November 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Multi-access to lua scripts
by Alex Lake 11 Nov '13

11 Nov '13

Got a dialplan which calls a lua script that does a bridge. That bridge may well end up calling some bit of dialplan that calls the same lua script. I've found that fs seems to complain that there is "No such file or directory" when the script is being called for the second time. Am I smoking something odd here, or is this WAD?

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ORIGINATOR_CANCEL doesn't stop the ringing at the called party.
by Vincent Xia 11 Nov '13

11 Nov '13

for a vanilla call process, the called party will hangup immediately after the caller cancels the call before it's answered, this is true when im using <action application="bridge" data="user/${destination_number}" /> in the dialplan xml. however, if the call is handled by a lua script with a pattern like sessionB = freeswitch.Session('user/xxxx'); if sessionB:ready() then freeswitch.bridge(session, sessionB); else cause = sessionB:hangupCause(); if (cause == "NO_USER_RESPONSE" or cause=="NO_ANSWER") then session:streamFile("user_no_answer.wav"); end session:hangup(); end the called party is still ringing after the caller cancels the call before it's answered，this lua call process pattern is from wiki so im pretty sure it's being widely used, but why im having this problem? could someone figure this out?

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About the hardware fax detection on Sangoma cads
by sparklezou 11 Nov '13

11 Nov '13

Hi All, When configure the card and freetdm via wancfg_fs command, there is a question. Would you like to enable hardware fax detection? I always choose YES. I don't know how to use it. Is there any predefined dialplan context,such as "fax",to handle the incoming the fax? Thanks! 2013-11-11 sparklezou

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blind transfer back to caller on failure
by Tiago Sampaio 11 Nov '13

11 Nov '13

Hi all. I'm wondering if is possible to return the call if no answer or user busy? i tryed this: <context name="INTERNAL"> <extension name="INTER_RAMAL"> <condition field="destination_number" expression="^(\d{4})$"> <action application="set" data="continue_on_fail=true" /> <action application="bridge" data="[originate_timeout=10]user/$1" /> </condition> </extension> <extension name="BLIND_TRANSFER"> <condition field="destination_number" expression="^blind_transfer$"> <action application="answer"/> <action application="read" data="3 4 'tone_stream://%(10000,0,350,440)' digits 5000 #"/> <action application="set" data="continue_on_fail=true" /> <action application="execute_extension" data="is_transfer XML INTERNAL" />  <action application="bridge" data="user/${caller_id_number}" /> <action application="hangup" /> </condition> </extension> <extension name="is_transfer"> <condition field="destination_number" expression="^is_transfer$"/> <condition field="${digits}" expression="^(\d+)$"> <action application="set" data="continue_on_fail=true" /> <action application="transfer" data="-bleg ${digits} XML INTERNAL"/> <anti-action application="eval" data="cancel transfer"/> </condition> </extension> </context> Example: User A call user B, user B answer and do blind transfer to user C. User B hangup when call blind transfer and C rings 30 seconds and then transfer again to B before hangup A... Now if i do blind transfer and no answer or user busy, freeswitch hangup all sides...

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Trying to determine why bind_digit_action does not work in a conference created by outbound socket
by Simon Ditner 11 Nov '13

11 Nov '13

I am trying to use outbound socket to set up a conference, unfortunately I can not get my bind_digit_action calls to work in-conference. My setup is pretty boiler-plate: autoload_configs/conference.conf.xml <configuration name="conference.conf" description="Audio Conference"> ... <profiles> <profile name="simonconf"> <param name="rate" value="8000"/> <param name="interval" value="20"/> <param name="energy-level" value="300"/> <param name="caller-controls" value="none"/> </profile> </profiles> </configuration> dialplan/default.xml <include> <context name="default> <extension name="simon1"> <condition field="destination_number" expression="^800$"> <action application="bind_digit_action" data="random_context,*3,exec:playback,ivr/ivr-welcome_to_freeswitch.wav" /> <action application="digit_action_set_realm" data="random_context"/> <action application="answer"/> <action application="conference" data="random_conference@simonconf "/> </condition> </extension> <extension name="simon2"> <condition field="destination_number" expression="^900$"> <action application="bind_digit_action" data="random_context,*3,exec:playback,ivr/ivr-welcome_to_freeswitch.wav" /> <action application="digit_action_set_realm" data="random_context"/> <action application="socket" data="127.0.0.1:8888 async full"/> </condition> </extension> </context> </include> Listening on port 8888, I have a server which issues: connect sendmsg call-command: execute execute-app-name: answer sendmsg call-command: execute execute-app-name: conference execute-app-arg: random_conference@simonconf event-lock: true Calling into simon1, the *3 will work. Calling in to simon2, *3 will not work. If I do not join the user into the conference, *3 works in both setups. Is there any particular reason that *3 would not work in the simon2 setup? Thanks, spd

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enabling tls cause sip_profile error
by adahary 10 Nov '13

10 Nov '13

I have enabled tls/ssl in vars.xml and restarted freeswitch. when reloading mod_sofia all profiles got into errors. I have read that openssl-devel should be install before compiling FS. I did that before building FS without adding anything special for the ssl on the ./configure command (like --enable-zrtp). I'm about to repeat the build procedure but before that I would like to ask if there is something elese to be done beside installing openssl-devel (done) and # ./configure. regards assaf -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/enabling-tls-cause-sip-profil… Sent from the freeswitch-users mailing list archive at Nabble.com.

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Build Problem.
by dotnetdub 10 Nov '13

10 Nov '13

Hey All I am following the instructions to install from the wiki. Running Squeeze 32 bit for testing. git clone git://git.freeswitch.org/freeswitch.git cd freeswitch/ ./bootstrap.sh ./configure this step fails because of this line in configure AM_INIT_AUTOMAKE(libfreeswitch,0.1) If I remove that it fails at: AM_PROG_CC_C_O AC_PROG_LIBTOOL I cloned the repo and did the exact same install two weeks ago on Debian Squeeze and it worked so I copied the source tree from that machine to this machine and its compiling now as I type this.. I think there may be an issue at the moment with the git repo? Brian

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FS with SSL/TLS issues!
by mehroz 10 Nov '13

10 Nov '13

Hi, it seems like i am unable to get it done. Here is what i have done 1. created CA (Root) certificates: ./gentls_cert setup -cn pbx.freeswitch.org -alt DNS:pbx.freeswitch.org -org freeswitch.org 2. created Server Certificate: ./gentls_cert create_server -cn pbx.freeswitch.org -alt DNS:pbx.freeswitch.org -org freeswitch.org *I replaced pbx.freeswitch.org and freeswitch.org with my FS IP address in which i am listening.* 3. Enabled /internal_ssl_enable=true/ and/ ip_tls_version=tlsv1/ in vars.xml . 4. When i try to Register 3CX on TLS over port 5061, all i see is : tport_tls.c:869 tls_connect() tls_connect(0x7fbd94e7a5e0): events NEGOTIATING tport_tls.c:869 tls_connect() tls_connect(0x7fbd94e7a5e0): events NEGOTIATING tport_tls.c:958 tls_connect() tls_connect(0x7fbd94e7a5e0): TLS setup failed (error:00000001:lib(0):func(0):reason(1)) tport.c:2084 tport_close() tport_close(0x7fbd94e7a5e0): tls/59.103.76.82:56863/sips tport.c:2730 tport_wakeup_pri() tport_wakeup_pri(0x7fbd94005090): events IN tport.c:869 tport_alloc_secondary() tport_alloc_secondary(0x7fbd94005090): new secondary tport 0x7fbd94f53c70 tport_type_tls.c:603 tport_tls_accept() tport_tls_accept(0x7fbd94f53c70): new connection from tls/59.103.76.82:56864/sips. PLease help!! -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/FS-with-SSL-TLS-issues-tp7587… Sent from the freeswitch-users mailing list archive at Nabble.com.

3 2

Diagnosing locked calls
by Alex Lake 10 Nov '13

10 Nov '13

Got a few calls that are seeming to last indefinitely. The last entries in the log are 632b92fd-3729-4e5e-af82-152192f86533 2013-10-21 16:44:04.553092 [DEBUG] switch_core_state_machine.c:92 sofia/internal/002347067124224(a)87.238.73.160 Standard REPORTING, cause: NORMAL_CLEARING 632b92fd-3729-4e5e-af82-152192f86533 2013-10-21 16:44:04.553092 [DEBUG] switch_core_state_machine.c:761 (sofia/internal/002347067124224(a)87.238.73.160) State REPORTING going to sleep 632b92fd-3729-4e5e-af82-152192f86533 2013-10-21 16:44:04.553092 [DEBUG] switch_core_state_machine.c:440 (sofia/internal/002347067124224(a)87.238.73.160) State Change CS_REPORTING -> CS_DESTROY 632b92fd-3729-4e5e-af82-152192f86533 2013-10-21 16:44:04.553092 [DEBUG] switch_core_session.c:1338 Send signal sofia/internal/002347067124224(a)87.238.73.160 [BREAK] 632b92fd-3729-4e5e-af82-152192f86533 2013-10-21 16:44:04.553092 [DEBUG] switch_core_session.c:1546 Session 4 (sofia/internal/002347067124224(a)87.238.73.160) Locked, Waiting on external entities I need a strategy to work out what it's waiting for!

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Taking audio samples from bug
by Anton 10 Nov '13

10 Nov '13

When I check the audio sample contained in switch_frame::data array, it always has the same value -1, regardless of the position in the array and sound level. In order to get these samples I use different applications containing the following line: frame = switch_core_media_bug_get_read_replace_frame(bug); and immediately after insert this code: switch_log_printf( SWITCH_CHANNEL_LOG, SWITCH_LOG_NOTICE, "data[0]=%d\n", ((int16_t *)(frame->data))[0] ); then been rebuilt with the appropriate module and initiating its launch. After that, expect to see on the console different sample values, but it is always the same -1. Someone explain to me why?

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