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5 sec delay in FreeSWITCH when caller and callee tries to place the call off hold at the same time
by Jyotshna Cherukuri 15 May '18

15 May '18

Hello, I have a scenario where caller and callee both place the call on hold and when they both try to un hold the call at the same time , I've noticed FreeSWITCH introducing 5 sec delay to unhold both the legs. I have tracked the 5 second delay in switch_core_media.c to switch_channel_wait_for_flag... if (b_channel) { if (switch_channel_test_flag(session->channel, CF_HOLD)) { switch_ivr_unhold(b_session); switch_channel_clear_flag(session->channel, CF_SUSPEND); switch_channel_clear_flag(session->channel, CF_HOLD); } else { switch_channel_stop_broadcast(b_channel); switch_channel_wait_for_flag(b_channel, CF_BROADCAST, SWITCH_FALSE, 5000, NULL); } } switch_channel_wait_for_flag is not returning back and its timing out I believe . I am not sure if this a valid bug and I want to understand why it's not clearing CF_BROADCAST flag on one leg . Any help is appreciated . Also, under perfect working scenarios I see "done playing file local_stream://xxx" on the console , but I dont see this log for the above scenario . Appreciate all your help ! ThanksJyotshna

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Set default ENUM resolver to the ENUMER: enum.enumer.org
by Oleg Khovayko 03 May '18

03 May '18

Hello, 1. Currently, ENUM module is almost not supported in FreeSWITCH, and default lookup site (e164.org) specified in the mod_enum, is our of service for long time. 2. We started blockchain-based distributed ENUM service, and it is grows these days, and several commercial switches already using it. This document contains pptx presentation, and several articles about ENUMER: http://bit.ly/ENUMER 3. I would like contact someone from FS community, who will change default configs for mod_enum to our public resolver enum.enumer.org I believe, ENUMER is a great chance to get united SIP/VOIP world, which is currently fragmented. And united we will be strong! 4. I also open to all community members, to help you adopt ENUMER in your FreeSWITCH based PBX, or to other cooperation activities. Thanks, Oleg Khovayko, ENUMER inventor.

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Using extern variable - undefined reference while loading module
by Vidhya S R 24 Apr '18

24 Apr '18

Hi, I'm using an extern variable (switch_hash_t *) declared in a header file under src/include. This variable is defined in another .c file under src. Now, I'm including this header file in my module under src/mod/my_mod and trying to initialise the hash. It compiles fine but when freeswitch is run, the module is not loaded and throws the error undefined reference. Im not sure why. Sorry if this is a basic question.

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switch_poll time out value is not honored
by kaiduan xie 19 Apr '18

19 Apr '18

Hi, For my application, I made some changes in switch_rtp to use poll to read from network instead of synchronising on timer, switch_rtp_clear_flag(rtp_session, SWITCH_RTP_FLAG_USE_TIMER); I observed a weird behaviour with regarding to switch_poll, it blocks with 20ms always even the timer value is 10 ms when there are no packet received on network. + pt = 10 * 1000; // 10ms+ switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG, "+poll reading:%d\n", pt); } poll_status = switch_poll(rtp_session->read_pollfd, 1, &fdr, pt);+ switch_log_printf(SWITCH_CHANNEL_SESSION_LOG(rtp_session->session), SWITCH_LOG_DEBUG, "-poll reading:%d:%d\n", pt, poll_status); The following is from the log, 0c19d3aa-429e-11e8-bbd0-fb8396b8383d 2018-04-17 20:21:31.453604 [DEBUG] [23807] switch_rtp.c:7001 +poll reading:100000c19d3aa-429e-11e8-bbd0-fb8396b8383d 2018-04-17 20:21:31.473649 [DEBUG] [23807] switch_rtp.c:7005 -poll reading:10000:2 Any idea? This is really weird, I expect switch_poll works as select. Many thanks for help. /Kaiduan

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PRESENCE IN
by Marcelo Araujo 10 Apr '18

10 Apr '18

Hi all. I'm trying to use PRESENCE IN event to make BLF signalize some events to my endpoints I'm following example in: https://wiki.freeswitch.org/wiki/PRESENCE_IN_event_example but nothing happens. Any other functional examples usage of PRESENCE IN? -- MARCELO ARAUJO LPIC-3, CLE, dCAA 48 40528322 marcelo(a)gnovit.com http://gnovit.com Gnovit .:. Open source consulting .:.

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BFCP Support in Freeswitch
by Sagar Joshi 05 Apr '18

05 Apr '18

Hi Ken and Brian, I would like to quickly introduce myself . I am working as developer in VoIP and Web Conferencing domain for over 9 years now. Currently my company GS Lab. Pvt. Ltd.. Pune , India is actively using and customizing Freeswitch for different projects. Apart from that , we have developed the BFCP feature for Opal Voip. This will allow OPAL to have content sharing in SIP calls. We are in the process contributing it with Opal. We wanted to check with you if we can contribute the same to FreeSwitch . Thanks and Regards Sagar Joshi Associate Technical Manager Convergence Practice

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unexported apr_file_unlock
by Paul Zillmann 08 Mar '18

08 Mar '18

Hello, in the APR wrapper switch_file_unlock(switch_file_t *thefile) is missing. Due to the high error setting on the compiler, the usage of apr_file_unlock() is not possible. Is this by mistake? Sincerely, Paul

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Send sound in a conference
by Cédric Clavier 23 Jan '18

23 Jan '18

Hello everybody, I’m in trouble with freeswitch, there must be something i didn’t understood but i don’t know what. I want to add a user (no importance who it is (phone-number, name)) into a conference and this user job is to send sound into the conference (based on the http://lists.freeswitch.org/pipermail/freeswitch-users/2015-July/114783.html) Another part of his job is to keep the conference alive even if there is nobody inside So I did the following In the internal dialplan i put {{{ <extension name="playmedia"> <condition field="destination_number" expression="^ playmedia $"> <action application="endless_playback" data="/home/toto/medias/media.mp3" /> </condition> </extension> }}} And then I try in the conference to make a call to this dialplan from a conference {{{ conference 146806@[sip_profile_in_use] dial sofia/internal/playmedia }}} Normally the default user is used to make the call no ? The result of my command line is {{{ 2018-01-23 17:14:49.282641 [DEBUG] switch_ivr_originate.c:2142 Parsing global variables 2018-01-23 17:14:49.282641 [WARNING] mod_sofia.c:4664 Cannot locate registered user playmedia@internal 2018-01-23 17:14:49.282641 [NOTICE] mod_sofia.c:4918 Close Channel N/A [CS_NEW] 2018-01-23 17:14:49.282641 [DEBUG] switch_core_state_machine.c:741 () Running State Change CS_DESTROY (Cur 0 Tot 27) 2018-01-23 17:14:49.282641 [DEBUG] switch_core_state_machine.c:751 (N/A) State DESTROY 2018-01-23 17:14:49.282641 [DEBUG] mod_sofia.c:343 N/A SOFIA DESTROY 2018-01-23 17:14:49.282641 [DEBUG] switch_core_state_machine.c:751 (N/A) State DESTROY going to sleep 2018-01-23 17:14:49.282641 [NOTICE] switch_ivr_originate.c:2851 Cannot create outgoing channel of type [sofia] cause: [USER_NOT_REGISTERED] 2018-01-23 17:14:49.282641 [DEBUG] switch_ivr_originate.c:3848 Originate Resulted in Error Cause: 606 [USER_NOT_REGISTERED] }}} I don’t understand what happen. Thank in advance for your answers. Cédric Clavier

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Any prompts y’all want Callie to record?
by Chad Phillips 07 Dec '17

07 Dec '17

My company is scheduling some time with Callie from GM Voices to record a few voiceover pieces. Since the setup cost is always the most expensive part of the whole deal, I always like to reach out to the FreeSWITCH community and see if there are any new phrases we’d like to add to the sound files. Anybody have ideas? As long as it’s not too crazy many words, I’ll just eat the cost :) Let me know by Friday December 1st — we’ll probably submit everything the following Monday. Chad

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Fwd: FreeSWITCH Video Conferencing Blackout issue
by Richard Hsu 06 Nov '17

06 Nov '17

Hi all, [Issue] I am experimenting with the FreeSWITCH version 1.6.2 video calling capability. In particular, we are seeing an issue where sometimes the video element in HTML will start streaming videos just fine, but most of the time, one side of the video call, or both, will have a screen being pitch black. The audio is totally fine though. [Self-diagnosis] Looking into chrome://webrtc-internal, it indicates that there is no frame being sent, and looking into the FreeSWITCH log, it appears this line: "[ERR] switch_rtp.c:8098 Error: video SRTP protection failed with code 10". is always present. I looked into the source code, which corresponds to the switch_rtp.c file, it seems like the "srtp_protect()" function, which transforms/encrypts an RTP packet into SRTP packet, returns an error code. I tried by patching it with the changes here: https://freeswitch.org/jira/br owse/FS-10667, since it seemed like a similar issue, but it didn't work either. I understand there are several ways to go about this: 1) stick a debugger or 2) try out the latest version. [Question] Without other major changes, what else would you recommend me to look at? More importantly, has anyone seen this strange behavior of FreeSWITCH before? I appreciate your effort with FreeSWITCH so far and would love to hear your thoughts! Thanks, Richard -- Richard Hsu Massachusetts Institute of Technology Department of Electrical Engineering and Computer Science Class of 2016

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