December 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Feedback
by Chad Vogel 23 Dec '11

23 Dec '11

Hello, I wanted to reach out and get some feedback from everyone because my company started a project using FS as base several weeks ago, our goal was to optimize and integrate FS more directly into Windows environments. Here is my question, we would like to share our changes and where should we host our project? Because we understand our build breaks compatibility with the UNIX environments and fails to meet the objectives cross platform compatibility, however it could contribute greatly and be beneficial to other windows users. Here are some of the changes we have made: * We converted FS core to C++ and having it compiling to .NET 4.0 CLI * FS now runs inside its own namespace * We replace FS_CLI with a PowerShell shell app * Supports SQL Server support via the Native SQL Server Client API * Support for windows clustering (up to 32 node active/active cluster) * Ties more directly into the Win32 API and has less reliance on open source Libs * Replaces OpenSSL sockets with Windows encrypted sockets * Added windows performance monitoring * No longer need to use mod_managed for managed modules * Memory management relies on .Net garbage collection * Added support for Microsoft Speech * VoiceXML 3.0 and SCXML support * No longer supports JS, LUA, PHP development - Modules can be only developed in .NET, C or C++( we feel support for other languages can be added back in but falls outside the scope of our objectives at this time, support for SCXML should make limit the need for other scripting languages) * Fixed RTP clock timing issues in virtualized environments * New configuration file format using .Net App Configuration files (kind of looks like an IIS config file)

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faxing and file permissions
by Tim Titov 23 Dec '11

23 Dec '11

When faxing with freeswitch the directory and containing file need to be readable by "other". Otherwise, mod_spandsp complains about the file being inaccessible. You can also see on mod_spandsp wiki people doing chmod o+r $TMPFAX. Why can't this be run under the freeswitch user only, so that you can do basic access control?

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Some questions regarding IPs and gateway
by georg＠riseup.net 22 Dec '11

22 Dec '11

Hello all, I'm a asterisk user, and would like to try FS now. I studied already the documentation. However, I still got some problems and questions, any help would be greatly appreciated. 1.) I've got a server with a IP reachable from the open, evil internet. I would like to disable this IP for FreeSWITCH, so that it's not possible to contact FreeSWITCH from there. Do I have to change the routing table of the operating system for this, or is it possible to exclude this ip in FS? 2.) I've got SIP trunk trough a german provider without registration. Authentication is based on IPs in a MPLS-net. So I set my IP in this net as the "external ip" in vars.xml After this, I setup the SIP trunk as a gateway sip-profiles/external.xml using the IP of the voice-switch of my provider as the proxy value and setting register to false. I think I'm still missing something, because sofia_gateway_data on the CLI is telling me "-ERR Parameter missing". I searched for this problem already, but found no solution. 3.) I would like to disable NAT completely, because I think I don't need this (and never used this in Asterisk). Actually there is no NAT involved in my setup. I achieved this using -nonat at startup. But I'm wondering, if there isn't a parameter to disable it by default. 4.) I've got another private (192.168.x.x) net connected to my server. All my phones are in there. So it set the internal ip in sip-profiles/internal to this. Is this correct? Thanks and Greetings from Berlin, Georg

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Odp: transfer to conference from conference / caller controls
by Rodney 22 Dec '11

22 Dec '11

Mariusz, using your suggestion, i worked out a solution that works. although i have to drop the bleg with sched_hangup in order for it to work with my hangup back to ivr method. it still looks like crap in my cdr but I can deal with it for now since i am crazy for wanting to transfer from static conference right to dynamic :) but doing it with this loopback method seems to give me the caller controls back. thanks for your help. ~rodney condition destination_number ^760$ 1 action answer 3 action set conference_user_list=| 10 action play_and_get_digits 4 4 3 5000 # ask4digit.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+ 20 action phrase spell,${target_num} 30 action sched_hangup +10 normal_clearing bleg 35 action bridge loopback/app=conference:${target_num}-127.0.0.1@dynamic_conf 4. Odp: transfer to conference from conference / caller controls (Mariusz Czulada) > Perhaps this will be useful for you: Dialplan: <extension name="bind_and_join"> ... <action application="bind_meta_app" data="4 a s execute_extension::sub_conf"/> ... <action application="conference" data="mainconf@cx"/> ... </extension> <extension name="sub_conf"> ... <action application="bridge" data="loopback/app=conference:${conference_name}_sub@cx_sub"/> ... </extension> conference.conf.xml: <group name="cx_controls"> ... <control action="hangup" digits="9"/> </group> <group name="cx_sub_controls"> ... <control action="hangup" digits="4"/> </group> <profile name="cx"> ... <param name="caller-controls" value="cx_controls"/> </profile> <profile name="cx_sub"> ... <param name="caller-controls" value="cx_sub_controls"/> </profile> Dnia 20-12-2011 o godz. 18:21 Rodney napisał(a): I am trying to transfer to a dynamic conference from within a static conference but when I do my callers loose control and can not use any dtmf caller control options and must hang up to get back. I tried using clear digit action but do not think this works on the "conf" realm. the dynamic conference works fine with all controls if i send them directly from the ivr but for what i am trying it is important to do this transfer from another conference. is there a method transfer and release all previous controls? i even tried using a difference conference profile so i transferred from static 501@default to <digits>@dynamic to see if giving new controls would work, to no avail. is there a log or something i can post that will help you in understanding how this is happening? or is it even fixable? thanks rodney > > > <group name="default"> > <control action="mute" digits="1"/> > <control action="hangup" digits="0"/> > <control action="execute_application" digits="4" data="playback C:/Program Files/FreeSWITCH/recordings/conf_help.wav"/> > <control action="execute_application" digits="8" data="execute_extension ANNOUNCE_CONF_COUNT_PRIVATE XML default"/> > <control action="execute_application" digits="#" data="execute_extension 759 xml default"/> > >- <group name="dynamic"> > <control action="mute" digits="1"/> > <control action="execute_application" digits="0" data="execute_extension 401 XML default"/> (main ivr menu extension) > <control action="execute_application" digits="8" data="execute_extension ANNOUNCE_CONF_COUNT_PRIVATE XML default"/> > > > > > >condition destination_number ^759$ 1 > action answer 2 > action set conference_user_list=| 11 > action play_and_get_digits 4 4 3 5000 # askprivateroom.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+ 15 > action phrase spell,${target_num} 16 > action conference ${target_num}-127.0.0.1@default > _______________________________________________ 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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transfer conference to conference
by Rodney 22 Dec '11

22 Dec '11

actually i just tried this and even though it looked good on screen, i was still just in the static conference i tried to transfer from, so i am back to square one :( condition destination_number ^760$ 1 action answer 3 action set conference_user_list=| 10 action play_and_get_digits 4 4 3 5000 # askprivateroom2.wav ivr/ivr-that_was_an_invalid_entry.wav target_num \d+ 20 action phrase spell,${target_num} 30 action sched_hangup +10 normal_clearing bleg 35 action bridge loopback/app=conference:${target_num}-127.0.0.1@dynamic_conf

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Problems in using mod_xml_curl at user authentication.
by rakib 0000 22 Dec '11

22 Dec '11

Hello list, I'm a newbie FreeSwitch user, trying to learn how mod_xml_curl works. I've done the basics things mentioned in FreeSwitch wiki pages about using mod_xml_curl. Now my plan to use mod_xml_curl for user registration. So, what I've done is - I wrote a server end application which reads POST request from mod_xml_curl and tries to find whether it's a valid post request (REGISTER) or not. If the request is a REGISTER request - then the application extracts the user id, domain. Then, it queries the database and extracts password and creates xml response. One of my sample response are like below (the following response has been taken from /tmp): <?xml version="1.0" encoding="UTF-8" standalone="no"?> <document type="freeswitch/xml"> <section name="directory"> <domain name="192.168.0.108"> <groups> <group name="default"> <users> <user id="1100"> <params> <param name="password" value="sakib"/> </params> <variables> <variable name="user_context" value="default"/> </variables> </user> </users> </group> </groups> </domain> </section> </document> With above response, I didn't had any luck. It fails to register, with following response: 2011-12-18 14:51:29.253474 [WARNING] sofia_reg.c:2446 Can't find user [1100(a)192.168.0.108] You must define a domain called '192.168.0.108' in your directory and add a user with the id="1100" attribute and you must configure your device to use the proper domain in it's authentication credentials. To figure it out, I've tried in different ways too. Like, I've used the above response patterns to create a user under conf/directory/default/ directory, but didn't work. Then I tried the following xml response pattern : <include> <user id="1003"> <params> <param name="password" value="$${default_password}"/> </params> <variables> <variable name="user_context" value="default"/> </variables> </user> </include> with the above xml I manage to create a user and authenticate perfectly. But, I've modified my server end application to give a that kind of response, but didn't work though. I've copied the server response (from the content dumped under /tmp) to create an user under conf/directory/default/ directory - but it works. So, my question is what am I missing here? FreeSwitch tries to get the response from the server, it gets the response too, but somehow not interpreting the response (or something else?) and gives "can't find user" error. Any hint's , suggestion will be very helpful. (note, I'm not using any webserver here.) Thanks in advance, Rakib

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Re: [Freeswitch-users] FreeSWITCH-users Digest, Vol 66, Issue 140
by Arnie Goodstein 22 Dec '11

22 Dec '11

Thanks Sent from my Android phone using TouchDown (www.nitrodesk.com) -----Original Message----- From: freeswitch-users-request(a)lists.freeswitch.org [freeswitch-users-request(a)lists.freeswitch.org] Received: Thursday, 22 Dec 2011, 3:41pm To: freeswitch-users(a)lists.freeswitch.org [freeswitch-users(a)lists.freeswitch.org] Subject: FreeSWITCH-users Digest, Vol 66, Issue 140 Send FreeSWITCH-users mailing list submissions to freeswitch-users(a)lists.freeswitch.org To subscribe or unsubscribe via the World Wide Web, visit http://lists.freeswitch.org/mailman/listinfo/freeswitch-users or, via email, send a message with subject or body 'help' to freeswitch-users-request(a)lists.freeswitch.org You can reach the person managing the list at freeswitch-users-owner(a)lists.freeswitch.org When replying, please edit your Subject line so it is more specific than "Re: Contents of FreeSWITCH-users digest..." Today's Topics: 1. Re: Number of calls (Michael Collins) 2. Re: Freeswitch Crashing again and again (Michael Collins) 3. Odp: transfer to conference from conference / caller controls (Rodney) ---------------------------------------------------------------------- Message: 1 Date: Thu, 22 Dec 2011 12:22:55 -0800 From: Michael Collins <msc(a)freeswitch.org> Subject: Re: [Freeswitch-users] Number of calls To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Message-ID: <CAKzWOxX8+962TC7L4YQWruj5ki=wV_4zRczQ+KH9JY+WuPzFKA(a)mail.gmail.com> Content-Type: text/plain; charset="iso-8859-1" That looks like a decent 4-core processor. You can probably do several hundred concurrent calls w/o much issue. Of course, "it depends" on what other stuff you have going on inside this server. :) -MC On Wed, Dec 21, 2011 at 11:44 PM, Voip service <voipservice911(a)gmail.com>wrote: > Hi, > > I am new in voip, how many calls can one freeswitch box handle with 30 % > of trans-coded calls and system configuration as > 8GB RAM > X3430 Xeon Processor, 2.4GHz, 8M Cache, Turbo > > > _________________________________________________________________________ > Professional FreeSWITCH Consulting Services: > consulting(a)freeswitch.org > http://www.freeswitchsolutions.com > > FreeSWITCH-powered IP PBX: The CudaTel Communication Server > http://www.cudatel.com > > Official FreeSWITCH Sites > http://www.freeswitch.org > http://wiki.freeswitch.org > http://www.cluecon.com > > 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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Freeswitch Crashing again and again
by Faisal Rehman 22 Dec '11

22 Dec '11

Dear FS users Our freeswitch has been working stable for around 1.5 years. Currently we are getting Freeswitch crashing without any apperent reasons with following coredump Core was generated by `/usr/local/freeswitch/bin/freeswitch -nc'.Program terminated with signal 11, Segmentation fault.#0 0x0000003d922790d0 in strchr () from /lib64/libc.so.6(gdb) bt#0 0x0000003d922790d0 in strchr () from /lib64/libc.so.6#1 0x00002b7677883008 in switch_ivr_originate (session=0x2aaad020cab8, bleg=0x54091ce0, cause=0x54091cec, bridgeto=0x2b7677950557 "true", timelimit_sec=60, table=0x0, cid_name_override=0x0, cid_num_override=0x0, caller_profile_override=0x0, ovars=0x0, flags=<value optimized out>, cancel_cause=0x0) at ./src/include/switch_utils.h:429#2 0x00002aaac8cf68f8 in audio_bridge_function (session=0x2aaad020cab8, data=0x2aaaac754480 "sofia/x8888/47756508496197@10.47.78.170:5654") at /usr/src/freeswitch/src/mod/applications/mod_dptools/mod_dptools.c:2683#3 0x00002b7677853952 in switch_core_session_exec (session=0x2aaad020cab8, application_interface=0x2aaab00553b8, arg=0x2aaad0006b28 "sofia/${flag_auth}") at src/switch_core_session.c:2121#4 0x00002b7677853f7f in switch_core_session_execute_application_get_flags (session=0x0, app=0x2aaad0006b20 "bridge", arg=0x2aaad0006b28 "sofia/${flag_auth}", flags=0x0) at src/switch_core_session.c:2006#5 0x00002b7677856963 in switch_core_session_run (session=0x2aaad020cab8) at src/switch_core_state_machine.c:181#6 0x00002b7677850d30 in switch_core_session_thread (thread=<value optimized out>, obj=0x2aaad020cab8) at src/switch_core_session.c:1311#7 0x0000003d92a0673d in start_thread () from /lib64/libpthread.so.0#8 0x0000003d922d3d1d in clone () from /lib64/libc.so.6 Please help to fix this issue as it is very urgent. Thanks and Regards, Faisal

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how to know which gateway used after continue_on_fail=true
by Muhammad Naseer Bhatti 22 Dec '11

22 Dec '11

Hello, I am using LUA to generate a DP and set channel variables for the bridge. All the variables are set, and LUA exits and the bridge is made outside the LUA script and the call is connected. I am using mod_xml_cdr for CDRs. If I set 1 gateway and bridge the call, I can see that gateway used in the CDR. Problem comes in when I use continue_on_fail=true, and have multiple gateways in bridge. The call fails from the first gateway, and is successfully connected via the second gateway. But I am not able to see the real gateway used in the CDR. I tried enabling log-b-leg in mod_xml_cdr to see what it shows, but I just can't see the gateway which was actually used to bridge the call. In the last_asg, I see both of the gateways passed. How can I know which gateway was used for the bridge? And no, I am NOT using mod_lcr. -Muhammad

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Need help with PortAudio
by Rob Hutton 22 Dec '11

22 Dec '11

I am trying to get portaudio working, and have run into a couple of problems 1) If portaudio cannot access the device, it causes freeswitch to segfault 2) I have set up a dialplan following the intercom example which seems to be working, but I get no audio. I have tried setting every device shown in devlist as the output device. 3) When I issue a "pa play". the cli hangs, even if I give it a timeout. Any thoughts?

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