September 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

lua startup scripts (possibly!)
by Pete Kelly 02 Sep '10

02 Sep '10

Hi I am new to freeswitch, so please excuse me if I have some terminology slightly wrong! I am writing an IVR application using mod_lua, to do simple auth then enter a menu tree etc. I need to "preprocess" the incoming SIP INVITE by running a series of regular expressions (which are stored in a mysql database) against the request URI (and/or From header), to add an extra header to the INVITE for internal use. I only want to read in the regular expressions from the database at "startup" (or by a controlled reload), as the query to get them is quite heavy and the IVR will be handling upwards of 100 concurrent calls - so to query for the regex list on each INVITE is impractical. Is this startup scenario possible? Two options that I think I might have are: 1) Have a lua startup script which reads the regex list in at startup, and listens for incoming call events. I would then apply the header within the lua script before the dialplan extension is looked up. Does anyone have any examples, or know of any documentation which shows something similar to this happening. I have searched the wiki high and low but can only find very basic documentation on event handling in lua, and startup scripts in lua. 2) Have the regular expressions stored somewhere within the dialplan (global variables?) and make it part of a dialplan extension to add this header. Any advice or tips anybody could give me will be very helpful. Thanks Pete

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Calls Disconnecting under 30 seconds
by Khurram Shafi 02 Sep '10

02 Sep '10

Hello List, After running FS for 5-6 hours we get calls disconnecting under 30 seconds. We are using enable-100rel=true After talking in irc with Matthew we disabled 100rel but now getting calls drop exactly after 32 seconds due to no response of ACK packet from FS. We are using 1 profile (external) for incoming and outgoing calls. Settings of the profile can be found here : http://pastebin.freeswitch.org/13783 we have all the logs and traces of the problem. Attached is the trace file of the problematic call. There are hundred's of calls like this. We have now again enabled 100rel. Regards

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Re: [Freeswitch-users] Problems getting dtmf events - code and debug now attached
by Nigel Kent 02 Sep '10

02 Sep '10

It has not made a difference when I used the playback_terminators . My problem happens when the code gets to the while loop, this is past the playback command. As can been seen from the debug log the code does go into the while loop and waits for events it catches all events apart from dtmf. It seems the detection of dtmf events is not happening after the playback command [DEBUG] src/esl.c:1042 esl_recv_event() RECV MESSAGE Event-Name: SOCKET_DATA Content-Length: 5015 Content-Type: text/event-plain got event test event wait for event The full test script: require 'rubygems' require "ESL" require 'socket' include Socket::Constants #Log ESL::eslSetLogLevel(7) @ring_ring = "tone_stream://%(400,200,400,450);%(400,2200,400,450);loops=2" def pick_up_call @con.execute("answer") @con.setEventLock("true") @con.execute("sleep", "1000") @con.execute("playback", @ring_ring) end bind_address = "127.0.0.1" bind_port = 8084 socket = Socket.new(AF_INET, SOCK_STREAM, 0) sockaddr = Socket.sockaddr_in(bind_port, bind_address) socket.bind(sockaddr) socket.listen(5) puts "Listening for connections on #{bind_address}:#{bind_port}" loop do client_socket, client_sockaddr = socket.accept pid = fork do @con = ESL::ESLconnection.new(client_socket.fileno) info = @con.getInfo @uuid = info.getHeader("UNIQUE-ID") @con.sendRecv("myevents") @con.sendRecv("divert_events on") pick_up_call while @con.connected puts "wait for event" e = @con.recvEvent puts "got event" if (!e.nil?) puts "test event" name = e.getHeader("Event-Name") break if name == "SERVER_DISCONNECTED" if name == "DTMF" digit = e.getHeader("DTMF-DIGIT") duration = e.getHeader("DTMF-DURATION") puts "*** dtmf #{digit} #{duration}" end end end puts "Connection closed" end Process.detach(pid) end --- On Thu, 2/9/10, Michael Collins <msc(a)freeswitch.org> wrote: From: Michael Collins <msc(a)freeswitch.org> Subject: Re: [Freeswitch-users] Problems getting dtmf events - code and debug now attached To: "FreeSWITCH Users Help" <freeswitch-users(a)lists.freeswitch.org> Date: Thursday, 2 September, 2010, 18:54 Did you do this? http://wiki.freeswitch.org/wiki/Channel_Variables#playback_terminators -MC On Thu, Sep 2, 2010 at 6:26 AM, Nigel Kent <ktngl(a)yahoo.co.uk> wrote: code below and debug log now attached. It seems once playback file is played no further dtmf events come through? Any ideas! while @con.connected puts "wait for event" e = @con.recvEvent puts "got event" if (!e.nil?) puts "test event" name = e.getHeader("Event-Name") break if name == "SERVER_DISCONNECTED" if name == "DTMF" digit = e.getHeader("DTMF-DIGIT") duration = e.getHeader("DTMF-DURATION") puts "*** dtmf #{digit} #{duration}" end end end --- On Wed, 1/9/10, Nigel Kent <ktngl(a)yahoo.co.uk> wrote: From: Nigel Kent <ktngl(a)yahoo.co.uk> Subject: [Freeswitch-users] Problems getting dtmf events To: freeswitch-users(a)lists.freeswitch.org Date: Wednesday, 1 September, 2010, 21:15 I am trying to catch dtmf events but am having problems. It seems no dtmfs are being recieved. I can filter to other events and it gets picked up so this section of the code is doing the indented action of getting events. However I notice that during the execution of the playback command while the sound file is playing if a key press is done at this time the dtmf does get received. So my thought are what conditions does the playback command set in order to be able to have dtmf get recognized. -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Re: [Freeswitch-users] VoiceMail to Email troubles
by Peter Hinman 02 Sep '10

02 Sep '10

D'oh! Sorry about that. I did try the nullmailer. I've had a hard time finding details on getting it setup correctly, so I don't know if FS actually tried to use it and nullmailer failed, or if FS didn't even make the call. Given that the php/python/lua scripts don't appear to be called at all, I'm inclined to think that nullmailer never had a chance either. Any other ideas? On 9/1/2010 10:56 AM, Brian West wrote: > Please DO NOT click reply, change the subject and hijack a thread. You have without knowing hijacked the "Early Media Troubles" thread. Click New message and input the address of the list. If you notice the list archives will also show that your thread is in the middle of another thread that is already in progress. Doing this increases the chances of your email being ignored or lost due to how some clients thread lists. > > Have you tried this: > > http://wiki.freeswitch.org/wiki/Mod_voicemail#nullmailer_settings > > > I know it works I use it daily... but I use exim 4 and the perl script in scripts/perl/sendmail > > /b > > > On Sep 1, 2010, at 11:48 AM, Peter Hinman wrote: > >> I'm struggling with getting VoiceMail messages sent out via email. >> I've tried the python and PHP scripts but it doesn't look like either of >> them actually get run when a voice mail is left. I've also created a >> lua script for testing purposes to show that the script is actually >> being called, and it doesn't look like it's getting called either. >> >> Currently in switch.conf.xml I have the following: >> >> <param name="mailer-app" value="/usr/local/bin/lua51/lua"/> >> <param name="mailer-app-args" >> value="/usr/local/freeswitch/scripts/test.lua"/> >> >> >> My directory.xml contains the following: >> <user id="1001"> >> <params> >> <param name="vm-email-all-messages" value="true"/> >> <param name="vm-attach-file" value="true"/> >> <param name="vm-keep-local-after-email" value="false"/> >> <param name="vm-mailto" value="Peter.Hinman(a)ParcelPool.com"/> >> </params> >> </user> >> >> The session log for the call shows the following: >> 2010-09-01 16:28:55.588613 [DEBUG] mod_voicemail.c:2403 Deliver VM to >> 1001(a)ParcelPool.com >> 2010-09-01 16:28:55.607187 [DEBUG] switch_utils.c:631 Emailed file >> [/tmp/mail.12833585354d4c] to [Peter.Hinman(a)ParcelPool.com] >> 2010-09-01 16:28:55.607187 [DEBUG] mod_voicemail.c:2571 Sending message >> to Peter.Hinman(a)ParcelPool.com >> 2010-09-01 16:28:55.609671 [DEBUG] switch_cpp.cpp:602 CoreSession::hangup >> >> >> If I call the lua script from the command line it runs correctly, but it >> doesn't appear to get called by FS when I leave a voice mail msg. >> >> Any ideas what I'm missing here? >> >> Thanks, >> Peter > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org -- Peter Hinman ParcelPool.com Office: 801-434-7400 Cell: 801-830-8292

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Re: [Freeswitch-users] Problems getting dtmf events - code and debug now attached
by Nigel Kent 02 Sep '10

02 Sep '10

code below and debug log now attached. It seems once playback file is played no further dtmf events come through? Any ideas! while @con.connected puts "wait for event" e = @con.recvEvent puts "got event" if (!e.nil?) puts "test event" name = e.getHeader("Event-Name") break if name == "SERVER_DISCONNECTED" if name == "DTMF" digit = e.getHeader("DTMF-DIGIT") duration = e.getHeader("DTMF-DURATION") puts "*** dtmf #{digit} #{duration}" end end end --- On Wed, 1/9/10, Nigel Kent <ktngl(a)yahoo.co.uk> wrote: From: Nigel Kent <ktngl(a)yahoo.co.uk> Subject: [Freeswitch-users] Problems getting dtmf events To: freeswitch-users(a)lists.freeswitch.org Date: Wednesday, 1 September, 2010, 21:15 I am trying to catch dtmf events but am having problems. It seems no dtmfs are being recieved. I can filter to other events and it gets picked up so this section of the code is doing the indented action of getting events. However I notice that during the execution of the playback command while the sound file is playing if a key press is done at this time the dtmf does get received. So my thought are what conditions does the playback command set in order to be able to have dtmf get recognized. -----Inline Attachment Follows----- _______________________________________________ FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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setting switch_r_sdp with outbound event socket
by Christian Löschenkohl 02 Sep '10

02 Sep '10

hello i try to moditfy the sip sdp to get some user agent working, but setting switch_r_sdp doesn't work the way i try it on the loopback interface i do see the command of setting the variable like this sendmsg call-command: execute execute-app-name: set execute-app-arg: <![CDATA[switch_r_sdp=v=0 o=root 713636917 713636917 IN IP4 x.x.x.x s=call c=IN IP4 x.x.x.x t=0 0 m=audio 17300 RTP/AVP 0 9 99 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:9 g722/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=fmtp:18 annexb=no a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=ptime:20 ]]> (btw this example sdp is ok) but on the console i get the error [ERR] switch_ivr.c:469 Invalid Command! and the b-leg doesn't get the modified sdp. what could i get wrong here? br -- Ing. Christian Löschenkohl Technische Leitung, Forschung & Entwicklung VoIP xpirio Telekommunikation & Service GmbH Lakeside B04 9020 Klagenfurt Austria T +43 (0) 5 77 11 - 1000 F +43 (0) 5 77 11 - 1002 E christian.loeschenkohl(a)xpirio.com

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error loading module 'luasql.mysql'
by Adam Wilt 02 Sep '10

02 Sep '10

Hi, I followed the instructions in the Lua documentation for setting up luasql, but when I try to run my script I get: 2010-01-27 19:08:14.799250 [ERR] mod_lua.cpp:182 error loading module 'luasql.mysql' from file '/usr/local/lib/lua/5.1/luasql/mysql.so': /usr/local/lib/lua/5.1/luasql/mysql.so: cannot restore segment prot after reloc: Permission denied stack traceback: [C]: ? [C]: in function 'require' /usr/local/freeswitch/scripts/l.lua:2: in main chunk I'm running FreeSWITCH version 1.4 and luasql version 2.1.1. I changed the permissions for mysql.so and for my script to 777, so I'm not sure where the permission problem could be. I'd appreciate any suggestions. Thanks, Adam

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Re: [Freeswitch-users] Calls Disconnecting under 32 seconds
by Chris Chen 02 Sep '10

02 Sep '10

You have NAT issues, simple as that, try to fix your settings around NAT. On Thu, Sep 2, 2010 at 4:38 AM, Ahmed Ajmal <ahmed.ajmal(a)breezecom.ae>wrote: > Hello, > > > > After running FS for 5-6 hours we get calls disconnecting under 30 seconds. > > We are using enable-100rel=true > > After talking in irc with Matthew we disabled 100rel but now getting calls > drop exactly after 32 seconds due to no response of ACK packet from FS. > > > > We are using 1 profile (external) for incoming and outgoing calls. > > Settings of the profile can be found here : > > http://pastebin.freeswitch.org/13783 > > > > There are hundred's of calls like this. > > > > We have now again enabled 100rel. > > > > Regards > > > > _______________________________________________ > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > >

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Turn off Remote-Party-ID
by Fraser Redmond 02 Sep '10

02 Sep '10

I have a gateway that requires the Remote-Party-ID to be removed from the outbound/B leg's INVITE. I have this working on an older version of FreeSwitch (from about march probably: 1.0.head-svn-17188) by setting in the dialplan: sip_cid_type=none That variable seems to be being ignored now though. I've tried the gateway setting which also doesn't work: <param name="cid-type" value="none"/> However setting the external-profile setting *does *work: <param name="caller-id-type" value="none"/> I guess that's my work around for now, but it'd be nice to be able to set it per-gateway/call again instead. There was an email to the list about a week ago (Remote-Party-ID manipulation in SIP 200 OK) which looked like it had the same problem, but it didn't get any replies. Cheers, Fraser

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Calls Disconnecting under 32 seconds
by Ahmed Ajmal 02 Sep '10

02 Sep '10

Hello, After running FS for 5-6 hours we get calls disconnecting under 30 seconds. We are using enable-100rel=true After talking in irc with Matthew we disabled 100rel but now getting calls drop exactly after 32 seconds due to no response of ACK packet from FS. We are using 1 profile (external) for incoming and outgoing calls. Settings of the profile can be found here : http://pastebin.freeswitch.org/13783 There are hundred's of calls like this. We have now again enabled 100rel. Regards

1 0