May 2013 - FreeSWITCH-users - FreeSWITCH Mailing Lists

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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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.

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Is it necessary to call hangup....
by Moiz Chinoy 07 Nov '13

07 Nov '13

Hello, Is it necessary to call hangup in dialplan after say playing a file through playback application. -- Regards, Moiz Chinoy.

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enable-100rel question
by Mike Burlingame 24 Oct '13

24 Oct '13

I was looking to turn on 100rel however based on the below info from the wiki I am not sure I want to run the risk of it crashing FS is this information still accurate? >From http://wiki.freeswitch.org/wiki/Sofia_Configuration_Files enable-100rel This enable support for 100rel (100% reliability - PRACK message as defined in RFC3262) This fixes a problem with SIP where provisional messages like "180 Ringing" are not ACK'd and therefore could be dropped over a poor connection without retransmission. *2009-07-08:* Enabling this may cause FreeSWITCH to crash, see FSCORE-392. <param name="enable-100rel" value="true"/>

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Prevent A leg from hangup after bridge with inbound ESL socket
by Alex Massover 09 Oct '13

09 Oct '13

Hi, I have a very simple dialplan that just do park for incoming calls. All rest of leg management is done via ESL inbound socket. I'm trying to do the same behavior like in this dialplan example, but from ESL inbound socket: <action application="set" data="hangup_after_bridge=false"/> <action application="bridge" data="sofia/internal/1001@$${domain}"/> <action application="bridge" data="sofia/internal/1002@$${domain}"/> The problem is with bridge API, if B leg doesn't answer (e.g. 404, or busy), A leg disconnects. But I'm trying to prevent A leg from disconnecting in order to do bridge to other place. Looks like hangup_after_bridge=false, park_after_bridge=true, transfer_after_bridge etc don't have any effect when bridge done from inbound socket. A leg disconnects always. Is there any way to keep A leg after bridge with inbound socket? I'm aware of originate, but prefer to user bridge. -- Best Regards, Alex Massover

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execute_on_fax_success/failure
by Darcy Primrose 07 Oct '13

07 Oct '13

I am using excute_on_fax_success to call a script when a fax is received, the script converts the fax to pdf and emails it, Is there a way to pass the number of pages received, ie, a variable created by spandsp. Everything works, but I do not seem to be able to pass the variables. <condition field="destination_number" expression="^1(\d{10})$|(\d{10})$"> <action application="set" data="faxtempdirectory=/faxfiles/${caller_id_number}"/> <action application="system" data="mkdir ${faxtempdirectory}"/> <action application="set" data="faxDirection=outbound"/> <action application="set" data="emailAddress=$1$2@${fax2net_server}"/> <action application="set" data="faxFile=$1$2-${caller_id_number}-${strftime(%Y-%m-%d-%H-%M-%S)-Success}"/> <action application="set" data="execute_on_fax_success=system /./var/mail/sentfaxes/fax2mail.sh ${faxFile} $1$2@${fax2net_server} ${fax_security_code} ${fromRc} fax2net_orig=${fax_email}= ${fax_transfer_rate}"/> <action application="answer"/> <action application="playback" data="silence_stream://2000"/> <action application="set" data="fax_enable_t38_request=true"/> <action application="set" data="fax_enable_t38=true"/> <action application="rxfax" data="${faxFile}.tif"/> <action application="hangup"/> </condition> Thanks Darcy Primrose

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Error Creating SIP UA for profile: X
by Henrik Aagaard Sørensen 04 Oct '13

04 Oct '13

I've had a server running only FreeSWITCH for about 6 months now. I have had it turned off for some while. Now, when trying to start FreeSWTICH, I get the errors: Error Creating SIP UA for profile: X where X differs between internal, external and others. When starting my CentOS 6 server I run: service iptables stop service ip6tables stop just to make sure that they aren't blocking anything. I have nothing else running on CentOS. I've tried to log everything from console, warning etc. but there's nothing indicating what the problem is. Can someone help me out?

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Cordless phones
by Jonas Gauffin 19 Sep '13

19 Sep '13

Hi, I'm looking for IP-dects where a base station can handle 5 phones (which all should be able to talk at the same time). All suggestions are welcome. Regards, Jonas

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IM messaging on FreeSwitch
by mehroz 10 Sep '13

10 Sep '13

Hi, Anyone with an experience of SIP SIMPLE IM Messaging implemention? I dont see the stuff well documented on http://wiki.freeswitch.org/wiki/Mod_sms <http://wiki.freeswitch.org/wiki/Mod_sms> I followed simple steps to test it with reply back as in default chatplan. <?xml version="1.0" encoding="utf-8"?> <include> <context name="default"> <extension name="demo"> <condition field="to" expression="^(.*)$"> <action application="reply" data="Hello, you said: ${_body}"/> </condition> </extension> </context> </include> I tested it with eyeBeam, Jitsi, it doesnt work, and i do not see reply back. and i see in logs: 2013-03-07 14:58:49.090936 [INFO] mod_sms.c:334 Processing text message 9999->9999 in context default Chatplan: 9999 parsing [default->demo] continue=false Chatplan: 9999(a)198.84.61.52 Regex (PASS) [demo] to(9999(a)198.84.61.52) =~ /^(.*)$/ break=on-false Chatplan: 9999(a)198.84.61.52 Action reply(Hello, you said: ${_body}) -- View this message in context: http://freeswitch-users.2379917.n2.nabble.com/IM-messaging-on-FreeSwitch-tp… Sent from the freeswitch-users mailing list archive at Nabble.com.

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eavesdrop not sending video
by D D 08 Sep '13

08 Sep '13

Hi, I tried to use eavesdrop to capture the video of a parked call, but I only receive the audio. I have followed the directions described at http://wiki.freeswitch.org/wiki/Video-recording (including a rebuild after doing the git checkout of video-media-bug), and I can successfully use record_fsv to record the video to a file. Any ideas why the eavesdrop does not send the video? Thanks, David

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