April 2021 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Forking with one leg using a redirect server.
by David VILLAUME 30 Apr '21

30 Apr '21

Hello, We are having troubles with following scenario : Freeswitch receives an incoming call and forks to one user (B1) and a sip redirect-server (B2). If the contact inside the redirect (B2) is local, the leg B1 is is cancelled when the 302 reaches the server. d521d2f2-9c69-11eb-aec3-b7e8effe21d8 2021-04-13 17:06:27.569223 [NOTICE] sofia.c:6506 Hangup sofia/external/+33123456789 [CS_CONSUME_MEDIA] [REDIRECTION_TO_NEW_DESTINATION] d521c17c-9c69-11eb-aebe-b7e8effe21d8 2021-04-13 17:06:27.589224 [NOTICE] switch_ivr_originate.c:3601 Hangup sofia/internal-centrex/internaluser@172.16.33.40:5062<mailto:sofia/internal-centrex/internaluser@172.16.33.40:5062> [CS_CONSUME_MEDIA] [ORIGINATOR_CANCEL] This occurs even if manual mode. Does anyone have idea about how to avoid this cancel ? Regards, David

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Re: [Freeswitch-users] disable MOH
by Juan Nino 27 Apr '21

27 Apr '21

I am trying to get freeswitch to ignore re invites and don't play any MOH as >>> the MOH already comes from the PBX, even when disabling it I still see in >>> the logs that is trying to play MOH >>> 2019-02-18 17:24:49.427916 [DEBUG] switch_ivr.c:623 >>> sofia/rtp-dirty/%2B12345@x.x.x.x:5060 Command Execute >>> playback(local_stream://moh) >>> >>> EXECUTE sofia/rtp-dirty/%2B12345@x.x.x.x:5060 >>> playback(local_stream://moh) >>> >>> any idea what has to be done to get freeswitch to continue passing RTP >>> without trying to impose its own? >>> >>> thanks >>> >> >> >

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WSS error and disconnection
by Kakiman Expert 27 Apr '21

27 Apr '21

Hello, I am working on freeSWITCH 1.10.5 with webRTC configured. Im connecting through WSS on port 7443 with SIPJS client and Chrome. The issue seems to be random but when it happens, the websocket is closed. Resulting in websocket disconnected (1006). It is happening very often and today, I had a lot of messages such as "registrar_tport_error() tport error 0 " and then all the users were unregistered. I had to restart FS to get back the possibility to register and make calls. Did you already met this issue ? how could it be solved ? thanks franck

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Multiple DIDs per extension
by John Tuxies 27 Apr '21

27 Apr '21

I got an isolated network and i got Freeswitch in it. I took user 1000 and registered it in an Asterisk server. The same happened for accounts 1001-1003 to different servers. Here is an icon of what i am trying to accomplish [image: ITSP_Scenario.jpg] The network is not connected to Internet or any other Voice service. I would like Freeswitch to act as an ITSP for the Asterisk servers and route in/out numbers. For example Asterisk's_1 server user which has the DID +493087710024 to be able and dial the number +49308771360 and reach another user on Asterisk Server 4. All the communication will go through the Server (Freeswitch). May i have some help on how to configure this thing please?

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(no subject)
by Kakiman Expert 26 Apr '21

26 Apr '21

Hello, I am working on freeSWITCH 1.10.5 with webRTC configured. Im connecting through WSS on port 7443 with SIPJS client and Chrome. The issue seems to be random but when it happens, the websocket is closed. Resulting in websocket disconnected (1006). It is happening very often and today, I had a lot of messages such as "registrar_tport_error() tport error 0 " and then all the users were unregistered. I had to restart FS to get back the possibility to register and make calls. Did you already met this issue ? how could it be solved ? thanks franck

1 0

Github FreeSwitch bug report should have more labels
by mario_fs 26 Apr '21

26 Apr '21

Back in the Jira days, we could specify the type of bug, for instance build error. Now, cannot find a way to categorize a bug. I think this is useful because for instance, a build error may get priority because it does not allow you to continue testing, at least that’s my case. In Jira, I could specify a build error. I found no way to add a label to the type of bug, and assume that is reserved to the developers. I know it can be done because if you look at the bottom of 1145 <https://github.com/signalwire/freeswitch/issues/1145> and click the reference to 75728, you will see that issue has a label of “build failure” (homebrew labels), which I think can then be filtered so only those errors can be found easily. Just a thought, Mario G

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disable music on hold when in a conference?
by David Villasmil 26 Apr '21

26 Apr '21

Hello guys, I have this case where 3 users are in a conference, and if one of the puts the call on hold, the others will hear his usual music on hold. Is there a way of disabling MOH while in a conference? thanks, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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[ANNOUNCE]: sipnagios, a Nagios Plugin to check Call Quality in SIP VoIP (compatible with checkmk, etc)
by Giovanni Maruzzelli 22 Apr '21

22 Apr '21

Hello fellow VoIPers and RTCers, on GitHub there is an early release of sipnagios, opensource. check it out: https://github.com/gmaruzz/sipnagios sipnagios is a Nagios Plugin to check Call Quality in SIP VoIP (compatible with checkmk, etc) sipnagios implements the Nagios plugin API for monitoring and performance data.sipnagios.c is a modification of the original siprtp.c sample in pjproject distribution. Supposedly, it works on Linux, Windows, and anywhere you can compile pjproject on.It makes a call, checks all the various resulting values (mos, rtt, pdd, tta, jitter, packet loss, bytes and packets transferred, and so on). It verifies these values are included into acceptable, warning, or critical ranges.If the call has gone well, sipnagios print performance data for Nagios graphs, and returns 0.If the call fails, or if its measured values are not inside acceptable ranges, it exits with Nagios conventional WARNING or CRITICAL values. mos calculation is scraped from Julien Chavanton work (VoIP Patrol, on GitHub too) I can't even understand :) (merci Julien!) Enjoy! -giovanni -- Sincerely, Giovanni Maruzzelli OpenTelecom.IT cell: +39 347 266 56 18

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Problem getting custom voicemail script to run
by Alex Lake 19 Apr '21

19 Apr '21

FreeSWITCH Version 1.10.6-release-18 running on Debian 10. I need voicemail to run an alternative script when a message is left, and not just send an email. I've changed /etc/freeswitch/autoload_configs/switch.conf.xml to have: <param name="mailer-app" value="/usr/bin/php /usr/local/freeswitch/scripts/processvm.php"/> <param name="mailer-app-args" value=""/> But I don't believe it's even trying to use it. Instead I get entries in the log like this: 2021-04-19 16:49:04.272054 [DEBUG] mod_voicemail.c:2832 Deliver VM to 6378101(a)w.x.y.z 2021-04-19 16:49:04.872034 [DEBUG] switch_utils.c:1308 Emailed file [/tmp/mail.1618850944b82e] to [email(a)domain.com] 2021-04-19 16:49:04.872034 [DEBUG] mod_voicemail.c:3042 Sending message to email(a)domain.com What am I missing?! Can this behaviour be overridden?

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Lua API call delays the packet sending out
by VERMA Vineet (BCS/SPS) 19 Apr '21

19 Apr '21

Dears, I am facing the issue that when REST API triggers the freeswitch I am sending the reply with Following code: stream:write("HTTP/1.1 200 OK\r\n") if (accept == "application/json" or accept == "*/*") then stream:write("Content-Type: application/json\r\n") else stream:write("Content-Type: text/plain\r\n") end stream:write("\r\n") but it seems before packet it out , there is as deplay of 2-3 sec . can you please suggest how to solve this? I already tried stream:shutdown(); and stream:close() but did not helped. Thanks & Regards Vineet Verma This e-mail cannot be used for other purposes than BICS business use.

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