- FreeSWITCH-users - FreeSWITCH Mailing Lists

Is there any archive or list of topics discussed
by Hemrajsinh Gharia 06 Jun '20

06 Jun '20

Hello guys, I am new to FreeSWITCH and landed in this community group, which is the mailing list. I am kind of disappointed not finding easy to use google group for FreeSWITCH. There are Google groups for other WebRTC, like discuss-webrtc (general group), kurento, and even freeswitch-ru (Russian version). So is there such a group where I can go and see/search the previous discussion before posting anything? It is handy, isn't it? Thanks, Hemrajsinh Gharia https://ghariaonline.wordpress.com/ <https://github.com/gharia/> <https://www.linkedin.com/in/hemraj-gharia-53597769> <https://www.facebook.com/hemraj.gharia>

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[Announcement] SIP Chronicles #4, featuring Mark J. Crane @ FusionPBX
by Maxim Sobolev 06 Jun '20

06 Jun '20

SIP Folks, First of all thanks everyone who participated in or just watched our first 3 episodes of SIP Chronicles! I think it's safe to say that both myself and my partner in crime Giovanni have been pleasantly surprised by the turn-around numbers and general interest our little initiative has generated! We have had time to refine and think over our concept and if you ask me now, SIP Chronicles is not about a particular product, or technology - it's first and foremost about stories of amazing, passionate people doing exciting things with SIP technology! In that spirit, this Saturday we'll bring another story: the story of Mark J. Crane and his FusionPBX project. Even if you have never heard about FusionPBX or FreeSWITCH (which is unlikely) that might be a great opportunity to learn and hear Mark's story. Or if you are using either one already - ask Mark a question! Please join us this Saturday @ 4:30pm UTC: https://youtu.be/CJB7RjiIJ4s Thanks and see you soon! :) -Max

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Jitterbuffer documentation "https://freeswitch.org/confluence/display/FREESWITCH/jitterbuffer_msec"
by Alexander Haugg 05 Jun '20

05 Jun '20

Hi, is it possible, the "max drift" parameter is obsolete? I was looking into the switch_core_media.c in the function check_jb. I can't see that the "max drift" parameter have any effect. Is this correct? Thanks a lot Alex

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why FS respons BYE to Transferor afer REFER
by Yuriy Nasida 04 Jun '20

04 Jun '20

Guys, do you know why FS respons with 202/NOTIFY/BYE to REFER ? In my opinioin FS should not sends BYE to Transferor at once. That is very weird because (after refer) FS sends INVITE to Transfer Target (like it should be) and this part of call is fine. Why FS sends BYE to Transferor... I checked rcf5589 / rfc3892 and Transferee (FS) never does so. In fs_cli FS just said (after new leg bridge is done) that last dialplan instruction is done, hung up. Please advice.

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Permission to update FreeSwitch doc
by Sergey Safarov 03 Jun '20

03 Jun '20

Hi Could you provide permission for Konstantin Markelov to update FreeSwitch confluence documentation. Username: konmar89 Full name: Konstantin Email: konmar89(a)gmail.com Sergey Safarov

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SIPP Test scenario for REPLACES?
by David Villasmil 02 Jun '20

02 Jun '20

Hello guys, I'm trying to test a REPLACES scenario, but don't have the right scenarios... Would anyone by any chance have a SIPP scenario xml file i can use? I'd appreciate it! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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Remove REPLACES from Supported?
by David Villasmil 02 Jun '20

02 Jun '20

Is this possible? Regards, David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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outbound calls from one freeswitch to another
by Joli Martinez 01 Jun '20

01 Jun '20

Hello, I have two freeswitch boxes. They are in two different locations, but both are connected through a VPN. I am looking to send all calls from one freeswitch to another. I was able to get inbound routing working with each other. The problem is when freeswitch1 wants to make an outbound call to the PSTN out of freeswitch2. What do I have tried to add the IP 10.41.100.2 to the ACL and that did not work. What am I missing? 2020-05-27 10:50:37.774914 [WARNING] sofia_reg.c:2906 Can't find user [ FreeSWITCH(a)10.21.10.4] from 10.41.100.2 You must define a domain called '10.21.10.4' in your directory and add a user with the id="FreeSWITCH" attribute and you must configure your device to use the proper domain in it's authentication credentials. send 678 bytes to udp/[10.41.100.2]:5080 at 10:50:37.773507: ------------------------------------------------------------------------ SIP/2.0 403 Forbidden Via: SIP/2.0/UDP 10.41.100.2:5080;rport=5080;branch=z9hG4bKjr90SBDZmUQKj From: "CallerID" <sip:1112223333@10.41.100.2>;tag=90SHD9m8r6X2N To: <sip:4445556666@10.21.10.4>;tag=rmB10y0mDg5yD Call-ID: 44b8eb5a-1acc-1239-f884-005056873da3 CSeq: 20741727 INVITE User-Agent: FreeSWITCH Accept: application/sdp Allow: INVITE, ACK, BYE, CANCEL, OPTIONS, MESSAGE, INFO, UPDATE, REGISTER, REFER, NOTIFY, PUBLISH, SUBSCRIBE Supported: timer, path, replaces Allow-Events: talk, hold, conference, presence, as-feature-event, dialog, line-seize, call-info, sla, include-session-description, presence.winfo, message-summary, refer Content-Length: 0 ------------------------------------------------------------------------ 2020-05-27 10:50:37.774914 [NOTICE] sofia.c:2332 Hangup sofia/internal/ 1112223333(a)10.41.100.2 [CS_NEW] [CALL_REJECTED]

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Mod Call Center Behavior
by Vishal Pai 01 Jun '20

01 Jun '20

Hello Everyone I am using call center module for multiple tenant but when i use command callcenter_config queue reload [queue_name] it reload/update the status to default assigned to agents. Is it possible to reload the particular queue of tenant not all queues in call center. Thanks V.P

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playback to soundcard TML instead ringfile with portaudio
by Tamer Higazi 01 Jun '20

01 Jun '20

Hi people, I am using on my linux system a lot portaudio to place and make phone calls. Inside the portaudio.conf I am using: <param name="ring-file" value="/opt/freeswitch10/sounds/ring_dd.wav"/> that if a phone call comes in it respectively rings on my headset. Instead the file, i'd like freeswitch to playback tones via TML: https://freeswitch.org/confluence/display/FREESWITCH/TGML now what I want is to disable the ring-file and make freeswitch in the dialplan playback to myself (the soundcard) for example this UK ring: <action application="playback" data="tone_stream://%(400,200,400,450);%(400,2200,400,450)"/> not the callee should get this ring, freeswitch should playback this sequence before I execute "pa answer". Any ideas ?

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