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mod_event_socket 'linger timeout, closing socket'
by Alejandro Recarey 31 May '17

31 May '17

Hi all, thanks for taking the time to read, I'm using an outbound socket server (plivo open source) to control my FS calls. The problem I'm having is that when using the 'record' command sometimes the outbound socket app is not receiving the events that signal the end of the recording. This is a somewhat random event. Sometimes it works perfectly, and sometimes it doesnt. In the cases where it doesnt work, I get the following message in the FS console: 2017-05-30 16:55:53.013021 [DEBUG] mod_event_socket.c:1431 linger timeout, closing socket I added a little bit of extra logging to mod_event_socket.c and recompiled and I got this info: 2017-05-31 12:46:47.009854 [DEBUG] mod_event_socket.c:1431 Extra Debug: Linger timeout: 1496234806 - Epoch Time Now: 1496234807 2017-05-31 12:46:47.009854 [DEBUG] mod_event_socket.c:1432 linger timeout, closing socket I wanted to print those two variables, because they're used in mod_event_socket in the if that determines whether to throw the timeout error or not: if (switch_test_flag(listener, LFLAG_HANDLE_DISCO) && listener->linger_timeout != (time_t) -1 && switch_epoch_time_now(NULL) > listener->linger_timeout) { switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(listener->session), SWITCH_LOG_DEBUG, "linger timeout, closing socket\n"); status = SWITCH_STATUS_FALSE; break; } Notice that if the epoch has advanced just 1 second (hence epoch > linger_timeout) the socket is dropped. I would hope that the socket would linger for longer than one second. Is there any way to raise this timeout? Thanks again, Alex

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Mailing list issues.
by Michael Jerris 31 May '17

31 May '17

It looks like the mailing list was having issues stemming from our recent server maintenance. It should be corrected now. Mike

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Subscription to different conferences at a time
by Deepika Yadav 31 May '17

31 May '17

Hi, I have a scenario where mutliple conferences are hosted through Freeswitch. I programmed a python thread that subscribes to Freeswitch conference events using following filters: freeswitch_connection.events('plain', 'all') if event_name == "CHANNEL_ANSWER": ---do something...... if event_name == "CUSTOM": action = conference_event.getHeader('Event-Subclass') if action == 'conference::maintenance': conf_action = conference_event.getHeader('Action') if conf_action == 'conference-destroy': <do something> elif (conf_action == 'unmute-member'): <do something> This works perfect only when there is one conference going on. I am worried when mutiple conferences will be there then this subscription to events of other conferences also. Is there any way for subscribing to conferences based on their IDs or uuids? Regards Deepika https://www.iiitd.edu.in/~deepikay/

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test
by Matt Broad 31 May '17

31 May '17

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Codec list after restarting freeswitch
by Fred Schulz 31 May '17

31 May '17

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mod_voicemail greeting_path file permissions
by Jonathan Hunter 30 May '17

30 May '17

Hi Guys, Hope all are well? With mod_voicemail, can the greeting_path be set as a variable? And if so can the file permissions also be defined? I see there was a patch for file_path sometime ago but just wondered if the same can be set for greeting_path, as I need to allow both apache and freeswitch to access the same directory and fileset and permissions seems to be playing a part. thanks jon

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Testing
by Brian West 30 May '17

30 May '17

Oh and don't forget to register for ClueCon :) -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com Book a phone call (CST) <https://freeswitch.com/appointment> Allison prompts for FreeSWITCH: *https://www.gofundme.com/allison-prompts-for-freeswitch* <https://www.gofundme.com/allison-prompts-for-freeswitch> Got Bugs? Report them here <https://freeswitch.org/jira>! | Reddit: /r/freeswitch <https://www.reddit.com/r/freeswitch> *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *Skype:*briankwest

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Turn configuration mod_sofia
by Alexander Haugg 30 May '17

30 May '17

Hi all, the stun configuration for the mod_sofia profile is very easy "<param name="ext-rtp-ip" value=" stun:stun.freeswitch.org"/>" but what is to do, if i need a relay candidate in the sdp? How can i set the turn address and the login credentials? Thanks a lot

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Fax receive issue with t30 codec
by Hardik Patel 30 May '17

30 May '17

Hello, I am using opensips as entry point using dispatcher. opensips( 127.0.0.1), i am routing call to freeswitch server (127.0.0.3). Now I am trying to receive fax, my issue is when i try to send fax in softphone(Zoiper) from the log i am seeing that it is sending fax using t30 codec. and i am not receiving the fax at destination, is it because of codec, should it only work with t38 codec? if that is the issue than how am i be able to send the fax using t38 from zoiper? Here i am attaching the fs log with loglevel 9 and sip trace is also enabled. 127.0.0.2 => carrier/provider IP 123456789 => Fax number test(a)gamil.com => Email Address 127.0.0.4 =>UI IP Pastebin link:https://pastebin.freeswitch.org/view/9ec52715 -- Hardik Patel iNextrix Technologies Pvt Ltd

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Windows build - mod_lua
by Gregor Nanger 30 May '17

30 May '17

I've compiled FS with visual studio and there is no mod_lua.dll generated. Hence I get error when FS starts that it can not find dll. Windows build went fine (with excluded modv8) Anyone also find this problem? Best regards, Gregor

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