- FreeSWITCH-users - FreeSWITCH Mailing Lists

FreeSWITCH Stops Responding
by Cary Shouse 06 Oct '21

06 Oct '21

We recently started having a problem with our FreeSWITCH servers where calls are working well, with about 200 calls operating sometimes for a couple of days, and then something happens and all call progress stops. Then any new inbound calls that come into the server get stuck in initial status, new inbound get stuck at RINGING. This happens until the max sessions are hit and of course nothing else works from there. Our thought is something is creating a lock and stopping anything else from progressing. We are seeing the stuck sessions using "show channels". Doing a uuid_kill on the calls seems to make no difference to the output of "show channels", but we have noticed in the logs that some events do happen to some of the calls. We had ran into a similar issue when writing a custom C module and made some mistakes with session_rwlock etc. But in this case, all of this is using LUA. Some general information: FreeSWITCH Versions 1.10.6 and 1.10.1 Modules used: LUA, xml_curl(dialplan), event_socket, Most of this is just application playing audio files and or creating a new call to bridge to. Really looking for some help on what debugging we can enable to help identify what is getting stuck to identify where a bug might be on our usage or possible issue with FreeSWITCH. We have captured a core dump of the running process when it got into the bad state as well, if that is helpful, we can send it in. Thank you for your help.

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jitters generated from freeswitch to trunk in conference call
by Achintha 06 Oct '21

06 Oct '21

Hi all, I configured 2 freeswitch servers (FreeSWITCH version: 1.10.5-release-17-25569c1631~64bit) on Debian 10.4 as media servers. one opensips server is located in front of the freeswitch servers and opensips act as SIP load balancer. media directly connect with sip trunk and freeswitch.We use dynamic dialplan to generate calls and use g711. Anyway i generate below mentioned scenario 1. registered as extension 2. dial outbound to mobile call through the sip trunk and talk 3. put hold the first call 4. dial another outbound call to another mobile through the same sip trunk and talk. 5. connect both calls (conference). in the 5th step jitter buffers generated from freeswitch to trunk. voice not clear. I have changed codecs and tests (opus,G711a,G711u) but the issue is not fixed. how to solve this problem. Best Regards.. Achintha Udukumbura

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stop channel from entering conference
by David Villasmil 05 Oct '21

05 Oct '21

Hello all, I have a scenario where: A calls B (i.e.: +19995551212) and puts it on hold. A calls C (i.e.: +99911111) and when it answers: The client sends a refer to B to connect to a conference, and then transfers C to the conference. All this works well, but i need to block specific called numbers from entering the conference. I can identify the channels as they come in and i'm trying to hangup the leg i want. I tried with <action application="hangup"/> and also with <action application="set" data="res=${uuid_kill(${uuid})}"/> but all channels are disconnected. Not sure if i'm doing something wrong. Any help would be appreciated Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Multiple api_on_answer?
by Avi Marcus 05 Oct '21

05 Oct '21

Docs show that you can have multiple execute_on, specifically: execute_on_answer_1 Does that also work for api_on_answer? I searched the code for execute_on_answer_1 and couldn't find that or a loop for 1-9. https://freeswitch.org/confluence/pages/viewpage.action?pageId=1048892#id-_… Thanks, -Avi Marcus

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ios 15 issue
by Nathan Stratton 05 Oct '21

05 Oct '21

We recently encountered an issue where all socket connection attempts are repeatedly disconnected and retried upon attempting to connect to FreeSWITCH. Freeswitch does not contain logs of these connection attempts and they seem to occur immediately upon the initial request (likely during the SSL handshake). A look at the Apple developer forums has yielded a number of people experiencing the same issue, pointing to a problem with a transition to a new NSURLSession implementation in ios15. Unfortunately, there is no expected timeline for a fix to address these issues. This is the link to the forum thread relevant to the problem: https://developer.apple.com/forums/thread/685403 ><> nathan stratton

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system call inside lua
by Mehdi Shirazi 04 Oct '21

04 Oct '21

Hi I can run this query on linux cli: mssql-cli -S 192.168.69.60 -d IVR_SH -U IVRSH1 -P IVRSH -Q "SELECT * FROM "dbo"."IVR_REQEST" " but the same query inside lua script with os.execute does not work. Itried: -os.execute(' mssql-cli -S 192.168.69.60 -d IVR_SH -U IVRSH1 -P IVRSH -Q "SELECT * FROM "dbo"."IVR_REQEST" " '); -os.execute([[ mssql-cli -S 192.168.69.60 -d IVR_SH -U IVRSH1 -P IVRSH -Q "SELECT * FROM "dbo"."IVR_REQEST" " ]]); ... Please help. Regards M.Shirazi

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Re: [Freeswitch-users] iOS 15 blocking WSS login on port 8082
by David P 01 Oct '21

01 Oct '21

> > Good to know, Chris. It's hard to tell where the official repos are for mod_verto and its js library in github. Would you mind providing a link? From: cmrienzo(a)gmail.com > > This is fixed in latest mod_verto. > > Chris >

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iOS 15 blocking WSS login on port 8082
by David P 30 Sep '21

30 Sep '21

When testing with an iPhone running iOS 14.8, there is no problem with verto logging-into FS 10.5 via WSS to port 8082. But when the same device is upgraded to iOS 15 and used in the same wifi network hours later, verto repeatedly gets "Socket is not connected" and there is no sign of the request in the FS log at debug level. This happens even with a phone that is not enrolled in the new "Private Relay" feature of iOS 15. Anyone else seeing this? I don't see any mention on tech sites, and I suspect it involves the port which is a default for verto.

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Outbound event_socket memory usage
by Adrien Martin 30 Sep '21

30 Sep '21

Hello, I'm currently benchmarking an ESL program that use both outbound and inbound mode and notice a quite large memory usage for the FreeSWITCH process compared to a dialplan that has similar actions (8 set (half multiset), 1 sched_hangup, 1 bridge). The FreeSWITCH process use about 2 Mo per call and about 180 Ko without ESL. This does not look like a leak, the memory usage is stable and proportional to the current calls, Valgrind does not show anything special either. What was tested: - Pure dialplan while running only the inbound ESL subscribing to some events (CHANNEL_HANGUP, CHANNEL_HANGUP_COMPLETE, CHANNEL_EXECUTE_COMPLETE, SESSION_HEARTBEAT) for all channels, inbound ESL does not increase memory consumption, - Outbound ESL that just close the connection at the beginning of the dialplan, same memory usage as a complete ESL program, - Outbound ESL linger set/not set, no change, - Outbound ESL parameters async/full present or not, no change, - Versions 1.10.5~release~17 and 1.10.6~release~18, no change. Does this kind of resource penalty is to be expected? What could cause these differences? For those using outbound ESL, how much memory does your FreeSWITCH use per call?

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SRTP negotiation on reinvite
by Alberto 30 Sep '21

30 Sep '21

Hi, I have 2 end users, connected to freeswitch and configured with optional SRTP. Both users can make and receive encrypted calls, but when one of them sends a reinvite to make a transfer, freeswitch replies with a 488. I've spotted these lines in the logs: 2021-09-15 14:37:45.880223 [WARNING] switch_core_media.c:4331 Declining invite with only SAVP because secure media is administratively disabled 2021-09-15 14:37:45.880223 [DEBUG] sofia.c:8007 Reinvite resulted in codec negotiation failure. What seems strange to me is that users are configured to use PCMA only, and are both able to use SRTP, so why would the codec negotiation fail? How do I enable optional SRTP for re-invites? Thanks

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