July 2011 - FreeSWITCH-users - FreeSWITCH Mailing Lists

.Net interface
by Roger Castaldo 18 Jul '11

18 Jul '11

Hi everyone this is just to let everyone know that I have developed a C# based .Net socket library to allow a freeswitch server to be interfaced with using the event socket and outbound socket. The library is hosted on google code, the link is http://code.google.com/p/freeswitch-socket-dotnet/ If anyone is interested in helping with the development please let me know. Feel free to use it and expect updates to it as time permits, my goal is to try and make the library somewhat idiot proof by implementing all available calls that could be handled on the server side when controlling a phone call.

4 5

Transfers and billsecs
by Isaac Jurado 18 Jul '11

18 Jul '11

Hello all, First of all, I want to apologize for the length of this message. I wouldn't like to waste anyone's time. I'm new to this list and also fairly new to the VoIP world. I work for a company that provides some basic telephony services using FreeSWITCH. Recelty, we sketched up a plan to extend the features of our service to the customers; so we started experimenting first. Our setup is a basic HTTP server that controls FreeSWITCH through mod_xml_curl and processes call information afterwards, with the information provided by mod_xml_cdr. In general, each customer has its own context and we are trying to come along with a general billing system that tries to be as fair as possible. In order to reduce billing system complexity, we are completing channel information from the dialplan so we can inspect each CDR generated independently, and be able to discard it or process it on its own, without having to check other related CDRs. Obviously, this is becoming harder and harder after we started dealing with call transfers, either attended or blind. Consider the following scenario: 1. We have four customers: A, B, C and D. 2. A calls B, the complete billing information is in the a-leg (from A to FS) and it will be charged as a call to B. 3. After some time (say 20 seconds), B makes a blind transfer to C. So a new b-leg is created (from FS to C), the channel between FS and B is closed and a CDR is generated. Conceptually, this is a call from B to C which should be charged to B, not to A. 4. Again, after some time (say 30 seconds), C makes a blind transfer to D. Similar procedure: new channel from FS to D and the channel from FS to C terminates, generating a CDR. This time, we should charge a call from C to D. 5. After 15 seconds, A or D hang up and two more CDRs are generated. In the end, the billing system should charge: - The call between A and B, which had a duration of 65 seconds. - The call between B and C, which had a duration of 45 seconds. - The call between C and D, which had a duration of 15 seconds. And here is where the problems arise, because only the a-leg (from A to FS) contains the proper value of <billsec> for the "logical" call between A and B. The CDR of the FS to C channel has a <billsec> of 30 seconds and the one for the FS to D channel has 15 seconds (all aprox). However, all CDRs appear to have the same <call_uuid> so they can be easily related. The problem is that we would like to avoid this CDR dependency if we can, mainly because the background process that treats with CDR information is stateless at the moment. The purpose of this message is to survey about what other people is doing about this, as it seems to be the source of some headaches according to the list archives. I think that, in the end, we will have to mix up different CDRs but we may be missing some hidden functionality of FreeSWITCH. Thank you for your time :-) P.S: Testing the same call flow, but with attended transfers it is even more complex in terms of CDR processing, but we would like to have some idea on how to solve this one first. -- Isaac Jurado Internet Busines Solutions eConcept

1 0

Error While Installing free switch on UBuntu 11.04
by Ravindra Kondiparthi 18 Jul '11

18 Jul '11

Hi, I am getting the below error while installing free switch on UBuntu 11.04.Any help in resolving the error is appreciated. Compiling src/switch_core_sqldb.c ... cc1: warnings being treated as errors src/switch_core_sqldb.c: In function '_switch_cache_db_get_db_handle': src/switch_core_sqldb.c:314:47: error: comparison between 'switch_odbc_status_t' and 'enum <anonymous>' src/switch_core_sqldb.c: In function 'switch_cache_db_execute_sql_real': src/switch_core_sqldb.c:389:90: error: comparison between 'switch_status_t' and 'enum <anonymous>' make[1]: *** [libfreeswitch_la-switch_core_sqldb.lo] Error 1 make: *** [all] Error 2 thanks Ravi

2 1

FreeSwitch does not send HOLD Reinvite on other leg.
by Manjiri Deshpande 18 Jul '11

18 Jul '11

Scenario: A -> FS -> B A puts the calls on hold ( ie sending a reINVITE with media ip 0.0.0.0 ). FS does not send this re INVITE to leg B.Leg B continues to send RTP to FS and call disconnects sometime afterwards if call is not removed from hold. Disconnect times have been as short as 3 minutes and as long as 30 minutes. Test calls have included background noise from Leg B and seen noise in trace as normal RTP packets. Call does not seem to disconnect if this background noise is present. Once background noise disappears from Leg B ,FreeSwitch call will disconnect. Why Leg B sends RTP packets sometimes ,so that call is not getting disconnected before 30 mins. Also If RTP packets are not there,then also call should not be disconnected before 30 mins. Thanks, Manjiri

2 2

tdm support
by Michel Daggelinckx 17 Jul '11

17 Jul '11

Is there a module in the works to configure freetdm cards also with support for dahdi tools and wanpipeconfig. Michel

2 2

Music on hold spec file
by Joseph L. Casale 16 Jul '11

16 Jul '11

I whipped up a quick spec file to build these, there's an empty base package which pulls in the 8/16/32/48 Khz sub packages. How does one go abouts submitting this? Thanks, jlc

2 1

playing audio to the *bridged* legs
by Mi Ke 16 Jul '11

16 Jul '11

Hi All ! Is it possible to play notification audio (wav or tgml) to one of *bridged* legs (without conferencing them) with Lua or any other way? session:streamFile executed on the leg gives audio neither on A or B while log shows file was played OK ... Thanks / Mike

2 1

Same peace of code works with Javascript and not LUA
by Boris Kovalenko 16 Jul '11

16 Jul '11

Hello! I'm trying to migrate DISA from Javascript to LUA. This peace of code works fine in Javascript: ostr = "{ignore_early_media=true" + ",origination_caller_id_number=" + session.getVariable("caller_id_number") + "}sofia/ipbx/50004#" + digits + "@192.168.1.1:5060"; osession = new Session( ostr ); bridge(session, osession); osession.hangup(); With LUA not: originate_string = "{ignore_early_media=true" .. ",origination_caller_id_number=" .. session:getVariable("caller_id_number") .. "}" .. "sofia/ipbx/50004#" .. digits .. "@192.168.1.1:5060"; originate_session = freeswitch.Session( originate_string ); freeswitch.bridge(session, originate_session); originate_session:hangup(); The call is droped when remote answers. There is an error in log: bridge: session not ready. I tried to use if( originate_session:ready() ) then freeswitch.bridge(session, originate_session); originate_session:hangup(); end with no luck :(. Please, help me. What am I doing wrong? -- Regards, Boris

3 12

Re: [Freeswitch-users] non-blocking bridge in Lua
by Mi Ke 15 Jul '11

15 Jul '11

Many thanks, Avi ! The following code worked for me: local ob_session = freeswitch.Session("{return_ring_ready=true,execute_on_answer=lua ob_answer.lua}" .. called_id,session) if (ob_session:ready()) then freeswitch.bridge(session,ob_session) else --- process failure here end All channel data related to the bridge and both legs are available from ob_answer.lua via session:getVariable() All the best / Mike ----- Original Message ----- From: Avi Marcus Sent: 07/15/11 01:48 AM To: FreeSWITCH Users Help Subject: Re: [Freeswitch-users] non-blocking bridge in Lua Some options: You can set an execute_on_answer on the leg B to run your new lua script http://wiki.freeswitch.org/wiki/Variable_execute_on_answer Or, you can schedule all this stuff in the originate string, with either execute_on_answer or export nolocal:api_on_answer=sched_transfer or the like. -Avi Marcus On Fri, Jul 15, 2011 at 1:31 AM, Mi Ke <http://mi.ke @http://null.net > wrote: Hi all ! I want to make a Lua app which will start with Leg A, collect input, initiate outbound Leg B and when it answers, periodically play decreasing series of beeps to Leg A depending on how many minutes left to disconnect (remaining time queried via odbc). My initial idea was to issue a non-blocking bridge command after Leg B session originate returns ring-ready and then to do the rest in while loop, but the bridge called by api:executeString("bgapi bridge legB_session legA_session") just returns Job UUID without actually bridging sessions. A usual freeswitch.bridge call blocks further script execution until unbridged, disallowing me to check Leg B states when it's active. Is there any elegant way to acheive my goals using Lua/Dialplan apps ? To create an app starting with Leg A and constantly watching for Leg B existense/states, playing corresponding feedback messages to Leg A ? Thanks in advance for yours hints/responses Mike _______________________________________________ Join us at ClueCon 2011, Aug 9-11, Chicago http://www.cluecon.com 877-7-4ACLUE 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

1 0

Re: [Freeswitch-users] rtmp_url folder
by Jack 15 Jul '11

15 Jul '11

Thanks, it does connect with just the sip ip . My client shows connected, but I am not getting a keypad and nothing happens when I click new call or login.??? I am serving the web page from windows iis6 . I am getting an error in jquery.hotkeys-0.7.9.min.js line 124: 1. Uncaught TypeError: Cannot read property 'combi' of null 1. Has anyone else experienced this and found a solution? On 7/14/2011 10:21 PM, Gohar Ahmed wrote: > > It doesn't matter , you can remove the "/blabla" part from > rtmp://ip.of.server and it'll still connect. Atleast It worked for me. > > *From:*freeswitch-users-bounces@lists.freeswitch.org > [mailto:freeswitch-users-bounces@lists.freeswitch.org] *On Behalf Of > *Seven Du > *Sent:* Friday, July 15, 2011 5:05 AM > *To:* FreeSWITCH Users Help > *Subject:* Re: [Freeswitch-users] rtmp_url folder > > I think point to /phone as in the example should be ok > > -- > Seven Du > > About: http://about.me/dujinfang > > Blog: http://www.dujinfang.com > > Proj: http://www.freeswitch.org.cn > > Sent with Sparrow <http://www.sparrowmailapp.com> > > On Thursday, July 14, 2011 at 4:45 AM, Jack wrote: > > Freeswitch.html has a spot to set the variable rtmp_url. What should > be in the folder it points to? > Thanks, > Jack > > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > FreeSWITCH-users mailing list > FreeSWITCH-users(a)lists.freeswitch.org > <mailto:FreeSWITCH-users@lists.freeswitch.org> > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > > _______________________________________________ > Join us at ClueCon 2011, Aug 9-11, Chicago > http://www.cluecon.com 877-7-4ACLUE > > 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

1 0