April 2015 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Current production release 1.4.15 or 1.4.18?
by Andrew Keil 22 Apr '15

22 Apr '15

To FreeSWITCH Users, Simple question. Is the current FreeSWITCH release to use in production 1.4.15 or 1.4.18? I noticed that 1.4.18 was just released on March 26th 2015, however the main FreeSWITCH.org webpage still shows Release: 1.4.15. Kind Regards, Andrew Keil Visytel Pty Ltd

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att_xfer and blind transfer
by Ali Pey 22 Apr '15

22 Apr '15

Hello, Is there anyway to do blind transfer with att_xfer? Also, based on wiki for att_xfer, to cancel transfer and go back to the caller, you can press # ONLY after C-leg (last party) answers. However, most users would need to be able cancel transfer while the last party is ringing and hasn't answered. Is that possible? Thanks, Ali Pey

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patch for sofia_sip_i_invite to support replaces with action
by Luis Azedo 22 Apr '15

22 Apr '15

Hi, i need a inline custom action passed with an INVITE msg with a replaces header or a better to use this scenario. right now, for calls with replaces header there is support for sla, intercept and attendant transfer. my problem is that intercept requires the bridge_uuid which is not set on one leg calls (on hold playing music) so the intercept condition won't be hit. the attendant transfer takes app/data applies and then hangs up the call. i need a simple answer,intercept:(sip->sip_replaces->rp_call_id) instead of bridge_uuid. i tried to set signal_bond and originate_signal_bound on the channel to see if i could bypass this, but with no luck. i created a small patch that i am attaching here with the hope that FreeSWITCH will look and possibly merge it. Best

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att_xfer + leg A
by Денис 22 Apr '15

22 Apr '15

How to make att_xfer from leg A? It makes for B only...

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ClueCon Weekly Today at 1PM EST!
by Ken Rice 22 Apr '15

22 Apr '15

Greetings FreeSWITCHers! Today Wed, April 22nd @ 1PM EST Join us at https://conference.freeswitch.org/ for ClueCon Weekly. Click the ³Call 888² button to join us. This week Chris Rienze (@crienzo), author of mod_rayo, and regular contributor to FreeSWITCH, will be showing off some good things with lua, graylog, and pcapsipdump. Also have your Headset and WebCams ready and use Chrome when calling 888 to get in on the Video side of the conf! -- Ken http://www.FreeSWITCH.org http://www.ClueCon.com http://www.OSTAG.org irc.freenode.net #freeswitch Twitter: @FreeSWITCH

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Challenges
by Tony Mendez 22 Apr '15

22 Apr '15

Hi all, I have installed the latest version of freeswitch, with fusionpbx. It is an upgraded install. However, some of the extensions do not generate a ring tone when calling in. Specifically, it is one extension and one ring group. Everything else is fine ( I think). Would be good if someone can help with 1) direct me to the relevant sections in the freeswitch manual, or 2) just tell me how to make it ring. In fusionpbx, it is set to “default” for that specific ring group, same as the other ring groups. Gracias, Tony

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Duplicate request from FreeSWITCH in the radius server
by Густаво Силва 22 Apr '15

22 Apr '15

I am getting duplicate request from my freeswitch: Info: rlm_radutmp: Login entry for NAS localhost port 0 duplicate I am using freeswitch and freeradius in the same machine. The radius is authenticating the request but why the freeswitch is sending twice? Is something in the radiusclient that I shoud change? Regards, Gustavo

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Trunk Calls
by Ali Jibran 22 Apr '15

22 Apr '15

Hi all. I have voipfone trunked to freeswitch. Everything works fine. But now I've bought another number from the same provider. The call comes in but executes the same IVR at 1234. How do I change the dialplan that the call executes different IVR depending on the number dialed. here is the gateway conf: <include> <gateway name="voipfone"> <param name="username" value="XXXXX"/> <param name="realm" value="sip.voipfone.net"/> <param name="password" value="XXXXX"/> <param name="extension" value="1234"/> <param name="expire-seconds" value="60"/> </gateway> </include>

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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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perl ESL outbound socket, no DTMF if not async ???
by Giovanni Maruzzelli 21 Apr '15

21 Apr '15

Ciao FreeSWITCHers, on FS 1.4.18: using perl ESL outbound socket, I see FS receives DTMF 2833 only when socket is declared "async" in dialplan. NB: not only the outbound socket does not receives DTMF events, is FreeSWITCH itself that does not receives DTMFs. Is this correct? Is this the expected behavior? If I delete "async" from dialplan, DTMFs are not received es: <action application="socket" data="127.0.0.1:8040 async"/> works, while with <action application="socket" data="127.0.0.1:8040"/> no DTMFs at all. ==== btw, this is an example of outbound_socket in perl ESL #!/usr/bin/perl require ESL; use IO::Socket::INET; my $ip = "127.0.0.1"; my $sock = new IO::Socket::INET ( LocalHost => $ip, LocalPort => '8040', Proto => 'tcp', Listen => 1, Reuse => 1 ); die "Could not create socket: $!\n" unless $sock; for(;;) { my $new_sock = $sock->accept(); my $pid = fork(); if ($pid) { print "New child pid $pid created...\n"; close($new_sock); next; } my $fd = fileno($new_sock); my $con = new ESL::ESLconnection($fd); my $info = $con->getInfo(); my $uuid = $info->getHeader("unique-id"); printf "Connected call %s, from %s\n", $uuid, $info->getHeader("caller-caller-id-number"); $con->sendRecv("myevents $uuid"); $con->setEventLock("1"); # need this for forcing execution order on async, has no effect if no "async" in dialplan $con->execute("sleep","500"); $con->execute("answer"); $con->execute("sleep","500"); $con->execute("playback", "ivr/ivr-welcome_to_freeswitch.wav"); $con->execute("sleep","500"); $con->execute("playback", "ivr/ivr-finished_pound_hash_key.wav"); while($con->connected()) { my $e = $con->recvEvent(); if ($e) { my $name = $e->getHeader("event-name"); print "EVENT [$name]\n"; if ($name eq "DTMF") { my $digit = $e->getHeader("dtmf-digit"); my $duration = $e->getHeader("dtmf-duration"); print "DTMF digit $digit ($duration)\n"; $con->execute("hangup"); } } } print "BYE\n"; close($new_sock); } -- Sincerely, Giovanni Maruzzelli Cell : +39-347-2665618

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