- FreeSWITCH-users - FreeSWITCH Mailing Lists

How to run linux shell command from within
by Dmitry Saratsky 01 Mar '11

01 Mar '11

Thanks Christopher, I already have found why this was not working for me (it was quotes wrong usage). However I already have finished with my script with LUA’s (not FS’s) function: os.execute(execute_command); that also runs shell commands, but dose it directly from within LUA, but not through FreeSWITCH’s API. Do you see any problems with that? Thanks, Dmitry.

2 1

Disconnect Call Timer
by Max Clark 01 Mar '11

01 Mar '11

Hello, I'd like to automatically disconnect a call after x minutes (say 4 hours). Is this something that can be set in the dialplan before the bridge? What is the best way to do this? Thanks, Max

2 1

disabling ptime warning message
by Malay Thakershi 01 Mar '11

01 Mar '11

Hello, I updated to the latest FS version last week. I started getting the following warning when speech / sound is played on the call. "2011-02-28 10:15:55.985930 [WARNING] sofia_glue.c:213 Codec LPC payload 7 added to sdp wanting ptime 90 but it's already 20 (G7221:115:20), disabling ptime." I read sections on codecs and negotiations. Following are the settings from vars.xml (I have not changed them): <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G7221@32000h ,G7221@16000h,G722,PCMU,PCMA,GSM"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,GSM"/> Also, there is no codec related setting in sip_profiles files and sofia.conf.xml file. I am playing audio files using Cepstral TTS during the call. Can someone please help me understand these settings? And if they are appropriate? Thank you. Malay

3 16

t38_gateway self - no re-Invite?
by Clemens Ebentheuer 01 Mar '11

01 Mar '11

Hi to all from austria, I´m very new to freeswitch and I´m trying to get t38_gateway self working - but no success. for testing I use zoiper or spa2102 as t38 "device" - i have troubles in outbound direction: t38 device(internal) -> FS:t38_gateway self -> sip-provider (external - no t38 support) freeswitch sends no re-invite, I guess and Zoiper BYEs with Reason: SIP;description="Stale re-Invite" also see: http://pastebin.freeswitch.org/15439 inbound (t38_gateway peer) works fine - also when I do t38 device -> FS:rxfax my simple dialplan for the outbound call: <extension name="scpa" > <condition field="destination_number" expression="^1(.*)$"> <action application="set" data="fax_enable_t38_request=true"/> <action application="set" data="fax_enable_t38=true"/> <action application="set" data="execute_on_answer=t38_gateway self"/> <action application="bridge" data="sofia/gateway/scpa567/$1"/> </condition> </extension> Maybe I have to mention that I´m using Fusionpbx?! Thanks for any advice! I allready tried different examples I found, but no luck... Clemens

5 15

FreeSwitch Installation Error
by George Lee 28 Feb '11

28 Feb '11

Hi, I changed the installation directory to /opt/freeswitch > ./configure --prefix=/opt/freeswitch Then I (as root) ran make; make install and it gave me this error: /bin/bash ./libtool --mode=install /usr/bin/install -c 'ftmod_zt.la' '/opt/freeswitch/mod/ftmod_zt.la' libtool: install: error: cannot install `ftmod_zt.la' to a directory not ending in /usr/local/freeswitch/mod /bin/bash ./libtool --mode=install /usr/bin/install -c 'ftmod_skel.la' '/opt/freeswitch/mod/ftmod_skel.la' libtool: install: error: cannot install `ftmod_skel.la' to a directory not ending in /usr/local/freeswitch/mod /bin/bash ./libtool --mode=install /usr/bin/install -c 'ftmod_analog.la' '/opt/freeswitch/mod/ftmod_analog.la' libtool: install: error: cannot install `ftmod_analog.la' to a directory not ending in /usr/local/freeswitch/mod /bin/bash ./libtool --mode=install /usr/bin/install -c 'ftmod_analog_em.la' '/opt/freeswitch/mod/ftmod_analog_em.la' libtool: install: error: cannot install `ftmod_analog_em.la' to a directory not ending in /usr/local/freeswitch/mod What did I setup wrong? Thanks, George

3 4

ESL socket outbound: fail to stop tone detect
by Stephen Wilde 28 Feb '11

28 Feb '11

In an originated outbound session that is in progressing phase, I do, via esl socket outbound, a "tone_detect". When I receive the event "DETECTED_TONE" I do the "stop_tone_detect" but sometimes the stop has no effect and I see in the log the warning: "switch_core_session.c:1955 Cannot execute app 'stop_tone_detect' media required on an outbound channel that does not have media established" But the media is present (the tone is correctly detected!). The event I receive on this channel are (in sequence) CHANNEL_UUID CHANNEL_OUTGOING CHANNEL_ORIGINATE CHANNEL_STATE - CS_INIT CHANNEL_STATE - CS_ROUTING CHANNEL_STATE - CS_CONSUME_MEDIA CHANNEL_PROGRESS CHANNEL_PROGRESS_MEDIA CHANNEL_STATE - CS_EXECUTE CHANNEL_EXECUTE CHANNEL_PARK then I do the "tone_detect" and I receive: CHANNEL_EXECUTE MEDIA_BUG_START CHANNEL_EXECUTE_COMPLETE DETECTED_TONE so I do the "stop_tone_detect" that fails in the log Please, can anyone address me to solve this issue?

2 2

FreeSwithc not handling no SDP INVITE properly for video call
by George Lee 28 Feb '11

28 Feb '11

Hi all, I am having trouble with FreeSwitch handling late codec negotiation for video calls. The call logs are here: http://pastebin.freeswitch.org/15481 I have this line: <param name="enable-3pcc" value="proxy"/> added to the external sip_profile for handling late codec negotiation. I also have <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA,H263"/> <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA,H263"/> in vars.xml for codec support list. The initial INVITE contains no SDP for the caller and once the callee answers the call, it sends the 200 OK with audio (pcma, pcmu) and video (H263) codec capabilities. FreeSwitch forwards the 200 OK to the caller without video codec capabilities. As a result, the video call does not establish properly. Could someone give me a pointer? Thanks, George

2 5

freetdm with a400p segfaults
by Jesse Cloutier 28 Feb '11

28 Feb '11

Im having trouble with getting freetdm to work with a openvox a400p with 4 fxo modules using the latest freeswitch from GIT dahdi installs fine and 'dahdi_hardware' lists it as a TDM400P When I configure freetdm.conf only I can start freeswitch and do a ftdm dump and it displays my span with 4 channels When I configure freetdm.conf.xml freeswitch segfaults on startup I don't know if I need them but ftmod_analog.so and ftmod_zt.so have been built this is my freetdm.conf: [general] [span zt myFX0] fxo-channel => 1 fxo-channel => 2 fxo-channel => 3 fxo-channel => 4 This is my freetdm.conf.xml: <configuration name="freetdm.conf" description="FreeTDM Configuration"> <settings> <param name="debug" value="6"/>     </settings>  <analog_spans>  <span name="myFXO">  <param name="dialplan" value="XML"/> <param name="context" value="Incoming-FXO"/> <param name="tonegroup" value="us"/> <param name="enable_callerid" value="true"/> <param name="enable-analog-option" value="call-swap"/>  </span> </analog_spans> </configuration> Any idea what I am doing wrong? -- Jesse Cloutier Network Administrator Cronomagic Canada 5143411579 x210 jesse(a)cronomagic.com

2 1

mod_event_socket message pizza received from BSD machines
by Mauritz Løvgren 28 Feb '11

28 Feb '11

Hi, We are experiencing weird behaviour when using socket connection to mod_event_socket while running FS on FreeBSD and Mac OSX. We send a lot of bgapi and api commands to the FreeSwitch while controlling and monitoring hundreds of sessions. We have created a simple load-test that displays the problem we face. The load-test simply fires the following command continuosly to freeswitch: “bgapi sofia status”, generally as fast as freeswitch can receive it. The following occurs, but only on FreeBSD or MacOSX (10.6.6) (both 64-bit) while using latest version of FreeSwitch: - Replies and events are inter-mixed on the receive stream (the output from freeswitch), causing protocol errors in the receiving client. This does _not_ happen on CentOS or Windows, so we wonder what could be causing this. Is there a problem with threading or socket libs here? We assume that FreeSwitch protect a mod_event_socket inbound connection output stream by locking (or other means) to make sure not any two messages are written ‘simultaneously’ causing garbage in the receiving end? Regards, Mauritz Lovgren System Architect IPLink Inc. http://www.iplink.no

3 4

How to config sip profiles and add them into the freeswitch system
by Hareem Haque 28 Feb '11

28 Feb '11

I understand that sip profiles are managed via xml files.. i can make them.. but when i use sofia profile start command.. it does not pick up the .xml profile so how can i setup profiles and add them into the system. Your help is greatly appreciated Best Regards Hareem. Haque

2 1