freeswitch

What is the purpose of FreeSWITCH's `mod_esl`?

由 放肆的年华 提交于 2019-12-13 08:57:58
问题 There is another question trying to figure out the connection between mod_event_socket and the Event Socket Library (ESL). The Event Socket Library itself is a C library (libesl) that can be used to build external C applications to control FreeSWITCH via its event system. As far as I understand, mod_esl is only a FreeSWITCH module wrapper around libesl, used to extend programming languages with the same functionality (i.e., to control a FreeSWITCH instance). The documentation doesn't mention

How to read Call-Info Header from Invite Message using sipml5

由 我只是一个虾纸丫 提交于 2019-12-13 07:19:36
问题 I use sipml5 with freeswitch and I need to detect when call should be answered automatically. The only part where I can get it from is SIP Invite message: recv=INVITE sip:username@IP:50598;transport=ws;intercom=true SIP/2.0 Via: SIP/2.0/WSS IP;branch=z9hG4bKd451.8dc49598935d4ebdf937de014cf1d922.0 From: "Device QuickCall"<sip:NUMBER@DOMAIN>;tag=68rtr6c12v9em To: <sip:michaltesar2@IP:50598;transport=ws> Contact: <sip:mod_sofia@IP:11000> Call-ID: dcd8fb4d69f0850840a743c152f4f7358a21-quickcall

Freeswitch performance improved after changing default password

由 梦想的初衷 提交于 2019-12-13 07:14:47
问题 I am new to Freeswitch. After default installation of Freswitch 1.4, I noticed big latency while performing simple telephony operations. i.e. when I called one extension from another, the called number started ringing 10 seconds after initialization of call. Same happened with conference bridge, it took 10 seconds for a callee to enter in conference. Same time my colleague's Freeswitch was performing well. I inquired him and came to know he only changed default password. To my surprise when I

Convert pdf to Tiff with same quality

由 强颜欢笑 提交于 2019-12-13 03:34:45
问题 We are using following shell script to convert pdf attachment in tiff but having some issue with quality, So can you please check below shell script and let us know anything where we can improve quality as well as compress file size too as much as possible while conversation. shell_exec('/usr/bin/gs -q -sDEVICE=tiffg4 -r204x392 -dBATCH -dPDFFitPage -dNOPAUSE -sOutputFile=america_out7.tif america_test.pdf'); We have tried following command and seems quality is better but when we are going to

Freeswitch pauses on check_ip at boot on centos 7.1

由 落爺英雄遲暮 提交于 2019-12-12 06:24:03
问题 During an investigation into a different problem (Inconsistent systemd startup of freeswitch) I discovered that both the latest freeswitch 1.6 and 1.7 paused for several minutes at a time (between 4 and 14) during boot up on centos 7.1. Whilst it was intermittent, it was as often as one time in 3 or 4. Running this from the command line : /usr/bin/freeswitch -nonat -db /dev/shm -log /usr/local/freeswitch/log -conf /usr/local/freeswitch/conf -run /usr/local/freeswitch/run caused the following

Freeswitch and webRTC: media rejected with 488

由 ▼魔方 西西 提交于 2019-12-11 20:35:08
问题 I can register from my webclient to my freeswitch. But, when I try to make call the call gets rejected with 488 not acceptable here. From freeswitch console log im getting. 2014-07-22 22:03:59.673585 [DEBUG] switch_core_state_machine.c:53 sofia/internal/alice@192.168.146.133 Standard REPORTING, cause: INCOMPATIBLE_DESTINATION I added < action application="export" data="rtp_secure_media=true" /> with my extension; but no luck. below is the SDP of my INVITE v=0 o=Mozilla-SIPUA-31.0 26508 1 IN

sip 180 183区别

由 淺唱寂寞╮ 提交于 2019-12-11 10:07:22
sip 180 183区别： 180 不带sdp, 183带sdp信息； 如果A的SIP终端收到183，它就协商媒体，将B端发过来的Early Media在自己的扬声器里放出来；但如果收到的是180，没有SDP就没法协商媒体，因此，B就没法给A发Early Media了。怎么办，总不能让主叫用户干等着啊，所以，A的话机在这种情况下能自己产生一个回铃音，或任何用户在A话机上设置的音乐 首先，我们先看一种熟悉的情况。FreeSWITCH可以假装它就是B，这样，配置方法跟上面讲的基本一样，只是它在假装后还要假戏真做，要用bridge这个Application再去呼叫B，并把电话接通。 <action application="ring_ready"/> <action application="sleep" data="2000"/> <action application="answer"/> <action application="playback" data="/tmp/hello.wav"/> <action application="bridge" data="user/B"/> 所以在上面的配置中，至于是回180还是183，配置方式跟上面讲的一模一样，就没必要多说了。 其次，FreeSWITCH心情好，想听听B的意见。如果它即不执行ring_ready，也不执行answer

FreeSwitch - It doesn't connect MySQL database using xml_mod_curl

由 偶尔善良 提交于 2019-12-11 09:22:14
问题 After install FreeSwitch, I tried to connect user account using xml_mod_curl. I followed below instruction to connect MySQL database. http://saevolgo.blogspot.kr/2012/07/freeswitch-with-sip-users-in-mysql-mod.html I could see that it actually read XML to sign in SIP account with 1000-1019 username. However, it doesn't look like it connects MySQL database. I failed to sign in new user and new password that is saved in MySQL column. here is global_defines.php if (basename($_SERVER['PHP_SELF'])

How to record in async mode in freeswitch

由 喜夏-厌秋 提交于 2019-12-11 09:11:05
问题 I am trying to execute record command in async mode from Java esl, the reason is I have to play music on hold when ever request is processing, and play the wave file to theuser, stopping the moh, I have tried in sync mode but it did not work, I have tried the same in async mode using event lock for play and record, play and moh works fine but the problem is while recording,record command is called but not being executed, the same record command executed properly in sync mode, Please help I

Get session variable using session uuid

由 社会主义新天地 提交于 2019-12-11 08:09:16
问题 How am i able to get session variable using just session uuid (using lua). So for example, we have session of leg_a (when someone connect sip phone). When other side answers, we will be within other session (session of leg_b). Using just, session:getVariable("variable_name") will not help in that case because session refers to the current session. How am i able to use lua to get variable name when i know session uuid. (so i can get other sessions variable, even we are within current session
订阅 freeswitch