >;tag=8aQNK2QyUKjXa..Call-ID: >> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 INVITE..Contact: >> >> > ..User-Agent: >> >> > FreeSWITCH-mod_sofia/1.0.head-git-1086cba 2011-05-2 >> >> > 3 22-51-43 -0500..Accept: application/sdp..Allow: INVITE, ACK, BYE, >> >> > CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, >> >> > PUBLISH, >> >> > SUBSCRIBE..Supported: timer, precondition, path, replaces..A >> >> > llow-Events: talk, hold, presence, dialog, line-seize, call-info, >> sla, >> >> > include-session-description, presence.winfo, message-summary, >> >> > refer..Content-Length: 0..Remote-Party-ID: "Outbound Call" > >> > _213>;party=calling;privacy=off;screen=no.... >> >> > # >> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060 >> >> > CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: >> SIP/2.0/UDP >> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" >> >> > ;tag=F5263502-30CB1633..To: > >> > 13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID: >> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact: >> >> > ..Allow: INVITE, ACK, BYE, CANCEL, >> >> > OPTIONS, >> >> > INFO, MESSAGE, SUBSCR >> >> > IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: >> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: >> Digest >> >> > username="pbxjker_2564", realm="10.10.78.36", >> >> > nonce="ab4a9421-fbb9-41fd-943d-462 >> >> > 89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002, >> >> > uri="sip:213 at 10.10.78.36:5060;user=phone", >> >> > response="d47848babd8c396d8da15145503875d0", >> >> > algorithm=MD5..Max-Forwards: >> >> > 70..Content-Length: 0.. >> >> > .. >> >> > # >> >> > U 10.10.78.36:5060 -> 10.10.78.24:5060 >> >> > SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP >> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" >> >> > ;tag=F5263502-30CB1633..To: >> > >> > 3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID: >> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 >> CANCEL..Content-Length: >> >> > 0.... >> >> > # >> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060 >> >> > CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: >> SIP/2.0/UDP >> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" >> >> > ;tag=F5263502-30CB1633..To: > >> > 13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID: >> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact: >> >> > ..Allow: INVITE, ACK, BYE, CANCEL, >> >> > OPTIONS, >> >> > INFO, MESSAGE, SUBSCR >> >> > IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: >> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: >> Digest >> >> > username="pbxjker_2564", realm="10.10.78.36", >> >> > nonce="ab4a9421-fbb9-41fd-943d-462 >> >> > 89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002, >> >> > uri="sip:213 at 10.10.78.36:5060;user=phone", >> >> > response="d47848babd8c396d8da15145503875d0", >> >> > algorithm=MD5..Max-Forwards: >> >> > 70..Content-Length: 0.. >> >> > .. >> >> > # >> >> > U 10.10.78.36:5060 -> 10.10.78.24:5060 >> >> > SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP >> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" >> >> > ;tag=F5263502-30CB1633..To: >> > >> > 3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID: >> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 >> CANCEL..Content-Length: >> >> > 0.... >> >> > # >> >> > U 10.10.78.24:55479 -> 10.10.78.36:5060 >> >> > CANCEL sip:213 at 10.10.78.36:5060;user=phone SIP/2.0..Via: >> SIP/2.0/UDP >> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" >> >> > ;tag=F5263502-30CB1633..To: > >> > 13 at 10.10.78.36;user=phone>..CSeq: 2 CANCEL..Call-ID: >> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..Contact: >> >> > ..Allow: INVITE, ACK, BYE, CANCEL, >> >> > OPTIONS, >> >> > INFO, MESSAGE, SUBSCR >> >> > IBE, NOTIFY, PRACK, UPDATE, REFER..User-Agent: >> >> > PolycomSoundPointIP-SPIP_650-UA/3.2.2.0477..Proxy-Authorization: >> Digest >> >> > username="pbxjker_2564", realm="10.10.78.36", >> >> > nonce="ab4a9421-fbb9-41fd-943d-462 >> >> > 89eee03ac", qop=auth, cnonce="KnfhF00Oi0GfJCi", nc=00000002, >> >> > uri="sip:213 at 10.10.78.36:5060;user=phone", >> >> > response="d47848babd8c396d8da15145503875d0", >> >> > algorithm=MD5..Max-Forwards: >> >> > 70..Content-Length: 0.. >> >> > .. >> >> > # >> >> > U 10.10.78.36:5060 -> 10.10.78.24:5060 >> >> > SIP/2.0 481 Call/Transaction Does Not Exist..Via: SIP/2.0/UDP >> >> > 10.10.78.24;branch=z9hG4bKc391a06d627A0FBA..From: "Polycom One" >> >> > ;tag=F5263502-30CB1633..To: >> > >> > 3.78.36;user=phone>;tag=8aQNK2QyUKjXa..Call-ID: >> >> > 424791be-fbadec9f-83bb2654 at 192.168.1.72..CSeq: 2 >> CANCEL..Content-Length: >> >> > 0.... >> >> > # >> >> > >> >> > >> >> > >> >> > FreeSWITCH-users mailing list >> >> > FreeSWITCH-users at lists.freeswitch.org >> >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> > UNSUBSCRIBE: >> http://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 >> >> >> >> >> >> FreeSWITCH-users mailing list >> >> FreeSWITCH-users at lists.freeswitch.org >> >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> >> UNSUBSCRIBE: >> http://lists.freeswitch.org/mailman/options/freeswitch-users >> >> http://www.freeswitch.org >> > >> > >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE: >> http://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 >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/aa601dae/attachment-0001.html From chris at ghosttelecom.com Wed Dec 21 12:32:42 2011 From: chris at ghosttelecom.com (Chris Martineau) Date: Wed, 21 Dec 2011 09:32:42 -0000 Subject: [Freeswitch-users] rtp timestamp issue on pickup Message-ID: <1D10AB188D6CCA46BB4369E3268E36EF402E39@SVR01.ghosttelecom.local> Hi, My specific application requires an incoming call to pickup an existing waiting call. The first call is terminated and parked. The second call rings in and uses intercept to then connect it to the waiting call. The problem I have is that the second call has a lower rtp timestamp than the waiting call causing the client codec to wait until it catches up? i.e client incoming rtp stream timestamp effectively looks like this ... existing stream from freeswitch 37920 38080 38240 On intercept the timestamps then become that of the other calls stream 3040 3200 ... The client then doesn't get any incoming voice until the new timestamps catch up! Is there any way to set freeswitch to maintain the timestamps? Should the client allow for this? Any ideas would be greatly appreciated. Regards Chris -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/203ca19b/attachment.html From avi at avimarcus.net Wed Dec 21 12:41:19 2011 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 21 Dec 2011 11:41:19 +0200 Subject: [Freeswitch-users] G723 usage - need something special for transcoding? Message-ID: Is g723 supposed to work out of the box? I added G723 to the codec negotiation.. and I see it chose it on leg B: Set Codec sofia/external/551137119355 G723/8000 30 ms 240 samples 6300 bits Leg A is... Set Codec sofia/internal/1000 at sip.domain.com PCMU/8000 20 ms 160 samples 64000 bits Leg A can't hear leg B... I don't know about leg B. Is the ptimes an issue..? Shouldn't FS just transcode the call? -Avi Marcus -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/488522b5/attachment.html From peter.olsson at visionutveckling.se Wed Dec 21 12:50:42 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 21 Dec 2011 10:50:42 +0100 Subject: [Freeswitch-users] rtp timestamp issue on pickup In-Reply-To: <1D10AB188D6CCA46BB4369E3268E36EF402E39@SVR01.ghosttelecom.local> References: <1D10AB188D6CCA46BB4369E3268E36EF402E39@SVR01.ghosttelecom.local> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5B279DBC79@cooper> If FS lowers the timestamp FS activates the mark bit, which should cause the client to be able to handle this correctly. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Chris Martineau Skickat: den 21 december 2011 10:33 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] rtp timestamp issue on pickup Hi, My specific application requires an incoming call to pickup an existing waiting call. The first call is terminated and parked. The second call rings in and uses intercept to then connect it to the waiting call. The problem I have is that the second call has a lower rtp timestamp than the waiting call causing the client codec to wait until it catches up? i.e client incoming rtp stream timestamp effectively looks like this ... existing stream from freeswitch 37920 38080 38240 On intercept the timestamps then become that of the other calls stream 3040 3200 ... The client then doesn't get any incoming voice until the new timestamps catch up! Is there any way to set freeswitch to maintain the timestamps? Should the client allow for this? Any ideas would be greatly appreciated. Regards Chris !DSPAM:4ef1a81732761796211860! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/66c8e091/attachment.html From freeswitch-list at puzzled.xs4all.nl Wed Dec 21 14:01:13 2011 From: freeswitch-list at puzzled.xs4all.nl (Patrick Lists) Date: Wed, 21 Dec 2011 12:01:13 +0100 Subject: [Freeswitch-users] Need help with PortAudio In-Reply-To: <201112202245.57010.justlikeef@gmail.com> References: <201112202245.57010.justlikeef@gmail.com> Message-ID: <4EF1BC79.4000903@puzzled.xs4all.nl> On 21-12-11 04:45, Rob Hutton wrote: > I am trying to get portaudio working, and have run into a couple of problems > > > 1) If portaudio cannot access the device, it causes freeswitch to segfault > > > 2) I have set up a dialplan following the intercom example which seems > to be working, but I get no audio. I have tried setting every device > shown in devlist as the output device. > > > 3) When I issue a "pa play". the cli hangs, even if I give it a timeout. > > > Any thoughts? I think someone reported on the mailing list a while back that he solved his troubles with portaudio by upgrading the portaudio source to the latest stable release which currently is: http://www.portaudio.com/archives/pa_stable_v19_20111121.tgz Regards, Patrick From lloydie.t at gmail.com Wed Dec 21 14:20:44 2011 From: lloydie.t at gmail.com (lloyd thomas) Date: Wed, 21 Dec 2011 11:20:44 +0000 Subject: [Freeswitch-users] Problem with make current In-Reply-To: References: Message-ID: I have done bootstrap and configure already, but it made no difference On 21 December 2011 07:00, Peter Olsson wrote: > Try "make spandsp-reconf" > > Or just do a ./bootstrap.sh and ./configure. > > /Peter > > ----- Reply message ----- > Fr?n: "lloyd thomas" > Datum: ons, dec 21, 2011 03:05 > Rubrik: [Freeswitch-users] Problem with make current > Till: "freeswitch-users" > > Since I am not getting anywhere with my sip registration problem I decided > to update FS from git using make current. > Unfortunately I am getting the following errors. What can I do to resolve? > > In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:101, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch_apr.c:37: > /usr/src/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:145: error: expected > declaration specifiers or ?...? before ?tz_t? > make[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1 > make[2]: Leaving directory `/usr/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/usr/src/freeswitch' > make: *** [current] Error 2 > > !DSPAM:4ef13c9932768811635433! > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/afe7f1c0/attachment-0001.html From valter at fastway.com.br Wed Dec 21 14:39:52 2011 From: valter at fastway.com.br (Valter Nogueira) Date: Wed, 21 Dec 2011 09:39:52 -0200 Subject: [Freeswitch-users] Freeswitch inside-out Message-ID: I have pointed out my emacs+cscope to freeswtich source code and I am trying to figure out how does it work? So far, I figure out that switch_loadable_module_init creates one thread per runtime module and that task_thread_loop plays an important role in running freeswitch. Is there any doc about freeswitch internals, since I yet want to discover how channels are initiated and bridged together (and how the "talk" flows between them) thanks, Valter * * -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/d2e3a926/attachment.html From gmaruzz at gmail.com Wed Dec 21 14:44:57 2011 From: gmaruzz at gmail.com (Giovanni Maruzzelli) Date: Wed, 21 Dec 2011 12:44:57 +0100 Subject: [Freeswitch-users] Freeswitch inside-out In-Reply-To: References: Message-ID: wiki! http://wiki.freeswitch.org/wiki/Core_Outline_of_FreeSWITCH http://wiki.freeswitch.org/wiki/Documentation/Developer_Documentation http://wiki.freeswitch.org/wiki/Authoring_Freeswitch_Modules http://wiki.freeswitch.org/wiki/Creating_a_new_Abstraction_Interface -giovanni On Wed, Dec 21, 2011 at 12:39 PM, Valter Nogueira wrote: > I have pointed out my emacs+cscope to freeswtich source code and I am > trying to figure out how does it work? > > So far, I figure out that switch_loadable_module_init creates one thread > per runtime module and that task_thread_loop plays an important role in > running > freeswitch. > > Is there any doc about freeswitch internals, since I yet want to discover > how channels are initiated and bridged together (and how the "talk" flows > between them) > > thanks, > > Valter > > > * > * > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/66bbe891/attachment.html From chris at ghosttelecom.com Wed Dec 21 15:32:49 2011 From: chris at ghosttelecom.com (Chris Martineau) Date: Wed, 21 Dec 2011 12:32:49 -0000 Subject: [Freeswitch-users] rtp timestamp issue on pickup In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5B279DBC79@cooper> References: <1D10AB188D6CCA46BB4369E3268E36EF402E39@SVR01.ghosttelecom.local> <549CFEF87AEDE841A38E9D15EAB4C04C5B279DBC79@cooper> Message-ID: <1D10AB188D6CCA46BB4369E3268E36EF402E77@SVR01.ghosttelecom.local> Thanks for the reply, Looking at the wireshark it doesn't seem to activate the mark bit again? It's not that freeswitch lowers the ts it is just forwarding the incoming timestamp which is out of synch with the timestamp of the parked call, maybe this is why it is not being set? Any other ideas? Thanks Chris From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson Sent: 21 December 2011 09:51 To: 'freeswitch-users at lists.freeswitch.org' Subject: Re: [Freeswitch-users] rtp timestamp issue on pickup If FS lowers the timestamp FS activates the mark bit, which should cause the client to be able to handle this correctly. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Chris Martineau Skickat: den 21 december 2011 10:33 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] rtp timestamp issue on pickup Hi, My specific application requires an incoming call to pickup an existing waiting call. The first call is terminated and parked. The second call rings in and uses intercept to then connect it to the waiting call. The problem I have is that the second call has a lower rtp timestamp than the waiting call causing the client codec to wait until it catches up? i.e client incoming rtp stream timestamp effectively looks like this ... existing stream from freeswitch 37920 38080 38240 On intercept the timestamps then become that of the other calls stream 3040 3200 ... The client then doesn't get any incoming voice until the new timestamps catch up! Is there any way to set freeswitch to maintain the timestamps? Should the client allow for this? Any ideas would be greatly appreciated. Regards Chris !DSPAM:4ef1a81732761796211860! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/ab4974e7/attachment.html From peter.olsson at visionutveckling.se Wed Dec 21 15:46:27 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 21 Dec 2011 13:46:27 +0100 Subject: [Freeswitch-users] rtp timestamp issue on pickup In-Reply-To: <1D10AB188D6CCA46BB4369E3268E36EF402E77@SVR01.ghosttelecom.local> References: <1D10AB188D6CCA46BB4369E3268E36EF402E39@SVR01.ghosttelecom.local> <549CFEF87AEDE841A38E9D15EAB4C04C5B279DBC79@cooper> <1D10AB188D6CCA46BB4369E3268E36EF402E77@SVR01.ghosttelecom.local> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5B279DBDEC@cooper> Try to enable rtp-rewrite-timestamps according to http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio, that might help. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Chris Martineau Skickat: den 21 december 2011 13:33 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] rtp timestamp issue on pickup Thanks for the reply, Looking at the wireshark it doesn't seem to activate the mark bit again? It's not that freeswitch lowers the ts it is just forwarding the incoming timestamp which is out of synch with the timestamp of the parked call, maybe this is why it is not being set? Any other ideas? Thanks Chris From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson Sent: 21 December 2011 09:51 To: 'freeswitch-users at lists.freeswitch.org' Subject: Re: [Freeswitch-users] rtp timestamp issue on pickup If FS lowers the timestamp FS activates the mark bit, which should cause the client to be able to handle this correctly. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Chris Martineau Skickat: den 21 december 2011 10:33 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] rtp timestamp issue on pickup Hi, My specific application requires an incoming call to pickup an existing waiting call. The first call is terminated and parked. The second call rings in and uses intercept to then connect it to the waiting call. The problem I have is that the second call has a lower rtp timestamp than the waiting call causing the client codec to wait until it catches up? i.e client incoming rtp stream timestamp effectively looks like this ... existing stream from freeswitch 37920 38080 38240 On intercept the timestamps then become that of the other calls stream 3040 3200 ... The client then doesn't get any incoming voice until the new timestamps catch up! Is there any way to set freeswitch to maintain the timestamps? Should the client allow for this? Any ideas would be greatly appreciated. Regards Chris !DSPAM:4ef1d21e32762117429169! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/f5648361/attachment-0001.html From ib-freeswitch at bzsolutions.it Wed Dec 21 16:54:44 2011 From: ib-freeswitch at bzsolutions.it (ib-freeswitch at bzsolutions.it) Date: Wed, 21 Dec 2011 14:54:44 +0100 (CET) Subject: [Freeswitch-users] G723 usage - need something special for transcoding? In-Reply-To: <32467084.17381324475577103.JavaMail.javamailuser@localhost> Message-ID: <6251787.17401324475684086.JavaMail.javamailuser@localhost> Hi, freeswitch supports G.723 only in pass through, if you want to transcode, there is a sangoma card that can do it, commercial license is avaiable only for G.729. G723 / G729 / AMR are patented, and you can't use use the algorithm without permission of patent holders. If ALEG and BLEG support G.723 you can change codec negotiation mode in freeswitch. You can found usefull info here on WiKi: http://wiki.freeswitch.org/wiki/Codec_negotiation ----- Original Message ----- From: "Avi Marcus" To: "FreeSWITCH Users Help" Sent: Wednesday, December 21, 2011 10:41:19 AM GMT +01:00 Amsterdam / Berlin / Bern / Rome / Stockholm / Vienna Subject: [Freeswitch-users] G723 usage - need something special for transcoding? Is g723 supposed to work out of the box? I added G723 to the codec negotiation.. and I see it chose it on leg B: Set Codec sofia/external/551137119355 G723/8000 30 ms 240 samples 6300 bits Leg A is... Set Codec sofia/internal/ 1000 at sip.domain.com PCMU/8000 20 ms 160 samples 64000 bits Leg A can't hear leg B... I don't know about leg B. Is the ptimes an issue..? Shouldn't FS just transcode the call? -Avi Marcus _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From avi at avimarcus.net Wed Dec 21 17:06:37 2011 From: avi at avimarcus.net (Avi Marcus) Date: Wed, 21 Dec 2011 16:06:37 +0200 Subject: [Freeswitch-users] G723 usage - need something special for transcoding? In-Reply-To: <6251787.17401324475684086.JavaMail.javamailuser@localhost> References: <32467084.17381324475577103.JavaMail.javamailuser@localhost> <6251787.17401324475684086.JavaMail.javamailuser@localhost> Message-ID: Ah, well, that would explain it then.. are there any other common open licenses other than pcmu/pcma for PSTN interconnects? e.g. they don't offer speex.. heck, the underlying carrier doesn't have g711. -Avi Marcus On Wed, Dec 21, 2011 at 3:54 PM, wrote: > Hi, > freeswitch supports G.723 only in pass through, if you want to transcode, > there is a sangoma card that can do it, commercial license is avaiable only > for G.729. > > G723 / G729 / AMR are patented, and you can't use use the algorithm > without permission of patent holders. > > If ALEG and BLEG support G.723 you can change codec negotiation mode in > freeswitch. > > You can found usefull info here on WiKi: > http://wiki.freeswitch.org/wiki/Codec_negotiation > > > ----- Original Message ----- > From: "Avi Marcus" > To: "FreeSWITCH Users Help" > Sent: Wednesday, December 21, 2011 10:41:19 AM GMT +01:00 Amsterdam / > Berlin / Bern / Rome / Stockholm / Vienna > Subject: [Freeswitch-users] G723 usage - need something special for > transcoding? > > > > Is g723 supposed to work out of the box? > I added G723 to the codec negotiation.. and I see it chose it on leg B: > Set Codec sofia/external/551137119355 G723/8000 30 ms 240 samples 6300 bits > Leg A is... Set Codec sofia/internal/ 1000 at sip.domain.com PCMU/8000 20 ms > 160 samples 64000 bits > > > Leg A can't hear leg B... I don't know about leg B. Is the ptimes an > issue..? Shouldn't FS just transcode the call? > > > -Avi Marcus > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/59d816ad/attachment.html From ib-freeswitch at bzsolutions.it Wed Dec 21 17:27:56 2011 From: ib-freeswitch at bzsolutions.it (ib-freeswitch at bzsolutions.it) Date: Wed, 21 Dec 2011 15:27:56 +0100 (CET) Subject: [Freeswitch-users] G723 usage - need something special for transcoding? In-Reply-To: <3273741.17461324477590875.JavaMail.javamailuser@localhost> Message-ID: <32239653.17481324477676272.JavaMail.javamailuser@localhost> Usually G729 it's the standard, if you phone support it, you don't need to do transcoding, test it in dialplan: .... bridge bla bla bla.... http://wiki.freeswitch.org/wiki/Variable_inherit_codec Set on the phone only the codecs supperted by providers and it will work... An usefull tools to debug what happen it's SIPGREP You can find it in __WHERE_YOU_HAVE_FS_SOURCES/freeswitch/scripts/trace/sipgrep ----- Original Message ----- From: "Avi Marcus" To: "FreeSWITCH Users Help" Sent: Wednesday, December 21, 2011 3:06:37 PM GMT +01:00 Amsterdam / Berlin / Bern / Rome / Stockholm / Vienna Subject: Re: [Freeswitch-users] G723 usage - need something special for transcoding? Ah, well, that would explain it then.. are there any other common open licenses other than pcmu/pcma for PSTN interconnects? e.g. they don't offer speex.. heck, the underlying carrier doesn't have g711. -Avi Marcus On Wed, Dec 21, 2011 at 3:54 PM, < ib-freeswitch at bzsolutions.it > wrote: Hi, freeswitch supports G.723 only in pass through, if you want to transcode, there is a sangoma card that can do it, commercial license is avaiable only for G.729. G723 / G729 / AMR are patented, and you can't use use the algorithm without permission of patent holders. If ALEG and BLEG support G.723 you can change codec negotiation mode in freeswitch. You can found usefull info here on WiKi: http://wiki.freeswitch.org/wiki/Codec_negotiation ----- Original Message ----- From: "Avi Marcus" < avi at avimarcus.net > To: "FreeSWITCH Users Help" < FreeSWITCH-users at lists.freeswitch.org > Sent: Wednesday, December 21, 2011 10:41:19 AM GMT +01:00 Amsterdam / Berlin / Bern / Rome / Stockholm / Vienna Subject: [Freeswitch-users] G723 usage - need something special for transcoding? Is g723 supposed to work out of the box? I added G723 to the codec negotiation.. and I see it chose it on leg B: Set Codec sofia/external/551137119355 G723/8000 30 ms 240 samples 6300 bits Leg A is... Set Codec sofia/internal/ 1000 at sip.domain.com PCMU/8000 20 ms 160 samples 64000 bits Leg A can't hear leg B... I don't know about leg B. Is the ptimes an issue..? Shouldn't FS just transcode the call? -Avi Marcus _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE: http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From boris at tagnet.ru Wed Dec 21 17:34:13 2011 From: boris at tagnet.ru (Boris Kovalenko) Date: Wed, 21 Dec 2011 20:34:13 +0600 Subject: [Freeswitch-users] Linksys PAP2-T and echo In-Reply-To: References: <4EAEB77C.9060901@tagnet.ru> <4EAEBAB7.1030405@coppice.org> <3B1EABFB29D34F61BC96746169E2E2C4@DWP> <4EAECFFD.2090502@tagnet.ru> <4EAEDDCA.4070907@coppice.org> <4EAF5BE0.4050305@tagnet.ru> Message-ID: <4EF1EE65.40703@tagnet.ru> Hello! I found the root of echo problem - it is DECT. Echo present only when we are talking via DECT phones. Yes, I know DECT means additional coding + delay. So... it is possible to eleminate echo for DECT phones too? > Hi Boris, what kind of analogue handsets do you have attached to the > PAP2-T's? I have found with other ATA's (mostly Grandstreams) that it is > the impedance settings for the handset that contributes the most to > echo. Different handsets can be seup for different countries and > sometimes you have to adjust the ATA to match. > > > Regards > Brian > > > > On 01/11/2011 02:39, Boris Kovalenko wrote: >> Hello! >> >> Steve, my configuration is: >> >> PAP2-T (A) ---- FreeSwitch --- PAP-2T (B) >> I tried Freeswitch in all modes: default, proxy_media, bypass_media. The >> echo is always present. User A hears its own echo, and user B hear its own. >> >>> Hi, >>> >>> The echo canceller in the SPA3102 is a disaster, but the ones in the >>> PAP2T and SPA2102 shouldn't give you problems. If you use the default >>> gains you shouldn't really notice echo. If you set some wild gains you >>> might bee poor results. Are you sure the echo originates from the PAP2T, >>> and not some other part of the signal chain? >>> >>> Steve >>> >>> >>> On 11/01/2011 12:42 AM, Boris Kovalenko wrote: >>>> Hello! >>>> >>>> Steve, Darcy, I've played ... no success :( Setting FXS to -15 do echo >>>> acceptable but I still can hear it. So there is no way to eleminate echo >>>> with Linksys devices? Only to buy more professional devices like Addpac? >>>>> on your pap2t web interface, go to voice, advanced, select the line, the >>>>> settings below will allow you to play with echo cancellation. However, you >>>>> also need to go the the regional tab in voice and play with the fxs input >>>>> and output gains, the voice gain on the pap2t or any linksys gateway, will >>>>> impact the echo on the line. This is most likely where your problem is. >>>>> You might not be able to cure the echo on every call, but you should be able >>>>> to make it acceptable. >>>>> >>>>> Darcy >>>>> >>>>> >>>>> Audio Configuration >>>>> Preferred Codec: Second Preferred Codec: >>>>> Third Preferred Codec: Use Pref Codec Only: >>>>> Silence Supp Enable: Silence Threshold: >>>>> G729a Enable: Echo Canc Enable: >>>>> G723 Enable: Echo Canc Adapt Enable: >>>>> G726-16 Enable: Echo Supp Enable: >>>>> G726-24 Enable: FAX CED Detect Enable: >>>>> G726-32 Enable: FAX CNG Detect Enable: >>>>> G726-40 Enable: FAX Passthru Codec: >>>>> DTMF Process INFO: FAX Codec Symmetric: >>>>> DTMF Process AVT: FAX Passthru Method: >>>>> DTMF Tx Method: DTMF Tx Mode: >>>>> DTMF Tx Strict Hold Off Time: FAX Process NSE: >>>>> Hook Flash Tx Method: FAX Disable ECAN: >>>>> Release Unused Codec: FAX Enable T38: >>>>> FAX T38 Redundancy: FAX Tone Detect Mode: >>>>> >>>>> -----Original Message----- >>>>> From: Steve Underwood >>>>> Sent: Monday, October 31, 2011 11:11 AM >>>>> To: FreeSWITCH Users Help >>>>> Subject: Re: [Freeswitch-users] Linksys PAP2-T and echo >>>>> >>>>> On 10/31/2011 10:58 PM, Boris Kovalenko wrote: >>>>>> Hello! >>>>>> >>>>>> >>>>>> I'm wondering if it is possible to solve echo with Linksys PAP2-T >>>>>> devices. Does anybody here uses it? I read many articles and still can't >>>>>> understand how to eleminate echo :( I read >>>>>> http://www.voip-info.org/wiki/view/Asterisk+echo+cancellation and found >>>>>> that Asterisk has its own EC. Does Freeswitch has it too? I found this >>>>>> article http://docs.freeswitch.org/echo_can_page.html but can't >>>>>> understand - has Freeswitch EC or not? If yes - how to turn it on and off? >>>>> Neither Asterisk or Freeswitch will echo cancel your PAP2T. Echo >>>>> cancellation over IP is very problematic, and hardly ever attempted. The >>>>> PAP2T should be echo cancelling for itself, and they usually do a fairly >>>>> good job of this. I don't think it is configurable. Its always on. >>>>> >>>>> Steve > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Regards, Boris From xing2kin at yahoo.com Wed Dec 21 17:50:43 2011 From: xing2kin at yahoo.com (king2kin) Date: Wed, 21 Dec 2011 06:50:43 -0800 (PST) Subject: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) In-Reply-To: References: Message-ID: <1324479043.53899.YahooMailNeo@web39705.mail.mud.yahoo.com> No, I didn't remove any tables. I just compiled and run it with few settings updated (e.g. sounds, odbc). From: Peter Olsson To: "freeswitch-users at lists.freeswitch.org" Sent: Monday, December 19, 2011 2:50 PM Subject: Re: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) Did you try to remove all tables, so FS can create everything from scratch? /Peter ----- Reply message ----- Fr?n: "king2kin" Datum: m?n, dec 19, 2011 07:11 Rubrik: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) Till: "FreeSWITCH Users Help" No, this occurs on the latest GIT version of freeswitch that I downloaded on Dec. 18, 2011 by command: { git clone git://git.freeswitch.org/freeswitch.git } The early GIT version (e.g. 2011-08-31) doesn't have such problems. The latest version can run on windows 2003, but keep printing out error messages in freeswitch.log and console, for example: [DEBUG] switch_core_sqldb.c:890 SQL ERR [no such table: sip_registrations] [delete from sip_registrations where (contact like '%TCP%' or status like '%TCP%' or status like '%TLS%') and hostname='W2k3T602' and network_ip like '%' and network_port like '%' and sip_username like '%' and mwi_user? like '%' and mwi_host like '%' and orig_server_host like '%' and orig_hostname like '%'] Auto Generating Table! [DEBUG] switch_core_sqldb.c:897 SQL ERR [no such table: sip_subscriptions] [DEBUG] switch_core_sqldb.c:897 SQL ERR [no such table: sip_presence] [DEBUG] switch_core_sqldb.c:890 SQL ERR [no such table: sip_dialogs] ...... also, while running? the new version of freeswitch, the following error messages keep coming up in freeswitch.log: { 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it } From: Brian West To: FreeSWITCH Users Help Sent: Monday, December 19, 2011 9:24 AM Subject: Re: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) You're using a very old version of FreeSWITCH... why ? /b On Dec 18, 2011, at 7:11 PM, king2kin wrote: No, these tables were not created automatically by freeswitch on windows 2003 while MySQL is running. During the past three months, I had been playing around the GIT version of 2011-08-31, whose log file [freeswitch.log] didn't complain those database tables that I listed in my previous email. It seems to me that these db tables come up only in more recent GIT version. From: Jeff Lenk > To: freeswitch-users at lists.freeswitch.org Sent: Monday, December 19, 2011 1:30 AM Subject: Re: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) All the required tables should be created on first run. Same as always. Is that not happening? -- 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 !DSPAM:4eeed36a32763536013952! _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/3df240d5/attachment-0001.html From peter.olsson at visionutveckling.se Wed Dec 21 18:02:25 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 21 Dec 2011 16:02:25 +0100 Subject: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) In-Reply-To: <1324479043.53899.YahooMailNeo@web39705.mail.mud.yahoo.com> References: <1324479043.53899.YahooMailNeo@web39705.mail.mud.yahoo.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5B279DBEC8@cooper> So... That's a good start then - just to see if it's created correctly. It's supposed to auto-recreate everything though. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r king2kin Skickat: den 21 december 2011 15:51 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) No, I didn't remove any tables. I just compiled and run it with few settings updated (e.g. sounds, odbc). From: Peter Olsson > To: "freeswitch-users at lists.freeswitch.org" > Sent: Monday, December 19, 2011 2:50 PM Subject: Re: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) Did you try to remove all tables, so FS can create everything from scratch? /Peter ----- Reply message ----- Fr?n: "king2kin" > Datum: m?n, dec 19, 2011 07:11 Rubrik: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) Till: "FreeSWITCH Users Help" > No, this occurs on the latest GIT version of freeswitch that I downloaded on Dec. 18, 2011 by command: { git clone git://git.freeswitch.org/freeswitch.git } The early GIT version (e.g. 2011-08-31) doesn't have such problems. The latest version can run on windows 2003, but keep printing out error messages in freeswitch.log and console, for example: [DEBUG] switch_core_sqldb.c:890 SQL ERR [no such table: sip_registrations] [delete from sip_registrations where (contact like '%TCP%' or status like '%TCP%' or status like '%TLS%') and hostname='W2k3T602' and network_ip like '%' and network_port like '%' and sip_username like '%' and mwi_user like '%' and mwi_host like '%' and orig_server_host like '%' and orig_hostname like '%'] Auto Generating Table! [DEBUG] switch_core_sqldb.c:897 SQL ERR [no such table: sip_subscriptions] [DEBUG] switch_core_sqldb.c:897 SQL ERR [no such table: sip_presence] [DEBUG] switch_core_sqldb.c:890 SQL ERR [no such table: sip_dialogs] ...... also, while running the new version of freeswitch, the following error messages keep coming up in freeswitch.log: { 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it 2011-12-18 03:21:13.578125 [WARNING] switch_xml.c:2328 Invalid UTF-8 character to ampersand, skip it } From: Brian West > To: FreeSWITCH Users Help > Sent: Monday, December 19, 2011 9:24 AM Subject: Re: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) You're using a very old version of FreeSWITCH... why ? /b On Dec 18, 2011, at 7:11 PM, king2kin wrote: No, these tables were not created automatically by freeswitch on windows 2003 while MySQL is running. During the past three months, I had been playing around the GIT version of 2011-08-31, whose log file [freeswitch.log] didn't complain those database tables that I listed in my previous email. It seems to me that these db tables come up only in more recent GIT version. From: Jeff Lenk >> To: freeswitch-users at lists.freeswitch.org> Sent: Monday, December 19, 2011 1:30 AM Subject: Re: [Freeswitch-users] Where are table definitions of database for freewsitch (latest GIT version) All the required tables should be created on first run. Same as always. Is that not happening? -- 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 !DSPAM:4ef1f26932769813811637! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/098fcfd1/attachment.html From peter.olsson at visionutveckling.se Wed Dec 21 18:47:58 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 21 Dec 2011 16:47:58 +0100 Subject: [Freeswitch-users] Problem with make current In-Reply-To: References: Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5B279DBF1C@cooper> Try to remove the autogenerated file "/usr/src/freeswitch/libs/spandsp/src/spandsp.h", then do a "make spandsp-reconf", see if that helps. Last way out is probably to remove the source and do a fresh git clone, but it should work anyway. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r lloyd thomas Skickat: den 21 december 2011 12:21 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Problem with make current I have done bootstrap and configure already, but it made no difference On 21 December 2011 07:00, Peter Olsson > wrote: Try "make spandsp-reconf" Or just do a ./bootstrap.sh and ./configure. /Peter ----- Reply message ----- Fr?n: "lloyd thomas" > Datum: ons, dec 21, 2011 03:05 Rubrik: [Freeswitch-users] Problem with make current Till: "freeswitch-users" > Since I am not getting anywhere with my sip registration problem I decided to update FS from git using make current. Unfortunately I am getting the following errors. What can I do to resolve? In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:101, from ./src/include/private/switch_core_pvt.h:35, from src/switch_apr.c:37: /usr/src/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:145: error: expected declaration specifiers or ?...? before ?tz_t? make[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1 make[2]: Leaving directory `/usr/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at 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:4ef1c05f32768994868157! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/6d8b15cf/attachment-0001.html From ndavis at inetwork.com Wed Dec 21 19:16:41 2011 From: ndavis at inetwork.com (Neil Davis) Date: Wed, 21 Dec 2011 09:16:41 -0700 Subject: [Freeswitch-users] Threads remain after calling close on Java client In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD0225507041@NY1-EXMB-01.ip-soft.net> References: <8b41de351c0d1365e3786e7a60645275@mail.gmail.com> <6A6B4C284AD15042B429EB9D904544AD0225507041@NY1-EXMB-01.ip-soft.net> Message-ID: <895731f6d2c9f7220f752e977f485386@mail.gmail.com> Here is a thread dump of my Tomcat process at the point when it is hanging on shutdown. There are a number of threads in a ?waiting on condition? state that appear to have to do with the netty package on which the Freeswitch client is dependent. 2011-12-21 09:05:17 Full thread dump Java HotSpot(TM) Client VM (14.3-b01 mixed mode): "DestroyJavaVM" prio=6 tid=0x546f7400 nid=0x640 waiting on condition [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "pool-5-thread-16" prio=6 tid=0x544b5800 nid=0x1098 waiting on condition [0x55e4f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-15" prio=6 tid=0x54f7f400 nid=0x10f0 waiting on condition [0x54e5f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-14" prio=6 tid=0x544b3c00 nid=0xdb4 waiting on condition [0x5670f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-13" prio=6 tid=0x544b4000 nid=0x12c8 waiting on condition [0x566bf000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-12" prio=6 tid=0x544b4800 nid=0xe18 waiting on condition [0x5666f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-11" prio=6 tid=0x544b3000 nid=0x620 waiting on condition [0x5661f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-10" prio=6 tid=0x552a1800 nid=0x131c waiting on condition [0x565cf000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-9" prio=6 tid=0x552a1400 nid=0x1710 waiting on condition [0x5657f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-8" prio=6 tid=0x552a0c00 nid=0x1094 waiting on condition [0x5652f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-7" prio=6 tid=0x552a0800 nid=0x1040 waiting on condition [0x564df000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-6" prio=6 tid=0x552a0000 nid=0x179c waiting on condition [0x5648f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-5" prio=6 tid=0x5529fc00 nid=0x3e4 waiting on condition [0x5643f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-4" prio=6 tid=0x5529f400 nid=0x63c waiting on condition [0x563ef000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-3" prio=6 tid=0x5529f000 nid=0x17e8 waiting on condition [0x5639f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-2" prio=6 tid=0x5529e800 nid=0x574 waiting on condition [0x5634f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-4-thread-1" prio=6 tid=0x5529e400 nid=0x10cc waiting on condition [0x562ff000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x09970378> (a java.util.concurrent.SynchronousQueue$TransferStack) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) at java.util.concurrent.SynchronousQueue.poll(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-3-thread-1" prio=6 tid=0x546f8800 nid=0x654 waiting on condition [0x562af000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x09970580> (a java.util.concurrent.SynchronousQueue$TransferStack) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) at java.util.concurrent.SynchronousQueue.poll(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-1" prio=6 tid=0x546f8000 nid=0x594 waiting on condition [0x5625f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI TCP Connection(2)-10.0.0.22" daemon prio=6 tid=0x551db800 nid=0xf6c runnable [0x5593f000] java.lang.Thread.State: RUNNABLE at java.net.SocketInputStream.socketRead0(Native Method) at java.net.SocketInputStream.read(Unknown Source) at java.io.BufferedInputStream.fill(Unknown Source) at java.io.BufferedInputStream.read(Unknown Source) - locked <0x08f5c2d0> (a java.io.BufferedInputStream) at java.io.FilterInputStream.read(Unknown Source) at sun.rmi.transport.tcp.TCPTransport.handleMessages(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$ConnectionHandler.run0(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$ConnectionHandler.run(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.runTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - <0x08f62b10> (a java.util.concurrent.locks.ReentrantLock$NonfairSync) "JMX server connection timeout 22" daemon prio=6 tid=0x5474fc00 nid=0x424 in Object.wait() [0x558ef000] java.lang.Thread.State: TIMED_WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x08e9bdb0> (a [I) at com.sun.jmx.remote.internal.ServerCommunicatorAdmin$Timeout.run(Unknown Source) - locked <0x08e9bdb0> (a [I) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI Scheduler(0)" daemon prio=6 tid=0x5511d400 nid=0x1420 waiting on condition [0x5589f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x08e9bdd0> (a java.util.concurrent.locks.AbstractQueuedSynchronizer$ConditionObject) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.locks.AbstractQueuedSynchronizer$ConditionObject.awaitNanos(Unknown Source) at java.util.concurrent.DelayQueue.take(Unknown Source) at java.util.concurrent.ScheduledThreadPoolExecutor$DelayedWorkQueue.take(Unknown Source) at java.util.concurrent.ScheduledThreadPoolExecutor$DelayedWorkQueue.take(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI TCP Connection(idle)" daemon prio=6 tid=0x55176800 nid=0x884 waiting on condition [0x557ff000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x08ebd088> (a java.util.concurrent.SynchronousQueue$TransferStack) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) at java.util.concurrent.SynchronousQueue.poll(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI TCP Accept-0" daemon prio=6 tid=0x5518bc00 nid=0x163c runnable [0x557af000] java.lang.Thread.State: RUNNABLE at java.net.PlainSocketImpl.socketAccept(Native Method) at java.net.PlainSocketImpl.accept(Unknown Source) - locked <0x08e9e158> (a java.net.SocksSocketImpl) at java.net.ServerSocket.implAccept(Unknown Source) at java.net.ServerSocket.accept(Unknown Source) at sun.management.jmxremote.LocalRMIServerSocketFactory$1.accept(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$AcceptLoop.executeAcceptLoop(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$AcceptLoop.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "GC Daemon" daemon prio=2 tid=0x5464b000 nid=0x1718 in Object.wait() [0x5497f000] java.lang.Thread.State: TIMED_WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x089b1270> (a sun.misc.GC$LatencyLock) at sun.misc.GC$Daemon.run(Unknown Source) - locked <0x089b1270> (a sun.misc.GC$LatencyLock) Locked ownable synchronizers: - None "Low Memory Detector" daemon prio=6 tid=0x01a12c00 nid=0x14fc runnable [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "CompilerThread0" daemon prio=10 tid=0x01a0f800 nid=0x1ec waiting on condition [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "JDWP Event Helper Thread" daemon prio=6 tid=0x01a01400 nid=0x173c runnable [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "Attach Listener" daemon prio=10 tid=0x019f5000 nid=0x13a4 waiting on condition [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "Signal Dispatcher" daemon prio=10 tid=0x019ea000 nid=0x17b8 runnable [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "Finalizer" daemon prio=8 tid=0x019bb400 nid=0x1720 in Object.wait() [0x53fcf000] java.lang.Thread.State: WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x089b2858> (a java.lang.ref.ReferenceQueue$Lock) at java.lang.ref.ReferenceQueue.remove(Unknown Source) - locked <0x089b2858> (a java.lang.ref.ReferenceQueue$Lock) at java.lang.ref.ReferenceQueue.remove(Unknown Source) at java.lang.ref.Finalizer$FinalizerThread.run(Unknown Source) Locked ownable synchronizers: - None "Reference Handler" daemon prio=10 tid=0x019ba000 nid=0x53c in Object.wait() [0x53f7f000] java.lang.Thread.State: WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x089b2878> (a java.lang.ref.Reference$Lock) at java.lang.Object.wait(Object.java:485) at java.lang.ref.Reference$ReferenceHandler.run(Unknown Source) - locked <0x089b2878> (a java.lang.ref.Reference$Lock) Locked ownable synchronizers: - None "VM Thread" prio=10 tid=0x019b7400 nid=0xab8 runnable "VM Periodic Task Thread" prio=10 tid=0x01a1bc00 nid=0xc7c waiting on condition JNI global references: 21240 *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Hector Geraldino *Sent:* Monday, December 19, 2011 8:29 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Threads remain after calling close on Java client Hi Neil, Can you get a thread dump of the tomcat process to try to figure out what this problem is about? Or at least, try to connect the jconsole to the tomcat process and get the StackTrace of one of these threads to have a better idea of what is going on. IIRC I?ve fixed a couple of bugs for this library, but the patches haven?t been tested by the main developer (dvarnes) nor integrated on the repository (freeswitch-contrib). If this problem can be fixed with my patched code, I would be happy to share it with you. Good luck! *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Neil Davis *Sent:* Friday, December 16, 2011 7:09 PM *To:* FreeSWITCH-users at lists.freeswitch.org *Subject:* [Freeswitch-users] Threads remain after calling close on Java client Hi, I built a web application that connects to Freeswitch using the org.freeswitch.esl.client.Client. I connect the Client object from a Spring annotated service that I call from a Spring controller. I put the connected client in my ServletContext, so I can access it later to call client.cancelEventSubscriptions() and client.close() from my ServletContextListener contextDestroyed method when Tomcat is shutting down. The problem I'm having is that even after I call close on the client, there are still a bunch of active threads that the client has spawned in the background. These threads are causing Tomcat to hang when I'm shutting down. Can anyone suggest an approach that would enable my application to disconnect the Freeswitch client when Tomcat is shutting down that would allow Tomcat to shutdown gracefully? Below are errors from my Tomcat log for the threads that I have identified as being related to the Freeswitch client. I don't know how I can get to these threads to interrupt them and Client.close() seems to leave them hanging. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-3-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-4-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-2] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-3] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-4] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-5] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-6] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-7] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-8] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-9] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-10] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-11] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-12] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-13] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-14] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-15] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-16] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.internal.ThreadLocalBoolean] (value [org.jboss.netty.util.internal.ThreadLocalBoolean at 186e192]) and a value of type [java.lang.Boolean] (value [false]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.CharsetUtil$1] (value [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type [java.util.IdentityHashMap] (value [{windows-1252=sun.nio.cs.MS1252$Encoder at 373f86}]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.internal.ThreadLocalRandom$1] (value [org.jboss.netty.util.internal.ThreadLocalRandom$1 at 12bb519]) and a value of type [org.jboss.netty.util.internal.ThreadLocalRandom] (value [org.jboss.netty.util.internal.ThreadLocalRandom at 7e9dbc]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.CharsetUtil$1] (value [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type [java.util.IdentityHashMap] (value [{windows-1252=sun.nio.cs.MS1252$Encoder at a5b041}]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Thanks, Neil Davis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/93481d7b/attachment-0001.html From msc at freeswitch.org Wed Dec 21 19:16:17 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Dec 2011 08:16:17 -0800 Subject: [Freeswitch-users] Error "Channels not ready" when bridge call in Lua In-Reply-To: <1324431091351-7113903.post@n2.nabble.com> References: <1324393096164-7112050.post@n2.nabble.com> <4EF0A757.8070404@tagnet.ru> <1324431091351-7113903.post@n2.nabble.com> Message-ID: Also, this isn't "support" - this is the freeswitch-users list. Support can be reached at: consulting at freeswitch.org. -MC On Tue, Dec 20, 2011 at 5:31 PM, dma wrote: > Hi Boris, > > Thanks a lot for the information! I shall try to upgrade! > > Best regards, > D.Ma > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Error-Channels-not-ready-when-bridge-call-in-Lua-tp7112050p7113903.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/20111221/24053c96/attachment.html From alessandro.illiano at neexa.it Wed Dec 21 15:15:09 2011 From: alessandro.illiano at neexa.it (Alessandro Illiano) Date: Wed, 21 Dec 2011 13:15:09 +0100 Subject: [Freeswitch-users] R: Sofia late-negotiation on re-INVITE(codec-modification) In-Reply-To: Message-ID: Hi All, I've some issue, B send a re-invite changing the codec and fs hangup the call with 488 , Without reinvite a-leg?. Regards, Alessandro Da: Visciano Giovanni Risposta: FreeSWITCH Users Help Data: Thu, 1 Dec 2011 16:35:54 +0100 A: FreeSWITCH Users Help Oggetto: [Freeswitch-users] R: Sofia late-negotiation on re-INVITE(codec-modification) R: [Freeswitch-users] Sofia late-negotiation on re-INVITE(codec-modification) Finally we are back to our test. I updated my FS installation to last GIT (FreeSWITCH Version 1.0.head (git-eae86e0 2011-11-30 18-14-24 -0600)) TEST ---- SIP vs SIP basic audio call, then re-INVITE for codec modification. FS configuration B2B, avoid transcoding. CONF ---- In sofia SIP profile I have: and in my dialplan XML I hit: Loaded codec modules: freeswitch at internal> show codec type,name,ikey codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.729,mod_g729 codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE TEST 1) ------- 1001 ----invite(pcma)----> FS --invite(pcma)--> 1000 1001 <----200OK(pcma)---- FS <--200OK(pcma)-- 1000 1001 --re/invite(g729)---> FS 1001 <----488------------ FS Full log: http://pastebin.freeswitch.org/17906 Note: FS does not negotiate end to end the reINVITE O/A codec modification. It is closed locally on the 1001->FS leg. From the log I see 2011-12-01 15:54:13.871332 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2011-12-01 15:54:13.871332 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] 2011-12-01 15:54:13.871332 [ERR] sofia.c:5876 Reinvite Codec Error! TEST 2) ------- 1000 ----invite(pcma)----> FS --invite(pcma)--> 1001 1000 <----200OK(pcma)---- FS <--200OK(pcma)-- 1001 1000 FS <--re/invite(g729)- 1001 1000 FS ------------------> 1001 Full log: http://pastebin.freeswitch.org/17907 Note: FS does not negotiate end to end the reINVITE O/A codec modification. It is closed locally on the 1001->FS leg. FS select locally G729 but this result in transcoding! I hate transcoding! >From the log I see 2011-12-01 16:00:12.971244 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2011-12-01 16:00:12.971244 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [G729:18:8000:20:8000]/[G729:18:8000:20:8000] 2011-12-01 16:00:12.971244 [DEBUG] sofia_glue.c:2806 Changing Codec from PCMA at 20ms@8000hz to G729 at 20ms@8000hz 2011-12-01 16:00:13.011337 [DEBUG] sofia_glue.c:2888 Set Codec sofia/internal/1001 at 138.132.110.64:5070 G729/8000 20 ms 160 samples 8000 bits 2011-12-01 16:00:13.011337 [DEBUG] switch_core_codec.c:116 sofia/internal/1001 at 138.132.110.64:5070 Push codec G729:18 2011-12-01 16:00:13.031252 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode! 2011-12-01 16:00:13.031252 [ERR] switch_core_io.c:1077 Codec G.729 encoder error! Regard, Giovanni PS: next week I won't be at work. My collegue Nevio will follow the problem, so if you need more data/details just tell us. Internet Email Confidentiality Footer **************************************************************************** **************************************************************** La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. **************************************************************************** **************************************************************** _________________________________________________________________________ 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/2f24c7ee/attachment.html From msc at freeswitch.org Wed Dec 21 19:22:00 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Dec 2011 08:22:00 -0800 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today Message-ID: Hello all! Today's conference call agenda is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_12_21 We will be talking about CDRs today. DRK will be sharing some of his experience with parsing and rating. If anyone else has experience with CDR parsing and rating please give us your input. We'll also be discussing some CDR collection techniques. Lastly, we have a few updates for the community, including Mitch Capper discussing some TLS changes in FreeSWITCH. Talk to you soon! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/eaca8db3/attachment.html From msc at freeswitch.org Wed Dec 21 19:23:28 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 21 Dec 2011 08:23:28 -0800 Subject: [Freeswitch-users] Need help with PortAudio In-Reply-To: <201112202245.57010.justlikeef@gmail.com> References: <201112202245.57010.justlikeef@gmail.com> Message-ID: Also, if you can reproduce the segfault on latest git then please open a Jira ticket so that the devs can address it. -MC On Tue, Dec 20, 2011 at 7:45 PM, Rob Hutton wrote: > ** > > I am trying to get portaudio working, and have run into a couple of > problems > > > 1) If portaudio cannot access the device, it causes freeswitch to segfault > > > 2) I have set up a dialplan following the intercom example which seems to > be working, but I get no audio. I have tried setting every device shown in > devlist as the output device. > > > 3) When I issue a "pa play". the cli hangs, even if I give it a timeout. > > > Any thoughts? > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/354ed9a7/attachment-0001.html From Giovanni.Visciano at italtel.it Wed Dec 21 20:26:16 2011 From: Giovanni.Visciano at italtel.it (Visciano Giovanni) Date: Wed, 21 Dec 2011 18:26:16 +0100 Subject: [Freeswitch-users] R: Sofia late-negotiation on re-INVITE(codec-modification) References: Message-ID: I've opened a BUG. http://jira.freeswitch.org/browse/FS-3739 I don't know what's you FS configuration. What I know is: - SIP "media proxy mode" is OK. - re-INVITE for codec modification to image T.38 (t38-passthrough) is OK in all SIP "media mode". The "late_negotiation" + "inherit_codec" trick to avoid transcoding in a Back2Back (that is no media proxy) configuration works only for the very first call setup SDP offer/answer. http://wiki.freeswitch.org/wiki/Codec_negotiation Once the call is established, further codec modification via re-INVITE are usually closed on each leg (except for t38) and this can led to transcoding (and obvious errors if you can't or don't want to transcode). Hope this helps you find out your problem. Giovanni -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Alessandro Illiano Sent: Wed 12/21/2011 1:15 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] R: Sofia late-negotiation on re-INVITE(codec-modification) Hi All, I've some issue, B send a re-invite changing the codec and fs hangup the call with 488 , Without reinvite a-legS. Regards, Alessandro Da: Visciano Giovanni Risposta: FreeSWITCH Users Help Data: Thu, 1 Dec 2011 16:35:54 +0100 A: FreeSWITCH Users Help Oggetto: [Freeswitch-users] R: Sofia late-negotiation on re-INVITE(codec-modification) R: [Freeswitch-users] Sofia late-negotiation on re-INVITE(codec-modification) Finally we are back to our test. I updated my FS installation to last GIT (FreeSWITCH Version 1.0.head (git-eae86e0 2011-11-30 18-14-24 -0600)) TEST ---- SIP vs SIP basic audio call, then re-INVITE for codec modification. FS configuration B2B, avoid transcoding. CONF ---- In sofia SIP profile I have: and in my dialplan XML I hit: Loaded codec modules: freeswitch at internal> show codec type,name,ikey codec,G.711 alaw,CORE_PCM_MODULE codec,G.711 ulaw,CORE_PCM_MODULE codec,G.729,mod_g729 codec,PROXY PASS-THROUGH,CORE_PCM_MODULE codec,PROXY VIDEO PASS-THROUGH,CORE_PCM_MODULE codec,RAW Signed Linear (16 bit),CORE_PCM_MODULE TEST 1) ------- 1001 ----invite(pcma)----> FS --invite(pcma)--> 1000 1001 <----200OK(pcma)---- FS <--200OK(pcma)-- 1000 1001 --re/invite(g729)---> FS 1001 <----488------------ FS Full log: http://pastebin.freeswitch.org/17906 Note: FS does not negotiate end to end the reINVITE O/A codec modification. It is closed locally on the 1001->FS leg. From the log I see 2011-12-01 15:54:13.871332 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2011-12-01 15:54:13.871332 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [telephone-event:101:8000:20:0]/[PCMA:8:8000:20:64000] 2011-12-01 15:54:13.871332 [ERR] sofia.c:5876 Reinvite Codec Error! TEST 2) ------- 1000 ----invite(pcma)----> FS --invite(pcma)--> 1001 1000 <----200OK(pcma)---- FS <--200OK(pcma)-- 1001 1000 FS <--re/invite(g729)- 1001 1000 FS ------------------> 1001 Full log: http://pastebin.freeswitch.org/17907 Note: FS does not negotiate end to end the reINVITE O/A codec modification. It is closed locally on the 1001->FS leg. FS select locally G729 but this result in transcoding! I hate transcoding! >From the log I see 2011-12-01 16:00:12.971244 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [G729:18:8000:20:8000]/[PCMA:8:8000:20:64000] 2011-12-01 16:00:12.971244 [DEBUG] sofia_glue.c:4767 Audio Codec Compare [G729:18:8000:20:8000]/[G729:18:8000:20:8000] 2011-12-01 16:00:12.971244 [DEBUG] sofia_glue.c:2806 Changing Codec from PCMA at 20ms@8000hz to G729 at 20ms@8000hz 2011-12-01 16:00:13.011337 [DEBUG] sofia_glue.c:2888 Set Codec sofia/internal/1001 at 138.132.110.64:5070 G729/8000 20 ms 160 samples 8000 bits 2011-12-01 16:00:13.011337 [DEBUG] switch_core_codec.c:116 sofia/internal/1001 at 138.132.110.64:5070 Push codec G729:18 2011-12-01 16:00:13.031252 [ERR] mod_g729.c:102 This codec is only usable in passthrough mode! 2011-12-01 16:00:13.031252 [ERR] switch_core_io.c:1077 Codec G.729 encoder error! Regard, Giovanni PS: next week I won't be at work. My collegue Nevio will follow the problem, so if you need more data/details just tell us. Internet Email Confidentiality Footer **************************************************************************** **************************************************************** La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. **************************************************************************** **************************************************************** _________________________________________________________________________ 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 -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 6150 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/7b87cc90/attachment.bin From brian at freeswitch.org Wed Dec 21 20:30:59 2011 From: brian at freeswitch.org (Brian West) Date: Wed, 21 Dec 2011 11:30:59 -0600 Subject: [Freeswitch-users] Sofia late-negotiation on re-INVITE(codec-modification) In-Reply-To: References: Message-ID: <98718589-93E6-4378-B849-BF4E453C2F8B@freeswitch.org> I don't think this is a bug... I think its by design. /b On Dec 21, 2011, at 11:26 AM, Visciano Giovanni wrote: > I've opened a BUG. > http://jira.freeswitch.org/browse/FS-3739 > > I don't know what's you FS configuration. > What I know is: > - SIP "media proxy mode" is OK. > - re-INVITE for codec modification to image T.38 (t38-passthrough) is OK in all SIP "media mode". > > The "late_negotiation" + "inherit_codec" trick to avoid transcoding in a Back2Back > (that is no media proxy) configuration works only for the very first call setup SDP offer/answer. > http://wiki.freeswitch.org/wiki/Codec_negotiation > > Once the call is established, further codec modification via re-INVITE are usually closed on > each leg (except for t38) and this can led to transcoding (and obvious errors if you can't or don't want > to transcode). > > Hope this helps you find out your problem. > > Giovanni -- 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/20111221/352f15af/attachment.html From Hector.Geraldino at ip-soft.net Wed Dec 21 22:26:07 2011 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Wed, 21 Dec 2011 14:26:07 -0500 Subject: [Freeswitch-users] Threads remain after calling close on Java client In-Reply-To: <895731f6d2c9f7220f752e977f485386@mail.gmail.com> References: <8b41de351c0d1365e3786e7a60645275@mail.gmail.com> <6A6B4C284AD15042B429EB9D904544AD0225507041@NY1-EXMB-01.ip-soft.net> <895731f6d2c9f7220f752e977f485386@mail.gmail.com> Message-ID: <6A6B4C284AD15042B429EB9D904544AD02255071D4@NY1-EXMB-01.ip-soft.net> Hi Neil, This doesn't seem to be the same concurrency issue I had, but I'm attaching the patch that fixes my issue anyway. Feel free to test it and send me back the restuls. In case it doesn't work you might try to "manually" close the channel by modifying the close() method on the org.freeswitch.esl.client.inbound.Client. Try to do a channel.disconnect(); and channel=null; and see what happens. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Neil Davis Sent: Wednesday, December 21, 2011 11:17 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Threads remain after calling close on Java client Here is a thread dump of my Tomcat process at the point when it is hanging on shutdown. There are a number of threads in a "waiting on condition" state that appear to have to do with the netty package on which the Freeswitch client is dependent. 2011-12-21 09:05:17 Full thread dump Java HotSpot(TM) Client VM (14.3-b01 mixed mode): "DestroyJavaVM" prio=6 tid=0x546f7400 nid=0x640 waiting on condition [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "pool-5-thread-16" prio=6 tid=0x544b5800 nid=0x1098 waiting on condition [0x55e4f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-15" prio=6 tid=0x54f7f400 nid=0x10f0 waiting on condition [0x54e5f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-14" prio=6 tid=0x544b3c00 nid=0xdb4 waiting on condition [0x5670f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-13" prio=6 tid=0x544b4000 nid=0x12c8 waiting on condition [0x566bf000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-12" prio=6 tid=0x544b4800 nid=0xe18 waiting on condition [0x5666f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-11" prio=6 tid=0x544b3000 nid=0x620 waiting on condition [0x5661f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-10" prio=6 tid=0x552a1800 nid=0x131c waiting on condition [0x565cf000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-9" prio=6 tid=0x552a1400 nid=0x1710 waiting on condition [0x5657f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-8" prio=6 tid=0x552a0c00 nid=0x1094 waiting on condition [0x5652f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-7" prio=6 tid=0x552a0800 nid=0x1040 waiting on condition [0x564df000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-6" prio=6 tid=0x552a0000 nid=0x179c waiting on condition [0x5648f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-5" prio=6 tid=0x5529fc00 nid=0x3e4 waiting on condition [0x5643f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-4" prio=6 tid=0x5529f400 nid=0x63c waiting on condition [0x563ef000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-3" prio=6 tid=0x5529f000 nid=0x17e8 waiting on condition [0x5639f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-2" prio=6 tid=0x5529e800 nid=0x574 waiting on condition [0x5634f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-4-thread-1" prio=6 tid=0x5529e400 nid=0x10cc waiting on condition [0x562ff000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x09970378> (a java.util.concurrent.SynchronousQueue$TransferStack) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) at java.util.concurrent.SynchronousQueue.poll(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-3-thread-1" prio=6 tid=0x546f8800 nid=0x654 waiting on condition [0x562af000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x09970580> (a java.util.concurrent.SynchronousQueue$TransferStack) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) at java.util.concurrent.SynchronousQueue.poll(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-1" prio=6 tid=0x546f8000 nid=0x594 waiting on condition [0x5625f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI TCP Connection(2)-10.0.0.22" daemon prio=6 tid=0x551db800 nid=0xf6c runnable [0x5593f000] java.lang.Thread.State: RUNNABLE at java.net.SocketInputStream.socketRead0(Native Method) at java.net.SocketInputStream.read(Unknown Source) at java.io.BufferedInputStream.fill(Unknown Source) at java.io.BufferedInputStream.read(Unknown Source) - locked <0x08f5c2d0> (a java.io.BufferedInputStream) at java.io.FilterInputStream.read(Unknown Source) at sun.rmi.transport.tcp.TCPTransport.handleMessages(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$ConnectionHandler.run0(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$ConnectionHandler.run(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.runTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - <0x08f62b10> (a java.util.concurrent.locks.ReentrantLock$NonfairSync) "JMX server connection timeout 22" daemon prio=6 tid=0x5474fc00 nid=0x424 in Object.wait() [0x558ef000] java.lang.Thread.State: TIMED_WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x08e9bdb0> (a [I) at com.sun.jmx.remote.internal.ServerCommunicatorAdmin$Timeout.run(Unknown Source) - locked <0x08e9bdb0> (a [I) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI Scheduler(0)" daemon prio=6 tid=0x5511d400 nid=0x1420 waiting on condition [0x5589f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x08e9bdd0> (a java.util.concurrent.locks.AbstractQueuedSynchronizer$ConditionObject) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.locks.AbstractQueuedSynchronizer$ConditionObject.awaitNanos(Unknown Source) at java.util.concurrent.DelayQueue.take(Unknown Source) at java.util.concurrent.ScheduledThreadPoolExecutor$DelayedWorkQueue.take(Unknown Source) at java.util.concurrent.ScheduledThreadPoolExecutor$DelayedWorkQueue.take(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI TCP Connection(idle)" daemon prio=6 tid=0x55176800 nid=0x884 waiting on condition [0x557ff000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x08ebd088> (a java.util.concurrent.SynchronousQueue$TransferStack) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) at java.util.concurrent.SynchronousQueue.poll(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI TCP Accept-0" daemon prio=6 tid=0x5518bc00 nid=0x163c runnable [0x557af000] java.lang.Thread.State: RUNNABLE at java.net.PlainSocketImpl.socketAccept(Native Method) at java.net.PlainSocketImpl.accept(Unknown Source) - locked <0x08e9e158> (a java.net.SocksSocketImpl) at java.net.ServerSocket.implAccept(Unknown Source) at java.net.ServerSocket.accept(Unknown Source) at sun.management.jmxremote.LocalRMIServerSocketFactory$1.accept(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$AcceptLoop.executeAcceptLoop(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$AcceptLoop.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "GC Daemon" daemon prio=2 tid=0x5464b000 nid=0x1718 in Object.wait() [0x5497f000] java.lang.Thread.State: TIMED_WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x089b1270> (a sun.misc.GC$LatencyLock) at sun.misc.GC$Daemon.run(Unknown Source) - locked <0x089b1270> (a sun.misc.GC$LatencyLock) Locked ownable synchronizers: - None "Low Memory Detector" daemon prio=6 tid=0x01a12c00 nid=0x14fc runnable [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "CompilerThread0" daemon prio=10 tid=0x01a0f800 nid=0x1ec waiting on condition [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "JDWP Event Helper Thread" daemon prio=6 tid=0x01a01400 nid=0x173c runnable [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "Attach Listener" daemon prio=10 tid=0x019f5000 nid=0x13a4 waiting on condition [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "Signal Dispatcher" daemon prio=10 tid=0x019ea000 nid=0x17b8 runnable [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "Finalizer" daemon prio=8 tid=0x019bb400 nid=0x1720 in Object.wait() [0x53fcf000] java.lang.Thread.State: WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x089b2858> (a java.lang.ref.ReferenceQueue$Lock) at java.lang.ref.ReferenceQueue.remove(Unknown Source) - locked <0x089b2858> (a java.lang.ref.ReferenceQueue$Lock) at java.lang.ref.ReferenceQueue.remove(Unknown Source) at java.lang.ref.Finalizer$FinalizerThread.run(Unknown Source) Locked ownable synchronizers: - None "Reference Handler" daemon prio=10 tid=0x019ba000 nid=0x53c in Object.wait() [0x53f7f000] java.lang.Thread.State: WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x089b2878> (a java.lang.ref.Reference$Lock) at java.lang.Object.wait(Object.java:485) at java.lang.ref.Reference$ReferenceHandler.run(Unknown Source) - locked <0x089b2878> (a java.lang.ref.Reference$Lock) Locked ownable synchronizers: - None "VM Thread" prio=10 tid=0x019b7400 nid=0xab8 runnable "VM Periodic Task Thread" prio=10 tid=0x01a1bc00 nid=0xc7c waiting on condition JNI global references: 21240 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hector Geraldino Sent: Monday, December 19, 2011 8:29 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Threads remain after calling close on Java client Hi Neil, Can you get a thread dump of the tomcat process to try to figure out what this problem is about? Or at least, try to connect the jconsole to the tomcat process and get the StackTrace of one of these threads to have a better idea of what is going on. IIRC I've fixed a couple of bugs for this library, but the patches haven't been tested by the main developer (dvarnes) nor integrated on the repository (freeswitch-contrib). If this problem can be fixed with my patched code, I would be happy to share it with you. Good luck! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Neil Davis Sent: Friday, December 16, 2011 7:09 PM To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] Threads remain after calling close on Java client Hi, I built a web application that connects to Freeswitch using the org.freeswitch.esl.client.Client. I connect the Client object from a Spring annotated service that I call from a Spring controller. I put the connected client in my ServletContext, so I can access it later to call client.cancelEventSubscriptions() and client.close() from my ServletContextListener contextDestroyed method when Tomcat is shutting down. The problem I'm having is that even after I call close on the client, there are still a bunch of active threads that the client has spawned in the background. These threads are causing Tomcat to hang when I'm shutting down. Can anyone suggest an approach that would enable my application to disconnect the Freeswitch client when Tomcat is shutting down that would allow Tomcat to shutdown gracefully? Below are errors from my Tomcat log for the threads that I have identified as being related to the Freeswitch client. I don't know how I can get to these threads to interrupt them and Client.close() seems to leave them hanging. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-3-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-4-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-2] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-3] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-4] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-5] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-6] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-7] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-8] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-9] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-10] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-11] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-12] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-13] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-14] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-15] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-16] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.internal.ThreadLocalBoolean] (value [org.jboss.netty.util.internal.ThreadLocalBoolean at 186e192]) and a value of type [java.lang.Boolean] (value [false]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.CharsetUtil$1] (value [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type [java.util.IdentityHashMap] (value [{windows-1252=sun.nio.cs.MS1252$Encoder at 373f86}]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.internal.ThreadLocalRandom$1] (value [org.jboss.netty.util.internal.ThreadLocalRandom$1 at 12bb519]) and a value of type [org.jboss.netty.util.internal.ThreadLocalRandom] (value [org.jboss.netty.util.internal.ThreadLocalRandom at 7e9dbc]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.CharsetUtil$1] (value [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type [java.util.IdentityHashMap] (value [{windows-1252=sun.nio.cs.MS1252$Encoder at a5b041}]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Thanks, Neil Davis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/99041e99/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: Client.java.diff Type: application/octet-stream Size: 1196 bytes Desc: Client.java.diff Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/99041e99/attachment-0001.obj From Giovanni.Visciano at italtel.it Wed Dec 21 22:34:57 2011 From: Giovanni.Visciano at italtel.it (Visciano Giovanni) Date: Wed, 21 Dec 2011 20:34:57 +0100 Subject: [Freeswitch-users] Sofia late-negotiation onre-INVITE(codec-modification) References: <98718589-93E6-4378-B849-BF4E453C2F8B@freeswitch.org> Message-ID: I used to say to my boss: ... it's not a bug it's a feature! I'm kidding :) By the way, even in this "design choice", al least the first test scenario is a bug (described in the first half on the Jira BUG): http://pastebin.freeswitch.org/17906 Moreover this "by design" behavior create some problem not only for transcoding issue but also for fax upspeed codec modification (no T38). Years ago, when hardware resources where poor, I had a lot of discussions with my colleagues to avoid the allocation of transcoding resources on our softswitch and we always ended up to use a transcoding resource if and only if it's really needed, if the endpoints can speak the same language let them do all the stuff. I know easy to say but ... Back to FS, what I think is we miss an endpoint independent codec negotiation signaling. I mean something like we have in the call setup phase via "absolute_codec_string" / "ep_codec_string" / SWITCH_ORIGINATOR_CODEC_VARIABLE. As I'm playing/prototyping a new endpoint module I used these channel variable to successfully interworking with SIP endpoint gateway in codec passthrough... but now that the test-suite is going over the SIP basic call codec modification I found that "by design problem". In the (working) spare time I'm trying to write a patch to extend the core interworking opening the way I suggested few lines over. As I'm very new to FS, for now it's a very embryonal and poor implementation in my opinion, but enough for the endpoint module prototype evaluation. I hope to give that back to you if I can find more time to move the patch to a presentable level. Giovanni -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org on behalf of Brian West Sent: Wed 12/21/2011 6:30 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Sofia late-negotiation onre-INVITE(codec-modification) I don't think this is a bug... I think its by design. /b On Dec 21, 2011, at 11:26 AM, Visciano Giovanni wrote: > I've opened a BUG. > http://jira.freeswitch.org/browse/FS-3739 > > I don't know what's you FS configuration. > What I know is: > - SIP "media proxy mode" is OK. > - re-INVITE for codec modification to image T.38 (t38-passthrough) is OK in all SIP "media mode". > > The "late_negotiation" + "inherit_codec" trick to avoid transcoding in a Back2Back > (that is no media proxy) configuration works only for the very first call setup SDP offer/answer. > http://wiki.freeswitch.org/wiki/Codec_negotiation > > Once the call is established, further codec modification via re-INVITE are usually closed on > each leg (except for t38) and this can led to transcoding (and obvious errors if you can't or don't want > to transcode). > > Hope this helps you find out your problem. > > Giovanni -- Brian West FreeSWITCH Solutions, LLC Phone: +1 (918) 420-9266 Fax: +1 (918) 420-9267 brian at freeswitch.org http://www.freeswitch.org Internet Email Confidentiality Footer ----------------------------------------------------------------------------------------------------- La presente comunicazione, con le informazioni in essa contenute e ogni documento o file allegato, e' rivolta unicamente alla/e persona/e cui e' indirizzata ed alle altre da questa autorizzata/e a riceverla. Se non siete i destinatari/autorizzati siete avvisati che qualsiasi azione, copia, comunicazione, divulgazione o simili basate sul contenuto di tali informazioni e' vietata e potrebbe essere contro la legge (art. 616 C.P., D.Lgs n. 196/2003 Codice in materia di protezione dei dati personali). Se avete ricevuto questa comunicazione per errore, vi preghiamo di darne immediata notizia al mittente e di distruggere il messaggio originale e ogni file allegato senza farne copia alcuna o riprodurne in alcun modo il contenuto. This e-mail and its attachments are intended for the addressee(s) only and are confidential and/or may contain legally privileged information. If you have received this message by mistake or are not one of the addressees above, you may take no action based on it, and you may not copy or show it to anyone; please reply to this e-mail and point out the error which has occurred. ----------------------------------------------------------------------------------------------------- -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: application/ms-tnef Size: 4634 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/0e18e85b/attachment.bin From Ryan at ocens.com Wed Dec 21 22:45:36 2011 From: Ryan at ocens.com (Ryan Watkins) Date: Wed, 21 Dec 2011 19:45:36 +0000 Subject: [Freeswitch-users] Issue adding a SIP Gateway In-Reply-To: <4EF0581B.30701@integrafin.co.uk> References: <44E5C0A9D48A3246966A4AE04692014D102A6FB5@CH1PRD0604MB109.namprd06.prod.outlook.com> <201112151954.01880.justlikeef@gmail.com> <44E5C0A9D48A3246966A4AE04692014D102A803C@CH1PRD0604MB109.namprd06.prod.outlook.com> <4EEDBBB2.50302@integrafin.co.uk> <44E5C0A9D48A3246966A4AE04692014D102AA3C6@CH1PRD0604MB109.namprd06.prod.outlook.com> <4EEF8EF0.6010207@integrafin.co.uk> <44E5C0A9D48A3246966A4AE04692014D102AA5E4@CH1PRD0604MB109.namprd06.prod.outlook.com> <4EEFA3BD.5010201@integrafin.co.uk> <44E5C0A9D48A3246966A4AE04692014D102AA623@CH1PRD0604MB109.namprd06.prod.outlook.com> <4EF0581B.30701@integrafin.co.uk> Message-ID: <44E5C0A9D48A3246966A4AE04692014D102ABCAE@CH1PRD0604MB109.namprd06.prod.outlook.com> Alex, I rebuild my virtual server... love virtual servers for dev work :p It appears that, after the rebuild and reinstall of FreeSWITCH that I'm now able to register an external gateway.... So it must have been something that didn't install correctly on the last go. Thank you for your help... able to at least make outgoing calls, incoming are still an issue :P but I'm learning and working through it. Ryan From: Alex Crow [mailto:acrow at integrafin.co.uk] Sent: Tuesday, December 20, 2011 1:41 AM To: Ryan Watkins Cc: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Issue adding a SIP Gateway On 19/12/11 21:05, Ryan Watkins wrote: Alex, I've seen that as well... but where I figured it was part of the example.xml gateway I didn't pay much mind. I have, however, renamed the example.xml to example.xml.noload, stopped and started the daemon, and attempted to reload the external profile within freeswitch... and it's STILL loading the example gateway... and will not load iptel, I haven't ever touched the external.xml, but I'm wondering if maybe it needed to be edited? Or perhaps the freeswitch install put yet another location in that this setup that it might be looking to?? Really lost at this point... don't know how it's loading the example gateway when it's been renamed (even renamed the example.xml in /usr/src/freeswitch* for S&Gs) Ryan, Do you have exactly this at the top of /opt/freeswitch/conf/sip_profiles/external.xml? If you do, and it's still not loading, I am stumped too. Can you post your entire external.xml? Cheers Alex From: Alex Crow [mailto:acrow at integrafin.co.uk] Sent: Monday, December 19, 2011 12:51 PM To: Ryan Watkins Cc: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Issue adding a SIP Gateway Ryan, I see this: 2011-12-19 12:28:30.388149 [DEBUG] sofia.c:1736 Launching worker thread for external 2011-12-19 12:28:30.388202 [ERR] sofia.c:2490 ERROR: password param is REQUIRED! This could actually be from your "example.com" gateway, it is indeed an error relating to gateway defs. I suggest you delete that gateway or move it to example.com.noload (or the like). It could be stopping your proper gateway from loading. I'm sure you don't need it anyway. Also check, if you've edited your external.xml file that you haven't got messed up tags before the include of external/*.xml. Cheers Alex On 19/12/11 20:38, Ryan Watkins wrote: Here's the log file... thanks again Alex and all From: Alex Crow [mailto:acrow at integrafin.co.uk] Sent: Monday, December 19, 2011 11:22 AM To: FreeSWITCH Users Help Cc: Ryan Watkins Subject: Re: [Freeswitch-users] Issue adding a SIP Gateway OK, As long as those "funny quotes" in your last post aren't present in your XML file I can't see what is the problem. I had an issue copying dialplans from the web with these, had to go through and change them all by hand. If you shut down freeswitch, clear your logfile, and start it again, can you post the contents of /opt/freeswitch/log/freeswitch.log to the list (or to pastebin etc and link here). There must be something in the logs. BTW you should not need to change things in /usr/src/*, once you've installed that doesn't make any difference. Cheers Alex On 19/12/11 18:36, Ryan Watkins wrote: Thanks for the reply Alex, I followed the example in the FreeSWITCH 1.0.6 book for iptel, which is as follows: (yes, I supplied my iptel username in this line) (again, I supplied my iptel password on this line) I've checked the file permissions and changed them so that all users have rwx for the iptel.org.xml file; as well as every folder up to /external for both /usr/src/freeswitch and the /opt/freeswitch paths I've also changed the iptel.org.xml files in those paths to the example that you linked from the wiki However, I'm still getting the same result.... Any other suggestions? Thanks again! From: Alex Crow [mailto:acrow at integrafin.co.uk] Sent: Sunday, December 18, 2011 2:09 AM To: FreeSWITCH Users Help Cc: Ryan Watkins Subject: Re: [Freeswitch-users] Issue adding a SIP Gateway On 16/12/11 01:05, Ryan Watkins wrote: I did run the" sofia profile external restart reloadxml" command.... It didn't load the new gateway, so that's why I tried registering the gateway specifically. Ryan, Did you follow this example: http://wiki.freeswitch.org/wiki/Provider_Configuration:_iptel and replace the usename and password with your own? Check that the permissions on the new XML file allow it to be read by the user freeswitch is running as. Also double-check your closing tags on the file. This can cause your gateway to be skipped, hence the "invalid gateway" when you try to use it. Cheers Alex -- This message is intended only for the addressee and may contain confidential information. Unless you are that person, you may not disclose its contents or use it in any way and are requested to delete the message along with any attachments and notify us immediately. "Transact" is operated by Integrated Financial Arrangements plc Domain House, 5-7 Singer Street, London EC2A 4BQ Tel: (020) 7608 4900 Fax: (020) 7608 5300 (Registered office: as above; Registered in England and Wales under number: 3727592) Authorised and regulated by the Financial Services Authority (entered on the FSA Register; number: 190856) -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/a39cd1df/attachment-0001.html From justlikeef at gmail.com Wed Dec 21 22:58:36 2011 From: justlikeef at gmail.com (Rob Hutton) Date: Wed, 21 Dec 2011 14:58:36 -0500 Subject: [Freeswitch-users] Need help with PortAudio In-Reply-To: References: <201112202245.57010.justlikeef@gmail.com> Message-ID: <201112211458.37234.justlikeef@gmail.com> I will do so. I am seeing the segfault in several freeswitch modules when they can't access resources that they need access to. I have been building an AppArmor profile. If there is a way to protect from this in the core, it might be a good addition... On Wednesday 21 December 2011 11:23:28 Michael Collins wrote: > Also, if you can reproduce the segfault on latest git then please open a > Jira ticket so that the devs can address it. > > -MC > > On Tue, Dec 20, 2011 at 7:45 PM, Rob Hutton wrote: > > > ** > > > > I am trying to get portaudio working, and have run into a couple of > > problems > > > > > > 1) If portaudio cannot access the device, it causes freeswitch to segfault > > > > > > 2) I have set up a dialplan following the intercom example which seems to > > be working, but I get no audio. I have tried setting every device shown in > > devlist as the output device. > > > > > > 3) When I issue a "pa play". the cli hangs, even if I give it a timeout. > > > > > > Any thoughts? > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/dd80b128/attachment.html From anthony.minessale at gmail.com Thu Dec 22 00:13:53 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 21 Dec 2011 15:13:53 -0600 Subject: [Freeswitch-users] Threads remain after calling close on Java client In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD02255071D4@NY1-EXMB-01.ip-soft.net> References: <8b41de351c0d1365e3786e7a60645275@mail.gmail.com> <6A6B4C284AD15042B429EB9D904544AD0225507041@NY1-EXMB-01.ip-soft.net> <895731f6d2c9f7220f752e977f485386@mail.gmail.com> <6A6B4C284AD15042B429EB9D904544AD02255071D4@NY1-EXMB-01.ip-soft.net> Message-ID: if this patch is necessary can you post it to a jira? On Wed, Dec 21, 2011 at 1:26 PM, Hector Geraldino < Hector.Geraldino at ip-soft.net> wrote: > Hi Neil,**** > > ** ** > > This doesn?t seem to be the same concurrency issue I had, but I?m > attaching the patch that fixes my issue anyway. Feel free to test it and > send me back the restuls.**** > > ** ** > > In case it doesn?t work you might try to ?manually? close the channel by > modifying the close() method on the > org.freeswitch.esl.client.inbound.Client. Try to do a channel.disconnect(); > and channel=null; and see what happens.**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Neil Davis > *Sent:* Wednesday, December 21, 2011 11:17 AM > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Threads remain after calling close on > Java client**** > > ** ** > > Here is a thread dump of my Tomcat process at the point when it is hanging > on shutdown. There are a number of threads in a ?waiting on condition? > state that appear to have to do with the netty package on which the > Freeswitch client is dependent.**** > > **** > > **** > > 2011-12-21 09:05:17**** > > Full thread dump Java HotSpot(TM) Client VM (14.3-b01 mixed mode):**** > > **** > > "DestroyJavaVM" prio=6 tid=0x546f7400 nid=0x640 waiting on condition > [0x00000000]**** > > java.lang.Thread.State: RUNNABLE**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-16" prio=6 tid=0x544b5800 nid=0x1098 waiting on condition > [0x55e4f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-15" prio=6 tid=0x54f7f400 nid=0x10f0 waiting on condition > [0x54e5f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-14" prio=6 tid=0x544b3c00 nid=0xdb4 waiting on condition > [0x5670f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-13" prio=6 tid=0x544b4000 nid=0x12c8 waiting on condition > [0x566bf000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-12" prio=6 tid=0x544b4800 nid=0xe18 waiting on condition > [0x5666f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-11" prio=6 tid=0x544b3000 nid=0x620 waiting on condition > [0x5661f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-10" prio=6 tid=0x552a1800 nid=0x131c waiting on condition > [0x565cf000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-9" prio=6 tid=0x552a1400 nid=0x1710 waiting on condition > [0x5657f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-8" prio=6 tid=0x552a0c00 nid=0x1094 waiting on condition > [0x5652f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-7" prio=6 tid=0x552a0800 nid=0x1040 waiting on condition > [0x564df000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-6" prio=6 tid=0x552a0000 nid=0x179c waiting on condition > [0x5648f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-5" prio=6 tid=0x5529fc00 nid=0x3e4 waiting on condition > [0x5643f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-4" prio=6 tid=0x5529f400 nid=0x63c waiting on condition > [0x563ef000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-3" prio=6 tid=0x5529f000 nid=0x17e8 waiting on condition > [0x5639f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-2" prio=6 tid=0x5529e800 nid=0x574 waiting on condition > [0x5634f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-4-thread-1" prio=6 tid=0x5529e400 nid=0x10cc waiting on condition > [0x562ff000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > - parking to wait for <0x09970378> (a > java.util.concurrent.SynchronousQueue$TransferStack)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown > Source)**** > > at > java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) > **** > > at java.util.concurrent.SynchronousQueue.poll(Unknown > Source)**** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-3-thread-1" prio=6 tid=0x546f8800 nid=0x654 waiting on condition > [0x562af000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > - parking to wait for <0x09970580> (a > java.util.concurrent.SynchronousQueue$TransferStack)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown > Source)**** > > at > java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) > **** > > at java.util.concurrent.SynchronousQueue.poll(Unknown > Source)**** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-1" prio=6 tid=0x546f8000 nid=0x594 waiting on condition > [0x5625f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "RMI TCP Connection(2)-10.0.0.22" daemon prio=6 tid=0x551db800 nid=0xf6c > runnable [0x5593f000]**** > > java.lang.Thread.State: RUNNABLE**** > > at java.net.SocketInputStream.socketRead0(Native Method)** > ** > > at java.net.SocketInputStream.read(Unknown Source)**** > > at java.io.BufferedInputStream.fill(Unknown Source)**** > > at java.io.BufferedInputStream.read(Unknown Source)**** > > - locked <0x08f5c2d0> (a java.io.BufferedInputStream)**** > > at java.io.FilterInputStream.read(Unknown Source)**** > > at > sun.rmi.transport.tcp.TCPTransport.handleMessages(Unknown Source)**** > > at > sun.rmi.transport.tcp.TCPTransport$ConnectionHandler.run0(Unknown Source)* > *** > > at > sun.rmi.transport.tcp.TCPTransport$ConnectionHandler.run(Unknown Source)** > ** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.runTask(Unknown Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - <0x08f62b10> (a > java.util.concurrent.locks.ReentrantLock$NonfairSync)**** > > **** > > "JMX server connection timeout 22" daemon prio=6 tid=0x5474fc00 nid=0x424 > in Object.wait() [0x558ef000]**** > > java.lang.Thread.State: TIMED_WAITING (on object monitor)**** > > at java.lang.Object.wait(Native Method)**** > > - waiting on <0x08e9bdb0> (a [I)**** > > at > com.sun.jmx.remote.internal.ServerCommunicatorAdmin$Timeout.run(Unknown > Source)**** > > - locked <0x08e9bdb0> (a [I)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "RMI Scheduler(0)" daemon prio=6 tid=0x5511d400 nid=0x1420 waiting on > condition [0x5589f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > - parking to wait for <0x08e9bdd0> (a > java.util.concurrent.locks.AbstractQueuedSynchronizer$ConditionObject)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > java.util.concurrent.locks.AbstractQueuedSynchronizer$ConditionObject.awaitNanos(Unknown > Source)**** > > at java.util.concurrent.DelayQueue.take(Unknown Source)*** > * > > at > java.util.concurrent.ScheduledThreadPoolExecutor$DelayedWorkQueue.take(Unknown > Source)**** > > at > java.util.concurrent.ScheduledThreadPoolExecutor$DelayedWorkQueue.take(Unknown > Source)**** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "RMI TCP Connection(idle)" daemon prio=6 tid=0x55176800 nid=0x884 waiting > on condition [0x557ff000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > - parking to wait for <0x08ebd088> (a > java.util.concurrent.SynchronousQueue$TransferStack)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown > Source)**** > > at > java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) > **** > > at java.util.concurrent.SynchronousQueue.poll(Unknown > Source)**** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "RMI TCP Accept-0" daemon prio=6 tid=0x5518bc00 nid=0x163c runnable > [0x557af000]**** > > java.lang.Thread.State: RUNNABLE**** > > at java.net.PlainSocketImpl.socketAccept(Native Method)*** > * > > at java.net.PlainSocketImpl.accept(Unknown Source)**** > > - locked <0x08e9e158> (a java.net.SocksSocketImpl)**** > > at java.net.ServerSocket.implAccept(Unknown Source)**** > > at java.net.ServerSocket.accept(Unknown Source)**** > > at > sun.management.jmxremote.LocalRMIServerSocketFactory$1.accept(Unknown > Source)**** > > at > sun.rmi.transport.tcp.TCPTransport$AcceptLoop.executeAcceptLoop(Unknown > Source)**** > > at > sun.rmi.transport.tcp.TCPTransport$AcceptLoop.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "GC Daemon" daemon prio=2 tid=0x5464b000 nid=0x1718 in Object.wait() > [0x5497f000]**** > > java.lang.Thread.State: TIMED_WAITING (on object monitor)**** > > at java.lang.Object.wait(Native Method)**** > > - waiting on <0x089b1270> (a sun.misc.GC$LatencyLock)**** > > at sun.misc.GC$Daemon.run(Unknown Source)**** > > - locked <0x089b1270> (a sun.misc.GC$LatencyLock)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "Low Memory Detector" daemon prio=6 tid=0x01a12c00 nid=0x14fc runnable > [0x00000000]**** > > java.lang.Thread.State: RUNNABLE**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "CompilerThread0" daemon prio=10 tid=0x01a0f800 nid=0x1ec waiting on > condition [0x00000000]**** > > java.lang.Thread.State: RUNNABLE**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "JDWP Event Helper Thread" daemon prio=6 tid=0x01a01400 nid=0x173c > runnable [0x00000000]**** > > java.lang.Thread.State: RUNNABLE**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "Attach Listener" daemon prio=10 tid=0x019f5000 nid=0x13a4 waiting on > condition [0x00000000]**** > > java.lang.Thread.State: RUNNABLE**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "Signal Dispatcher" daemon prio=10 tid=0x019ea000 nid=0x17b8 runnable > [0x00000000]**** > > java.lang.Thread.State: RUNNABLE**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "Finalizer" daemon prio=8 tid=0x019bb400 nid=0x1720 in Object.wait() > [0x53fcf000]**** > > java.lang.Thread.State: WAITING (on object monitor)**** > > at java.lang.Object.wait(Native Method)**** > > - waiting on <0x089b2858> (a > java.lang.ref.ReferenceQueue$Lock)**** > > at java.lang.ref.ReferenceQueue.remove(Unknown Source)**** > > - locked <0x089b2858> (a java.lang.ref.ReferenceQueue$Lock) > **** > > at java.lang.ref.ReferenceQueue.remove(Unknown Source)**** > > at java.lang.ref.Finalizer$FinalizerThread.run(Unknown > Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "Reference Handler" daemon prio=10 tid=0x019ba000 nid=0x53c in > Object.wait() [0x53f7f000]**** > > java.lang.Thread.State: WAITING (on object monitor)**** > > at java.lang.Object.wait(Native Method)**** > > - waiting on <0x089b2878> (a java.lang.ref.Reference$Lock) > **** > > at java.lang.Object.wait(Object.java:485)**** > > at java.lang.ref.Reference$ReferenceHandler.run(Unknown > Source)**** > > - locked <0x089b2878> (a java.lang.ref.Reference$Lock)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "VM Thread" prio=10 tid=0x019b7400 nid=0xab8 runnable **** > > **** > > "VM Periodic Task Thread" prio=10 tid=0x01a1bc00 nid=0xc7c waiting on > condition **** > > **** > > JNI global references: 21240**** > > **** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Hector > Geraldino > *Sent:* Monday, December 19, 2011 8:29 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Threads remain after calling close on > Java client**** > > **** > > Hi Neil,**** > > **** > > Can you get a thread dump of the tomcat process to try to figure out what > this problem is about? Or at least, try to connect the jconsole to the > tomcat process and get the StackTrace of one of these threads to have a > better idea of what is going on.**** > > **** > > IIRC I?ve fixed a couple of bugs for this library, but the patches haven?t > been tested by the main developer (dvarnes) nor integrated on the > repository (freeswitch-contrib). If this problem can be fixed with my > patched code, I would be happy to share it with you.**** > > **** > > Good luck!**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Neil > Davis > *Sent:* Friday, December 16, 2011 7:09 PM > *To:* FreeSWITCH-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Threads remain after calling close on Java > client**** > > **** > > Hi,**** > > **** > > I built a web application that connects to Freeswitch using the > org.freeswitch.esl.client.Client. I connect the Client object from a > Spring annotated service that I call from a Spring controller. I put the > connected client in my ServletContext, so I can access it later to call > client.cancelEventSubscriptions() and client.close() from my > ServletContextListener contextDestroyed method when Tomcat is shutting down. > **** > > **** > > The problem I'm having is that even after I call close on the client, > there are still a bunch of active threads that the client has spawned in > the background. These threads are causing Tomcat to hang when I'm shutting > down. Can anyone suggest an approach that would enable my application to > disconnect the Freeswitch client when Tomcat is shutting down that would > allow Tomcat to shutdown gracefully?**** > > **** > > Below are errors from my Tomcat log for the threads that I have identified > as being related to the Freeswitch client. I don't know how I can get to > these threads to interrupt them and Client.close() seems to leave them > hanging.**** > > **** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-1] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-3-thread-1] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-4-thread-1] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-2] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-3] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-4] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-5] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-6] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-7] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-8] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-9] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-10] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-11] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-12] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-13] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-14] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-15] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-16] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > checkThreadLocalMapForLeaks**** > > SEVERE: The web application [/socketspy] created a ThreadLocal with key of > type [org.jboss.netty.util.internal.ThreadLocalBoolean] (value > [org.jboss.netty.util.internal.ThreadLocalBoolean at 186e192]) and a value > of type [java.lang.Boolean] (value [false]) but failed to remove it when > the web application was stopped. Threads are going to be renewed over time > to try and avoid a probable memory leak. **** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > checkThreadLocalMapForLeaks**** > > SEVERE: The web application [/socketspy] created a ThreadLocal with key of > type [org.jboss.netty.util.CharsetUtil$1] (value > [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type > [java.util.IdentityHashMap] (value > [{windows-1252=sun.nio.cs.MS1252$Encoder at 373f86}]) but failed to remove > it when the web application was stopped. Threads are going to be renewed > over time to try and avoid a probable memory leak. **** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > checkThreadLocalMapForLeaks**** > > SEVERE: The web application [/socketspy] created a ThreadLocal with key of > type [org.jboss.netty.util.internal.ThreadLocalRandom$1] (value > [org.jboss.netty.util.internal.ThreadLocalRandom$1 at 12bb519]) and a value > of type [org.jboss.netty.util.internal.ThreadLocalRandom] (value > [org.jboss.netty.util.internal.ThreadLocalRandom at 7e9dbc]) but failed to > remove it when the web application was stopped. Threads are going to be > renewed over time to try and avoid a probable memory leak. **** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > checkThreadLocalMapForLeaks**** > > SEVERE: The web application [/socketspy] created a ThreadLocal with key of > type [org.jboss.netty.util.CharsetUtil$1] (value > [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type > [java.util.IdentityHashMap] (value > [{windows-1252=sun.nio.cs.MS1252$Encoder at a5b041}]) but failed to remove > it when the web application was stopped. Threads are going to be renewed > over time to try and avoid a probable memory leak. **** > > **** > > **** > > Thanks,**** > > **** > > Neil Davis**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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/20111221/cca6bf0e/attachment-0001.html From sherifomran2000 at yahoo.com Thu Dec 22 00:25:19 2011 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Wed, 21 Dec 2011 13:25:19 -0800 (PST) Subject: [Freeswitch-users] how is codec selected In-Reply-To: Message-ID: <1324502719.74670.YahooMailClassic@web110810.mail.gq1.yahoo.com> hello , does anybody know how to set up the codec selection negotiation list in freeswitch? How can you know which codec has been selected for such a call and which codecs are offered from telephone A and B. I am using log 7 thank you in advance k.regards Sherif -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/689c681d/attachment.html From lloydie.t at gmail.com Thu Dec 22 01:08:45 2011 From: lloydie.t at gmail.com (lloyd thomas) Date: Wed, 21 Dec 2011 22:08:45 +0000 Subject: [Freeswitch-users] Problem with make current In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5B279DBF1C@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C5B279DBF1C@cooper> Message-ID: Nearly. New errors below. not sure what they mean cc1: warnings being treated as errors src/switch_core.c: In function ?switch_system_fork?: src/switch_core.c:2475: error: ignoring return value of ?system?, declared with attribute warn_unused_result make[2]: *** [libfreeswitch_la-switch_core.lo] Error 1 make[2]: Leaving directory `/usr/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2 On 21 December 2011 15:47, Peter Olsson wrote: > Try to remove the autogenerated file > ?/usr/src/freeswitch/libs/spandsp/src/spandsp.h?, then do a ?make > spandsp-reconf?, see if that helps.**** > > ** ** > > Last way out is probably to remove the source and do a fresh git clone, > but it should work anyway.**** > > ** ** > > /Peter **** > > ** ** > > ** ** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *lloyd thomas > *Skickat:* den 21 december 2011 12:21 > *Till:* FreeSWITCH Users Help > *?mne:* Re: [Freeswitch-users] Problem with make current**** > > ** ** > > I have done bootstrap and configure already, but it made no difference**** > > On 21 December 2011 07:00, Peter Olsson > wrote:**** > > Try "make spandsp-reconf" > > Or just do a ./bootstrap.sh and ./configure. > > /Peter**** > > > ----- Reply message ----- > Fr?n: "lloyd thomas" > Datum: ons, dec 21, 2011 03:05 > Rubrik: [Freeswitch-users] Problem with make current > Till: "freeswitch-users" > > Since I am not getting anywhere with my sip registration problem I decided > to update FS from git using make current. > Unfortunately I am getting the following errors. What can I do to resolve? > > In file included from /usr/src/freeswitch/libs/spandsp/src/spandsp.h:101, > from ./src/include/private/switch_core_pvt.h:35, > from src/switch_apr.c:37: > /usr/src/freeswitch/libs/spandsp/src/spandsp/t4_tx.h:145: error: expected > declaration specifiers or ?...? before ?tz_t? > make[2]: *** [libfreeswitch_la-switch_apr.lo] Error 1 > make[2]: Leaving directory `/usr/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/usr/src/freeswitch' > make: *** [current] Error 2**** > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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:4ef1c05f32768994868157! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/54c8ed0f/attachment.html From jkomar at jbox.ca Thu Dec 22 01:11:19 2011 From: jkomar at jbox.ca (Komar, Jason) Date: Wed, 21 Dec 2011 15:11:19 -0700 Subject: [Freeswitch-users] Problem with make current In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C5B279DBF1C@cooper> Message-ID: I'm getting this one at the moment as well when updating on Gentoo. Jason On Wed, Dec 21, 2011 at 3:08 PM, lloyd thomas wrote: > Nearly. > New errors below. not sure what they mean > cc1: warnings being treated as errors > src/switch_core.c: In function ?switch_system_fork?: > src/switch_core.c:2475: error: ignoring return value of ?system?, declared > with attribute warn_unused_result > make[2]: *** [libfreeswitch_la-switch_core.lo] Error 1 > > make[2]: Leaving directory `/usr/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/usr/src/freeswitch' > make: *** [current] Error 2 From Hector.Geraldino at ip-soft.net Thu Dec 22 01:33:21 2011 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Wed, 21 Dec 2011 17:33:21 -0500 Subject: [Freeswitch-users] Threads remain after calling close on Java client In-Reply-To: References: <8b41de351c0d1365e3786e7a60645275@mail.gmail.com> <6A6B4C284AD15042B429EB9D904544AD0225507041@NY1-EXMB-01.ip-soft.net> <895731f6d2c9f7220f752e977f485386@mail.gmail.com> <6A6B4C284AD15042B429EB9D904544AD02255071D4@NY1-EXMB-01.ip-soft.net> Message-ID: <6A6B4C284AD15042B429EB9D904544AD02255071E6@NY1-EXMB-01.ip-soft.net> Sure, I can post it to Jira. Can you please help me to find an appropriate category to create the bug report? There is no project category for freeswitch-contrib projects. Also I'm not sure the patch format is compatible with git (I personally use SVN, don't know if the diff format is different between these two). From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, December 21, 2011 4:14 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Threads remain after calling close on Java client if this patch is necessary can you post it to a jira? On Wed, Dec 21, 2011 at 1:26 PM, Hector Geraldino > wrote: Hi Neil, This doesn't seem to be the same concurrency issue I had, but I'm attaching the patch that fixes my issue anyway. Feel free to test it and send me back the restuls. In case it doesn't work you might try to "manually" close the channel by modifying the close() method on the org.freeswitch.esl.client.inbound.Client. Try to do a channel.disconnect(); and channel=null; and see what happens. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Neil Davis Sent: Wednesday, December 21, 2011 11:17 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Threads remain after calling close on Java client Here is a thread dump of my Tomcat process at the point when it is hanging on shutdown. There are a number of threads in a "waiting on condition" state that appear to have to do with the netty package on which the Freeswitch client is dependent. 2011-12-21 09:05:17 Full thread dump Java HotSpot(TM) Client VM (14.3-b01 mixed mode): "DestroyJavaVM" prio=6 tid=0x546f7400 nid=0x640 waiting on condition [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "pool-5-thread-16" prio=6 tid=0x544b5800 nid=0x1098 waiting on condition [0x55e4f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-15" prio=6 tid=0x54f7f400 nid=0x10f0 waiting on condition [0x54e5f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-14" prio=6 tid=0x544b3c00 nid=0xdb4 waiting on condition [0x5670f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-13" prio=6 tid=0x544b4000 nid=0x12c8 waiting on condition [0x566bf000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-12" prio=6 tid=0x544b4800 nid=0xe18 waiting on condition [0x5666f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-11" prio=6 tid=0x544b3000 nid=0x620 waiting on condition [0x5661f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-10" prio=6 tid=0x552a1800 nid=0x131c waiting on condition [0x565cf000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-9" prio=6 tid=0x552a1400 nid=0x1710 waiting on condition [0x5657f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-8" prio=6 tid=0x552a0c00 nid=0x1094 waiting on condition [0x5652f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-7" prio=6 tid=0x552a0800 nid=0x1040 waiting on condition [0x564df000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-6" prio=6 tid=0x552a0000 nid=0x179c waiting on condition [0x5648f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-5" prio=6 tid=0x5529fc00 nid=0x3e4 waiting on condition [0x5643f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-4" prio=6 tid=0x5529f400 nid=0x63c waiting on condition [0x563ef000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-3" prio=6 tid=0x5529f000 nid=0x17e8 waiting on condition [0x5639f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-2" prio=6 tid=0x5529e800 nid=0x574 waiting on condition [0x5634f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-4-thread-1" prio=6 tid=0x5529e400 nid=0x10cc waiting on condition [0x562ff000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x09970378> (a java.util.concurrent.SynchronousQueue$TransferStack) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) at java.util.concurrent.SynchronousQueue.poll(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-3-thread-1" prio=6 tid=0x546f8800 nid=0x654 waiting on condition [0x562af000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x09970580> (a java.util.concurrent.SynchronousQueue$TransferStack) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) at java.util.concurrent.SynchronousQueue.poll(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-1" prio=6 tid=0x546f8000 nid=0x594 waiting on condition [0x5625f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI TCP Connection(2)-10.0.0.22" daemon prio=6 tid=0x551db800 nid=0xf6c runnable [0x5593f000] java.lang.Thread.State: RUNNABLE at java.net.SocketInputStream.socketRead0(Native Method) at java.net.SocketInputStream.read(Unknown Source) at java.io.BufferedInputStream.fill(Unknown Source) at java.io.BufferedInputStream.read(Unknown Source) - locked <0x08f5c2d0> (a java.io.BufferedInputStream) at java.io.FilterInputStream.read(Unknown Source) at sun.rmi.transport.tcp.TCPTransport.handleMessages(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$ConnectionHandler.run0(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$ConnectionHandler.run(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.runTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - <0x08f62b10> (a java.util.concurrent.locks.ReentrantLock$NonfairSync) "JMX server connection timeout 22" daemon prio=6 tid=0x5474fc00 nid=0x424 in Object.wait() [0x558ef000] java.lang.Thread.State: TIMED_WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x08e9bdb0> (a [I) at com.sun.jmx.remote.internal.ServerCommunicatorAdmin$Timeout.run(Unknown Source) - locked <0x08e9bdb0> (a [I) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI Scheduler(0)" daemon prio=6 tid=0x5511d400 nid=0x1420 waiting on condition [0x5589f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x08e9bdd0> (a java.util.concurrent.locks.AbstractQueuedSynchronizer$ConditionObject) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.locks.AbstractQueuedSynchronizer$ConditionObject.awaitNanos(Unknown Source) at java.util.concurrent.DelayQueue.take(Unknown Source) at java.util.concurrent.ScheduledThreadPoolExecutor$DelayedWorkQueue.take(Unknown Source) at java.util.concurrent.ScheduledThreadPoolExecutor$DelayedWorkQueue.take(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI TCP Connection(idle)" daemon prio=6 tid=0x55176800 nid=0x884 waiting on condition [0x557ff000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x08ebd088> (a java.util.concurrent.SynchronousQueue$TransferStack) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) at java.util.concurrent.SynchronousQueue.poll(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI TCP Accept-0" daemon prio=6 tid=0x5518bc00 nid=0x163c runnable [0x557af000] java.lang.Thread.State: RUNNABLE at java.net.PlainSocketImpl.socketAccept(Native Method) at java.net.PlainSocketImpl.accept(Unknown Source) - locked <0x08e9e158> (a java.net.SocksSocketImpl) at java.net.ServerSocket.implAccept(Unknown Source) at java.net.ServerSocket.accept(Unknown Source) at sun.management.jmxremote.LocalRMIServerSocketFactory$1.accept(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$AcceptLoop.executeAcceptLoop(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$AcceptLoop.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "GC Daemon" daemon prio=2 tid=0x5464b000 nid=0x1718 in Object.wait() [0x5497f000] java.lang.Thread.State: TIMED_WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x089b1270> (a sun.misc.GC$LatencyLock) at sun.misc.GC$Daemon.run(Unknown Source) - locked <0x089b1270> (a sun.misc.GC$LatencyLock) Locked ownable synchronizers: - None "Low Memory Detector" daemon prio=6 tid=0x01a12c00 nid=0x14fc runnable [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "CompilerThread0" daemon prio=10 tid=0x01a0f800 nid=0x1ec waiting on condition [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "JDWP Event Helper Thread" daemon prio=6 tid=0x01a01400 nid=0x173c runnable [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "Attach Listener" daemon prio=10 tid=0x019f5000 nid=0x13a4 waiting on condition [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "Signal Dispatcher" daemon prio=10 tid=0x019ea000 nid=0x17b8 runnable [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "Finalizer" daemon prio=8 tid=0x019bb400 nid=0x1720 in Object.wait() [0x53fcf000] java.lang.Thread.State: WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x089b2858> (a java.lang.ref.ReferenceQueue$Lock) at java.lang.ref.ReferenceQueue.remove(Unknown Source) - locked <0x089b2858> (a java.lang.ref.ReferenceQueue$Lock) at java.lang.ref.ReferenceQueue.remove(Unknown Source) at java.lang.ref.Finalizer$FinalizerThread.run(Unknown Source) Locked ownable synchronizers: - None "Reference Handler" daemon prio=10 tid=0x019ba000 nid=0x53c in Object.wait() [0x53f7f000] java.lang.Thread.State: WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x089b2878> (a java.lang.ref.Reference$Lock) at java.lang.Object.wait(Object.java:485) at java.lang.ref.Reference$ReferenceHandler.run(Unknown Source) - locked <0x089b2878> (a java.lang.ref.Reference$Lock) Locked ownable synchronizers: - None "VM Thread" prio=10 tid=0x019b7400 nid=0xab8 runnable "VM Periodic Task Thread" prio=10 tid=0x01a1bc00 nid=0xc7c waiting on condition JNI global references: 21240 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hector Geraldino Sent: Monday, December 19, 2011 8:29 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Threads remain after calling close on Java client Hi Neil, Can you get a thread dump of the tomcat process to try to figure out what this problem is about? Or at least, try to connect the jconsole to the tomcat process and get the StackTrace of one of these threads to have a better idea of what is going on. IIRC I've fixed a couple of bugs for this library, but the patches haven't been tested by the main developer (dvarnes) nor integrated on the repository (freeswitch-contrib). If this problem can be fixed with my patched code, I would be happy to share it with you. Good luck! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Neil Davis Sent: Friday, December 16, 2011 7:09 PM To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] Threads remain after calling close on Java client Hi, I built a web application that connects to Freeswitch using the org.freeswitch.esl.client.Client. I connect the Client object from a Spring annotated service that I call from a Spring controller. I put the connected client in my ServletContext, so I can access it later to call client.cancelEventSubscriptions() and client.close() from my ServletContextListener contextDestroyed method when Tomcat is shutting down. The problem I'm having is that even after I call close on the client, there are still a bunch of active threads that the client has spawned in the background. These threads are causing Tomcat to hang when I'm shutting down. Can anyone suggest an approach that would enable my application to disconnect the Freeswitch client when Tomcat is shutting down that would allow Tomcat to shutdown gracefully? Below are errors from my Tomcat log for the threads that I have identified as being related to the Freeswitch client. I don't know how I can get to these threads to interrupt them and Client.close() seems to leave them hanging. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-3-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-4-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-2] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-3] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-4] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-5] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-6] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-7] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-8] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-9] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-10] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-11] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-12] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-13] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-14] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-15] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-16] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.internal.ThreadLocalBoolean] (value [org.jboss.netty.util.internal.ThreadLocalBoolean at 186e192]) and a value of type [java.lang.Boolean] (value [false]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.CharsetUtil$1] (value [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type [java.util.IdentityHashMap] (value [{windows-1252=sun.nio.cs.MS1252$Encoder at 373f86}]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.internal.ThreadLocalRandom$1] (value [org.jboss.netty.util.internal.ThreadLocalRandom$1 at 12bb519]) and a value of type [org.jboss.netty.util.internal.ThreadLocalRandom] (value [org.jboss.netty.util.internal.ThreadLocalRandom at 7e9dbc]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.CharsetUtil$1] (value [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type [java.util.IdentityHashMap] (value [{windows-1252=sun.nio.cs.MS1252$Encoder at a5b041}]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Thanks, Neil Davis _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://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/20111221/3b994a90/attachment-0001.html From anthony.minessale at gmail.com Thu Dec 22 01:36:20 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Wed, 21 Dec 2011 16:36:20 -0600 Subject: [Freeswitch-users] Threads remain after calling close on Java client In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD02255071E6@NY1-EXMB-01.ip-soft.net> References: <8b41de351c0d1365e3786e7a60645275@mail.gmail.com> <6A6B4C284AD15042B429EB9D904544AD0225507041@NY1-EXMB-01.ip-soft.net> <895731f6d2c9f7220f752e977f485386@mail.gmail.com> <6A6B4C284AD15042B429EB9D904544AD02255071D4@NY1-EXMB-01.ip-soft.net> <6A6B4C284AD15042B429EB9D904544AD02255071E6@NY1-EXMB-01.ip-soft.net> Message-ID: you can just file it under esl or general, the patch should be ok as long as its a unified diff or you can get the FS tree and patch your local copy and submit a git diff from that if all else fails. On Wed, Dec 21, 2011 at 4:33 PM, Hector Geraldino < Hector.Geraldino at ip-soft.net> wrote: > Sure, I can post it to Jira. Can you please help me to find an > appropriate category to create the bug report? There is no project category > for freeswitch-contrib projects. **** > > ** ** > > Also I?m not sure the patch format is compatible with git (I personally > use SVN, don?t know if the diff format is different between these two).*** > * > > ** ** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Anthony > Minessale > *Sent:* Wednesday, December 21, 2011 4:14 PM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Threads remain after calling close on > Java client**** > > ** ** > > if this patch is necessary can you post it to a jira?**** > > On Wed, Dec 21, 2011 at 1:26 PM, Hector Geraldino < > Hector.Geraldino at ip-soft.net> wrote:**** > > Hi Neil,**** > > **** > > This doesn?t seem to be the same concurrency issue I had, but I?m > attaching the patch that fixes my issue anyway. Feel free to test it and > send me back the restuls.**** > > **** > > In case it doesn?t work you might try to ?manually? close the channel by > modifying the close() method on the > org.freeswitch.esl.client.inbound.Client. Try to do a channel.disconnect(); > and channel=null; and see what happens.**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Neil Davis > *Sent:* Wednesday, December 21, 2011 11:17 AM**** > > > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Threads remain after calling close on > Java client**** > > **** > > Here is a thread dump of my Tomcat process at the point when it is hanging > on shutdown. There are a number of threads in a ?waiting on condition? > state that appear to have to do with the netty package on which the > Freeswitch client is dependent.**** > > **** > > **** > > 2011-12-21 09:05:17**** > > Full thread dump Java HotSpot(TM) Client VM (14.3-b01 mixed mode):**** > > **** > > "DestroyJavaVM" prio=6 tid=0x546f7400 nid=0x640 waiting on condition > [0x00000000]**** > > java.lang.Thread.State: RUNNABLE**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-16" prio=6 tid=0x544b5800 nid=0x1098 waiting on condition > [0x55e4f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-15" prio=6 tid=0x54f7f400 nid=0x10f0 waiting on condition > [0x54e5f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-14" prio=6 tid=0x544b3c00 nid=0xdb4 waiting on condition > [0x5670f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-13" prio=6 tid=0x544b4000 nid=0x12c8 waiting on condition > [0x566bf000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-12" prio=6 tid=0x544b4800 nid=0xe18 waiting on condition > [0x5666f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-11" prio=6 tid=0x544b3000 nid=0x620 waiting on condition > [0x5661f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-10" prio=6 tid=0x552a1800 nid=0x131c waiting on condition > [0x565cf000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-9" prio=6 tid=0x552a1400 nid=0x1710 waiting on condition > [0x5657f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-8" prio=6 tid=0x552a0c00 nid=0x1094 waiting on condition > [0x5652f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-7" prio=6 tid=0x552a0800 nid=0x1040 waiting on condition > [0x564df000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-6" prio=6 tid=0x552a0000 nid=0x179c waiting on condition > [0x5648f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-5" prio=6 tid=0x5529fc00 nid=0x3e4 waiting on condition > [0x5643f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-4" prio=6 tid=0x5529f400 nid=0x63c waiting on condition > [0x563ef000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-3" prio=6 tid=0x5529f000 nid=0x17e8 waiting on condition > [0x5639f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-2" prio=6 tid=0x5529e800 nid=0x574 waiting on condition > [0x5634f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-4-thread-1" prio=6 tid=0x5529e400 nid=0x10cc waiting on condition > [0x562ff000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > - parking to wait for <0x09970378> (a > java.util.concurrent.SynchronousQueue$TransferStack)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown > Source)**** > > at > java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) > **** > > at java.util.concurrent.SynchronousQueue.poll(Unknown > Source)**** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-3-thread-1" prio=6 tid=0x546f8800 nid=0x654 waiting on condition > [0x562af000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > - parking to wait for <0x09970580> (a > java.util.concurrent.SynchronousQueue$TransferStack)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown > Source)**** > > at > java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) > **** > > at java.util.concurrent.SynchronousQueue.poll(Unknown > Source)**** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "pool-5-thread-1" prio=6 tid=0x546f8000 nid=0x594 waiting on condition > [0x5625f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) > **** > > at > org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) > **** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "RMI TCP Connection(2)-10.0.0.22" daemon prio=6 tid=0x551db800 nid=0xf6c > runnable [0x5593f000]**** > > java.lang.Thread.State: RUNNABLE**** > > at java.net.SocketInputStream.socketRead0(Native Method)** > ** > > at java.net.SocketInputStream.read(Unknown Source)**** > > at java.io.BufferedInputStream.fill(Unknown Source)**** > > at java.io.BufferedInputStream.read(Unknown Source)**** > > - locked <0x08f5c2d0> (a java.io.BufferedInputStream)**** > > at java.io.FilterInputStream.read(Unknown Source)**** > > at > sun.rmi.transport.tcp.TCPTransport.handleMessages(Unknown Source)**** > > at > sun.rmi.transport.tcp.TCPTransport$ConnectionHandler.run0(Unknown Source)* > *** > > at > sun.rmi.transport.tcp.TCPTransport$ConnectionHandler.run(Unknown Source)** > ** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.runTask(Unknown Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - <0x08f62b10> (a > java.util.concurrent.locks.ReentrantLock$NonfairSync)**** > > **** > > "JMX server connection timeout 22" daemon prio=6 tid=0x5474fc00 nid=0x424 > in Object.wait() [0x558ef000]**** > > java.lang.Thread.State: TIMED_WAITING (on object monitor)**** > > at java.lang.Object.wait(Native Method)**** > > - waiting on <0x08e9bdb0> (a [I)**** > > at > com.sun.jmx.remote.internal.ServerCommunicatorAdmin$Timeout.run(Unknown > Source)**** > > - locked <0x08e9bdb0> (a [I)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "RMI Scheduler(0)" daemon prio=6 tid=0x5511d400 nid=0x1420 waiting on > condition [0x5589f000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > - parking to wait for <0x08e9bdd0> (a > java.util.concurrent.locks.AbstractQueuedSynchronizer$ConditionObject)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > java.util.concurrent.locks.AbstractQueuedSynchronizer$ConditionObject.awaitNanos(Unknown > Source)**** > > at java.util.concurrent.DelayQueue.take(Unknown Source)*** > * > > at > java.util.concurrent.ScheduledThreadPoolExecutor$DelayedWorkQueue.take(Unknown > Source)**** > > at > java.util.concurrent.ScheduledThreadPoolExecutor$DelayedWorkQueue.take(Unknown > Source)**** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "RMI TCP Connection(idle)" daemon prio=6 tid=0x55176800 nid=0x884 waiting > on condition [0x557ff000]**** > > java.lang.Thread.State: TIMED_WAITING (parking)**** > > at sun.misc.Unsafe.park(Native Method)**** > > - parking to wait for <0x08ebd088> (a > java.util.concurrent.SynchronousQueue$TransferStack)**** > > at > java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source)**** > > at > java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown > Source)**** > > at > java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) > **** > > at java.util.concurrent.SynchronousQueue.poll(Unknown > Source)**** > > at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown > Source)**** > > at > java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "RMI TCP Accept-0" daemon prio=6 tid=0x5518bc00 nid=0x163c runnable > [0x557af000]**** > > java.lang.Thread.State: RUNNABLE**** > > at java.net.PlainSocketImpl.socketAccept(Native Method)*** > * > > at java.net.PlainSocketImpl.accept(Unknown Source)**** > > - locked <0x08e9e158> (a java.net.SocksSocketImpl)**** > > at java.net.ServerSocket.implAccept(Unknown Source)**** > > at java.net.ServerSocket.accept(Unknown Source)**** > > at > sun.management.jmxremote.LocalRMIServerSocketFactory$1.accept(Unknown > Source)**** > > at > sun.rmi.transport.tcp.TCPTransport$AcceptLoop.executeAcceptLoop(Unknown > Source)**** > > at > sun.rmi.transport.tcp.TCPTransport$AcceptLoop.run(Unknown Source)**** > > at java.lang.Thread.run(Unknown Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "GC Daemon" daemon prio=2 tid=0x5464b000 nid=0x1718 in Object.wait() > [0x5497f000]**** > > java.lang.Thread.State: TIMED_WAITING (on object monitor)**** > > at java.lang.Object.wait(Native Method)**** > > - waiting on <0x089b1270> (a sun.misc.GC$LatencyLock)**** > > at sun.misc.GC$Daemon.run(Unknown Source)**** > > - locked <0x089b1270> (a sun.misc.GC$LatencyLock)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "Low Memory Detector" daemon prio=6 tid=0x01a12c00 nid=0x14fc runnable > [0x00000000]**** > > java.lang.Thread.State: RUNNABLE**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "CompilerThread0" daemon prio=10 tid=0x01a0f800 nid=0x1ec waiting on > condition [0x00000000]**** > > java.lang.Thread.State: RUNNABLE**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "JDWP Event Helper Thread" daemon prio=6 tid=0x01a01400 nid=0x173c > runnable [0x00000000]**** > > java.lang.Thread.State: RUNNABLE**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "Attach Listener" daemon prio=10 tid=0x019f5000 nid=0x13a4 waiting on > condition [0x00000000]**** > > java.lang.Thread.State: RUNNABLE**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "Signal Dispatcher" daemon prio=10 tid=0x019ea000 nid=0x17b8 runnable > [0x00000000]**** > > java.lang.Thread.State: RUNNABLE**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "Finalizer" daemon prio=8 tid=0x019bb400 nid=0x1720 in Object.wait() > [0x53fcf000]**** > > java.lang.Thread.State: WAITING (on object monitor)**** > > at java.lang.Object.wait(Native Method)**** > > - waiting on <0x089b2858> (a > java.lang.ref.ReferenceQueue$Lock)**** > > at java.lang.ref.ReferenceQueue.remove(Unknown Source)**** > > - locked <0x089b2858> (a java.lang.ref.ReferenceQueue$Lock) > **** > > at java.lang.ref.ReferenceQueue.remove(Unknown Source)**** > > at java.lang.ref.Finalizer$FinalizerThread.run(Unknown > Source)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "Reference Handler" daemon prio=10 tid=0x019ba000 nid=0x53c in > Object.wait() [0x53f7f000]**** > > java.lang.Thread.State: WAITING (on object monitor)**** > > at java.lang.Object.wait(Native Method)**** > > - waiting on <0x089b2878> (a java.lang.ref.Reference$Lock) > **** > > at java.lang.Object.wait(Object.java:485)**** > > at java.lang.ref.Reference$ReferenceHandler.run(Unknown > Source)**** > > - locked <0x089b2878> (a java.lang.ref.Reference$Lock)**** > > **** > > Locked ownable synchronizers:**** > > - None**** > > **** > > "VM Thread" prio=10 tid=0x019b7400 nid=0xab8 runnable **** > > **** > > "VM Periodic Task Thread" prio=10 tid=0x01a1bc00 nid=0xc7c waiting on > condition **** > > **** > > JNI global references: 21240**** > > **** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Hector > Geraldino > *Sent:* Monday, December 19, 2011 8:29 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] Threads remain after calling close on > Java client**** > > **** > > Hi Neil,**** > > **** > > Can you get a thread dump of the tomcat process to try to figure out what > this problem is about? Or at least, try to connect the jconsole to the > tomcat process and get the StackTrace of one of these threads to have a > better idea of what is going on.**** > > **** > > IIRC I?ve fixed a couple of bugs for this library, but the patches haven?t > been tested by the main developer (dvarnes) nor integrated on the > repository (freeswitch-contrib). If this problem can be fixed with my > patched code, I would be happy to share it with you.**** > > **** > > Good luck!**** > > **** > > *From:* freeswitch-users-bounces at lists.freeswitch.org > [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Neil > Davis > *Sent:* Friday, December 16, 2011 7:09 PM > *To:* FreeSWITCH-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Threads remain after calling close on Java > client**** > > **** > > Hi,**** > > **** > > I built a web application that connects to Freeswitch using the > org.freeswitch.esl.client.Client. I connect the Client object from a > Spring annotated service that I call from a Spring controller. I put the > connected client in my ServletContext, so I can access it later to call > client.cancelEventSubscriptions() and client.close() from my > ServletContextListener contextDestroyed method when Tomcat is shutting down. > **** > > **** > > The problem I'm having is that even after I call close on the client, > there are still a bunch of active threads that the client has spawned in > the background. These threads are causing Tomcat to hang when I'm shutting > down. Can anyone suggest an approach that would enable my application to > disconnect the Freeswitch client when Tomcat is shutting down that would > allow Tomcat to shutdown gracefully?**** > > **** > > Below are errors from my Tomcat log for the threads that I have identified > as being related to the Freeswitch client. I don't know how I can get to > these threads to interrupt them and Client.close() seems to leave them > hanging.**** > > **** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-1] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-3-thread-1] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-4-thread-1] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-2] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-3] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-4] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-5] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-6] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-7] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-8] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-9] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-10] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-11] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-12] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-13] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-14] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-15] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > clearReferencesThreads**** > > SEVERE: The web application [/socketspy] appears to have started a thread > named [pool-5-thread-16] but has failed to stop it. This is very likely to > create a memory leak.**** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > checkThreadLocalMapForLeaks**** > > SEVERE: The web application [/socketspy] created a ThreadLocal with key of > type [org.jboss.netty.util.internal.ThreadLocalBoolean] (value > [org.jboss.netty.util.internal.ThreadLocalBoolean at 186e192]) and a value > of type [java.lang.Boolean] (value [false]) but failed to remove it when > the web application was stopped. Threads are going to be renewed over time > to try and avoid a probable memory leak. **** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > checkThreadLocalMapForLeaks**** > > SEVERE: The web application [/socketspy] created a ThreadLocal with key of > type [org.jboss.netty.util.CharsetUtil$1] (value > [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type > [java.util.IdentityHashMap] (value > [{windows-1252=sun.nio.cs.MS1252$Encoder at 373f86}]) but failed to remove > it when the web application was stopped. Threads are going to be renewed > over time to try and avoid a probable memory leak. **** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > checkThreadLocalMapForLeaks**** > > SEVERE: The web application [/socketspy] created a ThreadLocal with key of > type [org.jboss.netty.util.internal.ThreadLocalRandom$1] (value > [org.jboss.netty.util.internal.ThreadLocalRandom$1 at 12bb519]) and a value > of type [org.jboss.netty.util.internal.ThreadLocalRandom] (value > [org.jboss.netty.util.internal.ThreadLocalRandom at 7e9dbc]) but failed to > remove it when the web application was stopped. Threads are going to be > renewed over time to try and avoid a probable memory leak. **** > > Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader > checkThreadLocalMapForLeaks**** > > SEVERE: The web application [/socketspy] created a ThreadLocal with key of > type [org.jboss.netty.util.CharsetUtil$1] (value > [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type > [java.util.IdentityHashMap] (value > [{windows-1252=sun.nio.cs.MS1252$Encoder at a5b041}]) but failed to remove > it when the web application was stopped. Threads are going to be renewed > over time to try and avoid a probable memory leak. **** > > **** > > **** > > Thanks,**** > > **** > > Neil Davis**** > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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 > > -- 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/20111221/3e23591f/attachment-0001.html From gcd at i.ph Thu Dec 22 01:54:29 2011 From: gcd at i.ph (Nandy Dagondon) Date: Thu, 22 Dec 2011 06:54:29 +0800 Subject: [Freeswitch-users] how is codec selected In-Reply-To: <1324502719.74670.YahooMailClassic@web110810.mail.gq1.yahoo.com> References: <1324502719.74670.YahooMailClassic@web110810.mail.gq1.yahoo.com> Message-ID: it's found here: http://wiki.freeswitch.org/wiki/Codec_negotiation hope this answers your question. On Thu, Dec 22, 2011 at 5:25 AM, Sherif Omran wrote: > hello , > > does anybody know how to set up the codec selection negotiation list in > freeswitch? How can you know which codec has been selected for such a call > and which codecs are offered from telephone A and B. > > I am using log 7 > > > thank you in advance > > k.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/20111222/6f3e4d8b/attachment.html From jeff at jefflenk.com Thu Dec 22 02:12:36 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 21 Dec 2011 15:12:36 -0800 (PST) Subject: [Freeswitch-users] Problem with make current In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C5B279DBF1C@cooper> Message-ID: <1324509156638-7116980.post@n2.nabble.com> thats crazy that a compiler should care about checking a return parameter. anyways you should open a Jira on this so it gets fixed. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-make-current-tp7113961p7116980.html Sent from the freeswitch-users mailing list archive at Nabble.com. From ndavis at inetwork.com Thu Dec 22 02:18:50 2011 From: ndavis at inetwork.com (Neil Davis) Date: Wed, 21 Dec 2011 16:18:50 -0700 Subject: [Freeswitch-users] Threads remain after calling close on Java client In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD02255071D4@NY1-EXMB-01.ip-soft.net> References: <8b41de351c0d1365e3786e7a60645275@mail.gmail.com> <6A6B4C284AD15042B429EB9D904544AD0225507041@NY1-EXMB-01.ip-soft.net> <895731f6d2c9f7220f752e977f485386@mail.gmail.com> <6A6B4C284AD15042B429EB9D904544AD02255071D4@NY1-EXMB-01.ip-soft.net> Message-ID: <9ffbcd24295d4daec70f26135a902515@mail.gmail.com> Hector, Thanks for your suggestions. I tried both your fix and modifying the close method as suggested, but I still have threads hanging around in the TIMED-WAITING status. They eventually terminate after about a minute. For the time being, I?m just stalling the Tomcat shutdown with Thread.sleeps until all the hanging threads have terminated. Tomcat is then able to shutdown gracefully. I?ll look a little more at the esl code as I have time and update the mailing list if I find a solution. Thanks, Neil Davis *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Hector Geraldino *Sent:* Wednesday, December 21, 2011 12:26 PM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Threads remain after calling close on Java client Hi Neil, This doesn?t seem to be the same concurrency issue I had, but I?m attaching the patch that fixes my issue anyway. Feel free to test it and send me back the restuls. In case it doesn?t work you might try to ?manually? close the channel by modifying the close() method on the org.freeswitch.esl.client.inbound.Client. Try to do a channel.disconnect(); and channel=null; and see what happens. *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Neil Davis *Sent:* Wednesday, December 21, 2011 11:17 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Threads remain after calling close on Java client Here is a thread dump of my Tomcat process at the point when it is hanging on shutdown. There are a number of threads in a ?waiting on condition? state that appear to have to do with the netty package on which the Freeswitch client is dependent. 2011-12-21 09:05:17 Full thread dump Java HotSpot(TM) Client VM (14.3-b01 mixed mode): "DestroyJavaVM" prio=6 tid=0x546f7400 nid=0x640 waiting on condition [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "pool-5-thread-16" prio=6 tid=0x544b5800 nid=0x1098 waiting on condition [0x55e4f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-15" prio=6 tid=0x54f7f400 nid=0x10f0 waiting on condition [0x54e5f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-14" prio=6 tid=0x544b3c00 nid=0xdb4 waiting on condition [0x5670f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-13" prio=6 tid=0x544b4000 nid=0x12c8 waiting on condition [0x566bf000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-12" prio=6 tid=0x544b4800 nid=0xe18 waiting on condition [0x5666f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-11" prio=6 tid=0x544b3000 nid=0x620 waiting on condition [0x5661f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-10" prio=6 tid=0x552a1800 nid=0x131c waiting on condition [0x565cf000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-9" prio=6 tid=0x552a1400 nid=0x1710 waiting on condition [0x5657f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-8" prio=6 tid=0x552a0c00 nid=0x1094 waiting on condition [0x5652f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-7" prio=6 tid=0x552a0800 nid=0x1040 waiting on condition [0x564df000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-6" prio=6 tid=0x552a0000 nid=0x179c waiting on condition [0x5648f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-5" prio=6 tid=0x5529fc00 nid=0x3e4 waiting on condition [0x5643f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-4" prio=6 tid=0x5529f400 nid=0x63c waiting on condition [0x563ef000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-3" prio=6 tid=0x5529f000 nid=0x17e8 waiting on condition [0x5639f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-2" prio=6 tid=0x5529e800 nid=0x574 waiting on condition [0x5634f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-4-thread-1" prio=6 tid=0x5529e400 nid=0x10cc waiting on condition [0x562ff000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x09970378> (a java.util.concurrent.SynchronousQueue$TransferStack) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) at java.util.concurrent.SynchronousQueue.poll(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-3-thread-1" prio=6 tid=0x546f8800 nid=0x654 waiting on condition [0x562af000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x09970580> (a java.util.concurrent.SynchronousQueue$TransferStack) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) at java.util.concurrent.SynchronousQueue.poll(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-1" prio=6 tid=0x546f8000 nid=0x594 waiting on condition [0x5625f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI TCP Connection(2)-10.0.0.22" daemon prio=6 tid=0x551db800 nid=0xf6c runnable [0x5593f000] java.lang.Thread.State: RUNNABLE at java.net.SocketInputStream.socketRead0(Native Method) at java.net.SocketInputStream.read(Unknown Source) at java.io.BufferedInputStream.fill(Unknown Source) at java.io.BufferedInputStream.read(Unknown Source) - locked <0x08f5c2d0> (a java.io.BufferedInputStream) at java.io.FilterInputStream.read(Unknown Source) at sun.rmi.transport.tcp.TCPTransport.handleMessages(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$ConnectionHandler.run0(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$ConnectionHandler.run(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.runTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - <0x08f62b10> (a java.util.concurrent.locks.ReentrantLock$NonfairSync) "JMX server connection timeout 22" daemon prio=6 tid=0x5474fc00 nid=0x424 in Object.wait() [0x558ef000] java.lang.Thread.State: TIMED_WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x08e9bdb0> (a [I) at com.sun.jmx.remote.internal.ServerCommunicatorAdmin$Timeout.run(Unknown Source) - locked <0x08e9bdb0> (a [I) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI Scheduler(0)" daemon prio=6 tid=0x5511d400 nid=0x1420 waiting on condition [0x5589f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x08e9bdd0> (a java.util.concurrent.locks.AbstractQueuedSynchronizer$ConditionObject) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.locks.AbstractQueuedSynchronizer$ConditionObject.awaitNanos(Unknown Source) at java.util.concurrent.DelayQueue.take(Unknown Source) at java.util.concurrent.ScheduledThreadPoolExecutor$DelayedWorkQueue.take(Unknown Source) at java.util.concurrent.ScheduledThreadPoolExecutor$DelayedWorkQueue.take(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI TCP Connection(idle)" daemon prio=6 tid=0x55176800 nid=0x884 waiting on condition [0x557ff000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x08ebd088> (a java.util.concurrent.SynchronousQueue$TransferStack) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) at java.util.concurrent.SynchronousQueue.poll(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI TCP Accept-0" daemon prio=6 tid=0x5518bc00 nid=0x163c runnable [0x557af000] java.lang.Thread.State: RUNNABLE at java.net.PlainSocketImpl.socketAccept(Native Method) at java.net.PlainSocketImpl.accept(Unknown Source) - locked <0x08e9e158> (a java.net.SocksSocketImpl) at java.net.ServerSocket.implAccept(Unknown Source) at java.net.ServerSocket.accept(Unknown Source) at sun.management.jmxremote.LocalRMIServerSocketFactory$1.accept(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$AcceptLoop.executeAcceptLoop(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$AcceptLoop.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "GC Daemon" daemon prio=2 tid=0x5464b000 nid=0x1718 in Object.wait() [0x5497f000] java.lang.Thread.State: TIMED_WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x089b1270> (a sun.misc.GC$LatencyLock) at sun.misc.GC$Daemon.run(Unknown Source) - locked <0x089b1270> (a sun.misc.GC$LatencyLock) Locked ownable synchronizers: - None "Low Memory Detector" daemon prio=6 tid=0x01a12c00 nid=0x14fc runnable [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "CompilerThread0" daemon prio=10 tid=0x01a0f800 nid=0x1ec waiting on condition [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "JDWP Event Helper Thread" daemon prio=6 tid=0x01a01400 nid=0x173c runnable [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "Attach Listener" daemon prio=10 tid=0x019f5000 nid=0x13a4 waiting on condition [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "Signal Dispatcher" daemon prio=10 tid=0x019ea000 nid=0x17b8 runnable [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "Finalizer" daemon prio=8 tid=0x019bb400 nid=0x1720 in Object.wait() [0x53fcf000] java.lang.Thread.State: WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x089b2858> (a java.lang.ref.ReferenceQueue$Lock) at java.lang.ref.ReferenceQueue.remove(Unknown Source) - locked <0x089b2858> (a java.lang.ref.ReferenceQueue$Lock) at java.lang.ref.ReferenceQueue.remove(Unknown Source) at java.lang.ref.Finalizer$FinalizerThread.run(Unknown Source) Locked ownable synchronizers: - None "Reference Handler" daemon prio=10 tid=0x019ba000 nid=0x53c in Object.wait() [0x53f7f000] java.lang.Thread.State: WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x089b2878> (a java.lang.ref.Reference$Lock) at java.lang.Object.wait(Object.java:485) at java.lang.ref.Reference$ReferenceHandler.run(Unknown Source) - locked <0x089b2878> (a java.lang.ref.Reference$Lock) Locked ownable synchronizers: - None "VM Thread" prio=10 tid=0x019b7400 nid=0xab8 runnable "VM Periodic Task Thread" prio=10 tid=0x01a1bc00 nid=0xc7c waiting on condition JNI global references: 21240 *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Hector Geraldino *Sent:* Monday, December 19, 2011 8:29 AM *To:* FreeSWITCH Users Help *Subject:* Re: [Freeswitch-users] Threads remain after calling close on Java client Hi Neil, Can you get a thread dump of the tomcat process to try to figure out what this problem is about? Or at least, try to connect the jconsole to the tomcat process and get the StackTrace of one of these threads to have a better idea of what is going on. IIRC I?ve fixed a couple of bugs for this library, but the patches haven?t been tested by the main developer (dvarnes) nor integrated on the repository (freeswitch-contrib). If this problem can be fixed with my patched code, I would be happy to share it with you. Good luck! *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Neil Davis *Sent:* Friday, December 16, 2011 7:09 PM *To:* FreeSWITCH-users at lists.freeswitch.org *Subject:* [Freeswitch-users] Threads remain after calling close on Java client Hi, I built a web application that connects to Freeswitch using the org.freeswitch.esl.client.Client. I connect the Client object from a Spring annotated service that I call from a Spring controller. I put the connected client in my ServletContext, so I can access it later to call client.cancelEventSubscriptions() and client.close() from my ServletContextListener contextDestroyed method when Tomcat is shutting down. The problem I'm having is that even after I call close on the client, there are still a bunch of active threads that the client has spawned in the background. These threads are causing Tomcat to hang when I'm shutting down. Can anyone suggest an approach that would enable my application to disconnect the Freeswitch client when Tomcat is shutting down that would allow Tomcat to shutdown gracefully? Below are errors from my Tomcat log for the threads that I have identified as being related to the Freeswitch client. I don't know how I can get to these threads to interrupt them and Client.close() seems to leave them hanging. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-3-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-4-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-2] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-3] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-4] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-5] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-6] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-7] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-8] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-9] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-10] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-11] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-12] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-13] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-14] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-15] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-16] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.internal.ThreadLocalBoolean] (value [org.jboss.netty.util.internal.ThreadLocalBoolean at 186e192]) and a value of type [java.lang.Boolean] (value [false]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.CharsetUtil$1] (value [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type [java.util.IdentityHashMap] (value [{windows-1252=sun.nio.cs.MS1252$Encoder at 373f86}]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.internal.ThreadLocalRandom$1] (value [org.jboss.netty.util.internal.ThreadLocalRandom$1 at 12bb519]) and a value of type [org.jboss.netty.util.internal.ThreadLocalRandom] (value [org.jboss.netty.util.internal.ThreadLocalRandom at 7e9dbc]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.CharsetUtil$1] (value [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type [java.util.IdentityHashMap] (value [{windows-1252=sun.nio.cs.MS1252$Encoder at a5b041}]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Thanks, Neil Davis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/90af41fa/attachment-0001.html From justlikeef at gmail.com Thu Dec 22 02:35:43 2011 From: justlikeef at gmail.com (Rob Hutton) Date: Wed, 21 Dec 2011 18:35:43 -0500 Subject: [Freeswitch-users] Need help with PortAudio In-Reply-To: <4EF1BC79.4000903@puzzled.xs4all.nl> References: <201112202245.57010.justlikeef@gmail.com> <4EF1BC79.4000903@puzzled.xs4all.nl> Message-ID: <201112211835.44158.justlikeef@gmail.com> It looks like portaudio has completely changed, and pablio no longer works. Has anyone started an effort to rewrite mod_portaudio for portaudio v19? I have seen a couple of notes of new code on the list, but haven't seen anything... On Wednesday 21 December 2011 06:01:13 Patrick Lists wrote: > On 21-12-11 04:45, Rob Hutton wrote: > > I am trying to get portaudio working, and have run into a couple of problems > > > > > > 1) If portaudio cannot access the device, it causes freeswitch to segfault > > > > > > 2) I have set up a dialplan following the intercom example which seems > > to be working, but I get no audio. I have tried setting every device > > shown in devlist as the output device. > > > > > > 3) When I issue a "pa play". the cli hangs, even if I give it a timeout. > > > > > > Any thoughts? > > I think someone reported on the mailing list a while back that he solved > his troubles with portaudio by upgrading the portaudio source to the > latest stable release which currently is: > > http://www.portaudio.com/archives/pa_stable_v19_20111121.tgz > > Regards, > Patrick > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/c0edd9ce/attachment.html From lloydie.t at gmail.com Thu Dec 22 02:36:15 2011 From: lloydie.t at gmail.com (lloyd thomas) Date: Wed, 21 Dec 2011 23:36:15 +0000 Subject: [Freeswitch-users] Problem with make current In-Reply-To: <1324509156638-7116980.post@n2.nabble.com> References: <549CFEF87AEDE841A38E9D15EAB4C04C5B279DBF1C@cooper> <1324509156638-7116980.post@n2.nabble.com> Message-ID: Oh Dear. I was hoping to fix my registration problems with this. Will open a ticket On 21 December 2011 23:12, Jeff Lenk wrote: > thats crazy that a compiler should care about checking a return parameter. > anyways you should open a Jira on this so it gets fixed. > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Problem-with-make-current-tp7113961p7116980.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/20111221/9d4f183f/attachment.html From Hector.Geraldino at ip-soft.net Thu Dec 22 02:42:02 2011 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Wed, 21 Dec 2011 18:42:02 -0500 Subject: [Freeswitch-users] Threads remain after calling close on Java client In-Reply-To: References: <8b41de351c0d1365e3786e7a60645275@mail.gmail.com> <6A6B4C284AD15042B429EB9D904544AD0225507041@NY1-EXMB-01.ip-soft.net> <895731f6d2c9f7220f752e977f485386@mail.gmail.com> <6A6B4C284AD15042B429EB9D904544AD02255071D4@NY1-EXMB-01.ip-soft.net> <6A6B4C284AD15042B429EB9D904544AD02255071E6@NY1-EXMB-01.ip-soft.net> Message-ID: <6A6B4C284AD15042B429EB9D904544AD02255071F0@NY1-EXMB-01.ip-soft.net> Done. http://jira.freeswitch.org/browse/ESL-61 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, December 21, 2011 5:36 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Threads remain after calling close on Java client you can just file it under esl or general, the patch should be ok as long as its a unified diff or you can get the FS tree and patch your local copy and submit a git diff from that if all else fails. On Wed, Dec 21, 2011 at 4:33 PM, Hector Geraldino > wrote: Sure, I can post it to Jira. Can you please help me to find an appropriate category to create the bug report? There is no project category for freeswitch-contrib projects. Also I'm not sure the patch format is compatible with git (I personally use SVN, don't know if the diff format is different between these two). From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Wednesday, December 21, 2011 4:14 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Threads remain after calling close on Java client if this patch is necessary can you post it to a jira? On Wed, Dec 21, 2011 at 1:26 PM, Hector Geraldino > wrote: Hi Neil, This doesn't seem to be the same concurrency issue I had, but I'm attaching the patch that fixes my issue anyway. Feel free to test it and send me back the restuls. In case it doesn't work you might try to "manually" close the channel by modifying the close() method on the org.freeswitch.esl.client.inbound.Client. Try to do a channel.disconnect(); and channel=null; and see what happens. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Neil Davis Sent: Wednesday, December 21, 2011 11:17 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Threads remain after calling close on Java client Here is a thread dump of my Tomcat process at the point when it is hanging on shutdown. There are a number of threads in a "waiting on condition" state that appear to have to do with the netty package on which the Freeswitch client is dependent. 2011-12-21 09:05:17 Full thread dump Java HotSpot(TM) Client VM (14.3-b01 mixed mode): "DestroyJavaVM" prio=6 tid=0x546f7400 nid=0x640 waiting on condition [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "pool-5-thread-16" prio=6 tid=0x544b5800 nid=0x1098 waiting on condition [0x55e4f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-15" prio=6 tid=0x54f7f400 nid=0x10f0 waiting on condition [0x54e5f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-14" prio=6 tid=0x544b3c00 nid=0xdb4 waiting on condition [0x5670f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-13" prio=6 tid=0x544b4000 nid=0x12c8 waiting on condition [0x566bf000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-12" prio=6 tid=0x544b4800 nid=0xe18 waiting on condition [0x5666f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-11" prio=6 tid=0x544b3000 nid=0x620 waiting on condition [0x5661f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-10" prio=6 tid=0x552a1800 nid=0x131c waiting on condition [0x565cf000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-9" prio=6 tid=0x552a1400 nid=0x1710 waiting on condition [0x5657f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-8" prio=6 tid=0x552a0c00 nid=0x1094 waiting on condition [0x5652f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-7" prio=6 tid=0x552a0800 nid=0x1040 waiting on condition [0x564df000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-6" prio=6 tid=0x552a0000 nid=0x179c waiting on condition [0x5648f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-5" prio=6 tid=0x5529fc00 nid=0x3e4 waiting on condition [0x5643f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-4" prio=6 tid=0x5529f400 nid=0x63c waiting on condition [0x563ef000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-3" prio=6 tid=0x5529f000 nid=0x17e8 waiting on condition [0x5639f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-2" prio=6 tid=0x5529e800 nid=0x574 waiting on condition [0x5634f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-4-thread-1" prio=6 tid=0x5529e400 nid=0x10cc waiting on condition [0x562ff000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x09970378> (a java.util.concurrent.SynchronousQueue$TransferStack) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) at java.util.concurrent.SynchronousQueue.poll(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-3-thread-1" prio=6 tid=0x546f8800 nid=0x654 waiting on condition [0x562af000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x09970580> (a java.util.concurrent.SynchronousQueue$TransferStack) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) at java.util.concurrent.SynchronousQueue.poll(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-1" prio=6 tid=0x546f8000 nid=0x594 waiting on condition [0x5625f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI TCP Connection(2)-10.0.0.22" daemon prio=6 tid=0x551db800 nid=0xf6c runnable [0x5593f000] java.lang.Thread.State: RUNNABLE at java.net.SocketInputStream.socketRead0(Native Method) at java.net.SocketInputStream.read(Unknown Source) at java.io.BufferedInputStream.fill(Unknown Source) at java.io.BufferedInputStream.read(Unknown Source) - locked <0x08f5c2d0> (a java.io.BufferedInputStream) at java.io.FilterInputStream.read(Unknown Source) at sun.rmi.transport.tcp.TCPTransport.handleMessages(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$ConnectionHandler.run0(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$ConnectionHandler.run(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.runTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - <0x08f62b10> (a java.util.concurrent.locks.ReentrantLock$NonfairSync) "JMX server connection timeout 22" daemon prio=6 tid=0x5474fc00 nid=0x424 in Object.wait() [0x558ef000] java.lang.Thread.State: TIMED_WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x08e9bdb0> (a [I) at com.sun.jmx.remote.internal.ServerCommunicatorAdmin$Timeout.run(Unknown Source) - locked <0x08e9bdb0> (a [I) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI Scheduler(0)" daemon prio=6 tid=0x5511d400 nid=0x1420 waiting on condition [0x5589f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x08e9bdd0> (a java.util.concurrent.locks.AbstractQueuedSynchronizer$ConditionObject) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.locks.AbstractQueuedSynchronizer$ConditionObject.awaitNanos(Unknown Source) at java.util.concurrent.DelayQueue.take(Unknown Source) at java.util.concurrent.ScheduledThreadPoolExecutor$DelayedWorkQueue.take(Unknown Source) at java.util.concurrent.ScheduledThreadPoolExecutor$DelayedWorkQueue.take(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI TCP Connection(idle)" daemon prio=6 tid=0x55176800 nid=0x884 waiting on condition [0x557ff000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x08ebd088> (a java.util.concurrent.SynchronousQueue$TransferStack) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) at java.util.concurrent.SynchronousQueue.poll(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI TCP Accept-0" daemon prio=6 tid=0x5518bc00 nid=0x163c runnable [0x557af000] java.lang.Thread.State: RUNNABLE at java.net.PlainSocketImpl.socketAccept(Native Method) at java.net.PlainSocketImpl.accept(Unknown Source) - locked <0x08e9e158> (a java.net.SocksSocketImpl) at java.net.ServerSocket.implAccept(Unknown Source) at java.net.ServerSocket.accept(Unknown Source) at sun.management.jmxremote.LocalRMIServerSocketFactory$1.accept(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$AcceptLoop.executeAcceptLoop(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$AcceptLoop.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "GC Daemon" daemon prio=2 tid=0x5464b000 nid=0x1718 in Object.wait() [0x5497f000] java.lang.Thread.State: TIMED_WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x089b1270> (a sun.misc.GC$LatencyLock) at sun.misc.GC$Daemon.run(Unknown Source) - locked <0x089b1270> (a sun.misc.GC$LatencyLock) Locked ownable synchronizers: - None "Low Memory Detector" daemon prio=6 tid=0x01a12c00 nid=0x14fc runnable [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "CompilerThread0" daemon prio=10 tid=0x01a0f800 nid=0x1ec waiting on condition [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "JDWP Event Helper Thread" daemon prio=6 tid=0x01a01400 nid=0x173c runnable [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "Attach Listener" daemon prio=10 tid=0x019f5000 nid=0x13a4 waiting on condition [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "Signal Dispatcher" daemon prio=10 tid=0x019ea000 nid=0x17b8 runnable [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "Finalizer" daemon prio=8 tid=0x019bb400 nid=0x1720 in Object.wait() [0x53fcf000] java.lang.Thread.State: WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x089b2858> (a java.lang.ref.ReferenceQueue$Lock) at java.lang.ref.ReferenceQueue.remove(Unknown Source) - locked <0x089b2858> (a java.lang.ref.ReferenceQueue$Lock) at java.lang.ref.ReferenceQueue.remove(Unknown Source) at java.lang.ref.Finalizer$FinalizerThread.run(Unknown Source) Locked ownable synchronizers: - None "Reference Handler" daemon prio=10 tid=0x019ba000 nid=0x53c in Object.wait() [0x53f7f000] java.lang.Thread.State: WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x089b2878> (a java.lang.ref.Reference$Lock) at java.lang.Object.wait(Object.java:485) at java.lang.ref.Reference$ReferenceHandler.run(Unknown Source) - locked <0x089b2878> (a java.lang.ref.Reference$Lock) Locked ownable synchronizers: - None "VM Thread" prio=10 tid=0x019b7400 nid=0xab8 runnable "VM Periodic Task Thread" prio=10 tid=0x01a1bc00 nid=0xc7c waiting on condition JNI global references: 21240 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hector Geraldino Sent: Monday, December 19, 2011 8:29 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Threads remain after calling close on Java client Hi Neil, Can you get a thread dump of the tomcat process to try to figure out what this problem is about? Or at least, try to connect the jconsole to the tomcat process and get the StackTrace of one of these threads to have a better idea of what is going on. IIRC I've fixed a couple of bugs for this library, but the patches haven't been tested by the main developer (dvarnes) nor integrated on the repository (freeswitch-contrib). If this problem can be fixed with my patched code, I would be happy to share it with you. Good luck! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Neil Davis Sent: Friday, December 16, 2011 7:09 PM To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] Threads remain after calling close on Java client Hi, I built a web application that connects to Freeswitch using the org.freeswitch.esl.client.Client. I connect the Client object from a Spring annotated service that I call from a Spring controller. I put the connected client in my ServletContext, so I can access it later to call client.cancelEventSubscriptions() and client.close() from my ServletContextListener contextDestroyed method when Tomcat is shutting down. The problem I'm having is that even after I call close on the client, there are still a bunch of active threads that the client has spawned in the background. These threads are causing Tomcat to hang when I'm shutting down. Can anyone suggest an approach that would enable my application to disconnect the Freeswitch client when Tomcat is shutting down that would allow Tomcat to shutdown gracefully? Below are errors from my Tomcat log for the threads that I have identified as being related to the Freeswitch client. I don't know how I can get to these threads to interrupt them and Client.close() seems to leave them hanging. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-3-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-4-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-2] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-3] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-4] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-5] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-6] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-7] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-8] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-9] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-10] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-11] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-12] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-13] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-14] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-15] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-16] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.internal.ThreadLocalBoolean] (value [org.jboss.netty.util.internal.ThreadLocalBoolean at 186e192]) and a value of type [java.lang.Boolean] (value [false]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.CharsetUtil$1] (value [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type [java.util.IdentityHashMap] (value [{windows-1252=sun.nio.cs.MS1252$Encoder at 373f86}]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.internal.ThreadLocalRandom$1] (value [org.jboss.netty.util.internal.ThreadLocalRandom$1 at 12bb519]) and a value of type [org.jboss.netty.util.internal.ThreadLocalRandom] (value [org.jboss.netty.util.internal.ThreadLocalRandom at 7e9dbc]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.CharsetUtil$1] (value [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type [java.util.IdentityHashMap] (value [{windows-1252=sun.nio.cs.MS1252$Encoder at a5b041}]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Thanks, Neil Davis _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://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 -- 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/20111221/2cfc795b/attachment-0001.html From Hector.Geraldino at ip-soft.net Thu Dec 22 02:49:58 2011 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Wed, 21 Dec 2011 18:49:58 -0500 Subject: [Freeswitch-users] Threads remain after calling close on Java client In-Reply-To: <9ffbcd24295d4daec70f26135a902515@mail.gmail.com> References: <8b41de351c0d1365e3786e7a60645275@mail.gmail.com> <6A6B4C284AD15042B429EB9D904544AD0225507041@NY1-EXMB-01.ip-soft.net> <895731f6d2c9f7220f752e977f485386@mail.gmail.com> <6A6B4C284AD15042B429EB9D904544AD02255071D4@NY1-EXMB-01.ip-soft.net> <9ffbcd24295d4daec70f26135a902515@mail.gmail.com> Message-ID: <6A6B4C284AD15042B429EB9D904544AD02255071F1@NY1-EXMB-01.ip-soft.net> Yeah, I was going to suggest you that, as there's a chance for those threads to come to an end after a few seconds. The last thing I can think about is to create your own variable instances of Executors and pass them to the ClientBootstrap constructor on the connect() method of the Client class: ClientBootstrap bootstrap = new ClientBootstrap( new NioClientSocketChannelFactory( Executors.newCachedThreadPool(), <- replace this with your own instances Executors.newCachedThreadPool() ) ); By doing so, you can later call the ExecutorService.shutdown() method to immediately terminate all running threads. I think it's worth to give it a shot. Good luck! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Neil Davis Sent: Wednesday, December 21, 2011 6:19 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Threads remain after calling close on Java client Hector, Thanks for your suggestions. I tried both your fix and modifying the close method as suggested, but I still have threads hanging around in the TIMED-WAITING status. They eventually terminate after about a minute. For the time being, I'm just stalling the Tomcat shutdown with Thread.sleeps until all the hanging threads have terminated. Tomcat is then able to shutdown gracefully. I'll look a little more at the esl code as I have time and update the mailing list if I find a solution. Thanks, Neil Davis From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hector Geraldino Sent: Wednesday, December 21, 2011 12:26 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Threads remain after calling close on Java client Hi Neil, This doesn't seem to be the same concurrency issue I had, but I'm attaching the patch that fixes my issue anyway. Feel free to test it and send me back the restuls. In case it doesn't work you might try to "manually" close the channel by modifying the close() method on the org.freeswitch.esl.client.inbound.Client. Try to do a channel.disconnect(); and channel=null; and see what happens. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Neil Davis Sent: Wednesday, December 21, 2011 11:17 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Threads remain after calling close on Java client Here is a thread dump of my Tomcat process at the point when it is hanging on shutdown. There are a number of threads in a "waiting on condition" state that appear to have to do with the netty package on which the Freeswitch client is dependent. 2011-12-21 09:05:17 Full thread dump Java HotSpot(TM) Client VM (14.3-b01 mixed mode): "DestroyJavaVM" prio=6 tid=0x546f7400 nid=0x640 waiting on condition [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "pool-5-thread-16" prio=6 tid=0x544b5800 nid=0x1098 waiting on condition [0x55e4f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-15" prio=6 tid=0x54f7f400 nid=0x10f0 waiting on condition [0x54e5f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-14" prio=6 tid=0x544b3c00 nid=0xdb4 waiting on condition [0x5670f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-13" prio=6 tid=0x544b4000 nid=0x12c8 waiting on condition [0x566bf000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-12" prio=6 tid=0x544b4800 nid=0xe18 waiting on condition [0x5666f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-11" prio=6 tid=0x544b3000 nid=0x620 waiting on condition [0x5661f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-10" prio=6 tid=0x552a1800 nid=0x131c waiting on condition [0x565cf000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-9" prio=6 tid=0x552a1400 nid=0x1710 waiting on condition [0x5657f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-8" prio=6 tid=0x552a0c00 nid=0x1094 waiting on condition [0x5652f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-7" prio=6 tid=0x552a0800 nid=0x1040 waiting on condition [0x564df000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-6" prio=6 tid=0x552a0000 nid=0x179c waiting on condition [0x5648f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-5" prio=6 tid=0x5529fc00 nid=0x3e4 waiting on condition [0x5643f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-4" prio=6 tid=0x5529f400 nid=0x63c waiting on condition [0x563ef000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-3" prio=6 tid=0x5529f000 nid=0x17e8 waiting on condition [0x5639f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-2" prio=6 tid=0x5529e800 nid=0x574 waiting on condition [0x5634f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-4-thread-1" prio=6 tid=0x5529e400 nid=0x10cc waiting on condition [0x562ff000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x09970378> (a java.util.concurrent.SynchronousQueue$TransferStack) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) at java.util.concurrent.SynchronousQueue.poll(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-3-thread-1" prio=6 tid=0x546f8800 nid=0x654 waiting on condition [0x562af000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x09970580> (a java.util.concurrent.SynchronousQueue$TransferStack) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) at java.util.concurrent.SynchronousQueue.poll(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "pool-5-thread-1" prio=6 tid=0x546f8000 nid=0x594 waiting on condition [0x5625f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at org.jboss.netty.util.internal.LinkedTransferQueue.awaitMatch(LinkedTransferQueue.java:766) at org.jboss.netty.util.internal.LinkedTransferQueue.xfer(LinkedTransferQueue.java:673) at org.jboss.netty.util.internal.LinkedTransferQueue.poll(LinkedTransferQueue.java:1164) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI TCP Connection(2)-10.0.0.22" daemon prio=6 tid=0x551db800 nid=0xf6c runnable [0x5593f000] java.lang.Thread.State: RUNNABLE at java.net.SocketInputStream.socketRead0(Native Method) at java.net.SocketInputStream.read(Unknown Source) at java.io.BufferedInputStream.fill(Unknown Source) at java.io.BufferedInputStream.read(Unknown Source) - locked <0x08f5c2d0> (a java.io.BufferedInputStream) at java.io.FilterInputStream.read(Unknown Source) at sun.rmi.transport.tcp.TCPTransport.handleMessages(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$ConnectionHandler.run0(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$ConnectionHandler.run(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.runTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - <0x08f62b10> (a java.util.concurrent.locks.ReentrantLock$NonfairSync) "JMX server connection timeout 22" daemon prio=6 tid=0x5474fc00 nid=0x424 in Object.wait() [0x558ef000] java.lang.Thread.State: TIMED_WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x08e9bdb0> (a [I) at com.sun.jmx.remote.internal.ServerCommunicatorAdmin$Timeout.run(Unknown Source) - locked <0x08e9bdb0> (a [I) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI Scheduler(0)" daemon prio=6 tid=0x5511d400 nid=0x1420 waiting on condition [0x5589f000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x08e9bdd0> (a java.util.concurrent.locks.AbstractQueuedSynchronizer$ConditionObject) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.locks.AbstractQueuedSynchronizer$ConditionObject.awaitNanos(Unknown Source) at java.util.concurrent.DelayQueue.take(Unknown Source) at java.util.concurrent.ScheduledThreadPoolExecutor$DelayedWorkQueue.take(Unknown Source) at java.util.concurrent.ScheduledThreadPoolExecutor$DelayedWorkQueue.take(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI TCP Connection(idle)" daemon prio=6 tid=0x55176800 nid=0x884 waiting on condition [0x557ff000] java.lang.Thread.State: TIMED_WAITING (parking) at sun.misc.Unsafe.park(Native Method) - parking to wait for <0x08ebd088> (a java.util.concurrent.SynchronousQueue$TransferStack) at java.util.concurrent.locks.LockSupport.parkNanos(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.awaitFulfill(Unknown Source) at java.util.concurrent.SynchronousQueue$TransferStack.transfer(Unknown Source) at java.util.concurrent.SynchronousQueue.poll(Unknown Source) at java.util.concurrent.ThreadPoolExecutor.getTask(Unknown Source) at java.util.concurrent.ThreadPoolExecutor$Worker.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "RMI TCP Accept-0" daemon prio=6 tid=0x5518bc00 nid=0x163c runnable [0x557af000] java.lang.Thread.State: RUNNABLE at java.net.PlainSocketImpl.socketAccept(Native Method) at java.net.PlainSocketImpl.accept(Unknown Source) - locked <0x08e9e158> (a java.net.SocksSocketImpl) at java.net.ServerSocket.implAccept(Unknown Source) at java.net.ServerSocket.accept(Unknown Source) at sun.management.jmxremote.LocalRMIServerSocketFactory$1.accept(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$AcceptLoop.executeAcceptLoop(Unknown Source) at sun.rmi.transport.tcp.TCPTransport$AcceptLoop.run(Unknown Source) at java.lang.Thread.run(Unknown Source) Locked ownable synchronizers: - None "GC Daemon" daemon prio=2 tid=0x5464b000 nid=0x1718 in Object.wait() [0x5497f000] java.lang.Thread.State: TIMED_WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x089b1270> (a sun.misc.GC$LatencyLock) at sun.misc.GC$Daemon.run(Unknown Source) - locked <0x089b1270> (a sun.misc.GC$LatencyLock) Locked ownable synchronizers: - None "Low Memory Detector" daemon prio=6 tid=0x01a12c00 nid=0x14fc runnable [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "CompilerThread0" daemon prio=10 tid=0x01a0f800 nid=0x1ec waiting on condition [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "JDWP Event Helper Thread" daemon prio=6 tid=0x01a01400 nid=0x173c runnable [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "Attach Listener" daemon prio=10 tid=0x019f5000 nid=0x13a4 waiting on condition [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "Signal Dispatcher" daemon prio=10 tid=0x019ea000 nid=0x17b8 runnable [0x00000000] java.lang.Thread.State: RUNNABLE Locked ownable synchronizers: - None "Finalizer" daemon prio=8 tid=0x019bb400 nid=0x1720 in Object.wait() [0x53fcf000] java.lang.Thread.State: WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x089b2858> (a java.lang.ref.ReferenceQueue$Lock) at java.lang.ref.ReferenceQueue.remove(Unknown Source) - locked <0x089b2858> (a java.lang.ref.ReferenceQueue$Lock) at java.lang.ref.ReferenceQueue.remove(Unknown Source) at java.lang.ref.Finalizer$FinalizerThread.run(Unknown Source) Locked ownable synchronizers: - None "Reference Handler" daemon prio=10 tid=0x019ba000 nid=0x53c in Object.wait() [0x53f7f000] java.lang.Thread.State: WAITING (on object monitor) at java.lang.Object.wait(Native Method) - waiting on <0x089b2878> (a java.lang.ref.Reference$Lock) at java.lang.Object.wait(Object.java:485) at java.lang.ref.Reference$ReferenceHandler.run(Unknown Source) - locked <0x089b2878> (a java.lang.ref.Reference$Lock) Locked ownable synchronizers: - None "VM Thread" prio=10 tid=0x019b7400 nid=0xab8 runnable "VM Periodic Task Thread" prio=10 tid=0x01a1bc00 nid=0xc7c waiting on condition JNI global references: 21240 From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Hector Geraldino Sent: Monday, December 19, 2011 8:29 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] Threads remain after calling close on Java client Hi Neil, Can you get a thread dump of the tomcat process to try to figure out what this problem is about? Or at least, try to connect the jconsole to the tomcat process and get the StackTrace of one of these threads to have a better idea of what is going on. IIRC I've fixed a couple of bugs for this library, but the patches haven't been tested by the main developer (dvarnes) nor integrated on the repository (freeswitch-contrib). If this problem can be fixed with my patched code, I would be happy to share it with you. Good luck! From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Neil Davis Sent: Friday, December 16, 2011 7:09 PM To: FreeSWITCH-users at lists.freeswitch.org Subject: [Freeswitch-users] Threads remain after calling close on Java client Hi, I built a web application that connects to Freeswitch using the org.freeswitch.esl.client.Client. I connect the Client object from a Spring annotated service that I call from a Spring controller. I put the connected client in my ServletContext, so I can access it later to call client.cancelEventSubscriptions() and client.close() from my ServletContextListener contextDestroyed method when Tomcat is shutting down. The problem I'm having is that even after I call close on the client, there are still a bunch of active threads that the client has spawned in the background. These threads are causing Tomcat to hang when I'm shutting down. Can anyone suggest an approach that would enable my application to disconnect the Freeswitch client when Tomcat is shutting down that would allow Tomcat to shutdown gracefully? Below are errors from my Tomcat log for the threads that I have identified as being related to the Freeswitch client. I don't know how I can get to these threads to interrupt them and Client.close() seems to leave them hanging. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-3-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-4-thread-1] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-2] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-3] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-4] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-5] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-6] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-7] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-8] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-9] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-10] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-11] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-12] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-13] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-14] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-15] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader clearReferencesThreads SEVERE: The web application [/socketspy] appears to have started a thread named [pool-5-thread-16] but has failed to stop it. This is very likely to create a memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.internal.ThreadLocalBoolean] (value [org.jboss.netty.util.internal.ThreadLocalBoolean at 186e192]) and a value of type [java.lang.Boolean] (value [false]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.CharsetUtil$1] (value [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type [java.util.IdentityHashMap] (value [{windows-1252=sun.nio.cs.MS1252$Encoder at 373f86}]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.internal.ThreadLocalRandom$1] (value [org.jboss.netty.util.internal.ThreadLocalRandom$1 at 12bb519]) and a value of type [org.jboss.netty.util.internal.ThreadLocalRandom] (value [org.jboss.netty.util.internal.ThreadLocalRandom at 7e9dbc]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Dec 16, 2011 12:04:59 PM org.apache.catalina.loader.WebappClassLoader checkThreadLocalMapForLeaks SEVERE: The web application [/socketspy] created a ThreadLocal with key of type [org.jboss.netty.util.CharsetUtil$1] (value [org.jboss.netty.util.CharsetUtil$1 at 14d8e1]) and a value of type [java.util.IdentityHashMap] (value [{windows-1252=sun.nio.cs.MS1252$Encoder at a5b041}]) but failed to remove it when the web application was stopped. Threads are going to be renewed over time to try and avoid a probable memory leak. Thanks, Neil Davis -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/f194225c/attachment-0001.html From jeff at jefflenk.com Thu Dec 22 04:34:44 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Wed, 21 Dec 2011 17:34:44 -0800 (PST) Subject: [Freeswitch-users] Feedback In-Reply-To: <004501cc98a8$698763b0$3c962b10$@google.hm> References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM> <004501cc98a8$698763b0$3c962b10$@google.hm> Message-ID: <1324517684859-7117251.post@n2.nabble.com> Chad, Any update on this? Looking forward to seeing some of your contributions and perhaps being able to merge them into the mainline code. Sincerely, Jeff -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Feedback-tp6941381p7117251.html Sent from the freeswitch-users mailing list archive at Nabble.com. From gburca-freeswitch at ebixio.com Wed Dec 21 23:06:04 2011 From: gburca-freeswitch at ebixio.com (Gabriel Burca) Date: Wed, 21 Dec 2011 14:06:04 -0600 Subject: [Freeswitch-users] Forwarding MWI from gateway to internal extensions Message-ID: <4EF23C2C.3070300@ebixio.com> I just started using FreeSWITCH yesterday. Everything works perfectly, except I can't figure how to forward a SIP message waiting indication from a gateway to an internal extension. This is a simple home use-case. I have FS connected externally to a VoIP provider. Incoming calls (from the provider) are routed to a group extension which rings all phones. VM is handled by the VoIP provider. I see the SIP NOTIFY message come in to FS, but I don't know how to pass that notification message on to extension 1000 for example, or better yet, to all the extensions that are part of a group. I've added: to my external gateway config file, but it doesn't do the trick, and even if it did, I don't see how it would know where to forward the MWI to. Don't know if this is related, but I also see the following in the logs: [DEBUG] sofia.c:445 Gateway information missing Subscription Event: message-summary I've gone through most of the relevant documentation, and searched on-line, but came up empty handed. Is this something that can be accomplished through configuration files (and if so, how?) or does this require scripting? -- Gabriel Burca There's nothing remarkable about it. All one has to do is hit the right keys at the right time and the instrument plays itself. -- J. S. Bach From siobhan at pluggedin-tech.com Wed Dec 21 23:12:06 2011 From: siobhan at pluggedin-tech.com (Siobhan Hamilton) Date: Wed, 21 Dec 2011 15:12:06 -0500 Subject: [Freeswitch-users] Freeswitch help - calling between internal lines failing with NO_ROUTE_DESTINATION error Message-ID: <59634B10-4A88-49F7-A01B-A1DF8F8433A6@pluggedin-tech.com> Hi there, I have a Freeswitch install I'm resurrecting from a while back that once worked, but seems to not be anymore. What I wanted to do, to start, is make sure that the machine was working properly. I fired it up and registered two clients on softphone apps, extensions 1002 and 1003; when I tried to get them to call one another, I got back a 404 error and a "No route, aborting"/NO_ROUTE_DESTINATION message in the logs. Does anyone have any clues? I am pretty much a complete newbie at troubleshooting these sorts of things. Thanks in advance for any help... My debug log is at the bottom, but first here is what I see when I check sofia status profile external; note that the clients appear and are registered (see below): ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP xx.xxx.142.187 SIP-IP xx.xxx.142.187 URL sip:mod_sofia at xx.xxx.142.187:5060 BIND-URL sip:mod_sofia at xx.xxx.142.187:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN iLBC at 30i,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM CODECS OUT iLBC at 30i,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 19 FAILED-CALLS-IN 19 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: NDc0YWE1YTU1YjJhOTEwMGNmMGMxYjY2N2ZlYzMyMWY. User: 1003 at xx.xxx.142.187 Contact: "1003" Agent: Zoiper rev.11619 Status: Registered(UDP)(unknown) EXP(2011-12-21 15:50:06) EXPSECS(3038) Host: fswitch IP: 72.43.209.146 Port: 5060 Auth-User: 1003 Auth-Realm: xx.xxx.142.187 MWI-Account: 1003 at xx.xxx.142.187 Call-ID: PfxBVFYxNIg.pWaY.CpEFO1EhEEqpMHJ User: 1002 at xx.xxx.142.187 Contact: "user" Agent: CSipSimple r1108 / vivow-10 Status: Registered(UDP)(unknown) EXP(2011-12-21 15:08:44) EXPSECS(556) Host: fswitch IP: 72.43.209.146 Port: 44803 Auth-User: 1002 Auth-Realm: xx.xxx.142.187 MWI-Account: 1002 at xx.xxx.142.187 Total items returned: 2 ================================================================================================= Log (In debug mode) 2011-12-21 14:53:18.132608 [NOTICE] switch_channel.c:812 New Channel sofia/external/1003 at xx.xxx.142.187 [96716f86-500e-445c-8196-59e754c023c0] 2011-12-21 14:53:18.132608 [DEBUG] sofia.c:4659 Channel sofia/external/1003 at xx.xxx.142.187 entering state [received][100] 2011-12-21 14:53:18.132608 [DEBUG] sofia.c:4670 Remote SDP: v=0 o=Z 0 0 IN IP4 192.168.1.108 s=Z c=IN IP4 192.168.1.108 t=0 0 m=audio 8000 RTP/AVP 3 110 98 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[iLBC:97:8000:30:13330] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:115:32000:20:48000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:107:16000:20:32000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[GSM:3:8000:20:13200] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:2757 Set Codec sofia/external/1003 at xx.xxx.142.187 GSM/8000 20 ms 160 samples 13200 bits 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4616 Set 2833 dtmf send/recv payload to 101 2011-12-21 14:53:18.132608 [DEBUG] sofia.c:4837 (sofia/external/1003 at xx.xxx.142.187) State Change CS_NEW -> CS_INIT 2011-12-21 14:53:18.132608 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003 at xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.132608 [DEBUG] switch_core_state_machine.c:320 (sofia/external/1003 at xx.xxx.142.187) Running State Change CS_INIT 2011-12-21 14:53:18.132608 [DEBUG] switch_core_state_machine.c:356 (sofia/external/1003 at xx.xxx.142.187) State INIT 2011-12-21 14:53:18.132608 [DEBUG] mod_sofia.c:84 sofia/external/1003 at xx.xxx.142.187 SOFIA INIT 2011-12-21 14:53:18.132608 [DEBUG] mod_sofia.c:124 (sofia/external/1003 at xx.xxx.142.187) State Change CS_INIT -> CS_ROUTING 2011-12-21 14:53:18.132608 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003 at xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.132608 [DEBUG] switch_core_state_machine.c:356 (sofia/external/1003 at xx.xxx.142.187) State INIT going to sleep 2011-12-21 14:53:18.132608 [DEBUG] switch_core_state_machine.c:320 (sofia/external/1003 at xx.xxx.142.187) Running State Change CS_ROUTING 2011-12-21 14:53:18.134757 [DEBUG] switch_channel.c:1662 (sofia/external/1003 at xx.xxx.142.187) Callstate Change DOWN -> RINGING 2011-12-21 14:53:18.134757 [DEBUG] switch_core_state_machine.c:359 (sofia/external/1003 at xx.xxx.142.187) State ROUTING 2011-12-21 14:53:18.134757 [DEBUG] mod_sofia.c:147 sofia/external/1003 at xx.xxx.142.187 SOFIA ROUTING 2011-12-21 14:53:18.134757 [DEBUG] switch_core_state_machine.c:77 sofia/external/1003 at xx.xxx.142.187 Standard ROUTING 2011-12-21 14:53:18.134757 [INFO] mod_dialplan_xml.c:331 Processing 1003 <1003>->1002 at xx.xxx.142.187 in context public Dialplan: sofia/external/1003 at xx.xxx.142.187 parsing [public->unloop] continue=false Dialplan: sofia/external/1003 at xx.xxx.142.187 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/1003 at xx.xxx.142.187 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/1003 at xx.xxx.142.187 parsing [public->public_extensions] continue=false Dialplan: sofia/external/1003 at xx.xxx.142.187 Regex (FAIL) [public_extensions] destination_number(1002 at xx.xxx.142.187) =~ /^conf_(.*)$/ break=on-false Dialplan: sofia/external/1003 at xx.xxx.142.187 parsing [public->public_did] continue=false Dialplan: sofia/external/1003 at xx.xxx.142.187 Regex (FAIL) [public_did] destination_number(1002 at xx.xxx.142.187) =~ /^(5551212)$/ break=on-false 2011-12-21 14:53:18.134757 [INFO] switch_core_state_machine.c:142 No Route, Aborting 2011-12-21 14:53:18.134757 [DEBUG] switch_channel.c:2540 (sofia/external/1003 at xx.xxx.142.187) Callstate Change RINGING -> HANGUP 2011-12-21 14:53:18.134757 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/external/1003 at xx.xxx.142.187 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2011-12-21 14:53:18.134757 [DEBUG] switch_channel.c:2556 Send signal sofia/external/1003 at xx.xxx.142.187 [KILL] 2011-12-21 14:53:18.134757 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003 at xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.134757 [DEBUG] switch_core_state_machine.c:359 (sofia/external/1003 at xx.xxx.142.187) State ROUTING going to sleep 2011-12-21 14:53:18.134757 [DEBUG] switch_core_state_machine.c:320 (sofia/external/1003 at xx.xxx.142.187) Running State Change CS_HANGUP 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:557 (sofia/external/1003 at xx.xxx.142.187) State HANGUP 2011-12-21 14:53:18.136791 [DEBUG] mod_sofia.c:457 Channel sofia/external/1003 at xx.xxx.142.187 hanging up, cause: NO_ROUTE_DESTINATION 2011-12-21 14:53:18.136791 [DEBUG] mod_sofia.c:519 Responding to INVITE with: 404 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:46 sofia/external/1003 at xx.xxx.142.187 Standard HANGUP, cause: NO_ROUTE_DESTINATION 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:557 (sofia/external/1003 at xx.xxx.142.187) State HANGUP going to sleep 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:351 (sofia/external/1003 at xx.xxx.142.187) State Change CS_HANGUP -> CS_REPORTING 2011-12-21 14:53:18.136791 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003 at xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:320 (sofia/external/1003 at xx.xxx.142.187) Running State Change CS_REPORTING 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:617 (sofia/external/1003 at xx.xxx.142.187) State REPORTING 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:53 sofia/external/1003 at xx.xxx.142.187 Standard REPORTING, cause: NO_ROUTE_DESTINATION 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:617 (sofia/external/1003 at xx.xxx.142.187) State REPORTING going to sleep 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:345 (sofia/external/1003 at xx.xxx.142.187) State Change CS_REPORTING -> CS_DESTROY 2011-12-21 14:53:18.136791 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003 at xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.136791 [DEBUG] switch_core_session.c:1288 Session 18 (sofia/external/1003 at xx.xxx.142.187) Locked, Waiting on external entities 2011-12-21 14:53:18.136791 [NOTICE] switch_core_session.c:1306 Session 18 (sofia/external/1003 at xx.xxx.142.187) Ended 2011-12-21 14:53:18.136791 [NOTICE] switch_core_session.c:1308 Close Channel sofia/external/1003 at xx.xxx.142.187 [CS_DESTROY] 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:449 (sofia/external/1003 at xx.xxx.142.187) Callstate Change HANGUP -> DOWN 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:452 (sofia/external/1003 at xx.xxx.142.187) Running State Change CS_DESTROY 2011-12-21 14:53:18.138730 [DEBUG] switch_core_state_machine.c:462 (sofia/external/1003 at xx.xxx.142.187) State DESTROY 2011-12-21 14:53:18.138730 [DEBUG] mod_sofia.c:362 sofia/external/1003 at xx.xxx.142.187 SOFIA DESTROY 2011-12-21 14:53:18.138730 [DEBUG] switch_core_state_machine.c:60 sofia/external/1003 at xx.xxx.142.187 Standard DESTROY 2011-12-21 14:53:18.138730 [DEBUG] switch_core_state_machine.c:462 (sofia/external/1003 at xx.xxx.142.187) State DESTROY going to sleep -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111221/609759e3/attachment-0001.html From georg at riseup.net Thu Dec 22 01:45:25 2011 From: georg at riseup.net (georg at riseup.net) Date: Wed, 21 Dec 2011 23:45:25 +0100 Subject: [Freeswitch-users] Some questions regarding IPs and gateway Message-ID: <9a4d8ee96b56592ac4250c9805cfb1a7.squirrel@fulvetta.riseup.net> Hello all, I'm a asterisk user, and would like to try FS now. I studied already the documentation. However, I still got some problems and questions, any help would be greatly appreciated. 1.) I've got a server with a IP reachable from the open, evil internet. I would like to disable this IP for FreeSWITCH, so that it's not possible to contact FreeSWITCH from there. Do I have to change the routing table of the operating system for this, or is it possible to exclude this ip in FS? 2.) I've got SIP trunk trough a german provider without registration. Authentication is based on IPs in a MPLS-net. So I set my IP in this net as the "external ip" in vars.xml After this, I setup the SIP trunk as a gateway sip-profiles/external.xml using the IP of the voice-switch of my provider as the proxy value and setting register to false. I think I'm still missing something, because sofia_gateway_data on the CLI is telling me "-ERR Parameter missing". I searched for this problem already, but found no solution. 3.) I would like to disable NAT completely, because I think I don't need this (and never used this in Asterisk). Actually there is no NAT involved in my setup. I achieved this using -nonat at startup. But I'm wondering, if there isn't a parameter to disable it by default. 4.) I've got another private (192.168.x.x) net connected to my server. All my phones are in there. So it set the internal ip in sip-profiles/internal to this. Is this correct? Thanks and Greetings from Berlin, Georg From georg at riseup.net Thu Dec 22 03:00:32 2011 From: georg at riseup.net (georg at riseup.net) Date: Thu, 22 Dec 2011 01:00:32 +0100 Subject: [Freeswitch-users] Some questions regarding IPs and gateway In-Reply-To: <9a4d8ee96b56592ac4250c9805cfb1a7.squirrel@fulvetta.riseup.net> References: <9a4d8ee96b56592ac4250c9805cfb1a7.squirrel@fulvetta.riseup.net> Message-ID: <75b9703f741bd0540f72c14d45725780.squirrel@fulvetta.riseup.net> Hi all, I solved 2., 3. and 4. by myself, just 1. is remaining... :) Georg From georg at riseup.net Thu Dec 22 03:02:27 2011 From: georg at riseup.net (georg at riseup.net) Date: Thu, 22 Dec 2011 01:02:27 +0100 Subject: [Freeswitch-users] Some questions regarding IPs and gateway In-Reply-To: <9a4d8ee96b56592ac4250c9805cfb1a7.squirrel@fulvetta.riseup.net> References: <9a4d8ee96b56592ac4250c9805cfb1a7.squirrel@fulvetta.riseup.net> Message-ID: <9d0848ccace3e1d34699e53d24f49e37.squirrel@fulvetta.riseup.net> Its already late...I solved 2. and 4. by myself, 1. and 2. remain open... Cheers, Georg From engineerzuhairraza at gmail.com Thu Dec 22 10:00:20 2011 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Thu, 22 Dec 2011 11:00:20 +0400 Subject: [Freeswitch-users] how is codec selected In-Reply-To: References: <1324502719.74670.YahooMailClassic@web110810.mail.gq1.yahoo.com> Message-ID: Hi, You can set codec for a phone from dialplan using absolute_codec_string parameter as stated in codec negotiation link. Trans-coding issues in case of G729 or G723 can be handled as: http://wiki.freeswitch.org/wiki/Sofia#sip_renegotiate_codec_on_reinvite See http://wiki.freeswitch.org/wiki/Channel_Variables#Codec_Related for codec related variables. Also, there is a configuration in internal and external profiles to define a list of codec in each profile which can be set in global configuration http://wiki.freeswitch.org/wiki/Config_external.xml you can see SIP dialogues on each phone IP to verify which codec are being offered ngrep -q -Wbyline '' port 5060 and host x.x.x.x Regards, Zohair Raza On Thu, Dec 22, 2011 at 2:54 AM, Nandy Dagondon wrote: > it's found here: > http://wiki.freeswitch.org/wiki/Codec_negotiation > > hope this answers your question. > > On Thu, Dec 22, 2011 at 5:25 AM, Sherif Omran wrote: > >> hello , >> >> does anybody know how to set up the codec selection negotiation list in >> freeswitch? How can you know which codec has been selected for such a call >> and which codecs are offered from telephone A and B. >> >> I am using log 7 >> >> >> thank you in advance >> >> k.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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/d9369fbe/attachment.html From peter.olsson at visionutveckling.se Thu Dec 22 10:20:50 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 22 Dec 2011 08:20:50 +0100 Subject: [Freeswitch-users] Freeswitch help - calling between internal lines failing with NO_ROUTE_DESTINATION error Message-ID: <48523B12-A5DF-4903-A404-BA832AC3D16C@visionutveckling.se> Any special reason you use the external profile for registrations? You will need to update the dialplan for the public context, so it knows how to route these calls, check the default dialplan for how It's handled there. /Peter ----- Reply message ----- Fr?n: "Siobhan Hamilton" Datum: tors, dec 22, 2011 07:44 Rubrik: [Freeswitch-users] Freeswitch help - calling between internal lines failing with NO_ROUTE_DESTINATION error Till: "freeswitch-users at lists.freeswitch.org" Hi there, I have a Freeswitch install I'm resurrecting from a while back that once worked, but seems to not be anymore. What I wanted to do, to start, is make sure that the machine was working properly. I fired it up and registered two clients on softphone apps, extensions 1002 and 1003; when I tried to get them to call one another, I got back a 404 error and a "No route, aborting"/NO_ROUTE_DESTINATION message in the logs. Does anyone have any clues? I am pretty much a complete newbie at troubleshooting these sorts of things. Thanks in advance for any help... My debug log is at the bottom, but first here is what I see when I check sofia status profile external; note that the clients appear and are registered (see below): ================================================================================================= Name external Domain Name N/A Auto-NAT false DBName sofia_reg_external Pres Hosts Dialplan XML Context public Challenge Realm auto_to RTP-IP xx.xxx.142.187 SIP-IP xx.xxx.142.187 URL sip:mod_sofia at xx.xxx.142.187:5060 BIND-URL sip:mod_sofia at xx.xxx.142.187:5060 HOLD-MUSIC local_stream://moh OUTBOUND-PROXY N/A CODECS IN iLBC at 30i,G7221 at 32000h,G7221 at 16000h,G722,PCMU,PCMA,GSM CODECS OUT iLBC at 30i,PCMU,PCMA,GSM TEL-EVENT 101 DTMF-MODE rfc2833 CNG 13 SESSION-TO 0 MAX-DIALOG 0 NOMEDIA false LATE-NEG false PROXY-MEDIA false AGGRESSIVENAT false STUN-ENABLED true STUN-AUTO-DISABLE false CALLS-IN 19 FAILED-CALLS-IN 19 CALLS-OUT 0 FAILED-CALLS-OUT 0 Registrations: ================================================================================================= Call-ID: NDc0YWE1YTU1YjJhOTEwMGNmMGMxYjY2N2ZlYzMyMWY. User: 1003 at xx.xxx.142.187 Contact: "1003" Agent: Zoiper rev.11619 Status: Registered(UDP)(unknown) EXP(2011-12-21 15:50:06) EXPSECS(3038) Host: fswitch IP: 72.43.209.146 Port: 5060 Auth-User: 1003 Auth-Realm: xx.xxx.142.187 MWI-Account: 1003 at xx.xxx.142.187 Call-ID: PfxBVFYxNIg.pWaY.CpEFO1EhEEqpMHJ User: 1002 at xx.xxx.142.187 Contact: "user" Agent: CSipSimple r1108 / vivow-10 Status: Registered(UDP)(unknown) EXP(2011-12-21 15:08:44) EXPSECS(556) Host: fswitch IP: 72.43.209.146 Port: 44803 Auth-User: 1002 Auth-Realm: xx.xxx.142.187 MWI-Account: 1002 at xx.xxx.142.187 Total items returned: 2 ================================================================================================= Log (In debug mode) 2011-12-21 14:53:18.132608 [NOTICE] switch_channel.c:812 New Channel sofia/external/1003 at xx.xxx.142.187 [96716f86-500e-445c-8196-59e754c023c0] 2011-12-21 14:53:18.132608 [DEBUG] sofia.c:4659 Channel sofia/external/1003 at xx.xxx.142.187 entering state [received][100] 2011-12-21 14:53:18.132608 [DEBUG] sofia.c:4670 Remote SDP: v=0 o=Z 0 0 IN IP4 192.168.1.108 s=Z c=IN IP4 192.168.1.108 t=0 0 m=audio 8000 RTP/AVP 3 110 98 8 0 101 a=rtpmap:3 GSM/8000 a=rtpmap:110 speex/8000 a=rtpmap:98 iLBC/8000 a=fmtp:98 mode=30 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[iLBC:97:8000:30:13330] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:115:32000:20:48000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[G7221:107:16000:20:32000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[G722:9:8000:20:64000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMU:0:8000:20:64000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[PCMA:8:8000:20:64000] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4504 Audio Codec Compare [GSM:3:8000:20:13200]/[GSM:3:8000:20:13200] 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:2757 Set Codec sofia/external/1003 at xx.xxx.142.187 GSM/8000 20 ms 160 samples 13200 bits 2011-12-21 14:53:18.132608 [DEBUG] sofia_glue.c:4616 Set 2833 dtmf send/recv payload to 101 2011-12-21 14:53:18.132608 [DEBUG] sofia.c:4837 (sofia/external/1003 at xx.xxx.142.187) State Change CS_NEW -> CS_INIT 2011-12-21 14:53:18.132608 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003 at xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.132608 [DEBUG] switch_core_state_machine.c:320 (sofia/external/1003 at xx.xxx.142.187) Running State Change CS_INIT 2011-12-21 14:53:18.132608 [DEBUG] switch_core_state_machine.c:356 (sofia/external/1003 at xx.xxx.142.187) State INIT 2011-12-21 14:53:18.132608 [DEBUG] mod_sofia.c:84 sofia/external/1003 at xx.xxx.142.187 SOFIA INIT 2011-12-21 14:53:18.132608 [DEBUG] mod_sofia.c:124 (sofia/external/1003 at xx.xxx.142.187) State Change CS_INIT -> CS_ROUTING 2011-12-21 14:53:18.132608 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003 at xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.132608 [DEBUG] switch_core_state_machine.c:356 (sofia/external/1003 at xx.xxx.142.187) State INIT going to sleep 2011-12-21 14:53:18.132608 [DEBUG] switch_core_state_machine.c:320 (sofia/external/1003 at xx.xxx.142.187) Running State Change CS_ROUTING 2011-12-21 14:53:18.134757 [DEBUG] switch_channel.c:1662 (sofia/external/1003 at xx.xxx.142.187) Callstate Change DOWN -> RINGING 2011-12-21 14:53:18.134757 [DEBUG] switch_core_state_machine.c:359 (sofia/external/1003 at xx.xxx.142.187) State ROUTING 2011-12-21 14:53:18.134757 [DEBUG] mod_sofia.c:147 sofia/external/1003 at xx.xxx.142.187 SOFIA ROUTING 2011-12-21 14:53:18.134757 [DEBUG] switch_core_state_machine.c:77 sofia/external/1003 at xx.xxx.142.187 Standard ROUTING 2011-12-21 14:53:18.134757 [INFO] mod_dialplan_xml.c:331 Processing 1003 <1003>->1002 at xx.xxx.142.187 in context public Dialplan: sofia/external/1003 at xx.xxx.142.187 parsing [public->unloop] continue=false Dialplan: sofia/external/1003 at xx.xxx.142.187 Regex (PASS) [unloop] ${unroll_loops}(true) =~ /^true$/ break=on-false Dialplan: sofia/external/1003 at xx.xxx.142.187 Regex (FAIL) [unloop] ${sip_looped_call}() =~ /^true$/ break=on-false Dialplan: sofia/external/1003 at xx.xxx.142.187 parsing [public->public_extensions] continue=false Dialplan: sofia/external/1003 at xx.xxx.142.187 Regex (FAIL) [public_extensions] destination_number(1002 at xx.xxx.142.187) =~ /^conf_(.*)$/ break=on-false Dialplan: sofia/external/1003 at xx.xxx.142.187 parsing [public->public_did] continue=false Dialplan: sofia/external/1003 at xx.xxx.142.187 Regex (FAIL) [public_did] destination_number(1002 at xx.xxx.142.187) =~ /^(5551212)$/ break=on-false 2011-12-21 14:53:18.134757 [INFO] switch_core_state_machine.c:142 No Route, Aborting 2011-12-21 14:53:18.134757 [DEBUG] switch_channel.c:2540 (sofia/external/1003 at xx.xxx.142.187) Callstate Change RINGING -> HANGUP 2011-12-21 14:53:18.134757 [NOTICE] switch_core_state_machine.c:143 Hangup sofia/external/1003 at xx.xxx.142.187 [CS_ROUTING] [NO_ROUTE_DESTINATION] 2011-12-21 14:53:18.134757 [DEBUG] switch_channel.c:2556 Send signal sofia/external/1003 at xx.xxx.142.187 [KILL] 2011-12-21 14:53:18.134757 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003 at xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.134757 [DEBUG] switch_core_state_machine.c:359 (sofia/external/1003 at xx.xxx.142.187) State ROUTING going to sleep 2011-12-21 14:53:18.134757 [DEBUG] switch_core_state_machine.c:320 (sofia/external/1003 at xx.xxx.142.187) Running State Change CS_HANGUP 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:557 (sofia/external/1003 at xx.xxx.142.187) State HANGUP 2011-12-21 14:53:18.136791 [DEBUG] mod_sofia.c:457 Channel sofia/external/1003 at xx.xxx.142.187 hanging up, cause: NO_ROUTE_DESTINATION 2011-12-21 14:53:18.136791 [DEBUG] mod_sofia.c:519 Responding to INVITE with: 404 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:46 sofia/external/1003 at xx.xxx.142.187 Standard HANGUP, cause: NO_ROUTE_DESTINATION 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:557 (sofia/external/1003 at xx.xxx.142.187) State HANGUP going to sleep 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:351 (sofia/external/1003 at xx.xxx.142.187) State Change CS_HANGUP -> CS_REPORTING 2011-12-21 14:53:18.136791 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003 at xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:320 (sofia/external/1003 at xx.xxx.142.187) Running State Change CS_REPORTING 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:617 (sofia/external/1003 at xx.xxx.142.187) State REPORTING 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:53 sofia/external/1003 at xx.xxx.142.187 Standard REPORTING, cause: NO_ROUTE_DESTINATION 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:617 (sofia/external/1003 at xx.xxx.142.187) State REPORTING going to sleep 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:345 (sofia/external/1003 at xx.xxx.142.187) State Change CS_REPORTING -> CS_DESTROY 2011-12-21 14:53:18.136791 [DEBUG] switch_core_session.c:1116 Send signal sofia/external/1003 at xx.xxx.142.187 [BREAK] 2011-12-21 14:53:18.136791 [DEBUG] switch_core_session.c:1288 Session 18 (sofia/external/1003 at xx.xxx.142.187) Locked, Waiting on external entities 2011-12-21 14:53:18.136791 [NOTICE] switch_core_session.c:1306 Session 18 (sofia/external/1003 at xx.xxx.142.187) Ended 2011-12-21 14:53:18.136791 [NOTICE] switch_core_session.c:1308 Close Channel sofia/external/1003 at xx.xxx.142.187 [CS_DESTROY] 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:449 (sofia/external/1003 at xx.xxx.142.187) Callstate Change HANGUP -> DOWN 2011-12-21 14:53:18.136791 [DEBUG] switch_core_state_machine.c:452 (sofia/external/1003 at xx.xxx.142.187) Running State Change CS_DESTROY 2011-12-21 14:53:18.138730 [DEBUG] switch_core_state_machine.c:462 (sofia/external/1003 at xx.xxx.142.187) State DESTROY 2011-12-21 14:53:18.138730 [DEBUG] mod_sofia.c:362 sofia/external/1003 at xx.xxx.142.187 SOFIA DESTROY 2011-12-21 14:53:18.138730 [DEBUG] switch_core_state_machine.c:60 sofia/external/1003 at xx.xxx.142.187 Standard DESTROY 2011-12-21 14:53:18.138730 [DEBUG] switch_core_state_machine.c:462 (sofia/external/1003 at xx.xxx.142.187) State DESTROY going to sleep !DSPAM:4ef2cf9132769289182587! From covici at ccs.covici.com Thu Dec 22 10:31:35 2011 From: covici at ccs.covici.com (covici at ccs.covici.com) Date: Thu, 22 Dec 2011 02:31:35 -0500 Subject: [Freeswitch-users] Need help with PortAudio In-Reply-To: <201112211835.44158.justlikeef@gmail.com> References: <201112202245.57010.justlikeef@gmail.com> <4EF1BC79.4000903@puzzled.xs4all.nl> <201112211835.44158.justlikeef@gmail.com> Message-ID: <17985.1324539095@ccs.covici.com> I am using portaudio 19_pre20110326 with no problems. Its not great under Linux, but I am not getting seg faults. Rob Hutton wrote: > It looks like portaudio has completely changed, and pablio no longer works. Has anyone started an effort to rewrite mod_portaudio for portaudio v19? I have seen a couple of notes of new code on the list, but haven't seen anything... > > On Wednesday 21 December 2011 06:01:13 Patrick Lists wrote: > > On 21-12-11 04:45, Rob Hutton wrote: > > > I am trying to get portaudio working, and have run into a couple of problems > > > > > > > > > 1) If portaudio cannot access the device, it causes freeswitch to segfault > > > > > > > > > 2) I have set up a dialplan following the intercom example which seems > > > to be working, but I get no audio. I have tried setting every device > > > shown in devlist as the output device. > > > > > > > > > 3) When I issue a "pa play". the cli hangs, even if I give it a timeout. > > > > > > > > > Any thoughts? > > > > I think someone reported on the mailing list a while back that he solved > > his troubles with portaudio by upgrading the portaudio source to the > > latest stable release which currently is: > > > > http://www.portaudio.com/archives/pa_stable_v19_20111121.tgz > > > > Regards, > > Patrick > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > ---------------------------------------------------- > 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 ryan at kaevee.com Thu Dec 22 10:59:38 2011 From: ryan at kaevee.com (Ryan V) Date: Thu, 22 Dec 2011 13:29:38 +0530 Subject: [Freeswitch-users] One way voice on incoming calls In-Reply-To: References: Message-ID: Hi, I reconfigured the PRI line and restarted the freeswitch. This solved the problem. I have taken the backup of the configuration which did not work and compare the current configuration and update the list. Thanks for all the help. Ryan, On Tue, Dec 20, 2011 at 10:56 PM, Ryan V wrote: > Thanks for the reply. Let me check first thing in morning. > > Ryan. > > > On Tue, Dec 20, 2011 at 8:52 PM, curriegrad2004 wrote: > >> I'm suggesting there may be mismatched rtp port ranges on the grandstream >> phone and on the FreeSWITCH server. Use wireshark or run a tcpdump and see >> what's going on. It's way easier to see problems from the inside than >> outside. >> On 2011-12-20 6:34 AM, "Ryan V" wrote: >> >>> Hi, >>> >>> We have configured a Sangoma A101DE. We are using Grandstream GXP280 >>> phones and no NAT involved. We are using example configuration files except >>> for one change in 00_inbound_did.xml. >>> >>> >>> >>> Calls between extensions works fine and outbound calls also work fine. >>> >>> Though we are able to receive inbound calls, incoming audio is missing. >>> Calling party hears us loud and clear. >>> >>> Any suggestions? >>> >>> Thanks, >>> >>> 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/20111222/0fbfc540/attachment.html From ryan at kaevee.com Thu Dec 22 11:06:47 2011 From: ryan at kaevee.com (Ryan V) Date: Thu, 22 Dec 2011 13:36:47 +0530 Subject: [Freeswitch-users] Ringback on PRI line Message-ID: Hi, We are running freeswitch with Sangoma PRI and Analog cards. There is no ring back for calls coming in on PRI line. Calls coming in on FXO lines are fine. Calling party gets ring back. I have added following to file conf/dialplan/public/00_inbound_did.xml Any suggestions to resolve this problem? Thanks, Venkatesh K -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/d8c00f60/attachment.html From lloydie.t at gmail.com Thu Dec 22 11:13:47 2011 From: lloydie.t at gmail.com (lloyd thomas) Date: Thu, 22 Dec 2011 08:13:47 +0000 Subject: [Freeswitch-users] Problem with make current In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C5B279DBF1C@cooper> <1324509156638-7116980.post@n2.nabble.com> Message-ID: maybe it's time to give up. tried 'make current', bootstrap and configure. new error ./configure: line 11073: syntax error near unexpected token `openssl,' ./configure: line 11073: ` PKG_CHECK_MODULES(openssl, openssl,' configure: error: ./configure.gnu failed for libs/iksemel On 21 December 2011 23:36, lloyd thomas wrote: > Oh Dear. I was hoping to fix my registration problems with this. Will open > a ticket > > > > On 21 December 2011 23:12, Jeff Lenk wrote: > >> thats crazy that a compiler should care about checking a return parameter. >> anyways you should open a Jira on this so it gets fixed. >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Problem-with-make-current-tp7113961p7116980.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/20111222/d5854189/attachment.html From peter.olsson at visionutveckling.se Thu Dec 22 11:30:08 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Thu, 22 Dec 2011 09:30:08 +0100 Subject: [Freeswitch-users] Problem with make current In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C5B279DBF1C@cooper> <1324509156638-7116980.post@n2.nabble.com> Message-ID: <549CFEF87AEDE841A38E9D15EAB4C04C5B279DC017@cooper> Check out issue http://jira.freeswitch.org/browse/FS-3642 for this one. Especially the first comment from Jeff - you're missing pkg-config. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r lloyd thomas Skickat: den 22 december 2011 09:14 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Problem with make current maybe it's time to give up. tried 'make current', bootstrap and configure. new error ./configure: line 11073: syntax error near unexpected token `openssl,' ./configure: line 11073: ` PKG_CHECK_MODULES(openssl, openssl,' configure: error: ./configure.gnu failed for libs/iksemel On 21 December 2011 23:36, lloyd thomas > wrote: Oh Dear. I was hoping to fix my registration problems with this. Will open a ticket On 21 December 2011 23:12, Jeff Lenk > wrote: thats crazy that a compiler should care about checking a return parameter. anyways you should open a Jira on this so it gets fixed. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Problem-with-make-current-tp7113961p7116980.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:4ef2e5fc32761950963548! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/623c181a/attachment-0001.html From beppe.grillo at gmail.com Thu Dec 22 11:36:33 2011 From: beppe.grillo at gmail.com (Beppe Grillo) Date: Thu, 22 Dec 2011 09:36:33 +0100 Subject: [Freeswitch-users] Accept-Contact in sip INVITE Message-ID: Hi, I need insert Accept-Contact in sip INVITE . I saw that RFC-3841 is supported . Do I configure something in the profile of gw to enter the Accept-contact in Sip Invite ? Regards, Giuseppe -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/a4cf9db3/attachment.html From chris at ghosttelecom.com Thu Dec 22 12:50:58 2011 From: chris at ghosttelecom.com (Chris Martineau) Date: Thu, 22 Dec 2011 09:50:58 -0000 Subject: [Freeswitch-users] rtp timestamp issue on pickup In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5B279DBDEC@cooper> References: <1D10AB188D6CCA46BB4369E3268E36EF402E39@SVR01.ghosttelecom.local><549CFEF87AEDE841A38E9D15EAB4C04C5B279DBC79@cooper><1D10AB188D6CCA46BB4369E3268E36EF402E77@SVR01.ghosttelecom.local> <549CFEF87AEDE841A38E9D15EAB4C04C5B279DBDEC@cooper> Message-ID: <1D10AB188D6CCA46BB4369E3268E36EF402EAE@SVR01.ghosttelecom.local> Thanks that worked great. Regards Chris From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson Sent: 21 December 2011 12:46 To: 'FreeSWITCH Users Help' Subject: Re: [Freeswitch-users] rtp timestamp issue on pickup Try to enable rtp-rewrite-timestamps according to http://wiki.freeswitch.org/wiki/RTP_Issues#Dropped_Audio, that might help. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Chris Martineau Skickat: den 21 december 2011 13:33 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] rtp timestamp issue on pickup Thanks for the reply, Looking at the wireshark it doesn't seem to activate the mark bit again? It's not that freeswitch lowers the ts it is just forwarding the incoming timestamp which is out of synch with the timestamp of the parked call, maybe this is why it is not being set? Any other ideas? Thanks Chris From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Peter Olsson Sent: 21 December 2011 09:51 To: 'freeswitch-users at lists.freeswitch.org' Subject: Re: [Freeswitch-users] rtp timestamp issue on pickup If FS lowers the timestamp FS activates the mark bit, which should cause the client to be able to handle this correctly. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Chris Martineau Skickat: den 21 december 2011 10:33 Till: freeswitch-users at lists.freeswitch.org ?mne: [Freeswitch-users] rtp timestamp issue on pickup Hi, My specific application requires an incoming call to pickup an existing waiting call. The first call is terminated and parked. The second call rings in and uses intercept to then connect it to the waiting call. The problem I have is that the second call has a lower rtp timestamp than the waiting call causing the client codec to wait until it catches up? i.e client incoming rtp stream timestamp effectively looks like this ... existing stream from freeswitch 37920 38080 38240 On intercept the timestamps then become that of the other calls stream 3040 3200 ... The client then doesn't get any incoming voice until the new timestamps catch up! Is there any way to set freeswitch to maintain the timestamps? Should the client allow for this? Any ideas would be greatly appreciated. Regards Chris !DSPAM:4ef1d21e32762117429169! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/70c9f0ab/attachment.html From lloydie.t at gmail.com Thu Dec 22 13:13:08 2011 From: lloydie.t at gmail.com (lloyd thomas) Date: Thu, 22 Dec 2011 10:13:08 +0000 Subject: [Freeswitch-users] Problem with make current In-Reply-To: <549CFEF87AEDE841A38E9D15EAB4C04C5B279DC017@cooper> References: <549CFEF87AEDE841A38E9D15EAB4C04C5B279DBF1C@cooper> <1324509156638-7116980.post@n2.nabble.com> <549CFEF87AEDE841A38E9D15EAB4C04C5B279DC017@cooper> Message-ID: Thanks for that. will try again On 22 December 2011 08:30, Peter Olsson wrote: > Check out issue http://jira.freeswitch.org/browse/FS-3642 for this one. > Especially the first comment from Jeff ? you?re missing pkg-config.**** > > ** ** > > /Peter**** > > ** ** > > ** ** > > *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *F?r *lloyd thomas > *Skickat:* den 22 december 2011 09:14 > > *Till:* FreeSWITCH Users Help > *?mne:* Re: [Freeswitch-users] Problem with make current**** > > ** ** > > maybe it's time to give up. > > tried 'make current', bootstrap and configure. > new error > ./configure: line 11073: syntax error near unexpected token `openssl,' > ./configure: line 11073: ` PKG_CHECK_MODULES(openssl, openssl,' > configure: error: ./configure.gnu failed for libs/iksemel > > **** > > On 21 December 2011 23:36, lloyd thomas wrote:**** > > Oh Dear. I was hoping to fix my registration problems with this. Will open > a ticket**** > > > > **** > > On 21 December 2011 23:12, Jeff Lenk wrote:**** > > thats crazy that a compiler should care about checking a return parameter. > anyways you should open a Jira on this so it gets fixed. > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Problem-with-make-current-tp7113961p7116980.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:4ef2e5fc32761950963548! **** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/9087c15c/attachment.html From nbhatti at gmail.com Thu Dec 22 13:50:54 2011 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Thu, 22 Dec 2011 13:50:54 +0300 Subject: [Freeswitch-users] how to know which gateway used after continue_on_fail=true Message-ID: Hello, I am using LUA to generate a DP and set channel variables for the bridge. All the variables are set, and LUA exits and the bridge is made outside the LUA script and the call is connected. I am using mod_xml_cdr for CDRs. If I set 1 gateway and bridge the call, I can see that gateway used in the CDR. Problem comes in when I use continue_on_fail=true, and have multiple gateways in bridge. The call fails from the first gateway, and is successfully connected via the second gateway. But I am not able to see the real gateway used in the CDR. I tried enabling log-b-leg in mod_xml_cdr to see what it shows, but I just can't see the gateway which was actually used to bridge the call. In the last_asg, I see both of the gateways passed. How can I know which gateway was used for the bridge? And no, I am NOT using mod_lcr. -Muhammad From sherifomran2000 at yahoo.com Thu Dec 22 14:02:55 2011 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Thu, 22 Dec 2011 03:02:55 -0800 (PST) Subject: [Freeswitch-users] how is codec selected In-Reply-To: Message-ID: <1324551775.39544.YahooMailClassic@web110807.mail.gq1.yahoo.com> Hi Zohair, In which file should i put this parameter How does this command work ngrep?-q -Wbyline '' port 5060 and host x.x.x.x Should i make a call and during the call type it? kind regards, Sherif --- On Thu, 12/22/11, Zohair Raza wrote: From: Zohair Raza Subject: Re: [Freeswitch-users] how is codec selected To: "FreeSWITCH Users Help" Date: Thursday, December 22, 2011, 9:00 AM Hi,? You can set codec?for a phone from dialplan using?absolute_codec_string parameter as stated in codec negotiation link. Trans-coding?issues in case of G729 or G723 can be handled as:?http://wiki.freeswitch.org/wiki/Sofia#sip_renegotiate_codec_on_reinvite See?http://wiki.freeswitch.org/wiki/Channel_Variables#Codec_Related?for codec related variables. Also, there is a configuration in internal and external profiles to define a list of codec in each profile which can be set in global configurationhttp://wiki.freeswitch.org/wiki/Config_external.xml you can see SIP dialogues on each phone IP to verify which codec are being offered? ngrep?-q -Wbyline '' port 5060 and host x.x.x.x ? Regards, Zohair Raza On Thu, Dec 22, 2011 at 2:54 AM, Nandy Dagondon wrote: it's found here: http://wiki.freeswitch.org/wiki/Codec_negotiation hope this answers your question. On Thu, Dec 22, 2011 at 5:25 AM, Sherif Omran wrote: hello , does anybody know how to set up the codec selection negotiation list in freeswitch? How can you know which codec has been selected for such a call and which codecs are offered from telephone A and B. I am using log 7 thank you in advance k.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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/4bf8185c/attachment.html From engineerzuhairraza at gmail.com Thu Dec 22 14:09:45 2011 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Thu, 22 Dec 2011 15:09:45 +0400 Subject: [Freeswitch-users] how is codec selected In-Reply-To: <1324551775.39544.YahooMailClassic@web110807.mail.gq1.yahoo.com> References: <1324551775.39544.YahooMailClassic@web110807.mail.gq1.yahoo.com> Message-ID: Hi, You can find it in sofia.conf.xml For ngrep, you need to have ngrep installed in your system try "yum install ngrep" if you are using CentOS/RedHat and "apt-get install ngrep" if Ubuntu Yes, you need to make a call, as you want to know which codecs are being carried from leg A to FS and FS to Log B Regards, Zohair Raza On Thu, Dec 22, 2011 at 3:02 PM, Sherif Omran wrote: > Hi Zohair, > > In which file should i put this parameter > > > How does this command work > > ngrep -q -Wbyline '' port 5060 and host x.x.x.x > > > Should i make a call and during the call type it? > > kind regards, > Sherif > > > > --- On *Thu, 12/22/11, Zohair Raza * wrote: > > > From: Zohair Raza > Subject: Re: [Freeswitch-users] how is codec selected > To: "FreeSWITCH Users Help" > Date: Thursday, December 22, 2011, 9:00 AM > > > Hi, > > You can set codec for a phone from dialplan using absolute_codec_string > parameter as stated in codec negotiation link. > > Trans-coding issues in case of G729 or G723 can be handled as: > http://wiki.freeswitch.org/wiki/Sofia#sip_renegotiate_codec_on_reinvite > > See http://wiki.freeswitch.org/wiki/Channel_Variables#Codec_Related for > codec related variables. > > Also, there is a configuration in internal and external profiles to define > a list of codec in each profile which can be set in global configuration > http://wiki.freeswitch.org/wiki/Config_external.xml > > > > > > you can see SIP dialogues on each phone IP to verify which codec are being > offered > > ngrep -q -Wbyline '' port 5060 and host x.x.x.x > > > Regards, > Zohair Raza > > > On Thu, Dec 22, 2011 at 2:54 AM, Nandy Dagondon > > wrote: > > it's found here: > http://wiki.freeswitch.org/wiki/Codec_negotiation > > hope this answers your question. > > On Thu, Dec 22, 2011 at 5:25 AM, Sherif Omran > > wrote: > > hello , > > does anybody know how to set up the codec selection negotiation list in > freeswitch? How can you know which codec has been selected for such a call > and which codecs are offered from telephone A and B. > > I am using log 7 > > > thank you in advance > > k.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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/99e233a7/attachment-0001.html From avi at avimarcus.net Thu Dec 22 14:35:03 2011 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 22 Dec 2011 13:35:03 +0200 Subject: [Freeswitch-users] how to know which gateway used after continue_on_fail=true In-Reply-To: References: Message-ID: I usually manually add [gateway=mycarrier] to each bridge leg, that it gets logged in leg B. -Avi On Thu, Dec 22, 2011 at 12:50 PM, Muhammad Naseer Bhatti wrote: > Hello, I am using LUA to generate a DP and set channel variables for > the bridge. All the variables are set, and LUA exits and the bridge is > made outside the LUA script and the call is connected. I am using > mod_xml_cdr for CDRs. If I set 1 gateway and bridge the call, I can > see that gateway used in the CDR. Problem comes in when I use > continue_on_fail=true, and have multiple gateways in bridge. The call > fails from the first gateway, and is successfully connected via the > second gateway. But I am not able to see the real gateway used in the > CDR. I tried enabling log-b-leg in mod_xml_cdr to see what it shows, > but I just can't see the gateway which was actually used to bridge the > call. In the last_asg, I see both of the gateways passed. > > How can I know which gateway was used for the bridge? And no, I am NOT > using mod_lcr. > > > -Muhammad > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/ffca9e76/attachment.html From nbhatti at gmail.com Thu Dec 22 15:09:44 2011 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Thu, 22 Dec 2011 15:09:44 +0300 Subject: [Freeswitch-users] how to know which gateway used after continue_on_fail=true In-Reply-To: References: Message-ID: So, I would have to parse xml_cdr, for hangup_cause=NORMAL_CLEARING and billsec > 0, and that will only be the last good leg for the call was connected? The rest of the legs will return hangup_cause != NORMAL_CLEARING -M On Thu, Dec 22, 2011 at 2:35 PM, Avi Marcus wrote: > I usually manually add [gateway=mycarrier] to each bridge leg, that it gets > logged in leg B. > > -Avi > > > On Thu, Dec 22, 2011 at 12:50 PM, Muhammad Naseer Bhatti > wrote: >> >> Hello, I am using LUA to generate a DP and set channel variables for >> the bridge. All the variables are set, and LUA exits and the bridge is >> made outside the LUA script and the call is connected. I am using >> mod_xml_cdr for CDRs. If I set 1 gateway and bridge the call, I can >> see that gateway used in the CDR. Problem comes in when I use >> continue_on_fail=true, and have multiple gateways in bridge. The call >> fails from the first gateway, and is successfully connected via the >> second gateway. But I am not able to see the real gateway used in the >> CDR. I tried enabling log-b-leg in mod_xml_cdr to see what it shows, >> but I just can't see the gateway which was actually used to bridge the >> call. In the last_asg, I see both of the gateways passed. >> >> How can I know which gateway was used for the bridge? And no, I am NOT >> using mod_lcr. >> >> >> -Muhammad >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From roger.castaldo at gmail.com Thu Dec 22 18:41:04 2011 From: roger.castaldo at gmail.com (Roger Castaldo) Date: Thu, 22 Dec 2011 10:41:04 -0500 Subject: [Freeswitch-users] Event Socket Message-ID: Hi everyone i have a simple question, is there an api command that will list all the available events that can be listened on by the event socket? i realize there is a wiki page containing a list but for my purposes an api command to list the available event names would be much better. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/f2d53f96/attachment.html From justlikeef at gmail.com Thu Dec 22 19:17:11 2011 From: justlikeef at gmail.com (Rob Hutton) Date: Thu, 22 Dec 2011 11:17:11 -0500 Subject: [Freeswitch-users] Need help with PortAudio In-Reply-To: <17985.1324539095@ccs.covici.com> References: <201112202245.57010.justlikeef@gmail.com> <201112211835.44158.justlikeef@gmail.com> <17985.1324539095@ccs.covici.com> Message-ID: <201112221117.12308.justlikeef@gmail.com> What module are you using? According to the wiki, pablio is no longer maintained or working... On Thursday 22 December 2011 02:31:35 covici at ccs.covici.com wrote: > I am using portaudio 19_pre20110326 with no problems. Its not great > under Linux, but I am not getting seg faults. > > Rob Hutton wrote: > > > It looks like portaudio has completely changed, and pablio no longer works. Has anyone started an effort to rewrite mod_portaudio for portaudio v19? I have seen a couple of notes of new code on the list, but haven't seen anything... > > > > On Wednesday 21 December 2011 06:01:13 Patrick Lists wrote: > > > On 21-12-11 04:45, Rob Hutton wrote: > > > > I am trying to get portaudio working, and have run into a couple of problems > > > > > > > > > > > > 1) If portaudio cannot access the device, it causes freeswitch to segfault > > > > > > > > > > > > 2) I have set up a dialplan following the intercom example which seems > > > > to be working, but I get no audio. I have tried setting every device > > > > shown in devlist as the output device. > > > > > > > > > > > > 3) When I issue a "pa play". the cli hangs, even if I give it a timeout. > > > > > > > > > > > > Any thoughts? > > > > > > I think someone reported on the mailing list a while back that he solved > > > his troubles with portaudio by upgrading the portaudio source to the > > > latest stable release which currently is: > > > > > > http://www.portaudio.com/archives/pa_stable_v19_20111121.tgz > > > > > > Regards, > > > Patrick > > > > > > _________________________________________________________________________ > > > Professional FreeSWITCH Consulting Services: > > > consulting at freeswitch.org > > > http://www.freeswitchsolutions.com > > > > > > > > > > > > > > > Official FreeSWITCH Sites > > > http://www.freeswitch.org > > > http://wiki.freeswitch.org > > > http://www.cluecon.com > > > > > > FreeSWITCH-users mailing list > > > FreeSWITCH-users at lists.freeswitch.org > > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > > http://www.freeswitch.org > > > > > > > ---------------------------------------------------- > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/11826794/attachment-0001.html From Hector.Geraldino at ip-soft.net Thu Dec 22 19:19:33 2011 From: Hector.Geraldino at ip-soft.net (Hector Geraldino) Date: Thu, 22 Dec 2011 11:19:33 -0500 Subject: [Freeswitch-users] Event Socket In-Reply-To: References: Message-ID: <6A6B4C284AD15042B429EB9D904544AD0225507272@NY1-EXMB-01.ip-soft.net> Hi Roger, If the set of available events is constant (any of the existing events can be received by the ESL application at any time), I don't see why you need to query FS to get this list. Let's suppose this api command exists: issuing this command will always return a list with the same values, which are the values already listed on the Events list page: http://wiki.freeswitch.org/wiki/Event_List Can you do that? I'm pretty sure you can't. But, if you *really* want to get this list of events after issuing an api command to freeswitch (non-negotiable feature), you can always send an "api echo STRING_LIST" with the list of possible values: api echo CHANNEL_CALLSTATE CHANNEL_CREATE CHANNEL_DESTROY CHANNEL_STATE CHANNEL_ANSWER CHANNEL_HANGUP CHANNEL_HANGUP_COMPLETE CHANNEL_EXECUTE CHANNEL_EXECUTE_COMPLETE Content-Type: api/response Content-Length: 157 CHANNEL_CALLSTATE CHANNEL_CREATE CHANNEL_DESTROY CHANNEL_STATE CHANNEL_ANSWER CHANNEL_HANGUP CHANNEL_HANGUP_COMPLETE CHANNEL_EXECUTE CHANNEL_EXECUTE_COMPLETE PS: I'm kidding From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Roger Castaldo Sent: Thursday, December 22, 2011 10:41 AM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Event Socket Hi everyone i have a simple question, is there an api command that will list all the available events that can be listened on by the event socket? i realize there is a wiki page containing a list but for my purposes an api command to list the available event names would be much better. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/046b9857/attachment.html From roger.castaldo at gmail.com Thu Dec 22 20:04:17 2011 From: roger.castaldo at gmail.com (Roger Castaldo) Date: Thu, 22 Dec 2011 12:04:17 -0500 Subject: [Freeswitch-users] Event Socket In-Reply-To: <6A6B4C284AD15042B429EB9D904544AD0225507272@NY1-EXMB-01.ip-soft.net> References: <6A6B4C284AD15042B429EB9D904544AD0225507272@NY1-EXMB-01.ip-soft.net> Message-ID: Okay thanks for the info I will just have to do it somewhat differently in my code then, as i said I was just curious thinking that there might be some obscure command to list them On Thu, Dec 22, 2011 at 11:19 AM, Hector Geraldino < Hector.Geraldino at ip-soft.net> wrote: > Hi Roger,**** > > ** ** > > If the set of available events is constant (any of the existing events can > be received by the ESL application at any time), I don?t see why you need > to query FS to get this list. Let?s suppose this api command exists: > issuing this command will always return a list with the same values, which > are the values already listed on the Events list page:**** > > ** ** > > http://wiki.freeswitch.org/wiki/Event_List**** > > ** ** > > Can you do that? I?m pretty sure you can?t. But, if you **really** want > to get this list of events after issuing an api command to freeswitch > (non-negotiable feature), you can always send an ?api echo STRING_LIST? > with the list of possible values:**** > > ** ** > > api echo CHANNEL_CALLSTATE CHANNEL_CREATE CHANNEL_DESTROY CHANNEL_STATE > CHANNEL_ANSWER CHANNEL_HANGUP CHANNEL_HANGUP_COMPLETE CHANNEL_EXECUTE > CHANNEL_EXECUTE_COMPLETE**** > > ** ** > > Content-Type: api/response**** > > Content-Length: 157**** > > ** ** > > CHANNEL_CALLSTATE CHANNEL_CREATE CHANNEL_DESTROY CHANNEL_STATE > CHANNEL_ANSWER CHANNEL_HANGUP CHANNEL_HANGUP_COMPLETE CHANNEL_EXECUTE > CHANNEL_EXECUTE_COMPLETE **** > > ** ** > > ** ** > > PS: I?m kidding**** > > ** ** > > *From:* freeswitch-users-bounces at lists.freeswitch.org [mailto: > freeswitch-users-bounces at lists.freeswitch.org] *On Behalf Of *Roger > Castaldo > *Sent:* Thursday, December 22, 2011 10:41 AM > *To:* freeswitch-users at lists.freeswitch.org > *Subject:* [Freeswitch-users] Event Socket**** > > ** ** > > Hi everyone i have a simple question, is there an api command that will > list all the available events that can be listened on by the event socket? > i realize there is a wiki page containing a list but for my purposes an api > command to list the available event names would be much better.**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/8b2a033a/attachment.html From vetali100 at gmail.com Thu Dec 22 21:19:18 2011 From: vetali100 at gmail.com (Vitalie Colosov) Date: Thu, 22 Dec 2011 10:19:18 -0800 Subject: [Freeswitch-users] Some questions regarding IPs and gateway In-Reply-To: <9d0848ccace3e1d34699e53d24f49e37.squirrel@fulvetta.riseup.net> References: <9a4d8ee96b56592ac4250c9805cfb1a7.squirrel@fulvetta.riseup.net> <9d0848ccace3e1d34699e53d24f49e37.squirrel@fulvetta.riseup.net> Message-ID: " .I solved 2. and 4. by myself, 1. and 2. remain open... " You have last try :) 2011/12/21 > Its already late...I solved 2. and 4. by myself, 1. and 2. remain open... > > Cheers, > Georg > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/812f6b9f/attachment.html From vetali100 at gmail.com Thu Dec 22 21:23:34 2011 From: vetali100 at gmail.com (Vitalie Colosov) Date: Thu, 22 Dec 2011 10:23:34 -0800 Subject: [Freeswitch-users] Some questions regarding IPs and gateway In-Reply-To: References: <9a4d8ee96b56592ac4250c9805cfb1a7.squirrel@fulvetta.riseup.net> <9d0848ccace3e1d34699e53d24f49e37.squirrel@fulvetta.riseup.net> Message-ID: Regarding #1: Do you have 2 network cards? One has global IP (evil internet) and one has local IP (the one which you want to use for FreeSWITCH) ? 2011/12/22 Vitalie Colosov > " .I solved 2. and 4. by myself, 1. and 2. remain open... " > > You have last try :) > > 2011/12/21 > > Its already late...I solved 2. and 4. by myself, 1. and 2. remain open... >> >> Cheers, >> Georg >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/2d464f6a/attachment-0001.html From patrick at sunsus.net Thu Dec 22 09:49:33 2011 From: patrick at sunsus.net (sunsus) Date: Wed, 21 Dec 2011 22:49:33 -0800 (PST) Subject: [Freeswitch-users] FreeSWITCH cluster with IM/Presence In-Reply-To: <9FD45932-DD45-4370-A08B-9C06C6D3289C@me.com> References: <9FD45932-DD45-4370-A08B-9C06C6D3289C@me.com> Message-ID: <1324536573917-7117749.post@n2.nabble.com> Hi Have you allready build your POC? How was it going? regards Patrick -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FreeSWITCH-cluster-with-IM-Presence-tp4697419p7117749.html Sent from the freeswitch-users mailing list archive at Nabble.com. From voipservice911 at gmail.com Thu Dec 22 10:44:27 2011 From: voipservice911 at gmail.com (Voip service) Date: Thu, 22 Dec 2011 11:44:27 +0400 Subject: [Freeswitch-users] Number of calls Message-ID: Hi, I am new in voip, how many calls can one freeswitch box handle with 30 % of trans-coded calls and system configuration as 8GB RAM X3430 Xeon Processor, 2.4GHz, 8M Cache, Turbo -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/51a085bd/attachment.html From faisal.rehman22 at hotmail.com Thu Dec 22 11:46:00 2011 From: faisal.rehman22 at hotmail.com (Faisal Rehman) Date: Thu, 22 Dec 2011 13:46:00 +0500 Subject: [Freeswitch-users] Freeswitch Crashing again and again Message-ID: Dear FS users Our freeswitch has been working stable for around 1.5 years. Currently we are getting Freeswitch crashing without any apperent reasons with following coredump Core was generated by `/usr/local/freeswitch/bin/freeswitch -nc'.Program terminated with signal 11, Segmentation fault.#0 0x0000003d922790d0 in strchr () from /lib64/libc.so.6(gdb) bt#0 0x0000003d922790d0 in strchr () from /lib64/libc.so.6#1 0x00002b7677883008 in switch_ivr_originate (session=0x2aaad020cab8, bleg=0x54091ce0, cause=0x54091cec, bridgeto=0x2b7677950557 "true", timelimit_sec=60, table=0x0, cid_name_override=0x0, cid_num_override=0x0, caller_profile_override=0x0, ovars=0x0, flags=, cancel_cause=0x0) at ./src/include/switch_utils.h:429#2 0x00002aaac8cf68f8 in audio_bridge_function (session=0x2aaad020cab8, data=0x2aaaac754480 "sofia/x8888/47756508496197 at 10.47.78.170:5654") at /usr/src/freeswitch/src/mod/applications/mod_dptools/mod_dptools.c:2683#3 0x00002b7677853952 in switch_core_session_exec (session=0x2aaad020cab8, application_interface=0x2aaab00553b8, arg=0x2aaad0006b28 "sofia/${flag_auth}") at src/switch_core_session.c:2121#4 0x00002b7677853f7f in switch_core_session_execute_application_get_flags (session=0x0, app=0x2aaad0006b20 "bridge", arg=0x2aaad0006b28 "sofia/${flag_auth}", flags=0x0) at src/switch_core_session.c:2006#5 0x00002b7677856963 in switch_core_session_run (session=0x2aaad020cab8) at src/switch_core_state_machine.c:181#6 0x00002b7677850d30 in switch_core_session_thread (thread=, obj=0x2aaad020cab8) at src/switch_core_session.c:1311#7 0x0000003d92a0673d in start_thread () from /lib64/libpthread.so.0#8 0x0000003d922d3d1d in clone () from /lib64/libc.so.6 Please help to fix this issue as it is very urgent. Thanks and Regards, Faisal -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/5f5b18b0/attachment.html From georg at riseup.net Thu Dec 22 14:32:09 2011 From: georg at riseup.net (georg at riseup.net) Date: Thu, 22 Dec 2011 12:32:09 +0100 Subject: [Freeswitch-users] Some questions regarding IPs and gateway In-Reply-To: <9d0848ccace3e1d34699e53d24f49e37.squirrel@fulvetta.riseup.net> References: <9a4d8ee96b56592ac4250c9805cfb1a7.squirrel@fulvetta.riseup.net> <9d0848ccace3e1d34699e53d24f49e37.squirrel@fulvetta.riseup.net> Message-ID: > Its already late...I solved 2. and 4. by myself, 1. and 2. remain open... Oh god...this is also not correct. So up until now, I found no solution to 1. and 3. Georg From paul at cupis.co.uk Thu Dec 22 22:28:07 2011 From: paul at cupis.co.uk (Paul Cupis) Date: Thu, 22 Dec 2011 19:28:07 +0000 Subject: [Freeswitch-users] how to know which gateway used after continue_on_fail=true In-Reply-To: References: Message-ID: <4EF384C7.6090106@cupis.co.uk> On 22/12/11 10:50, Muhammad Naseer Bhatti wrote: > Hello, I am using LUA to generate a DP and set channel variables for > the bridge. All the variables are set, and LUA exits and the bridge is > made outside the LUA script and the call is connected. I am using > mod_xml_cdr for CDRs. If I set 1 gateway and bridge the call, I can > see that gateway used in the CDR. Problem comes in when I use > continue_on_fail=true, and have multiple gateways in bridge. The call > fails from the first gateway, and is successfully connected via the > second gateway. But I am not able to see the real gateway used in the > CDR. I tried enabling log-b-leg in mod_xml_cdr to see what it shows, > but I just can't see the gateway which was actually used to bridge the > call. In the last_asg, I see both of the gateways passed. > > How can I know which gateway was used for the bridge? And no, I am NOT > using mod_lcr. The information should be available in the app_log section of the a_leg XML CDR. Have a look at one of your CDRs, and compare the section to the section. If you are still having trouble with this, perhaps post back with example CDRs (possibly redacted and on pastebin). Regards, From nbhatti at gmail.com Thu Dec 22 22:34:09 2011 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Thu, 22 Dec 2011 22:34:09 +0300 Subject: [Freeswitch-users] how to know which gateway used after continue_on_fail=true In-Reply-To: <4EF384C7.6090106@cupis.co.uk> References: <4EF384C7.6090106@cupis.co.uk> Message-ID: http://wiki.freeswitch.org/wiki/Variable_failed_xml_cdr_prefix seems to have fixed the issue. I sent 4 gateways, 3 of them failed. 4th was the good one. It can be seen in the CDR. On Thu, Dec 22, 2011 at 10:28 PM, Paul Cupis wrote: > On 22/12/11 10:50, Muhammad Naseer Bhatti wrote: >> Hello, I am using LUA to generate a DP and set channel variables for >> the bridge. All the variables are set, and LUA exits and the bridge is >> made outside the LUA script and the call is connected. I am using >> mod_xml_cdr for CDRs. If I set 1 gateway and bridge the call, I can >> see that gateway used in the CDR. Problem comes in when I use >> continue_on_fail=true, and have multiple gateways in bridge. The call >> fails from the first gateway, and is successfully connected via the >> second gateway. But I am not able to see the real gateway used in the >> CDR. I tried enabling log-b-leg in mod_xml_cdr to see what it shows, >> but I just can't see the gateway which was actually used to bridge the >> call. In the last_asg, I see both of the gateways passed. >> >> How can I know which gateway was used for the bridge? And no, I am NOT >> using mod_lcr. > > The information should be available in the app_log section of the a_leg > XML CDR. > > Have a look at one of your CDRs, and compare the section to > the section. > > If you are still having trouble with this, perhaps post back with > example CDRs (possibly redacted and on pastebin). > > 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 From msc at freeswitch.org Thu Dec 22 23:17:10 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Dec 2011 12:17:10 -0800 Subject: [Freeswitch-users] Some questions regarding IPs and gateway In-Reply-To: References: <9a4d8ee96b56592ac4250c9805cfb1a7.squirrel@fulvetta.riseup.net> <9d0848ccace3e1d34699e53d24f49e37.squirrel@fulvetta.riseup.net> Message-ID: Georg, Welcome to FreeSWITCH! You've been busy. I have a question and a comment. On #1 - is that IP address for a NIC on the FreeSWITCH server itself or do you have a firewall and have FreeSWITCH sitting behind it? I think if you give us the specific details about your network layout that it would help us help you. For #3 - the -nonat argument is correct for disabling FreeSWITCH's "autonat" stuff. Also, the SIP profiles have NAT settings in them that can be adjusted. There are two flavors of NAT settings: those for when FreeSWITCH itself is behind NAT and those for when the clients connecting to FreeSWITCH are behind NAT. If you want an example of a "plain" SIP profile then look at conf/sip_profiles/external.xml - it's pretty basic and assumes that FreeSWITCH itself is not behind NAT. -MC On Thu, Dec 22, 2011 at 3:32 AM, wrote: > > Its already late...I solved 2. and 4. by myself, 1. and 2. remain open... > > Oh god...this is also not correct. So up until now, I found no solution to > 1. and 3. > > Georg > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/1f843090/attachment-0001.html From msc at freeswitch.org Thu Dec 22 23:22:55 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Dec 2011 12:22:55 -0800 Subject: [Freeswitch-users] Number of calls In-Reply-To: References: Message-ID: That looks like a decent 4-core processor. You can probably do several hundred concurrent calls w/o much issue. Of course, "it depends" on what other stuff you have going on inside this server. :) -MC On Wed, Dec 21, 2011 at 11:44 PM, Voip service wrote: > Hi, > > I am new in voip, how many calls can one freeswitch box handle with 30 % > of trans-coded calls and system configuration as > 8GB RAM > X3430 Xeon Processor, 2.4GHz, 8M Cache, Turbo > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/56e6ac91/attachment.html From msc at freeswitch.org Thu Dec 22 23:24:16 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Dec 2011 12:24:16 -0800 Subject: [Freeswitch-users] Freeswitch Crashing again and again In-Reply-To: References: Message-ID: Are you able to reproduce this on the latest git head? If so, open a Jira ticket at jira.freeswitch.org. If it is truly urgent then you can get commercial/professional help by contacting consulting at freeswitch.org. -MC On Thu, Dec 22, 2011 at 12:46 AM, Faisal Rehman wrote: > *Dear FS users* > > Our freeswitch has been working stable for around 1.5 years. > > Currently we are getting Freeswitch crashing without any apperent reasons > with following coredump > > > Core was generated by `/usr/local/freeswitch/bin/freeswitch -nc'. > Program terminated with signal 11, Segmentation fault. > #0 0x0000003d922790d0 in strchr () from /lib64/libc.so.6 > (gdb) bt > #0 0x0000003d922790d0 in strchr () from /lib64/libc.so.6 > #1 0x00002b7677883008 in switch_ivr_originate (session=0x2aaad020cab8, > bleg=0x54091ce0, cause=0x54091cec, > bridgeto=0x2b7677950557 "true", timelimit_sec=60, table=0x0, > cid_name_override=0x0, cid_num_override=0x0, > caller_profile_override=0x0, ovars=0x0, flags=, > cancel_cause=0x0) > at ./src/include/switch_utils.h:429 > #2 0x00002aaac8cf68f8 in audio_bridge_function (session=0x2aaad020cab8, > data=0x2aaaac754480 "sofia/x8888/47756508496197 at 10.47.78.170:5654") > at > /usr/src/freeswitch/src/mod/applications/mod_dptools/mod_dptools.c:2683 > #3 0x00002b7677853952 in switch_core_session_exec > (session=0x2aaad020cab8, application_interface=0x2aaab00553b8, > arg=0x2aaad0006b28 "sofia/${flag_auth}") at > src/switch_core_session.c:2121 > #4 0x00002b7677853f7f in > switch_core_session_execute_application_get_flags (session=0x0, > app=0x2aaad0006b20 "bridge", > arg=0x2aaad0006b28 "sofia/${flag_auth}", flags=0x0) at > src/switch_core_session.c:2006 > #5 0x00002b7677856963 in switch_core_session_run (session=0x2aaad020cab8) > at src/switch_core_state_machine.c:181 > #6 0x00002b7677850d30 in switch_core_session_thread (thread= optimized out>, obj=0x2aaad020cab8) > at src/switch_core_session.c:1311 > #7 0x0000003d92a0673d in start_thread () from /lib64/libpthread.so.0 > #8 0x0000003d922d3d1d in clone () from /lib64/libc.so.6 > > > *Please help to fix this issue as it is very urgent.* > > > Thanks and Regards, > > > *Faisal* > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/aec1feea/attachment.html From notlikeme75 at yahoo.com Thu Dec 22 23:41:19 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Thu, 22 Dec 2011 12:41:19 -0800 (PST) Subject: [Freeswitch-users] Odp: transfer to conference from conference / caller controls In-Reply-To: References: Message-ID: <1324586479.30344.YahooMailNeo@web65305.mail.ac2.yahoo.com> Mariusz, using your suggestion, i worked out a solution that works. although i have to drop the bleg with sched_hangup in order for it to work with my hangup back to ivr method. it still looks like crap in my cdr but I can deal with it for now since i am crazy for wanting to transfer from static conference right to dynamic :) but doing it with this loopback method seems to give me the caller controls back.? thanks for your help. ~rodney ?condition??? ? destination_number??? ? ^760$??? ? 1??? ? ? action??? ? answer??? ? ??? ? 3??? ? ? action??? ? set??? ? conference_user_list=|??? ? 10??? ? ? action??? ? play_and_get_digits??? ? 4 4 3 5000 # ask4digit.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+??? ? 20??? ? ? action??? ? phrase??? ? spell,${target_num}??? ? 30??? ? ? action??? ? sched_hangup??? ? +10 normal_clearing bleg??? ? 35??? ? ? action??? ? bridge??? ? loopback/app=conference:${target_num}-127.0.0.1 at dynamic_conf ? 4. Odp: transfer to conference from conference / caller controls ? ? ? (Mariusz Czulada) > Perhaps this will be useful for you: ? Dialplan: ? ??? ???? ... ??????? ???? ... ??????? ?? ? ... ??? ??? ???? ... ??????? ???? ... ??? ? conference.conf.xml: ? ??? ???? ... ????? ??? ??? ???? ... ????? ??? ??? ???? ... ????? ??? ??? ???? ... ????? ??? ? ? Dnia 20-12-2011 o godz. 18:21 Rodney napisa?(a): I am trying to transfer to a dynamic conference from within a static conference but when I do my callers loose control and can not use any dtmf caller control options and must hang up to get back. I tried using clear digit action but do not think this works on the "conf" realm. the dynamic conference works fine with all controls if i send them directly from the ivr but for what i am trying it is important to do this transfer from another conference. is there a method transfer and release all previous controls? i even tried using a difference conference profile so i transferred from static 501 at default to @dynamic to see if giving new controls would work, to no avail. is there a log or something i can post that will help you in understanding how this is happening? or is it even fixable? thanks rodney > > >?? >????? >????? >????? >????? >????? > >-???? >??????? >??????? ? (main ivr menu extension) >??????? > > > > > >condition??? ? destination_number??? ? ^759$??? ? 1??? ? >? action??? ? answer??? ? ??? ? 2??? ? >? action??? ? set??? ? conference_user_list=|??? ? 11??? ? >? action??? ? play_and_get_digits??? ? 4 4 3 5000 # askprivateroom.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+??? ? 15??? ? >? action??? ? phrase??? ? spell,${target_num}??? ? 16??? ? >? action??? ? conference??? ? ${target_num}-127.0.0.1 at default > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/fe3c91ad/attachment-0001.html From notlikeme75 at yahoo.com Thu Dec 22 23:49:52 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Thu, 22 Dec 2011 12:49:52 -0800 (PST) Subject: [Freeswitch-users] transfer back to ivr after voicemail function Message-ID: <1324586992.63365.YahooMailNeo@web65313.mail.ac2.yahoo.com> I am trying to transfer someone back to the IVR after they leave a message for someone or check their own mailbox. currently it does a hangup and they have to call back to do more.? is there a method for this? or is it hard coded in the "press # to continue portion of voice mail? check voice mail condition??? ? destination_number??? ? ^757$??? ? 1??? ? ? action??? ? answer??? ? ??? ? 3??? ? ? action??? ? sleep??? ? 1000??? ? 4??? ? ? action??? ? play_and_get_digits??? ? 4 4 3 5000 # checkvoicemail.wav ivr/ivr-that_was_an_invalid_entry.wav vmb \d+??? ? 12??? ? ? action??? ? phrase??? ? spell,${vmb}??? ? 15??? ? ? action??? ? voicemail???? check default ${domain_name} ${vmb} leave message ? condition??? ? destination_number??? ? ^758$??? ? 1??? ? ? action??? ? answer??? ? ??? ? 3??? ? ? action??? ? sleep??? ? 1000??? ? 4??? ? ? action??? ? play_and_get_digits??? ? 4 4 3 5000 # checkvoicemail.wav ivr/ivr-that_was_an_invalid_entry.wav vmb \d+??? ? 12??? ? ? action??? ? phrase??? ? spell,${vmb}??? ? 15??? ? ? action??? ? voicemail???? default ${domain_name} ${vmb} -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/411ffe1b/attachment.html From lloydie.t at gmail.com Fri Dec 23 00:01:03 2011 From: lloydie.t at gmail.com (lloyd thomas) Date: Thu, 22 Dec 2011 21:01:03 +0000 Subject: [Freeswitch-users] Problem with make current In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C5B279DBF1C@cooper> <1324509156638-7116980.post@n2.nabble.com> <549CFEF87AEDE841A38E9D15EAB4C04C5B279DC017@cooper> Message-ID: Nope. Back to make[5]: *** [mod_spandsp.la] Error 1 make[5]: Leaving directory `/usr/src/freeswitch/src/mod/applications/mod_spandsp' make[4]: *** [mod_spandsp-all] Error 1 make[4]: Leaving directory `/usr/src/freeswitch/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/src/freeswitch/src' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/src/freeswitch' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch' make: *** [current] Error 2 On 22 December 2011 10:13, lloyd thomas wrote: > Thanks for that. will try again > > > On 22 December 2011 08:30, Peter Olsson wrote: > >> Check out issue http://jira.freeswitch.org/browse/FS-3642 for this one. >> Especially the first comment from Jeff ? you?re missing pkg-config.**** >> >> ** ** >> >> /Peter**** >> >> ** ** >> >> ** ** >> >> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >> freeswitch-users-bounces at lists.freeswitch.org] *F?r *lloyd thomas >> *Skickat:* den 22 december 2011 09:14 >> >> *Till:* FreeSWITCH Users Help >> *?mne:* Re: [Freeswitch-users] Problem with make current**** >> >> ** ** >> >> maybe it's time to give up. >> >> tried 'make current', bootstrap and configure. >> new error >> ./configure: line 11073: syntax error near unexpected token `openssl,' >> ./configure: line 11073: ` PKG_CHECK_MODULES(openssl, openssl,' >> configure: error: ./configure.gnu failed for libs/iksemel >> >> **** >> >> On 21 December 2011 23:36, lloyd thomas wrote:**** >> >> Oh Dear. I was hoping to fix my registration problems with this. Will >> open a ticket**** >> >> >> >> **** >> >> On 21 December 2011 23:12, Jeff Lenk wrote:**** >> >> thats crazy that a compiler should care about checking a return parameter. >> anyways you should open a Jira on this so it gets fixed. >> >> -- >> View this message in context: >> http://freeswitch-users.2379917.n2.nabble.com/Problem-with-make-current-tp7113961p7116980.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:4ef2e5fc32761950963548! **** >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/7013c61b/attachment.html From ga at steadfasttelecom.com Fri Dec 23 01:09:08 2011 From: ga at steadfasttelecom.com (Gilad Abada) Date: Thu, 22 Dec 2011 17:09:08 -0500 Subject: [Freeswitch-users] Can't quite get call screening to work In-Reply-To: References: <013301ccb521$b1e4eca0$15aec5e0$@com> Message-ID: Hi I am trying to get call screening to work too. http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_13:_Call_Screening The issue I am having is that if the called device hangs up even after a full conversation (accepting the call by pressing 1) the calling party gets sent to voicemail. Seems like being sent to voicemail should be an anti-action? Also the recording says to reject the call press 2 or send it to VM press 3. These are not defined in the dial plan and I am not sure how to do that. Thanks in advance! Gill On Wed, Dec 7, 2011 at 5:51 PM, Michael Collins wrote: > it looks like the person who posted that example did not post their sample > phrase macro file. However, do a git pull... > > commit 9ea3ce666fa7f021b5c2a7e2fbe153eb351c5734 > Author: Michael S Collins > Date: ? Wed Dec 7 14:49:16 2011 -0800 > > ? ? config: add screen_confirm macro to lang/en/ivr/sounds.xml > > snag that config file and drop it into conf/lang/en/ivr/ and then reloadxml. > I did this on the fly w/o testing so be sure to test it thoroughly to make > sure it works! Also, be sure to use the full path name to the sound file > that you are playing back. > > -MC > > On Wed, Dec 7, 2011 at 12:49 PM, Phil Quesinberry > wrote: >> >> I'm trying to use the call screening example in the wiki and can't get FS >> to >> play the caller's name back to the destination extension. >> example here: >> >> http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_13:_Call_Scr >> eening >> >> The initial announcement asking for the caller's name works fine: >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 phrase(voicemail_record_name) >> 2011-12-07 14:52:13.715494 [DEBUG] mod_dptools.c:2362 Execute >> voicemail_record_name() lang >> 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:67 No language >> specified - Using [en] >> 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:244 Handle >> play-file:[voicemail/vm-record_name1.wav] (en:en) >> 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:1302 Codec >> Activated L16 at 8000hz 1 channels 20ms >> 2011-12-07 14:52:18.695891 [DEBUG] switch_ivr_play_say.c:1672 done playing >> file >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_name1.wav >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 playback(tone_stream://%(500, >> 0, >> 640)) >> 2011-12-07 14:52:18.799791 [DEBUG] switch_ivr_play_say.c:1302 Codec >> Activated L16 at 8000hz 1 channels 20ms >> 2011-12-07 14:52:19.299814 [DEBUG] switch_ivr_play_say.c:1672 done playing >> file tone_stream://%(500, 0, 640) >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 >> set(playback_terminators=#*0123456789) >> 2011-12-07 14:52:19.299814 [DEBUG] mod_dptools.c:1263 >> sofia/internal/102 at 192.168.1.6:5060 SET >> [playback_terminators]=[#*0123456789] >> >> Then the caller's name is recorded, and I've verified that the recording >> is >> indeed saved in /tmp: >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 record(/tmp/102-name.wav 7 200 >> 2) >> 2011-12-07 14:52:19.299814 [DEBUG] switch_ivr_play_say.c:585 Raw Codec >> Activated >> 2011-12-07 14:52:19.299814 [DEBUG] switch_core_codec.c:116 >> sofia/internal/102 at 192.168.1.6:5060 Push codec L16:70 >> . >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(group_confirm_key=1) >> 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 >> sofia/internal/102 at 192.168.1.6:5060 SET [group_confirm_key]=[1] >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 >> set(fail_on_single_reject=true) >> 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 >> sofia/internal/102 at 192.168.1.6:5060 SET [fail_on_single_reject]=[true] >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 >> set(group_confirm_file=phrase:screen_confirm:/tmp/102-name.wav) >> 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 >> sofia/internal/102 at 192.168.1.6:5060 SET >> [group_confirm_file]=[phrase:screen_confirm:/tmp/102-name.wav] >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(continue_on_fail=true) >> 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 >> sofia/internal/102 at 192.168.1.6:5060 SET [continue_on_fail]=[true] >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bridge(user/102) >> . >> Then when attempting to play back the Output from the console showing the >> error is here: >> 2011-12-07 14:52:26.348406 [ERR] switch_ivr_play_say.c:142 Can't find >> macro >> screen_confirm. >> 2011-12-07 14:52:26.348406 [WARNING] switch_ivr_play_say.c:339 Macro >> [screen_confirm]: '/tmp/102-name.wav' did not match any patterns >> 2011-12-07 14:52:26.348406 [ERR] switch_ivr_originate.c:219 >> sofia/internal/sip:102 at 192.168.1.4:5060 Error Playing File! >> >> The call goes right to voicemail once the destination extension attempts >> to >> answer it. >> >> Where are these macros supposed to be stored? ?Somewhere under >> /usr/local/freeswitch/conf/lang/en? ?Do I need to create a macro for >> screen_confirm or is it just named incorrectly or in the wrong place? >> >> 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Gilad Abada SteadFast Telecommunications, Inc. Call us to find out how much you can save with VoIP! V: 212.589.1001 F: 212.589.1011 For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises.?? Make your virtual office a reality.? Enjoy the freedom to travel while remaining connected to your office. From georg at riseup.net Thu Dec 22 23:48:42 2011 From: georg at riseup.net (georg at riseup.net) Date: Thu, 22 Dec 2011 21:48:42 +0100 Subject: [Freeswitch-users] Some questions regarding IPs and gateway In-Reply-To: References: <9a4d8ee96b56592ac4250c9805cfb1a7.squirrel@fulvetta.riseup.net> <9d0848ccace3e1d34699e53d24f49e37.squirrel@fulvetta.riseup.net> Message-ID: Hello, sorry for all the confusion I did: > On #1 - is that IP address for a NIC on the FreeSWITCH server itself or do > you have a firewall and have FreeSWITCH sitting behind it? I think if you > give us the specific details about your network layout that it would help > us help you. Its a IP associated to one nic of the server. No firewall involved. I've got five nics, with five nets associated: - eth0 -> 192.168.0.X [using this for cluster-communication] - eth1 -> 192.168.2.X [using this for cluster-communication] - eth2 -> static ip, rechable from the internet - eth3 -> 192.168.1.X [using this for the phones] - eth4 -> private MPLS net to my provider, using this for the SIP trunk I would like exclude eth0/eth1/eth2 from the configs. I found the acl.conf where I'm able to configure netmasks and stuff like this. I will try this, hope this is correct. Also I will bind FS to 192.168.1.X trough the local ip setting in vars.xml. Comments? > For #3 - the -nonat argument is correct for disabling FreeSWITCH's > "autonat" stuff. Also, the SIP profiles have NAT settings in them that can > be adjusted. There are two flavors of NAT settings: those for when > FreeSWITCH itself is behind NAT and those for when the clients connecting > to FreeSWITCH are behind NAT. If you want an example of a "plain" SIP > profile then look at conf/sip_profiles/external.xml - it's pretty basic > and > assumes that FreeSWITCH itself is not behind NAT. Allright. Thanks! Georg From agoodstein at wcgltd.com Thu Dec 22 23:58:09 2011 From: agoodstein at wcgltd.com (Arnie Goodstein) Date: Thu, 22 Dec 2011 15:58:09 -0500 Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 66, Issue 140 Message-ID: <7A652B375DF11741B7C66038B60BEACE5C0E53BFB3@WCGS-EXCHANGE2.WCGS.local> Thanks Sent from my Android phone using TouchDown (www.nitrodesk.com) -----Original Message----- From: freeswitch-users-request at lists.freeswitch.org [freeswitch-users-request at lists.freeswitch.org] Received: Thursday, 22 Dec 2011, 3:41pm To: freeswitch-users at lists.freeswitch.org [freeswitch-users at lists.freeswitch.org] Subject: FreeSWITCH-users Digest, Vol 66, Issue 140 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: Number of calls (Michael Collins) 2. Re: Freeswitch Crashing again and again (Michael Collins) 3. Odp: transfer to conference from conference / caller controls (Rodney) ---------------------------------------------------------------------- Message: 1 Date: Thu, 22 Dec 2011 12:22:55 -0800 From: Michael Collins Subject: Re: [Freeswitch-users] Number of calls To: FreeSWITCH Users Help Message-ID: Content-Type: text/plain; charset="iso-8859-1" That looks like a decent 4-core processor. You can probably do several hundred concurrent calls w/o much issue. Of course, "it depends" on what other stuff you have going on inside this server. :) -MC On Wed, Dec 21, 2011 at 11:44 PM, Voip service wrote: > Hi, > > I am new in voip, how many calls can one freeswitch box handle with 30 % > of trans-coded calls and system configuration as > 8GB RAM > X3430 Xeon Processor, 2.4GHz, 8M Cache, Turbo > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/56e6ac91/attachment-0001.html ------------------------------ Message: 2 Date: Thu, 22 Dec 2011 12:24:16 -0800 From: Michael Collins Subject: Re: [Freeswitch-users] Freeswitch Crashing again and again To: FreeSWITCH Users Help Message-ID: Content-Type: text/plain; charset="iso-8859-1" Are you able to reproduce this on the latest git head? If so, open a Jira ticket at jira.freeswitch.org. If it is truly urgent then you can get commercial/professional help by contacting consulting at freeswitch.org. -MC On Thu, Dec 22, 2011 at 12:46 AM, Faisal Rehman wrote: > *Dear FS users* > > Our freeswitch has been working stable for around 1.5 years. > > Currently we are getting Freeswitch crashing without any apperent reasons > with following coredump > > > Core was generated by `/usr/local/freeswitch/bin/freeswitch -nc'. > Program terminated with signal 11, Segmentation fault. > #0 0x0000003d922790d0 in strchr () from /lib64/libc.so.6 > (gdb) bt > #0 0x0000003d922790d0 in strchr () from /lib64/libc.so.6 > #1 0x00002b7677883008 in switch_ivr_originate (session=0x2aaad020cab8, > bleg=0x54091ce0, cause=0x54091cec, > bridgeto=0x2b7677950557 "true", timelimit_sec=60, table=0x0, > cid_name_override=0x0, cid_num_override=0x0, > caller_profile_override=0x0, ovars=0x0, flags=, > cancel_cause=0x0) > at ./src/include/switch_utils.h:429 > #2 0x00002aaac8cf68f8 in audio_bridge_function (session=0x2aaad020cab8, > data=0x2aaaac754480 "sofia/x8888/47756508496197 at 10.47.78.170:5654") > at > /usr/src/freeswitch/src/mod/applications/mod_dptools/mod_dptools.c:2683 > #3 0x00002b7677853952 in switch_core_session_exec > (session=0x2aaad020cab8, application_interface=0x2aaab00553b8, > arg=0x2aaad0006b28 "sofia/${flag_auth}") at > src/switch_core_session.c:2121 > #4 0x00002b7677853f7f in > switch_core_session_execute_application_get_flags (session=0x0, > app=0x2aaad0006b20 "bridge", > arg=0x2aaad0006b28 "sofia/${flag_auth}", flags=0x0) at > src/switch_core_session.c:2006 > #5 0x00002b7677856963 in switch_core_session_run (session=0x2aaad020cab8) > at src/switch_core_state_machine.c:181 > #6 0x00002b7677850d30 in switch_core_session_thread (thread= optimized out>, obj=0x2aaad020cab8) > at src/switch_core_session.c:1311 > #7 0x0000003d92a0673d in start_thread () from /lib64/libpthread.so.0 > #8 0x0000003d922d3d1d in clone () from /lib64/libc.so.6 > > > *Please help to fix this issue as it is very urgent.* > > > Thanks and Regards, > > > *Faisal* > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/aec1feea/attachment-0001.html ------------------------------ Message: 3 Date: Thu, 22 Dec 2011 12:41:19 -0800 (PST) From: Rodney Subject: [Freeswitch-users] Odp: transfer to conference from conference / caller controls To: "freeswitch-users at lists.freeswitch.org" Message-ID: <1324586479.30344.YahooMailNeo at web65305.mail.ac2.yahoo.com> Content-Type: text/plain; charset="utf-8" Mariusz, using your suggestion, i worked out a solution that works. although i have to drop the bleg with sched_hangup in order for it to work with my hangup back to ivr method. it still looks like crap in my cdr but I can deal with it for now since i am crazy for wanting to transfer from static conference right to dynamic :) but doing it with this loopback method seems to give me the caller controls back.? thanks for your help. ~rodney ?condition??? ? destination_number??? ? ^760$??? ? 1??? ? ? action??? ? answer??? ? ??? ? 3??? ? ? action??? ? set??? ? conference_user_list=|??? ? 10??? ? ? action??? ? play_and_get_digits??? ? 4 4 3 5000 # ask4digit.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+??? ? 20??? ? ? action??? ? phrase??? ? spell,${target_num}??? ? 30??? ? ? action??? ? sched_hangup??? ? +10 normal_clearing bleg??? ? 35??? ? ? action??? ? bridge??? ? loopback/app=conference:${target_num}-127.0.0.1 at dynamic_conf ? 4. Odp: transfer to conference from conference / caller controls ? ? ? (Mariusz Czulada) > Perhaps this will be useful for you: ? Dialplan: ? ??? ???? ... ??????? ???? ... ??????? ?? ? ... ??? ??? ???? ... ??????? ???? ... ??? ? conference.conf.xml: ? ??? ???? ... ????? ??? ??? ???? ... ????? ??? ??? ???? ... ????? ??? ??? ???? ... ????? ??? ? ? Dnia 20-12-2011 o godz. 18:21 Rodney napisa?(a): I am trying to transfer to a dynamic conference from within a static conference but when I do my callers loose control and can not use any dtmf caller control options and must hang up to get back. I tried using clear digit action but do not think this works on the "conf" realm. the dynamic conference works fine with all controls if i send them directly from the ivr but for what i am trying it is important to do this transfer from another conference. is there a method transfer and release all previous controls? i even tried using a difference conference profile so i transferred from static 501 at default to @dynamic to see if giving new controls would work, to no avail. is there a log or something i can post that will help you in understanding how this is happening? or is it even fixable? thanks rodney > > >?? >????? >????? >????? >????? >????? > >-???? >??????? >??????? ? (main ivr menu extension) >??????? > > > > > >condition??? ? destination_number??? ? ^759$??? ? 1??? ? >? action??? ? answer??? ? ??? ? 2??? ? >? action??? ? set??? ? conference_user_list=|??? ? 11??? ? >? action??? ? play_and_get_digits??? ? 4 4 3 5000 # askprivateroom.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+??? ? 15??? ? >? action??? ? phrase??? ? spell,${target_num}??? ? 16??? ? >? action??? ? conference??? ? ${target_num}-127.0.0.1 at default > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/fe3c91ad/attachment.html ------------------------------ _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org End of FreeSWITCH-users Digest, Vol 66, Issue 140 ************************************************* -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/6fc495ca/attachment-0001.html From msc at freeswitch.org Fri Dec 23 01:20:50 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Dec 2011 14:20:50 -0800 Subject: [Freeswitch-users] Excluding busy extensions from intercom In-Reply-To: <201112190217.11783.justlikeef@gmail.com> References: <201112190217.11783.justlikeef@gmail.com> Message-ID: What kind of phones are these? -MC On Sun, Dec 18, 2011 at 11:17 PM, Rob Hutton wrote: > ** > > I am trying to get a basic "All Page" setup working and have used the wiki > and example configs to put together the following dialplan: > > > > > > > > > > > data="conference_auto_outcall_caller_id_name=${effective_caller_id_name}"/> > > data="conference_auto_outcall_caller_id_number=${effective_caller_id_number}"/> > > > > > > data="conference_auto_outcall_prefix={sip_auto_answer=true}"/> > > > > > > > > > > > > > > > > > > > Everything works as expected, except that intercom forces any active calls > to hold, and likewise forces the originating call on hold. > > > Is there a way of excluding any active users? > > > I have tried both ${network_addr} and ${presence_id} in the > sip_exclude_contact variable, and neither causes the orginator to be > excluded. > > > Console log is here: > > http://pastebin.freeswitch.org/18022 > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/0104f26b/attachment.html From msc at freeswitch.org Fri Dec 23 01:33:50 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Dec 2011 14:33:50 -0800 Subject: [Freeswitch-users] Problems in using mod_xml_curl at user authentication. In-Reply-To: References: Message-ID: Turn on debug from fs_cli: xml_curl debug_on Get a debug trace of what's happening and post the console log and the reply xml file to pastebin.freeswitch.org. (Note that the xml_curl debug will give you the full path to the xml log file.) -MC On Sun, Dec 18, 2011 at 9:02 PM, rakib 0000 wrote: > Hello list, > > I'm a newbie FreeSwitch user, trying to learn how mod_xml_curl works. I've > done the basics things mentioned in FreeSwitch wiki pages about using > mod_xml_curl. Now my plan to use mod_xml_curl for user registration. So, > what I've done is - I wrote a server end application which reads POST > request from mod_xml_curl and tries to find whether it's a valid post > request (REGISTER) or not. If the request is a REGISTER request - then the > application extracts the user id, domain. Then, it queries the database and > extracts password and creates xml response. One of my sample response are > like below (the following response has been taken from /tmp): > > > >
> > > > > > > > > > > > > > > > >
> > > With above response, I didn't had any luck. It fails to register, with > following response: > > 2011-12-18 14:51:29.253474 [WARNING] sofia_reg.c:2446 Can't find user [ > 1100 at 192.168.0.108] > You must define a domain called '192.168.0.108' in your directory and add > a user with the id="1100" attribute > and you must configure your device to use the proper domain in it's > authentication credentials. > > To figure it out, I've tried in different ways too. Like, I've used the > above response patterns to create a user under conf/directory/default/ > directory, but didn't work. Then I tried the following xml response pattern > : > > > > > > > > > > > > > with the above xml I manage to create a user and authenticate perfectly. > But, I've modified my server end application to give a that kind of > response, but didn't work though. I've copied the server response (from the > content dumped under /tmp) to create an user under conf/directory/default/ > directory - but it works. So, my question is what am I missing here? > FreeSwitch tries to get the response from the server, it gets the response > too, but somehow not interpreting the response (or something else?) and > gives "can't find user" error. Any hint's , suggestion will be very > helpful. (note, I'm not using any webserver here.) > > Thanks in advance, > Rakib > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/b13c334c/attachment.html From msc at freeswitch.org Fri Dec 23 01:39:07 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Dec 2011 14:39:07 -0800 Subject: [Freeswitch-users] Paging and Intercom In-Reply-To: <201112192335.04932.justlikeef@gmail.com> References: <201112192335.04932.justlikeef@gmail.com> Message-ID: I don't have GS phone but I have not experienced these symptoms on Polycom, Snom, Aastra or Cisco SPA5xx phones. -MC On Mon, Dec 19, 2011 at 8:35 PM, Rob Hutton wrote: > ** > > How is everyone doing intercom and paging? We are having problems with > Grandstream phones where they put an active call on hold to answer the > second call if they see the "auto answer" headers. Do all brands behave > this way or is this something unique to Grandstream? > > > I was thinking earlyier about handling this from the switch, but since > current phones allow you to log in to multiple accounts even across > multiple switches, it would be impossible to do this from the switch > level... > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/e27d89e2/attachment.html From msc at freeswitch.org Fri Dec 23 01:39:46 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Dec 2011 14:39:46 -0800 Subject: [Freeswitch-users] Extending the field size of core's db_data table? In-Reply-To: References: Message-ID: I would send this to the -dev list. -MC On Mon, Dec 19, 2011 at 10:35 PM, Yehavi Bourvine wrote: > Hello, > > We use the DB api for storing and caching various data. We need to store > data that is longer than 255 characters, which is the current limit on the > field size there. > > The backend for this API is MySQL (via ODBC). Can I just increase the > field size in MySQL, or is there some dependency in FreeSwitch on this size? > > Thanks, __Yehavi: > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/5583495d/attachment-0001.html From msc at freeswitch.org Fri Dec 23 01:42:29 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Dec 2011 14:42:29 -0800 Subject: [Freeswitch-users] Problems with polycom registering In-Reply-To: References: Message-ID: What's your topology here? Is FS on public IP or behind NAT? Are the Polys on same network as FS? If not, are they behind NAT? -MC On Tue, Dec 20, 2011 at 10:03 AM, lloyd thomas wrote: > I had what I thought was the perfect set up on my FS box. Until I changed > internet provider (BT) > All was fine for a week but all of a sudden some of my Polycom phones > won't register. > The polycom phones are behind a NAT and my FS box is on a static IP > address. > I am using a multi company setup > > One of my polycom phones seems to register OK > > Using 'sofia profile xxxxxxxx siptrace on' I can see that one of my > polycom phones will not register because FS is trying to send replies to > the private IP address (192.168.101.16) > Any Ideas? Am I missing something in one of the profiles > > Good one > ------------------------------------------------------------------------ > send 702 bytes to udp/[87.194.242.110]:65004 at 17:49:32.226606: > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 87.194.242.110:65004;branch=z9hG4bK15c264eE5B5988B > From: "Suze" ;tag=CA76AF43-ED2A5944 > To: ;tag=7Uy88X4pc8yHc > Call-ID: 918d5430-ffb07455-eb2121fe at 87.194.242.110 > CSeq: 888 REGISTER > Contact: ;methods="INVITE, ACK, BYE, > CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, > REFER";expires=30 > Date: Tue, 20 Dec 2011 17:49:32 GMT > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2c009dd 2011-03-15 > 14-29-04 -0500 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > Content-Length: 0 > > > ------------------------------------------------------------------------- > > Bad one > ------------------------------------------------------------------------ > send 680 bytes to udp/[81.137.114.169]:5060 at 17:50:17.659167: > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.101.16:5060 > ;branch=z9hG4bK2b8f42dA63D9E42;received=81.137.154.169 > From: "Lloyd" ;tag=621EA6CE-AF1D573 > To: ;tag=aQaKeFQ13219e > Call-ID: 52e98627-d24590dc-528d7361 at 192.168.101.16 > CSeq: 1 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2c009dd 2011-03-15 > 14-29-04 -0500 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="phisys.tele.phi.co.uk", > nonce="1319a358-2b33-11e1-b5e4-dd1099e4b2d2", algorithm=MD5, qop="auth" > 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/20111222/342b25d3/attachment.html From msc at freeswitch.org Fri Dec 23 02:07:18 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Dec 2011 15:07:18 -0800 Subject: [Freeswitch-users] Some questions regarding IPs and gateway In-Reply-To: References: <9a4d8ee96b56592ac4250c9805cfb1a7.squirrel@fulvetta.riseup.net> <9d0848ccace3e1d34699e53d24f49e37.squirrel@fulvetta.riseup.net> Message-ID: On Thu, Dec 22, 2011 at 12:48 PM, wrote: > Hello, > > sorry for all the confusion I did: > > > On #1 - is that IP address for a NIC on the FreeSWITCH server itself or > do > > you have a firewall and have FreeSWITCH sitting behind it? I think if you > > give us the specific details about your network layout that it would help > > us help you. > > Its a IP associated to one nic of the server. No firewall involved. I've > got five nics, with five nets associated: > > - eth0 -> 192.168.0.X [using this for cluster-communication] > - eth1 -> 192.168.2.X [using this for cluster-communication] > - eth2 -> static ip, rechable from the internet > - eth3 -> 192.168.1.X [using this for the phones] > - eth4 -> private MPLS net to my provider, using this for the SIP trunk > > I would like exclude eth0/eth1/eth2 from the configs. I found the acl.conf > where I'm able to configure netmasks and stuff like this. I will try this, > hope this is correct. Also I will bind FS to 192.168.1.X trough the local > ip setting in vars.xml. Comments? > You want to set the IP address in the SIP profile. Remember, FreeSWITCH can use multiple IP addresses and multiple ports, so it's not like Asterisk at all in that respect. You control which IP/Port each SIP profile uses. If you tell the SIP profile explicitly to use a specific IP address then that's what it will use and will ignore the other NICs. Note: You can also tell mod_event_socket to listen on a specific IP/Port that is different than what your SIP profiles are listening on. -MC > > For #3 - the -nonat argument is correct for disabling FreeSWITCH's > > "autonat" stuff. Also, the SIP profiles have NAT settings in them that > can > > be adjusted. There are two flavors of NAT settings: those for when > > FreeSWITCH itself is behind NAT and those for when the clients connecting > > to FreeSWITCH are behind NAT. If you want an example of a "plain" SIP > > profile then look at conf/sip_profiles/external.xml - it's pretty basic > > and > > assumes that FreeSWITCH itself is not behind NAT. > > Allright. > > Thanks! > > Georg > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/f7da25f1/attachment.html From georg at riseup.net Fri Dec 23 02:08:26 2011 From: georg at riseup.net (georg at riseup.net) Date: Fri, 23 Dec 2011 00:08:26 +0100 Subject: [Freeswitch-users] Some questions regarding IPs and gateway In-Reply-To: References: <9a4d8ee96b56592ac4250c9805cfb1a7.squirrel@fulvetta.riseup.net> <9d0848ccace3e1d34699e53d24f49e37.squirrel@fulvetta.riseup.net> Message-ID: <6057682159298280544a4c16e4430721.squirrel@fulvetta.riseup.net> > Its a IP associated to one nic of the server. No firewall involved. I've > got five nics, with five nets associated: > > - eth0 -> 192.168.0.X [using this for cluster-communication] > - eth1 -> 192.168.2.X [using this for cluster-communication] > - eth2 -> static ip, rechable from the internet > - eth3 -> 192.168.1.X [using this for the phones] > - eth4 -> private MPLS net to my provider, using this for the SIP trunk > > I would like exclude eth0/eth1/eth2 from the configs. I found the acl.conf > where I'm able to configure netmasks and stuff like this. I will try this, > hope this is correct. Also I will bind FS to 192.168.1.X trough the local > ip setting in vars.xml. Comments? I used in vars.xml to achieve this. Thanks for all the hints, Georg From georg at riseup.net Fri Dec 23 02:19:02 2011 From: georg at riseup.net (georg at riseup.net) Date: Fri, 23 Dec 2011 00:19:02 +0100 Subject: [Freeswitch-users] Some questions regarding IPs and gateway In-Reply-To: References: <9a4d8ee96b56592ac4250c9805cfb1a7.squirrel@fulvetta.riseup.net> <9d0848ccace3e1d34699e53d24f49e37.squirrel@fulvetta.riseup.net> Message-ID: <68a7b9ccccb91c509337516e0e1b33b3.squirrel@fulvetta.riseup.net> > You want to set the IP address in the SIP profile. Remember, FreeSWITCH > can > use multiple IP addresses and multiple ports, so it's not like Asterisk at > all in that respect. You control which IP/Port each SIP profile uses. If > you tell the SIP profile explicitly to use a specific IP address then > that's what it will use and will ignore the other NICs. I achieved this now trough another way, see my other mail for this. I this "a no-go"? I already set the ips in my internal and my external profile, however, FS was still binding itself to the wan-ip (which is the way it normally works, if I'm right?). My plan was to change this and exclude especially the wan-ip. Georg From msc at freeswitch.org Fri Dec 23 02:20:32 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Dec 2011 15:20:32 -0800 Subject: [Freeswitch-users] Ringback on PRI line In-Reply-To: References: Message-ID: This is probably a PRI config issue. Doublecheck the PRI protocol settings with your provider. Also, get a d-channel trace and drop it on pastebin. (see the freetdm wiki page for d-chan capture instructions.) I'm sure Moy or one of the guys here will be happy to take a look. -MC On Thu, Dec 22, 2011 at 12:06 AM, Ryan V wrote: > Hi, > > We are running freeswitch with Sangoma PRI and Analog cards. There is no > ring back for calls coming in on PRI line. Calls coming in on FXO lines are > fine. Calling party gets ring back. > > I have added following to file conf/dialplan/public/00_inbound_did.xml > > > data="ringback=%(2000,4000,440.0,480.0)"/> > > Any suggestions to resolve this problem? > > Thanks, > > Venkatesh 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/20111222/0b18df82/attachment-0001.html From msc at freeswitch.org Fri Dec 23 02:21:39 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Dec 2011 15:21:39 -0800 Subject: [Freeswitch-users] Some questions regarding IPs and gateway In-Reply-To: <68a7b9ccccb91c509337516e0e1b33b3.squirrel@fulvetta.riseup.net> References: <9a4d8ee96b56592ac4250c9805cfb1a7.squirrel@fulvetta.riseup.net> <9d0848ccace3e1d34699e53d24f49e37.squirrel@fulvetta.riseup.net> <68a7b9ccccb91c509337516e0e1b33b3.squirrel@fulvetta.riseup.net> Message-ID: The "domain" method you used is actually a good way to go if you're always wanting the same IP for all network comms. -MC On Thu, Dec 22, 2011 at 3:19 PM, wrote: > > You want to set the IP address in the SIP profile. Remember, FreeSWITCH > > can > > use multiple IP addresses and multiple ports, so it's not like Asterisk > at > > all in that respect. You control which IP/Port each SIP profile uses. If > > you tell the SIP profile explicitly to use a specific IP address then > > that's what it will use and will ignore the other NICs. > > I achieved this now trough another way, see my other mail for this. > I this "a no-go"? > > I already set the ips in my internal and my external profile, however, FS > was still binding itself to the wan-ip (which is the way it normally > works, if I'm right?). My plan was to change this and exclude especially > the wan-ip. > > Georg > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/5548d326/attachment.html From notlikeme75 at yahoo.com Fri Dec 23 02:22:41 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Thu, 22 Dec 2011 15:22:41 -0800 (PST) Subject: [Freeswitch-users] transfer conference to conference In-Reply-To: References: Message-ID: <1324596161.92420.YahooMailNeo@web65308.mail.ac2.yahoo.com> actually i just tried this and even though it looked good on screen, i was still just in the static conference i tried to transfer from, so i am back to square one :( ?condition??? ? destination_number??? ? ^760$??? ? 1??? ? ? action??? ? answer??? ? ??? ? 3??? ? ? action??? ? set??? ? conference_user_list=|??? ? 10??? ? ? action??? ? play_and_get_digits??? ? 4 4 3 5000 # askprivateroom2.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+??? ? 20??? ? ? action??? ? phrase??? ? spell,${target_num}??? ? 30??? ? ? action??? ? sched_hangup??? ? +10 normal_clearing bleg??? ? 35??? ? ? action??? ? bridge??? ? loopback/app=conference:${target_num}-127.0.0.1 at dynamic_conf -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/4d8792cd/attachment.html From anthony.minessale at gmail.com Fri Dec 23 02:30:10 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Thu, 22 Dec 2011 17:30:10 -0600 Subject: [Freeswitch-users] Odp: transfer to conference from conference / caller controls In-Reply-To: <1324586479.30344.YahooMailNeo@web65305.mail.ac2.yahoo.com> References: <1324586479.30344.YahooMailNeo@web65305.mail.ac2.yahoo.com> Message-ID: You should be using transfer not execute_extension or you are going into the new conference mid digit parse and blocking in the first conference waiting to return. If you use the transfer app then you will exit the current conference and transfer to the new one. On Dec 22, 2011 2:42 PM, "Rodney" wrote: > Mariusz, using your suggestion, i worked out a solution that works. > although i have to drop the bleg with sched_hangup in order for it to work > with my hangup back to ivr method. it still looks like crap in my cdr but I > can deal with it for now since i am crazy for wanting to transfer from > static conference right to dynamic :) but doing it with this loopback > method seems to give me the caller controls back. thanks for your help. > ~rodney > > condition destination_number ^760$ 1 > action answer 3 > action set conference_user_list=| 10 > action play_and_get_digits 4 4 3 5000 # ask4digit.wav > ivr/ivr-that_was_an_invalid_entry.wav target_num \d+ 20 > action phrase spell,${target_num} 30 > action sched_hangup +10 normal_clearing bleg 35 > action bridge > loopback/app=conference:${target_num}-127.0.0.1 at dynamic_conf > > > 4. Odp: transfer to conference from conference / caller controls > (Mariusz Czulada) > > > > Perhaps this will be useful for you: > > Dialplan: > > > ... > > ... > > ... > > > > ... > data="loopback/app=conference:${conference_name}_sub at cx_sub"/> > ... > > > conference.conf.xml: > > > ... > > > > > ... > > > > > ... > > > > ... > > > > > Dnia 20-12-2011 o godz. 18:21 Rodney napisa?(a): > > I am trying to transfer to a dynamic conference from within a static > conference but when I do my callers loose control and can not use any dtmf > caller control options and must hang up to get back. I tried using clear > digit action but do not think this works on the "conf" realm. the dynamic > conference works fine with all controls if i send them directly from the > ivr but for what i am trying it is important to do this transfer from > another conference. is there a method transfer and release all previous > controls? i even tried using a difference conference profile so i > transferred from static 501 at default to @dynamic to see if giving > new controls would work, to no avail. is there a log or something i can > post that will help you in understanding how this is happening? or is it > even fixable? thanks rodney > > > > > > data="execute_extension ANNOUNCE_CONF_COUNT_PRIVATE XML default"/> > data="execute_extension 759 xml default"/> > > - > > data="execute_extension 401 XML default"/> (main ivr menu extension) > data="execute_extension ANNOUNCE_CONF_COUNT_PRIVATE XML default"/> > > > condition destination_number ^759$ 1 > action answer 2 > action set conference_user_list=| 11 > action play_and_get_digits 4 4 3 5000 # askprivateroom.wav > ivr/ivr-that_was_an_invalid_entry.wav target_num \d+ 15 > action phrase spell,${target_num} 16 > action conference ${target_num}-127.0.0.1 at default > > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/732641eb/attachment-0001.html From georg at riseup.net Fri Dec 23 02:31:14 2011 From: georg at riseup.net (georg at riseup.net) Date: Fri, 23 Dec 2011 00:31:14 +0100 Subject: [Freeswitch-users] Some questions regarding IPs and gateway In-Reply-To: References: <9a4d8ee96b56592ac4250c9805cfb1a7.squirrel@fulvetta.riseup.net> <9d0848ccace3e1d34699e53d24f49e37.squirrel@fulvetta.riseup.net> <68a7b9ccccb91c509337516e0e1b33b3.squirrel@fulvetta.riseup.net> Message-ID: <9f88efbe5e4e8c3b30cc9000634af3bb.squirrel@fulvetta.riseup.net> > The "domain" method you used is actually a good way to go if you're always > wanting the same IP for all network comms. Allright, thanks. Its just my second days with FS, but I'm already impressed. I like the possibilities of the ACLs/defining IPs/which IPs should FS listen to a lot. For all what I know, this is not possible in Asterisk (at least not in this manner), which is a bit stupid (and insecure?) in my eyes. Cheers, Georg From notlikeme75 at yahoo.com Fri Dec 23 02:38:04 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Thu, 22 Dec 2011 15:38:04 -0800 (PST) Subject: [Freeswitch-users] transfer to dynamic con from static In-Reply-To: References: Message-ID: <1324597084.71238.YahooMailNeo@web65303.mail.ac2.yahoo.com> everytime i use transfer from caller controls i get "destination out of order" on console. so i use the execute extension and get problems. is this correct? `rodney ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Thursday, December 22, 2011 6:30 PM Subject: FreeSWITCH-users Digest, Vol 66, Issue 145 ----- Forwarded Message ----- 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: Some questions regarding IPs and gateway (Michael Collins) ? 2. transfer conference to conference (Rodney) ? 3. Re: Odp: transfer to conference from conference / caller ? ? ? controls (Anthony Minessale) The "domain" method you used is actually a good way to go if you're always wanting the same IP for all network comms. -MC On Thu, Dec 22, 2011 at 3:19 PM, wrote: > You want to set the IP address in the SIP profile. Remember, FreeSWITCH >> can >> use multiple IP addresses and multiple ports, so it's not like Asterisk at >> all in that respect. You control which IP/Port each SIP profile uses. If >> you tell the SIP profile explicitly to use a specific IP address then >> that's what it will use and will ignore the other NICs. > >I achieved this now trough another way, see my other mail for this. >I this "a no-go"? > >I already set the ips in my internal and my external profile, however, FS >was still binding itself to the wan-ip (which is the way it normally >works, if I'm right?). My plan was to change this and exclude especially >the wan-ip. > >Georg > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > actually i just tried this and even though it looked good on screen, i was still just in the static conference i tried to transfer from, so i am back to square one :( ?condition??? ? destination_number??? ? ^760$??? ? 1??? ? ? action??? ? answer??? ? ??? ? 3??? ? ? action??? ? set??? ? conference_user_list=|??? ? 10??? ? ? action??? ? play_and_get_digits??? ? 4 4 3 5000 # askprivateroom2.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+??? ? 20??? ? ? action??? ? phrase??? ? spell,${target_num}??? ? 30??? ? ? action??? ? sched_hangup??? ? +10 normal_clearing bleg??? ? 35??? ? ? action??? ? bridge??? ? loopback/app=conference:${target_num}-127.0.0.1 at dynamic_conf You should be using transfer not execute_extension or you are going into the new conference mid digit parse and blocking in the first conference waiting to return.? If you use the transfer app then you will exit the current conference and transfer to the new one. On Dec 22, 2011 2:42 PM, "Rodney" wrote: Mariusz, using your suggestion, i worked out a solution that works. although i have to drop the bleg with sched_hangup in order for it to work with my hangup back to ivr method. it still looks like crap in my cdr but I can deal with it for now since i am crazy for wanting to transfer from static conference right to dynamic :) but doing it with this loopback method seems to give me the caller controls back.? thanks for your help. ~rodney > >?condition??? ? destination_number??? ? ^760$??? ? 1??? ? >? action??? ? answer??? ? ??? ? 3??? ? >? action??? ? set??? ? conference_user_list=|??? ? 10??? ? >? action??? ? play_and_get_digits??? ? 4 4 3 5000 # ask4digit.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+??? ? 20??? ? >? action??? ? phrase??? ? spell,${target_num}??? ? 30??? ? >? action??? ? sched_hangup??? ? +10 normal_clearing bleg??? ? 35??? ? >? action??? ? bridge??? ? loopback/app=conference:${target_num}-127.0.0.1 at dynamic_conf > > > >? 4. Odp: transfer to conference from conference / caller controls >? ? ? (Mariusz Czulada) > > >> > >Perhaps this will be useful for you: >? >Dialplan: >? >??? >???? ... >??????? >???? ... >??????? >?? ? ... >??? > >??? >???? ... >??????? >???? ... >??? >? >conference.conf.xml: >? >??? >???? ... >????? >??? > >??? >???? ... >????? >??? > >??? >???? ... >????? >??? >??? >???? ... >????? >??? >? >? >Dnia 20-12-2011 o godz. 18:21 Rodney napisa?(a): >I am trying to transfer to a dynamic conference from within a static conference but when I do my callers loose control and can not use any dtmf caller control options and must hang up to get back. I tried using clear digit action but do not think this works on the "conf" realm. the dynamic conference works fine with all controls if i send them directly from the ivr but for what i am trying it is important to do this transfer from another conference. is there a method transfer and release all previous controls? i even tried using a difference conference profile so i transferred from static 501 at default to @dynamic to see if giving new controls would work, to no avail. is there a log or something i can post that will help you in understanding how this is happening? or is it even fixable? thanks rodney >> >> >>?? >>????? >>????? >>????? >>????? >>????? >> >>-???? >>??????? >>??????? ? (main ivr menu extension) >>??????? >> >> >> >> >> >>condition??? ? destination_number??? ? ^759$??? ? 1??? ? >>? action??? ? answer??? ? ??? ? 2??? ? >>? action??? ? set??? ? conference_user_list=|??? ? 11??? ? >>? action??? ? play_and_get_digits??? ? 4 4 3 5000 # askprivateroom.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+??? ? 15??? ? >>? action??? ? phrase??? ? spell,${target_num}??? ? 16??? ? >>? action??? ? conference??? ? ${target_num}-127.0.0.1 at default >> > > > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/bc418104/attachment-0001.html From lloydie.t at gmail.com Fri Dec 23 02:48:15 2011 From: lloydie.t at gmail.com (lloyd thomas) Date: Thu, 22 Dec 2011 23:48:15 +0000 Subject: [Freeswitch-users] Problems with polycom registering In-Reply-To: References: Message-ID: The FS box is assigned public IP address via a router in bridge mode (DMZ plus). The router (Draytek) which the polycoms are connected to is also connected to the same router in bridge mode. The reason why I have it this way is because I have a remote polycom phone which seems to be registering OK. This set worked great on my old broadband provider and for around 5 days with the new provider. example- polycom phone(192.168.101.16) <--> Router(81.137.114.169) <--> (bridged modem) <--> INTERNET FS box (81.137.114.171) -----------------------------------------------------------(bridged modem) <--> INTERNET gateway = 81.137.114.174 subnet = 255.255.255.248 Lloydie T On 22 December 2011 22:42, Michael Collins wrote: > What's your topology here? Is FS on public IP or behind NAT? Are the Polys > on same network as FS? If not, are they behind NAT? > > -MC > > On Tue, Dec 20, 2011 at 10:03 AM, lloyd thomas wrote: > >> I had what I thought was the perfect set up on my FS box. Until I changed >> internet provider (BT) >> All was fine for a week but all of a sudden some of my Polycom phones >> won't register. >> The polycom phones are behind a NAT and my FS box is on a static IP >> address. >> I am using a multi company setup >> >> One of my polycom phones seems to register OK >> >> Using 'sofia profile xxxxxxxx siptrace on' I can see that one of my >> polycom phones will not register because FS is trying to send replies to >> the private IP address (192.168.101.16) >> Any Ideas? Am I missing something in one of the profiles >> >> Good one >> >> ------------------------------------------------------------------------ >> send 702 bytes to udp/[87.194.242.110]:65004 at 17:49:32.226606: >> >> ------------------------------------------------------------------------ >> SIP/2.0 200 OK >> Via: SIP/2.0/UDP 87.194.242.110:65004;branch=z9hG4bK15c264eE5B5988B >> From: "Suze" ;tag=CA76AF43-ED2A5944 >> To: ;tag=7Uy88X4pc8yHc >> Call-ID: 918d5430-ffb07455-eb2121fe at 87.194.242.110 >> CSeq: 888 REGISTER >> Contact: ;methods="INVITE, ACK, BYE, >> CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, >> REFER";expires=30 >> Date: Tue, 20 Dec 2011 17:49:32 GMT >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2c009dd 2011-03-15 >> 14-29-04 -0500 >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY >> Supported: timer, precondition, path, replaces >> Content-Length: 0 >> >> >> ------------------------------------------------------------------------- >> >> Bad one >> >> ------------------------------------------------------------------------ >> send 680 bytes to udp/[81.137.114.169]:5060 at 17:50:17.659167: >> >> ------------------------------------------------------------------------ >> SIP/2.0 401 Unauthorized >> Via: SIP/2.0/UDP 192.168.101.16:5060 >> ;branch=z9hG4bK2b8f42dA63D9E42;received=81.137.154.169 >> From: "Lloyd" ;tag=621EA6CE-AF1D573 >> To: ;tag=aQaKeFQ13219e >> Call-ID: 52e98627-d24590dc-528d7361 at 192.168.101.16 >> CSeq: 1 REGISTER >> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2c009dd 2011-03-15 >> 14-29-04 -0500 >> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >> REGISTER, REFER, NOTIFY >> Supported: timer, precondition, path, replaces >> WWW-Authenticate: Digest realm="phisys.tele.phi.co.uk", >> nonce="1319a358-2b33-11e1-b5e4-dd1099e4b2d2", algorithm=MD5, qop="auth" >> 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/8c53c982/attachment.html From notlikeme75 at yahoo.com Fri Dec 23 03:00:36 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Thu, 22 Dec 2011 16:00:36 -0800 (PST) Subject: [Freeswitch-users] transfer to dynamic conf from static In-Reply-To: <1324597084.71238.YahooMailNeo@web65303.mail.ac2.yahoo.com> References: <1324597084.71238.YahooMailNeo@web65303.mail.ac2.yahoo.com> Message-ID: <1324598436.33675.YahooMailNeo@web65307.mail.ac2.yahoo.com> sorry i am using: it gives me destination out of order and hangs up call ________________________________ From: Rodney To: "freeswitch-users at lists.freeswitch.org" Sent: Thursday, December 22, 2011 6:38 PM Subject: transfer to dynamic con from static everytime i use transfer from caller controls i get "destination out of order" on console. so i use the execute extension and get problems. is this correct? `rodney ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Thursday, December 22, 2011 6:30 PM Subject: FreeSWITCH-users Digest, Vol 66, Issue 145 ----- Forwarded Message ----- 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: Some questions regarding IPs and gateway (Michael Collins) ? 2. transfer conference to conference (Rodney) ? 3. Re: Odp: transfer to conference from conference / caller ? ? ? controls (Anthony Minessale) The "domain" method you used is actually a good way to go if you're always wanting the same IP for all network comms. -MC On Thu, Dec 22, 2011 at 3:19 PM, wrote: > You want to set the IP address in the SIP profile. Remember, FreeSWITCH >> can >> use multiple IP addresses and multiple ports, so it's not like Asterisk at >> all in that respect. You control which IP/Port each SIP profile uses. If >> you tell the SIP profile explicitly to use a specific IP address then >> that's what it will use and will ignore the other NICs. > >I achieved this now trough another way, see my other mail for this. >I this "a no-go"? > >I already set the ips in my internal and my external profile, however, FS >was still binding itself to the wan-ip (which is the way it normally >works, if I'm right?). My plan was to change this and exclude especially >the wan-ip. > >Georg > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > actually i just tried this and even though it looked good on screen, i was still just in the static conference i tried to transfer from, so i am back to square one :( ?condition??? ? destination_number??? ? ^760$??? ? 1??? ? ? action??? ? answer??? ? ??? ? 3??? ? ? action??? ? set??? ? conference_user_list=|??? ? 10??? ? ? action??? ? play_and_get_digits??? ? 4 4 3 5000 # askprivateroom2.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+??? ? 20??? ? ? action??? ? phrase??? ? spell,${target_num}??? ? 30??? ? ? action??? ? sched_hangup??? ? +10 normal_clearing bleg??? ? 35??? ? ? action??? ? bridge??? ? loopback/app=conference:${target_num}-127.0.0.1 at dynamic_conf You should be using transfer not execute_extension or you are going into the new conference mid digit parse and blocking in the first conference waiting to return.? If you use the transfer app then you will exit the current conference and transfer to the new one. On Dec 22, 2011 2:42 PM, "Rodney" wrote: Mariusz, using your suggestion, i worked out a solution that works. although i have to drop the bleg with sched_hangup in order for it to work with my hangup back to ivr method. it still looks like crap in my cdr but I can deal with it for now since i am crazy for wanting to transfer from static conference right to dynamic :) but doing it with this loopback method seems to give me the caller controls back.? thanks for your help. ~rodney > >?condition??? ? destination_number??? ? ^760$??? ? 1??? ? >? action??? ? answer??? ? ??? ? 3??? ? >? action??? ? set??? ? conference_user_list=|??? ? 10??? ? >? action??? ? play_and_get_digits??? ? 4 4 3 5000 # ask4digit.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+??? ? 20??? ? >? action??? ? phrase??? ? spell,${target_num}??? ? 30??? ? >? action??? ? sched_hangup??? ? +10 normal_clearing bleg??? ? 35??? ? >? action??? ? bridge??? ? loopback/app=conference:${target_num}-127.0.0.1 at dynamic_conf > > > >? 4. Odp: transfer to conference from conference / caller controls >? ? ? (Mariusz Czulada) > > >> > >Perhaps this will be useful for you: >? >Dialplan: >? >??? >???? ... >??????? >???? ... >??????? >?? ? ... >??? > >??? >???? ... >??????? >???? ... >??? >? >conference.conf.xml: >? >??? >???? ... >????? >??? > >??? >???? ... >????? >??? > >??? >???? ... >????? >??? >??? >???? ... >????? >??? >? >? >Dnia 20-12-2011 o godz. 18:21 Rodney napisa?(a): >I am trying to transfer to a dynamic conference from within a static conference but when I do my callers loose control and can not use any dtmf caller control options and must hang up to get back. I tried using clear digit action but do not think this works on the "conf" realm. the dynamic conference works fine with all controls if i send them directly from the ivr but for what i am trying it is important to do this transfer from another conference. is there a method transfer and release all previous controls? i even tried using a difference conference profile so i transferred from static 501 at default to @dynamic to see if giving new controls would work, to no avail. is there a log or something i can post that will help you in understanding how this is happening? or is it even fixable? thanks rodney >> >> >>?? >>????? >>????? >>????? >>????? >>????? >> >>-???? >>??????? >>??????? ? (main ivr menu extension) >>??????? >> >> >> >> >> >>condition??? ? destination_number??? ? ^759$??? ? 1??? ? >>? action??? ? answer??? ? ??? ? 2??? ? >>? action??? ? set??? ? conference_user_list=|??? ? 11??? ? >>? action??? ? play_and_get_digits??? ? 4 4 3 5000 # askprivateroom.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+??? ? 15??? ? >>? action??? ? phrase??? ? spell,${target_num}??? ? 16??? ? >>? action??? ? conference??? ? ${target_num}-127.0.0.1 at default >> > > > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/911b532c/attachment-0001.html From msc at freeswitch.org Fri Dec 23 03:25:17 2011 From: msc at freeswitch.org (Michael Collins) Date: Thu, 22 Dec 2011 16:25:17 -0800 Subject: [Freeswitch-users] transfer to dynamic con from static In-Reply-To: <1324597084.71238.YahooMailNeo@web65303.mail.ac2.yahoo.com> References: <1324597084.71238.YahooMailNeo@web65303.mail.ac2.yahoo.com> Message-ID: Pastebin the actual console output, including the stuff leading up to when you press #. -MC On Thu, Dec 22, 2011 at 3:38 PM, Rodney wrote: > everytime i use transfer from caller controls i get "destination out of > order" on console. so i use the execute extension and get problems. is this > correct? > > > > `rodney > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/c5536190/attachment.html From sherifomran2000 at yahoo.com Fri Dec 23 04:00:42 2011 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Thu, 22 Dec 2011 17:00:42 -0800 (PST) Subject: [Freeswitch-users] Radio stream binding ? In-Reply-To: Message-ID: <1324602042.19229.YahooMailClassic@web110803.mail.gq1.yahoo.com> hello guys, can any body help me to use bind_meta_app to connect a channel to a radio stream using gnuradio.org project? thanks in advance regards, Sherif -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/f88abcdf/attachment.html From timtitov at hotmail.com Fri Dec 23 04:28:29 2011 From: timtitov at hotmail.com (Tim Titov) Date: Thu, 22 Dec 2011 17:28:29 -0800 Subject: [Freeswitch-users] faxing and file permissions Message-ID: When faxing with freeswitch the directory and containing file need to be readable by "other". Otherwise, mod_spandsp complains about the file being inaccessible. You can also see on mod_spandsp wiki people doing chmod o+r $TMPFAX. Why can't this be run under the freeswitch user only, so that you can do basic access control? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/412adbc3/attachment.html From georg at riseup.net Fri Dec 23 05:10:40 2011 From: georg at riseup.net (georg at riseup.net) Date: Fri, 23 Dec 2011 03:10:40 +0100 Subject: [Freeswitch-users] int/ext dial tones not synchronous Message-ID: Hello all, I made the first experiences with the dialplan, and after some while, I really got calls from the outside routed to one of my sip phones. However, my phone is ringing something like 1-3 seconds earlier than the dial tone the caller is hearing. I googled around, but found no solution. As a start, I just used this extension: Thanks, Georg From sherifomran2000 at yahoo.com Fri Dec 23 05:46:57 2011 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Thu, 22 Dec 2011 18:46:57 -0800 (PST) Subject: [Freeswitch-users] Radio stream binding ? In-Reply-To: <1324602042.19229.YahooMailClassic@web110803.mail.gq1.yahoo.com> Message-ID: <1324608417.85995.YahooMailClassic@web110802.mail.gq1.yahoo.com> Hello all, does mod_scout play asf streams? thanks --- On Fri, 12/23/11, Sherif Omran wrote: From: Sherif Omran Subject: [Freeswitch-users] Radio stream binding ? To: "FreeSWITCH Users Help" Date: Friday, December 23, 2011, 3:00 AM hello guys, can any body help me to use bind_meta_app to connect a channel to a radio stream using gnuradio.org project? thanks in advance regards, Sherif -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111222/39435160/attachment.html From jaybinks at gmail.com Fri Dec 23 10:18:52 2011 From: jaybinks at gmail.com (jay binks) Date: Fri, 23 Dec 2011 17:18:52 +1000 Subject: [Freeswitch-users] Feedback In-Reply-To: <1324517684859-7117251.post@n2.nabble.com> References: <15500FF8-DE01-4383-AA80-00769AC3DB86@LYONL.COM> <004501cc98a8$698763b0$3c962b10$@google.hm> <1324517684859-7117251.post@n2.nabble.com> Message-ID: Agreed... put what you have currently on Git Hub so we can check it out... you sounded like you were already fairly advanced in your progress, so throwing something that mostly works on Git Hub should only take an hour. can wait to check it out... Jay On 22 December 2011 11:34, Jeff Lenk wrote: > Chad, > > Any update on this? Looking forward to seeing some of your contributions > and > perhaps being able to merge them into the mainline code. > > Sincerely, > Jeff > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Feedback-tp6941381p7117251.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 > -- Sincerely Jay -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/38121bef/attachment.html From ryan at kaevee.com Fri Dec 23 10:19:45 2011 From: ryan at kaevee.com (Ryan V) Date: Fri, 23 Dec 2011 12:49:45 +0530 Subject: [Freeswitch-users] Ringback on PRI line In-Reply-To: References: Message-ID: Adding following entries into inbound dial plan solved the problem. Thanks, Venkatesh K On Fri, Dec 23, 2011 at 4:50 AM, Michael Collins wrote: > This is probably a PRI config issue. Doublecheck the PRI protocol settings > with your provider. Also, get a d-channel trace and drop it on pastebin. > (see the freetdm wiki page for d-chan capture instructions.) > > I'm sure Moy or one of the guys here will be happy to take a look. > > -MC > > On Thu, Dec 22, 2011 at 12:06 AM, Ryan V wrote: > >> Hi, >> >> We are running freeswitch with Sangoma PRI and Analog cards. There is no >> ring back for calls coming in on PRI line. Calls coming in on FXO lines are >> fine. Calling party gets ring back. >> >> I have added following to file conf/dialplan/public/00_inbound_did.xml >> >> >> > data="ringback=%(2000,4000,440.0,480.0)"/> >> >> Any suggestions to resolve this problem? >> >> Thanks, >> >> Venkatesh 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 >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/cad97844/attachment-0001.html From ryan at kaevee.com Fri Dec 23 10:22:14 2011 From: ryan at kaevee.com (Ryan V) Date: Fri, 23 Dec 2011 12:52:14 +0530 Subject: [Freeswitch-users] Restrict outbound calls based on extension Message-ID: Hello, I want to allow outbound calls to a limited number of users. I have gone through toll definitions in dial plan. But, I am struggling to find a way to achieve that. Please help. Thanks, Ryan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/6ba735a5/attachment.html From avi at avimarcus.net Fri Dec 23 10:29:03 2011 From: avi at avimarcus.net (Avi Marcus) Date: Fri, 23 Dec 2011 09:29:03 +0200 Subject: [Freeswitch-users] Restrict outbound calls based on extension In-Reply-To: References: Message-ID: In brief: 1 way would be to store a variable in each directory user with what type of calls are allowed. Domestic, International, etc. e.g. "domestic,international" Then in your extension that actually routes the outbound calls, you would also check for the condition that this variable you set has the permission in it's list. Sorry for not showing you the actual code.. -Avi On Fri, Dec 23, 2011 at 9:22 AM, Ryan V wrote: > Hello, > > I want to allow outbound calls to a limited number of users. I have gone > through toll definitions in dial plan. But, I am struggling to find a way > to achieve that. Please help. > > Thanks, > > 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/20111223/357daea3/attachment.html From justlikeef at gmail.com Fri Dec 23 10:41:17 2011 From: justlikeef at gmail.com (Rob Hutton) Date: Fri, 23 Dec 2011 02:41:17 -0500 Subject: [Freeswitch-users] Excluding busy extensions from intercom In-Reply-To: References: <201112190217.11783.justlikeef@gmail.com> Message-ID: <201112230241.18260.justlikeef@gmail.com> Grandstream On Thursday 22 December 2011 17:20:50 Michael Collins wrote: > What kind of phones are these? > -MC > > On Sun, Dec 18, 2011 at 11:17 PM, Rob Hutton wrote: > > > ** > > > > I am trying to get a basic "All Page" setup working and have used the wiki > > and example configs to put together the following dialplan: > > > > > > > > > > > > > > > > > > > > > > > data="conference_auto_outcall_caller_id_name=${effective_caller_id_name}"/> > > > > > data="conference_auto_outcall_caller_id_number=${effective_caller_id_number}"/> > > > > > > > > > > > > > data="conference_auto_outcall_prefix={sip_auto_answer=true}"/> > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > Everything works as expected, except that intercom forces any active calls > > to hold, and likewise forces the originating call on hold. > > > > > > Is there a way of excluding any active users? > > > > > > I have tried both ${network_addr} and ${presence_id} in the > > sip_exclude_contact variable, and neither causes the orginator to be > > excluded. > > > > > > Console log is here: > > > > http://pastebin.freeswitch.org/18022 > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/39ff5d82/attachment-0001.html From justlikeef at gmail.com Fri Dec 23 10:43:31 2011 From: justlikeef at gmail.com (Rob Hutton) Date: Fri, 23 Dec 2011 02:43:31 -0500 Subject: [Freeswitch-users] Paging and Intercom In-Reply-To: References: <201112192335.04932.justlikeef@gmail.com> Message-ID: <201112230243.32117.justlikeef@gmail.com> I am trying to get them to address this because that is kind of what I thought, also. Their various models behave differently depending on what code base they are running, also, so it is not consistent within the brand... On Thursday 22 December 2011 17:39:07 Michael Collins wrote: > I don't have GS phone but I have not experienced these symptoms on Polycom, > Snom, Aastra or Cisco SPA5xx phones. > > -MC > > On Mon, Dec 19, 2011 at 8:35 PM, Rob Hutton wrote: > > > ** > > > > How is everyone doing intercom and paging? We are having problems with > > Grandstream phones where they put an active call on hold to answer the > > second call if they see the "auto answer" headers. Do all brands behave > > this way or is this something unique to Grandstream? > > > > > > I was thinking earlyier about handling this from the switch, but since > > current phones allow you to log in to multiple accounts even across > > multiple switches, it would be impossible to do this from the switch > > level... > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/94dc54cd/attachment.html From ryan at kaevee.com Fri Dec 23 11:43:54 2011 From: ryan at kaevee.com (Ryan V) Date: Fri, 23 Dec 2011 14:13:54 +0530 Subject: [Freeswitch-users] Restrict outbound calls based on extension In-Reply-To: References: Message-ID: Thanks. I managed to restrict by doing this. I added a variable into the directory entry for user like and restricted in outgoing dial plan with Probably I should add an entry in wiki for people like me. Thanks again, Venkatesh K On Fri, Dec 23, 2011 at 12:59 PM, Avi Marcus wrote: > In brief: > 1 way would be to store a variable in each directory user with what type > of calls are allowed. Domestic, International, etc. e.g. > "domestic,international" > Then in your extension that actually routes the outbound calls, you would > also check for the condition that this variable you set has the permission > in it's list. > > Sorry for not showing you the actual code.. > -Avi > > > On Fri, Dec 23, 2011 at 9:22 AM, Ryan V wrote: > >> Hello, >> >> I want to allow outbound calls to a limited number of users. I have gone >> through toll definitions in dial plan. But, I am struggling to find a way >> to achieve that. Please help. >> >> Thanks, >> >> 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/20111223/4fc187f2/attachment.html From sherifomran2000 at yahoo.com Fri Dec 23 12:44:29 2011 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Fri, 23 Dec 2011 01:44:29 -0800 (PST) Subject: [Freeswitch-users] python does not support threading? In-Reply-To: <201112230243.32117.justlikeef@gmail.com> Message-ID: <1324633469.18009.YahooMailClassic@web110803.mail.gq1.yahoo.com> Hi My Python does not support threading, so compiling mod_python does not complete. I ve the following packages installed Installed Packages python.x86_64?????????????????????????????????????? 2.6.6-29.el6????????????????????????????? @base??? python-devel.x86_64???????????????????????????????? 2.6.6-29.el6????????????????????????????? @base??? python-ethtool.x86_64?????????????????????????????? 0.6-1.el6???????????????????????????????? @base??? python-iniparse.noarch????????????????????????????? 0.3.1-2.1.el6???????????????????????????? installed python-iwlib.x86_64???????????????????????????????? 0.1-1.2.el6?????????????????????????????? @base??? python-libs.x86_64????????????????????????????????? 2.6.6-29.el6????????????????????????????? @base??? python-pycurl.x86_64??????????????????????????????? 7.19.0-8.el6????????????????????????????? installed python-urlgrabber.noarch??????????????????????????? 3.9.1-8.el6?????????????????????????????? installed any idea thank you -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/57c60b21/attachment-0001.html From shaheryarkh at googlemail.com Fri Dec 23 14:55:31 2011 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Fri, 23 Dec 2011 16:55:31 +0500 Subject: [Freeswitch-users] Ringback on PRI line In-Reply-To: References: Message-ID: Here is the reason why you need this, http://wiki.sangoma.com/nbe-faq#no_ring Thank you. On Fri, Dec 23, 2011 at 12:19 PM, Ryan V wrote: > Adding following entries into inbound dial plan solved the problem. > > > data="ringback=data=ringback=${in-ring}"/> > > > Thanks, > > Venkatesh K > > > On Fri, Dec 23, 2011 at 4:50 AM, Michael Collins wrote: > >> This is probably a PRI config issue. Doublecheck the PRI protocol >> settings with your provider. Also, get a d-channel trace and drop it on >> pastebin. (see the freetdm wiki page for d-chan capture instructions.) >> >> I'm sure Moy or one of the guys here will be happy to take a look. >> >> -MC >> >> On Thu, Dec 22, 2011 at 12:06 AM, Ryan V wrote: >> >>> Hi, >>> >>> We are running freeswitch with Sangoma PRI and Analog cards. There is no >>> ring back for calls coming in on PRI line. Calls coming in on FXO lines are >>> fine. Calling party gets ring back. >>> >>> I have added following to file conf/dialplan/public/00_inbound_did.xml >>> >>> >>> >> data="ringback=%(2000,4000,440.0,480.0)"/> >>> >>> Any suggestions to resolve this problem? >>> >>> Thanks, >>> >>> Venkatesh 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20111223/94e7b0f3/attachment.html From notlikeme75 at yahoo.com Fri Dec 23 14:59:39 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Fri, 23 Dec 2011 03:59:39 -0800 (PST) Subject: [Freeswitch-users] FreeSWITCH-users Digest, Vol 66, Issue 148 In-Reply-To: References: Message-ID: <1324641579.38168.YahooMailNeo@web65316.mail.ac2.yahoo.com> MC- Here is the pastebin: http://pastebin.freeswitch.org/18064 I created small log and recreated my problem. 1. at main ivr, pressed 7, then at conf menu went to static 509 (ivr option 9), and then pressed # to get transfered to 759 which is my dynamic extension Rodney ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Friday, December 23, 2011 2:20 AM Subject: FreeSWITCH-users Digest, Vol 66, Issue 148 ----- Forwarded Message ----- 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: transfer to dynamic con from static (Michael Collins) ? 2. Radio stream binding ? (Sherif Omran) ? 3. faxing and file permissions (Tim Titov) ? 4. int/ext dial tones not synchronous (georg at riseup.net) ? 5. Re: Radio stream binding ? (Sherif Omran) ? 6. Re: Feedback (jay binks) ? 7. Re: Ringback on PRI line (Ryan V) Pastebin the actual console output, including the stuff leading up to when you press #. -MC On Thu, Dec 22, 2011 at 3:38 PM, Rodney wrote: everytime i use transfer from caller controls i get "destination out of order" on console. so i use the execute extension and get problems. is this correct? > > > > > >`rodney > > > hello guys, can any body help me to use bind_meta_app to connect a channel to a radio stream using gnuradio.org project? thanks in advance regards, Sherif When faxing with freeswitch the directory and containing file need to be readable by "other". Otherwise, mod_spandsp complains about the file being inaccessible. You can also see on mod_spandsp wiki people doing?chmod o+r $TMPFAX. Why can't this be run under the freeswitch user only, so that you can do basic access control? Hello all, I made the first experiences with the dialplan, and after some while, I really got calls from the outside routed to one of my sip phones. However, my phone is ringing something like 1-3 seconds earlier than the dial tone the caller is hearing. I googled around, but found no solution. As a start, I just used this extension: ? ? ? ? ? ? ? Thanks, Georg Hello all, does mod_scout play asf streams? thanks --- On Fri, 12/23/11, Sherif Omran wrote: >From: Sherif Omran >Subject: [Freeswitch-users] Radio stream binding ? >To: "FreeSWITCH Users Help" >Date: Friday, December 23, 2011, 3:00 AM > > >hello guys, > >can any body help me to use > >bind_meta_app > >to connect a channel to a radio stream using gnuradio.org project? > >thanks in advance > >regards, >Sherif > >-----Inline Attachment Follows----- > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > Agreed... put what you have currently on Git Hub so we can check it out... you sounded like you were already fairly advanced in your progress, so throwing something that mostly works on Git Hub should only take an hour. can wait to check it out... Jay On 22 December 2011 11:34, Jeff Lenk wrote: Chad, > >Any update on this? Looking forward to seeing some of your contributions and >perhaps being able to merge them into the mainline code. > >Sincerely, >Jeff > >-- >View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Feedback-tp6941381p7117251.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 > -- Sincerely Jay Adding following entries into inbound dial plan solved the problem. ??????? ??????? ??????? Thanks, Venkatesh K On Fri, Dec 23, 2011 at 4:50 AM, Michael Collins wrote: This is probably a PRI config issue. Doublecheck the PRI protocol settings with your provider. Also, get a d-channel trace and drop it on pastebin. (see the freetdm wiki page for d-chan capture instructions.) > > >I'm sure Moy or one of the guys here will be happy to take a look. > > >-MC > > >On Thu, Dec 22, 2011 at 12:06 AM, Ryan V wrote: > >Hi, >> >>We are running freeswitch with Sangoma PRI and Analog cards. There is no ring back for calls coming in on PRI line. Calls coming in on FXO lines are fine. Calling party gets ring back. >> >>I have added following to file conf/dialplan/public/00_inbound_did.xml >> >>??????? >>??????? >> >>Any suggestions to resolve this problem? >> >>Thanks, >> >>Venkatesh 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 >> >> > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/aeecd2c8/attachment-0001.html From nasida at live.ru Fri Dec 23 15:43:35 2011 From: nasida at live.ru (Yuriy Nasida) Date: Fri, 23 Dec 2011 16:43:35 +0400 Subject: [Freeswitch-users] failover with mod_lcr. how to set failover reasons with | in bridge ? Message-ID: Hello list.I have same issue and wondering if nobody still have not solution. In general I have possibility not to use the bridge with "|" but this issue looks strange. Anyone? I am still wondering if there is a documented/undocumented reason why continue_on_fail does not work with piped failover? From: Ivan Kovacevic [mailto:ivank at rogers.com] Sent: June-09-11 4:38 PM To: 'freeswitch-users at lists.freeswitch.org' Subject: continue_on_fail failover with mod_lcr Hi Everyone, I would like to implement fail-over for my outbound gateway, but I would like to be able to pick certain cause codes which qualify to stop trying next gateway (specifically when I have bad number and I am getting sip:404 NO_ROUTE_DESTINATION, UNALLOCATED_NUMBER).I was using 1.0.6, but I moved to the newest git about month ago. After searching through lists and spending several hours playing with continue_to_fail, failure_causes and fail_on_single_reject I was able to make it work by specifying cause codes for which I want to fail-over and omitting ones that qualify to stop trying next gateway in variable continue_to_fail: So this setup is working for me and in the case I have bad number and x.x.x.x returns sip:404 (NO_ROUTE_DESTINATION or UNALLOCATED_NUMBER) it is not trying y.y.y.y gateway. However if I want to use "|" between my gateways - and the example below is not working. And no matter what x.x.x.x returns - it will try y.y.y.y and eventually z.z.z.z. Unfortunately, we have to use pipe for fail-over since we are using mod_lcr to choose between outbound gateways. Any suggestions? Thanks, Ivan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/62153d0d/attachment.html From B.Tietz at pinguin.ag Fri Dec 23 16:39:24 2011 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Fri, 23 Dec 2011 14:39:24 +0100 Subject: [Freeswitch-users] Failover with Postgresql Message-ID: <07BF4904977CC645B485E970424193AD0E690EDB7E@localhost> Hi, I try to make sofia recover with two servers. Setup is slightly done like described in Wiki. But if I try to make a sofia recover the call is not recovered. Here is the Error Message from CLI: 2011-12-23 14:34:42.256728 [WARNING] sofia_glue.c:5418 Invalid cdr data, call not recovered Both server can reach the pgsql-server via odbc. Data is written in sip_recovery. I think the coding fort he database is wrong. Can anyone tell me what would be the best encoding? UTF8 and ANSII is bad! regards, Benjamin -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/44c91e61/attachment.html From ga at steadfasttelecom.com Fri Dec 23 17:08:05 2011 From: ga at steadfasttelecom.com (Gilad Abada) Date: Fri, 23 Dec 2011 09:08:05 -0500 Subject: [Freeswitch-users] Failover with Postgresql In-Reply-To: <07BF4904977CC645B485E970424193AD0E690EDB7E@localhost> References: <07BF4904977CC645B485E970424193AD0E690EDB7E@localhost> Message-ID: Hi Benjamin Psql cuts off the characters in the sofia recover table. To fix this you need to add: "MaxLongVarcharSize = 65536" under your psql ANSI in /etc/odbcinst.ini Good luck Gill On Fri, Dec 23, 2011 at 8:39 AM, wrote: > Hi, > > > > I try to make sofia recover with two servers. Setup is slightly done like > described in Wiki. > > But if I try to make a sofia recover the call is not recovered. Here is the > Error Message from CLI: > > > > 2011-12-23 14:34:42.256728 [WARNING] sofia_glue.c:5418 Invalid cdr data, > call not recovered > > > > Both server can reach the pgsql-server via odbc. Data is written in > sip_recovery. > > > > I think the coding fort he database is wrong. Can anyone tell me what would > be the best encoding? UTF8 and ANSII is bad! > > > > regards, > > Benjamin > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Gilad Abada SteadFast Telecommunications, Inc. Call us to find out how much you can save with VoIP! V: 212.589.1001 F: 212.589.1011 For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises.?? Make your virtual office a reality.? Enjoy the freedom to travel while remaining connected to your office. From davidwaf at gmail.com Fri Dec 23 17:22:26 2011 From: davidwaf at gmail.com (David Wafula) Date: Fri, 23 Dec 2011 16:22:26 +0200 Subject: [Freeswitch-users] Rent freeswitch box for production Message-ID: Hi Team, Anyone recommend where i can rent box for hosting freeswitch to be used as a conference server. I dont want a virtual box please. Regards, -- David Wafula From B.Tietz at pinguin.ag Fri Dec 23 17:36:37 2011 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Fri, 23 Dec 2011 15:36:37 +0100 Subject: [Freeswitch-users] Failover with Postgresql In-Reply-To: References: <07BF4904977CC645B485E970424193AD0E690EDB7E@localhost> Message-ID: <07BF4904977CC645B485E970424193AD0E690EDBC2@localhost> Hi, cdr data still invalid. I have a Master-Master-MySQL-Setup over both machines where recovery works. Database has latin1 coding to. I just think MySQL is not that stable... That's why I'd like to try pgsql. this is my odbcinst.ini [PostgreSQLUnicode] Description = PostgreSQL ODBC driver (Unicode version) Driver = /usr/lib/odbc/psqlodbcw.so Setup = /usr/lib/odbc/libodbcpsqlS.so Debug = 0 CommLog = 1 UsageCount = 1 Threading = 0 MaxLongVarcharSize=65536 Here my odbc.ini [fs_psql] Description = PostgreSQLUnicode Driver = PostgreSQLUnicode Trace = No TraceFile = /tmp/psqlodbc.log Database = freeswitch Servername = 1.2.3.4 UserName = freeswitch Password = xxx Port = 5432 ReadOnly = Yes RowVersioning = No ShowSystemTables = No ShowOidColumn = No FakeOidIndex = No ConnSettings = pgsql-Database is LATIN1 coding VG, Benjamin T. -----Urspr?ngliche Nachricht----- Hi Benjamin Psql cuts off the characters in the sofia recover table. To fix this you need to add: "MaxLongVarcharSize = 65536" under your psql ANSI in /etc/odbcinst.ini Good luck Gill On Fri, Dec 23, 2011 at 8:39 AM, wrote: > Hi, > I try to make sofia recover with two servers. Setup is slightly done > like described in Wiki. > But if I try to make a sofia recover the call is not recovered. Here > is the Error Message from CLI: > 2011-12-23 14:34:42.256728 [WARNING] sofia_glue.c:5418 Invalid cdr > data, call not recovered > Both server can reach the pgsql-server via odbc. Data is written in > sip_recovery. > I think the coding fort he database is wrong. Can anyone tell me what > would be the best encoding? UTF8 and ANSII is bad! > regards, > > Benjamin > From ga at steadfasttelecom.com Fri Dec 23 17:41:59 2011 From: ga at steadfasttelecom.com (Gilad Abada) Date: Fri, 23 Dec 2011 09:41:59 -0500 Subject: [Freeswitch-users] Failover with Postgresql In-Reply-To: <07BF4904977CC645B485E970424193AD0E690EDBC2@localhost> References: <07BF4904977CC645B485E970424193AD0E690EDB7E@localhost> <07BF4904977CC645B485E970424193AD0E690EDBC2@localhost> Message-ID: These are my settings for postgresql and they work. /etc/odbc.ini [$YOUR_DSN_NAME]Description ? ? ? ? = PostgreSQL UnicodeDriver ? ? ? ? ? ? ?= PostgreSQL UnicodeTrace ? ? ? ? ? ? ? = NoTraceFile ? ? ? ? ? = /tmp/psqlodbc.logDatabase ? ? ? ? ? ?= $YOUR_DSN_NAMEServername ? ? ? ? ?= 127.0.0.1UserName ? ? ? ? ? ?= $YOUR_DB_USERNAMEPassword ? ? ? ? ? ?= $YOUR_DB_PASSWORDPort ? ? ?= 5432ReadOnly ? ? ? ? ? ?= YesRowVersioning ? ? ? = NoShowSystemTables ? ?= NoShowOidColumn ? ? ? = NoFakeOidIndex = NoConnSettings ? ? ? ?=ODBC /etc/odbcinst.ini[PostgreSQL ANSI]?Description ? ? ? ? ? ?= PostgreSQL ODBC driver (ANSI version)Driver ? ? ? ? ?= /usr/lib/odbc/psqlodbca.soSetup ? ? ? ? ? = /usr/lib/odbc/libodbcpsqlS.soDebug ? ? ? ? ? = 0CommLog ? ? ? ? = 1UsageCount ? ? ? ? ? ? ?= 1Threading = 0[PostgreSQL Unicode]Description ? ? ? ? ? ? = PostgreSQL ODBC driver (Unicode version)Driver ? ? ? ? ?= /usr/lib/odbc/psqlodbcw.soSetup ? ? ? ? ? = /usr/lib/odbc/libodbcpsqlS.soDebug ? ? ? ? ? = 0CommLog ? ? ? ? = 1UsageCount ? ? ? ? ? ? ?= 1Threading = 0MaxLongVarcharSize = 65536 On Fri, Dec 23, 2011 at 9:36 AM, wrote: > Hi, > > cdr data still invalid. I have a Master-Master-MySQL-Setup over both machines where recovery works. Database has latin1 coding to. I just think MySQL is not that stable... That's why I'd like to try pgsql. > > this is my odbcinst.ini > > [PostgreSQLUnicode] > Description ? ? = PostgreSQL ODBC driver (Unicode version) > Driver ? ? ?= /usr/lib/odbc/psqlodbcw.so > Setup ? ? ? = /usr/lib/odbc/libodbcpsqlS.so > Debug ? ? ? = 0 > CommLog ? ? = 1 > UsageCount ? ? ?= 1 > Threading = 0 > MaxLongVarcharSize=65536 > > Here my odbc.ini > > [fs_psql] > Description ? ? ? ? = PostgreSQLUnicode > Driver ? ? ? ? ? ? ?= PostgreSQLUnicode > Trace ? ? ? ? ? ? ? = No > TraceFile ? ? ? ? ? = /tmp/psqlodbc.log > Database ? ? ? ? ? ?= freeswitch > Servername ? ? ? ? ?= 1.2.3.4 > UserName ? ? ? ? ? ?= freeswitch > Password ? ? ? ? ? ?= ?xxx > Port ? ? ? ? ? ? ? ?= 5432 > ReadOnly ? ? ? ? ? ?= Yes > RowVersioning ? ? ? = No > ShowSystemTables ? ?= No > ShowOidColumn ? ? ? = No > FakeOidIndex ? ? ? ?= No > ConnSettings ? ? ? ?= > > pgsql-Database is LATIN1 coding > > VG, > Benjamin T. > > > -----Urspr?ngliche Nachricht----- > > Hi Benjamin > > Psql cuts off the characters in the sofia recover table. To fix this you need to add: "MaxLongVarcharSize = 65536" > under your psql ANSI in /etc/odbcinst.ini > > Good luck > Gill > > On Fri, Dec 23, 2011 at 8:39 AM, ? wrote: >> Hi, >> I try to make sofia recover with two servers. Setup is slightly done >> like described in Wiki. >> But if I try to make a sofia recover the call is not recovered. Here >> is the Error Message from CLI: >> 2011-12-23 14:34:42.256728 [WARNING] sofia_glue.c:5418 Invalid cdr >> data, call not recovered >> Both server can reach the pgsql-server via odbc. Data is written in >> sip_recovery. >> I think the coding fort he database is wrong. Can anyone tell me what >> would be the best encoding? UTF8 and ANSII is bad! >> regards, >> >> Benjamin >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Gilad Abada SteadFast Telecommunications, Inc. Call us to find out how much you can save with VoIP! V: 212.589.1001 F: 212.589.1011 For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises.?? Make your virtual office a reality.? Enjoy the freedom to travel while remaining connected to your office. From B.Tietz at pinguin.ag Fri Dec 23 17:52:45 2011 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Fri, 23 Dec 2011 15:52:45 +0100 Subject: [Freeswitch-users] Failover with Postgresql In-Reply-To: References: <07BF4904977CC645B485E970424193AD0E690EDB7E@localhost> <07BF4904977CC645B485E970424193AD0E690EDBC2@localhost> Message-ID: <07BF4904977CC645B485E970424193AD0E690EDBD0@localhost> Hi Gilad, and what is the coding of your database for freeswitch? VG, Benjamin T. Betreff: Re: [Freeswitch-users] Failover with Postgresql These are my settings for postgresql and they work. /etc/odbc.ini [$YOUR_DSN_NAME]Description ? ? ? ? = PostgreSQL UnicodeDriver ? ? ? ? ? ? ?= PostgreSQL UnicodeTrace ? ? ? ? ? ? ? = NoTraceFile ? ? ? ? ? = /tmp/psqlodbc.logDatabase ? ? ? ? ? ?= $YOUR_DSN_NAMEServername ? ? ? ? ?= 127.0.0.1UserName ? ? ? ? ? ?= $YOUR_DB_USERNAMEPassword ? ? ? ? ? ?= $YOUR_DB_PASSWORDPort ? ? ?= 5432ReadOnly ? ? ? ? ? ?= YesRowVersioning ? ? ? = NoShowSystemTables ? ?= NoShowOidColumn ? ? ? = NoFakeOidIndex = NoConnSettings ? ? ? ?=ODBC /etc/odbcinst.ini[PostgreSQL ANSI]?Description ? ? ? ? ? ?= PostgreSQL ODBC driver (ANSI version)Driver ? ? ? ? ?= /usr/lib/odbc/psqlodbca.soSetup ? ? ? ? ? = /usr/lib/odbc/libodbcpsqlS.soDebug ? ? ? ? ? = 0CommLog ? ? ? ? = 1UsageCount ? ? ? ? ? ? ?= 1Threading = 0[PostgreSQL Unicode]Description ? ? ? ? ? ? = PostgreSQL ODBC driver (Unicode version)Driver ? ? ? ? ?= /usr/lib/odbc/psqlodbcw.soSetup ? ? ? ? ? = /usr/lib/odbc/libodbcpsqlS.soDebug ? ? ? ? ? = 0CommLog ? ? ? ? = 1UsageCount ? ? ? ? ? ? ?= 1Threading = 0MaxLongVarcharSize = 65536 On Fri, Dec 23, 2011 at 9:36 AM, wrote: > Hi, > > cdr data still invalid. I have a Master-Master-MySQL-Setup over both machines where recovery works. Database has latin1 coding to. I just think MySQL is not that stable... That's why I'd like to try pgsql. > > this is my odbcinst.ini > > [PostgreSQLUnicode] > Description ? ? = PostgreSQL ODBC driver (Unicode version) Driver ? ? ? > = /usr/lib/odbc/psqlodbcw.so Setup ? ? ? = > /usr/lib/odbc/libodbcpsqlS.so Debug ? ? ? = 0 CommLog ? ? = 1 > UsageCount ? ? ?= 1 Threading = 0 > MaxLongVarcharSize=65536 > > Here my odbc.ini > > [fs_psql] > Description ? ? ? ? = PostgreSQLUnicode Driver ? ? ? ? ? ? ?= > PostgreSQLUnicode Trace ? ? ? ? ? ? ? = No TraceFile ? ? ? ? ? = > /tmp/psqlodbc.log Database ? ? ? ? ? ?= freeswitch Servername ? ? ? ? ? > = 1.2.3.4 UserName ? ? ? ? ? ?= freeswitch Password ? ? ? ? ? ?= ?xxx > Port ? ? ? ? ? ? ? ?= 5432 ReadOnly ? ? ? ? ? ?= Yes RowVersioning ? ? ? > = No ShowSystemTables ? ?= No ShowOidColumn ? ? ? = No FakeOidIndex ? ? ? ? > = No ConnSettings ? ? ? ?= > > pgsql-Database is LATIN1 coding > > VG, > Benjamin T. > > > -----Urspr?ngliche Nachricht----- > > Hi Benjamin > > Psql cuts off the characters in the sofia recover table. To fix this you need to add: "MaxLongVarcharSize = 65536" > under your psql ANSI in /etc/odbcinst.ini > > Good luck > Gill > > On Fri, Dec 23, 2011 at 8:39 AM, ? wrote: >> Hi, >> I try to make sofia recover with two servers. Setup is slightly done >> like described in Wiki. >> But if I try to make a sofia recover the call is not recovered. Here >> is the Error Message from CLI: >> 2011-12-23 14:34:42.256728 [WARNING] sofia_glue.c:5418 Invalid cdr >> data, call not recovered Both server can reach the pgsql-server via >> odbc. Data is written in sip_recovery. >> I think the coding fort he database is wrong. Can anyone tell me what >> would be the best encoding? UTF8 and ANSII is bad! >> regards, >> >> Benjamin >> > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users 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 -- Gilad Abada SteadFast Telecommunications, Inc. Call us to find out how much you can save with VoIP! V: 212.589.1001 F: 212.589.1011 For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises.?? Make your virtual office a reality.? Enjoy the freedom to travel while remaining connected to your office. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ga at steadfasttelecom.com Fri Dec 23 18:00:34 2011 From: ga at steadfasttelecom.com (Gilad Abada) Date: Fri, 23 Dec 2011 10:00:34 -0500 Subject: [Freeswitch-users] Failover with Postgresql In-Reply-To: <07BF4904977CC645B485E970424193AD0E690EDBD0@localhost> References: <07BF4904977CC645B485E970424193AD0E690EDB7E@localhost> <07BF4904977CC645B485E970424193AD0E690EDBC2@localhost> <07BF4904977CC645B485E970424193AD0E690EDBD0@localhost> Message-ID: What do you mean? I am using postgresql On Fri, Dec 23, 2011 at 9:52 AM, wrote: > Hi Gilad, > > and what is the coding of your database for freeswitch? > > VG, > Benjamin T. > > > Betreff: Re: [Freeswitch-users] Failover with Postgresql > > These are my settings for postgresql and they work. > > ?/etc/odbc.ini [$YOUR_DSN_NAME]Description ? ? ? ? = PostgreSQL UnicodeDriver ? ? ? ? ? ? ?= PostgreSQL UnicodeTrace ? ? ? ? ? ? ? = NoTraceFile ? ? ? ? ? = /tmp/psqlodbc.logDatabase ? ? ? ? ? ?= $YOUR_DSN_NAMEServername ? ? ? ? ?= 127.0.0.1UserName ? ? ? ? ? ?= $YOUR_DB_USERNAMEPassword ? ? ? ? ? ?= $YOUR_DB_PASSWORDPort > ? ? ?= 5432ReadOnly ? ? ? ? ? ?= YesRowVersioning ? ? ? = NoShowSystemTables ? ?= NoShowOidColumn ? ? ? = NoFakeOidIndex = NoConnSettings ? ? ? ?=ODBC /etc/odbcinst.ini[PostgreSQL ANSI]?Description ? ? ? ? ? ?= PostgreSQL ODBC driver (ANSI version)Driver ? ? ? ? ?= /usr/lib/odbc/psqlodbca.soSetup ? ? ? ? ? = /usr/lib/odbc/libodbcpsqlS.soDebug ? ? ? ? ? = 0CommLog ? ? ? ? = 1UsageCount ? ? ? ? ? ? ?= 1Threading = 0[PostgreSQL Unicode]Description ? ? ? ? ? ? = PostgreSQL ODBC driver (Unicode version)Driver ? ? ? ? ?= /usr/lib/odbc/psqlodbcw.soSetup ? ? ? ? ? = /usr/lib/odbc/libodbcpsqlS.soDebug ? ? ? ? ? = 0CommLog ? ? ? ? = 1UsageCount ? ? ? ? ? ? ?= 1Threading = 0MaxLongVarcharSize = 65536 On Fri, Dec 23, 2011 at 9:36 AM, ? wrote: >> Hi, >> >> cdr data still invalid. I have a Master-Master-MySQL-Setup over both machines where recovery works. Database has latin1 coding to. I just think MySQL is not that stable... That's why I'd like to try pgsql. >> >> this is my odbcinst.ini >> >> [PostgreSQLUnicode] >> Description ? ? = PostgreSQL ODBC driver (Unicode version) Driver >> = /usr/lib/odbc/psqlodbcw.so Setup ? ? ? = >> /usr/lib/odbc/libodbcpsqlS.so Debug ? ? ? = 0 CommLog ? ? = 1 >> UsageCount ? ? ?= 1 Threading = 0 >> MaxLongVarcharSize=65536 >> >> Here my odbc.ini >> >> [fs_psql] >> Description ? ? ? ? = PostgreSQLUnicode Driver ? ? ? ? ? ? ?= >> PostgreSQLUnicode Trace ? ? ? ? ? ? ? = No TraceFile ? ? ? ? ? = >> /tmp/psqlodbc.log Database ? ? ? ? ? ?= freeswitch Servername >> = 1.2.3.4 UserName ? ? ? ? ? ?= freeswitch Password ? ? ? ? ? ?= ?xxx >> Port ? ? ? ? ? ? ? ?= 5432 ReadOnly ? ? ? ? ? ?= Yes RowVersioning >> = No ShowSystemTables ? ?= No ShowOidColumn ? ? ? = No FakeOidIndex >> = No ConnSettings ? ? ? ?= >> >> pgsql-Database is LATIN1 coding >> >> VG, >> Benjamin T. >> >> >> -----Urspr?ngliche Nachricht----- >> >> Hi Benjamin >> >> Psql cuts off the characters in the sofia recover table. To fix this you need to add: "MaxLongVarcharSize = 65536" >> under your psql ANSI in /etc/odbcinst.ini >> >> Good luck >> Gill >> >> On Fri, Dec 23, 2011 at 8:39 AM, ? wrote: >>> Hi, >>> I try to make sofia recover with two servers. Setup is slightly done >>> like described in Wiki. >>> But if I try to make a sofia recover the call is not recovered. Here >>> is the Error Message from CLI: >>> 2011-12-23 14:34:42.256728 [WARNING] sofia_glue.c:5418 Invalid cdr >>> data, call not recovered Both server can reach the pgsql-server via >>> odbc. Data is written in sip_recovery. >>> I think the coding fort he database is wrong. Can anyone tell me what >>> would be the best encoding? UTF8 and ANSII is bad! >>> regards, >>> >>> Benjamin >>> >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users 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 > > > > -- > Gilad Abada > > SteadFast Telecommunications, Inc. > > Call us to find out how much you can save with VoIP! > > V: 212.589.1001 > F: 212.589.1011 > > > For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises.?? Make your virtual office a reality.? Enjoy the freedom to travel while remaining connected to your office. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Gilad Abada SteadFast Telecommunications, Inc. Call us to find out how much you can save with VoIP! V: 212.589.1001 F: 212.589.1011 For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises.?? Make your virtual office a reality.? Enjoy the freedom to travel while remaining connected to your office. From B.Tietz at pinguin.ag Fri Dec 23 18:07:22 2011 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Fri, 23 Dec 2011 16:07:22 +0100 Subject: [Freeswitch-users] Failover with Postgresql In-Reply-To: References: <07BF4904977CC645B485E970424193AD0E690EDB7E@localhost> <07BF4904977CC645B485E970424193AD0E690EDBC2@localhost> <07BF4904977CC645B485E970424193AD0E690EDBD0@localhost> Message-ID: <07BF4904977CC645B485E970424193AD0E690EDBDD@localhost> If you are in the psql-cli make '\l' for listing the databases and you will see the coding of the databases... VG, Benjamin T. Betreff: Re: [Freeswitch-users] Failover with Postgresql What do you mean? I am using postgresql On Fri, Dec 23, 2011 at 9:52 AM, wrote: > Hi Gilad, > > and what is the coding of your database for freeswitch? > > VG, > Benjamin T. > > > Betreff: Re: [Freeswitch-users] Failover with Postgresql > > These are my settings for postgresql and they work. > > ?/etc/odbc.ini [$YOUR_DSN_NAME]Description ? ? ? ? = PostgreSQL > UnicodeDriver ? ? ? ? ? ? ?= PostgreSQL UnicodeTrace ? ? ? ? ? ? ? = > NoTraceFile ? ? ? ? ? = /tmp/psqlodbc.logDatabase ? ? ? ? ? ?= > $YOUR_DSN_NAMEServername ? ? ? ? ?= 127.0.0.1UserName ? ? ? ? ? ?= > $YOUR_DB_USERNAMEPassword ? ? ? ? ? ?= $YOUR_DB_PASSWORDPort > ? ? ?= 5432ReadOnly ? ? ? ? ? ?= YesRowVersioning ? ? ? = NoShowSystemTables ? ?= NoShowOidColumn ? ? ? = NoFakeOidIndex = NoConnSettings ? ? ? ?=ODBC /etc/odbcinst.ini[PostgreSQL ANSI]?Description ? ? ? ? ? ?= PostgreSQL ODBC driver (ANSI version)Driver ? ? ? ? ?= /usr/lib/odbc/psqlodbca.soSetup ? ? ? ? ? = /usr/lib/odbc/libodbcpsqlS.soDebug ? ? ? ? ? = 0CommLog ? ? ? ? = 1UsageCount ? ? ? ? ? ? ?= 1Threading = 0[PostgreSQL Unicode]Description ? ? ? ? ? ? = PostgreSQL ODBC driver (Unicode version)Driver ? ? ? ? ?= /usr/lib/odbc/psqlodbcw.soSetup ? ? ? ? ? = /usr/lib/odbc/libodbcpsqlS.soDebug ? ? ? ? ? = 0CommLog ? ? ? ? = 1UsageCount ? ? ? ? ? ? ?= 1Threading = 0MaxLongVarcharSize = 65536 On Fri, Dec 23, 2011 at 9:36 AM, ? wrote: >> Hi, >> >> cdr data still invalid. I have a Master-Master-MySQL-Setup over both machines where recovery works. Database has latin1 coding to. I just think MySQL is not that stable... That's why I'd like to try pgsql. >> >> this is my odbcinst.ini >> >> [PostgreSQLUnicode] >> Description ? ? = PostgreSQL ODBC driver (Unicode version) Driver = >> /usr/lib/odbc/psqlodbcw.so Setup ? ? ? = >> /usr/lib/odbc/libodbcpsqlS.so Debug ? ? ? = 0 CommLog ? ? = 1 >> UsageCount ? ? ?= 1 Threading = 0 >> MaxLongVarcharSize=65536 >> >> Here my odbc.ini >> >> [fs_psql] >> Description ? ? ? ? = PostgreSQLUnicode Driver ? ? ? ? ? ? ?= >> PostgreSQLUnicode Trace ? ? ? ? ? ? ? = No TraceFile ? ? ? ? ? = >> /tmp/psqlodbc.log Database ? ? ? ? ? ?= freeswitch Servername = >> 1.2.3.4 UserName ? ? ? ? ? ?= freeswitch Password ? ? ? ? ? ?= ?xxx >> Port ? ? ? ? ? ? ? ?= 5432 ReadOnly ? ? ? ? ? ?= Yes RowVersioning = >> No ShowSystemTables ? ?= No ShowOidColumn ? ? ? = No FakeOidIndex = >> No ConnSettings ? ? ? ?= >> >> pgsql-Database is LATIN1 coding >> >> VG, >> Benjamin T. >> >> >> -----Urspr?ngliche Nachricht----- >> >> Hi Benjamin >> >> Psql cuts off the characters in the sofia recover table. To fix this you need to add: "MaxLongVarcharSize = 65536" >> under your psql ANSI in /etc/odbcinst.ini >> >> Good luck >> Gill >> >> On Fri, Dec 23, 2011 at 8:39 AM, ? wrote: >>> Hi, >>> I try to make sofia recover with two servers. Setup is slightly done >>> like described in Wiki. >>> But if I try to make a sofia recover the call is not recovered. Here >>> is the Error Message from CLI: >>> 2011-12-23 14:34:42.256728 [WARNING] sofia_glue.c:5418 Invalid cdr >>> data, call not recovered Both server can reach the pgsql-server via >>> odbc. Data is written in sip_recovery. >>> I think the coding fort he database is wrong. Can anyone tell me >>> what would be the best encoding? UTF8 and ANSII is bad! >>> regards, >>> >>> Benjamin >>> >> >> _____________________________________________________________________ >> _ ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> e >> rs >> http://www.freeswitch.org > > > > -- > Gilad Abada > > SteadFast Telecommunications, Inc. > > Call us to find out how much you can save with VoIP! > > V: 212.589.1001 > F: 212.589.1011 > > > For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises.?? Make your virtual office a reality.? Enjoy the freedom to travel while remaining connected to your office. > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users 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-use > rs > http://www.freeswitch.org -- Gilad Abada SteadFast Telecommunications, Inc. Call us to find out how much you can save with VoIP! V: 212.589.1001 F: 212.589.1011 For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises.?? Make your virtual office a reality.? Enjoy the freedom to travel while remaining connected to your office. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From ga at steadfasttelecom.com Fri Dec 23 18:22:41 2011 From: ga at steadfasttelecom.com (Gilad Abada) Date: Fri, 23 Dec 2011 10:22:41 -0500 Subject: [Freeswitch-users] Failover with Postgresql In-Reply-To: <07BF4904977CC645B485E970424193AD0E690EDBDD@localhost> References: <07BF4904977CC645B485E970424193AD0E690EDB7E@localhost> <07BF4904977CC645B485E970424193AD0E690EDBC2@localhost> <07BF4904977CC645B485E970424193AD0E690EDBD0@localhost> <07BF4904977CC645B485E970424193AD0E690EDBDD@localhost> Message-ID: my encoding is UTF8 On Fri, Dec 23, 2011 at 10:07 AM, wrote: > If you are in the psql-cli make '\l' for listing the databases and you will see the coding of the databases... > > VG, > Benjamin T. > > > Betreff: Re: [Freeswitch-users] Failover with Postgresql > > What do you mean? > I am using postgresql > > > > On Fri, Dec 23, 2011 at 9:52 AM, ? wrote: >> Hi Gilad, >> >> and what is the coding of your database for freeswitch? >> >> VG, >> Benjamin T. >> >> >> Betreff: Re: [Freeswitch-users] Failover with Postgresql >> >> These are my settings for postgresql and they work. >> >> ?/etc/odbc.ini [$YOUR_DSN_NAME]Description ? ? ? ? = PostgreSQL >> UnicodeDriver ? ? ? ? ? ? ?= PostgreSQL UnicodeTrace ? ? ? ? ? ? ? = >> NoTraceFile ? ? ? ? ? = /tmp/psqlodbc.logDatabase ? ? ? ? ? ?= >> $YOUR_DSN_NAMEServername ? ? ? ? ?= 127.0.0.1UserName ? ? ? ? ? ?= >> $YOUR_DB_USERNAMEPassword ? ? ? ? ? ?= $YOUR_DB_PASSWORDPort >> ? ? ?= 5432ReadOnly ? ? ? ? ? ?= YesRowVersioning ? ? ? = NoShowSystemTables ? ?= NoShowOidColumn ? ? ? = NoFakeOidIndex = NoConnSettings ? ? ? ?=ODBC /etc/odbcinst.ini[PostgreSQL ANSI]?Description ? ? ? ? ? ?= PostgreSQL ODBC driver (ANSI version)Driver ? ? ? ? ?= /usr/lib/odbc/psqlodbca.soSetup ? ? ? ? ? = /usr/lib/odbc/libodbcpsqlS.soDebug ? ? ? ? ? = 0CommLog ? ? ? ? = 1UsageCount ? ? ? ? ? ? ?= 1Threading = 0[PostgreSQL Unicode]Description ? ? ? ? ? ? = PostgreSQL ODBC driver (Unicode version)Driver ? ? ? ? ?= /usr/lib/odbc/psqlodbcw.soSetup ? ? ? ? ? = /usr/lib/odbc/libodbcpsqlS.soDebug ? ? ? ? ? = 0CommLog ? ? ? ? = 1UsageCount ? ? ? ? ? ? ?= 1Threading = 0MaxLongVarcharSize = 65536 On Fri, Dec 23, 2011 at 9:36 AM, ? wrote: >>> Hi, >>> >>> cdr data still invalid. I have a Master-Master-MySQL-Setup over both machines where recovery works. Database has latin1 coding to. I just think MySQL is not that stable... That's why I'd like to try pgsql. >>> >>> this is my odbcinst.ini >>> >>> [PostgreSQLUnicode] >>> Description ? ? = PostgreSQL ODBC driver (Unicode version) Driver = >>> /usr/lib/odbc/psqlodbcw.so Setup ? ? ? = >>> /usr/lib/odbc/libodbcpsqlS.so Debug ? ? ? = 0 CommLog ? ? = 1 >>> UsageCount ? ? ?= 1 Threading = 0 >>> MaxLongVarcharSize=65536 >>> >>> Here my odbc.ini >>> >>> [fs_psql] >>> Description ? ? ? ? = PostgreSQLUnicode Driver ? ? ? ? ? ? ?= >>> PostgreSQLUnicode Trace ? ? ? ? ? ? ? = No TraceFile ? ? ? ? ? = >>> /tmp/psqlodbc.log Database ? ? ? ? ? ?= freeswitch Servername = >>> 1.2.3.4 UserName ? ? ? ? ? ?= freeswitch Password ? ? ? ? ? ?= ?xxx >>> Port ? ? ? ? ? ? ? ?= 5432 ReadOnly ? ? ? ? ? ?= Yes RowVersioning = >>> No ShowSystemTables ? ?= No ShowOidColumn ? ? ? = No FakeOidIndex = >>> No ConnSettings ? ? ? ?= >>> >>> pgsql-Database is LATIN1 coding >>> >>> VG, >>> Benjamin T. >>> >>> >>> -----Urspr?ngliche Nachricht----- >>> >>> Hi Benjamin >>> >>> Psql cuts off the characters in the sofia recover table. To fix this you need to add: "MaxLongVarcharSize = 65536" >>> under your psql ANSI in /etc/odbcinst.ini >>> >>> Good luck >>> Gill >>> >>> On Fri, Dec 23, 2011 at 8:39 AM, ? wrote: >>>> Hi, >>>> I try to make sofia recover with two servers. Setup is slightly done >>>> like described in Wiki. >>>> But if I try to make a sofia recover the call is not recovered. Here >>>> is the Error Message from CLI: >>>> 2011-12-23 14:34:42.256728 [WARNING] sofia_glue.c:5418 Invalid cdr >>>> data, call not recovered Both server can reach the pgsql-server via >>>> odbc. Data is written in sip_recovery. >>>> I think the coding fort he database is wrong. Can anyone tell me >>>> what would be the best encoding? UTF8 and ANSII is bad! >>>> regards, >>>> >>>> Benjamin >>>> >>> >>> _____________________________________________________________________ >>> _ ___ Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >>> e >>> rs >>> http://www.freeswitch.org >> >> >> >> -- >> Gilad Abada >> >> SteadFast Telecommunications, Inc. >> >> Call us to find out how much you can save with VoIP! >> >> V: 212.589.1001 >> F: 212.589.1011 >> >> >> For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises.?? Make your virtual office a reality.? Enjoy the freedom to travel while remaining connected to your office. >> >> ______________________________________________________________________ >> ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users 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-use >> rs >> http://www.freeswitch.org > > > > -- > Gilad Abada > > SteadFast Telecommunications, Inc. > > Call us to find out how much you can save with VoIP! > > V: 212.589.1001 > F: 212.589.1011 > > > For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises.?? Make your virtual office a reality.? Enjoy the freedom to travel while remaining connected to your office. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Gilad Abada SteadFast Telecommunications, Inc. Call us to find out how much you can save with VoIP! V: 212.589.1001 F: 212.589.1011 For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises.?? Make your virtual office a reality.? Enjoy the freedom to travel while remaining connected to your office. From ryan at kaevee.com Fri Dec 23 18:55:44 2011 From: ryan at kaevee.com (Ryan V) Date: Fri, 23 Dec 2011 21:25:44 +0530 Subject: [Freeswitch-users] Ringback on PRI line In-Reply-To: References: Message-ID: Thanks for pointing me in right direction. I have to learn a lot on telephony and soft switches. I am total newbie when it comes to telephony and soft switches. I have taken a huge risk of replacing our Siemens PBX with freeswitch for a 100 people office. The only consolation is except for my partner, no one will scream at me :). Thanks to great guys developing freeswitch, I am having a great time working with freeswitch. Thanks, Venkatesh K On Fri, Dec 23, 2011 at 5:25 PM, Muhammad Shahzad < shaheryarkh at googlemail.com> wrote: > Here is the reason why you need this, > > http://wiki.sangoma.com/nbe-faq#no_ring > > Thank you. > > > > On Fri, Dec 23, 2011 at 12:19 PM, Ryan V wrote: > >> Adding following entries into inbound dial plan solved the problem. >> >> >> > data="ringback=data=ringback=${in-ring}"/> >> >> >> Thanks, >> >> Venkatesh K >> >> >> On Fri, Dec 23, 2011 at 4:50 AM, Michael Collins wrote: >> >>> This is probably a PRI config issue. Doublecheck the PRI protocol >>> settings with your provider. Also, get a d-channel trace and drop it on >>> pastebin. (see the freetdm wiki page for d-chan capture instructions.) >>> >>> I'm sure Moy or one of the guys here will be happy to take a look. >>> >>> -MC >>> >>> On Thu, Dec 22, 2011 at 12:06 AM, Ryan V wrote: >>> >>>> Hi, >>>> >>>> We are running freeswitch with Sangoma PRI and Analog cards. There is >>>> no ring back for calls coming in on PRI line. Calls coming in on FXO lines >>>> are fine. Calling party gets ring back. >>>> >>>> I have added following to file conf/dialplan/public/00_inbound_did.xml >>>> >>>> >>>> >>> data="ringback=%(2000,4000,440.0,480.0)"/> >>>> >>>> Any suggestions to resolve this problem? >>>> >>>> Thanks, >>>> >>>> Venkatesh 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 >>>> >>>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/f71aaf54/attachment.html From vetali100 at gmail.com Fri Dec 23 19:19:22 2011 From: vetali100 at gmail.com (Vitalie Colosov) Date: Fri, 23 Dec 2011 08:19:22 -0800 Subject: [Freeswitch-users] Rent freeswitch box for production In-Reply-To: References: Message-ID: Where it should be located? USA/Europe/Etc... This is important in the VOIP world because of packets delay. 2011/12/23 David Wafula > Hi Team, > Anyone recommend where i can rent box for hosting freeswitch to be > used as a conference server. I dont want a virtual box please. > > Regards, > -- > David Wafula > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/a3568d0c/attachment.html From lloydie.t at gmail.com Fri Dec 23 20:26:46 2011 From: lloydie.t at gmail.com (lloyd thomas) Date: Fri, 23 Dec 2011 17:26:46 +0000 Subject: [Freeswitch-users] Problem with make current In-Reply-To: References: <549CFEF87AEDE841A38E9D15EAB4C04C5B279DBF1C@cooper> <1324509156638-7116980.post@n2.nabble.com> <549CFEF87AEDE841A38E9D15EAB4C04C5B279DC017@cooper> Message-ID: Finally managed to do a make current. needed to install libtiff as well. In summary apt-get update apt-get install libtiff4-dev rm /usr/src/freeswitch/libs/spandsp/src/spandsp.h ./bootstrap.sh ./configure make current On 22 December 2011 21:01, lloyd thomas wrote: > Nope. Back to > > make[5]: *** [mod_spandsp.la] Error 1 > make[5]: Leaving directory > `/usr/src/freeswitch/src/mod/applications/mod_spandsp' > make[4]: *** [mod_spandsp-all] Error 1 > make[4]: Leaving directory `/usr/src/freeswitch/src/mod' > make[3]: *** [all-recursive] Error 1 > make[3]: Leaving directory `/usr/src/freeswitch/src' > make[2]: *** [all-recursive] Error 1 > > make[2]: Leaving directory `/usr/src/freeswitch' > make[1]: *** [all] Error 2 > make[1]: Leaving directory `/usr/src/freeswitch' > make: *** [current] Error 2 > > > On 22 December 2011 10:13, lloyd thomas wrote: > >> Thanks for that. will try again >> >> >> On 22 December 2011 08:30, Peter Olsson > > wrote: >> >>> Check out issue http://jira.freeswitch.org/browse/FS-3642 for this one. >>> Especially the first comment from Jeff ? you?re missing pkg-config.**** >>> >>> ** ** >>> >>> /Peter**** >>> >>> ** ** >>> >>> ** ** >>> >>> *Fr?n:* freeswitch-users-bounces at lists.freeswitch.org [mailto: >>> freeswitch-users-bounces at lists.freeswitch.org] *F?r *lloyd thomas >>> *Skickat:* den 22 december 2011 09:14 >>> >>> *Till:* FreeSWITCH Users Help >>> *?mne:* Re: [Freeswitch-users] Problem with make current**** >>> >>> ** ** >>> >>> maybe it's time to give up. >>> >>> tried 'make current', bootstrap and configure. >>> new error >>> ./configure: line 11073: syntax error near unexpected token `openssl,' >>> ./configure: line 11073: ` PKG_CHECK_MODULES(openssl, openssl,' >>> configure: error: ./configure.gnu failed for libs/iksemel >>> >>> **** >>> >>> On 21 December 2011 23:36, lloyd thomas wrote:**** >>> >>> Oh Dear. I was hoping to fix my registration problems with this. Will >>> open a ticket**** >>> >>> >>> >>> **** >>> >>> On 21 December 2011 23:12, Jeff Lenk wrote:**** >>> >>> thats crazy that a compiler should care about checking a return >>> parameter. >>> anyways you should open a Jira on this so it gets fixed. >>> >>> -- >>> View this message in context: >>> http://freeswitch-users.2379917.n2.nabble.com/Problem-with-make-current-tp7113961p7116980.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:4ef2e5fc32761950963548! **** >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/68d65b55/attachment-0001.html From lloydie.t at gmail.com Fri Dec 23 20:36:02 2011 From: lloydie.t at gmail.com (lloyd thomas) Date: Fri, 23 Dec 2011 17:36:02 +0000 Subject: [Freeswitch-users] Problems with polycom registering In-Reply-To: References: Message-ID: Not sure what the problem was, but I updated the FS box with 'make current' and I am back in business (for now) On 22 December 2011 23:48, lloyd thomas wrote: > The FS box is assigned public IP address via a router in bridge mode (DMZ > plus). The router (Draytek) which the polycoms are connected to is also > connected to the same router in bridge mode. The reason why I have it this > way is because I have a remote polycom phone which seems to be registering > OK. > This set worked great on my old broadband provider and for around 5 days > with the new provider. > > example- > polycom phone(192.168.101.16) <--> Router(81.137.114.169) <--> (bridged > modem) <--> INTERNET > FS box (81.137.114.171) > -----------------------------------------------------------(bridged modem) > <--> INTERNET > > gateway = 81.137.114.174 > subnet = 255.255.255.248 > > Lloydie T > > > On 22 December 2011 22:42, Michael Collins wrote: > >> What's your topology here? Is FS on public IP or behind NAT? Are the >> Polys on same network as FS? If not, are they behind NAT? >> >> -MC >> >> On Tue, Dec 20, 2011 at 10:03 AM, lloyd thomas wrote: >> >>> I had what I thought was the perfect set up on my FS box. Until I >>> changed internet provider (BT) >>> All was fine for a week but all of a sudden some of my Polycom phones >>> won't register. >>> The polycom phones are behind a NAT and my FS box is on a static IP >>> address. >>> I am using a multi company setup >>> >>> One of my polycom phones seems to register OK >>> >>> Using 'sofia profile xxxxxxxx siptrace on' I can see that one of my >>> polycom phones will not register because FS is trying to send replies to >>> the private IP address (192.168.101.16) >>> Any Ideas? Am I missing something in one of the profiles >>> >>> Good one >>> >>> ------------------------------------------------------------------------ >>> send 702 bytes to udp/[87.194.242.110]:65004 at 17:49:32.226606: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 200 OK >>> Via: SIP/2.0/UDP 87.194.242.110:65004;branch=z9hG4bK15c264eE5B5988B >>> From: "Suze" ;tag=CA76AF43-ED2A5944 >>> To: ;tag=7Uy88X4pc8yHc >>> Call-ID: 918d5430-ffb07455-eb2121fe at 87.194.242.110 >>> CSeq: 888 REGISTER >>> Contact: ;methods="INVITE, ACK, BYE, >>> CANCEL, OPTIONS, INFO, MESSAGE, SUBSCRIBE, NOTIFY, PRACK, UPDATE, >>> REFER";expires=30 >>> Date: Tue, 20 Dec 2011 17:49:32 GMT >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2c009dd 2011-03-15 >>> 14-29-04 -0500 >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY >>> Supported: timer, precondition, path, replaces >>> Content-Length: 0 >>> >>> >>> ------------------------------------------------------------------------- >>> >>> Bad one >>> >>> ------------------------------------------------------------------------ >>> send 680 bytes to udp/[81.137.114.169]:5060 at 17:50:17.659167: >>> >>> ------------------------------------------------------------------------ >>> SIP/2.0 401 Unauthorized >>> Via: SIP/2.0/UDP 192.168.101.16:5060 >>> ;branch=z9hG4bK2b8f42dA63D9E42;received=81.137.154.169 >>> From: "Lloyd" ;tag=621EA6CE-AF1D573 >>> To: ;tag=aQaKeFQ13219e >>> Call-ID: 52e98627-d24590dc-528d7361 at 192.168.101.16 >>> CSeq: 1 REGISTER >>> User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-2c009dd 2011-03-15 >>> 14-29-04 -0500 >>> Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, >>> REGISTER, REFER, NOTIFY >>> Supported: timer, precondition, path, replaces >>> WWW-Authenticate: Digest realm="phisys.tele.phi.co.uk", >>> nonce="1319a358-2b33-11e1-b5e4-dd1099e4b2d2", algorithm=MD5, qop="auth" >>> 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 >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/db81c9b9/attachment.html From voipservice911 at gmail.com Fri Dec 23 15:39:24 2011 From: voipservice911 at gmail.com (Voip service) Date: Fri, 23 Dec 2011 16:39:24 +0400 Subject: [Freeswitch-users] Number of calls In-Reply-To: References: Message-ID: Hi MC, Thanks for reply, I have a requirement of more than 1000 calls per sec. Can it support? On Fri, Dec 23, 2011 at 12:22 AM, Michael Collins wrote: > That looks like a decent 4-core processor. You can probably do several > hundred concurrent calls w/o much issue. Of course, "it depends" on what > other stuff you have going on inside this server. :) > > -MC > > On Wed, Dec 21, 2011 at 11:44 PM, Voip service wrote: > >> Hi, >> >> I am new in voip, how many calls can one freeswitch box handle with 30 % >> of trans-coded calls and system configuration as >> 8GB RAM >> X3430 Xeon Processor, 2.4GHz, 8M Cache, Turbo >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/f6ff70f3/attachment-0001.html From vhatz at kinetix.gr Fri Dec 23 17:51:52 2011 From: vhatz at kinetix.gr (Vlasis Hatzistavrou (KTI)) Date: Fri, 23 Dec 2011 16:51:52 +0200 Subject: [Freeswitch-users] Seasons greetings from Kinetix Tele.com Message-ID: <4EF49588.5090702@kinetix.gr> Seasons greetings from Kinetix Tele.com -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/e20677a5/attachment-0001.html -------------- next part -------------- A non-text attachment was scrubbed... Name: not available Type: image/jpeg Size: 188311 bytes Desc: not available Url : http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/e20677a5/attachment-0001.jpe From matt at partlytrue.com Fri Dec 23 19:35:53 2011 From: matt at partlytrue.com (Matt Piermarini) Date: Fri, 23 Dec 2011 11:35:53 -0500 Subject: [Freeswitch-users] Help with a Cisco 7970 phone. Message-ID: <4EF4ADE9.9030802@partlytrue.com> Hello all, I have been trying forever to get this Cisco 7970 to work properly with our freeswitch (latest git). The phone will register fine, and it will make outbound calls perfectly. The problem is with inbound calls to this phone, where FS tries to contact it on the wrong UDP port. The entire system here is on a local lan, and NAT has been disabled via --nonat cmd line FS parameter. Here is what it looks like when registered: sofia status profile internal reg Call-ID: 00181844-c24f0002-25f0dff2-c470b5a7 at 192.168.2.59 User: 15 at x.y.com Contact: "user" Agent: Cisco-CP7970G/8.5.2 Status: Registered(UDP-NAT)(unknown) EXP(2011-12-22 21:31:47) EXPSECS(724) Host: x.y.com IP: 192.168.2.59 Port: 49165 Auth-User: 15 Auth-Realm: x.y.com FS tries to contact the phone using the fs_path, which is getting set incorrectly. I'm not why FS thinks the fs_path should be different, as it appears the Cisco is sending proper Contact header in the sip requests/responses. Also, we have a bunch of ATA's on the same lan which all work fine (meaning the fs_path is correct). I tried chaning the NDLB-force-rport param in the internal context, but didn't seems to change anything. Here are the SIP traces which show what's happening. Any ideas of what's going on would be appreciated. Thanks, Matt 192.168.2.59 is the cisco phone. 192.168.2.1 is FS (listening on port 5070). recv 713 bytes from udp/[192.168.2.59]:49320 at 02:28:01.712338: ------------------------------------------------------------------------ REGISTER sip:x.y.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.59:5060;branch=z9hG4bKd9e43394 From: ;tag=00181844c24f0002a0bb4968-7b2b7dd4 To: Call-ID: 00181844-c24f0002-eb21e8a8-77681214 at 192.168.2.59 Max-Forwards: 70 Date: Fri, 23 Dec 2011 02:28:00 GMT CSeq: 101 REGISTER User-Agent: Cisco-CP7970G/8.5.2 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="30006" Supported: (null),X-cisco-xsi-7.0.1 Content-Length: 0 Reason: SIP;cause=200;text="cisco-alarm:20 Name=SEP00181844C24F Load=SIP70.8-5-2S Last=phone-keypad" Expires: 3600 send 689 bytes to udp/[192.168.2.59]:5060 at 02:28:01.715057: ------------------------------------------------------------------------ SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.2.59:5060;branch=z9hG4bKd9e43394 From: ;tag=00181844c24f0002a0bb4968-7b2b7dd4 To: ;tag=Qv2N52egZ2teH Call-ID: 00181844-c24f0002-eb21e8a8-77681214 at 192.168.2.59 CSeq: 101 REGISTER User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-8059cdc 2011-12-22 14-03-32 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces WWW-Authenticate: Digest realm="x.y.com", nonce="fae142ab-8b00-4ac7-89d7-04955e8714f6", algorithm=MD5, qop="auth" Content-Length: 0 recv 958 bytes from udp/[192.168.2.59]:49320 at 02:28:01.725802: ------------------------------------------------------------------------ REGISTER sip:x.y.com SIP/2.0 Via: SIP/2.0/UDP 192.168.2.59:5060;branch=z9hG4bK18d5a468 From: ;tag=00181844c24f0002a0bb4968-7b2b7dd4 To: Call-ID: 00181844-c24f0002-eb21e8a8-77681214 at 192.168.2.59 Max-Forwards: 70 Date: Fri, 23 Dec 2011 02:28:00 GMT CSeq: 102 REGISTER User-Agent: Cisco-CP7970G/8.5.2 Contact: ;+sip.instance="";+u.sip!model.ccm.cisco.com="30006" Authorization: Digest username="15",realm="x.y.com",uri="sip:x.y.com",response="xxxxxxxxxxxxxxxxxxxxxxxxxxxxxx",nonce="fae142ab-8b00-4ac7-89d7-04955e8714f6",cnonce="4f2a64d4",qop=auth,nc=00000001,algorithm=MD5 Supported: (null),X-cisco-xsi-7.0.1 Content-Length: 0 Reason: SIP;cause=200;text="cisco-alarm:20 Name=SEP00181844C24F Load=SIP70.8-5-2S Last=phone-keypad" Expires: 3600 send 649 bytes to udp/[192.168.2.59]:5060 at 02:28:01.737460: ------------------------------------------------------------------------ SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.2.59:5060;branch=z9hG4bK18d5a468 From: ;tag=00181844c24f0002a0bb4968-7b2b7dd4 To: ;tag=SeN78rgQSm7Kr Call-ID: 00181844-c24f0002-eb21e8a8-77681214 at 192.168.2.59 CSeq: 102 REGISTER Contact: ;expires=3600 Date: Fri, 23 Dec 2011 02:28:01 GMT User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-8059cdc 2011-12-22 14-03-32 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Content-Length: 0 *--------------- OK, so far so good.. phone is now registered.. Now look how FS tries to contact it.. sending notify's here: Note the port it tries to use, 49320 in this case.* send 1034 bytes to udp/[192.168.2.59]:49320 at 02:28:01.837149: ------------------------------------------------------------------------ NOTIFY sip:15 at 192.168.2.59:5060;transport=udp;fs_nat=yes;fs_path=sip:15%40192.168.2.59:49320%3Btransport%3Dudp SIP/2.0 Via: SIP/2.0/UDP 192.168.2.1:5070;rport;branch=z9hG4bKv92r0r53rQKNa Route: ;transport=udp Max-Forwards: 70 From: ;tag=U07rcFjyK6KSF To: Call-ID: 93e179d6-a7b0-122f-d396-5254003b348b CSeq: 21967576 NOTIFY Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-8059cdc 2011-12-22 14-03-32 -0600 Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, precondition, path, replaces Event: message-summary Allow-Events: talk, hold, presence, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Subscription-State: terminated;reason=timeout Content-Type: application/simple-message-summary Content-Length: 69 Messages-Waiting: no Message-Account: sip:15 at x.y.com *The phone never responds, as FS is supposed to contact it on port 5060, but FS is trying to use the same port that phone sent is SIP messages on. Here is a sample INVITE:* send 1229 bytes to udp/[192.168.2.59]:49320 at 02:38:15.217775: ------------------------------------------------------------------------ INVITE sip:15 at 192.168.2.59:5060;transport=udp SIP/2.0 Via: SIP/2.0/UDP 192.168.2.1:5070;rport;branch=z9hG4bK0D8U748HDUc0r Route: ;transport=udp Max-Forwards: 69 From: "x x x" ;tag=63X9ZrB5K886e To: Call-ID: 00e33727-a7b2-122f-d396-5254003b348b CSeq: 21967883 INVITE Contact: User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-8059cdc 2011-12-22 14-03-32 -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-Type: application/sdp Content-Disposition: session Content-Length: 204 X-FS-Support: update_display,send_info Remote-Party-ID: "x x x" ;party=calling;screen=yes;privacy=off v=0 o=FreeSWITCH 1324545412 1324545413 IN IP4 192.168.2.1 s=FreeSWITCH c=IN IP4 192.168.2.1 t=0 0 m=audio 62482 RTP/AVP 18 0 8 101 13 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/433fb967/attachment.html From msc at freeswitch.org Fri Dec 23 20:55:30 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Dec 2011 09:55:30 -0800 Subject: [Freeswitch-users] Number of calls In-Reply-To: References: Message-ID: 1000 *new* calls per second? I wouldn't go with anything less than an 8-core behemoth for that. -MC On Fri, Dec 23, 2011 at 4:39 AM, Voip service wrote: > Hi MC, > > Thanks for reply, I have a requirement of more than 1000 calls per sec. > Can it support? > > On Fri, Dec 23, 2011 at 12:22 AM, Michael Collins wrote: > >> That looks like a decent 4-core processor. You can probably do several >> hundred concurrent calls w/o much issue. Of course, "it depends" on what >> other stuff you have going on inside this server. :) >> >> -MC >> >> On Wed, Dec 21, 2011 at 11:44 PM, Voip service wrote: >> >>> Hi, >>> >>> I am new in voip, how many calls can one freeswitch box handle with 30 % >>> of trans-coded calls and system configuration as >>> 8GB RAM >>> X3430 Xeon Processor, 2.4GHz, 8M Cache, Turbo >>> >>> >>> _________________________________________________________________________ >>> Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/e3434e50/attachment-0001.html From msc at freeswitch.org Fri Dec 23 21:03:03 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Dec 2011 10:03:03 -0800 Subject: [Freeswitch-users] Radio stream binding ? In-Reply-To: <1324602042.19229.YahooMailClassic@web110803.mail.gq1.yahoo.com> References: <1324602042.19229.YahooMailClassic@web110803.mail.gq1.yahoo.com> Message-ID: there are thousands of stations you can get at shoutcast.com. those work very well. -MC On Thu, Dec 22, 2011 at 5:00 PM, Sherif Omran wrote: > hello guys, > > can any body help me to use > > bind_meta_app > > to connect a channel to a radio stream using gnuradio.org project? > > thanks in advance > > 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/20111223/41f0f7cf/attachment.html From msc at freeswitch.org Fri Dec 23 21:05:57 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Dec 2011 10:05:57 -0800 Subject: [Freeswitch-users] Restrict outbound calls based on extension In-Reply-To: References: Message-ID: There may actually be something in the wiki already. If not, be sure to mention that the toll_allow variable is used in the default.xml dialplan file. -MC On Fri, Dec 23, 2011 at 12:43 AM, Ryan V wrote: > Thanks. I managed to restrict by doing this. > > I added a variable into the directory entry for user like > > > > and restricted in outgoing dial plan with > > > > Probably I should add an entry in wiki for people like me. > > Thanks again, > > Venkatesh K > > > On Fri, Dec 23, 2011 at 12:59 PM, Avi Marcus wrote: > >> In brief: >> 1 way would be to store a variable in each directory user with what type >> of calls are allowed. Domestic, International, etc. e.g. >> "domestic,international" >> Then in your extension that actually routes the outbound calls, you would >> also check for the condition that this variable you set has the permission >> in it's list. >> >> Sorry for not showing you the actual code.. >> -Avi >> >> >> On Fri, Dec 23, 2011 at 9:22 AM, Ryan V wrote: >> >>> Hello, >>> >>> I want to allow outbound calls to a limited number of users. I have gone >>> through toll definitions in dial plan. But, I am struggling to find a way >>> to achieve that. Please help. >>> >>> Thanks, >>> >>> 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/20111223/e8af0dee/attachment.html From msc at freeswitch.org Fri Dec 23 21:08:29 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Dec 2011 10:08:29 -0800 Subject: [Freeswitch-users] Can't quite get call screening to work In-Reply-To: References: <013301ccb521$b1e4eca0$15aec5e0$@com> Message-ID: try setting this var to true prior to the bridge: http://wiki.freeswitch.org/wiki/Channel_Variables#hangup_after_bridge On Thu, Dec 22, 2011 at 2:09 PM, Gilad Abada wrote: > Hi > I am trying to get call screening to work too. > > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_13:_Call_Screening > > The issue I am having is that if the called device hangs up even after > a full conversation (accepting the call by pressing 1) the calling > party gets sent to voicemail. > > Seems like being sent to voicemail should be an anti-action? > Also the recording says to reject the call press 2 or send it to VM > press 3. These are not defined in the dial plan and I am not sure how > to do that. > > Thanks in advance! > Gill > > On Wed, Dec 7, 2011 at 5:51 PM, Michael Collins > wrote: > > it looks like the person who posted that example did not post their > sample > > phrase macro file. However, do a git pull... > > > > commit 9ea3ce666fa7f021b5c2a7e2fbe153eb351c5734 > > Author: Michael S Collins > > Date: Wed Dec 7 14:49:16 2011 -0800 > > > > config: add screen_confirm macro to lang/en/ivr/sounds.xml > > > > snag that config file and drop it into conf/lang/en/ivr/ and then > reloadxml. > > I did this on the fly w/o testing so be sure to test it thoroughly to > make > > sure it works! Also, be sure to use the full path name to the sound file > > that you are playing back. > > > > -MC > > > > On Wed, Dec 7, 2011 at 12:49 PM, Phil Quesinberry > > wrote: > >> > >> I'm trying to use the call screening example in the wiki and can't get > FS > >> to > >> play the caller's name back to the destination extension. > >> example here: > >> > >> > http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_13:_Call_Scr > >> eening > >> > >> The initial announcement asking for the caller's name works fine: > >> EXECUTE sofia/internal/102 at 192.168.1.6:5060phrase(voicemail_record_name) > >> 2011-12-07 14:52:13.715494 [DEBUG] mod_dptools.c:2362 Execute > >> voicemail_record_name() lang > >> 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:67 No language > >> specified - Using [en] > >> 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:244 Handle > >> play-file:[voicemail/vm-record_name1.wav] (en:en) > >> 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:1302 Codec > >> Activated L16 at 8000hz 1 channels 20ms > >> 2011-12-07 14:52:18.695891 [DEBUG] switch_ivr_play_say.c:1672 done > playing > >> file > >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_name1.wav > >> EXECUTE sofia/internal/102 at 192.168.1.6:5060playback(tone_stream://%(500, > >> 0, > >> 640)) > >> 2011-12-07 14:52:18.799791 [DEBUG] switch_ivr_play_say.c:1302 Codec > >> Activated L16 at 8000hz 1 channels 20ms > >> 2011-12-07 14:52:19.299814 [DEBUG] switch_ivr_play_say.c:1672 done > playing > >> file tone_stream://%(500, 0, 640) > >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 > >> set(playback_terminators=#*0123456789) > >> 2011-12-07 14:52:19.299814 [DEBUG] mod_dptools.c:1263 > >> sofia/internal/102 at 192.168.1.6:5060 SET > >> [playback_terminators]=[#*0123456789] > >> > >> Then the caller's name is recorded, and I've verified that the recording > >> is > >> indeed saved in /tmp: > >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 record(/tmp/102-name.wav 7 > 200 > >> 2) > >> 2011-12-07 14:52:19.299814 [DEBUG] switch_ivr_play_say.c:585 Raw Codec > >> Activated > >> 2011-12-07 14:52:19.299814 [DEBUG] switch_core_codec.c:116 > >> sofia/internal/102 at 192.168.1.6:5060 Push codec L16:70 > >> . > >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(group_confirm_key=1) > >> 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 > >> sofia/internal/102 at 192.168.1.6:5060 SET [group_confirm_key]=[1] > >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 > >> set(fail_on_single_reject=true) > >> 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 > >> sofia/internal/102 at 192.168.1.6:5060 SET [fail_on_single_reject]=[true] > >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 > >> set(group_confirm_file=phrase:screen_confirm:/tmp/102-name.wav) > >> 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 > >> sofia/internal/102 at 192.168.1.6:5060 SET > >> [group_confirm_file]=[phrase:screen_confirm:/tmp/102-name.wav] > >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(continue_on_fail=true) > >> 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 > >> sofia/internal/102 at 192.168.1.6:5060 SET [continue_on_fail]=[true] > >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bridge(user/102) > >> . > >> Then when attempting to play back the Output from the console showing > the > >> error is here: > >> 2011-12-07 14:52:26.348406 [ERR] switch_ivr_play_say.c:142 Can't find > >> macro > >> screen_confirm. > >> 2011-12-07 14:52:26.348406 [WARNING] switch_ivr_play_say.c:339 Macro > >> [screen_confirm]: '/tmp/102-name.wav' did not match any patterns > >> 2011-12-07 14:52:26.348406 [ERR] switch_ivr_originate.c:219 > >> sofia/internal/sip:102 at 192.168.1.4:5060 Error Playing File! > >> > >> The call goes right to voicemail once the destination extension attempts > >> to > >> answer it. > >> > >> Where are these macros supposed to be stored? Somewhere under > >> /usr/local/freeswitch/conf/lang/en? Do I need to create a macro for > >> screen_confirm or is it just named incorrectly or in the wrong place? > >> > >> 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 > > > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > -- > Gilad Abada > > SteadFast Telecommunications, Inc. > > Call us to find out how much you can save with VoIP! > > V: 212.589.1001 > F: 212.589.1011 > > > For 35 years, Steadfast Telecommunications has been providing > state-of-the-art communications technology to businesses and > government agencies - large and small. Steadfast Telecommunications > tailors Unified Communications and Voice-Over IP Solutions to > single-site offices or multi-site and worldwide enterprises. Make > your virtual office a reality. Enjoy the freedom to travel while > remaining connected to your office. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/eac4511c/attachment-0001.html From msc at freeswitch.org Fri Dec 23 21:10:24 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Dec 2011 10:10:24 -0800 Subject: [Freeswitch-users] int/ext dial tones not synchronous In-Reply-To: References: Message-ID: I'm not sure that I understand what the issue is here. What two "tones" are not in sync? You said "dial tone" but I'm pretty sure that's not what you mean. Are you talking about the ringback tone that the caller hears? -MC On Thu, Dec 22, 2011 at 6:10 PM, wrote: > Hello all, > > I made the first experiences with the dialplan, and after some while, I > really got calls from the outside routed to one of my sip phones. > > However, my phone is ringing something like 1-3 seconds earlier than the > dial tone the caller is hearing. > > I googled around, but found no solution. > > As a start, I just used this extension: > > > > > > > Thanks, > Georg > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/b6d127d2/attachment.html From ga at steadfasttelecom.com Fri Dec 23 21:12:13 2011 From: ga at steadfasttelecom.com (Gilad Abada) Date: Fri, 23 Dec 2011 13:12:13 -0500 Subject: [Freeswitch-users] Can't quite get call screening to work In-Reply-To: References: <013301ccb521$b1e4eca0$15aec5e0$@com> Message-ID: Thanks got it working and updated thre wiki On Fri, Dec 23, 2011 at 1:08 PM, Michael Collins wrote: > try setting this var to true prior to the bridge: > http://wiki.freeswitch.org/wiki/Channel_Variables#hangup_after_bridge > > > On Thu, Dec 22, 2011 at 2:09 PM, Gilad Abada > wrote: >> >> Hi >> I am trying to get call screening to work too. >> >> http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_13:_Call_Screening >> >> The issue I am having is that if the called device hangs up even after >> a full conversation (accepting the call by pressing 1) the calling >> party gets sent to voicemail. >> >> Seems like being sent to voicemail should be an anti-action? >> Also the recording says to reject the call press 2 or send it to VM >> press 3. These are not defined in the dial plan and I am not sure how >> to do that. >> >> Thanks in advance! >> Gill >> >> On Wed, Dec 7, 2011 at 5:51 PM, Michael Collins >> wrote: >> > it looks like the person who posted that example did not post their >> > sample >> > phrase macro file. However, do a git pull... >> > >> > commit 9ea3ce666fa7f021b5c2a7e2fbe153eb351c5734 >> > Author: Michael S Collins >> > Date: ? Wed Dec 7 14:49:16 2011 -0800 >> > >> > ? ? config: add screen_confirm macro to lang/en/ivr/sounds.xml >> > >> > snag that config file and drop it into conf/lang/en/ivr/ and then >> > reloadxml. >> > I did this on the fly w/o testing so be sure to test it thoroughly to >> > make >> > sure it works! Also, be sure to use the full path name to the sound file >> > that you are playing back. >> > >> > -MC >> > >> > On Wed, Dec 7, 2011 at 12:49 PM, Phil Quesinberry >> > wrote: >> >> >> >> I'm trying to use the call screening example in the wiki and can't get >> >> FS >> >> to >> >> play the caller's name back to the destination extension. >> >> example here: >> >> >> >> >> >> http://wiki.freeswitch.org/wiki/FreeSwitch_Dialplan_XML#Example_13:_Call_Scr >> >> eening >> >> >> >> The initial announcement asking for the caller's name works fine: >> >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 >> >> phrase(voicemail_record_name) >> >> 2011-12-07 14:52:13.715494 [DEBUG] mod_dptools.c:2362 Execute >> >> voicemail_record_name() lang >> >> 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:67 No language >> >> specified - Using [en] >> >> 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:244 Handle >> >> play-file:[voicemail/vm-record_name1.wav] (en:en) >> >> 2011-12-07 14:52:13.715494 [DEBUG] switch_ivr_play_say.c:1302 Codec >> >> Activated L16 at 8000hz 1 channels 20ms >> >> 2011-12-07 14:52:18.695891 [DEBUG] switch_ivr_play_say.c:1672 done >> >> playing >> >> file >> >> /usr/local/freeswitch/sounds/en/us/callie/voicemail/vm-record_name1.wav >> >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 >> >> playback(tone_stream://%(500, >> >> 0, >> >> 640)) >> >> 2011-12-07 14:52:18.799791 [DEBUG] switch_ivr_play_say.c:1302 Codec >> >> Activated L16 at 8000hz 1 channels 20ms >> >> 2011-12-07 14:52:19.299814 [DEBUG] switch_ivr_play_say.c:1672 done >> >> playing >> >> file tone_stream://%(500, 0, 640) >> >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 >> >> set(playback_terminators=#*0123456789) >> >> 2011-12-07 14:52:19.299814 [DEBUG] mod_dptools.c:1263 >> >> sofia/internal/102 at 192.168.1.6:5060 SET >> >> [playback_terminators]=[#*0123456789] >> >> >> >> Then the caller's name is recorded, and I've verified that the >> >> recording >> >> is >> >> indeed saved in /tmp: >> >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 record(/tmp/102-name.wav 7 >> >> 200 >> >> 2) >> >> 2011-12-07 14:52:19.299814 [DEBUG] switch_ivr_play_say.c:585 Raw Codec >> >> Activated >> >> 2011-12-07 14:52:19.299814 [DEBUG] switch_core_codec.c:116 >> >> sofia/internal/102 at 192.168.1.6:5060 Push codec L16:70 >> >> . >> >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(group_confirm_key=1) >> >> 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 >> >> sofia/internal/102 at 192.168.1.6:5060 SET [group_confirm_key]=[1] >> >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 >> >> set(fail_on_single_reject=true) >> >> 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 >> >> sofia/internal/102 at 192.168.1.6:5060 SET [fail_on_single_reject]=[true] >> >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 >> >> set(group_confirm_file=phrase:screen_confirm:/tmp/102-name.wav) >> >> 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 >> >> sofia/internal/102 at 192.168.1.6:5060 SET >> >> [group_confirm_file]=[phrase:screen_confirm:/tmp/102-name.wav] >> >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 set(continue_on_fail=true) >> >> 2011-12-07 14:52:21.623965 [DEBUG] mod_dptools.c:1263 >> >> sofia/internal/102 at 192.168.1.6:5060 SET [continue_on_fail]=[true] >> >> EXECUTE sofia/internal/102 at 192.168.1.6:5060 bridge(user/102) >> >> . >> >> Then when attempting to play back the Output from the console showing >> >> the >> >> error is here: >> >> 2011-12-07 14:52:26.348406 [ERR] switch_ivr_play_say.c:142 Can't find >> >> macro >> >> screen_confirm. >> >> 2011-12-07 14:52:26.348406 [WARNING] switch_ivr_play_say.c:339 Macro >> >> [screen_confirm]: '/tmp/102-name.wav' did not match any patterns >> >> 2011-12-07 14:52:26.348406 [ERR] switch_ivr_originate.c:219 >> >> sofia/internal/sip:102 at 192.168.1.4:5060 Error Playing File! >> >> >> >> The call goes right to voicemail once the destination extension >> >> attempts >> >> to >> >> answer it. >> >> >> >> Where are these macros supposed to be stored? ?Somewhere under >> >> /usr/local/freeswitch/conf/lang/en? ?Do I need to create a macro for >> >> screen_confirm or is it just named incorrectly or in the wrong place? >> >> >> >> 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 >> > >> > >> > >> > >> > _________________________________________________________________________ >> > Professional FreeSWITCH Consulting Services: >> > consulting at freeswitch.org >> > http://www.freeswitchsolutions.com >> > >> > >> > >> > >> > Official FreeSWITCH Sites >> > http://www.freeswitch.org >> > http://wiki.freeswitch.org >> > http://www.cluecon.com >> > >> > FreeSWITCH-users mailing list >> > FreeSWITCH-users at lists.freeswitch.org >> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> > http://www.freeswitch.org >> > >> >> >> >> -- >> Gilad Abada >> >> SteadFast Telecommunications, Inc. >> >> Call us to find out how much you can save with VoIP! >> >> V: 212.589.1001 >> F: 212.589.1011 >> >> >> For 35 years, Steadfast Telecommunications has been providing >> state-of-the-art communications technology to businesses and >> government agencies - large and small. Steadfast Telecommunications >> tailors Unified Communications and Voice-Over IP Solutions to >> single-site offices or multi-site and worldwide enterprises.?? Make >> your virtual office a reality.? Enjoy the freedom to travel while >> remaining connected to your office. >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -- Gilad Abada SteadFast Telecommunications, Inc. Call us to find out how much you can save with VoIP! V: 212.589.1001 F: 212.589.1011 For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises.?? Make your virtual office a reality.? Enjoy the freedom to travel while remaining connected to your office. From msc at freeswitch.org Fri Dec 23 21:12:38 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Dec 2011 10:12:38 -0800 Subject: [Freeswitch-users] Rent freeswitch box for production In-Reply-To: References: Message-ID: Talk to NormT or SwK in #freeswitch on irc.freenode.net if you need something in North America. -MC On Fri, Dec 23, 2011 at 6:22 AM, David Wafula wrote: > Hi Team, > Anyone recommend where i can rent box for hosting freeswitch to be > used as a conference server. I dont want a virtual box please. > > Regards, > -- > David Wafula > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/b9c3b722/attachment.html From notlikeme75 at yahoo.com Fri Dec 23 21:31:45 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Fri, 23 Dec 2011 10:31:45 -0800 (PST) Subject: [Freeswitch-users] transfer fail from conf to conf / desination out of order Message-ID: <1324665105.74396.YahooMailNeo@web65309.mail.ac2.yahoo.com> any idea why it just hangs up and never gives me the extension? extension works if i put it on the ivr. http://pastebin.freeswitch.org/18064 ~rodney -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/04bcf5fc/attachment.html From msc at freeswitch.org Fri Dec 23 23:43:10 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Dec 2011 12:43:10 -0800 Subject: [Freeswitch-users] transfer fail from conf to conf / desination out of order In-Reply-To: <1324665105.74396.YahooMailNeo@web65309.mail.ac2.yahoo.com> References: <1324665105.74396.YahooMailNeo@web65309.mail.ac2.yahoo.com> Message-ID: This looks like a possible bug. I can replicate on my system. I was able to use bind_digit_action to work around the issue. Open a Jira on this and report what you've got. Let me know if you need help trying out bind_digit_action (BDA). -MC On Fri, Dec 23, 2011 at 10:31 AM, Rodney wrote: > any idea why it just hangs up and never gives me the extension? extension > works if i put it on the ivr. > > http://pastebin.freeswitch.org/18064 > > > ~rodney > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/3d1dce8a/attachment-0001.html From msc at freeswitch.org Fri Dec 23 23:55:06 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Dec 2011 12:55:06 -0800 Subject: [Freeswitch-users] transfer back to ivr after voicemail function In-Reply-To: <1324586992.63365.YahooMailNeo@web65313.mail.ac2.yahoo.com> References: <1324586992.63365.YahooMailNeo@web65313.mail.ac2.yahoo.com> Message-ID: Just have more actions after voicemail in the dialplan. Also, you may wish to modify the voicemail_goodbye macro if you don't want it to say "thank you, goodbye" when exiting VM. -MC On Thu, Dec 22, 2011 at 12:49 PM, Rodney wrote: > I am trying to transfer someone back to the IVR after they leave a message > for someone or check their own mailbox. currently it does a hangup and they > have to call back to do more. is there a method for this? or is it hard > coded in the "press # to continue portion of voice mail? > > > check voice mail > > condition destination_number ^757$ 1 > action answer 3 > action sleep 1000 4 > action play_and_get_digits 4 4 3 5000 # checkvoicemail.wav > ivr/ivr-that_was_an_invalid_entry.wav vmb \d+ 12 > action phrase spell,${vmb} 15 > action voicemail check default ${domain_name} ${vmb} > > > leave message > > > condition destination_number ^758$ 1 > action answer 3 > action sleep 1000 4 > action play_and_get_digits 4 4 3 5000 # checkvoicemail.wav > ivr/ivr-that_was_an_invalid_entry.wav vmb \d+ 12 > action phrase spell,${vmb} 15 > action voicemail default ${domain_name} ${vmb} > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/1f7ecae1/attachment.html From notlikeme75 at yahoo.com Sat Dec 24 00:05:37 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Fri, 23 Dec 2011 13:05:37 -0800 (PST) Subject: [Freeswitch-users] transfer from static conf to dynamic destination out of order. In-Reply-To: References: Message-ID: <1324674337.10113.YahooMailNeo@web65308.mail.ac2.yahoo.com> MC, I will open up a jira for this possible bug. any help on the bind digit action would be great. i failed last time i tried. can bind digit action be single digit? this is what i am hoping for but i still want that digit to be "pound" i am thinking it wont work though. my extension into the static conference looks like this. where do put the bind digit action? before i have tried: backforward,7,exec:execute_extension,502 XML default? (so pressing dtmf 7 while in static conf 501 would transfer to 502) but didnt work. didnt even show up as binding the "backforward" realm in console. condition??? ? destination_number??? ? ^501$??? ? 1??? ? ? action??? ? set??? ? conference_user_list=|??? ? 4??? ? ? action??? ? playback??? ? C:/Program Files/FreeSWITCH/sounds/en/us/callie/digits/8000/1.wav??? ? 8??? ? ? action??? ? answer??? ? ??? ? 9??? ? ? action??? ? conference??? ? 501-127.0.0.1 at default any idea why it just hangs up and never gives me the extension? extension works if i put it on the ivr. http://pastebin.freeswitch.org/18064 ~rodney This looks like a possible bug. I can replicate on my system. I was able to use bind_digit_action to work around the issue. Open a Jira on this and report what you've got. Let me know if you need help trying out bind_digit_action (BDA). -MC On Fri, Dec 23, 2011 at 10:31 AM, Rodney wrote: any idea why it just hangs up and never gives me the extension? extension works if i put it on the ivr. > > >http://pastebin.freeswitch.org/18064 > > > >~rodney > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/c73685e6/attachment.html From msc at freeswitch.org Sat Dec 24 00:27:01 2011 From: msc at freeswitch.org (Michael Collins) Date: Fri, 23 Dec 2011 13:27:01 -0800 Subject: [Freeswitch-users] transfer from static conf to dynamic destination out of order. In-Reply-To: <1324674337.10113.YahooMailNeo@web65308.mail.ac2.yahoo.com> References: <1324674337.10113.YahooMailNeo@web65308.mail.ac2.yahoo.com> Message-ID: Edit your conference.conf.xml and remove the # action in the caller controls. Then add something like this to the dialplan prior to going into the conference: Note that you can change the BDA realm to something else if you want to change the meaning of the # key. That's why BDA is so awesome - you can do ANYTHING with it! -MC On Fri, Dec 23, 2011 at 1:05 PM, Rodney wrote: > MC, I will open up a jira for this possible bug. any help on the bind > digit action would be great. i failed last time i tried. can bind digit > action be single digit? this is what i am hoping for but i still want that > digit to be "pound" i am thinking it wont work though. my extension into > the static conference looks like this. where do put the bind digit action? > > before i have tried: > > backforward,7,exec:execute_extension,502 XML default (so pressing dtmf 7 > while in static conf 501 would transfer to 502) but didnt work. didnt even > show up as binding the "backforward" realm in console. > > condition destination_number ^501$ 1 > action set conference_user_list=| 4 > action playback C:/Program > Files/FreeSWITCH/sounds/en/us/callie/digits/8000/1.wav 8 > action answer 9 > action conference 501-127.0.0.1 at default > > > > > any idea why it just hangs up and never gives me the extension? extension > works if i put it on the ivr. > > http://pastebin.freeswitch.org/18064 > > > ~rodney > This looks like a possible bug. I can replicate on my system. I was able > to use bind_digit_action to work around the issue. Open a Jira on this and > report what you've got. Let me know if you need help trying out > bind_digit_action (BDA). > > -MC > > On Fri, Dec 23, 2011 at 10:31 AM, Rodney wrote: > > any idea why it just hangs up and never gives me the extension? extension > works if i put it on the ivr. > > http://pastebin.freeswitch.org/18064 > > > ~rodney > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/9f3b59e5/attachment-0001.html From notlikeme75 at yahoo.com Sat Dec 24 01:12:34 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Fri, 23 Dec 2011 14:12:34 -0800 (PST) Subject: [Freeswitch-users] bind digit action not working In-Reply-To: References: Message-ID: <1324678354.66975.YahooMailNeo@web65311.mail.ac2.yahoo.com> Michael, I appreciate all the help you have given me but I am still struggling on the BDA. I put it exactly how you did and still it doesn't show in console when i press # and I get no response. ?condition??? ? destination_number??? ? ^501$??? ? 1??? ? ? action??? ? answer??? ? ??? ? 2??? ? ? action??? ? set??? ? conference_user_list=|??? ? 4??? ? ? action??? ? bind_digit_action??? ? xfer,#,exec:transfer,759 XML default??? ? 5??? ? ? action??? ? digit_action_set_realm??? ? xfer??? ? 6??? ? ? action??? ? playback??? ? C:/Program Files/FreeSWITCH/sounds/en/us/callie/digits/8000/1.wav??? ? 8??? ? ? action??? ? conference??? ? 501-127.0.0.1 at default ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Friday, December 23, 2011 4:27 PM Subject: FreeSWITCH-users Digest, Vol 66, Issue 159 ----- Forwarded Message ----- 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: transfer back to ivr after voicemail function ? ? ? (Michael Collins) ? 2. transfer from static conf to dynamic destination??? out of ? ? ? order. (Rodney) ? 3. Re: transfer from static conf to dynamic destination out of ? ? ? order. (Michael Collins) Just have more actions after voicemail in the dialplan. Also, you may wish to modify the voicemail_goodbye macro if you don't want it to say "thank you, goodbye" when exiting VM. -MC On Thu, Dec 22, 2011 at 12:49 PM, Rodney wrote: I am trying to transfer someone back to the IVR after they leave a message for someone or check their own mailbox. currently it does a hangup and they have to call back to do more.? is there a method for this? or is it hard coded in the "press # to continue portion of voice mail? > > > > >check voice mail > > >condition??? ? destination_number??? ? ^757$??? ? 1??? ? >? action??? ? answer??? ? ??? ? 3??? ? >? action??? ? sleep??? ? 1000??? ? 4??? ? >? action??? ? play_and_get_digits??? ? 4 4 3 5000 # checkvoicemail.wav ivr/ivr-that_was_an_invalid_entry.wav vmb \d+??? ? 12??? ? >? action??? ? phrase??? ? spell,${vmb}??? ? 15??? ? >? action??? ? voicemail???? check default ${domain_name} ${vmb} > > > > >leave message > > > > >? condition??? ? destination_number??? ? ^758$??? ? 1??? ? >? action??? ? answer??? ? ??? ? 3??? ? >? action??? ? sleep??? ? 1000??? ? 4??? ? >? action??? ? play_and_get_digits??? ? 4 4 3 5000 # checkvoicemail.wav ivr/ivr-that_was_an_invalid_entry.wav vmb \d+??? ? 12??? ? >? action??? ? phrase??? ? spell,${vmb}??? ? 15??? ? >? action??? ? voicemail???? default ${domain_name} ${vmb} > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > MC, I will open up a jira for this possible bug. any help on the bind digit action would be great. i failed last time i tried. can bind digit action be single digit? this is what i am hoping for but i still want that digit to be "pound" i am thinking it wont work though. my extension into the static conference looks like this. where do put the bind digit action? before i have tried: backforward,7,exec:execute_extension,502 XML default? (so pressing dtmf 7 while in static conf 501 would transfer to 502) but didnt work. didnt even show up as binding the "backforward" realm in console. condition??? ? destination_number??? ? ^501$??? ? 1??? ? ? action??? ? set??? ? conference_user_list=|??? ? 4??? ? ? action??? ? playback??? ? C:/Program Files/FreeSWITCH/sounds/en/us/callie/digits/8000/1.wav??? ? 8??? ? ? action??? ? answer??? ? ??? ? 9??? ? ? action??? ? conference??? ? 501-127.0.0.1 at default any idea why it just hangs up and never gives me the extension? extension works if i put it on the ivr. http://pastebin.freeswitch.org/18064 ~rodney This looks like a possible bug. I can replicate on my system. I was able to use bind_digit_action to work around the issue. Open a Jira on this and report what you've got. Let me know if you need help trying out bind_digit_action (BDA). -MC On Fri, Dec 23, 2011 at 10:31 AM, Rodney wrote: any idea why it just hangs up and never gives me the extension? extension works if i put it on the ivr. > > >http://pastebin.freeswitch.org/18064 > > > >~rodney > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org Edit your conference.conf.xml and remove the # action in the caller controls. Then add something like this to the dialplan prior to going into the conference: Note that you can change the BDA realm to something else if you want to change the meaning of the # key. That's why BDA is so awesome - you can do ANYTHING with it! -MC On Fri, Dec 23, 2011 at 1:05 PM, Rodney wrote: MC, I will open up a jira for this possible bug. any help on the bind digit action would be great. i failed last time i tried. can bind digit action be single digit? this is what i am hoping for but i still want that digit to be "pound" i am thinking it wont work though. my extension into the static conference looks like this. where do put the bind digit action? > >before i have tried: > >backforward,7,exec:execute_extension,502 XML default? (so pressing dtmf 7 while in static conf 501 would transfer to 502) but didnt work. didnt even show up as binding the "backforward" realm in console. > >condition??? ? destination_number??? ? ^501$??? ? 1??? ? >? action??? ? set??? ? conference_user_list=|??? ? 4??? ? >? action??? ? playback??? ? C:/Program Files/FreeSWITCH/sounds/en/us/callie/digits/8000/1.wav??? ? 8??? ? >? action??? ? answer??? ? ??? ? 9??? ? >? action??? ? conference??? ? 501-127.0.0.1 at default > > > > > >any idea why it just hangs up and never gives me the extension? extension works if i put it on the ivr. > > >http://pastebin.freeswitch.org/18064 > > > > >~rodney > >This looks like a possible bug. I can replicate on my system. I was able to use bind_digit_action to work around the issue. Open a Jira on this and report what you've got. Let me know if you need help trying out bind_digit_action (BDA). > > >-MC > > >On Fri, Dec 23, 2011 at 10:31 AM, Rodney wrote: > >any idea why it just hangs up and never gives me the extension? extension works if i put it on the ivr. >> >> >>http://pastebin.freeswitch.org/18064 >> >> >> >>~rodney >> >>_________________________________________________________________________ >>Professional FreeSWITCH Consulting Services: >>consulting at freeswitch.org >>http://www.freeswitchsolutions.com >> >> >> >> >>Official FreeSWITCH Sites >>http://www.freeswitch.org >>http://wiki.freeswitch.org >>http://www.cluecon.com >> >>FreeSWITCH-users mailing list >>FreeSWITCH-users at lists.freeswitch.org >>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>http://www.freeswitch.org >> >> > >_______________________________________________ >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/743d11b0/attachment-0001.html From patrick at sunsus.net Sat Dec 24 00:57:00 2011 From: patrick at sunsus.net (sunsus) Date: Fri, 23 Dec 2011 13:57:00 -0800 (PST) Subject: [Freeswitch-users] CDRLogger Message-ID: <1324677420692-7122659.post@n2.nabble.com> Hello Do somebody know were I can finde the source code of this project: http://wiki.freeswitch.org/wiki/JavaCDRLogger I can't finde it on git. Is there a backup of the svn? Regards Patrick -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/CDRLogger-tp7122659p7122659.html Sent from the freeswitch-users mailing list archive at Nabble.com. From matt at partlytrue.com Sat Dec 24 05:19:08 2011 From: matt at partlytrue.com (Matt Piermarini) Date: Fri, 23 Dec 2011 21:19:08 -0500 Subject: [Freeswitch-users] Help with a Cisco 7970 phone. In-Reply-To: <4EF4ADE9.9030802@partlytrue.com> References: <4EF4ADE9.9030802@partlytrue.com> Message-ID: <4EF5369C.9080101@partlytrue.com> All, Typical as it is, as soon as I sent the help request, I figured it out. I had in the internal.xml, which I guess was forcing NAT on everything in the profile. I took that out and now everything works. That appears to break phones which talk to FS on a random source port, like the cisco, and when NOT running on a nat. I also assumed that running FS with the --nonat would essentially stop all NAT processing. Thank you for OSS. These lines of code in sofia_reg.c gave me the clue, as is_nat WAS being set in my system. if (is_nat && profile->local_network && switch_check_network_list_ip(network_ip, profile->local_network)) { if (profile->debug) { switch_log_printf(SWITCH_CHANNEL_LOG, SWITCH_LOG_DEBUG, "IP %s is on local network, not seting NAT mode.\n", network_ip); } is_nat = NULL; } Is there a possibility to also set is_nat = NULL when --nonat is specified on the cmdline? Thanks again, Matt On 12/23/2011 11:35 AM, Matt Piermarini wrote: > Hello all, > > I have been trying forever to get this Cisco 7970 to work properly > with our freeswitch (latest git). The phone will register fine, and > it will make outbound calls perfectly. The problem is with inbound > calls to this phone, where FS tries to contact it on the wrong UDP > port. The entire system here is on a local lan, and NAT has been > disabled via --nonat cmd line FS parameter. > > Here is what it looks like when registered: > > sofia status profile internal reg > > Call-ID: 00181844-c24f0002-25f0dff2-c470b5a7 at 192.168.2.59 > User: 15 at x.y.com > Contact: "user" > > Agent: Cisco-CP7970G/8.5.2 > Status: Registered(UDP-NAT)(unknown) EXP(2011-12-22 21:31:47) > EXPSECS(724) > Host: x.y.com > IP: 192.168.2.59 > Port: 49165 > Auth-User: 15 > Auth-Realm: x.y.com > > > FS tries to contact the phone using the fs_path, which is getting set > incorrectly. I'm not why FS thinks the fs_path should be different, > as it appears the Cisco is sending proper Contact header in the sip > requests/responses. > Also, we have a bunch of ATA's on the same lan which all work fine > (meaning the fs_path is correct). I tried chaning the > NDLB-force-rport param in the internal context, but didn't seems to > change anything. > Here are the SIP traces which show what's happening. Any ideas of > what's going on would be appreciated. > > Thanks, > Matt > > 192.168.2.59 is the cisco phone. > 192.168.2.1 is FS (listening on port 5070). > > recv 713 bytes from udp/[192.168.2.59]:49320 at 02:28:01.712338: > > ------------------------------------------------------------------------ > REGISTER sip:x.y.com SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.59:5060;branch=z9hG4bKd9e43394 > From: ;tag=00181844c24f0002a0bb4968-7b2b7dd4 > To: > Call-ID: 00181844-c24f0002-eb21e8a8-77681214 at 192.168.2.59 > Max-Forwards: 70 > Date: Fri, 23 Dec 2011 02:28:00 GMT > CSeq: 101 REGISTER > User-Agent: Cisco-CP7970G/8.5.2 > Contact: > ;+sip.instance="";+u.sip!model.ccm.cisco.com="30006" > Supported: (null),X-cisco-xsi-7.0.1 > Content-Length: 0 > Reason: SIP;cause=200;text="cisco-alarm:20 Name=SEP00181844C24F > Load=SIP70.8-5-2S Last=phone-keypad" > Expires: 3600 > > send 689 bytes to udp/[192.168.2.59]:5060 at 02:28:01.715057: > > ------------------------------------------------------------------------ > SIP/2.0 401 Unauthorized > Via: SIP/2.0/UDP 192.168.2.59:5060;branch=z9hG4bKd9e43394 > From: ;tag=00181844c24f0002a0bb4968-7b2b7dd4 > To: ;tag=Qv2N52egZ2teH > Call-ID: 00181844-c24f0002-eb21e8a8-77681214 at 192.168.2.59 > CSeq: 101 REGISTER > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-8059cdc 2011-12-22 > 14-03-32 -0600 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > WWW-Authenticate: Digest realm="x.y.com", > nonce="fae142ab-8b00-4ac7-89d7-04955e8714f6", algorithm=MD5, qop="auth" > Content-Length: 0 > > recv 958 bytes from udp/[192.168.2.59]:49320 at 02:28:01.725802: > > ------------------------------------------------------------------------ > REGISTER sip:x.y.com SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.59:5060;branch=z9hG4bK18d5a468 > From: ;tag=00181844c24f0002a0bb4968-7b2b7dd4 > To: > Call-ID: 00181844-c24f0002-eb21e8a8-77681214 at 192.168.2.59 > Max-Forwards: 70 > Date: Fri, 23 Dec 2011 02:28:00 GMT > CSeq: 102 REGISTER > User-Agent: Cisco-CP7970G/8.5.2 > Contact: > ;+sip.instance="";+u.sip!model.ccm.cisco.com="30006" > Authorization: Digest > username="15",realm="x.y.com",uri="sip:x.y.com",response="xxxxxxxxxxxxxxxxxxxxxxxxxxxxxx",nonce="fae142ab-8b00-4ac7-89d7-04955e8714f6",cnonce="4f2a64d4",qop=auth,nc=00000001,algorithm=MD5 > Supported: (null),X-cisco-xsi-7.0.1 > Content-Length: 0 > Reason: SIP;cause=200;text="cisco-alarm:20 Name=SEP00181844C24F > Load=SIP70.8-5-2S Last=phone-keypad" > Expires: 3600 > > send 649 bytes to udp/[192.168.2.59]:5060 at 02:28:01.737460: > > ------------------------------------------------------------------------ > SIP/2.0 200 OK > Via: SIP/2.0/UDP 192.168.2.59:5060;branch=z9hG4bK18d5a468 > From: ;tag=00181844c24f0002a0bb4968-7b2b7dd4 > To: ;tag=SeN78rgQSm7Kr > Call-ID: 00181844-c24f0002-eb21e8a8-77681214 at 192.168.2.59 > CSeq: 102 REGISTER > Contact: ;expires=3600 > Date: Fri, 23 Dec 2011 02:28:01 GMT > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-8059cdc 2011-12-22 > 14-03-32 -0600 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Content-Length: 0 > > *--------------- OK, so far so good.. phone is now registered.. Now > look how FS tries to contact it.. sending notify's here: Note the > port it tries to use, 49320 in this case.* > > send 1034 bytes to udp/[192.168.2.59]:49320 at 02:28:01.837149: > > ------------------------------------------------------------------------ > NOTIFY > sip:15 at 192.168.2.59:5060;transport=udp;fs_nat=yes;fs_path=sip:15%40192.168.2.59:49320%3Btransport%3Dudp > SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.1:5070;rport;branch=z9hG4bKv92r0r53rQKNa > Route: ;transport=udp > Max-Forwards: 70 > From: ;tag=U07rcFjyK6KSF > To: > Call-ID: 93e179d6-a7b0-122f-d396-5254003b348b > CSeq: 21967576 NOTIFY > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-8059cdc 2011-12-22 > 14-03-32 -0600 > Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, UPDATE, INFO, > REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE > Supported: timer, precondition, path, replaces > Event: message-summary > Allow-Events: talk, hold, presence, dialog, line-seize, call-info, > sla, include-session-description, presence.winfo, message-summary, refer > Subscription-State: terminated;reason=timeout > Content-Type: application/simple-message-summary > Content-Length: 69 > > Messages-Waiting: no > Message-Account: sip:15 at x.y.com > > *The phone never responds, as FS is supposed to contact it on port > 5060, but FS is trying to use the same port that phone sent is SIP > messages on. Here is a sample INVITE:* > > send 1229 bytes to udp/[192.168.2.59]:49320 at 02:38:15.217775: > > ------------------------------------------------------------------------ > INVITE sip:15 at 192.168.2.59:5060;transport=udp SIP/2.0 > Via: SIP/2.0/UDP 192.168.2.1:5070;rport;branch=z9hG4bK0D8U748HDUc0r > Route: ;transport=udp > Max-Forwards: 69 > From: "x x x" ;tag=63X9ZrB5K886e > To: > Call-ID: 00e33727-a7b2-122f-d396-5254003b348b > CSeq: 21967883 INVITE > Contact: > User-Agent: FreeSWITCH-mod_sofia/1.0.head-git-8059cdc 2011-12-22 > 14-03-32 -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-Type: application/sdp > Content-Disposition: session > Content-Length: 204 > X-FS-Support: update_display,send_info > Remote-Party-ID: "x x x" > ;party=calling;screen=yes;privacy=off > > v=0 > o=FreeSWITCH 1324545412 1324545413 IN IP4 192.168.2.1 > s=FreeSWITCH > c=IN IP4 192.168.2.1 > t=0 0 > m=audio 62482 RTP/AVP 18 0 8 101 13 > a=rtpmap:101 telephone-event/8000 > a=fmtp:101 0-16 > a=ptime:20 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/a957d49f/attachment-0001.html From georg at riseup.net Sat Dec 24 07:47:11 2011 From: georg at riseup.net (georg at riseup.net) Date: Sat, 24 Dec 2011 05:47:11 +0100 Subject: [Freeswitch-users] int/ext dial tones not synchronous In-Reply-To: References: Message-ID: Hi, > I'm not sure that I understand what the issue is here. What two "tones" > are > not in sync? You said "dial tone" but I'm pretty sure that's not what you > mean. Are you talking about the ringback tone that the caller hears? Exactly Michael, I was talking about the ringback tone. I didn't know the correct expression and used a translation engine... Georg From sherifomran2000 at yahoo.com Sat Dec 24 08:05:54 2011 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Fri, 23 Dec 2011 21:05:54 -0800 (PST) Subject: [Freeswitch-users] Radio stream binding ? In-Reply-To: Message-ID: <1324703154.9585.YahooMailClassic@web110808.mail.gq1.yahoo.com> but the station i need is this mmsh://live.sis.gov.eg/live?MSWMExt=.asf and it is not there. I would appreciate your help kind regards, Sherif Omran --- On Fri, 12/23/11, Michael Collins wrote: From: Michael Collins Subject: Re: [Freeswitch-users] Radio stream binding ? To: "FreeSWITCH Users Help" Date: Friday, December 23, 2011, 8:03 PM there are thousands of stations you can get at shoutcast.com. those work very well. -MC On Thu, Dec 22, 2011 at 5:00 PM, Sherif Omran wrote: hello guys, can any body help me to use bind_meta_app to connect a channel to a radio stream using gnuradio.org project? thanks in advance 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 -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/ec883f65/attachment.html From govoiper at gmail.com Sat Dec 24 16:22:03 2011 From: govoiper at gmail.com (Sammy Govind) Date: Sat, 24 Dec 2011 18:22:03 +0500 Subject: [Freeswitch-users] LUA transfer result:0 In-Reply-To: References: <2C9B38E79A44400D8F7509B1010A00C4@omni1.local> Message-ID: Hi, I'm facing same situation. I read the Lua transfer page: http://wiki.freeswitch.org/wiki/Mod_lua#session:transfer which states that I need to set AutoHangup to false. I did that too but my transfer command still fails :( I've to do looping between a list of numbers and each number could have different XML script to be run.So as soon as I parse CSV I find out the XML context on which to transfer the call and do a transfer, but it fails immediately. Hope to hear from you soon. Regards, Sammy On Sat, Oct 22, 2011 at 3:35 AM, Michael Collins wrote: > Don't forget that in a Lua dp script the script must "end" before the > transfer takes place. Make sure that there isn't a logic loop or something > in your code preventing it from ending. If you pastebin your script we'll > take a look. > -MC > > On Thu, Oct 20, 2011 at 2:27 PM, Anestis Mavro wrote: > >> Hi,**** >> >> ** ** >> >> I have tried two different ways of transfer in LUA:**** >> >> session:transfer(number,?XML?,?default?)**** >> >> ** ** >> >> And **** >> >> ** ** >> >> Session:execute(?transfer?,?12345 XML default?)**** >> >> ** ** >> >> And I can?t make it working.**** >> >> ** ** >> >> I get ?switch_cpp.cpp:797 transfer result: 0?**** >> >> ** ** >> >> The same transfer from within the dialplan works fine**** >> >> ** ** >> >> The call is not answered before the transfer.**** >> >> ** ** >> >> Does anybody have an idea?**** >> >> ** ** >> >> Thanks**** >> >> Anestis**** >> >> ** ** >> >> >> __________ Information from ESET NOD32 Antivirus, version of virus >> signature database 5054 (20100423) __________ >> >> The message was checked by ESET NOD32 Antivirus. >> >> http://www.eset.com >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111224/4c7bf5ad/attachment.html From david.villasmil.work at gmail.com Sat Dec 24 16:33:57 2011 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 24 Dec 2011 14:33:57 +0100 Subject: [Freeswitch-users] how many luas scripts are running? Message-ID: Hello gang, Is there any command that shows how many luas are running? Thanks David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111224/362317ba/attachment.html From notlikeme75 at yahoo.com Sat Dec 24 16:39:46 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Sat, 24 Dec 2011 05:39:46 -0800 (PST) Subject: [Freeswitch-users] bind digit action References: Message-ID: <1324733986.20908.YahooMailNeo@web65303.mail.ac2.yahoo.com> Michael, I got BDA to work by manually configuring the dialplan, seems fusionpbx was not doing it right when i added the actions. but now that I have it working, my issue is using 0 as a transfer back to main menu. is there a way to clear digit action on an IVR menu? scenario i have 9 static conference rooms bind digits are 0 transfer back to ivr room list 6 to move back in room list 7 to move up the room list to make this work, i clear bind digit on every conf room extension. the problem is my ivr has option 0 to go back to previous menu, but its not seen once you go into the room list bindings. I know making the IVR binding a different number would work, but I would like to keep with the consistent 0 to go back to previous menu. if there was a method to clear bind digit in the IVR, that would be great. I have tried before the ivr answer and after, with no luck. ie. pressing 0 , should go to room menu, and 0 again should go to main menu, but as of now, pressing 0 is binded to room menu only :( another good thing would be, i could set an expression with boundaries so i would only need 1 extension or 1 conference control like a low boundary of 501 and a high boundary of 509. so if someone presses 6 it would move down the list variable {6=current conf # - 1 but if in room 1, go to room 9, and if in room 9 pressing 7 would start over at 1} i accomplished this the long way with 9 separate extensions using bind digits to move up or down the list. i had to manually add a transfer for the next extension. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111224/dcfee54f/attachment-0001.html From ryan at kaevee.com Sat Dec 24 16:57:41 2011 From: ryan at kaevee.com (Ryan V) Date: Sat, 24 Dec 2011 19:27:41 +0530 Subject: [Freeswitch-users] Caller ID on Analog Lines Message-ID: Hi, Thanks to all the support we got in mailing lists and our telco, we put our freeswitch PBX online today. We are using Sangoma A101 and A200 Cards. We are successfully receiving the calls on our PRI and Analogue lines. But, we don't get caller-id on our analogue Lines. In India, Caller id comes in as DTMF tones and I could see the caller ID being detected correctly in debug. I have seen this issue raised earlier on the list and Moises Silva from Sangoma offered to fix it if given access to the server. Here is the link http://comments.gmane.org/gmane.comp.telephony.freeswitch.user/32340 I wanted to know whether anyone has come up with a solution. I am open to give access to our server if Moises Silva or anyone else in Sangoma would like to have a look at it. Thanks, Ryan -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111224/fd9e613a/attachment.html From david.villasmil.work at gmail.com Sat Dec 24 18:01:45 2011 From: david.villasmil.work at gmail.com (David Villasmil) Date: Sat, 24 Dec 2011 16:01:45 +0100 Subject: [Freeswitch-users] Continue after lua execute bridge Message-ID: Hello Gang, I'm executing from a lua script: session:execute("bridge","sofia/gateway/" .. gw .. "/" .. out_number .. "") But I want my script to END if the bridge is succesfull, i.e.: Ringing, Answer, Busy... I've been testing, but it doesn't seem like the script is aware of the result of the bridge until AFTER it has been release by the "bridge" application... Is there ANY way to end my script in any of the scenarious mentioned before? (Ringing, Answer, Busy) Thanks a lot oh, and MERRY CHRISTMAS or Happy Holidays, or whatever suits you! ;) David -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111224/3e615b6c/attachment.html From govoiper at gmail.com Sat Dec 24 19:04:38 2011 From: govoiper at gmail.com (Sammy Govind) Date: Sat, 24 Dec 2011 21:04:38 +0500 Subject: [Freeswitch-users] LUA transfer result:0 In-Reply-To: References: <2C9B38E79A44400D8F7509B1010A00C4@omni1.local> Message-ID: I got it working, FS requires a "return 0" in LUA script just after session:transfer() command. On Sat, Dec 24, 2011 at 6:22 PM, Sammy Govind wrote: > Hi, > I'm facing same situation. I read the Lua transfer page: > http://wiki.freeswitch.org/wiki/Mod_lua#session:transfer which states > that I need to set AutoHangup to false. I did that too but my transfer > command still fails :( > > I've to do looping between a list of numbers and each number could have > different XML script to be run.So as soon as I parse CSV I find out the XML > context on which to transfer the call and do a transfer, but it fails > immediately. > > Hope to hear from you soon. > Regards, > Sammy > > > On Sat, Oct 22, 2011 at 3:35 AM, Michael Collins wrote: > >> Don't forget that in a Lua dp script the script must "end" before the >> transfer takes place. Make sure that there isn't a logic loop or something >> in your code preventing it from ending. If you pastebin your script we'll >> take a look. >> -MC >> >> On Thu, Oct 20, 2011 at 2:27 PM, Anestis Mavro wrote: >> >>> Hi,**** >>> >>> ** ** >>> >>> I have tried two different ways of transfer in LUA:**** >>> >>> session:transfer(number,?XML?,?default?)**** >>> >>> ** ** >>> >>> And **** >>> >>> ** ** >>> >>> Session:execute(?transfer?,?12345 XML default?)**** >>> >>> ** ** >>> >>> And I can?t make it working.**** >>> >>> ** ** >>> >>> I get ?switch_cpp.cpp:797 transfer result: 0?**** >>> >>> ** ** >>> >>> The same transfer from within the dialplan works fine**** >>> >>> ** ** >>> >>> The call is not answered before the transfer.**** >>> >>> ** ** >>> >>> Does anybody have an idea?**** >>> >>> ** ** >>> >>> Thanks**** >>> >>> Anestis**** >>> >>> ** ** >>> >>> >>> __________ Information from ESET NOD32 Antivirus, version of virus >>> signature database 5054 (20100423) __________ >>> >>> The message was checked by ESET NOD32 Antivirus. >>> >>> http://www.eset.com >>> >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >>> http://www.freeswitch.org >>> >>> >> >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> >> > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111224/71c17544/attachment.html From shaheryarkh at googlemail.com Sat Dec 24 23:03:07 2011 From: shaheryarkh at googlemail.com (Muhammad Shahzad) Date: Sun, 25 Dec 2011 01:03:07 +0500 Subject: [Freeswitch-users] Caller ID on Analog Lines In-Reply-To: References: Message-ID: I think India uses polarity reversal to indicate start of caller id on PRI line, which i guess is not fully supported by FreeTDM yet. However, you can try the following FreeTDM options, though i am not sure if they will work. You may try to set last parameter to false and test again. Do let us know if it works or not. I am not much familiar with FreeTDM code, nor got the equipment to test, so for those who have, please see below link regarding same problem in Asterisk and how they solved it, https://issues.asterisk.org/print_bug_page.php?bug_id=6683 Hope this helps. Thank you. On Sat, Dec 24, 2011 at 6:57 PM, Ryan V wrote: > Hi, > > Thanks to all the support we got in mailing lists and our telco, we put > our freeswitch PBX online today. > > We are using Sangoma A101 and A200 Cards. We are successfully receiving > the calls on our PRI and Analogue lines. But, we don't get caller-id on our > analogue Lines. In India, Caller id comes in as DTMF tones and I could see > the caller ID being detected correctly in debug. > > I have seen this issue raised earlier on the list and Moises Silva from > Sangoma offered to fix it if given access to the server. Here is the link > http://comments.gmane.org/gmane.comp.telephony.freeswitch.user/32340 > > I wanted to know whether anyone has come up with a solution. I am open to > give access to our server if Moises Silva or anyone else in Sangoma would > like to have a look at it. > > Thanks, > > 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 > > -- 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/20111225/6d12c3d7/attachment.html From sherifomran2000 at yahoo.com Sat Dec 24 23:19:31 2011 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Sat, 24 Dec 2011 12:19:31 -0800 (PST) Subject: [Freeswitch-users] loopback number In-Reply-To: Message-ID: <1324757971.95056.YahooMailClassic@web110803.mail.gq1.yahoo.com> Hi, I want to create a loopback number but dont know how. Could any body tell me please? I installed bluebox and this resulted in having the default configurations overwritten. Any help is appreciated kind regards, Sherif -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111224/3f465a9f/attachment-0001.html From ryan at kaevee.com Sun Dec 25 05:12:25 2011 From: ryan at kaevee.com (Ryan V) Date: Sun, 25 Dec 2011 07:42:25 +0530 Subject: [Freeswitch-users] Caller ID on Analog Lines In-Reply-To: References: Message-ID: Caller ID on PRI line works fine. I have problem only with calls coming in on FXO. Thanks, Venkatesh K On Sun, Dec 25, 2011 at 1:33 AM, Muhammad Shahzad < shaheryarkh at googlemail.com> wrote: > I think India uses polarity reversal to indicate start of caller id on PRI > line, which i guess is not fully supported by FreeTDM yet. However, you can > try the following FreeTDM options, though i am not sure if they will work. > > > > > > You may try to set last parameter to false and test again. Do let us know > if it works or not. > > I am not much familiar with FreeTDM code, nor got the equipment to test, > so for those who have, please see below link regarding same problem in > Asterisk and how they solved it, > > https://issues.asterisk.org/print_bug_page.php?bug_id=6683 > > Hope this helps. > > Thank you. > > > On Sat, Dec 24, 2011 at 6:57 PM, Ryan V wrote: > >> Hi, >> >> Thanks to all the support we got in mailing lists and our telco, we put >> our freeswitch PBX online today. >> >> We are using Sangoma A101 and A200 Cards. We are successfully receiving >> the calls on our PRI and Analogue lines. But, we don't get caller-id on our >> analogue Lines. In India, Caller id comes in as DTMF tones and I could see >> the caller ID being detected correctly in debug. >> >> I have seen this issue raised earlier on the list and Moises Silva from >> Sangoma offered to fix it if given access to the server. Here is the link >> http://comments.gmane.org/gmane.comp.telephony.freeswitch.user/32340 >> >> I wanted to know whether anyone has come up with a solution. I am open to >> give access to our server if Moises Silva or anyone else in Sangoma would >> like to have a look at it. >> >> Thanks, >> >> 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 >> >> > > > -- > Muhammad Shahzad > ----------------------------------- > CISCO Rich Media Communication Specialist (CRMCS) > CISCO Certified Network Associate (CCNA) > Cell: +92 334 422 40 88 > MSN: shari_786pk at hotmail.com > Email: shaheryarkh at googlemail.com > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111225/4476efb1/attachment.html From notlikeme75 at yahoo.com Sun Dec 25 19:20:40 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Sun, 25 Dec 2011 08:20:40 -0800 (PST) Subject: [Freeswitch-users] conf count In-Reply-To: References: Message-ID: <1324830040.46726.YahooMailNeo@web65306.mail.ac2.yahoo.com> I can select the conf-alone sound in caller controls and even set up a bind digit for caller count once i am already in the conference using the Conference Announce Count Inline? extension. but what i cant figure out is how to speak the caller count when entering the conference room.? can anyone help with this? currently with conf-alone it only speaks the file when i am the only one left or there isnt anyone else in the room when i get there. i would like it to tell me the total callers of the room i am heading into. thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111225/aa10fbfd/attachment.html From notlikeme75 at yahoo.com Mon Dec 26 02:25:22 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Sun, 25 Dec 2011 15:25:22 -0800 (PST) Subject: [Freeswitch-users] mod_shout on win32 In-Reply-To: References: Message-ID: <1324855522.90986.YahooMailNeo@web65304.mail.ac2.yahoo.com> how do i get mod_shout on windows? it does not get installed with the msi. I would love to stream some internet stations to a conference. or use as MOH. thanks. windows server 2008 32bit. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111225/010f185d/attachment.html From sdevoy at bizfocused.com Sat Dec 24 02:54:54 2011 From: sdevoy at bizfocused.com (Sean Devoy) Date: Fri, 23 Dec 2011 18:54:54 -0500 Subject: [Freeswitch-users] ./configure still fails with error: libtermcap, libcurses or libncurses are required! Message-ID: <0a3101ccc1ce$44ad65e0$ce0831a0$@com> Hi, I am a NooB to FreeSwitch. Sorry if I should start elsewhere, I can't figure out where. I ran the recommended installation instructions, including installing Centos 5.7 from boot DVD. I installed and compiled GIT, regardless of the errors in the install guide. I ran: cd /usr/local/src git clone git://git.freeswitch.org/freeswitch.git I went to: cd /usr/local/src/freeswitch Ran: ./bootstrap.sh That required me to "yum install " autoconf, automake and lib??? (I dont recall), then it would run. And tried: ./configure Which of course terminates with: configure: error: libtermcap, libcurses or libncurses are required! I have "yum install"ed libtermcap, libcurse,libncurese,termcap,ncurses,curses and everything else I can possibly find a reference to. That help a lot (a.k.a. NONE). Now I get: configure: error: libtermcap, libcurses or libncurses are required! I have googled and read and searched. I was happy to see where this was a known problem. I was disappointed to see NO help on what to do since it was "fixed". I am trying not to ask dumb questions, but I am struggling here. It is possible for a mere mortal to install FreeSwitch on a new system? Is there a set of install instructions that match the current realease? I am fighting the urge to throw Linux out the window and go with the Windows build. I have made a living of MS products for over 20 years, but there seems to be nobody left there with a brain. I am trying to move on. Thanks in advance, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111223/dd241f13/attachment-0001.html From faheem_imt at yahoo.com Sun Dec 25 20:19:20 2011 From: faheem_imt at yahoo.com (Faheem) Date: Sun, 25 Dec 2011 09:19:20 -0800 (PST) Subject: [Freeswitch-users] help required regarding Bridge application In-Reply-To: <1324830040.46726.YahooMailNeo@web65306.mail.ac2.yahoo.com> References: <1324830040.46726.YahooMailNeo@web65306.mail.ac2.yahoo.com> Message-ID: <1324833560.786.YahooMailNeo@web161206.mail.bf1.yahoo.com> Hi all, I am new in this list and this is my first email to this group.? I need help regarding Bridge application using Lua script. I want to set the auto hangup for a bridge call in lua script. my script originate leg-A to local extension and then originate leg-b to a carrier and then bridge two legs. I want to hangup the leg-b after some seconds say 20 seconds. I did the?available?option, but nothing worked for me. Please help me.? Here is my lua code. ---------------------------------------------------------------------------------------------------------------- originate_str1 = "{origination_caller_id_number=".. caller.. ",effective_caller_id_number=".. caller ..",originate_timeout=45,ignore_early_media=true}sofia/internal/".. caller .."%" ..domain ..""; originate_str2 = "{origination_caller_id_number=".. callee.. ",effective_caller_id_number=".. callee ..",originate_timeout=60,sched_hangup=20,ignore_early_media=true}sofia/internal/".. carrier_dial_prefix1 .. callee .."%" ..carrier_gateway1 ..""; session1 = freeswitch.Session(originate_str1); session2 = freeswitch.Session(originate_str2); freeswitch.bridge(session1, session2); Thanks!? Faheem Muhammad -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111225/1c421be5/attachment-0001.html From vetali100 at gmail.com Mon Dec 26 02:51:45 2011 From: vetali100 at gmail.com (Vitalie Colosov) Date: Sun, 25 Dec 2011 15:51:45 -0800 Subject: [Freeswitch-users] ./configure still fails with error: libtermcap, libcurses or libncurses are required! In-Reply-To: <0a3101ccc1ce$44ad65e0$ce0831a0$@com> References: <0a3101ccc1ce$44ad65e0$ce0831a0$@com> Message-ID: You need to install few other packages - full list is: yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make ncurses-devel pkgconfig You can find this description at the following section of the installation guide: http://wiki.freeswitch.org/wiki/Installation_Guide#CentOS Also, you might want to install the optional packages: yum install unixODBC-devel openssl-devel gnutls-devel libogg-devel libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel expat-devel zlib zlib-devel bzip2 which Regards, Vitalie 2011/12/23 Sean Devoy > Hi,**** > > ** ** > > I am a NooB to FreeSwitch. Sorry if I should start elsewhere, I can?t > figure out where.**** > > ** ** > > I ran the recommended installation instructions, including installing > Centos 5.7 from boot DVD.**** > > ** ** > > I installed and compiled GIT, regardless of the errors in the install > guide.**** > > I ran:**** > > cd /usr/local/src**** > > git clone git://git.freeswitch.org/freeswitch.git**** > > ** ** > > ** ** > > I went to:**** > > cd /usr/local/src/freeswitch**** > > Ran:**** > > ./bootstrap.sh**** > > That required me to ?yum install ? autoconf, automake and lib??? (I dont > recall), then it would run.**** > > ** ** > > And tried:**** > > ./configure**** > > Which of course terminates with:**** > > configure: error: libtermcap, libcurses or libncurses are required!**** > > ** ** > > I have ?yum install?ed libtermcap, > libcurse,libncurese,termcap,ncurses,curses and everything else I can > possibly find a reference to. That help a lot (a.k.a. NONE). Now I get:* > *** > > configure: error: libtermcap, libcurses or libncurses are required!**** > > ** ** > > I have googled and read and searched. I was happy to see where this was a > known problem. I was disappointed to see NO help on what to do since it > was ?fixed?.**** > > ** ** > > I am trying not to ask dumb questions, but I am struggling here.**** > > ** ** > > It is possible for a mere mortal to install FreeSwitch on a new system? > Is there a set of install instructions that match the current realease?** > ** > > ** ** > > I am fighting the urge to throw Linux out the window and go with the > Windows build. I have made a living of MS products for over 20 years, but > there seems to be nobody left there with a brain. I am trying to move on. > **** > > ** ** > > Thanks in advance,**** > > Sean**** > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111225/0de594f5/attachment.html From curriegrad2004 at gmail.com Mon Dec 26 02:59:46 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Sun, 25 Dec 2011 15:59:46 -0800 Subject: [Freeswitch-users] ./configure still fails with error: libtermcap, libcurses or libncurses are required! In-Reply-To: References: <0a3101ccc1ce$44ad65e0$ce0831a0$@com> Message-ID: ncurses is a pre-requsite for FreeSWITCH to run properly in the first place. On Sun, Dec 25, 2011 at 3:51 PM, Vitalie Colosov wrote: > You need to install few other packages - full list is: > > yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make > ncurses-devel pkgconfig > > > You can find this description at the following section of the installation > guide: > http://wiki.freeswitch.org/wiki/Installation_Guide#CentOS > > Also, you might want to install the optional packages: > > yum install unixODBC-devel openssl-devel gnutls-devel libogg-devel > libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel > expat-devel zlib zlib-devel bzip2 which > > > Regards, > Vitalie > > 2011/12/23 Sean Devoy >> >> Hi, >> >> >> >> I am a NooB to FreeSwitch.? Sorry if I should start elsewhere, I can?t >> figure out where. >> >> >> >> I ran the recommended installation instructions, including installing >> Centos 5.7 from boot DVD. >> >> >> >> I installed and compiled GIT, regardless of the errors in the install >> guide. >> >> I ran: >> >> cd /usr/local/src >> >> git clone git://git.freeswitch.org/freeswitch.git >> >> >> >> >> >> I went to: >> >> cd /usr/local/src/freeswitch >> >> Ran: >> >> ./bootstrap.sh >> >> That required me to ?yum install ?? autoconf, automake and lib??? (I dont >> recall), then it would run. >> >> >> >> And tried: >> >> ./configure >> >> Which of course terminates with: >> >> configure: error: libtermcap, libcurses or libncurses are required! >> >> >> >> I have ?yum install?ed libtermcap, >> libcurse,libncurese,termcap,ncurses,curses and everything else I can >> possibly find a reference to.? That help a lot (a.k.a. NONE).? Now I get: >> >> configure: error: libtermcap, libcurses or libncurses are required! >> >> >> >> I have googled and read and searched.? I was happy to see where this was a >> known problem.? I was disappointed to see NO help on what to do since it was >> ?fixed?. >> >> >> >> I am trying not to ask dumb questions, but I am struggling here. >> >> >> >> It is possible for a mere mortal to install FreeSwitch on a new system? >> ?Is there a set of install instructions that match the current realease? >> >> >> >> I am fighting the urge to throw Linux out the window and go with the >> Windows build.? I have made a living of MS products for over 20 years, but >> there seems to be nobody left there with a brain.? I am trying to move on. >> >> >> >> Thanks in advance, >> >> Sean >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From jeff at jefflenk.com Mon Dec 26 06:19:54 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Sun, 25 Dec 2011 19:19:54 -0800 (PST) Subject: [Freeswitch-users] mod_shout on win32 In-Reply-To: <1324855522.90986.YahooMailNeo@web65304.mail.ac2.yahoo.com> References: <1324855522.90986.YahooMailNeo@web65304.mail.ac2.yahoo.com> Message-ID: <1324869594666-7127192.post@n2.nabble.com> that module is not built in the default package. you will have to build from source to get that module as it stands. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-shout-on-win32-tp7126980p7127192.html Sent from the freeswitch-users mailing list archive at Nabble.com. From govoiper at gmail.com Mon Dec 26 08:19:47 2011 From: govoiper at gmail.com (Sammy Govind) Date: Mon, 26 Dec 2011 10:19:47 +0500 Subject: [Freeswitch-users] help required regarding Bridge application In-Reply-To: <1324833560.786.YahooMailNeo@web161206.mail.bf1.yahoo.com> References: <1324830040.46726.YahooMailNeo@web65306.mail.ac2.yahoo.com> <1324833560.786.YahooMailNeo@web161206.mail.bf1.yahoo.com> Message-ID: Hi, Please see the below script by MC posted a few days earlier. Change the data accordingly. Also see this http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_sched_hangup Try setting the schedule hangup parameter before calling Bridge, I don't think this has any effect while in originate string. -- test_call -- create first leg, wait, then connect 2nd leg -- local calling_user = argv[1]; local called_num = argv[2]; freeswitch.consoleLog("INFO","Attempting to contact user " .. calling_user .. "\n"); local session1 = freeswitch.Session("{origination_caller_id_number=9876}user/"..calling_user); leg1_dispo = 'None'; while (session1:ready() == true and leg1_dispo ~= 'ANSWER') do leg1_dispo = session1:getVariable("endpoint_disposition"); freeswitch.consoleLog("INFO","Leg 1 disposition:" .. leg1_dispo .. "\n"); os.execute("sleep 1"); end if ( not session1:ready() ) then -- Oops, leg 1 hung up. Bummer. freeswitch.consoleLog("INFO","It appears that " .. calling_user .. " disconnected...\n") else freeswitch.consoleLog("INFO","Playing a prompt to " .. calling_user .. "\n"); session1:streamFile('ivr/ivr-hold_connect_call.wav'); session2 = freeswitch.Session("{origination_caller_id_number=" .. calling_user .."}user/" .. called_num); leg2_dispo = 'None'; while(session1:ready() and session2:ready() and leg2_dispo ~= "ANSWER") do if ( not session1:ready() ) then -- oops, leg 1 hung up freeswitch.consoleLog("INFO","Well, it appears that " .. calling_user .. " has disconnected.\n"); else os.execute("sleep 1"); leg2_dispo = session2:getVariable("endpoint_disposition"); freeswitch.consoleLog("INFO","Leg 2 disposition: " .. leg2_dispo .. "\n"); end end -- While if ( session1:ready() and session2:ready() ) then -- Looks good, bridge 'em freeswitch.bridge(session1,session2); else -- Uh oh, someone went away freeswitch.consoleLog("INFO","Somebody hung up :(\n"); end end Regards, Sammy. On Sun, Dec 25, 2011 at 10:19 PM, Faheem wrote: > Hi all, I am new in this list and this is my first email to this group. > I need help regarding Bridge application using Lua script. > I want to set the auto hangup for a bridge call in lua script. my script > originate leg-A to local extension and then originate leg-b to a carrier > and then bridge two legs. I want to hangup the leg-b after some seconds say > 20 seconds. > I did the available option, but nothing worked for me. Please help me. > > Here is my lua code. > > > ---------------------------------------------------------------------------------------------------------------- > originate_str1 = "{origination_caller_id_number=".. caller.. > ",effective_caller_id_number=".. caller > ..",originate_timeout=45,ignore_early_media=true}sofia/internal/".. caller > .."%" ..domain ..""; > > originate_str2 = "{origination_caller_id_number=".. callee.. > ",effective_caller_id_number=".. callee ..",originate_timeout=60, > sched_hangup=20,ignore_early_media=true}sofia/internal/".. > carrier_dial_prefix1 .. callee .."%" ..carrier_gateway1 ..""; > > session1 = freeswitch.Session(originate_str1); > session2 = freeswitch.Session(originate_str2); > freeswitch.bridge(session1, session2); > > Thanks! > Faheem Muhammad > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111226/cb5e5c51/attachment-0001.html From tculjaga at gmail.com Mon Dec 26 11:13:13 2011 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 26 Dec 2011 09:13:13 +0100 Subject: [Freeswitch-users] CUSTOM events Message-ID: hello, please, can anyone tell how do i subscribe to these events ? i tried "events plain CUSTOM", fs acepted it but i got no events... using "events plain all" i can get them but with all others i don't really want. Event-Subclass: spandsp%3A%3Arxfaxpageresult Event-Name: CUSTOM Core-UUID: df6035f5-552b-40ce-a66f-d4de0b0f9527 FreeSWITCH-Hostname: l01sipindirpp FreeSWITCH-Switchname: l01sipindirpp FreeSWITCH-IPv4: 85.114.35.241 FreeSWITCH-IPv6: 2a02%3Aac8%3Ad%3A924%3A211%3Aaff%3Afe5b%3Ada1c Event-Date-Local: 2011-12-25%2019%3A29%3A50 Event-Date-GMT: Sun,%2025%20Dec%202011%2020%3A29%3A50%20GMT Event-Date-Timestamp: 1324844990773140 Event-Calling-File: mod_spandsp_fax.c Event-Calling-Function: phase_d_handler Event-Calling-Line-Number: 439 fax-document-transferred-pages: 6 fax-image-resolution: 8031x3850 fax-image-size: 19764 fax-image-pixel-size: 1728x1094 fax-bad-rows: 0 fax-longest-bad-row-run: 0 fax-encoding: 3 fax-encoding-name: T.6 Content-Length: 734 Content-Type: text/event-plain Event-Subclass: spandsp%3A%3Atxfaxpageresult Event-Name: CUSTOM Core-UUID: df6035f5-552b-40ce-a66f-d4de0b0f9527 FreeSWITCH-Hostname: l01sipindirpp FreeSWITCH-Switchname: l01sipindirpp FreeSWITCH-IPv4: 85.114.35.241 FreeSWITCH-IPv6: 2a02%3Aac8%3Ad%3A924%3A211%3Aaff%3Afe5b%3Ada1c Event-Date-Local: 2011-12-25%2019%3A29%3A51 Event-Date-GMT: Sun,%2025%20Dec%202011%2020%3A29%3A51%20GMT Event-Date-Timestamp: 1324844991976624 Event-Calling-File: mod_spandsp_fax.c Event-Calling-Function: phase_d_handler Event-Calling-Line-Number: 439 fax-document-transferred-pages: 6 fax-image-resolution: 8031x3850 fax-image-size: 19763 fax-image-pixel-size: 1728x1078 fax-bad-rows: 0 fax-longest-bad-row-run: 0 fax-encoding: 3 fax-encoding-name: T.6 Event-Subclass: spandsp%3A%3Arxfaxresult Event-Name: CUSTOM Core-UUID: df6035f5-552b-40ce-a66f-d4de0b0f9527 FreeSWITCH-Hostname: l01sipindirpp FreeSWITCH-Switchname: l01sipindirpp FreeSWITCH-IPv4: 85.114.35.241 FreeSWITCH-IPv6: 2a02%3Aac8%3Ad%3A924%3A211%3Aaff%3Afe5b%3Ada1c Event-Date-Local: 2011-12-25%2019%3A29%3A54 Event-Date-GMT: Sun,%2025%20Dec%202011%2020%3A29%3A54%20GMT Event-Date-Timestamp: 1324844994164770 Event-Calling-File: mod_spandsp_fax.c Event-Calling-Function: phase_e_handler Event-Calling-Line-Number: 573 Channel-State: CS_EXECUTE Channel-Call-State: ACTIVE Channel-State-Number: 4 Channel-Name: sofia/external/123456%4085.114.35.241 Unique-ID: 4abfa488-ac1d-4382-b492-1e7529ac40ee Call-Direction: inbound Presence-Call-Direction: inbound Channel-HIT-Dialplan: true Channel-Call-UUID: 4abfa488-ac1d-4382-b492-1e7529ac40ee Answer-State: answered Channel-Read-Codec-Name: L16 Channel-Read-Codec-Rate: 8000 Channel-Read-Codec-Bit-Rate: 128000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Channel-Write-Codec-Bit-Rate: 64000 Caller-Direction: inbound Caller-Username: 123456 Caller-Dialplan: XML Caller-Caller-ID-Name: 123456 Caller-Caller-ID-Number: 123456 Caller-Network-Addr: 85.114.35.241 Caller-ANI: 123456 Caller-Destination-Number: getFAX Caller-Unique-ID: 4abfa488-ac1d-4382-b492-1e7529ac40ee Caller-Source: mod_sofia Caller-Transfer-Source: 1324844899%3A8784497d-6524-408f-bdde-7db53c068a06%3Abl_xfer%3AgetFAX/default/XML Caller-Context: default Caller-RDNIS: 38515492122 Caller-Channel-Name: sofia/external/123456%4085.114.35.241 Caller-Profile-Index: 2 Caller-Profile-Created-Time: 1324844899737667 Caller-Channel-Created-Time: 1324844899717382 Caller-Channel-Answered-Time: 1324844899737667 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false variable_direction: inbound variable_uuid: 4abfa488-ac1d-4382-b492-1e7529ac40ee variable_session_id: 21 variable_sip_local_network_addr: 85.114.35.241 variable_sip_network_ip: 85.114.35.241 variable_sip_network_port: 5080 variable_sip_received_ip: 85.114.35.241 variable_sip_received_port: 5080 variable_sip_via_protocol: udp variable_sip_from_user: 123456 variable_sip_from_uri: 123456%4085.114.35.241 variable_sip_from_host: 85.114.35.241 variable_sip_from_user_stripped: 123456 variable_sofia_profile_name: external variable_sip_Remote-Party-ID: %22123456%22%20%3Csip%3A123456%4085.114.35.241%3E%3Bparty%3Dcalling%3Bscreen%3Dyes%3Bprivacy%3Doff variable_sip_cid_type: rpid variable_sip_req_user: 38515492122 variable_sip_req_port: 5080 variable_sip_req_uri: 38515492122%4085.114.35.241%3A5080 variable_sip_req_host: 85.114.35.241 variable_sip_to_user: 38515492122 variable_sip_to_port: 5080 variable_sip_to_uri: 38515492122%4085.114.35.241%3A5080 variable_sip_to_host: 85.114.35.241 variable_sip_contact_user: mod_sofia variable_sip_contact_port: 5080 variable_sip_contact_uri: mod_sofia%4085.114.35.241%3A5080 variable_sip_contact_host: 85.114.35.241 variable_channel_name: sofia/external/123456%4085.114.35.241 variable_sip_user_agent: FreeSWITCH-mod_sofia/1.0.head-git- variable_sip_via_host: 85.114.35.241 variable_sip_via_port: 5080 variable_sip_via_rport: 5080 variable_switch_r_sdp: v%3D0%0D%0Ao%3DFreeSWITCH%201324824001%201324824002%20IN%20IP4%2085.114.35.241%0D%0As%3DFreeSWITCH%0D%0Ac%3DIN%20IP4%2085.114.35.241%0D%0At%3D0%200%0D%0Am%3Daudio%2020898%20RTP/AVP%200%208%203%20101%2013%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3Dptime%3A20%0D%0A variable_sip_audio_recv_pt: 0 variable_sip_use_codec_name: PCMU variable_sip_use_codec_rate: 8000 variable_sip_use_codec_ptime: 20 variable_write_codec: PCMU variable_write_rate: 8000 variable_DP_MATCH: ARRAY%3A%3A38515492122%7C%3A38515492122 variable_max_forwards: 69 variable_transfer_history: ARRAY%3A%3A1324844899%3A8784497d-6524-408f-bdde-7db53c068a06%3Abl_xfer%3AgetFAX/default/XML variable_transfer_source: 1324844899%3A8784497d-6524-408f-bdde-7db53c068a06%3Abl_xfer%3AgetFAX/default/XML variable_call_uuid: 4abfa488-ac1d-4382-b492-1e7529ac40ee variable_sip_local_sdp_str: v%3D0%0Ao%3DFreeSWITCH%201324819829%201324819830%20IN%20IP4%2085.114.35.241%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%2085.114.35.241%0At%3D0%200%0Am%3Daudio%2025070%20RTP/AVP%200%20101%2013%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Drtpmap%3A13%20CN/8000%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A variable_local_media_ip: 85.114.35.241 variable_local_media_port: 25070 variable_advertised_media_ip: 85.114.35.241 variable_sip_use_pt: 0 variable_rtp_use_ssrc: 1938434566 variable_sip_2833_send_payload: 101 variable_sip_2833_recv_payload: 101 variable_remote_media_ip: 85.114.35.241 variable_remote_media_port: 20898 variable_endpoint_disposition: ANSWER variable_sip_to_tag: KF59FF72Xm35m variable_sip_from_tag: j6BHempZ0BDKS variable_sip_cseq: 22086385 variable_sip_call_id: d32f024c-a9d9-122f-d58d-00110a5bda1c variable_sip_from_display: 123456 variable_sip_full_from: %22123456%22%20%3Csip%3A123456%4085.114.35.241%3E%3Btag%3Dj6BHempZ0BDKS variable_sip_full_to: %3Csip%3A38515492122%4085.114.35.241%3A5080%3E%3Btag%3DKF59FF72Xm35m variable_playback_seconds: 2 variable_playback_ms: 2000 variable_playback_samples: 16000 variable_current_application_data: /tmp/FAX-4abfa488-ac1d-4382-b492-1e7529ac40ee.tif variable_current_application: rxfax variable_fax_v17_disabled: 0 variable_fax_ecm_requested: 1 variable_fax_filename: /tmp/FAX-4abfa488-ac1d-4382-b492-1e7529ac40ee.tif variable_jitterbuffer_msec: 0 variable_read_codec: L16 variable_read_rate: 8000 variable_fax_image_pixel_size: 1728x1094 variable_fax_longest_bad_row_run: 0 variable_fax_encoding: 3 variable_fax_encoding_name: T.6 variable_fax_success: 1 variable_fax_result_code: 0 variable_fax_result_text: OK variable_fax_ecm_used: on variable_fax_local_station_id: SpanDSP%20Fax%20Ident variable_fax_remote_station_id: SpanDSP%20Fax%20Ident variable_fax_document_transferred_pages: 6 variable_fax_document_total_pages: 6 variable_fax_image_resolution: 8031x3850 variable_fax_image_size: 0 variable_fax_bad_rows: 0 variable_fax_transfer_rate: 14400 fax-success: 1 fax-result-code: 0 fax-result-text: OK fax-document-transferred-pages: 6 fax-document-total-pages: 6 fax-image-resolution: 8031x3850 fax-image-size: 0 fax-bad-rows: 0 fax-transfer-rate: 14400 fax-ecm-used: on fax-local-station-id: SpanDSP%20Fax%20Ident fax-remote-station-id: SpanDSP%20Fax%20Ident Content-Length: 6284 Content-Type: text/event-plain Event-Subclass: spandsp%3A%3Atxfaxresult Event-Name: CUSTOM Core-UUID: df6035f5-552b-40ce-a66f-d4de0b0f9527 FreeSWITCH-Hostname: l01sipindirpp FreeSWITCH-Switchname: l01sipindirpp FreeSWITCH-IPv4: 85.114.35.241 FreeSWITCH-IPv6: 2a02%3Aac8%3Ad%3A924%3A211%3Aaff%3Afe5b%3Ada1c Event-Date-Local: 2011-12-25%2019%3A29%3A54 Event-Date-GMT: Sun,%2025%20Dec%202011%2020%3A29%3A54%20GMT Event-Date-Timestamp: 1324844994185061 Event-Calling-File: mod_spandsp_fax.c Event-Calling-Function: phase_e_handler Event-Calling-Line-Number: 573 Channel-State: CS_EXECUTE Channel-Call-State: ACTIVE Channel-State-Number: 4 Channel-Name: sofia/external/38515492122%4085.114.35.241%3A5080 Unique-ID: efb1f19d-4298-409b-b975-06ebbb8403ee Call-Direction: outbound Presence-Call-Direction: outbound Channel-HIT-Dialplan: true Channel-Call-UUID: efb1f19d-4298-409b-b975-06ebbb8403ee Answer-State: answered Channel-Read-Codec-Name: L16 Channel-Read-Codec-Rate: 8000 Channel-Read-Codec-Bit-Rate: 128000 Channel-Write-Codec-Name: PCMU Channel-Write-Codec-Rate: 8000 Channel-Write-Codec-Bit-Rate: 64000 Caller-Direction: outbound Caller-Caller-ID-Name: 38515492122 Caller-Caller-ID-Number: 38515492122 Caller-Network-Addr: 85.114.35.241 Caller-Destination-Number: 38515492122 Caller-Unique-ID: efb1f19d-4298-409b-b975-06ebbb8403ee Caller-Source: src/switch_ivr_originate.c Caller-Context: default Caller-Channel-Name: sofia/external/38515492122%4085.114.35.241%3A5080 Caller-Profile-Index: 1 Caller-Profile-Created-Time: 1324844899717382 Caller-Channel-Created-Time: 1324844899717382 Caller-Channel-Answered-Time: 1324844899737667 Caller-Channel-Progress-Time: 0 Caller-Channel-Progress-Media-Time: 0 Caller-Channel-Hangup-Time: 0 Caller-Channel-Transfer-Time: 0 Caller-Screen-Bit: true Caller-Privacy-Hide-Name: false Caller-Privacy-Hide-Number: false variable_direction: outbound variable_is_outbound: true variable_uuid: efb1f19d-4298-409b-b975-06ebbb8403ee variable_session_id: 20 variable_sip_profile_name: external variable_channel_name: sofia/external/38515492122%4085.114.35.241%3A5080 variable_sip_destination_url: sip%3A38515492122%4085.114.35.241%3A5080 variable_origination_caller_id_name: 123456 variable_origination_caller_id_number: 123456 variable_originate_early_media: true variable_sip_local_sdp_str: v%3D0%0Ao%3DFreeSWITCH%201324824001%201324824002%20IN%20IP4%2085.114.35.241%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%2085.114.35.241%0At%3D0%200%0Am%3Daudio%2020898%20RTP/AVP%200%208%203%20101%2013%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A variable_sip_outgoing_contact_uri: %3Csip%3Amod_sofia%4085.114.35.241%3A5080%3E variable_sip_req_uri: 38515492122%4085.114.35.241%3A5080 variable_sofia_profile_name: external variable_sip_local_network_addr: 85.114.35.241 variable_sip_reply_host: 85.114.35.241 variable_sip_reply_port: 5080 variable_sip_network_ip: 85.114.35.241 variable_sip_network_port: 5080 variable_sip_user_agent: FreeSWITCH-mod_sofia/1.0.head-git- variable_sip_recover_contact: %3Csip%3A38515492122%4085.114.35.241%3A5080%3Btransport%3Dudp%3E variable_sip_recover_via: SIP/2.0/UDP%2085.114.35.241%3A5080%3Brport%3D5080%3Bbranch%3Dz9hG4bKttHcapKerBjrS variable_sip_from_display: 123456 variable_sip_full_from: %22123456%22%20%3Csip%3A123456%4085.114.35.241%3E%3Btag%3Dj6BHempZ0BDKS variable_sip_full_to: %3Csip%3A38515492122%4085.114.35.241%3A5080%3E%3Btag%3DKF59FF72Xm35m variable_sip_from_user: 123456 variable_sip_from_uri: 123456%4085.114.35.241 variable_sip_from_host: 85.114.35.241 variable_sip_to_user: 38515492122 variable_sip_to_port: 5080 variable_sip_to_uri: 38515492122%4085.114.35.241%3A5080 variable_sip_to_host: 85.114.35.241 variable_sip_contact_params: transport%3Dudp variable_sip_contact_user: 38515492122 variable_sip_contact_port: 5080 variable_sip_contact_uri: 38515492122%4085.114.35.241%3A5080 variable_sip_contact_host: 85.114.35.241 variable_sip_to_tag: KF59FF72Xm35m variable_sip_from_tag: j6BHempZ0BDKS variable_sip_cseq: 22086385 variable_sip_call_id: d32f024c-a9d9-122f-d58d-00110a5bda1c variable_switch_r_sdp: v%3D0%0D%0Ao%3DFreeSWITCH%201324819829%201324819830%20IN%20IP4%2085.114.35.241%0D%0As%3DFreeSWITCH%0D%0Ac%3DIN%20IP4%2085.114.35.241%0D%0At%3D0%200%0D%0Am%3Daudio%2025070%20RTP/AVP%200%20101%2013%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3Drtpmap%3A13%20CN/8000%0D%0Aa%3Dptime%3A20%0D%0A variable_sip_audio_recv_pt: 0 variable_sip_use_codec_name: PCMU variable_sip_use_codec_rate: 8000 variable_sip_use_codec_ptime: 20 variable_write_codec: PCMU variable_write_rate: 8000 variable_local_media_ip: 85.114.35.241 variable_local_media_port: 20898 variable_advertised_media_ip: 85.114.35.241 variable_sip_use_pt: 0 variable_rtp_use_ssrc: 3547204870 variable_sip_2833_send_payload: 101 variable_sip_2833_recv_payload: 101 variable_remote_media_ip: 85.114.35.241 variable_remote_media_port: 25070 variable_endpoint_disposition: ANSWER variable_pre_transfer_caller_id_name: 123456 variable_pre_transfer_caller_id_number: 123456 variable_call_uuid: efb1f19d-4298-409b-b975-06ebbb8403ee variable_current_application_data: /tmp/txfax.tiff variable_current_application: txfax variable_fax_v17_disabled: 0 variable_fax_ecm_requested: 1 variable_fax_filename: /tmp/txfax.tiff variable_jitterbuffer_msec: 0 variable_read_codec: L16 variable_read_rate: 8000 variable_fax_image_pixel_size: 1728x1078 variable_fax_longest_bad_row_run: 0 variable_fax_encoding: 3 variable_fax_encoding_name: T.6 variable_fax_success: 1 variable_fax_result_code: 0 variable_fax_result_text: OK variable_fax_ecm_used: on variable_fax_local_station_id: SpanDSP%20Fax%20Ident variable_fax_remote_station_id: SpanDSP%20Fax%20Ident variable_fax_document_transferred_pages: 6 variable_fax_document_total_pages: 6 variable_fax_image_resolution: 8031x3850 variable_fax_image_size: 19763 variable_fax_bad_rows: 0 variable_fax_transfer_rate: 14400 fax-success: 1 fax-result-code: 0 fax-result-text: OK fax-document-transferred-pages: 6 fax-document-total-pages: 6 fax-image-resolution: 8031x3850 fax-image-size: 19763 fax-bad-rows: 0 fax-transfer-rate: 14400 fax-ecm-used: on fax-local-station-id: SpanDSP%20Fax%20Ident fax-remote-station-id: SpanDSP%20Fax%20Ident -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111226/27762ba5/attachment-0001.html From tculjaga at gmail.com Mon Dec 26 11:20:19 2011 From: tculjaga at gmail.com (Tihomir Culjaga) Date: Mon, 26 Dec 2011 09:20:19 +0100 Subject: [Freeswitch-users] CUSTOM events In-Reply-To: References: Message-ID: well, ashamed and blind .... events plain CUSTOM spandsp::txfaxresult spandsp::rxfaxresult spandsp::rxfaxpageresult spandsp::txfaxpageresult pls ignore this post... T. On Mon, Dec 26, 2011 at 9:13 AM, Tihomir Culjaga wrote: > hello, > > please, can anyone tell how do i subscribe to these events ? > > i tried "events plain CUSTOM", fs acepted it but i got no events... > using "events plain all" i can get them but with all others i don't really > want. > > > Event-Subclass: spandsp%3A%3Arxfaxpageresult > Event-Name: CUSTOM > Core-UUID: df6035f5-552b-40ce-a66f-d4de0b0f9527 > FreeSWITCH-Hostname: l01sipindirpp > FreeSWITCH-Switchname: l01sipindirpp > FreeSWITCH-IPv4: 85.114.35.241 > FreeSWITCH-IPv6: 2a02%3Aac8%3Ad%3A924%3A211%3Aaff%3Afe5b%3Ada1c > Event-Date-Local: 2011-12-25%2019%3A29%3A50 > Event-Date-GMT: Sun,%2025%20Dec%202011%2020%3A29%3A50%20GMT > Event-Date-Timestamp: 1324844990773140 > Event-Calling-File: mod_spandsp_fax.c > Event-Calling-Function: phase_d_handler > Event-Calling-Line-Number: 439 > fax-document-transferred-pages: 6 > fax-image-resolution: 8031x3850 > fax-image-size: 19764 > fax-image-pixel-size: 1728x1094 > fax-bad-rows: 0 > fax-longest-bad-row-run: 0 > fax-encoding: 3 > fax-encoding-name: T.6 > > Content-Length: 734 > Content-Type: text/event-plain > > Event-Subclass: spandsp%3A%3Atxfaxpageresult > Event-Name: CUSTOM > Core-UUID: df6035f5-552b-40ce-a66f-d4de0b0f9527 > FreeSWITCH-Hostname: l01sipindirpp > FreeSWITCH-Switchname: l01sipindirpp > FreeSWITCH-IPv4: 85.114.35.241 > FreeSWITCH-IPv6: 2a02%3Aac8%3Ad%3A924%3A211%3Aaff%3Afe5b%3Ada1c > Event-Date-Local: 2011-12-25%2019%3A29%3A51 > Event-Date-GMT: Sun,%2025%20Dec%202011%2020%3A29%3A51%20GMT > Event-Date-Timestamp: 1324844991976624 > Event-Calling-File: mod_spandsp_fax.c > Event-Calling-Function: phase_d_handler > Event-Calling-Line-Number: 439 > fax-document-transferred-pages: 6 > fax-image-resolution: 8031x3850 > fax-image-size: 19763 > fax-image-pixel-size: 1728x1078 > fax-bad-rows: 0 > fax-longest-bad-row-run: 0 > fax-encoding: 3 > fax-encoding-name: T.6 > > > > > > Event-Subclass: spandsp%3A%3Arxfaxresult > Event-Name: CUSTOM > Core-UUID: df6035f5-552b-40ce-a66f-d4de0b0f9527 > FreeSWITCH-Hostname: l01sipindirpp > FreeSWITCH-Switchname: l01sipindirpp > FreeSWITCH-IPv4: 85.114.35.241 > FreeSWITCH-IPv6: 2a02%3Aac8%3Ad%3A924%3A211%3Aaff%3Afe5b%3Ada1c > Event-Date-Local: 2011-12-25%2019%3A29%3A54 > Event-Date-GMT: Sun,%2025%20Dec%202011%2020%3A29%3A54%20GMT > Event-Date-Timestamp: 1324844994164770 > Event-Calling-File: mod_spandsp_fax.c > Event-Calling-Function: phase_e_handler > Event-Calling-Line-Number: 573 > Channel-State: CS_EXECUTE > Channel-Call-State: ACTIVE > Channel-State-Number: 4 > Channel-Name: sofia/external/123456%4085.114.35.241 > Unique-ID: 4abfa488-ac1d-4382-b492-1e7529ac40ee > Call-Direction: inbound > Presence-Call-Direction: inbound > Channel-HIT-Dialplan: true > Channel-Call-UUID: 4abfa488-ac1d-4382-b492-1e7529ac40ee > Answer-State: answered > Channel-Read-Codec-Name: L16 > Channel-Read-Codec-Rate: 8000 > Channel-Read-Codec-Bit-Rate: 128000 > Channel-Write-Codec-Name: PCMU > Channel-Write-Codec-Rate: 8000 > Channel-Write-Codec-Bit-Rate: 64000 > Caller-Direction: inbound > Caller-Username: 123456 > Caller-Dialplan: XML > Caller-Caller-ID-Name: 123456 > Caller-Caller-ID-Number: 123456 > Caller-Network-Addr: 85.114.35.241 > Caller-ANI: 123456 > Caller-Destination-Number: getFAX > Caller-Unique-ID: 4abfa488-ac1d-4382-b492-1e7529ac40ee > Caller-Source: mod_sofia > Caller-Transfer-Source: > 1324844899%3A8784497d-6524-408f-bdde-7db53c068a06%3Abl_xfer%3AgetFAX/default/XML > Caller-Context: default > Caller-RDNIS: 38515492122 > Caller-Channel-Name: sofia/external/123456%4085.114.35.241 > Caller-Profile-Index: 2 > Caller-Profile-Created-Time: 1324844899737667 > Caller-Channel-Created-Time: 1324844899717382 > Caller-Channel-Answered-Time: 1324844899737667 > Caller-Channel-Progress-Time: 0 > Caller-Channel-Progress-Media-Time: 0 > Caller-Channel-Hangup-Time: 0 > Caller-Channel-Transfer-Time: 0 > Caller-Screen-Bit: true > Caller-Privacy-Hide-Name: false > Caller-Privacy-Hide-Number: false > variable_direction: inbound > variable_uuid: 4abfa488-ac1d-4382-b492-1e7529ac40ee > variable_session_id: 21 > variable_sip_local_network_addr: 85.114.35.241 > variable_sip_network_ip: 85.114.35.241 > variable_sip_network_port: 5080 > variable_sip_received_ip: 85.114.35.241 > variable_sip_received_port: 5080 > variable_sip_via_protocol: udp > variable_sip_from_user: 123456 > variable_sip_from_uri: 123456%4085.114.35.241 > variable_sip_from_host: 85.114.35.241 > variable_sip_from_user_stripped: 123456 > variable_sofia_profile_name: external > variable_sip_Remote-Party-ID: > %22123456%22%20%3Csip%3A123456%4085.114.35.241%3E%3Bparty%3Dcalling%3Bscreen%3Dyes%3Bprivacy%3Doff > variable_sip_cid_type: rpid > variable_sip_req_user: 38515492122 > variable_sip_req_port: 5080 > variable_sip_req_uri: 38515492122%4085.114.35.241%3A5080 > variable_sip_req_host: 85.114.35.241 > variable_sip_to_user: 38515492122 > variable_sip_to_port: 5080 > variable_sip_to_uri: 38515492122%4085.114.35.241%3A5080 > variable_sip_to_host: 85.114.35.241 > variable_sip_contact_user: mod_sofia > variable_sip_contact_port: 5080 > variable_sip_contact_uri: mod_sofia%4085.114.35.241%3A5080 > variable_sip_contact_host: 85.114.35.241 > variable_channel_name: sofia/external/123456%4085.114.35.241 > variable_sip_user_agent: FreeSWITCH-mod_sofia/1.0.head-git- > variable_sip_via_host: 85.114.35.241 > variable_sip_via_port: 5080 > variable_sip_via_rport: 5080 > variable_switch_r_sdp: > v%3D0%0D%0Ao%3DFreeSWITCH%201324824001%201324824002%20IN%20IP4%2085.114.35.241%0D%0As%3DFreeSWITCH%0D%0Ac%3DIN%20IP4%2085.114.35.241%0D%0At%3D0%200%0D%0Am%3Daudio%2020898%20RTP/AVP%200%208%203%20101%2013%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3Dptime%3A20%0D%0A > variable_sip_audio_recv_pt: 0 > variable_sip_use_codec_name: PCMU > variable_sip_use_codec_rate: 8000 > variable_sip_use_codec_ptime: 20 > variable_write_codec: PCMU > variable_write_rate: 8000 > variable_DP_MATCH: ARRAY%3A%3A38515492122%7C%3A38515492122 > variable_max_forwards: 69 > variable_transfer_history: > ARRAY%3A%3A1324844899%3A8784497d-6524-408f-bdde-7db53c068a06%3Abl_xfer%3AgetFAX/default/XML > variable_transfer_source: > 1324844899%3A8784497d-6524-408f-bdde-7db53c068a06%3Abl_xfer%3AgetFAX/default/XML > variable_call_uuid: 4abfa488-ac1d-4382-b492-1e7529ac40ee > variable_sip_local_sdp_str: > v%3D0%0Ao%3DFreeSWITCH%201324819829%201324819830%20IN%20IP4%2085.114.35.241%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%2085.114.35.241%0At%3D0%200%0Am%3Daudio%2025070%20RTP/AVP%200%20101%2013%0Aa%3Drtpmap%3A0%20PCMU/8000%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Drtpmap%3A13%20CN/8000%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A > variable_local_media_ip: 85.114.35.241 > variable_local_media_port: 25070 > variable_advertised_media_ip: 85.114.35.241 > variable_sip_use_pt: 0 > variable_rtp_use_ssrc: 1938434566 > variable_sip_2833_send_payload: 101 > variable_sip_2833_recv_payload: 101 > variable_remote_media_ip: 85.114.35.241 > variable_remote_media_port: 20898 > variable_endpoint_disposition: ANSWER > variable_sip_to_tag: KF59FF72Xm35m > variable_sip_from_tag: j6BHempZ0BDKS > variable_sip_cseq: 22086385 > variable_sip_call_id: d32f024c-a9d9-122f-d58d-00110a5bda1c > variable_sip_from_display: 123456 > variable_sip_full_from: > %22123456%22%20%3Csip%3A123456%4085.114.35.241%3E%3Btag%3Dj6BHempZ0BDKS > variable_sip_full_to: > %3Csip%3A38515492122%4085.114.35.241%3A5080%3E%3Btag%3DKF59FF72Xm35m > variable_playback_seconds: 2 > variable_playback_ms: 2000 > variable_playback_samples: 16000 > variable_current_application_data: > /tmp/FAX-4abfa488-ac1d-4382-b492-1e7529ac40ee.tif > variable_current_application: rxfax > variable_fax_v17_disabled: 0 > variable_fax_ecm_requested: 1 > variable_fax_filename: /tmp/FAX-4abfa488-ac1d-4382-b492-1e7529ac40ee.tif > variable_jitterbuffer_msec: 0 > variable_read_codec: L16 > variable_read_rate: 8000 > variable_fax_image_pixel_size: 1728x1094 > variable_fax_longest_bad_row_run: 0 > variable_fax_encoding: 3 > variable_fax_encoding_name: T.6 > variable_fax_success: 1 > variable_fax_result_code: 0 > variable_fax_result_text: OK > variable_fax_ecm_used: on > variable_fax_local_station_id: SpanDSP%20Fax%20Ident > variable_fax_remote_station_id: SpanDSP%20Fax%20Ident > variable_fax_document_transferred_pages: 6 > variable_fax_document_total_pages: 6 > variable_fax_image_resolution: 8031x3850 > variable_fax_image_size: 0 > variable_fax_bad_rows: 0 > variable_fax_transfer_rate: 14400 > fax-success: 1 > fax-result-code: 0 > fax-result-text: OK > fax-document-transferred-pages: 6 > fax-document-total-pages: 6 > fax-image-resolution: 8031x3850 > fax-image-size: 0 > fax-bad-rows: 0 > fax-transfer-rate: 14400 > fax-ecm-used: on > fax-local-station-id: SpanDSP%20Fax%20Ident > fax-remote-station-id: SpanDSP%20Fax%20Ident > > Content-Length: 6284 > Content-Type: text/event-plain > > Event-Subclass: spandsp%3A%3Atxfaxresult > Event-Name: CUSTOM > Core-UUID: df6035f5-552b-40ce-a66f-d4de0b0f9527 > FreeSWITCH-Hostname: l01sipindirpp > FreeSWITCH-Switchname: l01sipindirpp > FreeSWITCH-IPv4: 85.114.35.241 > FreeSWITCH-IPv6: 2a02%3Aac8%3Ad%3A924%3A211%3Aaff%3Afe5b%3Ada1c > Event-Date-Local: 2011-12-25%2019%3A29%3A54 > Event-Date-GMT: Sun,%2025%20Dec%202011%2020%3A29%3A54%20GMT > Event-Date-Timestamp: 1324844994185061 > Event-Calling-File: mod_spandsp_fax.c > Event-Calling-Function: phase_e_handler > Event-Calling-Line-Number: 573 > Channel-State: CS_EXECUTE > Channel-Call-State: ACTIVE > Channel-State-Number: 4 > Channel-Name: sofia/external/38515492122%4085.114.35.241%3A5080 > Unique-ID: efb1f19d-4298-409b-b975-06ebbb8403ee > Call-Direction: outbound > Presence-Call-Direction: outbound > Channel-HIT-Dialplan: true > Channel-Call-UUID: efb1f19d-4298-409b-b975-06ebbb8403ee > Answer-State: answered > Channel-Read-Codec-Name: L16 > Channel-Read-Codec-Rate: 8000 > Channel-Read-Codec-Bit-Rate: 128000 > Channel-Write-Codec-Name: PCMU > Channel-Write-Codec-Rate: 8000 > Channel-Write-Codec-Bit-Rate: 64000 > Caller-Direction: outbound > Caller-Caller-ID-Name: 38515492122 > Caller-Caller-ID-Number: 38515492122 > Caller-Network-Addr: 85.114.35.241 > Caller-Destination-Number: 38515492122 > Caller-Unique-ID: efb1f19d-4298-409b-b975-06ebbb8403ee > Caller-Source: src/switch_ivr_originate.c > Caller-Context: default > Caller-Channel-Name: sofia/external/38515492122%4085.114.35.241%3A5080 > Caller-Profile-Index: 1 > Caller-Profile-Created-Time: 1324844899717382 > Caller-Channel-Created-Time: 1324844899717382 > Caller-Channel-Answered-Time: 1324844899737667 > Caller-Channel-Progress-Time: 0 > Caller-Channel-Progress-Media-Time: 0 > Caller-Channel-Hangup-Time: 0 > Caller-Channel-Transfer-Time: 0 > Caller-Screen-Bit: true > Caller-Privacy-Hide-Name: false > Caller-Privacy-Hide-Number: false > variable_direction: outbound > variable_is_outbound: true > variable_uuid: efb1f19d-4298-409b-b975-06ebbb8403ee > variable_session_id: 20 > variable_sip_profile_name: external > variable_channel_name: sofia/external/38515492122%4085.114.35.241%3A5080 > variable_sip_destination_url: sip%3A38515492122%4085.114.35.241%3A5080 > variable_origination_caller_id_name: 123456 > variable_origination_caller_id_number: 123456 > variable_originate_early_media: true > variable_sip_local_sdp_str: > v%3D0%0Ao%3DFreeSWITCH%201324824001%201324824002%20IN%20IP4%2085.114.35.241%0As%3DFreeSWITCH%0Ac%3DIN%20IP4%2085.114.35.241%0At%3D0%200%0Am%3Daudio%2020898%20RTP/AVP%200%208%203%20101%2013%0Aa%3Drtpmap%3A101%20telephone-event/8000%0Aa%3Dfmtp%3A101%200-16%0Aa%3Dptime%3A20%0Aa%3Dsendrecv%0A > variable_sip_outgoing_contact_uri: > %3Csip%3Amod_sofia%4085.114.35.241%3A5080%3E > variable_sip_req_uri: 38515492122%4085.114.35.241%3A5080 > variable_sofia_profile_name: external > variable_sip_local_network_addr: 85.114.35.241 > variable_sip_reply_host: 85.114.35.241 > variable_sip_reply_port: 5080 > variable_sip_network_ip: 85.114.35.241 > variable_sip_network_port: 5080 > variable_sip_user_agent: FreeSWITCH-mod_sofia/1.0.head-git- > variable_sip_recover_contact: > %3Csip%3A38515492122%4085.114.35.241%3A5080%3Btransport%3Dudp%3E > variable_sip_recover_via: > SIP/2.0/UDP%2085.114.35.241%3A5080%3Brport%3D5080%3Bbranch%3Dz9hG4bKttHcapKerBjrS > variable_sip_from_display: 123456 > variable_sip_full_from: > %22123456%22%20%3Csip%3A123456%4085.114.35.241%3E%3Btag%3Dj6BHempZ0BDKS > variable_sip_full_to: > %3Csip%3A38515492122%4085.114.35.241%3A5080%3E%3Btag%3DKF59FF72Xm35m > variable_sip_from_user: 123456 > variable_sip_from_uri: 123456%4085.114.35.241 > variable_sip_from_host: 85.114.35.241 > variable_sip_to_user: 38515492122 > variable_sip_to_port: 5080 > variable_sip_to_uri: 38515492122%4085.114.35.241%3A5080 > variable_sip_to_host: 85.114.35.241 > variable_sip_contact_params: transport%3Dudp > variable_sip_contact_user: 38515492122 > variable_sip_contact_port: 5080 > variable_sip_contact_uri: 38515492122%4085.114.35.241%3A5080 > variable_sip_contact_host: 85.114.35.241 > variable_sip_to_tag: KF59FF72Xm35m > variable_sip_from_tag: j6BHempZ0BDKS > variable_sip_cseq: 22086385 > variable_sip_call_id: d32f024c-a9d9-122f-d58d-00110a5bda1c > variable_switch_r_sdp: > v%3D0%0D%0Ao%3DFreeSWITCH%201324819829%201324819830%20IN%20IP4%2085.114.35.241%0D%0As%3DFreeSWITCH%0D%0Ac%3DIN%20IP4%2085.114.35.241%0D%0At%3D0%200%0D%0Am%3Daudio%2025070%20RTP/AVP%200%20101%2013%0D%0Aa%3Drtpmap%3A0%20PCMU/8000%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3Drtpmap%3A13%20CN/8000%0D%0Aa%3Dptime%3A20%0D%0A > variable_sip_audio_recv_pt: 0 > variable_sip_use_codec_name: PCMU > variable_sip_use_codec_rate: 8000 > variable_sip_use_codec_ptime: 20 > variable_write_codec: PCMU > variable_write_rate: 8000 > variable_local_media_ip: 85.114.35.241 > variable_local_media_port: 20898 > variable_advertised_media_ip: 85.114.35.241 > variable_sip_use_pt: 0 > variable_rtp_use_ssrc: 3547204870 > variable_sip_2833_send_payload: 101 > variable_sip_2833_recv_payload: 101 > variable_remote_media_ip: 85.114.35.241 > variable_remote_media_port: 25070 > variable_endpoint_disposition: ANSWER > variable_pre_transfer_caller_id_name: 123456 > variable_pre_transfer_caller_id_number: 123456 > variable_call_uuid: efb1f19d-4298-409b-b975-06ebbb8403ee > variable_current_application_data: /tmp/txfax.tiff > variable_current_application: txfax > variable_fax_v17_disabled: 0 > variable_fax_ecm_requested: 1 > variable_fax_filename: /tmp/txfax.tiff > variable_jitterbuffer_msec: 0 > variable_read_codec: L16 > variable_read_rate: 8000 > variable_fax_image_pixel_size: 1728x1078 > variable_fax_longest_bad_row_run: 0 > variable_fax_encoding: 3 > variable_fax_encoding_name: T.6 > variable_fax_success: 1 > variable_fax_result_code: 0 > variable_fax_result_text: OK > variable_fax_ecm_used: on > variable_fax_local_station_id: SpanDSP%20Fax%20Ident > variable_fax_remote_station_id: SpanDSP%20Fax%20Ident > variable_fax_document_transferred_pages: 6 > variable_fax_document_total_pages: 6 > variable_fax_image_resolution: 8031x3850 > variable_fax_image_size: 19763 > variable_fax_bad_rows: 0 > variable_fax_transfer_rate: 14400 > fax-success: 1 > fax-result-code: 0 > fax-result-text: OK > fax-document-transferred-pages: 6 > fax-document-total-pages: 6 > fax-image-resolution: 8031x3850 > fax-image-size: 19763 > fax-bad-rows: 0 > fax-transfer-rate: 14400 > fax-ecm-used: on > fax-local-station-id: SpanDSP%20Fax%20Ident > fax-remote-station-id: SpanDSP%20Fax%20Ident > > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111226/83d9c320/attachment-0001.html From sharad at coraltele.com Mon Dec 26 12:51:42 2011 From: sharad at coraltele.com (sharad) Date: Mon, 26 Dec 2011 15:21:42 +0530 Subject: [Freeswitch-users] Originate a SIP INVITE with T38 References: Message-ID: Hi friends Need to know if there is a way to generate a SIP call with T38 Invite. Our receiver end can not send REINVITE with T38 but it can accept the T38 INVITE. Plz advice. regards Sharad From notlikeme75 at yahoo.com Mon Dec 26 15:29:34 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Mon, 26 Dec 2011 04:29:34 -0800 (PST) Subject: [Freeswitch-users] mod shout on win32 In-Reply-To: References: Message-ID: <1324902574.53784.YahooMailNeo@web65303.mail.ac2.yahoo.com> jeff, I figured as much. are there any instructions anywhere that I can use to help me build mod_shout on windows?? thanks that module is not built in the default package. you will have to build from source to get that module as it stands. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-shout-on-win32-tp7126980p7127192.html Sent from the freeswitch-users mailing list archive at Nabble.com. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111226/e2e499ad/attachment.html From sherifomran2000 at yahoo.com Mon Dec 26 15:58:46 2011 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Mon, 26 Dec 2011 04:58:46 -0800 (PST) Subject: [Freeswitch-users] Gateway Caller ID Help required In-Reply-To: <1324902574.53784.YahooMailNeo@web65303.mail.ac2.yahoo.com> Message-ID: <1324904326.42477.YahooMailClassic@web110802.mail.gq1.yahoo.com> Hi I have setup an IPkall gateway, so that any call to my number in USA be connected to my extension. However, the caller ID that appear is 10 digits and i want it to show +1 stating for usa. In the mean time, I have another german gateway, the number is transmitted fully including the country code, but it appears on my phone in the form of last 10 digits only. any help is appreciated thank you in advance -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111226/b1bb009a/attachment.html From engineerzuhairraza at gmail.com Mon Dec 26 16:02:18 2011 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Mon, 26 Dec 2011 17:02:18 +0400 Subject: [Freeswitch-users] Gateway Caller ID Help required In-Reply-To: <1324904326.42477.YahooMailClassic@web110802.mail.gq1.yahoo.com> References: <1324902574.53784.YahooMailNeo@web65303.mail.ac2.yahoo.com> <1324904326.42477.YahooMailClassic@web110802.mail.gq1.yahoo.com> Message-ID: Hi, use effective_caller_id variable in dialplan to append + Regards, Zohair Raza On Mon, Dec 26, 2011 at 4:58 PM, Sherif Omran wrote: > Hi > > I have setup an IPkall gateway, so that any call to my number in USA be > connected to my extension. However, the caller ID that appear is 10 digits > and i want it to show +1 stating for usa. > > In the mean time, I have another german gateway, the number is transmitted > fully including the country code, but it appears on my phone in the form of > last 10 digits only. > > any help is appreciated > > thank you in advance > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111226/077a63f9/attachment.html From juanito1982 at gmail.com Mon Dec 26 19:48:26 2011 From: juanito1982 at gmail.com (=?ISO-8859-1?Q?Juan_Antonio_Iba=F1ez_Santorum?=) Date: Mon, 26 Dec 2011 17:48:26 +0100 Subject: [Freeswitch-users] require in LUA Message-ID: Hello! I have one doubt about requier in Lua. I have one module coded in a fille called logging.lua. Why must I create '/usr/local/share/lua/5.1/logging/lua.lua' to be included in my main lua script instead creating '/usr/local/share/lua/5.1/logging.lua'? Regards -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111226/5f0815c4/attachment.html From nasir at ictinnovations.com Mon Dec 26 21:21:02 2011 From: nasir at ictinnovations.com (Nasir Iqbal) Date: Mon, 26 Dec 2011 23:21:02 +0500 Subject: [Freeswitch-users] Web based IVR Designer Message-ID: Working on web based IVR Designer for asterisk as well as Freeswitch http://sourcecodemania.com/ivr-designer-using-raphaeljs-for-asterisk/ Looking for your suggestions Regards Nasir Iqbal ICTBroadcast SMS, Fax and Voice broadcasting solution http://www.ictbroadcast.com/ -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111226/23ceb5b9/attachment.html From jeff at jefflenk.com Mon Dec 26 21:48:07 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Mon, 26 Dec 2011 10:48:07 -0800 (PST) Subject: [Freeswitch-users] mod shout on win32 In-Reply-To: <1324902574.53784.YahooMailNeo@web65303.mail.ac2.yahoo.com> References: <1324902574.53784.YahooMailNeo@web65303.mail.ac2.yahoo.com> Message-ID: <1324925287667-7128669.post@n2.nabble.com> You can use the free version on Visual Studio 2010 Express and msysgit and tortoisegit frontend which is what I use. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-shout-on-win32-tp7127898p7128669.html Sent from the freeswitch-users mailing list archive at Nabble.com. From georg at riseup.net Tue Dec 27 00:06:02 2011 From: georg at riseup.net (georg at riseup.net) Date: Mon, 26 Dec 2011 22:06:02 +0100 Subject: [Freeswitch-users] Searching for an application similar to Asterisks 'Authenticate' Message-ID: Hi all, As the subject states: I'm searching for an application similar to Asterisks 'Authenticate' [1]. The description writes: "The application requires a user to enter a password in order to continue execution." I've found on the wiki the function used for conference rooms, however I need this for a public reachable phone number, which then leads to a call redirection system. Thanks, Georg [1] http://www.voip-info.org/wiki/view/Asterisk+cmd+Authenticate From msc at freeswitch.org Tue Dec 27 00:36:42 2011 From: msc at freeswitch.org (Michael Collins) Date: Mon, 26 Dec 2011 13:36:42 -0800 Subject: [Freeswitch-users] Searching for an application similar to Asterisks 'Authenticate' In-Reply-To: References: Message-ID: You can do this with play_and_get_digits. Just put the correct "pin" value in the regex value of pagd. If they enter the incorrect value then just have the "failure" value be a transfer to a "thank you, goodbye" extension or something like that. This is your basic pagd syntax: (Note that you need to use real file names - see the play_and_get_digits syntax for all the details.) Then you need an "oops" extension for when they enter the wrong value" (Again, you'll need a real sound file with some sort of message telling the caller that he/she entered too many failures. Check our list of sound files in the ivr subdir - there might be one you can use.) Have fun! -MC On Mon, Dec 26, 2011 at 1:06 PM, wrote: > Hi all, > > As the subject states: I'm searching for an application similar to > Asterisks 'Authenticate' [1]. The description writes: "The application > requires a user to enter a password in order to continue execution." I've > found on the wiki the function used for conference rooms, however I need > this for a public reachable phone number, which then leads to a call > redirection system. > > Thanks, > Georg > > [1] http://www.voip-info.org/wiki/view/Asterisk+cmd+Authenticate > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111226/5d4ba2cc/attachment-0001.html From anthony.minessale at gmail.com Tue Dec 27 03:13:52 2011 From: anthony.minessale at gmail.com (Anthony Minessale) Date: Mon, 26 Dec 2011 18:13:52 -0600 Subject: [Freeswitch-users] Searching for an application similar to Asterisks 'Authenticate' In-Reply-To: References: Message-ID: In recent versions you can use the voicemail app using "check auth_only " args to auth against a user and it will either continue in the dp or hangup on you with an error message. On Mon, Dec 26, 2011 at 3:36 PM, Michael Collins wrote: > You can do this with play_and_get_digits. Just put the correct "pin" value > in the regex value of pagd. If they enter the incorrect value then just > have the "failure" value be a transfer to a "thank you, goodbye" extension > or something like that. > > This is your basic pagd syntax: > > > (Note that you need to use real file names - see the play_and_get_digits > syntax for all the details.) > > Then you need an "oops" extension for when they enter the wrong value" > > > > > > > > > (Again, you'll need a real sound file with some sort of message telling > the caller that he/she entered too many failures. Check our list of sound > files in the ivr subdir - there might be one you can use.) > > Have fun! > > -MC > > > On Mon, Dec 26, 2011 at 1:06 PM, wrote: > >> Hi all, >> >> As the subject states: I'm searching for an application similar to >> Asterisks 'Authenticate' [1]. The description writes: "The application >> requires a user to enter a password in order to continue execution." I've >> found on the wiki the function used for conference rooms, however I need >> this for a public reachable phone number, which then leads to a call >> redirection system. >> >> Thanks, >> Georg >> >> [1] http://www.voip-info.org/wiki/view/Asterisk+cmd+Authenticate >> >> > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://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/20111226/e4fdb648/attachment.html From chad at apartmentlines.com Tue Dec 27 04:11:56 2011 From: chad at apartmentlines.com (Chad Phillips) Date: Mon, 26 Dec 2011 17:11:56 -0800 Subject: [Freeswitch-users] require in LUA In-Reply-To: References: Message-ID: <557A845F0C90470AAA963523C3BC20B1@gmail.com> check conf/autoload_configs/lua.conf.xml. the module-directory and script-directory params contain the various paths that require will search for c modules and lua scripts respectively. the question marks are replaced by what you pass to require, with the exception that dots are converted to path separators. so if you have '/path/to/scripts/?.lua', and you do 'require "foo.lua"', it will look for '/path/to/scripts/foo/lua.lua' (i.e., replacing the ? with foo/lua). if you 'require "foo"', then you'll of course get '/path/to/scripts/foo.lua'. hope this helps. hunmonk Sent with Sparrow (http://www.sparrowmailapp.com/?sig) On Monday, December 26, 2011 at 8:48 AM, Juan Antonio Iba?ez Santorum wrote: > Hello! > > I have one doubt about requier in Lua. I have one module coded in a fille called logging.lua. Why must I create '/usr/local/share/lua/5.1/logging/lua.lua' to be included in my main lua script instead creating '/usr/local/share/lua/5.1/logging.lua'? > > 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/20111226/d205e704/attachment.html From sherifomran2000 at yahoo.com Tue Dec 27 04:14:49 2011 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Mon, 26 Dec 2011 17:14:49 -0800 (PST) Subject: [Freeswitch-users] Gateway Caller ID Help required In-Reply-To: Message-ID: <1324948489.3565.YahooMailClassic@web110815.mail.gq1.yahoo.com> Hi Zohair, I found it ? but i have 2 gateways, this works with the usa gateway. However, I need to put a condition in the dialplan if it comes from german gateway, add 0049 if it comes from usa gateway, add 001 I don't know the variable name of the gateway best regards and many thanks Sherif --- On Mon, 12/26/11, Zohair Raza wrote: From: Zohair Raza Subject: Re: [Freeswitch-users] Gateway Caller ID Help required To: "FreeSWITCH Users Help" Date: Monday, December 26, 2011, 3:02 PM Hi,?use effective_caller_id variable in dialplan to append + Regards, Zohair Raza On Mon, Dec 26, 2011 at 4:58 PM, Sherif Omran wrote: Hi I have setup an IPkall gateway, so that any call to my number in USA be connected to my extension. However, the caller ID that appear is 10 digits and i want it to show +1 stating for usa. In the mean time, I have another german gateway, the number is transmitted fully including the country code, but it appears on my phone in the form of last 10 digits only. any help is appreciated thank you in advance _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111226/d6c364a5/attachment-0001.html From sherifomran2000 at yahoo.com Tue Dec 27 06:20:05 2011 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Mon, 26 Dec 2011 19:20:05 -0800 (PST) Subject: [Freeswitch-users] Call log - multiple entries CDR?? Billing? In-Reply-To: <557A845F0C90470AAA963523C3BC20B1@gmail.com> Message-ID: <1324956005.92462.YahooMailClassic@web110816.mail.gq1.yahoo.com> Hi, I have the CDR enabled and see multiple logs for the same call. Can any body recommend a call log that works fine and could be extended to be used for billing? thanks in advance regards, Sherif Omran -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111226/3b4562cb/attachment.html From fieldpeak at gmail.com Tue Dec 27 06:58:57 2011 From: fieldpeak at gmail.com (fieldpeak) Date: Tue, 27 Dec 2011 11:58:57 +0800 Subject: [Freeswitch-users] Using xmpp to control conference Message-ID: Dear friends, Could you any one give any tips or an example how to use xmpp protocol to control FS to realize conference. i see the wiki, there is few docs for mod_xmpp_event ( http://wiki.freeswitch.org/wiki/Mod_xmpp_event) also i've considered using mod_socket_event to control conferece, while i prefer using xmpp since i've deployed xmpp server... Thanks! -- Regards, Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111227/715e59fe/attachment.html From curriegrad2004 at gmail.com Tue Dec 27 07:33:25 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Mon, 26 Dec 2011 20:33:25 -0800 Subject: [Freeswitch-users] Call log - multiple entries CDR?? Billing? In-Reply-To: <1324956005.92462.YahooMailClassic@web110816.mail.gq1.yahoo.com> References: <557A845F0C90470AAA963523C3BC20B1@gmail.com> <1324956005.92462.YahooMailClassic@web110816.mail.gq1.yahoo.com> Message-ID: xml_cdr does the job just fine... uuid_bridge is what you may want to be looking for On Mon, Dec 26, 2011 at 7:20 PM, Sherif Omran wrote: > Hi, > > I have the CDR enabled and see multiple logs for the same call. Can any > body recommend a call log that works fine and could be extended to be used > for billing? > > thanks in advance > > 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 > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111226/5d1cb34f/attachment.html From moises.silva at gmail.com Tue Dec 27 09:08:55 2011 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 26 Dec 2011 23:08:55 -0700 Subject: [Freeswitch-users] Freeswitch core dump on ftmod_wanpipe In-Reply-To: <4EF106C3.2060806@tiendalinux.com> References: <4EE7B2DE.6080906@tiendalinux.com> <4EE8E6B6.2090102@integrafin.co.uk> <4EF106C3.2060806@tiendalinux.com> Message-ID: On Tue, Dec 20, 2011 at 3:05 PM, Nestor A Diaz wrote: > Hi, i did an upgrade to the latest freeswitch sources (2011-12-16) and > it haven't core dumped for now, hoewever i haven't put the server in production, so still crossing fingers :) but right now i have a new set > of warnings : > > 2011-12-19 23:12:14.732393 [WARNING] switch_xml.c:2328 Invalid UTF-8 > character to ampersand, skip it > > while i don't care about them since the system is running i will take > care ot them later. > Let me know if you have any more core dumps. I'd need ssh access to troubleshoot. *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 VegaStream is now part of Sangoma! Ask us about both Gateway Appliances and Internal Gateways -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111226/16d2b395/attachment.html From moises.silva at gmail.com Tue Dec 27 09:11:20 2011 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 26 Dec 2011 23:11:20 -0700 Subject: [Freeswitch-users] FreeTDM & libisdn question In-Reply-To: <2B272E37A1D44FD19BFF1E2FF0AB4928@hpelite> References: <67F1171E7AB3444193F564B86EC7C988@hpelite> <4EE92688.8020901@freeswitch.org> <2B272E37A1D44FD19BFF1E2FF0AB4928@hpelite> Message-ID: On Wed, Dec 14, 2011 at 6:11 PM, Stuart Mills wrote: > Thanks for the advise Stefan, it's working using the config you suggested, > had a little fun with the freetdm config as there's not many examples using > this card, apart from that its all good. More examples in the FreeSWITCH wiki or patches to the default freetdm.conf.xml sample to add comments for libpri options are more than welcomed :-) *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 VegaStream is now part of Sangoma! Ask us about both Gateway Appliances and Internal Gateways -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111226/e4739a97/attachment-0001.html From moises.silva at gmail.com Tue Dec 27 09:13:20 2011 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 26 Dec 2011 23:13:20 -0700 Subject: [Freeswitch-users] Fwd: Configure openVOX card FXO card. In-Reply-To: References: Message-ID: On Fri, Dec 9, 2011 at 12:36 AM, Farooq Hussain wrote: > > Hello, > > I am new to freeswtich. Please let me know how to configure FXO openvox > 4ports card configure on freeswtich. > > This is the first resource you should read: http://wiki.freeswitch.org/wiki/FreeTDM *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 VegaStream is now part of Sangoma! Ask us about both Gateway Appliances and Internal Gateways -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111226/23d08133/attachment.html From moises.silva at gmail.com Tue Dec 27 09:16:23 2011 From: moises.silva at gmail.com (Moises Silva) Date: Mon, 26 Dec 2011 23:16:23 -0700 Subject: [Freeswitch-users] Sangoma A200 FXO Outgoing Problem In-Reply-To: References: <14AA2B32-ACE0-429F-AB60-2E9CAF98BD96@freeswitch.org> Message-ID: On Thu, Dec 8, 2011 at 6:49 PM, Ryan V wrote: > On Fri, Dec 9, 2011 at 2:13 AM, Brian West wrote: > >> its freetdm/spanname/channelnumber/numbertodial >> >> You can't do a on those as far as I knew. >> >> > I could see it trying to dial out on span 3 channel 1 even though I had > "a" in > > > > Here is the excerpt from logs. > > [DEBUG] switch_core_state_machine.c:362 (FreeTDM/3:1/9449905000) Running > State Change CS_HANGUP > > Thanks, > > Hi Ryan, Try disabling dial tone detection just to see if it works. You can also record the dial tone using "ftdm trace" command and then use wavesurfer (or perhaps audacity) to see the pair of frequencies used for dial tone and try to match them in your tones.conf if they are not already. *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 VegaStream is now part of Sangoma! Ask us about both Gateway Appliances and Internal Gateways -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111226/1b2e0284/attachment.html From engineerzuhairraza at gmail.com Tue Dec 27 10:09:49 2011 From: engineerzuhairraza at gmail.com (Zohair Raza) Date: Tue, 27 Dec 2011 11:09:49 +0400 Subject: [Freeswitch-users] Gateway Caller ID Help required In-Reply-To: <1324951168.54023.YahooMailClassic@web110802.mail.gq1.yahoo.com> References: <1324948489.3565.YahooMailClassic@web110815.mail.gq1.yahoo.com> <1324951168.54023.YahooMailClassic@web110802.mail.gq1.yahoo.com> Message-ID: Hi, See these links http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_10:_Route_to_a_gateway_extension_with_custom_caller_id http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_18:_Add_international_call_prefix_to_effective_caller_id_number_on_incoming_BRI_calls http://wiki.freeswitch.org/wiki/Dialplan_XML#Example_20:_Fix_invalid_caller_ID Your debug message doesn't show caller id, so what I've understood is, you need to make a check for 10 digit caller id, so that could be, and leave caller id as it is similarly you can make other checks as per your requirements. Regards, Zohair Raza On Tue, Dec 27, 2011 at 5:59 AM, Sherif Omran wrote: > Hi Zohair, > > I made it differently, each gateway forwards to an extension and in the > dial plan i control the incoming caller id. I think, this may not be > correct because of using 2 numbers instead of only 1 external number that > is connected to the same extension. > If you have an idea, i would appreciate. > > Regarding the german gateway, I can not add 0049 because i am calling it > from my swiss number and the number is transmitted correctly, however what > i see is 10 digits only. see the following output: > > 2011-12-27 01:53:32.275392 [DEBUG] switch_core_state_machine.c:362 > (sofia/sipinterface_1/+41793940965 at bluesip.net) Running State Change > CS_REPORTING > 2011-12-27 01:53:32.275392 [DEBUG] switch_core_state_machine.c:662 > (sofia/sipinterface_1/+41793940965 at bluesip.net) State REPORTING > 2011-12-27 01:53:32.275392 [DEBUG] switch_core_state_machine.c:362 > (sofia/sipinterface_1/sip:1000 at 88.64.50.28:61000) Running State Change > CS_HANGUP > > kind regards, > Sherif > > --- On *Tue, 12/27/11, Sherif Omran * wrote: > > > From: Sherif Omran > > Subject: Re: [Freeswitch-users] Gateway Caller ID Help required > To: "FreeSWITCH Users Help" > Date: Tuesday, December 27, 2011, 3:14 AM > > > Hi Zohair, > > I found it > > data="effective_caller_id_number=001${caller_id_number}"/> > > but i have 2 gateways, this works with the usa gateway. However, I need to > put a condition in the dialplan > > if it comes from german gateway, add 0049 > if it comes from usa gateway, add 001 > > I don't know the variable name of the gateway > > best regards and many thanks > > Sherif > > > --- On *Mon, 12/26/11, Zohair Raza * wrote: > > > From: Zohair Raza > Subject: Re: [Freeswitch-users] Gateway Caller ID Help required > To: "FreeSWITCH Users Help" > Date: Monday, December 26, 2011, 3:02 PM > > Hi, > use effective_caller_id variable in dialplan to append + > > Regards, > Zohair Raza > > > > > On Mon, Dec 26, 2011 at 4:58 PM, Sherif Omran wrote: > > Hi > > I have setup an IPkall gateway, so that any call to my number in USA be > connected to my extension. However, the caller ID that appear is 10 digits > and i want it to show +1 stating for usa. > > In the mean time, I have another german gateway, the number is transmitted > fully including the country code, but it appears on my phone in the form of > last 10 digits only. > > any help is appreciated > > thank you in advance > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > -----Inline Attachment Follows----- > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111227/bad33aac/attachment-0001.html From B.Tietz at pinguin.ag Tue Dec 27 12:21:28 2011 From: B.Tietz at pinguin.ag (B.Tietz at pinguin.ag) Date: Tue, 27 Dec 2011 10:21:28 +0100 Subject: [Freeswitch-users] Failover with Postgresql In-Reply-To: References: <07BF4904977CC645B485E970424193AD0E690EDB7E@localhost> <07BF4904977CC645B485E970424193AD0E690EDBC2@localhost> <07BF4904977CC645B485E970424193AD0E690EDBD0@localhost> <07BF4904977CC645B485E970424193AD0E690EDBDD@localhost> Message-ID: <07BF4904977CC645B485E970424193AD0F975C67CF@localhost> Hi, failover with PGSQL now working. I had to change the datatype for the "metadata"-coloumn from the sip_recovery-table to character varying (65536). regards, Benjamin T. Betreff: Re: [Freeswitch-users] Failover with Postgresql my encoding is UTF8 On Fri, Dec 23, 2011 at 10:07 AM, wrote: > If you are in the psql-cli make '\l' for listing the databases and you will see the coding of the databases... > > VG, > Benjamin T. > > > Betreff: Re: [Freeswitch-users] Failover with Postgresql > > What do you mean? > I am using postgresql > > > > On Fri, Dec 23, 2011 at 9:52 AM, ? wrote: >> Hi Gilad, >> >> and what is the coding of your database for freeswitch? >> >> VG, >> Benjamin T. >> >> >> Betreff: Re: [Freeswitch-users] Failover with Postgresql >> >> These are my settings for postgresql and they work. >> >> ?/etc/odbc.ini [$YOUR_DSN_NAME]Description ? ? ? ? = PostgreSQL >> UnicodeDriver ? ? ? ? ? ? ?= PostgreSQL UnicodeTrace ? ? ? ? ? ? ? = >> NoTraceFile ? ? ? ? ? = /tmp/psqlodbc.logDatabase ? ? ? ? ? ?= >> $YOUR_DSN_NAMEServername ? ? ? ? ?= 127.0.0.1UserName ? ? ? ? ? ?= >> $YOUR_DB_USERNAMEPassword ? ? ? ? ? ?= $YOUR_DB_PASSWORDPort >> ? ? ?= 5432ReadOnly ? ? ? ? ? ?= YesRowVersioning ? ? ? = NoShowSystemTables ? ?= NoShowOidColumn ? ? ? = NoFakeOidIndex = NoConnSettings ? ? ? ?=ODBC /etc/odbcinst.ini[PostgreSQL ANSI]?Description ? ? ? ? ? ?= PostgreSQL ODBC driver (ANSI version)Driver ? ? ? ? ?= /usr/lib/odbc/psqlodbca.soSetup ? ? ? ? ? = /usr/lib/odbc/libodbcpsqlS.soDebug ? ? ? ? ? = 0CommLog ? ? ? ? = 1UsageCount ? ? ? ? ? ? ?= 1Threading = 0[PostgreSQL Unicode]Description ? ? ? ? ? ? = PostgreSQL ODBC driver (Unicode version)Driver ? ? ? ? ?= /usr/lib/odbc/psqlodbcw.soSetup ? ? ? ? ? = /usr/lib/odbc/libodbcpsqlS.soDebug ? ? ? ? ? = 0CommLog ? ? ? ? = 1UsageCount ? ? ? ? ? ? ?= 1Threading = 0MaxLongVarcharSize = 65536 On Fri, Dec 23, 2011 at 9:36 AM, ? wrote: >>> Hi, >>> >>> cdr data still invalid. I have a Master-Master-MySQL-Setup over both machines where recovery works. Database has latin1 coding to. I just think MySQL is not that stable... That's why I'd like to try pgsql. >>> >>> this is my odbcinst.ini >>> >>> [PostgreSQLUnicode] >>> Description ? ? = PostgreSQL ODBC driver (Unicode version) Driver = >>> /usr/lib/odbc/psqlodbcw.so Setup ? ? ? = >>> /usr/lib/odbc/libodbcpsqlS.so Debug ? ? ? = 0 CommLog ? ? = 1 >>> UsageCount ? ? ?= 1 Threading = 0 >>> MaxLongVarcharSize=65536 >>> >>> Here my odbc.ini >>> >>> [fs_psql] >>> Description ? ? ? ? = PostgreSQLUnicode Driver ? ? ? ? ? ? ?= >>> PostgreSQLUnicode Trace ? ? ? ? ? ? ? = No TraceFile ? ? ? ? ? = >>> /tmp/psqlodbc.log Database ? ? ? ? ? ?= freeswitch Servername = >>> 1.2.3.4 UserName ? ? ? ? ? ?= freeswitch Password ? ? ? ? ? ?= ?xxx >>> Port ? ? ? ? ? ? ? ?= 5432 ReadOnly ? ? ? ? ? ?= Yes RowVersioning = >>> No ShowSystemTables ? ?= No ShowOidColumn ? ? ? = No FakeOidIndex = >>> No ConnSettings ? ? ? ?= >>> >>> pgsql-Database is LATIN1 coding >>> >>> VG, >>> Benjamin T. >>> >>> >>> -----Urspr?ngliche Nachricht----- >>> >>> Hi Benjamin >>> >>> Psql cuts off the characters in the sofia recover table. To fix this you need to add: "MaxLongVarcharSize = 65536" >>> under your psql ANSI in /etc/odbcinst.ini >>> >>> Good luck >>> Gill >>> >>> On Fri, Dec 23, 2011 at 8:39 AM, ? wrote: >>>> Hi, >>>> I try to make sofia recover with two servers. Setup is slightly >>>> done like described in Wiki. >>>> But if I try to make a sofia recover the call is not recovered. >>>> Here is the Error Message from CLI: >>>> 2011-12-23 14:34:42.256728 [WARNING] sofia_glue.c:5418 Invalid cdr >>>> data, call not recovered Both server can reach the pgsql-server via >>>> odbc. Data is written in sip_recovery. >>>> I think the coding fort he database is wrong. Can anyone tell me >>>> what would be the best encoding? UTF8 and ANSII is bad! >>>> regards, >>>> >>>> Benjamin >>>> >>> >>> ____________________________________________________________________ >>> _ _ ___ Professional FreeSWITCH Consulting Services: >>> consulting at freeswitch.org >>> http://www.freeswitchsolutions.com >>> >>> >>> >>> >>> Official FreeSWITCH Sites >>> http://www.freeswitch.org >>> http://wiki.freeswitch.org >>> http://www.cluecon.com >>> >>> FreeSWITCH-users mailing list >>> FreeSWITCH-users at lists.freeswitch.org >>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-u >>> s >>> e >>> rs >>> http://www.freeswitch.org >> >> >> >> -- >> Gilad Abada >> >> SteadFast Telecommunications, Inc. >> >> Call us to find out how much you can save with VoIP! >> >> V: 212.589.1001 >> F: 212.589.1011 >> >> >> For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises.?? Make your virtual office a reality.? Enjoy the freedom to travel while remaining connected to your office. >> >> _____________________________________________________________________ >> _ ___ Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-us >> e >> 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-us >> e >> rs >> http://www.freeswitch.org > > > > -- > Gilad Abada > > SteadFast Telecommunications, Inc. > > Call us to find out how much you can save with VoIP! > > V: 212.589.1001 > F: 212.589.1011 > > > For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises.?? Make your virtual office a reality.? Enjoy the freedom to travel while remaining connected to your office. > > ______________________________________________________________________ > ___ Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users 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-use > rs > http://www.freeswitch.org -- Gilad Abada SteadFast Telecommunications, Inc. Call us to find out how much you can save with VoIP! V: 212.589.1001 F: 212.589.1011 For 35 years, Steadfast Telecommunications has been providing state-of-the-art communications technology to businesses and government agencies - large and small. Steadfast Telecommunications tailors Unified Communications and Voice-Over IP Solutions to single-site offices or multi-site and worldwide enterprises.?? Make your virtual office a reality.? Enjoy the freedom to travel while remaining connected to your office. _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From sdevoy at bizfocused.com Mon Dec 26 06:47:17 2011 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sun, 25 Dec 2011 22:47:17 -0500 Subject: [Freeswitch-users] ./configure still fails with error: libtermcap, libcurses or libncurses are required! In-Reply-To: <0a3101ccc1ce$44ad65e0$ce0831a0$@com> References: <0a3101ccc1ce$44ad65e0$ce0831a0$@com> Message-ID: <05bf01ccc381$10533db0$30f9b910$@com> I got past this since it took 2 days to post. The problem was in the link that showed how/what to install during the CENTOS install. I went back and used some common sense. I installed server with very little default add-ons, but I added several developer library options. I ended up with a bigger Centos install than I probably needed, but then FreeSwitch went much better. Sean From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Sean Devoy Sent: Friday, December 23, 2011 6:55 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] ./configure still fails with error: libtermcap, libcurses or libncurses are required! Hi, I am a NooB to FreeSwitch. Sorry if I should start elsewhere, I can't figure out where. I ran the recommended installation instructions, including installing Centos 5.7 from boot DVD. I installed and compiled GIT, regardless of the errors in the install guide. I ran: cd /usr/local/src git clone git://git.freeswitch.org/freeswitch.git I went to: cd /usr/local/src/freeswitch Ran: ./bootstrap.sh That required me to "yum install " autoconf, automake and lib??? (I dont recall), then it would run. And tried: ./configure Which of course terminates with: configure: error: libtermcap, libcurses or libncurses are required! I have "yum install"ed libtermcap, libcurse,libncurese,termcap,ncurses,curses and everything else I can possibly find a reference to. That help a lot (a.k.a. NONE). Now I get: configure: error: libtermcap, libcurses or libncurses are required! I have googled and read and searched. I was happy to see where this was a known problem. I was disappointed to see NO help on what to do since it was "fixed". I am trying not to ask dumb questions, but I am struggling here. It is possible for a mere mortal to install FreeSwitch on a new system? Is there a set of install instructions that match the current realease? I am fighting the urge to throw Linux out the window and go with the Windows build. I have made a living of MS products for over 20 years, but there seems to be nobody left there with a brain. I am trying to move on. Thanks in advance, Sean -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111225/92c85d6e/attachment-0001.html From sdevoy at bizfocused.com Mon Dec 26 06:49:51 2011 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sun, 25 Dec 2011 22:49:51 -0500 Subject: [Freeswitch-users] ./configure still fails with error: libtermcap, libcurses or libncurses are required! In-Reply-To: References: <0a3101ccc1ce$44ad65e0$ce0831a0$@com> Message-ID: <05c401ccc381$6be307a0$43a916e0$@com> Thanks Vitalie. I did much the same by re-installing Centos w/ more options and adding the packages you showed. There is so much misinformation out there on this. Thank you for your time. From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Vitalie Colosov Sent: Sunday, December 25, 2011 6:52 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] ./configure still fails with error: libtermcap, libcurses or libncurses are required! You need to install few other packages - full list is: yum install autoconf automake gcc-c++ git-core libjpeg-devel libtool make ncurses-devel pkgconfig You can find this description at the following section of the installation guide: http://wiki.freeswitch.org/wiki/Installation_Guide#CentOS Also, you might want to install the optional packages: yum install unixODBC-devel openssl-devel gnutls-devel libogg-devel libvorbis-devel curl-devel libtiff-devel libjpeg-devel python-devel expat-devel zlib zlib-devel bzip2 which Regards, Vitalie 2011/12/23 Sean Devoy Hi, I am a NooB to FreeSwitch. Sorry if I should start elsewhere, I can't figure out where. I ran the recommended installation instructions, including installing Centos 5.7 from boot DVD. I installed and compiled GIT, regardless of the errors in the install guide. I ran: cd /usr/local/src git clone git://git.freeswitch.org/freeswitch.git I went to: cd /usr/local/src/freeswitch Ran: ./bootstrap.sh That required me to "yum install " autoconf, automake and lib??? (I dont recall), then it would run. And tried: ./configure Which of course terminates with: configure: error: libtermcap, libcurses or libncurses are required! I have "yum install"ed libtermcap, libcurse,libncurese,termcap,ncurses,curses and everything else I can possibly find a reference to. That help a lot (a.k.a. NONE). Now I get: configure: error: libtermcap, libcurses or libncurses are required! I have googled and read and searched. I was happy to see where this was a known problem. I was disappointed to see NO help on what to do since it was "fixed". I am trying not to ask dumb questions, but I am struggling here. It is possible for a mere mortal to install FreeSwitch on a new system? Is there a set of install instructions that match the current realease? I am fighting the urge to throw Linux out the window and go with the Windows build. I have made a living of MS products for over 20 years, but there seems to be nobody left there with a brain. I am trying to move on. Thanks in advance, Sean _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111225/053196a4/attachment-0001.html From sdevoy at bizfocused.com Mon Dec 26 06:50:56 2011 From: sdevoy at bizfocused.com (Sean Devoy) Date: Sun, 25 Dec 2011 22:50:56 -0500 Subject: [Freeswitch-users] mod_shout on win32 In-Reply-To: <1324869594666-7127192.post@n2.nabble.com> References: <1324855522.90986.YahooMailNeo@web65304.mail.ac2.yahoo.com> <1324869594666-7127192.post@n2.nabble.com> Message-ID: <05c901ccc381$930c34f0$b9249ed0$@com> Thanks. FYI, I did build from source after getting all the pre-reqs. Sean -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk Sent: Sunday, December 25, 2011 10:20 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_shout on win32 that module is not built in the default package. you will have to build from source to get that module as it stands. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-shout-on-win32-tp7126980p7 127192.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 peter at uringme.com Tue Dec 27 00:13:24 2011 From: peter at uringme.com (peter at uringme.com) Date: Mon, 26 Dec 2011 13:13:24 -0800 (PST) Subject: [Freeswitch-users] does streamFile callback provide file position? Message-ID: <1324934004.78516.YahooMailClassic@web2810.biz.mail.ne1.yahoo.com> I'm looking at the examples for mod_spidermonkey where streamFile can be interrupted by a DTMF.? There's dtmfcallback.js (http://wiki.freeswitch.org/wiki/Examples_dtmfcallback.js) and the Session_streamFile doc itself (http://wiki.freeswitch.org/wiki/Session_streamFile) They both use the third argument to the callback to get digits.digit or .duration, which I'm assuming is the length that the DTMF was pressed.? But, do the third or fourth arguments also include the position within the file?? I'm looking for a way to save the location in a voice prompt where a person pressed the DTMF -- either in seconds or samples or bytes from the beginning of the file. Also, for my own knowledge, is there a doc somewhere that explains all the possibile members of the third and fourth arguments (usually data and arg) in the callback from streamFile?? Thank you. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111226/7c478255/attachment.html From th982a at googlemail.com Tue Dec 27 07:00:48 2011 From: th982a at googlemail.com (Tamer Higazi) Date: Tue, 27 Dec 2011 05:00:48 +0100 Subject: [Freeswitch-users] sangoma A200 with mod_freetdm on windows?! Message-ID: <4EF942F0.4010404@googlemail.com> Hi people! I got my A200 board running with 1 FXS module on Linux along with mod_freetdm, but I am facing problems getting it to run on Windows. From Sangoma I followed the instructions to set up the board on Windows7 winpipe module, which works so far. How do I get freeswitch with mod_freetdm to run on Windows that I can make use of the board (pbx) on a win machine? for any ideas, I would thank you. Tamer From freeswitch at scottisheyes.com Tue Dec 27 08:37:14 2011 From: freeswitch at scottisheyes.com (James) Date: Mon, 26 Dec 2011 21:37:14 -0800 Subject: [Freeswitch-users] Building Freeswitch on OSX Lion - ./configure openssl error In-Reply-To: References: Message-ID: I'm using the latest git commit (01267cd6f5b9a243c42c571f5d161849a66b3c82) and running into an error configuring Freeswitch on OSX Lion: ./configure: line 12283: syntax error near unexpected token `openssl,' ./configure: line 12283: ` PKG_CHECK_MODULES(openssl, openssl,' configure: error: ./configure.gnu failed for libs/iksemel ./bootstrap.sh runs fine; ./configure eventually spits out the error above. Any tips for me? I am mainly trying to set it up for local development purposes, not production. Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111226/39c2e89e/attachment.html From sherifomran2000 at yahoo.com Tue Dec 27 15:10:50 2011 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Tue, 27 Dec 2011 04:10:50 -0800 (PST) Subject: [Freeswitch-users] Call log - multiple entries CDR?? Billing? In-Reply-To: Message-ID: <1324987850.98586.YahooMailClassic@web110814.mail.gq1.yahoo.com> out10021002100112/27/2011 11:30:48 am0:00:47Detailsout10021002100112/27/2011 11:30:48 am0:00:47Details Call logs are duplicated for 1 call? How can i prevent this? --- On Tue, 12/27/11, curriegrad2004 wrote: From: curriegrad2004 Subject: Re: [Freeswitch-users] Call log - multiple entries CDR?? Billing? To: "FreeSWITCH Users Help" Date: Tuesday, December 27, 2011, 6:33 AM xml_cdr does the job just fine... uuid_bridge is what you may want to be looking for On Mon, Dec 26, 2011 at 7:20 PM, Sherif Omran wrote: Hi, I have the CDR enabled and see multiple logs for the same call. Can any body recommend a call log that works fine and could be extended to be used for billing? thanks in advance 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 -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111227/4df7506b/attachment.html From grsingh750 at gmail.com Tue Dec 27 15:19:39 2011 From: grsingh750 at gmail.com (guru singh) Date: Tue, 27 Dec 2011 17:49:39 +0530 Subject: [Freeswitch-users] Caller ID on Analog Lines In-Reply-To: References: Message-ID: Hi, I am the person whose thread you mentioned. I've also had somebody contact me off the list regarding this. What I did in my case was jut get some sort of dtmf converter. It costs Rs 200 and is Made in China. You plug in your FXO line in it and take another wire from it to your sangoma card. Caller ID is detected then. The only labeling it has on it is NetSonic. Somebody who contacted me tried to find it online but wasn't able to I think. I was referred to a shop here by the guy who I buy my Sangoma cards from. Muhammad Shahzad is right about polarity reversal, I confirmed this with the telco (Airtel). I do have hardware to test this or give access to somebody who wants to test a fix. But I just moved to a new place and dont have anything set up right now. It'll take a week before I have servers set up. Regards guru On Sun, Dec 25, 2011 at 7:42 AM, Ryan V wrote: > Caller ID on PRI line works fine. I have problem only with calls coming in > on FXO. > > Thanks, > > Venkatesh K > > > On Sun, Dec 25, 2011 at 1:33 AM, Muhammad Shahzad > wrote: >> >> I think India uses polarity reversal to indicate start of caller id on PRI >> line, which i guess is not fully supported by FreeTDM yet. However,?you can >> try the following FreeTDM options, though i am not sure if they will work. >> >> >> >> >> >> You may try to set last parameter to false and test again. Do let us know >> if it works or not. >> >> I am not much familiar with FreeTDM code, nor got the?equipment?to test, >> so for those who have, please see below link regarding same problem in >> Asterisk and how they solved it, >> >> https://issues.asterisk.org/print_bug_page.php?bug_id=6683 >> >> Hope this helps. >> >> Thank you. >> >> >> On Sat, Dec 24, 2011 at 6:57 PM, Ryan V wrote: >>> >>> Hi, >>> >>> Thanks to all the support we got in mailing lists and our telco, we put >>> our freeswitch PBX online today. >>> >>> We are using Sangoma A101 and A200 Cards. We are successfully receiving >>> the calls on our PRI and Analogue lines. But, we don't get caller-id on our >>> analogue Lines. In India, Caller id comes in as DTMF tones and I could see >>> the caller ID being detected correctly in debug. >>> >>> I have seen this issue raised earlier on the list and Moises Silva from >>> Sangoma offered to fix it if given access to the server. Here is the link >>> http://comments.gmane.org/gmane.comp.telephony.freeswitch.user/32340 >>> >>> I wanted to know whether anyone has come up with a solution. I am open to >>> give access to our server if Moises Silva or anyone else in Sangoma would >>> like to have a look at it. >>> >>> Thanks, >>> >>> 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 >>> >> >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +92 334 422 40 88 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From sharad at coraltele.com Tue Dec 27 15:34:53 2011 From: sharad at coraltele.com (sharad) Date: Tue, 27 Dec 2011 18:04:53 +0530 Subject: [Freeswitch-users] Caller ID on Analog Lines References: Message-ID: <708B8A2349964E7CACABA8E13A854C5C@sharad> Hi, I think if you add some external module to capture the CLI, it wont help you because CLI is not detected by Freeeswitch. saying this bacause I am from PBX line & have good experience of the said problem. I would like to suggest to check the CLI type. It may be in FSK mode. Eaelier I had used Grandstream FXO gateways to detect the FSK CLI. Can'nt say about DTMF. You can confirm with grandstream people. Having said this, another check point is - Check the CLI type whether it is before ring or between ring. This is important if your CLI is in dtmf. If it is so, you may have to give some time to detect the CLI. We call it CLIP timer. At last, someone mentioned the CLI relation with polarity reversal. i do not think there is any relation between CLI & polarity reversal. Feel free to discuss further. Regards Sharad ----- Original Message ----- From: "guru singh" To: "FreeSWITCH Users Help" Sent: Tuesday, December 27, 2011 5:49 PM Subject: Re: [Freeswitch-users] Caller ID on Analog Lines Hi, I am the person whose thread you mentioned. I've also had somebody contact me off the list regarding this. What I did in my case was jut get some sort of dtmf converter. It costs Rs 200 and is Made in China. You plug in your FXO line in it and take another wire from it to your sangoma card. Caller ID is detected then. The only labeling it has on it is NetSonic. Somebody who contacted me tried to find it online but wasn't able to I think. I was referred to a shop here by the guy who I buy my Sangoma cards from. Muhammad Shahzad is right about polarity reversal, I confirmed this with the telco (Airtel). I do have hardware to test this or give access to somebody who wants to test a fix. But I just moved to a new place and dont have anything set up right now. It'll take a week before I have servers set up. Regards guru On Sun, Dec 25, 2011 at 7:42 AM, Ryan V wrote: > Caller ID on PRI line works fine. I have problem only with calls coming in > on FXO. > > Thanks, > > Venkatesh K > > > On Sun, Dec 25, 2011 at 1:33 AM, Muhammad Shahzad > wrote: >> >> I think India uses polarity reversal to indicate start of caller id on >> PRI >> line, which i guess is not fully supported by FreeTDM yet. However, you >> can >> try the following FreeTDM options, though i am not sure if they will >> work. >> >> >> >> >> >> You may try to set last parameter to false and test again. Do let us know >> if it works or not. >> >> I am not much familiar with FreeTDM code, nor got the equipment to test, >> so for those who have, please see below link regarding same problem in >> Asterisk and how they solved it, >> >> https://issues.asterisk.org/print_bug_page.php?bug_id=6683 >> >> Hope this helps. >> >> Thank you. >> >> >> On Sat, Dec 24, 2011 at 6:57 PM, Ryan V wrote: >>> >>> Hi, >>> >>> Thanks to all the support we got in mailing lists and our telco, we put >>> our freeswitch PBX online today. >>> >>> We are using Sangoma A101 and A200 Cards. We are successfully receiving >>> the calls on our PRI and Analogue lines. But, we don't get caller-id on >>> our >>> analogue Lines. In India, Caller id comes in as DTMF tones and I could >>> see >>> the caller ID being detected correctly in debug. >>> >>> I have seen this issue raised earlier on the list and Moises Silva from >>> Sangoma offered to fix it if given access to the server. Here is the >>> link >>> http://comments.gmane.org/gmane.comp.telephony.freeswitch.user/32340 >>> >>> I wanted to know whether anyone has come up with a solution. I am open >>> to >>> give access to our server if Moises Silva or anyone else in Sangoma >>> would >>> like to have a look at it. >>> >>> Thanks, >>> >>> 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 >>> >> >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +92 334 422 40 88 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From dujinfang at gmail.com Tue Dec 27 16:08:28 2011 From: dujinfang at gmail.com (Seven Du) Date: Tue, 27 Dec 2011 21:08:28 +0800 Subject: [Freeswitch-users] Building Freeswitch on OSX Lion - ./configure openssl error In-Reply-To: References: Message-ID: <1C786A574DB84D9D91FBC6F16961CB23@gmail.com> 1. report on jira.freeswitch.og 2. for a workaround, try mv libs/iksemel to /tmp before configure and move back before make if you don't need dingaling support. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow (http://www.sparrowmailapp.com) On Tuesday, December 27, 2011 at 1:37 PM, James wrote: > I'm using the latest git commit (01267cd6f5b9a243c42c571f5d161849a66b3c82) and running into an error configuring Freeswitch on OSX Lion: > > ./configure: line 12283: syntax error near unexpected token `openssl,' > ./configure: line 12283: ` PKG_CHECK_MODULES(openssl, openssl,' > configure: error: ./configure.gnu failed for libs/iksemel > > > > ./bootstrap.sh (http://bootstrap.sh) runs fine; ./configure eventually spits out the error above. Any tips for me? I am mainly trying to set it up for local development purposes, not production. > > Thanks. > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org (mailto:consulting at freeswitch.org) > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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/20111227/eefe51a6/attachment.html From kerem.erciyes at gmail.com Tue Dec 27 16:12:02 2011 From: kerem.erciyes at gmail.com (Kerem Erciyes) Date: Tue, 27 Dec 2011 15:12:02 +0200 Subject: [Freeswitch-users] Caller ID on Analog Lines In-Reply-To: <708B8A2349964E7CACABA8E13A854C5C@sharad> References: <708B8A2349964E7CACABA8E13A854C5C@sharad> Message-ID: A trick I learned when dealing with Turkish Telekom and CLI problems etc.. on FXO lines. When dealing with Analog lines try to find out maker and model of the Central PBX you are connected to, then search about tha model on the internet, this might give you better technical knowledge (at least what is supported and what is not). Good luck, Kerem On Tue, Dec 27, 2011 at 2:34 PM, sharad wrote: > Hi, > > I think if you add some external module to capture the CLI, it wont help > you > because CLI is not detected by Freeeswitch. saying this bacause I am from > PBX line & have good experience of the said problem. > > I would like to suggest to check the CLI type. It may be in FSK mode. > Eaelier I had used Grandstream FXO gateways to detect the FSK CLI. Can'nt > say about DTMF. You can confirm with grandstream people. > > Having said this, another check point is - Check the CLI type whether it is > before ring or between ring. This is important if your CLI is in dtmf. If > it > is so, you may have to give some time to detect the CLI. We call it CLIP > timer. > > At last, someone mentioned the CLI relation with polarity reversal. i do > not > think there is any relation between CLI & polarity reversal. > > Feel free to discuss further. > > Regards > Sharad > > > ----- Original Message ----- > From: "guru singh" > To: "FreeSWITCH Users Help" > Sent: Tuesday, December 27, 2011 5:49 PM > Subject: Re: [Freeswitch-users] Caller ID on Analog Lines > > > Hi, > > I am the person whose thread you mentioned. I've also had somebody > contact me off the list regarding this. What I did in my case was jut > get some sort of dtmf converter. It costs Rs 200 and is Made in China. > You plug in your FXO line in it and take another wire from it to your > sangoma card. Caller ID is detected then. The only labeling it has on > it is NetSonic. Somebody who contacted me tried to find it online but > wasn't able to I think. I was referred to a shop here by the guy who I > buy my Sangoma cards from. > > > Muhammad Shahzad is right about polarity reversal, I confirmed this > with the telco (Airtel). I do have hardware to test this or give > access to somebody who wants to test a fix. But I just moved to a new > place and dont have anything set up right now. It'll take a week > before I have servers set up. > > Regards > guru > > On Sun, Dec 25, 2011 at 7:42 AM, Ryan V wrote: > > Caller ID on PRI line works fine. I have problem only with calls coming > in > > on FXO. > > > > Thanks, > > > > Venkatesh K > > > > > > On Sun, Dec 25, 2011 at 1:33 AM, Muhammad Shahzad > > wrote: > >> > >> I think India uses polarity reversal to indicate start of caller id on > >> PRI > >> line, which i guess is not fully supported by FreeTDM yet. However, you > >> can > >> try the following FreeTDM options, though i am not sure if they will > >> work. > >> > >> > >> > >> > >> > >> You may try to set last parameter to false and test again. Do let us > know > >> if it works or not. > >> > >> I am not much familiar with FreeTDM code, nor got the equipment to test, > >> so for those who have, please see below link regarding same problem in > >> Asterisk and how they solved it, > >> > >> https://issues.asterisk.org/print_bug_page.php?bug_id=6683 > >> > >> Hope this helps. > >> > >> Thank you. > >> > >> > >> On Sat, Dec 24, 2011 at 6:57 PM, Ryan V wrote: > >>> > >>> Hi, > >>> > >>> Thanks to all the support we got in mailing lists and our telco, we put > >>> our freeswitch PBX online today. > >>> > >>> We are using Sangoma A101 and A200 Cards. We are successfully receiving > >>> the calls on our PRI and Analogue lines. But, we don't get caller-id on > >>> our > >>> analogue Lines. In India, Caller id comes in as DTMF tones and I could > >>> see > >>> the caller ID being detected correctly in debug. > >>> > >>> I have seen this issue raised earlier on the list and Moises Silva from > >>> Sangoma offered to fix it if given access to the server. Here is the > >>> link > >>> http://comments.gmane.org/gmane.comp.telephony.freeswitch.user/32340 > >>> > >>> I wanted to know whether anyone has come up with a solution. I am open > >>> to > >>> give access to our server if Moises Silva or anyone else in Sangoma > >>> would > >>> like to have a look at it. > >>> > >>> Thanks, > >>> > >>> 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 > >>> > >> > >> > >> > >> -- > >> Muhammad Shahzad > >> ----------------------------------- > >> CISCO Rich Media Communication Specialist (CRMCS) > >> CISCO Certified Network Associate (CCNA) > >> Cell: +92 334 422 40 88 > >> MSN: shari_786pk at hotmail.com > >> Email: shaheryarkh at googlemail.com > >> > >> > _________________________________________________________________________ > >> Professional FreeSWITCH Consulting Services: > >> consulting at freeswitch.org > >> http://www.freeswitchsolutions.com > >> > >> > >> > >> > >> Official FreeSWITCH Sites > >> http://www.freeswitch.org > >> http://wiki.freeswitch.org > >> http://www.cluecon.com > >> > >> FreeSWITCH-users mailing list > >> FreeSWITCH-users at lists.freeswitch.org > >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > >> UNSUBSCRIBE: > http://lists.freeswitch.org/mailman/options/freeswitch-users > >> http://www.freeswitch.org > >> > > > > > > _________________________________________________________________________ > > Professional FreeSWITCH Consulting Services: > > consulting at freeswitch.org > > http://www.freeswitchsolutions.com > > > > > > > > > > Official FreeSWITCH Sites > > http://www.freeswitch.org > > http://wiki.freeswitch.org > > http://www.cluecon.com > > > > FreeSWITCH-users mailing list > > FreeSWITCH-users at lists.freeswitch.org > > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > > http://www.freeswitch.org > > > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at 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/20111227/de528bc0/attachment-0001.html From notlikeme75 at yahoo.com Tue Dec 27 16:49:50 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 27 Dec 2011 05:49:50 -0800 (PST) Subject: [Freeswitch-users] mod shout win32 In-Reply-To: References: Message-ID: <1324993790.43968.YahooMailNeo@web65312.mail.ac2.yahoo.com> sean, or anyone. what are the prerequisites and how do you actually build mod shout on windows machines? is there a document somewhere you can help me with? all i see on the wiki is linux. thanks. rodney ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Tuesday, December 27, 2011 7:20 AM Subject: FreeSWITCH-users Digest, Vol 66, Issue 173 ----- Forwarded Message ----- 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: mod_shout on win32 (Sean Devoy) ? 2. does streamFile callback provide file position? ? ? ? (peter at uringme.com) ? 3. sangoma A200 with mod_freetdm on windows?! (Tamer Higazi) ? 4. Building Freeswitch on OSX Lion - ./configure??? openssl error ? ? ? (James) ? 5. Re: Call log - multiple entries CDR?? Billing? (Sherif Omran) ? 6. Re: Caller ID on Analog Lines (guru singh) Thanks. FYI, I did build from source after getting all the pre-reqs. Sean -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Jeff Lenk Sent: Sunday, December 25, 2011 10:20 PM To: freeswitch-users at lists.freeswitch.org Subject: Re: [Freeswitch-users] mod_shout on win32 that module is not built in the default package. you will have to build from source to get that module as it stands. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-shout-on-win32-tp7126980p7 127192.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 I'm looking at the examples for mod_spidermonkey where streamFile can be interrupted by a DTMF.? There's dtmfcallback.js (http://wiki.freeswitch.org/wiki/Examples_dtmfcallback.js) and the Session_streamFile doc itself (http://wiki.freeswitch.org/wiki/Session_streamFile) They both use the third argument to the callback to get digits.digit or .duration, which I'm assuming is the length that the DTMF was pressed.? But, do the third or fourth arguments also include the position within the file?? I'm looking for a way to save the location in a voice prompt where a person pressed the DTMF -- either in seconds or samples or bytes from the beginning of the file. Also, for my own knowledge, is there a doc somewhere that explains all the possibile members of the third and fourth arguments (usually data and arg) in the callback from streamFile?? Thank you. Hi people! I got my A200 board running with 1 FXS module on Linux along with mod_freetdm, but I am facing problems getting it to run on Windows. From Sangoma I followed the instructions to set up the board on Windows7 winpipe module, which works so far. How do I get freeswitch with mod_freetdm to run on Windows that I can make use of the board (pbx) on a win machine? for any ideas, I would thank you. Tamer I'm using the latest git commit (01267cd6f5b9a243c42c571f5d161849a66b3c82) and running into an error configuring Freeswitch on OSX Lion: ? ./configure: line 12283: syntax error near unexpected token `openssl,' ./configure: line 12283: ` ? ?PKG_CHECK_MODULES(openssl, openssl,' configure: error: ./configure.gnu failed for libs/iksemel ./bootstrap.sh runs fine; ./configure eventually spits out the error above. ?Any tips for me? ?I am mainly trying to set it up for local development purposes, not production. ? Thanks. out 1002 1002 1001 12/27/2011 11:30:48 am 0:00:47 Details out 1002 1002 1001 12/27/2011 11:30:48 am 0:00:47 Details Call logs are duplicated for 1 call? How can i prevent this? --- On Tue, 12/27/11, curriegrad2004 wrote: >From: curriegrad2004 >Subject: Re: [Freeswitch-users] Call log - multiple entries CDR?? Billing? >To: "FreeSWITCH Users Help" >Date: Tuesday, December 27, 2011, 6:33 AM > > >xml_cdr does the job just fine... uuid_bridge is what you may want to be looking for > > >On Mon, Dec 26, 2011 at 7:20 PM, Sherif Omran wrote: > >Hi, >> >>I have the CDR enabled and see multiple logs for the same call. Can any body recommend a call log that works fine and could be extended to be used for billing? >> >>thanks in advance >> >>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 >> >> > >-----Inline Attachment Follows----- > > >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > Hi, I am the person whose thread you mentioned. I've also had somebody contact me off the list regarding this. What I did in my case was jut get some sort of dtmf converter. It costs Rs 200 and is Made in China. You plug in your FXO line in it and take another wire from it to your sangoma card. Caller ID is detected then. The only labeling it has on it is NetSonic. Somebody who contacted me tried to find it online but wasn't able to I think. I was referred to a shop here by the guy who I buy my Sangoma cards from. Muhammad Shahzad is right about polarity reversal, I confirmed this with the telco (Airtel). I do have hardware to test this or give access to somebody who wants to test a fix. But I just moved to a new place and dont have anything set up right now. It'll take a week before I have servers set up. Regards guru On Sun, Dec 25, 2011 at 7:42 AM, Ryan V wrote: > Caller ID on PRI line works fine. I have problem only with calls coming in > on FXO. > > Thanks, > > Venkatesh K > > > On Sun, Dec 25, 2011 at 1:33 AM, Muhammad Shahzad > wrote: >> >> I think India uses polarity reversal to indicate start of caller id on PRI >> line, which i guess is not fully supported by FreeTDM yet. However,?you can >> try the following FreeTDM options, though i am not sure if they will work. >> >> >> >> >> >> You may try to set last parameter to false and test again. Do let us know >> if it works or not. >> >> I am not much familiar with FreeTDM code, nor got the?equipment?to test, >> so for those who have, please see below link regarding same problem in >> Asterisk and how they solved it, >> >> https://issues.asterisk.org/print_bug_page.php?bug_id=6683 >> >> Hope this helps. >> >> Thank you. >> >> >> On Sat, Dec 24, 2011 at 6:57 PM, Ryan V wrote: >>> >>> Hi, >>> >>> Thanks to all the support we got in mailing lists and our telco, we put >>> our freeswitch PBX online today. >>> >>> We are using Sangoma A101 and A200 Cards. We are successfully receiving >>> the calls on our PRI and Analogue lines. But, we don't get caller-id on our >>> analogue Lines. In India, Caller id comes in as DTMF tones and I could see >>> the caller ID being detected correctly in debug. >>> >>> I have seen this issue raised earlier on the list and Moises Silva from >>> Sangoma offered to fix it if given access to the server. Here is the link >>> http://comments.gmane.org/gmane.comp.telephony.freeswitch.user/32340 >>> >>> I wanted to know whether anyone has come up with a solution. I am open to >>> give access to our server if Moises Silva or anyone else in Sangoma would >>> like to have a look at it. >>> >>> Thanks, >>> >>> 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 >>> >> >> >> >> -- >> Muhammad Shahzad >> ----------------------------------- >> CISCO Rich Media Communication Specialist (CRMCS) >> CISCO Certified Network Associate (CCNA) >> Cell: +92 334 422 40 88 >> MSN: shari_786pk at hotmail.com >> Email: shaheryarkh at googlemail.com >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org >> > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111227/f08ddd08/attachment-0001.html From jeff at jefflenk.com Tue Dec 27 18:34:52 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Tue, 27 Dec 2011 07:34:52 -0800 (PST) Subject: [Freeswitch-users] sangoma A200 with mod_freetdm on windows?! In-Reply-To: <4EF942F0.4010404@googlemail.com> References: <4EF942F0.4010404@googlemail.com> Message-ID: <1325000092695-7130382.post@n2.nabble.com> Thats a pretty open ended question. I can tell you that it is supported and it works well. You will have to build from source and you can use the freely available version of Visual Studio 2010 Express. The projects and solutions to do this are already built and available in Git. If you ask some more specific questions I will try to help. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/sangoma-A200-with-mod-freetdm-on-windows-tp7129722p7130382.html Sent from the freeswitch-users mailing list archive at Nabble.com. From jeff at jefflenk.com Tue Dec 27 18:42:26 2011 From: jeff at jefflenk.com (Jeff Lenk) Date: Tue, 27 Dec 2011 07:42:26 -0800 (PST) Subject: [Freeswitch-users] mod shout win32 In-Reply-To: <1324993790.43968.YahooMailNeo@web65312.mail.ac2.yahoo.com> References: <1324993790.43968.YahooMailNeo@web65312.mail.ac2.yahoo.com> Message-ID: <1325000546240-7130388.post@n2.nabble.com> Have you seen http://wiki.freeswitch.org/wiki/Installation_for_Windows otherwise post some more specific questions on what you are having problems with. The Git supplied VS projects will build mod_shout but are building some of the dependent libraries with an older version than on linux and this may cause you to see differences in behavior or other. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-shout-win32-tp7130183p7130388.html Sent from the freeswitch-users mailing list archive at Nabble.com. From notlikeme75 at yahoo.com Tue Dec 27 21:11:16 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Tue, 27 Dec 2011 10:11:16 -0800 (PST) Subject: [Freeswitch-users] prevention of duplicate calls. References: Message-ID: <1325009476.81990.YahooMailNeo@web65305.mail.ac2.yahoo.com> is there a method using xml that i can prevent callers three waying themselves. I find some idiots will do this so they can "produce" feedback into a conference room. I would like the system to automatically determine that they are already on the ivr and send them to a recorded message and hangup. or maybe auto hanging up the first call in case of "accidentals" from voips not hanging up? and continuing the first call. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111227/0f47600e/attachment.html From moises.silva at gmail.com Tue Dec 27 22:17:51 2011 From: moises.silva at gmail.com (Moises Silva) Date: Tue, 27 Dec 2011 12:17:51 -0700 Subject: [Freeswitch-users] sangoma A200 with mod_freetdm on windows?! In-Reply-To: <4EF942F0.4010404@googlemail.com> References: <4EF942F0.4010404@googlemail.com> Message-ID: On Mon, Dec 26, 2011 at 9:00 PM, Tamer Higazi wrote: > Hi people! > I got my A200 board running with 1 FXS module on Linux along with > mod_freetdm, but I am facing problems getting it to run on Windows. From > Sangoma I followed the instructions to set up the board on Windows7 > winpipe module, which works so far. > > How do I get freeswitch with mod_freetdm to run on Windows that I can > make use of the board (pbx) on a win machine? > > You may want to send an email to Sangoma support. They are working already in a wiki page for Windows setup, in the meantime they can help you with instructions via email. *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 VegaStream is now part of Sangoma! Ask us about both Gateway Appliances and Internal Gateways -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111227/148c3b72/attachment.html From darcy at primrose.ws Tue Dec 27 23:59:42 2011 From: darcy at primrose.ws (Darcy) Date: Tue, 27 Dec 2011 15:59:42 -0500 Subject: [Freeswitch-users] micrsoft voip plug in for fax Message-ID: Has anyone any experience using microsoft?s voip plug in for fax with the freeswitch, I only get about 1 in 10 completion, inbound works 100%, outbound will only work if the server is in the same building. Darcy -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111227/5a756ba0/attachment.html From th982a at googlemail.com Tue Dec 27 21:57:43 2011 From: th982a at googlemail.com (Tamer Higazi) Date: Tue, 27 Dec 2011 19:57:43 +0100 Subject: [Freeswitch-users] sangoma A200 with mod_freetdm on windows?! In-Reply-To: <1325000092695-7130382.post@n2.nabble.com> References: <4EF942F0.4010404@googlemail.com> <1325000092695-7130382.post@n2.nabble.com> Message-ID: <4EFA1527.4030309@googlemail.com> Hi Jeff! Thank you for your support- I saw there are batch files for 2008 and 2010 available. The batch file for 2008 processed with errors at the end. As well, I want to compile the freeswitch on 64 bit windows, and not on 32bit. what he did by default. I modified the patch file with the /p flag to use the "x64" bit profile. However, 53 errors were at the end available. I will post here what I did wrong. Are the procedures of compiling freeswitch on windows the same as on Linux?! I ment, in the modules.conf to comment out the stuff I want to have?! Do I have to be careful when compiling freeswitch along mod_freetdm ?! Tamer Am 27.12.2011 16:34, schrieb Jeff Lenk: > Thats a pretty open ended question. I can tell you that it is supported and > it works well. You will have to build from source and you can use the freely > available version of Visual Studio 2010 Express. The projects and solutions > to do this are already built and available in Git. If you ask some more > specific questions I will try to help. > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/sangoma-A200-with-mod-freetdm-on-windows-tp7129722p7130382.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 freeswitch at scottisheyes.com Tue Dec 27 21:27:28 2011 From: freeswitch at scottisheyes.com (James) Date: Tue, 27 Dec 2011 10:27:28 -0800 Subject: [Freeswitch-users] Building Freeswitch on OSX Lion - ./configure openssl error In-Reply-To: <1C786A574DB84D9D91FBC6F16961CB23@gmail.com> References: <1C786A574DB84D9D91FBC6F16961CB23@gmail.com> Message-ID: Thanks for the response, Seven - the openssl issue has been reported in JIRA already a couple of times. The workarounds there don't seem to work. See: http://jira.freeswitch.org/browse/FS-3642 Your workaround that you mentioned did not appear to work for me either - I actually ended up getting the error that you originally reported in this other JIRA issue: http://jira.freeswitch.org/browse/FS-3243 What do I need to do to fix that error? And by the way, I would say that last JIRA issue should be reopened, as it's happening on my Macbook Air (OSX Lion) too. On Tue, Dec 27, 2011 at 5:08 AM, Seven Du wrote: > 1. report on jira.freeswitch.og > > 2. for a workaround, try mv libs/iksemel to /tmp before configure and > move back before make if you don't need dingaling support. > > -- > About: http://about.me/dujinfang > Blog: http://www.dujinfang.com > Proj: http://www.freeswitch.org.cn > > Sent with Sparrow > > On Tuesday, December 27, 2011 at 1:37 PM, James wrote: > > I'm using the latest git commit (01267cd6f5b9a243c42c571f5d161849a66b3c82) > and running into an error configuring Freeswitch on OSX Lion: > > ./configure: line 12283: syntax error near unexpected token `openssl,' > ./configure: line 12283: ` PKG_CHECK_MODULES(openssl, openssl,' > configure: error: ./configure.gnu failed for libs/iksemel > > ./bootstrap.sh runs fine; ./configure eventually spits out the error > above. Any tips for me? I am mainly trying to set it up for local > development purposes, not production. > > 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/20111227/5d3ea7ba/attachment-0001.html From nvitaly at gmail.com Tue Dec 27 21:45:19 2011 From: nvitaly at gmail.com (Vitaly Nikolaev) Date: Tue, 27 Dec 2011 13:45:19 -0500 Subject: [Freeswitch-users] prevention of duplicate calls. In-Reply-To: <1325009476.81990.YahooMailNeo@web65305.mail.ac2.yahoo.com> References: <1325009476.81990.YahooMailNeo@web65305.mail.ac2.yahoo.com> Message-ID: Hello, You can try to use mod_limit http://wiki.freeswitch.org/wiki/Mod_limit hash by callerid+callid and set limit 1 call PS: i never used that this way but it might work On Tue, Dec 27, 2011 at 1:11 PM, Rodney wrote: > is there a method using xml that i can prevent callers three waying > themselves. I find some idiots will do this so they can "produce" feedback > into a conference room. I would like the system to automatically determine > that they are already on the ivr and send them to a recorded message and > hangup. or maybe auto hanging up the first call in case of "accidentals" > from voips not hanging up and continuing the first call. > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -- -- Vitaly Nikolaev -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111227/70b5b6bf/attachment.html From th982a at googlemail.com Wed Dec 28 02:51:11 2011 From: th982a at googlemail.com (Tamer Higazi) Date: Wed, 28 Dec 2011 00:51:11 +0100 Subject: [Freeswitch-users] sangoma A200 with mod_freetdm on windows?! In-Reply-To: References: <4EF942F0.4010404@googlemail.com> Message-ID: <4EFA59EF.3040001@googlemail.com> Hi Moises! I allready sent one: http://support.sangoma.com/index.php?/Tickets/Ticket/View/460 they haven't replied since 2 days :( Tamer Am 27.12.2011 20:17, schrieb Moises Silva: > On Mon, Dec 26, 2011 at 9:00 PM, Tamer Higazi > wrote: > > Hi people! > I got my A200 board running with 1 FXS module on Linux along with > mod_freetdm, but I am facing problems getting it to run on Windows. From > Sangoma I followed the instructions to set up the board on Windows7 > winpipe module, which works so far. > > How do I get freeswitch with mod_freetdm to run on Windows that I can > make use of the board (pbx) on a win machine? > > > You may want to send an email to Sangoma support. They are working > already in a wiki page for Windows setup, in the meantime they can help > you with instructions via email. > > *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 > > > VegaStream is now part of Sangoma! > > > Ask us about both Gateway Appliances > and Internal > Gateways > > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From freeswitch at earthspike.net Wed Dec 28 03:14:55 2011 From: freeswitch at earthspike.net (John) Date: Wed, 28 Dec 2011 00:14:55 +0000 Subject: [Freeswitch-users] sangoma A200 with mod_freetdm on windows?! In-Reply-To: <4EFA59EF.3040001@googlemail.com> References: <4EF942F0.4010404@googlemail.com> <4EFA59EF.3040001@googlemail.com> Message-ID: <4EFA5F7F.4040105@earthspike.net> Tamer, This is linked from the Sangoma Support front page: http://support.sangoma.com/index.php?/News/NewsItem/View/1/sangoma-holiday-schedule John On 27/12/11 23:51, Tamer Higazi wrote: > Hi Moises! > I allready sent one: > > http://support.sangoma.com/index.php?/Tickets/Ticket/View/460 > > they haven't replied since 2 days :( > > > Tamer > > Am 27.12.2011 20:17, schrieb Moises Silva: >> On Mon, Dec 26, 2011 at 9:00 PM, Tamer Higazi> > wrote: >> >> Hi people! >> I got my A200 board running with 1 FXS module on Linux along with >> mod_freetdm, but I am facing problems getting it to run on Windows. From >> Sangoma I followed the instructions to set up the board on Windows7 >> winpipe module, which works so far. >> >> How do I get freeswitch with mod_freetdm to run on Windows that I can >> make use of the board (pbx) on a win machine? >> >> >> You may want to send an email to Sangoma support. They are working >> already in a wiki page for Windows setup, in the meantime they can help >> you with instructions via email. >> >> *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 >> >> >> VegaStream is now part of Sangoma! >> >> >> Ask us about both Gateway Appliances >> and Internal >> Gateways >> >> >> >> >> _________________________________________________________________________ >> Professional FreeSWITCH Consulting Services: >> consulting at freeswitch.org >> http://www.freeswitchsolutions.com >> >> >> >> >> Official FreeSWITCH Sites >> http://www.freeswitch.org >> http://wiki.freeswitch.org >> http://www.cluecon.com >> >> FreeSWITCH-users mailing list >> FreeSWITCH-users at lists.freeswitch.org >> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >> http://www.freeswitch.org > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > From curriegrad2004 at gmail.com Wed Dec 28 04:31:30 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 27 Dec 2011 17:31:30 -0800 Subject: [Freeswitch-users] Call log - multiple entries CDR?? Billing? In-Reply-To: <1324987850.98586.YahooMailClassic@web110814.mail.gq1.yahoo.com> References: <1324987850.98586.YahooMailClassic@web110814.mail.gq1.yahoo.com> Message-ID: This looks like a misconfigured box. Try setting the cdr_csv.conf.xml conf file's legs param to a only: If it is using XML CDR's then you'll need to go into the xml_cdr.conf.xml and set this param: to false All of the configuration files can be found under the autoload_configs in the conf root of your FreeSWITCH configuration folder... assuming you are using the default dialplan configuration On Tue, Dec 27, 2011 at 4:10 AM, Sherif Omran wrote: > out10021002100112/27/2011 11:30:48 am0:00:47Details > out10021002100112/27/2011 11:30:48 am0:00:47Details > > Call logs are duplicated for 1 call? How can i prevent this? > > > > > > --- On *Tue, 12/27/11, curriegrad2004 * wrote: > > > From: curriegrad2004 > Subject: Re: [Freeswitch-users] Call log - multiple entries CDR?? Billing? > To: "FreeSWITCH Users Help" > Date: Tuesday, December 27, 2011, 6:33 AM > > xml_cdr does the job just fine... uuid_bridge is what you may want to be > looking for > > On Mon, Dec 26, 2011 at 7:20 PM, Sherif Omran > > wrote: > > Hi, > > I have the CDR enabled and see multiple logs for the same call. Can any > body recommend a call log that works fine and could be extended to be used > for billing? > > thanks in advance > > 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 > > > > -----Inline Attachment Follows----- > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111227/4297af92/attachment-0001.html From curriegrad2004 at gmail.com Wed Dec 28 04:32:55 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Tue, 27 Dec 2011 17:32:55 -0800 Subject: [Freeswitch-users] micrsoft voip plug in for fax In-Reply-To: References: Message-ID: Can you please elicit more details for the Microsoft VoIP fax plugin? There are a few of them out there, so detailing which plugin can really help us out here. On Tue, Dec 27, 2011 at 12:59 PM, Darcy wrote: > Has anyone any experience using microsoft?s voip plug in for fax with the > freeswitch, I only get about 1 in 10 completion, inbound works 100%, > outbound will only work if the server is in the same building. > > 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 > From darcy at primrose.ws Wed Dec 28 05:21:05 2011 From: darcy at primrose.ws (Darcy) Date: Tue, 27 Dec 2011 21:21:05 -0500 Subject: [Freeswitch-users] micrsoft voip plug in for fax In-Reply-To: References: Message-ID: <4E7432A64CFC4DB188E0B850F3D3BB19@DWP> It is the one actually funded by microsoft, created by FaxBax (www.faxback.com). I have used it for a few years with other products with good success. I am trying to capture some events the leads to the failure, it appears to happen in the handshake process. -----Original Message----- From: curriegrad2004 Sent: Tuesday, December 27, 2011 8:32 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] micrsoft voip plug in for fax Can you please elicit more details for the Microsoft VoIP fax plugin? There are a few of them out there, so detailing which plugin can really help us out here. On Tue, Dec 27, 2011 at 12:59 PM, Darcy wrote: > Has anyone any experience using microsoft?s voip plug in for fax with the > freeswitch, I only get about 1 in 10 completion, inbound works 100%, > outbound will only work if the server is in the same building. > > 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 From mays.david at gmail.com Wed Dec 28 06:27:53 2011 From: mays.david at gmail.com (dma) Date: Tue, 27 Dec 2011 19:27:53 -0800 (PST) Subject: [Freeswitch-users] Error "Channels not ready" when bridge call in Lua In-Reply-To: References: <1324393096164-7112050.post@n2.nabble.com> <4EF0A757.8070404@tagnet.ru> <1324431091351-7113903.post@n2.nabble.com> Message-ID: <1325042873388-7131678.post@n2.nabble.com> Hi Michael, Sorry for the confusion. The problem still occurring after upgrade the latest version of Feeswitch using git head. Simply, after bridge the 2 sessions (inbound and outboud), the error is "channels not ready" and the call hang up. 2011-12-13 18:08:00.674769 [ERR] switch_cpp.cpp:1247 Channels not ready 2011-12-13 18:08:00.674769 [DEBUG] switch_cpp.cpp:618 CoreSession::hangup The reason is the status of the channel for inbound call is NOT ready. Do you have any idea about the cause of this issue? Thanks, D.Ma -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-Channels-not-ready-when-bridge-call-in-Lua-tp7112050p7131678.html Sent from the freeswitch-users mailing list archive at Nabble.com. From darcy at primrose.ws Wed Dec 28 06:46:55 2011 From: darcy at primrose.ws (Darcy) Date: Tue, 27 Dec 2011 22:46:55 -0500 Subject: [Freeswitch-users] micrsoft voip plug in for fax In-Reply-To: <4E7432A64CFC4DB188E0B850F3D3BB19@DWP> References: <4E7432A64CFC4DB188E0B850F3D3BB19@DWP> Message-ID: By the way, I am running FreeSWITCH Version 1.0.head (git-d93ed90 2011-11-11 20-17-21 -0600) -----Original Message----- From: Darcy Sent: Tuesday, December 27, 2011 9:21 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] micrsoft voip plug in for fax It is the one actually funded by microsoft, created by FaxBax (www.faxback.com). I have used it for a few years with other products with good success. I am trying to capture some events the leads to the failure, it appears to happen in the handshake process. -----Original Message----- From: curriegrad2004 Sent: Tuesday, December 27, 2011 8:32 PM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] micrsoft voip plug in for fax Can you please elicit more details for the Microsoft VoIP fax plugin? There are a few of them out there, so detailing which plugin can really help us out here. On Tue, Dec 27, 2011 at 12:59 PM, Darcy wrote: > Has anyone any experience using microsoft?s voip plug in for fax with the > freeswitch, I only get about 1 in 10 completion, inbound works 100%, > outbound will only work if the server is in the same building. > > 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 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.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 Dec 28 07:22:29 2011 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Tue, 27 Dec 2011 20:22:29 -0800 (PST) Subject: [Freeswitch-users] nibblebil odbc error In-Reply-To: Message-ID: <1325046149.53723.YahooMailClassic@web110807.mail.gq1.yahoo.com> Hi, I am trying to install nibblebill, I have FS installed with ./configure module odbc was enabled When i try isql "freeswitch" root "password" I get the following [ISQL]ERROR: Could not SQLConnect My files are the same as here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066055.html any help is appreciated thank you in advance kind regards, Sherif Omran From govoiper at gmail.com Wed Dec 28 07:37:35 2011 From: govoiper at gmail.com (Sammy Govind) Date: Wed, 28 Dec 2011 09:37:35 +0500 Subject: [Freeswitch-users] nibblebil odbc error In-Reply-To: <1325046149.53723.YahooMailClassic@web110807.mail.gq1.yahoo.com> References: <1325046149.53723.YahooMailClassic@web110807.mail.gq1.yahoo.com> Message-ID: Hi, Can you confirm that you've _correctly_ setup the odbc connectors in your OS ?! If your odbc.ini and odbcinst.ini files are correclt configured then you should be able to test the odbc connection using isql command. How did you enable the odbc !? Regards, Sammy On Wed, Dec 28, 2011 at 9:22 AM, Sherif Omran wrote: > Hi, > > I am trying to install nibblebill, I have FS installed with ./configure > module odbc was enabled > > When i try > isql "freeswitch" root "password" > > I get the following > > [ISQL]ERROR: Could not SQLConnect > > My files are the same as here > > http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066055.html > > any help is appreciated > > thank you in advance > > kind 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 > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111228/29cf4987/attachment.html From steveu at coppice.org Wed Dec 28 08:16:00 2011 From: steveu at coppice.org (Steve Underwood) Date: Wed, 28 Dec 2011 13:16:00 +0800 Subject: [Freeswitch-users] micrsoft voip plug in for fax In-Reply-To: <4E7432A64CFC4DB188E0B850F3D3BB19@DWP> References: <4E7432A64CFC4DB188E0B850F3D3BB19@DWP> Message-ID: <4EFAA610.9050100@coppice.org> Hi Darcy, On 12/28/2011 10:21 AM, Darcy wrote: > It is the one actually funded by microsoft, created by FaxBax > (www.faxback.com). I have used it for a few years with other products with > good success. I am trying to capture some events the leads to the failure, > it appears to happen in the handshake process. So you are talking about using Faxback's VoIP plugin for Microsoft FAX in T.38 mode, and sending from Microsoft FAX and scan? That should work OK. Perhaps something has been altered in the negotiation in an incompatible way. Can you get a pcap file of an exchange with the Freeswitch server, and we can try to figure out what is going wrong. If I remember rightly Faxback has a lot of compatibility problems, because they try to start in T.38 mode, which a lot of VoIP systems don't like. However, I might be remembering an old problem that went away. Steve > -----Original Message----- > From: curriegrad2004 > Sent: Tuesday, December 27, 2011 8:32 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] micrsoft voip plug in for fax > > Can you please elicit more details for the Microsoft VoIP fax plugin? > There are a few of them out there, so detailing which plugin can > really help us out here. > > On Tue, Dec 27, 2011 at 12:59 PM, Darcy wrote: >> Has anyone any experience using microsoft?s voip plug in for fax with the >> freeswitch, I only get about 1 in 10 completion, inbound works 100%, >> outbound will only work if the server is in the same building. >> >> 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > From peter.olsson at visionutveckling.se Wed Dec 28 10:36:01 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 28 Dec 2011 07:36:01 +0000 Subject: [Freeswitch-users] Building Freeswitch on OSX Lion - ./configure openssl error In-Reply-To: References: <1C786A574DB84D9D91FBC6F16961CB23@gmail.com> Message-ID: <1FFF97C269757C458224B7C895F35F150183EB@cantor.std.visionutv.se> If the error in FS-3243 still exists (conflicting uuid_t types), please reopen again. I can?t reproduce this on my Lion installation, but it seems someone needs to dig in deeper on this one. OpenSSL has two issues, one in iksemel, which needs pkg-config to be installed to work properly (or apply the patch included in issue FS-3642). The other issue is deprecated SSL functions in OSX Lion, a workaround can be found in issue FS-3450. /Peter Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r James Skickat: den 27 december 2011 19:27 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] Building Freeswitch on OSX Lion - ./configure openssl error Thanks for the response, Seven - the openssl issue has been reported in JIRA already a couple of times. The workarounds there don't seem to work. See: http://jira.freeswitch.org/browse/FS-3642 Your workaround that you mentioned did not appear to work for me either - I actually ended up getting the error that you originally reported in this other JIRA issue: http://jira.freeswitch.org/browse/FS-3243 What do I need to do to fix that error? And by the way, I would say that last JIRA issue should be reopened, as it's happening on my Macbook Air (OSX Lion) too. On Tue, Dec 27, 2011 at 5:08 AM, Seven Du > wrote: 1. report on jira.freeswitch.og 2. for a workaround, try mv libs/iksemel to /tmp before configure and move back before make if you don't need dingaling support. -- About: http://about.me/dujinfang Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn Sent with Sparrow On Tuesday, December 27, 2011 at 1:37 PM, James wrote: I'm using the latest git commit (01267cd6f5b9a243c42c571f5d161849a66b3c82) and running into an error configuring Freeswitch on OSX Lion: ./configure: line 12283: syntax error near unexpected token `openssl,' ./configure: line 12283: ` PKG_CHECK_MODULES(openssl, openssl,' configure: error: ./configure.gnu failed for libs/iksemel ./bootstrap.sh runs fine; ./configure eventually spits out the error above. Any tips for me? I am mainly trying to set it up for local development purposes, not production. 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 !DSPAM:4efa5a9c32768776921018! -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111228/68803cd2/attachment.html From peter.olsson at visionutveckling.se Wed Dec 28 10:40:17 2011 From: peter.olsson at visionutveckling.se (Peter Olsson) Date: Wed, 28 Dec 2011 07:40:17 +0000 Subject: [Freeswitch-users] sangoma A200 with mod_freetdm on windows?! In-Reply-To: <4EFA1527.4030309@googlemail.com> References: <4EF942F0.4010404@googlemail.com> <1325000092695-7130382.post@n2.nabble.com> <4EFA1527.4030309@googlemail.com> Message-ID: <1FFF97C269757C458224B7C895F35F1501884F@cantor.std.visionutv.se> The procedure to build FS on Windows is usually quite simple. Follow the instructions on the wiki, especially how to configure git before checking out the code. Then you just need to open the solution file and build. modules.conf doesn't exist, it builds most modules by default. If you build freetdm stuff uoy will need to install the Sangome provided wanpipe and sng_isdn packages, and also make sure that the include/library-patchs will find the files provided by these packages. Also, freetdm is a separate package, the solution file can be found under libs\freetdm (it not included in FS mail build solution, since it's a standalone project). /Peter -----Ursprungligt meddelande----- Fr?n: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] F?r Tamer Higazi Skickat: den 27 december 2011 19:58 Till: FreeSWITCH Users Help ?mne: Re: [Freeswitch-users] sangoma A200 with mod_freetdm on windows?! Hi Jeff! Thank you for your support- I saw there are batch files for 2008 and 2010 available. The batch file for 2008 processed with errors at the end. As well, I want to compile the freeswitch on 64 bit windows, and not on 32bit. what he did by default. I modified the patch file with the /p flag to use the "x64" bit profile. However, 53 errors were at the end available. I will post here what I did wrong. Are the procedures of compiling freeswitch on windows the same as on Linux?! I ment, in the modules.conf to comment out the stuff I want to have?! Do I have to be careful when compiling freeswitch along mod_freetdm ?! Tamer Am 27.12.2011 16:34, schrieb Jeff Lenk: > Thats a pretty open ended question. I can tell you that it is > supported and it works well. You will have to build from source and > you can use the freely available version of Visual Studio 2010 > Express. The projects and solutions to do this are already built and > available in Git. If you ask some more specific questions I will try to help. > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/sangoma-A200-with-mod-fr > eetdm-on-windows-tp7129722p7130382.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-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:4efa5a9a32765031432928! From hynek.cihlar at gmail.com Wed Dec 28 14:20:09 2011 From: hynek.cihlar at gmail.com (Hynek Cihlar) Date: Wed, 28 Dec 2011 12:20:09 +0100 Subject: [Freeswitch-users] Silence ESL event Message-ID: <-3489375824995432417@unknownmsgid> Hi all, is there any way to receive a silence-detected event through ESL? The use case is to hangup the call in case nothing interesting is going on on the channel. Sent from my mobile device From sharad at coraltele.com Wed Dec 28 14:50:41 2011 From: sharad at coraltele.com (sharad) Date: Wed, 28 Dec 2011 17:20:41 +0530 Subject: [Freeswitch-users] Freeswitch Sendmail & Email template References: Message-ID: <5AE61AC00F8E450DAAC5ECB092816799@sharad> Hello Do we have any API to make to shoot an email using some predefined template. Regards Sharad From sherifomran2000 at yahoo.com Wed Dec 28 15:54:04 2011 From: sherifomran2000 at yahoo.com (Sherif Omran) Date: Wed, 28 Dec 2011 04:54:04 -0800 (PST) Subject: [Freeswitch-users] nibblebil odbc error In-Reply-To: Message-ID: <1325076844.46097.YahooMailClassic@web110804.mail.gq1.yahoo.com> Hi Sammy Thank you for your email. Yes it is now connected. I can load the reload mod_nibblebill isql -v FreeSWITCH psswd +---------------------------------------+ | Connected!??????????????????????????? | |?????????????????????????????????????? | | sql-statement???????????????????????? | | help [tablename]????????????????????? | | quit????????????????????????????????? | |?????????????????????????????????????? | +---------------------------------------+ But how can i adjust and edit the accounts? regards, Sherif --- On Wed, 12/28/11, Sammy Govind wrote: From: Sammy Govind Subject: Re: [Freeswitch-users] nibblebil odbc error To: "FreeSWITCH Users Help" Date: Wednesday, December 28, 2011, 6:37 AM Hi,Can you confirm that you've _correctly_ setup the odbc connectors in your OS ?! If your odbc.ini and odbcinst.ini files are correclt configured then you should be able to test the odbc connection using isql command. How did you enable the odbc !? Regards,Sammy On Wed, Dec 28, 2011 at 9:22 AM, Sherif Omran wrote: Hi, I am trying to install nibblebill, I have FS installed with ./configure module odbc was enabled When i try isql "freeswitch" root "password" I get the following [ISQL]ERROR: Could not SQLConnect My files are the same as here http://lists.freeswitch.org/pipermail/freeswitch-users/2010-December/066055.html any help is appreciated thank you in advance kind 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 -----Inline Attachment Follows----- _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111228/a049bfbf/attachment.html From arnuld at phonologies.com Wed Dec 28 15:54:48 2011 From: arnuld at phonologies.com (Arnuld Uttre (Phonologies)) Date: Wed, 28 Dec 2011 18:24:48 +0530 Subject: [Freeswitch-users] freeswitch versions/releases Message-ID: At my company we are using freeswitch 1.0.7. I just chatted on #freeswitch and came to know that 1.0.7 was never an offical release. Even files.freeswitch.org does not have it. Last offical release was 1.0.6 and as of now 1.0.8 is in pipeline. It is advised to get freeswitch from Git for now which is extremely stable. I just want to conform that 1.0.7 was not official release. And is there any document on versioning/release system of freeswitch ? -- Arnuld Uttre Systems Software Engineer arnuld at Phonologies.COM http://www.phonologies.com Phonologies (India) Private Limited West Wing, Marri Deep, M. C. H. No. 12-5-4, Lallaguda, Secunderabad 500017, INDIA. Ph:+91-40-2701 8993 / 36 Fax:+91-40-2701 8992 From darcy at primrose.ws Wed Dec 28 16:09:24 2011 From: darcy at primrose.ws (Darcy) Date: Wed, 28 Dec 2011 08:09:24 -0500 Subject: [Freeswitch-users] micrsoft voip plug in for fax In-Reply-To: <4EFAA610.9050100@coppice.org> References: <4E7432A64CFC4DB188E0B850F3D3BB19@DWP> <4EFAA610.9050100@coppice.org> Message-ID: <230D0961A4424A09BA55A37949C535DE@DWP> Thanks, what actually happens is during the negotiation and t38 discovery, the freeswitch switches to the inside address, I just found this, so I need to see what is different that causes it to do that, it does start in the pcmu mode then switches to t38. There is a setting that allows me to manually input 'public IP address of NAT' and when I do that it works. I am going to do a pcap between the spa2102 and the voip plug in to see the difference. Darcy -----Original Message----- From: Steve Underwood Sent: Wednesday, December 28, 2011 12:16 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] micrsoft voip plug in for fax Hi Darcy, On 12/28/2011 10:21 AM, Darcy wrote: > It is the one actually funded by microsoft, created by FaxBax > (www.faxback.com). I have used it for a few years with other products > with > good success. I am trying to capture some events the leads to the > failure, > it appears to happen in the handshake process. So you are talking about using Faxback's VoIP plugin for Microsoft FAX in T.38 mode, and sending from Microsoft FAX and scan? That should work OK. Perhaps something has been altered in the negotiation in an incompatible way. Can you get a pcap file of an exchange with the Freeswitch server, and we can try to figure out what is going wrong. If I remember rightly Faxback has a lot of compatibility problems, because they try to start in T.38 mode, which a lot of VoIP systems don't like. However, I might be remembering an old problem that went away. Steve > -----Original Message----- > From: curriegrad2004 > Sent: Tuesday, December 27, 2011 8:32 PM > To: FreeSWITCH Users Help > Subject: Re: [Freeswitch-users] micrsoft voip plug in for fax > > Can you please elicit more details for the Microsoft VoIP fax plugin? > There are a few of them out there, so detailing which plugin can > really help us out here. > > On Tue, Dec 27, 2011 at 12:59 PM, Darcy wrote: >> Has anyone any experience using microsoft?s voip plug in for fax with the >> freeswitch, I only get about 1 in 10 completion, inbound works 100%, >> outbound will only work if the server is in the same building. >> >> 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 > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From arnuld at phonologies.com Wed Dec 28 16:19:23 2011 From: arnuld at phonologies.com (Arnuld Uttre (Phonologies)) Date: Wed, 28 Dec 2011 18:49:23 +0530 Subject: [Freeswitch-users] Can not receive Events Message-ID: I am running freeswitch. I have connected to it suing telnet on event socket. Through the same machine, using a different x-terminal, I have subscribed to: event plain all Content-Type: command/reply Reply-Text: +OK event listener enabled plain I am able to make calls using "api originate" command and got the call fine on my phone. I got +OK uuid-here too but I did not receive any events. I event tried event plain CHANNEL_DESTROY from different terminal but still no luck. I am usng freeswitch 1.0.7 on CentOS 5.5 with my own application configured in autoload_configs/myapp.conf.xml >From where I should start solving the problem ? Here are the contents of event_socket.conf.xml : -- Arnuld Uttre Systems Software Engineer arnuld at Phonologies.COM http://www.phonologies.com Phonologies (India) Private Limited West Wing, Marri Deep, M. C. H. No. 12-5-4, Lallaguda, Secunderabad 500017, INDIA. Ph:+91-40-2701 8993 / 36 Fax:+91-40-2701 8992 From nbhatti at gmail.com Wed Dec 28 18:04:46 2011 From: nbhatti at gmail.com (Muhammad Naseer Bhatti) Date: Wed, 28 Dec 2011 18:04:46 +0300 Subject: [Freeswitch-users] Monitor performance of modules Message-ID: Hello, Is it possible to monitor performance of modules loaded by FS? For example, mod_nibble or say mod_distributor. Is there a way to know how much time the module took from the time when called by FS till the module released data back to FS? Thanks. From rhow at exemail.com.au Wed Dec 28 18:58:25 2011 From: rhow at exemail.com.au (Ryan How) Date: Wed, 28 Dec 2011 23:58:25 +0800 Subject: [Freeswitch-users] Nat Setup Help Message-ID: <4EFB3CA1.7000903@exemail.com.au> Hi, I've been playing with freeswitch and having a bit of trouble getting devices to register from outside my network. I want devices to be able to use a single IP and connect whether they are inside or outside of the network. Network setup is a NAT router / gateway with a static IP. I've tried a few different setups as follows: Server inside the NAT. Works well, but external devices can't register. Tried port forwarding 5060-5091 & 16384-32768 but can't get devices to authenticate properly, I'm thinking maybe the ACL or the "domain" is causing issues but I don't know how to debug this). Also I want devices to be able to roam from internal to external, so it needs to use the external IP address all the time, but NAT reflection and VOIP just don't seem to work at all (It works for mail server and web server, so I don't know why not for VOIP). Server on gateway and multihomed. Again the internal / external IP issue... but it seems to work better than behind the NAT. Have a big of trouble with "domains", clients cannot authenticate because they don't exist on a different domain or something, I don't really understand it or know how to debug other than trial error, usually fixing 1 thing breaks another thing. Server on gateway and bound just to the external IP (I think!). Everything works as I want, except internal clients hear no sound, I don't know why, they can register and call fine. I think it must be a NAT issue? I Wouldn't think NAT would be involved here... Has anyone got a good way or pointers to how I should be trying to set this up?, The only way I've got it working so far is to VPN in and use it over that, but that adds a lot of overhead and makes it less reliable... This stuff does my head in :). Thanks! Ryan From krice at freeswitch.org Wed Dec 28 19:10:42 2011 From: krice at freeswitch.org (Ken Rice) Date: Wed, 28 Dec 2011 10:10:42 -0600 Subject: [Freeswitch-users] freeswitch versions/releases In-Reply-To: Message-ID: FreeSWITCH hasn't had an official release in quite some time... In fact it is highly recommended that you check out a current copy from git, build it and test it to make sure it works for your specific environment. There are plans to change this in the near future K On 12/28/11 6:54 AM, "Arnuld Uttre (Phonologies)" wrote: > At my company we are using freeswitch 1.0.7. I just chatted on #freeswitch > and came to know that 1.0.7 was never an offical release. Even > files.freeswitch.org does not have it. Last offical release was 1.0.6 and > as of now 1.0.8 is in pipeline. It is advised to get freeswitch from Git > for now which is extremely stable. > > I just want to conform that 1.0.7 was not official release. And is there > any document on versioning/release system of freeswitch ? > > > From msc at freeswitch.org Wed Dec 28 20:14:19 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 28 Dec 2011 09:14:19 -0800 Subject: [Freeswitch-users] FreeSWITCH Conference Call Today Message-ID: Hello all! Today on the FreeSWITCH conference call we will be discussing the event socket and how to generate calls. Please join us! The agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_12_28 Talk to you soon! -Michael -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111228/796a3fe4/attachment.html From curriegrad2004 at gmail.com Wed Dec 28 20:29:03 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 28 Dec 2011 09:29:03 -0800 Subject: [Freeswitch-users] freeswitch versions/releases In-Reply-To: References: Message-ID: 1.0.7 was really a snapshot from the git tree :P On Wed, Dec 28, 2011 at 8:10 AM, Ken Rice wrote: > FreeSWITCH hasn't had an official release in quite some time... In fact it > is highly recommended that you check out a current copy from git, build it > and test it to make sure it works for your specific environment. > > There are plans to change this in the near future > > K > > > On 12/28/11 6:54 AM, "Arnuld Uttre (Phonologies)" > wrote: > >> At my company we are using freeswitch 1.0.7. I just chatted on #freeswitch >> and came to know that 1.0.7 was never an offical release. Even >> files.freeswitch.org does not have it. Last offical release was 1.0.6 and >> as of now 1.0.8 is in pipeline. It is advised to get freeswitch from Git >> for now which is extremely stable. >> >> I just want to conform that 1.0.7 was not official release. And is there >> any document on ?versioning/release system of freeswitch ? >> >> >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org From msc at freeswitch.org Wed Dec 28 20:53:11 2011 From: msc at freeswitch.org (Michael Collins) Date: Wed, 28 Dec 2011 09:53:11 -0800 Subject: [Freeswitch-users] bind digit action In-Reply-To: <1324733986.20908.YahooMailNeo@web65303.mail.ac2.yahoo.com> References: <1324733986.20908.YahooMailNeo@web65303.mail.ac2.yahoo.com> Message-ID: maybe you could just change the digit action realm prior to entering the IVR? -MC On Sat, Dec 24, 2011 at 5:39 AM, Rodney wrote: > > Michael, > > I got BDA to work by manually configuring the dialplan, seems fusionpbx > was not doing it right when i added the actions. but now that I have it > working, my issue is using 0 as a transfer back to main menu. is there a > way to clear digit action on an IVR menu? > > scenario > > i have 9 static conference rooms > > bind digits are > > 0 transfer back to ivr room list > 6 to move back in room list > 7 to move up the room list > > to make this work, i clear bind digit on every conf room extension. the > problem is my ivr has option 0 to go back to previous menu, but its not > seen once you go into the room list bindings. I know making the IVR binding > a different number would work, but I would like to keep with the consistent > 0 to go back to previous menu. if there was a method to clear bind digit in > the IVR, that would be great. I have tried before the ivr answer and after, > with no luck. > > ie. pressing 0 , should go to room menu, and 0 again should go to main > menu, but as of now, pressing 0 is binded to room menu only :( > > > another good thing would be, i could set an expression with boundaries so > i would only need 1 extension or 1 conference control like a low boundary > of 501 and a high boundary of 509. so if someone presses 6 it would move > down the list variable {6=current conf # - 1 but if in room 1, go to room > 9, and if in room 9 pressing 7 would start over at 1} > > i accomplished this the long way with 9 separate extensions using bind > digits to move up or down the list. i had to manually add a transfer for > the next extension. > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111228/7fdf3f69/attachment.html From curriegrad2004 at gmail.com Thu Dec 29 01:47:09 2011 From: curriegrad2004 at gmail.com (curriegrad2004) Date: Wed, 28 Dec 2011 14:47:09 -0800 Subject: [Freeswitch-users] Nat Setup Help In-Reply-To: <4EFB3CA1.7000903@exemail.com.au> References: <4EFB3CA1.7000903@exemail.com.au> Message-ID: That's why there's ext-rtp-ip and ext-sip-ip varibles in the sofia conf file! You're getting close, however :P On Wed, Dec 28, 2011 at 7:58 AM, Ryan How wrote: > Hi, > > I've been playing with freeswitch and having a bit of trouble getting > devices to register from outside my network. I want devices to be able > to use a single IP and connect whether they are inside or outside of the > network. Network setup is a NAT router / gateway with a static IP. I've > tried a few different setups as follows: > > Server inside the NAT. Works well, but external devices can't register. > Tried port forwarding 5060-5091 & 16384-32768 but can't get devices to > authenticate properly, I'm thinking maybe the ACL or the "domain" is > causing issues but I don't know how to debug this). Also I want devices > to be able to roam from internal to external, so it needs to use the > external IP address all the time, but NAT reflection and VOIP just don't > seem to work at all (It works for mail server and web server, so I don't > know why not for VOIP). > > Server on gateway and multihomed. Again the internal / external IP > issue... but it seems to work better than behind the NAT. Have a big of > trouble with "domains", clients cannot authenticate because they don't > exist on a different domain or something, I don't really understand it > or know how to debug other than trial error, usually fixing 1 thing > breaks another thing. > > Server on gateway and bound just to the external IP (I think!). > Everything works as I want, except internal clients hear no sound, I > don't know why, they can register and call fine. I think it must be a > NAT issue? I Wouldn't think NAT would be involved here... > > Has anyone got a good way or pointers to how I should be trying to set > this up?, The only way I've got it working so far is to VPN in and use > it over that, but that adds a lot of overhead and makes it less reliable... > > This stuff does my head in :). > > Thanks! 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 From hrqiang at gmail.com Thu Dec 29 02:10:13 2011 From: hrqiang at gmail.com (RuiQiang Huang) Date: Wed, 28 Dec 2011 15:10:13 -0800 Subject: [Freeswitch-users] Help on RTMP Message-ID: Hi, I'm trying to add RTMP to my freeswitch server. I compiled mod_rtmp.so, mod_rtmp.la and put it under freeswitch/mod/. My freeswitch version is FreeSWITCH version: 1.0.head (git-c8c94f0 2011-08-19 11-52-40 -0500) My rtmp.conf.xml is However, I didn't see freeswitch listening on 1935 port and in fs_cli when I run freeswitch > rtmp status It's showing unknown command. My question is, is there anything else needed to get RTMP working? How to know which modules are loaded? Thanks. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111228/bae0821a/attachment-0001.html From brian at freeswitch.org Thu Dec 29 02:13:13 2011 From: brian at freeswitch.org (Brian West) Date: Wed, 28 Dec 2011 17:13:13 -0600 Subject: [Freeswitch-users] Help on RTMP In-Reply-To: References: Message-ID: <9CD21E5D-581E-45AB-854E-2CD973A0CCF0@freeswitch.org> make mod_rtmp-install do not copy the .so or .la yourself, chances are you copied the bash script wrapper the real .so is in .libs/mod_rtmp.so Secondly did you put it in modules.conf.xml? /b On Dec 28, 2011, at 5:10 PM, RuiQiang Huang wrote: > > I'm trying to add RTMP to my freeswitch server. I compiled mod_rtmp.so, > mod_rtmp.la and put it under freeswitch/mod/. > My freeswitch version is > FreeSWITCH version: 1.0.head (git-c8c94f0 2011-08-19 11-52-40 -0500) > My rtmp.conf.xml is -- 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/20111228/856bda03/attachment.html From hrqiang at gmail.com Thu Dec 29 02:44:33 2011 From: hrqiang at gmail.com (RuiQiang Huang) Date: Wed, 28 Dec 2011 15:44:33 -0800 Subject: [Freeswitch-users] Help on RTMP In-Reply-To: <9CD21E5D-581E-45AB-854E-2CD973A0CCF0@freeswitch.org> References: <9CD21E5D-581E-45AB-854E-2CD973A0CCF0@freeswitch.org> Message-ID: Thanks Brian. It's working now. On Wed, Dec 28, 2011 at 3:13 PM, Brian West wrote: > make mod_rtmp-install > > do not copy the .so or .la yourself, chances are you copied the bash > script wrapper the real .so is in .libs/mod_rtmp.so > > Secondly did you put it in modules.conf.xml? > > /b > > On Dec 28, 2011, at 5:10 PM, RuiQiang Huang wrote: > > > I'm trying to add RTMP to my freeswitch server. I compiled mod_rtmp.so, > mod_rtmp.la and put it under freeswitch/mod/. > My freeswitch version is > FreeSWITCH version: 1.0.head (git-c8c94f0 2011-08-19 11-52-40 -0500) > My rtmp.conf.xml is > > > -- > 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/20111228/328d414e/attachment.html From hrqiang at gmail.com Thu Dec 29 02:57:53 2011 From: hrqiang at gmail.com (RuiQiang Huang) Date: Wed, 28 Dec 2011 15:57:53 -0800 Subject: [Freeswitch-users] Help on RTMP In-Reply-To: References: <9CD21E5D-581E-45AB-854E-2CD973A0CCF0@freeswitch.org> Message-ID: My RTMP is running now. However, I still cannot get it authenticated. I change rtmp.conf.xml to ... and I have for user id 1015 in directory/default/v_1015.xml. I can authenticate SIP user 1015. But while I'm trying to login from Flex, it's showing authentication error. How to solve this? Thanks. On Wed, Dec 28, 2011 at 3:44 PM, RuiQiang Huang wrote: > Thanks Brian. > > It's working now. > > On Wed, Dec 28, 2011 at 3:13 PM, Brian West wrote: > >> make mod_rtmp-install >> >> do not copy the .so or .la yourself, chances are you copied the bash >> script wrapper the real .so is in .libs/mod_rtmp.so >> >> Secondly did you put it in modules.conf.xml? >> >> /b >> >> On Dec 28, 2011, at 5:10 PM, RuiQiang Huang wrote: >> >> >> I'm trying to add RTMP to my freeswitch server. I compiled mod_rtmp.so, >> mod_rtmp.la and put it under freeswitch/mod/. >> My freeswitch version is >> FreeSWITCH version: 1.0.head (git-c8c94f0 2011-08-19 11-52-40 -0500) >> My rtmp.conf.xml is >> >> >> -- >> 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/20111228/76ee8b1e/attachment.html From rhow at exemail.com.au Thu Dec 29 04:08:14 2011 From: rhow at exemail.com.au (Ryan How) Date: Thu, 29 Dec 2011 09:08:14 +0800 Subject: [Freeswitch-users] Nat Setup Help In-Reply-To: References: <4EFB3CA1.7000903@exemail.com.au> Message-ID: <4EFBBD7E.60201@exemail.com.au> So if I use the default config and just set ext-rtp-ip and ext-sip-ip in the internal profile to the static IP it might work? I've messed up something in the config now and it tells me invalid profile, I guess I shoulda taken a copy before messing too much so I can reset it :). On 29/12/2011 6:47 AM, curriegrad2004 wrote: > That's why there's ext-rtp-ip and ext-sip-ip varibles in the sofia conf file! > > You're getting close, however :P > > On Wed, Dec 28, 2011 at 7:58 AM, Ryan How wrote: >> Hi, >> >> I've been playing with freeswitch and having a bit of trouble getting >> devices to register from outside my network. I want devices to be able >> to use a single IP and connect whether they are inside or outside of the >> network. Network setup is a NAT router / gateway with a static IP. I've >> tried a few different setups as follows: >> >> Server inside the NAT. Works well, but external devices can't register. >> Tried port forwarding 5060-5091& 16384-32768 but can't get devices to >> authenticate properly, I'm thinking maybe the ACL or the "domain" is >> causing issues but I don't know how to debug this). Also I want devices >> to be able to roam from internal to external, so it needs to use the >> external IP address all the time, but NAT reflection and VOIP just don't >> seem to work at all (It works for mail server and web server, so I don't >> know why not for VOIP). >> >> Server on gateway and multihomed. Again the internal / external IP >> issue... but it seems to work better than behind the NAT. Have a big of >> trouble with "domains", clients cannot authenticate because they don't >> exist on a different domain or something, I don't really understand it >> or know how to debug other than trial error, usually fixing 1 thing >> breaks another thing. >> >> Server on gateway and bound just to the external IP (I think!). >> Everything works as I want, except internal clients hear no sound, I >> don't know why, they can register and call fine. I think it must be a >> NAT issue? I Wouldn't think NAT would be involved here... >> >> Has anyone got a good way or pointers to how I should be trying to set >> this up?, The only way I've got it working so far is to VPN in and use >> it over that, but that adds a lot of overhead and makes it less reliable... >> >> This stuff does my head in :). >> >> Thanks! 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 From mays.david at gmail.com Thu Dec 29 05:52:38 2011 From: mays.david at gmail.com (dma) Date: Wed, 28 Dec 2011 18:52:38 -0800 (PST) Subject: [Freeswitch-users] Error "Channels not ready" when bridge call in Lua In-Reply-To: <1325042873388-7131678.post@n2.nabble.com> References: <1324393096164-7112050.post@n2.nabble.com> <4EF0A757.8070404@tagnet.ru> <1324431091351-7113903.post@n2.nabble.com> <1325042873388-7131678.post@n2.nabble.com> Message-ID: <1325127158488-7134642.post@n2.nabble.com> Hi Guys, Any idea on this problem? Regards, D.Ma -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-Channels-not-ready-when-bridge-call-in-Lua-tp7112050p7134642.html Sent from the freeswitch-users mailing list archive at Nabble.com. From vetali100 at gmail.com Thu Dec 29 08:23:29 2011 From: vetali100 at gmail.com (Vitalie Colosov) Date: Wed, 28 Dec 2011 21:23:29 -0800 Subject: [Freeswitch-users] Error "Channels not ready" when bridge call in Lua In-Reply-To: <1325127158488-7134642.post@n2.nabble.com> References: <1324393096164-7112050.post@n2.nabble.com> <4EF0A757.8070404@tagnet.ru> <1324431091351-7113903.post@n2.nabble.com> <1325042873388-7131678.post@n2.nabble.com> <1325127158488-7134642.post@n2.nabble.com> Message-ID: Sometimes it happened for me when session2 was not yet ready, and I fixed this by adding a small workaround: "session:sleep(500);" Please see below, and reply back if it works for you. -------------------- session:answer() session2 = freeswitch.Session(callstring) *session:sleep(500);* if (session2:ready()) then freeswitch.bridge(session, session2) end -------------------- 2011/12/28 dma > Hi Guys, > > Any idea on this problem? > > Regards, > D.Ma > > -- > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Error-Channels-not-ready-when-bridge-call-in-Lua-tp7112050p7134642.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/20111228/63a0ca50/attachment.html From boris at tagnet.ru Thu Dec 29 08:51:51 2011 From: boris at tagnet.ru (Boris Kovalenko) Date: Thu, 29 Dec 2011 11:51:51 +0600 Subject: [Freeswitch-users] Error "Channels not ready" when bridge call in Lua In-Reply-To: <1325042873388-7131678.post@n2.nabble.com> References: <1324393096164-7112050.post@n2.nabble.com> <4EF0A757.8070404@tagnet.ru> <1324431091351-7113903.post@n2.nabble.com> <1325042873388-7131678.post@n2.nabble.com> Message-ID: <4EFBFFF7.1050301@tagnet.ru> Hello! Try to add ignore_early_media=true to the originate string. > Hi Michael, > > Sorry for the confusion. > > The problem still occurring after upgrade the latest version of Feeswitch > using git head. > > Simply, after bridge the 2 sessions (inbound and outboud), the error is > "channels not ready" and the call hang up. > > 2011-12-13 18:08:00.674769 [ERR] switch_cpp.cpp:1247 Channels not ready > 2011-12-13 18:08:00.674769 [DEBUG] switch_cpp.cpp:618 CoreSession::hangup > > The reason is the status of the channel for inbound call is NOT ready. Do > you have any idea about the cause of this issue? > > Thanks, > D.Ma > > -- > View this message in context: http://freeswitch-users.2379917.n2.nabble.com/Error-Channels-not-ready-when-bridge-call-in-Lua-tp7112050p7131678.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 -- Regards, Boris From govoiper at gmail.com Thu Dec 29 09:14:08 2011 From: govoiper at gmail.com (Sammy Govind) Date: Thu, 29 Dec 2011 11:14:08 +0500 Subject: [Freeswitch-users] Error "Channels not ready" when bridge call in Lua In-Reply-To: <4EFBFFF7.1050301@tagnet.ru> References: <1324393096164-7112050.post@n2.nabble.com> <4EF0A757.8070404@tagnet.ru> <1324431091351-7113903.post@n2.nabble.com> <1325042873388-7131678.post@n2.nabble.com> <4EFBFFF7.1050301@tagnet.ru> Message-ID: Hi, Can you please pastebin the FS console debug logs for this script, just want to know whats going on in the Console when the script is executed. Just wanted to see that the session you are trying to create really is creating or there is some error in there!? Also sleep() is a nice way to wait for sometime if freeswitch is taking time to create a new session object. -- Regards, Sammy On Thu, Dec 29, 2011 at 10:51 AM, Boris Kovalenko wrote: > Hello! > > Try to add ignore_early_media=true to the originate string. > > Hi Michael, > > > > Sorry for the confusion. > > > > The problem still occurring after upgrade the latest version of Feeswitch > > using git head. > > > > Simply, after bridge the 2 sessions (inbound and outboud), the error is > > "channels not ready" and the call hang up. > > > > 2011-12-13 18:08:00.674769 [ERR] switch_cpp.cpp:1247 Channels not ready > > 2011-12-13 18:08:00.674769 [DEBUG] switch_cpp.cpp:618 CoreSession::hangup > > > > The reason is the status of the channel for inbound call is NOT ready. Do > > you have any idea about the cause of this issue? > > > > Thanks, > > D.Ma > > > > -- > > View this message in context: > http://freeswitch-users.2379917.n2.nabble.com/Error-Channels-not-ready-when-bridge-call-in-Lua-tp7112050p7131678.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 > > > -- > Regards, > Boris > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111229/2773c3eb/attachment.html From fieldpeak at gmail.com Thu Dec 29 15:37:27 2011 From: fieldpeak at gmail.com (fieldpeak) Date: Thu, 29 Dec 2011 20:37:27 +0800 Subject: [Freeswitch-users] Using xmpp to control conference In-Reply-To: References: Message-ID: Is there anyone can help advise ? ? 2011-12-27 ??11:58?"fieldpeak" ??? > Dear friends, > > Could you any one give any tips or an example how to use xmpp protocol to > control FS to realize conference. > > i see the wiki, there is few docs for mod_xmpp_event ( > http://wiki.freeswitch.org/wiki/Mod_xmpp_event) > > also i've considered using mod_socket_event to control conferece, while i > prefer using xmpp since i've deployed xmpp server... > > Thanks! > -- > Regards, > Charles > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111229/041e9257/attachment.html From anita.hall at simmortel.com Thu Dec 29 15:40:57 2011 From: anita.hall at simmortel.com (Anita Hall) Date: Thu, 29 Dec 2011 18:10:57 +0530 Subject: [Freeswitch-users] Asterisk and Freeswitch on the same machine! Message-ID: Hi friends Since my efforts to find an open source solution for SS7 on freeswitch did not meet any success, we had to resort to a cheap trick! chan_ss7 <-> Asterisk <-> FreeSWITCH where * and FS are on the same machine. Since we wanted to do only 120 calls per machine, it has worked out pretty well. Doing more than 150 calls is meeting a CPU bottleneck (surprisingly from FS side!) One interesting thing, we noted was that FS was taking considerably more CPU cycles than Asterisk (sometimes thrice as much!) when the calls were sent from Asterisk to FS. The dialplan on Asterisk was playing gsm audio (demo context) while FS dialplan was just sleeping! I believe, this is surprising? When doing SIP calls from FS to Asterisk, the CPU loads are comparable. I was under the impression that FS take less CPU cycles? My team member is writing a report with all performance stats, shall share when it is done. Bye, Anita. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111229/abeba4f6/attachment.html From avi at avimarcus.net Thu Dec 29 15:47:49 2011 From: avi at avimarcus.net (Avi Marcus) Date: Thu, 29 Dec 2011 14:47:49 +0200 Subject: [Freeswitch-users] Asterisk and Freeswitch on the same machine! In-Reply-To: References: Message-ID: 1) Sangoma cards with an SS7 stack for FS, a much less hacky solution.. 2) Does your server end up transcoding? Is that why it's using so much cpu? -Avi On Thu, Dec 29, 2011 at 2:40 PM, Anita Hall wrote: > Hi friends > > Since my efforts to find an open source solution for SS7 on freeswitch did > not meet any success, we had to resort to a cheap trick! > > chan_ss7 <-> Asterisk <-> FreeSWITCH > > where * and FS are on the same machine. > > Since we wanted to do only 120 calls per machine, it has worked out pretty > well. Doing more than 150 calls is meeting a CPU bottleneck (surprisingly > from FS side!) > > One interesting thing, we noted was that FS was taking considerably more > CPU cycles than Asterisk (sometimes thrice as much!) when the calls were > sent from Asterisk to FS. The dialplan on Asterisk was playing gsm audio > (demo context) while FS dialplan was just sleeping! I believe, this is > surprising? > > When doing SIP calls from FS to Asterisk, the CPU loads are comparable. I > was under the impression that FS take less CPU cycles? > > My team member is writing a report with all performance stats, shall share > when it is done. > > Bye, > Anita. > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111229/fe9e8392/attachment-0001.html From notlikeme75 at yahoo.com Fri Dec 30 05:03:40 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Thu, 29 Dec 2011 18:03:40 -0800 (PST) Subject: [Freeswitch-users] BDA on IVR In-Reply-To: References: Message-ID: <1325210620.19309.YahooMailNeo@web65315.mail.ac2.yahoo.com> MC. temporarily, i had to create an pagd extension in place of the ivr, but this caused issues with a bda i was using so i really need an ivr. what is the method to change or clear a realm before entering an IVR? thanks. ________________________________ From: "freeswitch-users-request at lists.freeswitch.org" To: freeswitch-users at lists.freeswitch.org Sent: Wednesday, December 28, 2011 6:10 PM Subject: FreeSWITCH-users Digest, Vol 66, Issue 181 ----- Forwarded Message ----- 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. Nat Setup Help (Ryan How) ? 2. Re: freeswitch versions/releases (Ken Rice) ? 3. FreeSWITCH Conference Call Today (Michael Collins) ? 4. Re: freeswitch versions/releases (curriegrad2004) ? 5. Re: bind digit action (Michael Collins) ? 6. Re: Nat Setup Help (curriegrad2004) ? 7. Help on RTMP (RuiQiang Huang) Hi, I've been playing with freeswitch and having a bit of trouble getting devices to register from outside my network. I want devices to be able to use a single IP and connect whether they are inside or outside of the network. Network setup is a NAT router / gateway with a static IP. I've tried a few different setups as follows: Server inside the NAT. Works well, but external devices can't register. Tried port forwarding 5060-5091 & 16384-32768 but can't get devices to authenticate properly, I'm thinking maybe the ACL or the "domain" is causing issues but I don't know how to debug this). Also I want devices to be able to roam from internal to external, so it needs to use the external IP address all the time, but NAT reflection and VOIP just don't seem to work at all (It works for mail server and web server, so I don't know why not for VOIP). Server on gateway and multihomed. Again the internal / external IP issue... but it seems to work better than behind the NAT. Have a big of trouble with "domains", clients cannot authenticate because they don't exist on a different domain or something, I don't really understand it or know how to debug other than trial error, usually fixing 1 thing breaks another thing. Server on gateway and bound just to the external IP (I think!). Everything works as I want, except internal clients hear no sound, I don't know why, they can register and call fine. I think it must be a NAT issue? I Wouldn't think NAT would be involved here... Has anyone got a good way or pointers to how I should be trying to set this up?, The only way I've got it working so far is to VPN in and use it over that, but that adds a lot of overhead and makes it less reliable... This stuff does my head in :). Thanks! Ryan FreeSWITCH hasn't had an official release in quite some time... In fact it is highly recommended that you check out a current copy from git, build it and test it to make sure it works for your specific environment. There are plans to change this in the near future K On 12/28/11 6:54 AM, "Arnuld Uttre (Phonologies)" wrote: > At my company we are using freeswitch 1.0.7. I just chatted on #freeswitch > and came to know that 1.0.7 was never an offical release. Even > files.freeswitch.org does not have it. Last offical release was 1.0.6 and > as of now 1.0.8 is in pipeline. It is advised to get freeswitch from Git > for now which is extremely stable. > > I just want to conform that 1.0.7 was not official release. And is there > any document on? versioning/release system of freeswitch ? > > > Hello all! Today on the FreeSWITCH conference call we will be discussing the event socket and how to generate calls. Please join us! The agenda page is here: http://wiki.freeswitch.org/wiki/FS_weekly_2011_12_28 Talk to you soon! -Michael1.0.7 was really a snapshot from the git tree :P On Wed, Dec 28, 2011 at 8:10 AM, Ken Rice wrote: > FreeSWITCH hasn't had an official release in quite some time... In fact it > is highly recommended that you check out a current copy from git, build it > and test it to make sure it works for your specific environment. > > There are plans to change this in the near future > > K > > > On 12/28/11 6:54 AM, "Arnuld Uttre (Phonologies)" > wrote: > >> At my company we are using freeswitch 1.0.7. I just chatted on #freeswitch >> and came to know that 1.0.7 was never an offical release. Even >> files.freeswitch.org does not have it. Last offical release was 1.0.6 and >> as of now 1.0.8 is in pipeline. It is advised to get freeswitch from Git >> for now which is extremely stable. >> >> I just want to conform that 1.0.7 was not official release. And is there >> any document on ?versioning/release system of freeswitch ? >> >> >> > > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting at freeswitch.org > http://www.freeswitchsolutions.com > > > > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > FreeSWITCH-users mailing list > FreeSWITCH-users at lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org maybe you could just change the digit action realm prior to entering the IVR? -MC On Sat, Dec 24, 2011 at 5:39 AM, Rodney wrote: > >Michael, > >I got BDA to work by manually configuring the dialplan, seems fusionpbx was not doing it right when i added the actions. but now that I have it working, my issue is using 0 as a transfer back to main menu. is there a way to clear digit action on an IVR menu? > >scenario > >i have 9 static conference rooms > >bind digits are > >0 transfer back to ivr room list >6 to move back in room list >7 to move up the room list > >to make this work, i clear bind digit on every conf room extension. the problem is my ivr has option 0 to go back to previous menu, but its not seen once you go into the room list bindings. I know making the IVR binding a different number would work, but I would like to keep with the consistent 0 to go back to previous menu. if there was a method to clear bind digit in the IVR, that would be great. I have tried before the ivr answer and after, with no luck. > >ie. pressing 0 , should go to room menu, and 0 again should go to main menu, but as of now, pressing 0 is binded to room menu only :( > > >another good thing would be, i could set an expression with boundaries so i would only need 1 extension or 1 conference control like a low boundary of 501 and a high boundary of 509. so if someone presses 6 it would move down the list variable {6=current conf # - 1 but if in room 1, go to room 9, and if in room 9 pressing 7 would start over at 1} > >i accomplished this the long way with 9 separate extensions using bind digits to move up or down the list. i had to manually add a transfer for the next extension. >_________________________________________________________________________ >Professional FreeSWITCH Consulting Services: >consulting at freeswitch.org >http://www.freeswitchsolutions.com > > > > >Official FreeSWITCH Sites >http://www.freeswitch.org >http://wiki.freeswitch.org >http://www.cluecon.com > >FreeSWITCH-users mailing list >FreeSWITCH-users at lists.freeswitch.org >http://lists.freeswitch.org/mailman/listinfo/freeswitch-users >UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users >http://www.freeswitch.org > > That's why there's ext-rtp-ip and ext-sip-ip varibles in the sofia conf file! You're getting close, however :P On Wed, Dec 28, 2011 at 7:58 AM, Ryan How wrote: > Hi, > > I've been playing with freeswitch and having a bit of trouble getting > devices to register from outside my network. I want devices to be able > to use a single IP and connect whether they are inside or outside of the > network. Network setup is a NAT router / gateway with a static IP. I've > tried a few different setups as follows: > > Server inside the NAT. Works well, but external devices can't register. > Tried port forwarding 5060-5091 & 16384-32768 but can't get devices to > authenticate properly, I'm thinking maybe the ACL or the "domain" is > causing issues but I don't know how to debug this). Also I want devices > to be able to roam from internal to external, so it needs to use the > external IP address all the time, but NAT reflection and VOIP just don't > seem to work at all (It works for mail server and web server, so I don't > know why not for VOIP). > > Server on gateway and multihomed. Again the internal / external IP > issue... but it seems to work better than behind the NAT. Have a big of > trouble with "domains", clients cannot authenticate because they don't > exist on a different domain or something, I don't really understand it > or know how to debug other than trial error, usually fixing 1 thing > breaks another thing. > > Server on gateway and bound just to the external IP (I think!). > Everything works as I want, except internal clients hear no sound, I > don't know why, they can register and call fine. I think it must be a > NAT issue? I Wouldn't think NAT would be involved here... > > Has anyone got a good way or pointers to how I should be trying to set > this up?, The only way I've got it working so far is to VPN in and use > it over that, but that adds a lot of overhead and makes it less reliable... > > This stuff does my head in :). > > Thanks! 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 Hi,? I'm trying to add RTMP to my freeswitch server. I compiled mod_rtmp.so, mod_rtmp.la and put it under freeswitch/mod/. My freeswitch version is? FreeSWITCH version: 1.0.head (git-c8c94f0 2011-08-19 11-52-40 -0500) My rtmp.conf.xml is? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? ? However, I didn't see freeswitch listening on 1935 port and in fs_cli when I run? freeswitch > rtmp status It's showing unknown command. My question is, is there anything else needed to get RTMP working? How to know which modules are loaded?? Thanks.? _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111229/38ddecbe/attachment-0001.html From notlikeme75 at yahoo.com Fri Dec 30 05:08:30 2011 From: notlikeme75 at yahoo.com (Rodney) Date: Thu, 29 Dec 2011 18:08:30 -0800 (PST) Subject: [Freeswitch-users] create temporary code for 1 on 1 conference Message-ID: <1325210910.94115.YahooMailNeo@web65316.mail.ac2.yahoo.com> Is there a method i can use to generate a 3 digit number that syncs to that crazy long uuid number that stays valid until that caller hangs up? I would like to create this code and play the code before menu options then play it on demand from an extension. I would also like a method to exchange this code with another current caller to allow them to speak in a 1 on 1 conference together. how do i put these two individuals into a uuid bridge without using an operator to do so? I would like them to exchange codes verbally and press an option that asks for the other persons code and "match" them into a conf. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111229/89ddd06f/attachment.html From fieldpeak at gmail.com Fri Dec 30 06:32:32 2011 From: fieldpeak at gmail.com (fieldpeak) Date: Fri, 30 Dec 2011 11:32:32 +0800 Subject: [Freeswitch-users] Freeswitch integrated with Openfire Message-ID: Dear friends, Could anyone advise how to integrate FS work with Openfire like Asterisk can do, thanks. -- Regards, Charles -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111230/41d1fa11/attachment.html From arnuld at phonologies.com Fri Dec 30 14:19:15 2011 From: arnuld at phonologies.com (Arnuld Uttre (Phonologies)) Date: Fri, 30 Dec 2011 16:49:15 +0530 Subject: [Freeswitch-users] Connecting to Event Socket Message-ID: I mostly connect to event-socket using telnet and then subscribe to receive events. I want to connect to event-socket using a simple TCP client from within a C Program. Is it a goo idea or there is a better way ? -- Arnuld Uttre Systems Software Engineer arnuld at Phonologies.COM http://www.phonologies.com Phonologies (India) Private Limited West Wing, Marri Deep, M. C. H. No. 12-5-4, Lallaguda, Secunderabad 500017, INDIA. Ph:+91-40-2701 8993 / 36 Fax:+91-40-2701 8992 From Lenin-Ruslan at yandex.ru Fri Dec 30 11:42:11 2011 From: Lenin-Ruslan at yandex.ru (Lenin-Ruslan) Date: Fri, 30 Dec 2011 12:42:11 +0400 Subject: [Freeswitch-users] problem with date-time Message-ID: <811071325234531@web151.yandex.ru> Please help Windows 2008R2 FreeSWITCH Version 1.0.head (git-a58742d 2011-12-16 09-16-37 -0600) condition does not work /log 7 2011-12-30 12:06:29.570395 [DEBUG] switch_xml.c:2870 XML DateTime Check: date time[2011-12-30 12:06:29] =~ 2011-12-29 17:00:01~2012-01-02 11:00:01 (FAIL) From drazen.blanusa at nth.ch Fri Dec 30 15:27:42 2011 From: drazen.blanusa at nth.ch (Drazen Blanusa) Date: Fri, 30 Dec 2011 13:27:42 +0100 Subject: [Freeswitch-users] Connecting to Event Socket In-Reply-To: References: Message-ID: <003201ccc6ee$6d979af0$48c6d0d0$@blanusa@nth.ch> I'm usig it from java. This is good approach, but there is a chance that you should create custom events or mods on FS side. It depends of your SW architecture. -----Original Message----- From: freeswitch-users-bounces at lists.freeswitch.org [mailto:freeswitch-users-bounces at lists.freeswitch.org] On Behalf Of Arnuld Uttre (Phonologies) Sent: Friday, December 30, 2011 12:19 PM To: freeswitch-users at lists.freeswitch.org Subject: [Freeswitch-users] Connecting to Event Socket I mostly connect to event-socket using telnet and then subscribe to receive events. I want to connect to event-socket using a simple TCP client from within a C Program. Is it a goo idea or there is a better way ? -- Arnuld Uttre Systems Software Engineer arnuld at Phonologies.COM http://www.phonologies.com Phonologies (India) Private Limited West Wing, Marri Deep, M. C. H. No. 12-5-4, Lallaguda, Secunderabad 500017, INDIA. Ph:+91-40-2701 8993 / 36 Fax:+91-40-2701 8992 _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting at freeswitch.org http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://wiki.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users at lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org From greg.buzzard at yahoo.com Fri Dec 30 20:59:33 2011 From: greg.buzzard at yahoo.com (Greg Buzzard) Date: Fri, 30 Dec 2011 09:59:33 -0800 (PST) Subject: [Freeswitch-users] how do I play music on hold for A-leg while IVRing with B-leg Message-ID: <1325267973.42182.YahooMailNeo@web39704.mail.mud.yahoo.com> Hi All, new to FreeSwitch, great software! ?Read earlier posts, bought FS book and am about half-way through it and still have a question. Problem context: ?I'm trying to do a form of call screening for home use using a Lua script. ?I record a "self-intro" message from calling party (A-leg), would then like to play hold music for A-leg while creating new session and doing IVR dance with called party (B-leg). ?IVR objective is to see if B-leg wishes to accept, add caller-ID to whitelist or blacklist, and/or send to voicemail, etc. ?I have all of this working except the ability to play hold music to A-leg before either bridging them, sending them to voicemail or playing a message and dropping them. Specific problem: ?I don't have my head wrapped around the "clean" approach to addressing this problem. ?I.e., seems like I may want to?"park" the A-leg (with music), while I do the IVR with B-leg. ?Then either: (1) unpark A-leg and bridge to B-leg or (2) hang-up B-leg, unpark A-leg and send to VM or (3) other embellishments where I add caller-ID (if it exists) to either white or black lists and/or play recording to telemarketers to leave me alone (with no VM), etc. ?If this is a good approach, getting a few pointers on key commands/apps to use would be great.? FWIW, my first simple-minded approach to transfer caller (A-leg) to hold-music extension while connecting/IVRing with B-leg didn't work as execution of the transfer was held until the IVRing was done). ?Similar to what I saw in some earlier postings. ?If I get this figured out, I'm happy to summarize and post useful snippets to the email archive, wiki page or wherever, when done. -g -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20111230/448968e0/attachment.html