- FreeSWITCH-users - FreeSWITCH Mailing Lists

ICE -> FreeSWITCH tries exclusively the first candidate
by Alexander Haugg 20 Aug '21

20 Aug '21

Hi, currently I have the problem that the FreeSWITCH only serves the first candidate from the candidate list via STUN request. I expected it to try to reach all candidates relatively soon (trickle). In the source code I saw that there are four ICE types: ICE_GOOGLE_JINGLE ICE_VANILLA ICE_CONTROLLED ICE_LITE What is behind them, and how can I select them from the "outside"? Or am I on the wrong track? What to do, that the FreeSwitch serves the candidates according to Trickle? I have the same result for mod_sofia and mod_verto (is clear, they use the same media stack). Thanks and greetings Alex

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Freeswitch doesn't start
by William Colls 19 Aug '21

19 Aug '21

Debian Buster on a pine64-lsb board. Trying to build from source current stable version. Everything seems to compile and install correctly, but when try to start freeswitch, it fails with the message: /usr/bin/freeswitch: error while loading shared libraries: libspandsp.so.3: cannot open shared object file: No such file or directory But the file does exist in /usr/local/lib. A link in /usr/local/freeswitch/lib points to /usr/local/lib/libspandsp.so.3 Both are owned by freeswitch:freeswitch. What am I missing? Thanks for your time. William

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inbound T.38 re-invite transaction stalls
by Jeff Pyle 19 Aug '21

19 Aug '21

Hello, This is on 1.10.6-release-18-1ff9d0a60e~64bit, the latest release version from the repo. I run the following API command to send a TIFF file with txfax: originate {sip_h_P-CustomHeader=value,ignore_early_media=true,fax_ident='Fax Test',origination_caller_id_name='Fax Test',origination_caller_id_number=2165551111,absolute_codec_string=PCMU@20i ,fax_verbose=true,fax_enable_t38=yes,fax_enable_t38_request=yes,fax_use_ecm=no,fax_end_page=1,jitterbuffer_msec=200:200:200}sofia/external/ 4045551111@1.2.3.4:5060 &txfax(/usr/local/etc/faxtiffs/smiley-fine.tif) The INVITE goes out as expected, and the call connects normally. One second later, the far side sends a re-invite for T.38. Local FreeSWITCH shows the remote SDP on the console, sends a 100 Trying, and this appears on the console: [DEBUG] switch_core_media.c:5192 sofia/external/4045551111@1.2.3.4:5060 T38 ACCEPT on request [DEBUG] switch_core_media.c:5296 sofia/external/4045551111@1.2.3.4:5060 T38 IS POSSIBLE on request And then nothing. No 200 OK, 488 Not Acceptable Here, nothing past the 100 Trying. I can't tell why. spandsp.conf.xml is vanilla with only ident, header and spool-dir updated. The environment is relatively simple with all static public IPs, no NAT, etc. This particular FS instance serves only as an endpoint for testing like this fax example. This configuration has been working in general for years and specifically on this box for almost a year. Apparently something has changed somewhere but I can't see what. I'm hoping someone might be able to point me in a direction to diagnose and solve this. Regards, Jeff

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timeout/inter-digit-timeout not working on IVR??
by Avi Marcus 18 Aug '21

18 Aug '21

I've got a weird IVR situation. I just went from 8 options to 10, so I bumped the digit-len to 2. On my phones, it's fine -- inter-digit-timeout kicks in when I dial only one digit. But I see someone else calling in and entering one digit and it just hangs. The timeout doesn't trigger, the inter-digit-timeout doesn't trigger... Any ideas? I can't see any reason for a difference in the logs... Not working log: https://pastebin.freeswitch.org/view/479f4fc3 Working log: https://pastebin.freeswitch.org/view/dc046d01 I'm not sure how old this is: FreeSWITCH Version 1.10.5-release-17-25569c1631~64bit (-release-17-25569c1631 64bit) but I can't simply reboot and kill live traffic to test it. The hard part is it's not failing from my phone so I can't reproduce at will. Thanks for any help! -Avi Marcus

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ICE -> FreeSWITCH tries exclusively the first candidate
by Alexander Haugg 13 Aug '21

13 Aug '21

Hi, currently I have the problem that the FreeSWITCH only serves the first candidate from the candidate list via STUN request. I expected it to try to reach all candidates relatively soon (trickle). In the source code I saw that there are four ICE types: ICE_GOOGLE_JINGLE ICE_VANILLA ICE_CONTROLLED ICE_LITE What is behind them, and how can I select them from the "outside"? Or am I on the wrong track? What to do, that the FreeSwitch serves the candidates according to Trickle? I have the same result for mod_sofia and mod_verto (is clear, they use the same media stack). Thanks and greetings Alex

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B-Leg Question
by Alexander Perkins 10 Aug '21

10 Aug '21

Hi All. I have a situation where I need to transfer the b-leg of a call to an IVR, or to a CS agent, even if the A leg hangs up. I am thinking I'd need to use the execute_after_hangup? Is that accurate? Can somebody point me in the right direction? Any help would be appreciated. Thank you, Alex

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Can calls be rejected if they are missing EITHER audio or video USABLE ice candidates?
by David P 09 Aug '21

09 Aug '21

We're evaluating a paid TURN service, and we've found that sometimes it provides only a STUN candidate for audio and neither STUN nor TURN for video. The user's host candidates are all private IPv4. FS 10.5 allows such a call, but we'd like to reject any call *lacking* usable ICE candidates for *either* audio or video. I see at https://freeswitch.org/confluence/display/FREESWITCH/Channel+Variables that there is channel var "variable_remote_media_ip", so if this is empty for both audio and video, we could set our dialplan to reject the call. But is this var set independently for audio and video so we can check that both are present? Or is there a better way?

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ICTAgent Ver 0.6 released, A chrome extension based web phone
by Tahir Almas 07 Aug '21

07 Aug '21

Just released a new version of ICTAgent with voice calling and Fax support using ICTCore as backend communications framework also tested with any other gateway. Here is project link https://github.com/ictinnovations/ictagent regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT **************************************************************************************************************** NOTICE OF CONFIDENTIALITY This communication including any information transmitted with it is intended only for the use of the addressees and is confidential and may be protected by legal privilege . If you are not an intended recipient, be aware that any disclosure, copying, distribution or use of this e-mail or any attachment is prohibited. If you have received this e-mail in error, please notify us immediately by returning it to the sender and delete this copy from your system. Thank you for your cooperation.

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Handling temporary WS breakages more gracefully
by David P 26 Jul '21

26 Jul '21

When there is a temporary break in verto's WSS login, it does verto.purge() and that seems to force a new createOffer/createAnswer negotiation and re-execution of the dialplan if the connection is re-established. Is there a reason why renegotiation of the media connections is necessary? In our case we've tried to minimize disruptions to the callers by using verto's support for reattaching/recovering sessions, but that seems to complicate things when the dialplan is re-executed...it seems to result in simultaneous verto calls with the same users. And some of these sessions seem to trigger media_timeout despite there being media (resulting in sudden loss of media in recordings). Does it seem that I'm interpreting things consistently with the design of verto and FS? If so, then it seems that we should disable the verto.purge() so the current ICE connections are kept and the dialplan is not re-executed, and rely on media_timeout to detect abandoned sessions.

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Verto WebRTC
by Praveenkumar T 23 Jul '21

23 Jul '21

Dear All, I am using Verto (WebRTC) as well as other standard sip phone which can establish normal extension calls b/w each other, however I notice in the fs_cli <show registrations> it's not listing the extension ID for the one which are logged/registered via Verto, where as I can see the extension registrations listed in case of other sip phone. Please confirm if any configuration to be done in order to list in show registrations (to list the extensions which are registered via Verto). Waiting for your kind response. Thanks & Regards Praveen

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