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JAVA CODE ERROR
by Freeswitch user 19 Sep '17

19 Sep '17

Error:- [ERR] switch_cpp.cpp:683 object is not initalized import org.freeswitch.*; import org.freeswitch.swig.*; public class PhoneTest implements FreeswitchScript, DTMFCallback, HangupHook { public PhoneTest() { } public String onDTMF(Object object, int i, String arg) { if (object instanceof String) freeswitch.console_log("notice", "DTMF: " + (String)object + " ARG: " + arg + "\n"); else freeswitch.console_log("notice", "WOW GOT AN EVENT: " + object.toString()); return "true"; } public void onHangup() { freeswitch.console_log("notice", "HANGUP!\n"); } public void run(String sessionUuid, String args) { freeswitch.console_log("notice", "UUID: " + sessionUuid + " ARGS: " + args + "\n"); JavaSession session = null; try { session = new JavaSession(sessionUuid); session.setDTMFCallback(this, "TEST"); // Getting error in this line. session object not initilizing session.setHangupHook(this); session.answer(); session.streamFile(args, 0); session.hangup(""); } finally { if (session != null) session.delete(); } } }

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install source error
by Макаревич М.А. 19 Sep '17

19 Sep '17

Hi, I am installing freeswitch 1.6.19 with RPI-3,but getting some errors Linux raspberrypi 4.9.41-v7+ #1023 SMP Tue Aug 8 16:00:15 BST 2017 armv7l GNU/Linux. pi@raspberrypi:~/freeswitch-1.6.19 $ make -j3 touch /home/pi/freeswitch-1.6.19/src/include/switch.h CC libfreeswitch_spandsp_la-plc.lo make[1]: Entering directory '/home/pi/freeswitch-1.6.19/libs/srtp' /bin/bash ./libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -g -O2 -Wall -std=c99 -pedantic -MT srtp.lo -MD -MP -MF .deps/srtp.Tpo -c -o srtp.lo `test -f 'srtp/srtp.c' || echo './'`srtp/srtp.c libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -g -O2 -Wall -std=c99 -pedantic -MT srtp.lo -MD -MP -MF .deps/srtp.Tpo -c srtp/srtp.c -fPIC -DPIC -o srtp.o In file included from ./crypto/include/prng.h:17:0, from ./crypto/include/crypto_kernel.h:50, from ./include/srtp.h:53, from srtp/srtp.c:46: ./crypto/include/aes_icm_ossl.h:66:20: error: field ‘ctx’ has incomplete type EVP_CIPHER_CTX ctx; ^~~ In file included from srtp/srtp.c:50:0: ./crypto/include/aes_gcm_ossl.h:58:18: error: field ‘ctx’ has incomplete type EVP_CIPHER_CTX ctx; ^~~ config.status: creating Makefile config.status: creating include/apr.h config.status: creating build/apr_rules.mk Makefile:513: recipe for target 'srtp.lo' failed make[1]: *** [srtp.lo] Error 1 Please suggest me.

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Dynamic Conference Calls
by Akshaya j 19 Sep '17

19 Sep '17

Hello FS users, I'm a fresher working around with FreeSwitch features and came across this conference room feature. This conference room actually acts like a rooms where anyone can join the room by dialling some number. There is a also moderator who can add any user into the conference room. What i need to know is whether the conventional conference calls is available in FreeSwitch? The conventional conference calls is what we use in our normal mobile phones where any person in a call can add any other person by dialling his number and then all 3 persons can talk together. That is anybody in a call can act as a moderator to add a person to join. I don't find any content regarding this conference in web, is there any other terminology for this feature in Freeswitch which i dont see or whether this is possible? If so how to proceed with this ?? Any help would be highly appreciated.!!! Thanks Advance, PurpleWinged

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Bug? Recording pauses when the recorded leg is held
by Björn Bylander 19 Sep '17

19 Sep '17

Hello, I've got an application using mod_erlang_event which does the following: 1. Originates a call to A and waits for A to answer. 2. Parks A. 3. Originates a call to B and waits for B to answer. 4. Bridges A to B. 5. Starts recording B using uuid_record (to a file with the "wav" extension). Audio from both A and B is recorded, as expected. 6. Stops recording when either leg is terminated. If B holds its call using a SIP re-INVITE with recvonly in the SDP, FreeSWITCH pauses recording instead of recording just the audio from the A leg. This seems like a bug to me given the documentation for uuid_record (from https://freeswitch.org/confluence/display/FREESWITCH/mod_commands) "The start command causes FreeSWITCH to start mixing all call legs together and saves the result as a file in the format that the file's extension dictates." Am I right in considering this behaviour to be a bug? I'm using FS v1.6.19. Regards, Björn Bylander

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rtmp and user registrations
by Madovsky 19 Sep '17

19 Sep '17

Hi everybody, is mod_rtmp registered users can be managed like sip profile with a postgresql table like rtmp_registrations? thanks Franck

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install error
by Макаревич М.А. 19 Sep '17

19 Sep '17

I am installing freeswitch 1.6.19 with RPI-3,but getting some errors. Linux raspberrypi 4.9.41-v7+ #1023 SMP Tue Aug 8 16:00:15 BST 2017 armv7l GNU/Linux pi@raspberrypi:~/freeswitch-1.6.19 $ make -j3 touch /home/pi/freeswitch-1.6.19/src/include/switch.h CC libfreeswitch_spandsp_la-plc.lo make[1]: Entering directory '/home/pi/freeswitch-1.6.19/libs/srtp' /bin/bash ./libtool --tag=CC --mode=compile gcc -DHAVE_CONFIG_H -I. -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -g -O2 -Wall -std=c99 -pedantic -MT srtp.lo -MD -MP -MF .deps/srtp.Tpo -c -o srtp.lo `test -f 'srtp/srtp.c' || echo './'`srtp/srtp.c libtool: compile: gcc -DHAVE_CONFIG_H -I. -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -I./src -Icrypto/include -I./include -I./crypto/include -g -O2 -Wall -std=c99 -pedantic -MT srtp.lo -MD -MP -MF .deps/srtp.Tpo -c srtp/srtp.c -fPIC -DPIC -o srtp.o In file included from ./crypto/include/prng.h:17:0, from ./crypto/include/crypto_kernel.h:50, from ./include/srtp.h:53, from srtp/srtp.c:46: ./crypto/include/aes_icm_ossl.h:66:20: error: field ‘ctx’ has incomplete type EVP_CIPHER_CTX ctx; ^~~ In file included from srtp/srtp.c:50:0: ./crypto/include/aes_gcm_ossl.h:58:18: error: field ‘ctx’ has incomplete type EVP_CIPHER_CTX ctx; ^~~ config.status: creating Makefile config.status: creating include/apr.h config.status: creating build/apr_rules.mk Makefile:513: recipe for target 'srtp.lo' failed make[1]: *** [srtp.lo] Error 1 make[1]: Leaving directory '/home/pi/freeswitch-1.6.19/libs/srtp' Makefile:3719: recipe for target 'libs/srtp/libsrtp.la' failed make: *** [libs/srtp/libsrtp.la] Error 2 Please suggest me.

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vm_cc weird behavior
by Yuriy Nasida 19 Sep '17

19 Sep '17

Hi, I noted that FS sends a lot of emails to my email box instead of only one in case I use vm_cc. It's very simple configuration. I just set vm_cc before mod_voicemail is called (according wiki page) and leave voicemail message. For main emails box I recieve only one message as expected. But I have ~60 exactly same messages at email box from vm_cc. In console I also noted that FS really sends emails many times without obvios reason. mod_voicemail.c:2832 Deliver VM to yuriytest2(a)mydomain.com mod_voicemail.c:1925 Update MWI: Processing for yuriytest2(a)mydomain.com in inbox mod_voicemail.c:1950 Update MWI: Messages Waiting yes mod_voicemail.c:1951 Update MWI: Update Reason NEW mod_voicemail.c:1952 Update MWI: Message Account yuriytest2(a)mydomain.com mod_voicemail.c:1953 Update MWI: Voice Message 69/0 switch_utils.c:1180 Emailed file [/tmp/mail.1505386894e5a8] to [ myemailbox(a)gmail.com] mod_voicemail.c:3042 Sending message to myemailbox(a)gmail.com Next FS does this again and tries to Deliver VM to yuriytest2(a)mydomain.com. Please advice.

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How to record User Voice and listen for the DTMF simultaneously
by Deepika Yadav 18 Sep '17

18 Sep '17

Hi, I want to give users an option to record their voice by presssing some digit in the IVR menu through my python script. I use the code as follows: session.recordFile(filename, 120, 500, 3) However, this code will block for 2 minutes and then proceed. In case a user has little to say and after 1 minute wants to stop the recording, how I can do that. Is there any function similar to "session.playAndGetDigits" to also hear for DTMFs at the same time? -- Regards, Deepika https://deepikay.wixsite.com/deepika

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Re: [Freeswitch-users] TLS and SRTP commercial certs go in which file?
by Shaun Stokes 18 Sep '17

18 Sep '17

Hi Bipin, We've found that these are the certs which FreeSWITCH will use, the pem file will need the public and private key (same as your wss cert). You'll also need to make sure the user for FreeSWITCH has read permission to the certs. agent.pem dtls-srtp.pem tls.pem wss.pem FreeSWITCH doesn't seem to need the intermediary and root cert of the CA. Here are some of the TLS parameters you might also want on your SIP profile. Name: tls Value: true Name: tls-bind-params Value: transport=tls Name: tls-cert-dir Value: "Your Cert Directory Path" Name: tls-sip-port Value: 5061 Name: tls-verify-date Value: true Name: tls-verify-depth Value: 2 Name: tls-verify-policy Value: all|subjects_all Name: tls-version Value: tlsv1.2 Shaun From: FreeSWITCH-users [mailto:freeswitch-users-bounces@lists.freeswitch.org] On Behalf Of Bipin Patel Sent: 16 September 2017 06:49 To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org<mailto:freeswitch-users@lists.freeswitch.org>> Subject: Re: [Freeswitch-users] TLS and SRTP commercial certs go in which file? hi, no one? Regards, Bipin ________________________________ -------- Original Message -------- Subject: [Freeswitch-users] TLS and SRTP commercial certs go in which file? From: Bipin Patel <bipin(a)xbipin.com><mailto:bipin@xbipin.com> To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org><mailto:freeswitch-users@lists.freeswitch.org> Date: 9/15/2017, 3:44:33 PM hi, when i setup verto on my server i used commercial certificates with wss.pem containing the following and all that works brilliant: -----BEGIN CERTIFICATE----- <lots of gibberish from the actual certificate> -----END CERTIFICATE----- -----BEGIN RSA PRIVATE KEY----- <lots of gibberish from the actual certificate> -----END RSA PRIVATE KEY----- -----BEGIN CERTIFICATE----- <lots of gibberish from the intermediate certificate> -----END CERTIFICATE----- -----BEGIN CERTIFICATE----- <even more gibberish from the root certificate> -----END CERTIFICATE----- now i want to use the same certificate for TLS and SRTP and i was reading the docs and it mentioned to create a agent.pem file with the actual server cert and key but where do i copy the intermediatory and root cert of the CA, which folders do both go in and with what filename? any help is appreciated -- Regards, Bipin ________________________________ _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)freeswitch.org<mailto:consulting@freeswitch.org> http://www.freeswitchsolutions.com Official FreeSWITCH Sites http://www.freeswitch.org http://confluence.freeswitch.org http://www.cluecon.com FreeSWITCH-users mailing list FreeSWITCH-users(a)lists.freeswitch.org<mailto:FreeSWITCH-users@lists.freeswitch.org> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________ [http://www.itec-support.co.uk/wp-content/uploads/2016/07/email_logo.jpg] Shaun Stokes - Infrastructure Analyst T : 01453 700713 E : shaun.stokes(a)itec-support.co.uk W : www.itec-support.co.uk Registered Address :- ITEC Support, Suite 2 Prospect House, Bath Road, Stroud, Gloucestershire GL5 3QF Company No. 06908001 CONFIDENTIALITY NOTICE This communication and the information it contains are intended for the person or organisation to which it is addressed. Its contents are confidential and may be protected in law. Unauthorised use, copying or disclosure of any of it may be unlawful. If you are not the intended recipient, please contact us immediately. The contents of any attachments in this e-mail may contain software viruses, which could damage your own computer system. While ITEC Support has taken every reasonable precaution to minimise this risk, we cannot accept liability for any damage which you sustain as a result of software viruses. You should carry out your own virus checking procedure before opening any attachment. ______________________________________________________________________ This message has been checked for all known viruses by MessageLabs Virus Scanning Service. ______________________________________________________________________

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488 INCOMPATIBLE DESTINATION with Firefox 55.03
by Agustí Ubalde Bellot 18 Sep '17

18 Sep '17

Hi all, I am testing WebRTC calls with Firefox 55.03 and I have a problem receiving calls. The case is that FreeSWITCH receives a call with G711 codec from an external Asterisk, but when it processes INVITE, it receives a 488 from the browser: ====================================================================== nta.c: 3297 agent_recv_response () nta: received 488 Not Acceptable Here for INVITE (112529915) ====================================================================== Any idea what may be going on? An additional note: calls from the navbegador to this same Asterisk are generated correctly, and in Chrome the 2 cases work correctly. Regards, Agustí Ubalde

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