February 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

So you wanna setup your own CA for WSS/SSL/TLS?
by Brian West 02 Jul '15

02 Jul '15

Someone should probably turn this into a nice how-to: Here is how I did it. wget http://www.openssl.org/contrib/ssl.ca-0.1.tar.gz tar zxfv ssl.ca-0.1.tar.gz cd ssl.ca-0.1/ perl -i -pe 's/md5/sha1/g' *.sh perl -i -pe 's/2048/2048/g' *.sh ./new-root-ca.sh ./new-server-cert.sh self.bkw.org ./sign-server-cert.sh self.bkw.org cat self.bkw.org.crt self.bkw.org.key > /usr/local/freeswitch/certs/wss.pem Setup Apache: default-ssl: SSLCertificateFile /usr/local/freeswitch/certs/wss.pem SSLCertificateKeyFile /usr/local/freeswitch/certs/wss.pem SSLCertificateChainFile /usr/local/freeswitch/certs/wss.pem Setup Sofia TLS: cat self.bkw.org.crt self.bkw.org.key > /usr/local/freeswitch/certs/agent.pem cat ca.crt > /usr/local/freeswitch/certs/cafile.pem vars.xml: <X-PRE-PROCESS cmd="set" data="internal_ssl_enable=true"/> <X-PRE-PROCESS cmd="set" data="external_ssl_enable=true"/> Restart FreeSWITCH. Now make sure your system has ca.crt imported so it will trust your new found hotness. TEST: openssl s_client -connect self.bkw.org:443 openssl s_client -connect self.bkw.org:8082 Depending on what you've setup you'll see: subject=/C=US/ST=Oklahoma/L=McAlester/O=Tonka Truck/OU=Secure Web Server/CN= self.bkw.org/emailAddress=brian@bkw.org issuer=/C=US/ST=Oklahoma/L=McAlester/O=Whizzzzzzy Bang Bang/OU=Certification Services Division/CN=WBB Root CA/emailAddress= brian(a)bkw.org Or there abouts. -- *Brian West* brian(a)freeswitch.org *Twitter: @FreeSWITCH , @briankwest* http://www.freeswitchbook.com http://www.freeswitchcookbook.com *T:*+19184209001 | *F:*+19184209002 | *M:*+1918424WEST (9378) *iNUM:*+883 5100 1420 9001 | *ISN:*410*543 | *Skype:*briankwest

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Re - British Telecom (BT) SIP test list before allowing a server on their IP Telephony network
by Andrew Keil 02 Jun '15

02 Jun '15

To FreeSWITCH users, Re - British Telecom (BT) SIP test list before allowing a server on their IP Telephony network. I wondered if someone can take a quick look at this test case spreadsheet from BT in the UK to see if a standard FreeSWITCH installation (current production release) will cover off all these test cases. I assume the answer is yes! However I thought that I would ask in advance. Kind Regards, Andrew Keil

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Update CLI on Freeswitch originated call
by Joshua Gigg 22 May '15

22 May '15

Hi, Is it possible to update the CLI at will once a Freeswitch originated call has been answered? I know it can update during a transfer, but I want to be able to control it directly myself. -- Joshua Gigg

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Receive chat
by Sameer Khan 05 May '15

05 May '15

Hi, Is there any command which can receive chat from sip channel so I can put that received message in to a variable.

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Unable to load mod_java.
by John 21 Apr '15

21 Apr '15

Hi All I have built mod_java and have mod_java.so in /usr/local/freeswitch/mod/ but when I am trying to load mod_java from freeswitch terminal like : load mod_java getting following error : 2015-02-07 15:01:13.175331 [INFO] mod_enum.c:880 ENUM Reloaded 2015-02-07 15:01:13.175331 [NOTICE] modjava.c:366 Java Framework Loading... 2015-02-07 15:01:13.175331 [INFO] switch_time.c:1411 Timezone reloaded 1781 definitions +OK Reloading XML -ERR [module load file routine returned an error] 2015-02-07 15:01:13.215310 [CRIT] switch_loadable_module.c:1447 Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** I am using OpenJDK version : java version "1.6.0_33" OpenJDK Runtime Environment (IcedTea6 1.13.5) (6b33-1.13.5-1ubuntu0.14.04) OpenJDK 64-Bit Server VM (build 23.25-b01, mixed mode) and my conf/autoload_configs/java. conf.xml reads as : <configuration name="java.conf" description="Java Plug-Ins"> <javavm path="/usr/lib/jvm/java-6- openjdk-amd64/jre/lib/amd64/jamvm/libjvm.so"/> <options> <option value="-Djava.class.path=/usr/local/freeswitch/scripts/freeswitch.jar"/> <option value="-agentlib:jdwp=transport=dt_socket,server=y,suspend=n,address=0.0.0.0:8000"/> </options> <startup class="org/freeswitch/example/ApplicationLauncher" method="startup"/> </configuration> Can someone help me to resolve this error? I tried googling this issue and accordingly made above config changes. Thanks, John

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eliminating fax header line
by Bruce Lefko 26 Mar '15

26 Mar '15

I'd like to completely eliminate the header line sent at the top of each fax page using spandsp. i've set fax-header to a null string, but that doesn't seem to be removing it, and it defaults back to 'Spandsp fax header'. This seems to be the default in the source code in 1.2. What do I have to do to eliminate the header line? Do I have to make changes to freeswitch source? Thanks!

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Port changes in multiple 183s cause no audio after 200
by Jeff Pyle 17 Mar '15

17 Mar '15

Hello, The following is on FreeSWITCH Version 1.5.15b+git~20150126T215733Z~c16f9ec1d9~64bit (git c16f9ec 2015-01-26 21:57:33Z 64bit). The design goal for this configuration is that of a simple transcoding SBC. SIP calls arrive with various supported codecs, SIP calls bridge out on PCMU. No users, no auth, etc. Overall, it seems to work but there is one call flow I'm struggling with. The B-leg of calls are bridged to a PSTN gateway. If the gateway's signaling follows 100, 183, 200, all is well. But if the gateway sends multiple 183s with different RTP ports, there is no audio when the call goes to 200. See the following example: - Gateway signals 183 with SDP indicating audio on port 16384. - Gateway signals 183 with SDP indicating audio on port 16386. - Gateway signals 200 with SDP indicating audio on port 16384 (same as original 183). In the debug I see where it detects the port change from 183 #1 to 183 #2. As such, I hear early media from both until the 200 OK. When the call connects, I see no such port change in the debug, and since it's still listening on the wrong port (from 183 #2), there is no audio. I've seen some older posts where Anthony seemed against even the port change from 183 #1 to 183 #2, yet that seems to work okay today. It just doesn't sense the port change from 183 --> 200. I don't know if this is a feature, bug, or misconfigured option. Thoughts are welcome! - Jeff

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Values of read_codec/write_codec on different FS versions...
by Vladimir Getmanshchuk 13 Mar '15

13 Mar '15

Hello Everyone! I have two installations of FS with absolutely identical configurations. Both has SIP profiles with proxy-media enabled. But on freeswitch@internal> version FreeSWITCH Version 1.4.15~64bit ( 64bit) I have values in read_codec/write_codec variables at CDRs: "read_codec":"PROXY","write_codec":"PROXY" but on another one freeswitch@internal> version FreeSWITCH Version 1.4.5~64bit ( 64bit) I have: "read_codec":"G729","write_codec":"G729", "read_codec":"PCMA","write_codec":"PCMA", etc... So Why? -- Yours sincerely, Vladimir Getmanshchuk

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any handset recommendations that operate like polycom+cisco?
by Nathan Neulinger 10 Mar '15

10 Mar '15

Looking for any handset/desk phone recommendations that have the "every line key can be it's own SIP account, even on expansion modules" behavior. Yealink units for example only allow a set number of lines, and the rest don't operate as line keys, so even though you can have a 30 + button phone, it's not usable for a receptionist or secretary that may handle calls for 15 different people with their own direct lines. With the polycom and cisco (sccp) phones, you can define N different SIP accounts. Are there any other phones out there that operate this way that anyone can recommend? -- Nathan ------------------------------------------------------------ Nathan Neulinger nneul(a)mst.edu Missouri S&T Information Technology (573) 612-1412 System Administrator - Architect

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Real-time billing application for the FreeSWITCH (mod_lua, mod_perl or ESL)
by Sami Montour 04 Mar '15

04 Mar '15

Hello Everyone, I am a long time user of OpenSIPS and have recently been tasked with developing a call control/real-time billing application for the FreeSWITCH. I would like to develop the application in either C/C++, Perl or lua (totally new to lua) in this order. I have used ESL to develop some basic functionality of the application using Perl. For instance, when the call comes in, I pass over control from FreeSWITCH to the call control application using outbound TCP socket. Then the call control application listens for CHANNEL_ANSWER event to start billing process and CHANNEL_HANGUP event to stop billing. The application does some database work as well to retrieve and check user status and balance and all that stuff. My main concern is with load testing because of the TCP connection between the FreeSWITCH and call control application and also TCP connection to database to retrieve subscriber info. The basic functionality seems to work fine but haven't tested it with high call volume, which I am about to do. The last couple of days I have been experimenting with mod_lua but can't find a method under "Session" to get information related to CHANNEL_ANSWER and CHANNEL_HANGUP events. The "session:geVariable" does not have variables related to session answer time or hangup time, something similar to "$e->getHeader("Caller-Channel-Answered-Time")" in ESL. I must be missing something very simple here. Again, my experience with mod_lua is only a couple of days old. I am new Freeswitch and thought I would seek the advice of all the experts and people with more experience with Freeswitch. For this type of application, am I better off with mod_lua, or mod_perl, or shall I stick with ESL using Perl? How about ESL with C/C++? Will I gain some performance gain there? Any feedback would be appreciated. Thanks. Sam

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