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Re: [Freeswitch-users] Verto Communicator error
by Mike Jerris 24 Nov '20

24 Nov '20

The client is https://github.com/freeswitch/verto-client <https://github.com/freeswitch/verto-client> We are looking for a maintainer, there currently isn’t one. Mike > On Oct 17, 2020, at 5:28 PM, Social Boh via FreeSWITCH-users <freeswitch-users(a)lists.freeswitch.org> wrote: > > > From: Social Boh <social(a)bohboh.info> > Subject: Verto Communicator error > Date: October 17, 2020 at 5:28:29 PM MDT > To: freeswitch-users(a)lists.freeswitch.org > > > Hello, > > this is the problem: > > bower bootstrap-material-design#~0.3.0 not-cached https://github.com/FezVrasta/bootstrap-material-design.git#~0.3.0 > bower bootstrap-material-design#~0.3.0 resolve https://github.com/FezVrasta/bootstrap-material-design.git#~0.3.0 > bower bootstrap-material-design#~0.3.0 ENORESTARGET No tag found that was able to satisfy ~0.3.0 > > the new location for this tool is: https://github.com/mdbootstrap/bootstrap-material-design > > Where can i change the new link? >

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Re: [Freeswitch-users] rtmp is sending as 160x120 inset within 320x240 instead of fullscreen
by David P 23 Nov '20

23 Nov '20

Thanks for your response, Seven. I collected all my info about this problem into https://github.com/signalwire/freeswitch/issues/955#issuecomment-732425630 and replied to you there. On Sun, Nov 22, 2020 at 4:53 PM < freeswitch-users-request(a)lists.freeswitch.org> wrote: > From: Seven Du <dujinfang(a)gmail.com> > To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> > Cc: > Bcc: > Date: Sun, 22 Nov 2020 10:49:51 +0800 > Subject: Re: [Freeswitch-users] rtmp is sending as 160x120 inset within > 320x240 instead of fullscreen > did you happen to set record_concat_video etc ? > > or maybe try conference record >

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Issue: INCOMPATIBLE_DESTINATION / AUDIO RTP REPORTS ERROR [Missing local host]
by Maciej Bylica 22 Nov '20

22 Nov '20

Hi all, I am working on 1.10.5 (today's build) with proxy_media=true configuration set in dialplan config. Around 2-3% of total call attempts are CANCELed with "INCOMPATIBLE_DESTINATION" / "AUDIO RTP REPORTS ERROR: [Missing local host]" notification. I assume that the problem is probably closely connected with codec negotiation, but in my case (proxy_media=true) FS does forward the packets onward and endpoints must agree on the same codec to accept the call. I have compared all the SIP call flows incoming and outgoing (INVITES, 183) and found no clue. Some of the important log snippets: - the call is CANCELed once FS receives 183 and gets first early media packets 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] switch_ivr_originate.c:3801 Sending early media 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:8729 PROXY AUDIO RTP [sofia/outside_1/ 4916222112233(a)10.20.30.20] 10.20.30.6:26502->10.20.30.6:26502 codec: 18 ms: 20 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [ERR] switch_core_media.c:9669 AUDIO RTP REPORTS ERROR: [Missing local host] 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [NOTICE] switch_core_media.c:9670 Hangup sofia/outside_1/4916222112233(a)10.20.30.20 [CS_EXECUTE] [INCOMPATIBLE_DESTINATION] a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3808 sofia/outside_1/4916222112233(a)10.20.30.20 Media Establishment Failed. a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [NOTICE] switch_ivr_originate.c:3810 Hangup sofia/outside_1/3144917111223344 [CS_CONSUME_MEDIA] [INCOMPATIBLE_DESTINATION] 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [DEBUG] switch_ivr_originate.c:3995 Originate Resulted in Error Cause: 88 [INCOMPATIBLE_DESTINATION] a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_channel.c:3565 (sofia/outside_1/3144917111223344) Callstate Change DOWN -> EARLY a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] sofia.c:7339 sofia/outside_1/3144917111223344 skip receive message [PROGRESS_EVENT] (channel is hungup already) 8b93b671-b90e-4a0e-aa73-6510a8b9caaf 2020-11-06 19:58:36.832494 [INFO] mod_dptools.c:3631 Originate Failed. Cause: INCOMPATIBLE_DESTINATION a0f2b4f7-8ecd-435d-b66e-444e4d2c65e6 2020-11-06 19:58:36.832494 [DEBUG] switch_core_media.c:12291 sofia/outside_1/3144917111223344 too late. - incoming (Originator -> FS) SIP INVITE has following SDP v=0 o=- 1604689110 1604689110 IN IP4 10.20.30.6 s=- c=IN IP4 10.20.30.6 t=0 0 m=audio 26502 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv a=silenceSupp:off - - - - - outgoing (FS -> Terminator) SDP of SIP INVITE looks like this (codec transparent) v=0 o=FreeSWITCH 4241000132 4241000133 IN IP4 10.20.30.20 s=FreeSWITCH c=IN IP4 10.20.30.20 t=0 0 m=audio 31034 RTP/AVP 18 0 8 101 a=rtpmap:18 G729/8000 a=fmtp:18 annexb=no a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=silenceSupp:off - - - - - 183 that is received (Terminator -> FS) v=0 o=JOANO_SDP 24930105 0 IN IP4 10.20.30.40 s=JOANO-SIP c=IN IP4 79.133.196.167 t=0 0 m=audio 18200 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=silenceSupp:off - - - - a=ptime:20 As a result FS tears down the call attempt by sending (FS -> Originator) SIP/2.0 488 Not Acceptable Here Via: SIP/2.0/UDP 10.20.30.10:5060 ;received=10.20.30.10;rport=5060;branch=z9hG4bK0609c6993b9c-30-16a290 Via: SIP/2.0/UDP 10.20.30.2:5060;branch=z9hG4bK5765.136934e2.0 Max-Forwards: 66 From: <sip:4916222112233@10.20.30.5:5061;user=phone>;tag=b6b8aec94s To: <sip:3144917111223344@10.20.30.20;user=phone>;tag=DcUe4343Uvj8Q Call-ID: 684d5998-d859dd12-ca30a685-551b1244 CSeq: 1 INVITE User-Agent: FreeSWITCH-mod_sofia/1.10.5-release~64bit Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY Supported: timer, path, replaces Allow-Events: talk, hold, conference, refer Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 and CANCEL (FS -> Terminator) CANCEL sip:3144917111223344@10.20.30.30:5060 SIP/2.0 Via: SIP/2.0/UDP 10.20.30.20:5080;rport;branch=z9hG4bKK01B9mcmpp5jS Max-Forwards: 70 From: <sip:4916222112233@10.20.30.20>;tag=g76r9mQeKQN0a To: <sip:3144917111223344@10.20.30.30:5060> Call-ID: e788fdfd-9b04-1239-6881-fc15b4264fc0 CSeq: 27790763 CANCEL Reason: Q.850;cause=88;text="INCOMPATIBLE_DESTINATION" Content-Length: 0 Could somebody help me to address the issue I am struggling with ? Detailed debugging logs and sip signalization might be found here: https://pastebin.com/G0sjGiw8 - FS logs https://pastebin.com/2Zt6r2mF - SIP logs Many thanks in advance, Maciej

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Bridge 2 conferences
by Joey Golan 22 Nov '20

22 Nov '20

Hello, I have 2 conferences with 50 users on each one of them and one speaker total. I would like to bridge between them so all the 99 will hear the speaker. How can I achieve that? Thanks, Joey.

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Channel flag meanings
by Jason Blood 22 Nov '20

22 Nov '20

Where can I find a description of what the different channel flags mean? I'm referring to the enum switch_channel_flag_t defined in switch_types.h -Jason

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Second incoming call terminates after 32 seconds
by Dion Phillips 22 Nov '20

22 Nov '20

Hi I have Freeswitch setup on a cloud server and Grandstream phones in the office connected successfully to the switch. I have never had any issues with NAT causing the phones or Freeswitch to lose their connection. The phones have "keep-alive" set so are always sending "OPTIONS" messages to Freeswitch to keep the ports open. I don't use the default 5060 port for the internal profile on the Freeswitch side. The office has a Fortigate firewall and a Opnsense box that is used to connect the office to a DC cloud server. The issue is that when a second call comes into the office, it will terminate after 32 seconds. There are only 2 voip lines so max 2 calls at a time. This only occurs if the second call is inbound. What is even more weird is that if the second call is answered by the phone that is already on a call, then this does not happen. If the second call is outbound, there is also no issue. I have done a sip trace and the calls progress correctly from CALL SETUP -> IN CALL -> COMPLETED. The logfile however has an ORIGINATOR_CANCEL message when the call is terminated. I have two phones at my house connected to the same PBX and when I use them to test, I cannot get the call to drop which suggest to me there is something in the office that is dropping packets but only on the second call. If the firewall was an issue, why would the first incoming call work and all outgoing calls work also? I have tried creating a rule on the firewall to allow all traffic from the Freeswitch IP but that make no difference. If the SIP trunk was the issue why would the home phones work. Can anyone give me some pointers on what I should be looking for in a SIP trace or tcpdump or loglfile or tea leaves? Thanks Dion.

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mod_amqp on Mac OS
by Phil Steitz 22 Nov '20

22 Nov '20

Has anyone succeeded at getting mod_amqp built and working on Mac OS? I am trying to get a Mac dev server working to test apps that use RabbitMQ to send and receive FS events. I first tried just enabling the module in a system installed using the MacFI installer. It did not load and showed as not present. Looking at the so's I could see it had not been built. I tried to hack the config in the deployed sources to get it to build (see below), but it was missing the amqp dependencies. Searching the archives, I found the thread starting at [1]. I tried using brew to install rabbitmq-c and then setting package path per suggestions in the responses in [2]. That worked to get get the module built, but I ended up getting a seg fault when I started FS. If I can get some pointers on how to do this correctly, I would be happy to submit a patch to the Mac wiki with instructions on how to do it. The dev server that I am trying to do the build on is OSX 10.15.7. I started by using the MacFI installer, which does not build the module. To get the module built, I modified freeswitch/src/mod/modules.inc to add it. Then config, make, make install after the rabbitmq-c brew install and with | | |PKG_CONFIG_PATH="/usr/local/Cellar/rabbitmq-c/0.10.0/lib/pkgconfig:/usr/local/opt/openssl@1.1/lib/pkgconfig" exported. This works, but the server fails on startup with Segmentation Fault 11.| Thanks! Phil [1] https://lists.freeswitch.org/pipermail/freeswitch-users/2016-July/121346.ht… [2] https://stackoverflow.com/questions/51818515/installing-amqp-on-mac-with-br…

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rtmp is sending as 160x120 inset within 320x240 instead of fullscreen
by David P 22 Nov '20

22 Nov '20

When we configure getUserMedia to send 320x240 (using recent Chrome), and then in our FS 1.10.5 dialplan we forward that as rtmp this way... <action application="record_session" data="{modname=mod_av,vw=320,vh=240}rtmp://mydomain.com"/> ...an ffmpeg capture reveals the user video to be only a 160x120 inset (against the left margin and centered vertically). The rest of the frame is black. I've looked in av.conf.xml, vars.xml, and conference.conf.xml for any setting that might explain this; it seems like a 2-person conference layout (and we are using a conference with a virtual agent with setting 'mintwo' and <param name="max-members" value="2"/> How can we fill the rtmp frames with 320x240 of only the verto user?

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Freeswitch (incorrectly?) reporting packet loss via RTCP
by David Horton 20 Nov '20

20 Nov '20

I have a freeswitch server v1.10.1 running on AWS EC2 which I have noticed is reporting lost packets in the RTCP reports it is sending out. This happens even with a single call. Meanwhile, the network stats on the same instances show no dropped packets and no packet receive errors. Is there anything which could explain this? Have others seen errors in the RTCP stats reported by freeswitch? dave

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'__min': is not a member of 'std' gsmlib
by Gregor Nanger 20 Nov '20

20 Nov '20

I want to build FS in Visual Studio 2019. I retarget projects to new C++ SDK and I get this error: Error C2039 '__min': is not a member of 'std' gsmlib C:\Program Files (x86)\Microsoft Visual Studio\2019\Community\VC\Tools\MSVC\14.28.29333\include\xlocale 2101 Error C2065 '__min': undeclared identifier gsmlib C:\Program Files (x86)\Microsoft Visual Studio\2019\Community\VC\Tools\MSVC\14.28.29333\include\xlocale 2101 Error C2039 '__min': is not a member of 'std' gsmlib C:\Program Files (x86)\Microsoft Visual Studio\2019\Community\VC\Tools\MSVC\14.28.29333\include\xlocale 2303 Error C2065 '__min': undeclared identifier gsmlib C:\Program Files (x86)\Microsoft Visual Studio\2019\Community\VC\Tools\MSVC\14.28.29333\include\xlocale 2303 It has something with including alogorithm.h, but I have lack of knowledge how to solve this problem. Best regards, Gregor

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