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22 Oct '13

Hello List, I'm a newbie to Freeswitch. I need to bridge an inbound call to another call, and detect pattren of DTMF. If the the pattern happen, I need to capture it to a variable, and hangup leg B (the bridged one), and continue doing stuff with Leg A. If Leg B was disconnected, I need hangup leg A as well. I wish to better understand how to look such thing on the wiki, prior for direct answer on how to do it, in order to better learn FS and it's usage :) So any help on this matter is more then welcome. Thanks Ido

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Can't receive CHANNEL_HANGUP_COMPLETE event when using mod_rtmp
by 王永涛 22 Oct '13

22 Oct '13

Hi I made some changes on testserver.c. Then I call a sipnumber from the flex client to a x-lite client. When I hang up from the flex side, can receive CHANNEL_HANGUP_COMPLETE event loop can be ended. When I hang up from the x-lite side, I can't receive CHANNEL_HANGUP_COMPLETE event and loop can not be ended, the variable status value is ESL_BREAK. Thanks. Code is as follows static void mycallback(esl_socket_t server_sock, esl_socket_t client_sock, struct sockaddr_in *addr) { esl_handle_t handle = {{0}}; int done = 0; esl_status_t status; time_t exp = 0; esl_attach_handle(&handle, client_sock, addr); esl_log(ESL_LOG_INFO, "Connected! %d\n", handle.sock); esl_filter(&handle, "unique-id", esl_event_get_header(handle.info_event, "caller-unique-id")); esl_events(&handle, ESL_EVENT_TYPE_PLAIN, "CHANNEL_HANGUP_COMPLETE DTMF CUSTOM"); esl_send_recv(&handle, "linger"); esl_execute(&handle, "set", "hangup_after_bridge=true", NULL); esl_execute(&handle, "bridge", "user/1001(a)192.168.1.128", NULL); int i=rand(); while((status = esl_recv_timed(&handle, 1000)) != ESL_FAIL) { printf("Thread:%d\n", i); if (done) { if (time(NULL) >= exp) { break; } } else if (status == ESL_SUCCESS) { const char *type = esl_event_get_header(handle.last_event, "content-type"); if (type && !strcasecmp(type, "text/disconnect-notice")) { const char *dispo = esl_event_get_header(handle.last_event, "content-disposition"); esl_log(ESL_LOG_INFO, "Got a disconnection notice dispostion: [%s]\n", dispo ? dispo : ""); if (dispo && !strcmp(dispo, "linger")) { done = 1; esl_log(ESL_LOG_INFO, "Waiting 5 seconds for any remaining events.\n"); exp = time(NULL) + 5; } } } } esl_log(ESL_LOG_INFO, "Disconnected! %d\n", handle.sock); esl_disconnect(&handle); }

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AdhearsionConf 2013
by Ben Langfeld 21 Oct '13

21 Oct '13

Apologies for interrupting Ladies and Gents, but if you would spare a moment of your time for a brief suggestion... Old School Telephony meets the New School Open Web: Adhearsion ( http://adhearsion.com) is one of the premiere choices of framework for building rich Voice and Real-Time Communications applications. The project is six years old and has a history of successful deployments of very powerful and sophisticated voice integrations. At AdhearsionConf (Dec 4-5, Atlanta GA, http://adhearsionconf.com) you can hear from experts in the space and learn how to integrate new tools like WebRTC, as well as advancements in hot topics like speech recognition and text-to-speech, and how to use Adhearsion in conjunction with FreeSWITCH to vastly expand the capabilities of your network both internally and shared with third-party developers to create applications which leave a lasting impression on your users. Tickets for the conference are available at http://adhearsionconf.com/tickets at the early-bird rate of $110 until Nov 11th. The FreeSWITCH community's very own Chris Rienzo will be there to talk about mod_rayo, along with a flock of other excellent speakers including Tim Wenhold (PHRG) and Matt Jordan (Digium). I'd personally love to see FreeSWITCH people making a strong attendance and showing Matt what for! The first round is also on me! Ben Langfeld Adhearsion Foundation Inc.

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How to prevent group_confirm_file from looping?
by Lyle Pratt 21 Oct '13

21 Oct '13

Hello! I'm trying to use the "group_confirm_file" functionality to prevent voicemail systems from answering my outbound calls to users. The problem I'm having is that there does not seem to be a way to prevent the "group_confirm_file" from looping/repeating. I only want the file to play ONE time then hang up if they do not press the "group_confirm_key". Currently, if someone ignores one of the calls (on their cell phone for example) and their voicemail system picks up, they will get a voicemail with the "group_confirm_file" audio. This is extremely annoying! How can I change this? Is it possible for me to write a module to change/correct this behavior? Thanks! Lyle

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How do I stop group_confirm_file from looping?
by Lyle Pratt 21 Oct '13

21 Oct '13

I'm trying to use the "group_confirm_file" functionality to prevent voicemail systems from answering my outbound bridge calls to users. The problem I'm having is that there does not seem to be a way to prevent the "group_confirm_file" from looping/repeating. I only want the file to play ONE time then hang up if they do not press the "group_confirm_key". Currently, if someone ignores one of the calls (on their cell phone for example) and their voicemail system picks up, they will get a voicemail with the "group_confirm_file" audio. This is extremely annoying! How can I change this? Is it possible for me to write a module to change/correct this behavior? Thanks! Lyle Pratt

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Same timestamp 161 for all RFC2833 events generated by freeswitch
by Yuri Ritvin 21 Oct '13

21 Oct '13

In Freeswitch 1.2.12 all RFC2833 events generated by freeswitch bear the same timestamp (161). Is there a way to send incremental timestamp as it's sent for the rest of the RTP packets ? Thank you. Yuri

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FreeSWITCH news and Notes - FreeSWITCH 1.2.14 Released!
by Ken Rice 21 Oct '13

21 Oct '13

FreeSWITCHers! Today we are proud to announce FreeSWITCH 1.2.14! Available today via git, http://files.freeswitch.org/freeswitch-1.2.14.tar.bz2, and the deb and yum repos! This is a maintenance release to address several bugs that have been identified since the last release. Also dont forget ClueCon Weekly Conference Call! Every Wed at 1PM EST! For more information on how to join see: http://wiki.freeswitch.org/wiki/Weekly_Conference_Call_Calling_Instruct ions -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch

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How to set channel variable per SIP profile
by Victor Chukalovskiy 20 Oct '13

20 Oct '13

Hi, Subject says it all. Is there a way to set channel variable for all calls passing through a SIP profile? For example, I want all calls authorized with ACL and hitting SIP profile to have country=foo. This has to be done before hitting the dialplan. There is a bunch of "params" in the SIP profile, but these aren't channel variables. Thanks! -Victor

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Freesiwtch 1.2.14 Windows
by Andre 20 Oct '13

20 Oct '13

Hi, when I compile 64 bit I get this error Error 1 error C1083: Cannot open include file: 'nametab.h': No such file or directory \freeswitch-1.2.14\freeswitch-1.2.14\libs\xmlrpc-c\lib\expat\xmltok\xmltok.c 10 1 xmltok Any idea what this means or how to fix it? Or does it even matter? Andre

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Limit
by Andre 19 Oct '13

19 Oct '13

Hi, how do I set the CPS and Port limits on a bridge Can I set both on one application or do I need to set 2 limits? 5 ports <action application="limit" data="hash outbound customerxxx 5 !NORMAL_TEMPORARY_FAILURE" /> 5 cps <action application="limit" data="hash outbound customerxxx 5/1 !NORMAL_TEMPORARY_FAILURE" /> Can I do this for both 5 ports and 5 cps? <action application="limit" data="hash outbound customerxxx 5 5/1 !NORMAL_TEMPORARY_FAILURE" />

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