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SIP for chat messages only
by mbo 23 Jan '15

23 Jan '15

Hello, We have implemented a web based client based on the jsSIP library to do audio/video and chat via SIP over websocket and freeswitch. With this client it is also possible to chat only. When chatting between two clients, the Call-ID in the message is different for each message. MESSAGE sip:1008@192.168.0.123 SIP/2.0 Via: SIP/2.0/WSS rfboa1c9gg6q.invalid;branch=z9hG4bK2512434 Max-Forwards: 69 To: <sip:1008@192.168.0.123> From: <sip:192.168.0.123>;tag=e472qkncp0 Call-ID: 0ggfl2u355ebrf77jsrq <—— always different CSeq: 521 MESSAGE sessionId: some uuid Content-Type: text/plain Allow: INVITE,ACK,CANCEL,BYE,UPDATE,MESSAGE,OPTIONS Supported: outbound User-Agent: JsSIP 0.6.0 Content-Length: 2 First problem: As the Call-ID is different for each message, I cannot group the messages to a conversation. Second problem: I want to store the chat conversation in the database and link it to a CDR. But there is no CDR written for the conversation. As I can see in the RFC (https://www.ietf.org/rfc/rfc3428.txt) on page 3, it should also be possible to create a dialog and then send all messages in context of this dialog. I assume this means, it is possible to establish a chat only (INVITE) call and then send all messages in context of the call. If there is a call then freeswitch will also write a cdr and if I can send messages in context of that call I can group them to a conversation, insert it to the db and link it to the cdr. How can I do this with freeswitch? Thanks Markus

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bridge after first call answers
by Necati Demir 23 Jan '15

23 Jan '15

Hello, When I run the following command, freeswitch make call to two of these numbers concurrently. fs_cli -x "originate loopback/5XXX/default & bridge(loopback/5YYY/default)" But I want a scenario like this: Freeswitch first calls 5XXX. If 5XXX answers, then freeswitch calls 5YYY. How to do this? -- Necati DEMİR --------------------

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Session Limit
by Ahmed Habiba 23 Jan '15

23 Jan '15

Dears, kindly Im facing some issue with concurrent session limit, below is the status showing the maximum sessions of 355, however I set maximum sessions to 10000, you kind help to identify why freeswitch use lower limit than the configured one will be appreciated. /etc/freeswitch# fs_cli -x "status" UP 0 years, 0 days, 0 hours, 19 minutes, 2 seconds, 490 milliseconds, 264 microseconds FreeSWITCH (Version 1.5.8b+git git 35f2bcc 2014-02-12 23:33:16Z 64bit) is ready 345 session(s) since startup 345 session(s) - peak 345, last 5min 345 0 session(s) per Sec out of max 1000, peak 66, last 5min 0 335 session(s) max min idle cpu 0.00/62.00 Current Stack Size/Max 240K/8192K switch.conf.xml: <param name="max-sessions" value="10000"/>  <param name="sessions-per-second" value="1000”/> Thanks , Ahmed Habiba.

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perl script excute hangup causes segfault on freeswitch
by kthofer 22 Jan '15

22 Jan '15

Hi There we want to hang up a channel after some time-out of not detecting DTMF or typing the wrong DTMF 3 times a perl script sends execute hangup and in deed sometime the channel is being terminated but sometime the free switch is eating all its memory and getting a segfault. we can not see why is FS not executing the hangup nor the reason for the segfault any suggestion/help would be relay appreciated. -- With best regards Karl Theo Hofer

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Issue in .NET call control using ESL DLL
by Grant Bagdasarian 22 Jan '15

22 Jan '15

Hello, I'm experiencing a rather weird issue when using the ESL libraries for .NET. When running the below code outside of Visual Studio in its own executable(Console Application) I'm receiving the below exception. When the code is run inside Visual Studio, it works perfect. Using ESL library contained in Freeswitch 1.2.13. Unhandled Exception: System.AccessViolationException: Attempted to read or write protected memory. This is often an indication that other memory is corrupt. at CSharp_ESLevent_Serialize(Void* jarg1, SByte* jarg2) in c:\users\gb\downloads\freeswitch-1.2.13\freeswitch-1.2.13\libs\esl\managed\esl_wrap.cpp:line 413 at ManagedEsl.ESLPINVOKE.ESLevent_Serialize(HandleRef jarg1, String jarg2) at VoiceEventSocketDemoClient.Program.Main(String[] args) in c:\Users\gb\Documents\Visual Studio 2012\Projects\VoiceEventSocketDemoClient\VoiceEventSocketDemoClient\Program.cs:line 26 TcpListener server = new TcpListener(IPAddress.Parse("LISTEN.IP.GOES.HERE"), 5000); server.Start(); TcpClient eslClient = server.AcceptTcpClient(); ESLconnection eslConnection = new ESLconnection(eslClient.Client.Handle.ToInt32()); ESLevent e = eslConnection.GetInfo(); Console.WriteLine(e.Serialize("json")); Does anyone know why this is happening? Maybe someone can reproduce the problem on their own PC using the above code example. Thank you! Regards, Grant

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Register more than 100 gateways on a host
by Regis M 22 Jan '15

22 Jan '15

Hi, I had a project where I could have to register between 100 and 500 gateways/trunk on a host. Does anyone have already done something similar ? Have you some problems with this configuration or recommandations ? For me, It"s equivalent to 250 sip phones register on me... it's just that : sofia status will give me some lines :) Regards Régis

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Re: [Freeswitch-users] How to Bridge To Local Extensions?
by George F. Phelps 22 Jan '15

22 Jan '15

Can someone help me with my question? Thanks, George From: George F. Phelps [mailto:GeorgePhelps@gfphelps.com] Sent: Saturday, January 10, 2015 12:02 PM To: freeswitch-users(a)lists.freeswitch.org Subject: How to Bridge To Local Extensions? The "transfer" statement, shown below, works (in my inbound dialplan): <action application="transfer" data="1001 XML default"/> What is the correct syntax for using "bridge" instead of "transfer"? The following statement does not work for me: <action application="bridge" data="sofia/internal/1001"/> My extensions are effectively default values and in the default directory location. For example: more /usr/local/freeswitch/conf/directory/default/1001.xml <include> <user id="1001"> <params> <param name="password" value="XXXX"/> <param name="vm-password" value="1001"/> </params> <variables> <variable name="toll_allow" value="domestic,international,local"/> <variable name="accountcode" value="1001"/> <variable name="user_context" value="default"/> <variable name="effective_caller_id_name" value="Extension 1001"/> <variable name="effective_caller_id_number" value="1001"/> <variable name="outbound_caller_id_name" value="$${outbound_caller_name}"/> <variable name="outbound_caller_id_number" value="$${outbound_caller_id}"/> <variable name="callgroup" value="techsupport"/> </variables> </user> </include> My goal is to configure simultaneous ringing for multiple extensions: <action application="bridge" data="{ignore_early_media=true}sofia/internal/1001,sofia/internal/1002"/> Thanks, George

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spandsp_start_dtmf detects # key many times
by Nagalenoj H. 22 Jan '15

22 Jan '15

Hi, When I use spandsp_start_dtmf along with play_and_get_digits, '#' is getting detected many times, though it is pressed once. Other DTMFs are getting detected properly, whereas the '#' is getting detected 2 or 3 times. I have checked the SIP phone's settings, I do not find anything wrong there. What could be the reason? Is this because of some SIP phone setting? Here is the freeswitch log, 2015-01-19 09:33:39.671681 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [1] 2015-01-19 09:33:39.671681 [DEBUG] switch_channel.c:488 RECV DTMF 1:2000 2015-01-19 09:33:39.851686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [1], duration = 191 ms 2015-01-19 09:33:40.071688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [4] 2015-01-19 09:33:40.071688 [DEBUG] switch_channel.c:488 RECV DTMF 4:2000 2015-01-19 09:33:40.251687 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [4], duration = 153 ms 2015-01-19 09:33:40.431699 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [7] 2015-01-19 09:33:40.431699 [DEBUG] switch_channel.c:488 RECV DTMF 7:2000 2015-01-19 09:33:40.631680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [7], duration = 204 ms 2015-01-19 09:33:40.851688 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [4] 2015-01-19 09:33:40.851688 [DEBUG] switch_channel.c:488 RECV DTMF 4:2000 2015-01-19 09:33:40.971695 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [4], duration = 127 ms 2015-01-19 09:33:41.151691 [DEBUG] switch_core_session.c:1053 Send signal sofia/internal/sip:278@192.168.2.86:5060 [BREAK] 2015-01-19 09:33:41.171701 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [7] 2015-01-19 09:33:41.171701 [DEBUG] switch_channel.c:488 RECV DTMF 7:2000 2015-01-19 09:33:41.351680 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [7], duration = 191 ms 2015-01-19 09:33:41.371686 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16@8000hz 1 channels 20ms 2015-01-19 09:33:41.691686 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [#] 2015-01-19 09:33:41.691686 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 EXECUTE sofia/internal/392(a)3427.vbiz.mundio.com flush_dtmf() 2015-01-19 09:33:41.691686 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16@8000hz 1 channels 30ms 2015-01-19 09:33:41.711684 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [#], duration = 38 ms 2015-01-19 09:33:41.771683 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [#] 2015-01-19 09:33:41.771683 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1747 done playing file file_string:///root/united_fone/voice_files/en//CR_enter_conf_pin.wav 2015-01-19 09:33:41.771683 [DEBUG] switch_ivr_play_say.c:1314 Codec Activated L16@8000hz 1 channels 30ms 2015-01-19 09:33:41.811686 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [#], duration = 25 ms 2015-01-19 09:33:41.871691 [DEBUG] mod_spandsp_dsp.c:373 DTMF BEGIN DETECTED: [#] 2015-01-19 09:33:41.871691 [DEBUG] switch_channel.c:488 RECV DTMF #:2000 2015-01-19 09:33:41.951681 [DEBUG] mod_spandsp_dsp.c:385 DTMF END DETECTED: [#], duration = 76 ms -- Regards, Nagalenoj H.

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Install FusionPBX in minutes on Centos 6 & 7
by Luis Daniel Lucio Quiroz 22 Jan '15

22 Jan '15

Hello everybody Just to share you this: https://okay.com.mx/en/entrepreneurs/install-fusionpbx-in-a-moment.html With this alternate repo you will be able to install FusionPBX in minutes, just with a yum command. Please read the link for all details. Luis Daniel Lucio Quiroz CISSP, CISM, CISA Linux, VoIP and much more fun www.okay.com.mx Need LCR? Check out LCR for FusionPBX with FreeSWITCH Need Billing? Check out Billing for FusionPBX with FreeSWITCH

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media bypass on freeswitch
by Achintha 21 Jan '15

21 Jan '15

I configured freeswitch (FreeSWITCH Version 1.5.15b+git~20141117T202539Z~424df19083~64bit) on Ubuntu 12.04. It works properly. I’m using xml_curl module to generate the dialplan. Then I set the media bypass parameter on freeswitch dialplan. But the freeswitch crashed when following dialplan was given. <document type="freeswitch/xml"> <section name="dialplan" description="RE Dial Plan For FreeSwitch"> <context name="130_131_SANASA"> <extension name="2022"> <condition field="destination_number" expression="[^\s]*"> <action application="set" data="ringback=${us-ring}" /> <action application="export" data="OperationType=User" /> <action application="export" data="CustomCompanyStr=131_130" /> <action application="export" data="FromGuUserId=14111215533245504" /> <action application="export" data="ToGuUserId=14111215495317100" /> <action application="export" data="FromUser=sanasains2023_sanasa_lk" /> <action application="export" data="FromNumber=2023" /> <action application="export" data="nolocal:ToNumber=2022" /> <action application="export" data="nolocal:ToUser=sanasains2022_sanasa_lk" /> <action application="set" data="continue_on_fail=true" /> <action application="set" data="hangup_after_bridge=true" /> <action application="set" data="ignore_early_media=false" /> <action application="set" data="bypass_media=true" /> <action application="bind_meta_app" data="3 ab s execute_extension::att_xfer XML PBXFeatures" /> <action application="bind_meta_app" data="4 ab s execute_extension::att_xfer_group XML PBXFeatures" /> <action application="bind_meta_app" data="6 ab s execute_extension::att_xfer_outbound XML PBXFeatures" /> <action application="bind_meta_app" data="5 ab s execute_extension::att_xfer_conference XML PBXFeatures" /> <action application="bridge" data="[leg_timeout=60,origination_caller_id_name=sanasains2023_sanasa_lk,origination_caller_id_number=2023]user/ 2022(a)192.168.51.X <2022(a)192.168.51.86>X,pickup/3003" /> <action application="hangup" /> </condition> </extension> </context> </section> </document In here I ‘m using attendant transfer feature. Here I have attached the print screen of freeswitch console just after it crashed. Kindly advise me to solve this problem. -- Thanking You.. Achintha

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