- FreeSWITCH-users - FreeSWITCH Mailing Lists

ESL help on remote machines
by Jungle Boogie 18 Jan '19

18 Jan '19

Hi All, I know I'm a little late to the ESL party, but I have a use for it and I want to try it out. As the inbound diagram depicts, I would like a perl script on a remote machine to trigger a simple lua script on the FreeSWITCH machine that generates an outbound call. https://freeswitch.org/confluence/display/FREESWITCH/mod_event_socket ----------- Machine A - perl esl ----------- | | \/ ---------- Freeswitch - lua originate ---------- | | \/ ---------- phone call --------- The perl esl page states the module must be compiled. Does this mean machine A needs to have the entire FS source code to compile the module? https://freeswitch.org/confluence/display/FREESWITCH/Perl+ESL If the entire src isn't needed, what's it take to install the ESL module successfully on machine A? Thanks, sean

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Shared presence not working with callcenter application of Freeswitch
by sagar malam 18 Jan '19

18 Jan '19

I am facing issue related to presence/ shared presence for agents in callcenter(mod_callcenter). Suppose we are dialling a user 1001(a)example.com and it is a shared user in two phones. Shared presence works perfectly if we directly dial extension 1001. But if same extension is configured as agent of callcenter and call is bridged to agent(extension) through callcenter, shared presence does not work. Freeswitch does not generate notify for agent state change. Is this expected behaviour ? If not then what can fix this ? Thanks in advance

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Screen pop application
by Moshe Rosenberg 18 Jan '19

18 Jan '19

Hi I am looking for a windows based application that will pop up a web browser with the url + caller I’d each time the phone rings -- Moshe Rosenberg Tel. 718 633 1444 Cel. 347 678 3993 www.data <http://www.dataday.consulting/>phone.cloud Moshe(a)dataphone.cloud

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Double NAT w/ same Private Network Address
by Nandy Dagondon 17 Jan '19

17 Jan '19

Hi everybody, I'm testing mod_verto in a Double NAT setup. The problem: no media. This is the setup: FS (192.168.1.x/24) <> NAT <> Internet <> NAT <> WebRTC Client (192.168.1.x/24) Both FS and the Client routers have static IP WAN addresses. I suspect that since Private Network addresses are the same (192.168.1.x/24), the ICE candidate chosen is always 192.168.1.x, thinking that it's in the Local Network. Examining the log file indeed showed that the WAN IP's are included in the candidate. But, the one chosen is always a private IP. Am I correct? What needs to be done to ensure that WAN candidates will be selected? Changing the Network Address of either side might solve the problem but I need the solution via FS/Verto configuration. (Btw, the local TURN service is not running.) Appreciate for your help. /Nandy

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Call Quality with Recording
by Jai Rangi 16 Jan '19

16 Jan '19

We experience poor call quality when recording is on. We are using record_session to do call recording. session:execute("record_session","$${recordings_dir}/${uuid}.mp3") Any one else experiencing similar issue? Any comment on recording wav vs mp3. Thank you, Jai

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Play the sound file to caller from the same location where it had left in previous call
by Aqs Younas 15 Jan '19

15 Jan '19

Greetings, I would like to play the sound file to the caller at the same location where he had left on the previous call. How I see is that, save call duration upon call end into database and check it again upon new call against DID and sleek the file to that point. I have seen a post from one guy on the same topic but could not able to find this. Any better thoughts? Best Regards, Aqs Younas

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How to enable the tls avaya J series phones
by Surender Singh 15 Jan '19

15 Jan '19

Hi Team, I am new for freeswitch and I installed the FS on my window7 machine . I want to register the avaya j100 series phone with FS with TLS . I have one CA.pem certificate which I generated through SSL tool or I need to generate the CA certificate using FS only. Can you please guide or share the steps to enable the TLS with FS for window based FS ? Regads Surender Singh ::DISCLAIMER:: -------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------- The contents of this e-mail and any attachment(s) are confidential and intended for the named recipient(s) only. E-mail transmission is not guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or may contain viruses in transmission. The e mail and its contents (with or without referred errors) shall therefore not attach any liability on the originator or HCL or its affiliates. Views or opinions, if any, presented in this email are solely those of the author and may not necessarily reflect the views or opinions of HCL or its affiliates. Any form of reproduction, dissemination, copying, disclosure, modification, distribution and / or publication of this message without the prior written consent of authorized representative of HCL is strictly prohibited. If you have received this email in error please delete it and notify the sender immediately. Before opening any email and/or attachments, please check them for viruses and other defects. --------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------------

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Basic Nagios script for Freeswitch
by Mickael Hubert 14 Jan '19

14 Jan '19

Hi all, today I developed a basic nagios script to check freeswitch instance. I publish it in github, maybe it can help you. https://github.com/Mickaelh51/pynagfreeswitch Ex: pynagfreeswitch.py -t -w 10 -c 20 ==> OK - Total calls count is 0 pynagfreeswitch.py -g gateway1 -G failedcallsout -w 10 -c 20 ==> OK - gateway1 failedcallsout is 2 pynagfreeswitch.py -g GWSCR1 -G pingtime -w 0.20 -c 0.56 ==> WARNING - GWSCR1 pingtime is 0.55 pynagfreeswitch.py -g GWSCR1 -G status -S up ==> OK - GWSCR1 status is up pynagfreeswitch.py -g GWSCR1 -G state -S noreg ==> OK - GWSCR1 state is noreg ./pynagfreeswitch.py -h usage: pynagfreeswitch.py [-h] [-a AUTH] [-s SERVER] [-p PORT] [-w WARNING] [-c CRITICAL] [-g GATEWAY] [-G GATEWAYCHECK] [-t] [-S STRINGOK] optional arguments: -h, --help show this help message and exit -a AUTH, --auth AUTH ESL password -s SERVER, --server SERVER FreeSWITCH server IP address -p PORT, --port PORT FreeSWITCH server event socket port -w WARNING, --warning WARNING threshold that generates a Nagios warning -c CRITICAL, --critical CRITICAL threshold that generates a Nagios critical warning -g GATEWAY, --gateway GATEWAY select gateway -G GATEWAYCHECK, --gatewaycheck GATEWAYCHECK type of checking (failedcallsout / status / state / ...) -t, --totalcallscount total calls count -S STRINGOK, --stringok STRINGOK if this string is found = OK ++

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Max participants join Conference bridge
by Ahmed Chohan 11 Jan '19

11 Jan '19

Hi, I'm currently exploring varity of products in market that support conference service, I would like to how many participants can join each conference in freeswitch? Please advise. -- Regards, Ahmed Munir Chohan

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Transfer call to Call Group cause LOSE_RACE
by Chhorm Chhatra 11 Jan '19

11 Jan '19

Hello, Currently, I faced a problem that when I transfer a call to call group, it causes *Originate Failed. Cause: LOSE_RACE.* My setup is that I export bind digit action on the Leg B so that he can transfer Leg A to another extension. > Transfer Configuration: <action application="bind_digit_action" data="my_transfer,~069\d{3},exec:execute_extension,digit_transfer XML default,peer,peer"/> <action application="set" data= "bridge_pre_execute_bleg_app=bind_digit_action"/> <action application="set" data="bridge_pre_execute_bleg_data=my_transfer"/> > Digit_transfer Dialplan Configuration: <extension name="Display digits dialled AND TRANSFER"> <condition field="destination_number" expression="^digit_transfer$"> <action application="log" data="INFO Called dialled ${last_matching_digits}"/> <action application="log" data="INFO en number iterated ${last_matching_digits}"/> <action application="transfer" data="-bleg ${last_matching_digits} XML default"/> </condition> </extension> >From this part, the digit_transfer extension will transfer based on last_matching_digits to the target extension bound by Leg B. Note: normal call to group call can be bridged successfully. Any help would be really appreaciated. Best Regards, Chhorm Chhatra

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