November 2022 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Building FreeSwitch with OpenSSL3 (Ubuntu 22)
by Yehavi Bourvine 02 Dec '22

02 Dec '22

Hello, I've done GIT CLONE today and tried to compile. It fails on OpenSSL3 changes. According to issue 1762 this has been fixed, but probably was not committed yet (sorry, I am not a GIT expert, so maybe I am using the wrong term). How can I get the updated code? Thanks, __Yehavi:

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critical error max queue size reached
by Kakiman Expert 30 Nov '22

30 Nov '22

Hello, on my freeswitch 1.10.6-release-18 , i have a lot of critical error in the freeswitch.log, and that's saturate the disk partition and stops freeswitch do you know how to solve this critical issue ? here is the error message i receive on freeswitch [CRIT] mod_event_socket.c:188 Log enqueue ERROR [11] | [Resource temporarily unavailable] Queue size: [100000/100000] Max queue size reached thanks

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Zoom Can't Render H.264 from Freeswitch 1.10
by ROBERT KIRNUM 30 Nov '22

30 Nov '22

Having trouble with transitioning from Freeswitch 1.6 to 1.10. When attempting to connect to a Zoom client the video from Freeswitch 1.10 fails to render in Zoom. Using Freeswitch 1.6 works as expected. The video from the Zoom client works fine on both cases. After much analysis of packet captures, the H.264 RTP sent to Zoom looks fine. It can be extracted from a packet capture and played using VLC. The RTP is being sent to the correct address / port, the payload type is correct as per SDP negotiation, same for packetization-mode. The stream sent to Zoom contains IDR frames preceded by SPS / PPS / SEI and is consistent with the stream from Freeswitch 1.6. Zoom sends frequent FIR and PLI requests as though it is not receiving any video / key frames. The capture shows no sequence number gaps, no jitter, no clock drift, etc. Has anyone experienced this? Any suggestions on what could possibly cause this? Thanks, Bob

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processing redirect
by Brian :: 28 Nov '22

28 Nov '22

Hello :) I have a particular scenario where a proxy replies with a 302 redirect to Freeswitch. In the 302 they add an X-header: with-some-info I have tried a few things but I can't seem to access this X-header from my redirected context. If I run the info app there it doesn't appear at all. And it doesn't get sent to the new host either. Is there any way to access this header? fs processes the redirect just fine but I could really do with accessing the info in this header in my redirected context. Thank you!

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extension forced public context if IP is in ACL
by Josh 26 Nov '22

26 Nov '22

Greetings, I came across a recent thread http://lists.freeswitch.org/pipermail/freeswitch-users/2022-November/136043… discussing similar problems but it is not clear what solution is. I my case incoming Gateway and ATA adapter, logged in to an Extension, are on the same IP address. I added my IP to an ACL to allow incoming calls from Gateway to an Extension. No question about this part. However, exactly as in above mentioned thread, attempt to make outgoing call fails because Extension IP matches ACL and is forced into public context instead of domain context. If I delete ACL then outgoing calls work but incoming rejected. Is there a solution for above scenario? For example, is it possible to configure incoming Gateway without a need for ACL? Josh. PS. Since I am using FusionPBX all configuration appears to be in postgresql db tables. PPS. Explicitly setting up domain context to an Extension in FusionPBX UI makes no difference whatsoever.

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Conflicting settings in SIP profile in vanilla demo
by TTT 26 Nov '22

26 Nov '22

I am working with a simple FreeSWITCH installation, with the vanilla demo configuration. I see that the Internal SIP profile contains: <param name="apply-inbound-acl" value="domains"/> <param name="auth-calls" value="$${internal_auth_calls}"/> The first line means that if a caller's IP is on the 'domains' list that they do NOT need to authenticate. The second line means that in order to use this SIP profile the user MUST be authenticated (internal_auth_calls is true). Aren't these two lines contradictory? Why allow a user to not authenticate for this SIP profile, and then say they must authenticate to use this SIP profile?

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maximum cps limit of freeswitch server
by Tahir Almas Dhesi 25 Nov '22

25 Nov '22

what is the maximum cps limit of a good freeswitch node (single node ) regards *Tahir Almas* Managing Partner ICT Innovations http://www.ictinnovations.com Leveraging open source in ICT

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Originate length issue via fs_cli
by David VILLAUME 24 Nov '22

24 Nov '22

Hello, I encounter a problem trying to trigger some originate via fs_cli while they work with my dial plan. It seems related to the length of call_url + extension : In example : This works fine: originate {test="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test2="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test3="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test4="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test5="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test6="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test7="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test8="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test9="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test10="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test11="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test12="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test13="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test14="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test15="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test16="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test17="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test18="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test19="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test20="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test21="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa"}sofia/activities/a@a.com;fs_path=sip:172.16.20.100:5080 &echo But if I add one char : originate {test="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test2="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test3="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test4="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test5="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test6="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test7="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test8="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test9="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test10="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test11="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test12="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test13="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test14="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test15="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test16="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test17="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test18="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test19="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test20="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa",test21="aaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaaa"}sofia/activities/a@a.com;fs_path=sip:172.16.20.100:5080 &echo I got a message : -USAGE: <call url> <exten>|&<application_name>(<app_args>) [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>] I tried to look at source code but cannot identify anything. Is that something you also reproduce ? Regards, David

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firefox and vpn with freeswitch
by Kakiman Expert 23 Nov '22

23 Nov '22

Hello, i am working with sip.js and freeswitch 1.10.6. On "end user" computer, when working from home, the VPN is connected and all the flows go through the VPN. With Firefox, and the VPN connected, it takes 60 seconds to have 2 ways audio whereas without VPN and chrome with/without VPN, the 2 ways audio is immediately establish. Is there any parameter/config that helps to have Firefox+VPN works ? thanks

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Xml-handler python
by Adrian Fretwell 23 Nov '22

23 Nov '22

Hello, Does anyone have a working xml handler written in python? I am trying to make one work, but the handler gets called many times with no meaningful arguments, I'm just wondering if I'm missing something obvious... Kind regards, Adrian Fretwell

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