May 2017 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Paging
by Rick Jarvis 14 Jun '17

14 Jun '17

Looking to set up paging (not multicast). What’s the best way of achieving this? Specifically, I want to have the receiving handset(s) answer muted for privacy reasons, so it’s literally like a PA system rather than just auto answer…?

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Open Source SIP\WebRTC clients compatible with FreeSWITCH
by Shaun Stokes 09 Jun '17

09 Jun '17

Hi All, Does anyone have any recommendations on a good open source SIP\WebRTC client which works on multiple platforms (Windows, Mac, Linux, Mobiles) to provide presence, voice, video, instant messaging, screen sharing and file sharing? This must be capable of integrating with FreeSWITCH for voice and video (presence via FreeSWITCH would be an advantage). Many Thanks, Shaun [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes(a)itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________

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Monitoring call quality using variable_rtp_audio_xxx
by Jonathan Hunter 07 Jun '17

07 Jun '17

Hi Guys, Sorry for the noise, we are looking to poll calls in progress to grab the rtp_audio variables when a call is in progress, and we want to understand if this is a good approach and to that end what values/variables we should consider and what ranges should we be working with? I understand mos and quality percentage but what other values are a good indicator? Again Im clear on jitter and its meaning just want to understand whats what as not everything understandably is not documented. Many thanks Jon "variable_rtp_audio_recv_pt": "8", "variable_rtp_audio_in_raw_bytes": "374960", "variable_rtp_audio_in_media_bytes": "374616", "variable_rtp_audio_in_packet_count": "2180", "variable_rtp_audio_in_media_packet_count": "2178", "variable_rtp_audio_in_skip_packet_count": "92", "variable_rtp_audio_in_jitter_packet_count": "0", "variable_rtp_audio_in_dtmf_packet_count": "0", "variable_rtp_audio_in_cng_packet_count": "0", "variable_rtp_audio_in_flush_packet_count": "2", "variable_rtp_audio_in_largest_jb_size": "0", "variable_rtp_audio_in_jitter_min_variance": "28.57", "variable_rtp_audio_in_jitter_max_variance": "116.33", "variable_rtp_audio_in_jitter_loss_rate": "0.02", "variable_rtp_audio_in_jitter_burst_rate": "0.98", "variable_rtp_audio_in_mean_interval": "20.39", "variable_rtp_audio_in_flaw_total": "43", "variable_rtp_audio_in_quality_percentage": "97.00", "variable_rtp_audio_in_mos": "4.47", "variable_rtp_audio_out_raw_bytes": "210012", "variable_rtp_audio_out_media_bytes": "210012", "variable_rtp_audio_out_packet_count": "1221", "variable_rtp_audio_out_media_packet_count": "1221", "variable_rtp_audio_out_skip_packet_count": "0", "variable_rtp_audio_out_dtmf_packet_count": "0", "variable_rtp_audio_out_cng_packet_count": "0", "variable_rtp_audio_rtcp_packet_count": "0", "variable_rtp_audio_rtcp_octet_count": "0",

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About proxy-hold codec re-negotiation behavior.
by Lợi Đặng 06 Jun '17

06 Jun '17

Hi list, I'm using FS with *proxy-hold*. Call flow is simple A -> FreeSWITCH -> B Assume FS, A and B support PCMA and PCMU, FS prefer PCMA in codec-prefs profile configuration. In affects of my configuration for *greedy early negotiation* and *disable transcoding*, PCMA is negotiated in both legs for initial INVITE, that's good, and expected. But then I remove PCMA from A and compose a re-INVITE for holding with PCMU only, the a leg is re-negotiated with PCMU, but the b leg have re-INVITE with PCMA due to *proxy-hold* variable, that causes transcoding happened after hold. My desire is to have FS to re-negotiate with b leg too, not so similar but as what I observed when I was using it with media proxy option ( A-A before hold, U-U after hold). I tried late negotiation in this case, but no luck. So is that not able for FS to re-negotiate codec on b leg, in using proxy-hold? Any advise is appreciated. rgds, Loi Dang

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testing 1234
by Ken Rice 02 Jun '17

02 Jun '17

This is just a test

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FS to FS
by Rick Jarvis 01 Jun '17

01 Jun '17

Sorry if this is a dumb question, but it’s something I’ve always struggled with that I’m convinced should be quite straightforward. Getting one FS to register with another as an extension (I know there are other ways of getting FS to talk together, but this is the situation I’m faced with). Box one (the host) is not behind NAT. Box two (the client) IS behind NAT. In my mind it should be good enough to have a directory entry on box one, and a SIP profile on box two, then something like a catchall dial plan entry on box 2 to accept the calls made to the ‘extension’ it’s registered as and handle the call accordingly. But I don’t think it’s that simple? I’ve looked through the soft phone configuration, which suggests using a directory entry with the register flag set to true, but this doesn’t seem to help me. Thanks R

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originated_legs
by Tihomir Culjaga 01 Jun '17

01 Jun '17

Hello, There is something that is bugging me pretty hard and i need to understand how variable_originated_legs in CHANNEL_HANGUP gets populated ? example: variable_originated_legs: ARRAY::faf71922-4526-11e7-b0f4-9fb65fdc7341;Outbound Call;888|:faf71922-4526-11e7-b0f4-9fb65fdc7341;Outbound Call;888|:faf76c56-4526-11e7-b0fd-9fb65fdc7341;Outbound Call;0916331550", the reason :=) i have a scenario like this: 1. Incoming calls (IN_CALL) are sent to PARK application 2. On CHANNEL_PARK i originate a call to an extension say EXT1 e.g. originate {ogirination_uuid=<ext1_uuid>}user/1002 & park() ... and eventually i do uuid_bridge <in_call_uuid> <ext1_uuid> 3. EXT1 makes a call transfer to EXT2 (using att_xfer) and hangs up the call 4. on CHANNEL_HANGUP from EXT1 i learn EXT2 uuid and.,,, IN_CALL is talking to EXT2 and we are happy :=) ... this is all good if the transfer destination is a single extension (i get enough info from signal_bond etc...). But if i have multiple destination behind EXT2 e.g. user/1002,user/1003,sofia/gateway/gw1/012345 the thing gets complicated. I was hoping i could exploit variable_originated_leg in CHANNEL_HANGUP from EXT1 to learn all origination legs uuid after an att_xfer to a multiple destination. is that feasible ? I mean, will this variable contain all UUIDs att_xfer generated on a multi-destination transfer ? please i would appreciate any heads up here :=) thanks, Tihomir.

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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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