December 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

mod_rtmp: "Read error" when attempting calls from FS to RTMP client.
by peely 03 Dec '11

03 Dec '11

Hi, I have a strange issue when I attempt to bridge a call from FS out to an RTMP client. As soon as the call is offered up to the client, the call is disconnected immediately i.e at the "ringing" stage. The strange thing is that as soon as I issue an "fsctl loglevel 9" then everything is fine. I don't know whether this is because of any additional load, as nothing above "ERR" is printed in each case. When the call is cleared immediately then I am getting "[ERR] rtmp.c:713 Read error" in the logs. mod_rtmp is compiled from today's git. Is there a possibility something is trying to happen too quickly in the handshake before data has been received from the client? Outbound calls from client to FS work flawlessly and incoming calls too, so long as I seem to have debug logging enabled. I'm not sure if it's relevant, but I'm bridging to the UUID of the session AKA "<action application="bridge" data="rtmp/cd421c76-1614-11e1-be92-397d873fd9b8"/>" Cheers, Neil. -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/mod-rtmp-Read-error-when-atte… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Recording with music (playing in background)
by Pankaj Upadhyay 03 Dec '11

03 Dec '11

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 .

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mod_callcenter ringback - help
by Jeferson Rodrigo Almeida 02 Dec '11

02 Dec '11

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

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Callcenter ringback - help
by Jeferson Rodrigo Almeida 02 Dec '11

02 Dec '11

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

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google voice connection going offline (mod_dingaling)
by Federico Beffa 02 Dec '11

02 Dec '11

Hi All, I've setup mod_dingaling to connect to my google voice account according to the instructions on the wiki http://wiki.freeswitch.org/wiki/Google_Voice#Setup_FreeSWITCH_-_Dingaling_t… with two modifications: <param name="rtp-ip" value="auto"/> <param name="ext-rtp-ip" value="auto-nat"/> I use the same IP settings for SIP (sofia) with no problem. The setup works fine for the first hour or two, but then I can see from another google account that the connection established by freeswitch goes offline. If I try anyway to make an outbound call, then the other end does not ring (even if I wait for a long time). However, after some minutes the freeswitch account returns online and works again for a couple of hours. I'm behind a nat and firewall. Is there any port which I have to open to keep freeswitch online? As far as I understand only the google XMMP server needs to listen to port 5222 and port 5269. Or, is there any kind of "keep alive" timer available? The google talk client running in the same environment works flawlessly. Thanks for any advise. Fede

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MOH in multi-tenant PBX
by DJB International 02 Dec '11

02 Dec '11

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

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is 100rel still a bad idea?
by Dave Horton 01 Dec '11

01 Dec '11

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?

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Build error: undefined reference to `Curl_setopt'
by Farhan Husain 01 Dec '11

01 Dec '11

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? Thanks, Mohammad

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How does one make dialplan changes current without restarting FS? L16 codec?
by Phil Quesinberry 01 Dec '11

01 Dec '11

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

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Mysql unixODBC problem
by Sanath Prasanna 01 Dec '11

01 Dec '11

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@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@test]# isql -v freeswitch_odbc root test +---------------------------------------+ | Connected! | | | | sql-statement | | help [tablename] | | quit | | | +---------------------------------------+ SQL> [root@test]# mysql -V mysql Ver 14.14 Distrib 5.5.18, for Linux (i686) using EditLine wrapper Br Sanath

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