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How to creat an outgoing media channal for non-SIP?
by max 18 Jul '11

18 Jul '11

Hello, I want to write a quick and dirty extension as a prove of concept for Freeswitch translating SIP into another protocol(e.g. proprietary RUF XML-RTP for calls). For that I chose to use mod_erlang_event to handle the handshake and the other standards server. The media should be handled completely by Freeswitch. Please see the attached picture. The focus on this project is to get used to Erlang as well as to Freeswitch. To the problem: While creating an outgoing media channel via Erlang to the non SIP I got stuck. Can I use originate in that context and if yes, how? If no, is there any other command? Thank you, Max

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Looking for US based providers offering t.38 that work with freeswitch
by Tony Graziano 18 Jul '11

18 Jul '11

So far we have only been able to find a provider using sonus switches for this. Is anyone successfully using a trunk provider to receive faxes via t.38 in freeswitch they are willing to share? Thanks, Tony

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Hanged up callers doubt
by Leonardo P. Bidinoto 18 Jul '11

18 Jul '11

Hi everyone! I having some troubles with my FS box(CentOS 5.3 64-bit, Xeon(R) CPU 2.66GHz with 8 cores , 8GbRAM, (git-bc19d28 2011-04-25 15-53-54 -0400) ). Sometimes, when a user calls to my box and hangs up, the user's channel remains on FS when i execute a "show channels" command. with "uuid_exists" command, im receiving "false". looking at freeswitch log, the last message for the channels is: 0eeb975f-2ae9-4f7d-9888-1478faf87d2d 2011-05-11 00:05:57.555142 [DEBUG] switch_core_session.c:1286 Session 38 (sofia/external/2562687580(a)10.0.70.6) Locked, Waiting on external entities Im not receiving "Ended" State after the previous message. did someone have seen this issue? or know a way to remove from freeswitch? Thanks all. -- Leonardo Pires Bidinoto Voice Technology www.voicetechnology.com.br

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How to creat an outgoing media channal for non-SIP?
by max 18 Jul '11

18 Jul '11

Hello, First of all I am sorry if you receive this mail double. I sent it but I didn't get a copy on the mailing list. I want to write a quick and dirty extension as a prove of concept for Freeswitch translating SIP into another protocol(e.g. proprietary RUF XML-RTP for calls). For that I chose to use mod_erlang_event to handle the handshake and the other standards server. The media should be handled completely by Freeswitch. The focus on this project is to get used to Erlang as well as to Freeswitch so solutions should contain this two parts. To the problem: While creating an outgoing media channel via Erlang to the non SIP I got stuck. Can I use originate in that context and if yes, how? If no, is there any other command to realise that? Also I would need to know how to control/close this media channel. Thank you, Max

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.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.

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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

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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

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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

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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

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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

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