November 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Another Concurrent Calls Monitor Question
by DJB 09 Nov '09

09 Nov '09

I am also curious whether you can recommend how I can get the info if I want to see concurrent calls by account code. Let's say if I am running FS as SBC and I want to monitor concurrent calls per customer. I've looked at the HEARTBEAT, but it only gives me overall session-count. How safe is it if I have a cron job running every 5 minutes, and get the data from core.db in the calls tables. For instance, if I issue the following query: select count(*) from calls where substr(callee_chan_name,27,15)='$gw_ip_addr'; I don't want to query directly from core.db, so it would be great if I can use something from event socket to monitor per customer (account code) or ip address. Thank you.

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PortAudio needs work on Mac OS X 10.6
by Bruce Fletcher 09 Nov '09

09 Nov '09

I'm trying to get through the noobie tutorial that c888 recommends in IRC, but PortAudio doesn't seem to build properly on Mac OS X 10.6. It failed due to some code that wasn't 64bit ready, apparently. The error I got was exactly the same as this 4 month old error from MacPorts: http://trac.macports.org/ticket/20338#comment:4 Someone there produced the following patch, which almost got me going again: http://trac.macports.org/changeset/53938 This has to be manually applied because the line numbers don't match up between the MacPorts and FreeSWITCH versions of PortAudio. At least, not in the configure script. This still doesn't quite get portaudio built, but I found a suggestion to configure it with -- disable-shared here: http://music.columbia.edu/pipermail/portaudio/2008-February/008188.html This finally got me to the point of having a compiled version of portaudio, but of course with that much hacking around nothing is going to go smoothly, is it. I can boot freeswitch fine and 'load mod_portaudio' seems to work, but when I try to hear some MOH output, I get this: freeswitch(a)Media-Centre.local> pa call 9999 2009-11-08 00:50:51.894512 [NOTICE] switch_channel.c:613 New Channel portaudio/9999 [3079018b-0df8-4c61-bffa-2f2eb681d06d] Assertion failed: (sizeof( UInt32 ) == sizeof( long )), function ringBufferIOProc, file src/hostapi/coreaudio/pa_mac_core.c, line 1713. I'd be happy to poke around on the PortAudio site and see if there is any useful information or patches there, but it's really late now so I just wanted to document where I've gotten to in case someone else has the same problem or a better clue on how to fix things. This is all using version 15396 out of subversion, if it matters. Thanks, - Bruce

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Right way to start FS on CentOS at boot-time?
by Fred-145 09 Nov '09

09 Nov '09

Hello For those of you running FS on CentOS (5.4) who compiled FS from SVN, I'd like to make sure I'm doing it right to have FS start automatically at boot-time: 1. cp /usr/src/freeswitch/build/freeswitch.init.redhat /etc/init.d/freeswitch 2. vi /etc/init.d/freeswitch: PID_FILE=${PID_FILE-/usr/local/freeswitch/log/freeswitch.pid} FS_FILE=${FS_FILE-/usr/local/freeswitch/bin/freeswitch} FS_HOME=${FS_HOME-/usr/local/freeswitch} 3. chmod 755 /etc/init.d/freeswitch 4. chkconfig --level 345 freeswitch on 5. chkconfig --list freeswitch 6. (why needed in addition to chkconfig?) ln -s /etc/init.d/freeswitch /usr/sbin/rcfreeswitch To manually start the server: Launch the server through the rc.d script: /etc/init.d/freeswitch start Is the above correct? Thank you. -- View this message in context: http://old.nabble.com/Right-way-to-start-FS-on-CentOS-at-boot-time--tp26272… Sent from the Freeswitch-users mailing list archive at Nabble.com.

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Monitoring via SNMP
by Dimitar Dechev 09 Nov '09

09 Nov '09

Dear All, I couldn't find much information about how to monitor Freeswitch via SNMP like how many calls/legs I have, how many CAPs, and etc. One of the thing I do currently is to make simple bash script which in general runs "fs_cli -x 'show calls count'" or some other command and call that script via snmpd.conf. I would appreciate if somebody tell me if they is snmp module in Freeswitch, or provide with link/method from where I can read some information. Thanks!

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javascript parameter
by god.nirvana 09 Nov '09

09 Nov '09

hi all: how can i get the value of the myArg1 myArg2 in test.js. like this originate sofia/example/1000(a)somewhere.com '&javascript(test.js myArg1 myArg2)' thanks! 2009-11-09 god.nirvana

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building on Fedora 12
by shouldbe q931 09 Nov '09

09 Nov '09

Hi, While I appreciate that Fedora 12 is still only in beta. Because I want to try FusionPBX and I've had no success getting pfSense to work in a single NIC environment, and FusionPBX needs PHP 5.3, and Fedora 12 appears to be the first distro with PHP5.3... Anyway, these are the steps that I've done yum install autoconf automake binutils bison curl-devel db4 db4-devel expat-devel flex gcc-c++ gettext gdb gdbm gdbm-devel gnutls-devel httpd kernel-devel libogg-devel libtiff libtiff-devel libtool libvorbis-devel make mkxauth mysql-server mysql mysql-devel ncurses ncurses-devel ncurses-libs openssl-devel perl-Apache2-SOAP perl-devel php php-common php-mysql php-pdo php-soap php-xmlrpc postfix python python-devel screen sqlite sqlite-devel sqlite2 sqlite2-devel subversion unixODBC-devel wget wireshark wireshark-gnome zlib zlib-devel mkdir /usr/src/freeswitch mkdir /usr/src/freeswitch/svn svn checkout http://svn.freeswitch.org/svn/freeswitch/trunk /usr/src/freeswitch/svn cd /usr/src/freeswitch/svn ./bootstrap.sh ./configure make SVN reports Checked out external at revision 849. Checked out revision 15396. I then get the following error at the end -------------------------------------------------------------- Making all in soa Making all in tport LTCOMPILE tport_tls.lo cc1: warnings being treated as errors tport_tls.c: In function âtls_init_contextâ: tport_tls.c:280: error: assignment discards qualifiers from pointer target type tport_tls.c:282: error: assignment discards qualifiers from pointer target type tport_tls.c: In function âtls_post_connection_checkâ: tport_tls.c:527: error: assignment discards qualifiers from pointer target type make[9]: *** [tport_tls.lo] Error 1 make[8]: *** [all] Error 2 Making all in nta Making all in nth Making all in nea Making all in iptsec Making all in nua make[8]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. make[7]: *** [all-recursive] Error 1 Making all in packages make[6]: *** [all-recursive] Error 1 make[5]: *** [all] Error 2 make[4]: *** [//usr/src/freeswitch/svn/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 make[3]: *** [mod_sofia-all] Error 1 make[2]: *** [all-recursive] Error 1 Making all in build +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[1]: *** [all-recursive] Error 1 make: *** [all] Error 2 [root@conf-2 svn]# -------------------------------------------------------------- I then tried using the "quick and dirty" method from http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install, but get a similar failure -------------------------------------------------------------- Making all in nua make[9]: Entering directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua/nua' make[10]: Entering directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua/nua' LTCOMPILE nua.lo LTCOMPILE nua_common.lo LTCOMPILE nua_stack.lo LTCOMPILE nua_server.lo LTCOMPILE nua_client.lo LTCOMPILE nua_extension.lo LTCOMPILE nua_dialog.lo LTCOMPILE outbound.lo LTCOMPILE nua_params.lo LTCOMPILE nua_register.lo LTCOMPILE nua_registrar.lo LTCOMPILE nua_session.lo LTCOMPILE nua_options.lo LTCOMPILE nua_message.lo LTCOMPILE nua_publish.lo LTCOMPILE nua_subnotref.lo LTCOMPILE nua_notifier.lo LTCOMPILE nua_event_server.lo LTCOMPILE nua_tag.lo LTCOMPILE nua_tag_ref.lo LINK libnua.la make[10]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua/nua' make[9]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua/nua' make[9]: Entering directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua' make[9]: *** No rule to make target `tport/libtport.la', needed by `libsofia-sip-ua.la'. Stop. make[9]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua' make[8]: *** [all-recursive] Error 1 make[8]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua' Making all in packages make[8]: Entering directory `/usr/src/freeswitch.trunk/libs/sofia-sip' make[8]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip' make[7]: *** [all-recursive] Error 1 make[7]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip' make[6]: *** [all] Error 2 make[6]: Leaving directory `/usr/src/freeswitch.trunk/libs/sofia-sip' make[5]: *** [/usr/src/freeswitch.trunk/libs/sofia-sip/libsofia-sip-ua/libsofia-sip-ua.la] Error 2 make[5]: Leaving directory `/usr/src/freeswitch.trunk/src/mod/endpoints/mod_sofia' make[4]: *** [mod_sofia-all] Error 1 make[4]: Leaving directory `/usr/src/freeswitch.trunk/src/mod' make[3]: *** [all-recursive] Error 1 make[3]: Leaving directory `/usr/src/freeswitch.trunk/src' Making all in build make[3]: Entering directory `/usr/src/freeswitch.trunk/build' +-------- FreeSWITCH Build Complete -----------+ + FreeSWITCH has been successfully built. + + Install by running: + + + + make install + +----------------------------------------------+ make[3]: Leaving directory `/usr/src/freeswitch.trunk/build' make[2]: *** [all-recursive] Error 1 make[2]: Leaving directory `/usr/src/freeswitch.trunk' make[1]: *** [all] Error 2 make[1]: Leaving directory `/usr/src/freeswitch.trunk' make: *** [freeswitch] Error 2 [root@conf-2 src]# -------------------------------------------------------------- I'm still very much a noob when it comes to building, but I'd welcome any constructive suggestions on how to resolve this. Cheers Arne

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SIP Provider with unlimited channels
by Shameem Shiek 09 Nov '09

09 Nov '09

Dear Freeswitch Users, I am looking for a SIP Provider who can provide a DID with unlimited channels. Currently I am using junction networks but they have a high 2.9c/minute charge. I am looking for someone who has a flat rate for X minutes. Any advise would be much appreciated. Thanks.

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Remote-Party-ID issue and call pickup information
by Yehavi Bourvine 09 Nov '09

09 Nov '09

Hello, While trying to display the *called party *name on SNOM phones I've found that the field sent to the phone needs to be changed slightly in order to make SNOM work: Insetad of sending P-Assterted-Identity SNOM expects Remote-Party-ID. I changed it in mod_sofia and now SNOM, Polycom and Cisco work ok. Just wanted to let the developers know... And now a question: We have SNOM phones monitoring other extensions (BLF feature). When a call comes in, the monitoring phones get notification, but the name field (identity display) contains the calling extension number and not its display name. Can this be fixed? Thanks! __Yehavi:

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Dialpan: try.. finally analogs
by Artem Shiyanov 09 Nov '09

09 Nov '09

Hello! I have to deal with classic problem: "Leaking stream handle" in FS console. I also know the reason - firstly channel is sent to the extension with "playback" and later it is transfered to another extensions with "execute_extension" or, another trouble-case - channel is bridged to some addres. I do not ask (but I wish to) why FS doesn't close stream automatically when channel is gone. I ask whether it is possible to use some "try.. finally" construction in diaplan? If "yes" then I can simply stop playback in the "finally" block.. Any thoughs?

3 3

Extension: No audio
by Mark Campbell-Smith 09 Nov '09

09 Nov '09

Hi! I have FS natted and am connecting with an 'external' extension that is registered to FS. ie the extension 2000 is registered on the internet with a public IP through my router to FS (192.168.1.120 IP address). uPnP works and I see that the extension is registered successfully. The problem is that I do not get any audio When looking at the SIP trace, I see the INVITE but do not see a TRYING or RINGING message. The extension is actually ringing. I modified the RTP port range on the remote end to match the RTP ports of freeswitch. I have put a sip trace in the pastebin at http://pastebin.freeswitch.org/11035 If anyone has an idea what needs to be set to get audio, help appreciated. Thanks!

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