June 2022 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Sharing session between 2 freeswitch
by Patrick Karton 12 Jun '22

12 Jun '22

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Re: [Freeswitch-users] debian repos for Raspberry Pi broken?
by Mohammad-Reza Tazari 10 Jun '22

10 Jun '22

Dear Andrey, any news here? Thanks and kind regards, -- Saied Andrey Volk wrote on 21-Dec-2021 21:36: > > Let me see what I can do. May take some time. > > вт, 21 дек. 2021 г. в 23:19, Johan Helsingius <julf at julf.com <http://lists.freeswitch.org/mailman/listinfo/freeswitch-users>>: > > > >* Hi, *> >> >* Trying to install freeswitch on a RPi 4 (Debian 11 Bullseye), *> >* my /etc/apt/sources.list.d/freeswitch.list has: *> >> >* deb http://files.freeswitch.org/repo/deb/rpi/debian-release/ <http://files.freeswitch.org/repo/deb/rpi/debian-release/> bullseye main *> >* deb-src http://files.freeswitch.org/repo/deb/rpi/debian-release/ <http://files.freeswitch.org/repo/deb/rpi/debian-release/> bullseye *> >* main *> >> >* but when I try *> >> >* apt-get update && apt-get install -y freeswitch-meta-all *> >> >* I get *> >> >* Unable to locate package freeswitch-meta-all *> >> >* apt-cache search freeswitch only gives me: *> >* libsipwitch-dev - secure peer-to-peer SIP VoIP server - development files *> >* libsipwitch1 - secure peer-to-peer SIP VoIP server - shared libraries *> >* libsipwitch1-dbg - secure peer-to-peer SIP VoIP server - debug symbols *> >* sipwitch - secure peer-to-peer VoIP server for the SIP protocol *> >* sipwitch-cgi - secure peer-to-peer SIP VoIP server - CGI XML-RPC interface *> >> >* Any suggestions? *> >> >* Julf *

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Mod_callcenter scalability
by Peiffer Eric 08 Jun '22

08 Jun '22

Hi, I'm new in freeswitch world, My goals is to study a call center system using freeswitch. System must be scalable in order to add new freeswitch instance. To do that I have created 2 freeswitch instances loadbalanced by opensips. The 2 freeswitch instances used the same postgres datasbase in order to shared configuration data. Xml files for directory, dialplan, config, etc .. are also the same. So I have one agent associated to one queue configured in the shared postgres database. When user call the callcenter if the agent is free then the call is offered to this agent. If a second user call occurs (call handled by the other fresswitch instance) this call is queued. So every think is working. Then I do not understand why call center is not scalable. If there is issue with callcenter scalablity please could you tell me what? and what would be the solution? Regards. Eric Peiffer

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force SIP INFO
by David Villasmil 06 Jun '22

06 Jun '22

Anyway to force SIP INFO on the outbound side even if telephone event is present? Thanks! David Villasmil email: david.villasmil.work(a)gmail.com phone: +34669448337

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freeswitch logs don't want to write when call setVariable() and Play TTS prompt
by Rajnikant Vanza 06 Jun '22

06 Jun '22

Hi Users, We have one scenario, in which we don't want to write any logs when calling set variable function during active call and playing TTS prompt (Refer highlighted string) 86e386f6-ff4c-4fa7-9b53-52836aa8e150 2022-06-06 06:53:19.530899 98.77% [DEBUG] switch_cpp.cpp:773 CoreSession::*setVariable(customerId, 123654789258)* ----------- 2022-06-06 07:09:06.830902 99.03% [INFO] mrcp_client_connection.c:530 (TTS-6) Send MRCPv2 Data 172.28.230.145:41052 <-> 172.28.199.43:20032 [304 bytes] MRCP/2.0 304 SPEAK 1 Channel-Identifier: 8de38aa0-6053-4b00-9@speechsynth Content-Type: text/plain Speech-Language: en-us Voice-Name: Jackie Vendor-Specific-Parameters: tts-engine=unimrcp:lumenvox;tts-voice=Jackie Content-Length: 62 *<say-as interpret-as="vxml:digits"> 147258369456 </say-as>* 2022-06-06 07:09:06.910901 99.03% [INFO] mrcp_client_connection.c:635 () Receive MRCPv2 Data 172.28.230.145:41052 <-> 172.28.199.43:20032 [125 bytes] Thank you in advance. Best Regards, Rajnikant Vanza

1 0

Extending mod_esf
by Support from NetworkedAudio LLC 02 Jun '22

02 Jun '22

I’m looking to do multicast send and receive as a module, I’ve looked at mod_esf and also using gstreamer but a mod_portaudio style module would really help. Looking through this list, there have been a few mentions, and a mod_rtp_stream that was begun – I have time and money to put into this now, does anyone know of any projects that already add multicast RTP send and receive capabilities to FreeSWITCH rather than starting fresh?

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Freeswitch on Kubernetes
by Mahmood Alkhalil 02 Jun '22

02 Jun '22

Would like to hear from anyone who managed to run Freeswitch on Kubernetes ? how did u setup the pod specs ? how did you expose the pod to public access ?

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User Variables from directory not accessible
by Örn Arnarson 02 Jun '22

02 Jun '22

Hello guys, I have a bit of a conundrum; I recently upgraded from FreeSWITCH 1.6.12 to 1.10.6. I copied most of the configuration over as-is and most everything works as expected. However, any variables set in the user directory that were previously accessible in the dialplan, such as accountcode and other custom variables that I set are no longer accessible from the dialplan when users originate calls. I can access them via the user_data function as a workaround, but I am struggling to understand why I need to do this. That is, I can access them like so: ${user_data(${caller_id_number}@${domain_name} var accountcode)} But not like so: ${accountcode} In the previous version (and on another FreeSWITCH installation of the same version), the latter is enough. Can anyone think of a reason why this would be the case? I have another FreeSWITCH instance running the same exact version where I can access directory variables without a problem, so it must be a configuration issue or a module that I am missing or something. Regards, Örn

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Programatically constructing switch_r_sdp
by Aditya Uppuluri 01 Jun '22

01 Jun '22

Hello Team, Is there a way to programmatically copy the SDP from one leg to the other? I have a situation, where i have two incoming legs and want to bridge them without answering them in freeswitch. I want to know if i can copy the SDP from one leg to the other programmatically (Not via XML but via ESL inbound) I Found that there is a way to do it in XML dialplan like this: <action application="set"> <![CDATA[switch_r_sdp=(sdp here) ]]> </action> How to achieve something similar using ESL inbound? Regards, Aditya

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