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choppy sound of voicemail recording files
by Ketan Kothari 29 Jun '17

29 Jun '17

Hi Guys we have implemented voicemail features on our system but we are facing an issue with choppy sound of recording files. When the files are playing an instructions the sounds is choppy and unable to listen it properly we use "rtp-autofix-timing" variable to resolve this issue and really it works fine for us but then its not allow us to record a message and hang up the call with message "Message is less than minimum record length".

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FreeSWITCH and OpenSIPS Training will be available this August at the end of ClueCon
by Anthony Minessale 28 Jun '17

28 Jun '17

There is limited availability so sign up early! https://www.cluecon.com/training.html -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA

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How to delay setInputCallback() while playing audio?
by Muhammad Naseer Bhatti 28 Jun '17

28 Jun '17

I am running an IVR which collects DTMF input while the file is being played, written in Lua. I set InputCallback and then listen for DTMF events. If event found, take actions etc. session:setInputCallback("eventHandler") playbackMessage (playback_delay) Facing a strange issues with Verizon where they send a quick RTP Event write after starting the audio, which triggers the DTMF event and bad things happen in my script. This only happens within first 1 to 2 seconds of RTP. How can I have my InputCallback listen for events only after a few seconds while still playing the audio? or do I have to switch to playAndGetDigits() ? -- Sent with Airmail

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tier change event
by Mimiko 28 Jun '17

28 Jun '17

Hello. mod_callcenter generate agent's state or status change events. Is there events generated when a tier is added, changed, removed?

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Is it possible to send an SIP UPDATE message via sendevent ?
by Jonathan Hunter 28 Jun '17

28 Jun '17

Hi Guys, I have a quick question, I can see from documentation you can send SIP messages via sendevent and the ESL. Is it possible to send a SIP UPDATE message with this method and if so are there any configuration options I need to be aware of? Essentially we want to send an UPDATE after answer of a call to flag a change. Thanks Jon

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keep limit counter active across dialplans when used in attended transfers
by Bipin Patel 28 Jun '17

28 Jun '17

hi, i have a requirement where i need to generate a different ringtone if that ext is on another call and i do it using limit by setting a counter for the caller and callee and then checking the counter if it has a value of 1 which would mean that ext is on anohter call so generate a different ringtone and this work fine. The problem being if A calls B and B attended transfers the call to C then the counter for C is incremented in the features dialplan but it doesnt reflect in the internal dialplan and i read that active limits dont remain global across dialplans so is there anyway to keep the limit counter active. i tried limit_ignore_transfer but that doesnt seem to work, im on the FS master Below is my internal and features dialplan, any pointers would be helpful INTERNAL DIALPLAN <extension name="general_stuff1" continue="true"> <condition> <action application="set" data="rtp_negotiate_near_match=true"/> <action application="set" data="inherit_codec=true"/> <action application="set" data="suppress_cng=true"/> <action application="set" data="limit_ignore_transfer=true" /> <action application="limit" data="hash max_calls ${sip_from_user} -1" /> </condition> </extension> <extension name="ChangeRingBack"> <condition field="destination_number" expression="^ChangeRingBack$"/> <condition field="rdnis" expression="^(.*$)"> <action application="set" data="transfer_ringback=%(250,250,400);%(250,3500,400)"/> <action application="set" data="ringback=%(250,250,400);%(250,3500,400)"/> <action application="bridge" data="sofia/internal/$1%$${domain}"/> </condition> </extension> <extension name="local_user_calling"> <condition field="destination_number" expression="^(.*)$"> <action application="limit" data="hash max_calls ${destination_number} -1" /> <action application="bind_meta_app" data="3 b s execute_extension::att_xfer XML features"/> <action application="set" data="transfer_ringback=%(400,200,400,375);%(400,2000,400,375)"/> <action application="set" data="hold_music=$${hold_music}" /> <action application="set" data="ringback=%(400,200,400,375);%(400,2000,400,375)"/> <action application="set" data="call_timeout=30"/> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=true"/> <action application="limit" data="hash max_calls ${destination_number} 1 ChangeRingBack"/> <action application="bridge" data="sofia/internal/$1%$${domain}"/> </condition> </extension> FEATURES DIALPLAN <extension name="att_xfer"> <condition field="destination_number" expression="^att_xfer$"> <action application="set" data="transfer_ringback=%(400,200,400,375);%(400,2000,400,375)"/> <action application="set" data="hold_music=$${hold_music}" /> <action application="set" data="ringback=%(400,200,400,375);%(400,2000,400,375)"/> <action application="set" data="continue_on_fail=true"/> <action application="read" data="2 11 ivr/ivr-enter_ext.wav digits 30000 #"/> <action application="set" data="origination_cancel_key=#"/> <action application="limit" data="hash max_calls ${digits} -1" /> <action application="att_xfer" data="sofia/internal/${digits}%$${domain}"/> </condition> </extension> -- Regards, Bipin ------------------------------------------------------------------------

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Problem with fs_encode
by Guillermo Ruiz Camauer 28 Jun '17

28 Jun '17

I have created a small utility program based on fs_encode which will take ALL files in a directory and convert them from .WAV to .G729 and .PCMU formats. The utility is called wavBatchEncode and takes just a directory path as an argument. This program was working very well until I upgraded FreeSwitch. Now the program seems to hang when it tries to write out the first converted files. I have to KILL the program from another terminal. It leaves 2 0 byte files with .PCMU and .G729 extension. I have run through a GIT bisect to find where things broke. The last working version is Version 1.6.16 git ae1cdce 2017-04-11. The first broken version is 38621e47bad3b63f03a0a27f6ca9ed92f6969032. I then used GIT DIFF to see if I could see what had happened. There are very few modifications between these two commits: root@fs3:/usr/src/freeswitch.git# git diff ae1cdce 38621e47bad3b63f03a0a27f6ca9ed92f6969032 diff --git a/src/include/switch_module_interfaces.h b/src/include/switch_module_interfaces.h index e0a5c20..7ca027d 100644 --- a/src/include/switch_module_interfaces.h +++ b/src/include/switch_module_interfaces.h @@ -329,6 +329,11 @@ typedef struct switch_mm_s { switch_video_profile_t vprofile; switch_video_encode_speed_t vencspd; uint8_t try_hardware_encoder; + int scale_w; + int scale_h; + switch_img_fmt_t fmt; + char *auth_username; + char *auth_password; } switch_mm_t; /*! an abstract representation of a file handle (some parameters based on compat with libsndfile) */ diff --git a/src/mod/applications/mod_av/avformat.c b/src/mod/applications/mod_av/avformat.c index b944625..4b92801 100644 --- a/src/mod/applications/mod_av/avformat.c +++ b/src/mod/applications/mod_av/avformat.c @@ -906,7 +906,7 @@ SWITCH_STANDARD_APP(record_av_function) char codec_str[256]; const AVCodecDescriptor *desc; - if (!strncmp(data, "rtmp://", 7)) { + if (!strncmp(data, "rtmp://", 7) || !strncmp(data, "rtsp://", 7)) { fmt->video_codec = AV_CODEC_ID_H264; fmt->audio_codec = AV_CODEC_ID_AAC; } @@ -1694,9 +1694,20 @@ static switch_status_t av_file_open(switch_file_handle_t *handle, const char *pa return SWITCH_STATUS_GENERR; } else if (handle->stream_name && (!strcasecmp(handle->stream_name, "rtmp") || !strcasecmp(handle->stream_name, "youtube"))) { format = "flv"; - switch_snprintf(file, sizeof(file), "rtmp://%s", path); + + // meh really silly format for the user / pass libav..... + if (handle->mm.auth_username && handle->mm.auth_password) { + switch_snprintf(file, sizeof(file), "rtmp://%s pubUser=%s pubPasswd=%s flashver=FMLE/3.0", path, handle->mm.auth_username, handle->mm.auth_password); + } else { + switch_snprintf(file, sizeof(file), "rtmp://%s", path); + } + + } else if (handle->stream_name && !strcasecmp(handle->stream_name, "rtsp")) { + format = "rtsp"; + switch_snprintf(file, sizeof(file), "rtsp://%s", path); } + ext++; if ((context = (av_file_context_t *)switch_core_alloc(handle->memory_pool, sizeof(av_file_context_t))) == 0) { @@ -1783,7 +1794,7 @@ static switch_status_t av_file_open(switch_file_handle_t *handle, const char *pa if (fmt->video_codec != AV_CODEC_ID_NONE) { const AVCodecDescriptor *desc; - if ((handle->stream_name && (!strcasecmp(handle->stream_name, "rtmp") || !strcasecmp(handle->stream_name, "youtube")))) { + if ((handle->stream_name && (!strcasecmp(handle->stream_name, "rtmp") || !strcasecmp(handle->stream_name, "rtsp") || !strcasecmp(handle->stream_name, "youtube")))) { if (fmt->video_codec != AV_CODEC_ID_H264 ) { fmt->video_codec = AV_CODEC_ID_H264; // force H264 @@ -2525,6 +2536,7 @@ SWITCH_MODULE_LOAD_FUNCTION(mod_avformat_load) supported_formats[i++] = "av"; supported_formats[i++] = "rtmp"; + supported_formats[i++] = "rtsp"; supported_formats[i++] = "mp4"; supported_formats[i++] = "m4a"; supported_formats[i++] = "mov"; diff --git a/src/mod/applications/mod_av/mod_av.c b/src/mod/applications/mod_av/mod_av.c index 3d3bd82..141fcdc 100644 --- a/src/mod/applications/mod_av/mod_av.c +++ b/src/mod/applications/mod_av/mod_av.c @@ -93,7 +93,7 @@ int mod_av_lockmgr_cb(void **m, enum AVLockOp op) static void log_callback(void *ptr, int level, const char *fmt, va_list vl) { switch_log_level_t switch_level = SWITCH_LOG_DEBUG; - + return; /* naggy messages */ if (level == AV_LOG_DEBUG || level == AV_LOG_WARNING) return; diff --git a/src/switch_core_file.c b/src/switch_core_file.c index 46ee539..aff9442 100644 --- a/src/switch_core_file.c +++ b/src/switch_core_file.c @@ -177,6 +177,14 @@ SWITCH_DECLARE(switch_status_t) switch_core_perform_file_open(const char *file, fh->mm.try_hardware_encoder = switch_true(val); } + if ((val = switch_event_get_header(fh->params, "auth_username"))) { + fh->mm.auth_username = switch_core_strdup(fh->memory_pool, val); + } + + if ((val = switch_event_get_header(fh->params, "auth_password"))) { + fh->mm.auth_password = switch_core_strdup(fh->memory_pool, val); + } + if ((val = switch_event_get_header(fh->params, "fps"))) { float ftmp = atof(val); if (ftmp > 0.0f) { (END) I believe that it is the changes in switch_core_file.c that break my utility program, but I can't be sure. Just o add some detail, I have a Digium TCE400 card in my system which handles the G729 conversion. I have no idea of what those auth_username and auth_password parameters are, but that seems to break things for me. Any idea of what I can do to get things working again? I have created a PasteBin with the code of my fs_encode derived utility: https://pastebin.freeswitch.org/view/a3d64c69 It is pretty short. -- Guillermo Ruiz Camauer

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Recording distorted on resume
by Igor Olhovskiy 27 Jun '17

27 Jun '17

Hi! I’m trying to make pause in recording via bind_digit_action record_session and stop_record_session RECORD_APPEND is set to true, format is WAV Problem I have is when I’m listening to recording, first part is distorted completely (sounds like 44kHz is transformed to 8kHz with slowdown and not with resampling) Recording is done via export(nolocal:api_on_answer=uuid_record 5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b start /var/lib/freeswitch/recordings/bw.1st-byte.com/archive/2017/Jun/27/5ae06306-70a5-4e0b-a9ca-cb44ebf0f95b.wav) Than bind_digit_action(local,*2,exec:record_session,/var/lib/freeswitch/recordin… and bind_digit_action(local,*1,exec:stop_record_session,/var/lib/freeswitch/rec… This is a full log of a call https://pastebin.com/4Nn7dhui Freeswitch version is 1.6.18-35-6e79667~64bit Regards, Igor

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The majority of ClueCon 2017 speakers have now been announced
by Anthony Minessale 26 Jun '17

26 Jun '17

ClueCon is the annual celebration of FreeSWITCH and all other open source telephony. It is produced for all 13 years by the FreeSWITCH core team. Our motto is "a conference for developers, by developers" I hope we will see you there in August. Have a look at the lineup for 2017 on our website! https://www.cluecon.com/speakers.html -- Anthony Minessale II ♬ @anthmfs ♬ @FreeSWITCH ♬ ☞ http://freeswitch.org/ ☞ http://cluecon.com/ ☞ http://twitter.com/FreeSWITCH ☞ irc.freenode.net #freeswitch ☞ *http://freeswitch.org/g+ <http://freeswitch.org/g+>* ClueCon Weekly Development Call ☎ sip:888@conference.freeswitch.org ☎ +19193869900 https://www.youtube.com/watch?v=9XXgW34t40s https://www.youtube.com/watch?v=NLaDpGQuZDA

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Refer with replaces - how to configure to support
by Magnus Kelly 26 Jun '17

26 Jun '17

Hello all, I am trying to configure the FS dial plan to fully support "Refer with Replaces" ? Call case is FS establishes the call successfully with an upstream server, but when the user behind the upstream server performs a transfer, a Refer with replaces is generated and sent to FS, FS accepts the request, but post the 202 the expected invite (FS to upstream) in support of replaces does not occur and then the transfer fails. My assumption is I am missing the correct configuration for Refer with Replaces, I have spent some time searching web but not yet found the correct way to handle this. My assumption is FS fully supports "Refer with Replaces" but confirmation and any tips on configuration most welcome. Appreciate any insight. Thanks Magnus

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