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DTMF
by Tihomir Culjaga 19 May '17

19 May '17

hello FS uses, On incoming calls, i got a carrier that advertises rfc2833 (telephone-event) while supporting inBand DTMF only. is there a chance FreeSWITCH can ignore telephone-event for incoming calls from a specific gateway ( profile ) and switch to inBand ? Regards, T.

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DTMF
by Brian : 19 May '17

19 May '17

Hello List, We have a PITA carrier that we do 2833 with 99.8% of the time. Sometimes if the carrier they are sending the call to doesn't support 2833 when they will send an OK back to us the SDP won't have 2833 in the RTP Map. Is it possible to detect this in dialog and start dtmf generate so we will send tones inband for these calls? For inbound calls we detect it with something like <condition field="${switch_r_sdp}" expression="a=rtpmap:(\d+)\stelephone-event/8000"> on the initial invite. Just not sure how to do this for an OK in dialog or if its possible. Thanks! Brian

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Verto calls
by Agustí Ubalde Bellot 19 May '17

19 May '17

Hi all, I am trying to call from Verto extension to another Verto extension. *Both are successfully registered* (Verto status show the successfully register) but the call between is not established. The call remains in ring state. This is the last dialplan function executed (calling 1000 to 1001): *EXECUTE verto.rtc/1001 bridge()* Regards, Tineli

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Uninstalling freeswitch after compiling and installing from source
by Ravi sanyal 18 May '17

18 May '17

Hi, I made a noob mistake and ran the script for installing freeswitch from binaries without doing logs on the process and fs_cli is complaining that commands are not found. specifically, sofia status, or sofia status profile <name> but im sure that it extends further than that. I'm dont know how to get a list of the commands available. typing in help (not /help) at the prompt says it can't find a command. So i've installed it and it doesn't seem to be working, and don't know how to uninstall it. Where can i find a list of acceptable commands for the configure command? Im trying to enable sctp, but when looking through the logs for it you have a line: checking whether SCTP is supported... yes is this the same as checking with the --enable-sctp flag? (when looking at the logs for the added command there's no extra lines pertaining to enable-sctp as if you didn't specify it). I'm running the latest build from the github repository. When running this command: ./configure --enable-sctp > /home/fleetchat/config.log it keeps on displaying errors on output, but don't know if they're serious or not. ./configure: line 31222: php-config: command not found ./configure: line 31223: php-config: command not found ./configure: line 31224: php-config: command not found ./configure: line 31225: php: command not found ./configure: line 31226: php-config: command not found ./configure: line 31227: php-config: command not found ./configure: line 5306: AX_COMPILER_VENDOR: command not found rm: cannot remove 'conftest*': No such file or directory ./configure: line 19537: test: 3.16.0-4-amd64: integer expression expected ./configure: line 19819: test: 3.16.0-4-amd64: integer expression expected config.status: WARNING: 'apr-config.in' seems to ignore the --datarootdir setting rm: cannot remove 'libtoolT': No such file or directory config.status: WARNING: 'Makefile.in' seems to ignore the --datarootdir setting config.status: WARNING: 'lib/Makefile.in' seems to ignore the --datarootdir setting configure: WARNING: ** STUN support disabled ** Im sure it's somewhere but i also don't know where the configuration files should be located when building from source. i placed them in /etc/freeswitch. From memory that was where the installation instructions said they should be located. The configuration files were downloaded from the ubuntu repository command freeswitch-meta-vanilla The reason they come from the ubuntu repository is because i had freeswitch set up on ubuntu 16.04 but found out debian was the recommended OS. I'm running freeswitch on debian 8.8 now. Nathan -- *Virtual Guard Ltd* *info(a)virtualguard.co.nz <info(a)virtualguard.co.nz>*

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WSS SSL errors "decryption failed or bad record mac" under load
by Luke Wahlmeier 18 May '17

18 May '17

I keep semi-regularly running into issues using the wss transport when using dtls/strp/ice. This is on the latest 1.6.17~34~0fc0946 on Debian jessie, but I am pretty sure it was happening on the last couple releases as well. It seems like something bad/wrong happens to the encrypted data going over the websocket coming from freeswitch when more then 1 websocket connection are going and so far ice/srtp/dtls also seem to be needed in the invite to duplicate it. I have tried many different languages and network/ssl stacks and keep running into this. It is always on data coming in from freeswitch on the websocket connection, and its very very random. Sometimes I will get it 20 times in a row, other times it takes thousands of connections/sessions before it happen. It also, obviously, completely goes away if I use plain ws instead wss. Here are the errors: python: SSLError: [SSL: DECRYPTION_FAILED_OR_BAD_RECORD_MAC] decryption failed or bad record mac (_ssl.c:1750) c/c++ (stunnel4): SSL_read: 1408F119: error:1408F119:SSL routines:SSL3_GET_RECORD:decryption failed or bad record mac Java: java.lang.IllegalArgumentException: Bad arguments at javax.crypto.Mac.update(Mac.java:509) at sun.security.ssl.MAC.compute(MAC.java:135) at sun.security.ssl.InputRecord.checkMacTags(InputRecord.java:265) at sun.security.ssl.InputRecord.decrypt(InputRecord.java:216) at sun.security.ssl.EngineInputRecord.decrypt(EngineInputRecord.java:177) at sun.security.ssl.SSLEngineImpl.readRecord(SSLEngineImpl.java:974) at sun.security.ssl.SSLEngineImpl.readNetRecord(SSLEngineImpl.java:907) at sun.security.ssl.SSLEngineImpl.unwrap(SSLEngineImpl.java:781) at javax.net.ssl.SSLEngine.unwrap(SSLEngine.java:624) Attached are a simple python script to do the load, my dialplan and sip_profile. The python script can take a few runs before it see the error, and I know its not completing the sip or rtp, but even if it does this still happens. I have also looked at libsofia-sip-ua/tport/ws.c and I dont see anything obvious. I am getting setup to build v1.6 head and test this any guidance on ways I can trouble shoot this better or requests for more info are very welcome. Thanks Luke

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REFER for transer call handling
by Yuriy Gorlichenko 18 May '17

18 May '17

Hi. I found some examples about features.xml but it is just a part of dialplan and will be executed on the DTMF sended in call processing (am i right?), so I trying to understand how Freeswitch handles REFER. Presume i have phone with button "transfer" that sends REFER message to server. How can I associate dialplan whith REFER message arrived to server (rewrite function for it or give additional functionality). Also can i bind some codes (like in conference mode) for every call? Because in coference mode it is very cool to use: session:execute("bind_digit_action","moderator,*1,exec:execute_extension,add") session:execute("bind_digit_action","moderator,*2,exec:execute_extension,remove") for rewrite feature codes for it. Thx you for response

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Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue
by sachin 18 May '17

18 May '17

Hello Michael, Thanks for the information. Can you please let me know what was the client used? Also I am using self signed certificates on my local LAN. Thanks and RegardsSD From: Michael Jerris <mike(a)jerris.com> Sent: Fri, 12 May 2017 03:58:08 GMT+0530 To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org> Subject: Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issue we just had confirmed on Jira that 1.6.17 windows is working fine with latest chrome.  If you are getting errors still with something that is other than sipml5, please confirm details of what exactly the issue is.On May 8, 2017, at 8:08 AM, sachin <snehsach(a)rediffmail.com> wrote:Hello Michael,I tried using Sip.js instead of sipml5 and with Linux as the server. I have installed the FreeSWITCH Version 1.9.0+git~20170501T171230Z~e3ef041517~64bit (git e3ef041 2017-05-01 17:12:30Z 64bit)I am getting INCOMPATIBLE  DESTINATION error. Also tried sipml5.. with bot the client I am getting the same error.Please let me know what I am missing.Thanks and Regards,SDFrom: Michael Jerris <mike(a)jerris.com>Sent: Fri, 05 May 2017 21:22:20To: FreeSWITCH Users Help <freeswitch-users(a)lists.freeswitch.org>Subject: Re: [Freeswitch-users] Changing audio DTLS state from HANDSHAKE to FAIL issuecan you try this using sip.js or something thats confirmed to work when using linux as a server please.  This may just be an issue with sipml5. On May 5, 2017, at 9:53 AM, Brian West <brian(a)freeswitch.org> wrote: Clearly that isn't the case, I suspect a define is missing to actually enable ECDSA at this point. On Fri, May 5, 2017 at 8:47 AM, Gregor Nanger <gregor@infomedia.si> wrote:Brian, isn't this solved in 1.6.16? FS-10037 [core] Update OpenSSL to version 1.0.2k for Windows 2017-05-05 15:26 GMT+02:00 Brian West <brian@freeswitch.org>:OpenSSL on the windows build needs to be updated. https://freeswitch.org/jira/browse/FS-9510 On Thu, May 4, 2017 at 2:31 PM, Gregor Nanger <gregor@infomedia.si> wrote:This error is familiar to me, I think so, if I remembered correctly.  I am using windows. I had this error when client was Chrome. I just updated openssl dlls. I think that openssl library is updated on windows build now with version 1.6.17. And this version is also compiled on FS FTP to just download and install it. 2017-05-04 19:15 GMT+02:00 Anthony Minessale <anthony.minessale@gmail.com>:FS 1.5 sounds like a bad plan.Try latest FS 1.6 or master.  On Thu, May 4, 2017 at 12:04 PM, Ítalo Rossi <italo@freeswitch.org> wrote:Old openssl version maybe?Em qui, 4 de mai de 2017 às 11:19, sachin <snehsach(a)rediffmail.com> escreveu:Hello All,I am trying to setup the Webrtc call between 2 SIPML5 client. I am using FreeSWITCH Version 1.6.14 on Windows. I am using internal LAN.The clients are getting registered over wss. I have created self signed certificates.  In var.xml I have set the codecs setting as follows  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=PCMU,PCMA">  <X-PRE-PROCESS cmd="set" data="outbound_codec_prefs=PCMU,PCMA">I am able to establish the call and there is 2 way voice when I call from Sipmpl5 to Linphone and it works. But when I call from Linphone to Firefox (Simpl5) then the call is not getting established and I am getting following error2017-05-04 11:29:04.543434 [ERR] switch_rtp.c:2907 audio Handshake failure 12017-05-04 11:29:04.543434 [INFO] switch_rtp.c:2908 Changing audio DTLS state from HANDSHAKE to FAIL2017-05-04 11:29:04.543434 [NOTICE] switch_rtp.c:2889 Hangup sofia/internal/sip:1001@df7jal23ls0d.invalid [CS_EXCHANGE_MEDIA] [DESTINATION_OUT_OF_ORDER]My setup is as followsSIPML5 (FireFox browser 53.0)<-----> FS (v1.5)(Windows) <----> LinphoneI am attaching the logs for the reference.fs-logleve9.txt : Debug trace with loglevel =9fs-sip-trace.txt : Sip tracePlease let me know what could the issue and pointers to resolve the same.Thanks and RegardsSD _________________________________________________________________________ Professional FreeSWITCH Consulting Services: consulting(a)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 http://lists.freeswitch.org/mailman/listinfo/freeswitch-users UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users http://www.freeswitch.org

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Dialplan conditions
by Raúl Alexis Betancor Santana 18 May '17

18 May '17

Now that I have solved the problem of loading the dialplan from the xml files ... I get to the point that it doesn't work as I expect. This is the only extension defined: <context name="default"> <extension name="t38_transcode"> <condition field="sip_h_X-T38-Transcode" expresion="^True"/> <condition field="destination_number" expresion="^(\+1|1)([2-9]\d\d[2-9]\d{6})$"> <action application="set" data="fax_enable_t38=true"/> <action application="set" data="sip_execute_on_image=t38_gateway peer nocng"/> <action application="bridge" data="sofia/external/$1$2(a)proxy.server.tld"/> </condition> </extension> </context> I expect not to call the bridge app if there is no T38-Transcode header on the incoming INVITE request and if it is present ... that $1$2 will be filled ... but what I get on the traces is that any call fires the call to the bridge app and also that the dialled number is exactly $1$2 ... instead of the values they should have. I'm loading a minimal setup of modules, did I need to load someother one to get the conditions and the expresions to work on this simple dialplan? My modules.conf.xml file: <configuration name="modules.conf" description="Modules"> <modules> <load module="mod_syslog"/> <load module="mod_logfile"/> <load module="mod_event_socket"/> <load module="mod_sofia"/> <load module="mod_commands"/> <load module="mod_dptools"/> <load module="mod_spandsp"/> <load module="mod_tone_stream"/> <load module="mod_db"/> <load module="mod_xml_rpc"/> <load module="mod_expr"/> <load module="mod_dialplan_xml"/> </modules> </configuration> Best regards

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Noob question about xml dialplan
by Raúl Alexis Betancor Santana 18 May '17

18 May '17

Hi all, till now I'm been working with pre-made setup files for FS, and now I'm trying to get a deep knowleadge of how the dialplan works. So I modifed my autoload_configs/sofia.conf.xml file and changed my dialplan param to something like this: <param name="dialplan" value="XML:/etc/freeswitch/ dialplans/t38_transcode.xml,inline:socket:127.0.0.1:8022 async full"/> The Idea is that it loads another .xml file especific task and also use a dialplan throught a socket to a daemon that handle the rest. On my t38_transcode.xml file ... very simple: <extension name="t38_transcode"> <condition field="sip_h_X-T38-Transcode" expresion="^True"/> <condition field="destination_number" expresion="^(\+1|1)?([2-9]\d\ d[2-9]\d{6})$"> <action application="set" data="fax_enable_t38=true"/> <action application="set" data="t38_trace=true"/> <action application="set" data="sip_execute_on_image=t38_gateway peer nocng"/> <action application="bridge" data="sofia/external/$1$2(a)proxy.server.tld"/> </condition> </extension> But If I fire a call to FS like [number]@[FS_IP] from a sip testing client I get [INFO] switch_core_state_machine.c:311 No Route, Aborting What I'm doing wrong here? My target its just I want to ANY call that came in with a SIP header of X-T39-Transcode=True (market by a sip proxy elsewhere on the net) just do a T38->Ulaw transcoding saving the T38 trace, so I could inspect it later. The rest of calls coming in ... as they don't have the sip-header should end on other app. For doing the testing I disabled the socket_inline part of the dialplan string, so it have only this: <param name="dialplan" value="XML:/etc/freeswitch/dialplans/t38_transcode.xml"/> Did I miss something? ... or maybe missundestood who the xml dialplan works ?

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Fax result when using mail to fax
by Bipin Patel 17 May '17

17 May '17

Hi, I have a python script that checks a mailbox for faxes to be sent, when it finds a new mail it downloads the PDF and converts to Tiff and sends via the originate command, now the problem is I can't seem to find a way to get the fax report, nor in cdr it says what happened and the originate just says ok followed by uuid of the bridge. I'm sending the fax over a fxo gateway device. Due to it going over fxo, I have no clue if the remote fax machine answered call and the only place the fax report comes is in the console. Is there a variable or anything else that can be resent to the python script telling what the result was. I tried the execute on fax successful and fail variable but they don't seem to work. My python script generates the originate command going to loopback which goes to dial plan where the actual bridge to fxo gateway happens. How are others dealing with fax report I would love to hear. Regards, Bipin

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