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Examples
by Robert Clayton 14 Aug '08

14 Aug '08

Are there any examples of using a Dialplan to allow a single incoming extension to bridge to simultaneous extensions which are linked to an application (example recording)? Thanks

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google talk and freeswitch
by jordan pan 13 Aug '08

13 Aug '08

Hi everyone, I want to know whether freeswich can become a xmpp server for gtalk,I means gtalk don't need to login on the talk.google.com. then, I also want to konw the difference between asterisk and freeswtich to the theory of work together with google talk. Thanks! Best regards! Eric

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Don't hang up after application has ended
by Alois Komenda 12 Aug '08

12 Aug '08

Hello, I have written an implementation of the switch_application_interface. Now I noticed, that after the execution of the application_function the session is hung up automatically. Can I take any influence on this behavior? I don't want the session to be closed in some cases. Applications, which are called in the future, should be able to work with this session. For example the first application decides what to do with a session. And the second one bridges it to an endpoint. Is that possible at all? Thanks in advance! Alois Komenda Fraunhofer-Einrichtung für Systeme der Kommunikationstechnik ESK

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Rhino Equipment
by James Finstrom 12 Aug '08

12 Aug '08

Greetings Devs, A brief introduction, I am James one of the nice folks here at Rhino and I am making a goal to have integration of our hardware into freeswitch documented and published within the next 30 days. This seems to have been an issue or there has been some problem with communication that has made this not happen yet. Anyhow if you have any questions or concerns email me. I know there are folks in the FS world using our stuff so I am confident it works but we sort of have to do it here to "officialy support it." If you ave had success setting up one of our cards or if you are an openzap guru please also get ahold of me. I also have a php hardware abstraction deal where I should be able to auto configure our cards once I understand the base configurations. Regards, James Finstrom Rhino Equipment Corp. All Rhino products are made in America, 100% Money Back Guarantee, and have a 5 Year warranty. Sometimes for it to "just work" you have to think outside the box! Phone: 1-877-RHINO-T1 ~ FAX: +1 (480) 961-1826 IP: asterisk.rhinoequipment.com ~ FWD: 633686 THIS COMMUNICATION MAY CONTAIN CONFIDENTIAL AND/OR OTHERWISE PROPRIETARY MATERIAL and is thus for use only by the intended recipient. If you received this in error, please contact the sender and delete the email and its attachments from all computers.

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If you are interested to develop applications for Freeswitch using Win32 please send email to paulh@instruments.com .
by Paul Herring 11 Aug '08

11 Aug '08

Paul Herring Computer Instruments, Inc. 9901 West 87th Street Overland Park, Kansas 66212 USA DID 913-307-8860 CELL 913-526-9158 paulh(a)instruments.com www.eivrsolutions.com <http://www.eivrsolutions.com/> -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

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Originating channel of a DTMF event
by Juan Jose Comellas 11 Aug '08

11 Aug '08

I currently have the need to know which channel originated a DTMF event when inside a bridge. I'm monitoring events through mod_event_socket. The problem is that, as far as I can see, the originating channel ID is never carried with the DTMF event. Is there any reason for this? I've been looking for ways to modify FreeSWITCH to support this and I wanted to discuss it with the core developers to make sure that whatever modification I make is acceptable to be merged back into FS. The options I'm contemplating right now are: 1. Add the originating channel ID to the switch_dtmf_t struct. Pros: don't need to change the interfaces in any of the functions that are used to send DTMFs; cons: this change would make the struct almost 8x bigger than what it currently is. 2. Change switch_core_session_send_dtmf() (switch_core_io.c) to receive the channel ID as an argument and have the function pass the channel ID to all the event hooks when they are called. Is any of these options acceptable? Does anybody have any recommendations about this?

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Typo in stfu.c ?
by Vassil Panayotov 10 Aug '08

10 Aug '08

Hi list, I asked Anthony about this in IRC, but he seemed to be quite busy at cluecon and I really don't understand how this works. If I understand correctly, the function stfu_n_measure_interval tries to estimate what the difference between timestamps of two consecutive RTP packets should be and records the frequency of occurrence? I don't understand the following(stfu.c:128) : if (track[index] > most) { most = index; } What is the point of comparing the frequency of occurrence with an index? Maybe it should be: if (track[index] > track[most]) { most = index; } Best regards. Vassil

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conference with background sound + some details on t
by t 10 Aug '08

10 Aug '08

i already asked this on the fs-users list. so i like to ask the developers if they can give me some hints. %< i like to setup conferences with some ambient music in the background. is there a background sound parameter? when i use the perpetual sound with local_stream its fine. but all the users have unmute themselve first. how can i do things with a user after he entered a conference? in the dialplan? via script? ... like unmuting him. a more general question is how to play no-sound for certain events. if there is something like: <param name="kicked-sound" value="conference/conf-kicked.wav"/> can i set it to something like: <param name="kicked-sound" value="nosound"/> ?? %< ... some more background about me ... i thought it is a really common setup to have conferences with backgroundchannels. by thinking deeper about the idea with queuebased conferencing and room-oriented ivr-system (voice-muds) i started the project blabla (but its still a on a very beginning) [python + yate] -- https://blabla.bountysource.com/ i also took part on the idea of VARE (virtual a rendering engines) which is hyperlinked conferencing. or a base for some "audio-web". http://blog.justdigits.de/?cat=11 <-- sorry german but basically a audiointerpretation of documents like http://justdigits.de/vare/Erdgeschoss.vare#S by some 16 channel mixer ontop of asterisk and xmpp. since 1 year i work on something called dialstation. which is php+rubyonrails+freeswitch. more about this can you see on my talk on asterisktag. [where i also announced the release of FreeSWITCH 1.0 ;-) ] http://www.asterisk-tag.org/wiki/Programm_27.05.2008_(Mixed_Track) thats for now. lets see who i meet here on this list. t.

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Unit testing
by Simon Shaw 09 Aug '08

09 Aug '08

Does freeswitch have a unit testing policy? If so what is the code coverage like? Thanks, Simon

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Core Dump: Potential Double Free
by Patrick Grondin 07 Aug '08

07 Aug '08

Hi, I've had a problem last night where Freeswitch core dumped complaining about a potential double free. At that time, it was doing an all night automated test batch with a load averaging 230 simultaneous SIP channels. If nobody has seen this yet, I'll file a jira about that. Here is the bt of the dump : #0 0x000000359c830155 in raise () from /lib64/libc.so.6 #1 0x000000359c831bf0 in abort () from /lib64/libc.so.6 #2 0x000000359c86a3db in __libc_message () from /lib64/libc.so.6 #3 0x000000359c871684 in _int_free () from /lib64/libc.so.6 #4 0x000000359c874ccc in free () from /lib64/libc.so.6 #5 0x00002ab676bf7cb5 in switch_buffer_destroy (buffer=0x1b569c10) at src/switch_buffer.c:287 #6 0x00002ab676c0084d in switch_core_media_bug_destroy (bug=0x1b569c08) at src/switch_core_media_bug.c:40 #7 0x00002ab676c00af6 in switch_core_media_bug_remove_all (session=0x1d2f7a38) at src/switch_core_media_bug.c:283 #8 0x00002ab676c090ca in switch_core_session_thread (thread=<value optimized out>, obj=<value optimized out>) at src/switch_core_session.c:788 #9 0x000000359d406307 in start_thread () from /lib64/libpthread.so.0 #10 0x000000359c8d1ded in clone () from /lib64/libc.so.6

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