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Bridge in event socket - async mode
by Nagalenoj H. 14 Oct '09

14 Oct '09

Hi, I'm using event outboud socket(perl) in async mode. Scenario where I face problem: When a call comes to an extension(1000), my program will play some message to the user, and get some DTMF. I'll get the DTMF event(as I'm in async mode) and store the digits in a variable. When he presses "#", I call $con->execute("bridge","user/$dtmf"), to bridge him to the dialed number. It is bridging sometimes, and sometimes it is not bridging. Is it the problem with async mode?? I also set $con->setEventLock("true"), before executing the bridge. But still the problem persists. I've seen the logs too, I didn't find the bridge application getting executed. What makes the problem here..?? The freeswitch log, when bridge is happening: http://pastebin.freeswitch.org/10704 The freeswitch log, when bridging is not happening: http://pastebin.freeswitch.org/10705 Code: # some statements if($name eq "CHANNEL_BRIDGE") { # checking for events . } elsif ($name eq "DTMF") { my $digit = $e->getHeader("dtmf-digit"); if($digit eq '#') { print "Going to bridge\n"; $con->execute("bridge","user/$dtmf_digit"); } else { $dtmf_digit.=$digit; } } -- Regards, Nagalenoj H.

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openzap Failure opening channel error
by lakshmanan ganapathy 14 Oct '09

14 Oct '09

Hi, Again I was struck in a problem, Here is the scenario. On incomming call, I just call an event outboud socket. But what happens is, for the first 15 call, it is working fine. But from the 16th call to 30th call, it says the below error. 2009-10-07 15:07:48.201846 [WARNING] ozmod_libpri.c:761 --Failure opening channel 1:17 (ignored) 2009-10-07 15:07:55.381861 [WARNING] ozmod_libpri.c:761 --Failure opening channel 1:18 (ignored) 2009-10-07 15:07:58.569774 [WARNING] ozmod_libpri.c:761 --Failure opening channel 1:19 (ignored) 2009-10-07 15:08:01.37824 [WARNING] ozmod_libpri.c:761 --Failure opening channel 1:20 (ignored) 2009-10-07 15:08:03.129846 [WARNING] ozmod_libpri.c:761 --Failure opening channel 1:21 (ignored) 2009-10-07 15:08:04.825851 [WARNING] ozmod_libpri.c:761 --Failure opening channel 1:22 (ignored) 2009-10-07 15:08:06.289977 [WARNING] ozmod_libpri.c:761 --Failure opening channel 1:23 (ignored) 2009-10-07 15:08:07.761961 [WARNING] ozmod_libpri.c:761 --Failure opening channel 1:24 (ignored) 2009-10-07 15:08:09.737944 [WARNING] ozmod_libpri.c:761 --Failure opening channel 1:25 (ignored) 2009-10-07 15:08:11.462018 [WARNING] ozmod_libpri.c:761 --Failure opening channel 1:26 (ignored) 2009-10-07 15:08:13.566024 [WARNING] ozmod_libpri.c:761 --Failure opening channel 1:27 (ignored) 2009-10-07 15:08:15.430163 [WARNING] ozmod_libpri.c:761 --Failure opening channel 1:28 (ignored) 2009-10-07 15:08:17.446103 [WARNING] ozmod_libpri.c:761 --Failure opening channel 1:29 (ignored) 2009-10-07 15:08:19.430118 [WARNING] ozmod_libpri.c:761 --Failure opening channel 1:30 (ignored) 2009-10-07 15:08:21.358121 [WARNING] ozmod_libpri.c:761 --Failure opening channel 1:31 (ignored) But for the next call, when it is opening channel 1:1, it is executing my dial plans. I don't know why it failed when it is choosing 1:17-1:31. Any one has any idea. Below are my configuration details: openzap.conf [span zt PRI_1] trunk_type => e1 b-channel => 1:1-15 d-channel=> 1:16 b-channel => 1:17-31 openzap.conf.xml <configuration name="openzap.conf" description="OpenZAP Configuration"> <settings> <param name="debug" value="7"/> </settings> <libpri_spans> <span name="PRI_1"> <param name="node" value="cpe"/> <param name="switch" value="euroisdn"/> <param name="dialplan" value="XML"/> <param name="context" value="default"/> </span> </libpri_spans> </configuration> zaptel.conf span=1,1,0,ccs,hdb3 bchan=1-15,17-31 dchan=16 loadzone = us defaultzone = us oz list span: 1 (PRI_1) type: isdn chan_count: 47 dialplan: XML context: default dial_regex: fail_dial_regex: hold_music: analog_options none

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Re: [Freeswitch-users] Mod_java loading error
by dipen＠entvoice.com 14 Oct '09

14 Oct '09

Hi Baskar, Can you tell us how did you solve your problem of Loading mod_java. I mean which steps to follow to solve the problem. Currently i am getting the error.. [ERR] modjava.c:133 load_config() Error loading /opt/SDK/jdk/jre/lib/i386/client/libjvm.so [CRIT] switch_loadable_module.c:871 switch_loadable_module_load_file() Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** Thanks & Regards Dipen Velani. On Wed 15/04/09 3:15 AM , Baskar <yudha2008(a)gmail.com> wrote: > Hi, > > Now i can able to load the mod_java in the freeswitch console. > > After that i have followed these method to run the PhoneTest.java > 1) VERIFIED MY CLASSPATH IN THE JAVA.CONF.XML: > > 2)MY PHONETEST.CLASS IS LOCATED IN /USR/LOCAL/FREESWITCH/SCRIPT - > DIRECTORY, SAME AS WHERE FREESWITCH.JAR IS LOCATED > > 3) I CREATED PHONETEST.JAR FILE FROM PHONETEST.CLASS > > 4)CHANGED MY DIALPLAN TO: > Then i have complied and Run the PhoneTest.java by this methods > > Method1: > [localhost scripts]# javac -cp freeswitch.jar PhoneTest.java > [localhost scripts]# java -cp freeswitch.jar PhoneTest > Exception in thread "main" java.lang.NoClassDefFoundError: PhoneTest > Caused by: java.lang.ClassNotFoundException: PhoneTest > at > java.net.URLClassLoader$1.run(URLClassLoader.java:200) > at java.security.AccessController.doPrivileged(Native > Method) > at > java.net.URLClassLoader.findClass(URLClassLoader.java:188) > at > java.lang.ClassLoader.loadClass(ClassLoader.java:306) > at > sun.misc.Launcher$AppClassLoader.loadClass(Launcher.java:276) > at > java.lang.ClassLoader.loadClass(ClassLoader.java:251) > at > java.lang.ClassLoader.loadClassInternal(ClassLoader.java:319) > > Method2: > [localhost scripts]# javac -cp PhoneTest.jar:freeswitch.jar > PhoneTest.java > [localhost scripts]# java -cp PhoneTest.jar:freeswitch.jar PhoneTest > Exception in thread "main" java.lang.NoSuchMethodError: main > > But I get these above errors .So Please Correct me where i am wrong > and assist me how to compile and run the PhoneTest.java. > -- > Warm Regards, > N.Baskar > > > ---- Msg sent via @Mail - http://atmail.com/

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Preannouncing a message before making a call
by Simon J Mudd 14 Oct '09

14 Oct '09

I'm looking to migrate to Freeswitch from Asterisk (SOHO usage). One thing I have configured at the moment which I'm unsure how to do in FreeSwitch is the following. When the call's dial pattern is recognised I've been getting Asterisk to say: "Dialing <destination> via <gateway>" where <destination> might be a country, or area and <gateway> may be the name of one of several SIP gateways I have configured. Each phrase has been a prerecorded wav file played in sequence. [macro-XXXXX] exten => s,1,Playback(/etc/asterisk/wav/dialing) exten => s,2,Playback(/etc/asterisk/wav/${ARG2}) # Name of destination exten => s,3,Playback(/etc/asterisk/wav/via) exten => s,4,Playback(/etc/asterisk/wav/XXXXX) # Gateway name exten => s,5,Dial(SIP/XXXXX/${ARG1},60,tT) # ARG1 = Destination number # actual usage exten => _0044.,1,Macro(XXXXX,${EXTEN},touk) How would I approach doing something like this in FreeSwtich? Thanks for any pointers. Simon

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T.38 in sip header
by Dome Charoenyost 14 Oct '09

14 Oct '09

Dear all, Is posible to check T.38 in sip header? i want to disconnect call when T.38 not support in gateway and try next gateway BG Dome C.

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FS Slide deck?
by Gabriel Gunderson 14 Oct '09

14 Oct '09

Hey all, The 3rd annual UTOSC (Utah Open Source Conference) [1] starts today. It's a conference that gives Open Source users/hackers a chance gather together to mingle and share. It's a lot of fun. Anyway, this year I've offered to work with a friend of mine to lead a BoF (Birds of a Feather) discussion on Open Source Telephony [2]. While it should mostly be group discussion, you never know when people are going to get shy and clam up. *If* they do, it would be nice to have some slides to fill that time. Rather than make some from scratch (that are unlikely to get used), I thought I'd ask here and see if anyone has a FS slide deck that they wouldn't mind sharing. Thanks, Gabe 1) http://2009.utosc.com/pages/home/ 2) http://2009.utosc.com/presentation/136/

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FreeSWITCH Conf Call Agenda For October 16th
by Michael Collins 13 Oct '09

13 Oct '09

FYI, The agenda for this week's call is here: http://wiki.freeswitch.org/wiki/FS_weekly_2009_10_16 Please feel free to add agenda items and updates as you see fit. -Michael

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FS Extension Groups Documentation
by Jerry Richards 13 Oct '09

13 Oct '09

Is there a link to the documentation of FS groups and what can be done with a group (i.e. capabilities)? Thanks And Best Regards, Jerry

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Registering a large number of SIP users
by Muhammad Shahzad 13 Oct '09

13 Oct '09

Hi, I am creating a load test setup for FreeSWITCH using Sofia SIP. I have two machines both configured with mod_xml_curl, one machine (lets call it SIP Server) has 100 SIP accounts. Now i want to register second machine (lets call it SIP Client) to all these 100 SIP accounts on first machine. How can i do that? One approach that i can think of is that to create a new profile (or use existing external profile) on SIP Client and add all accounts to it as gateways having, *<param name="register" value="true">* define in their configuration. Is my approach correct? Are there any better ways to do this? Thank you. -- ________________________________________________________ | | | FATAL ERROR --- O X | |_______________________________________________________| | You have moved the mouse. | | Windows must be restarted for the changes to take effect. | | <OK> | ####################################/ Muhammad Shahzad ----------------------------------- CISCO Rich Media Communication Specialist (CRMCS) CISCO Certified Network Associate (CCNA) Cell: +92 334 422 40 88 MSN: shari_786pk(a)hotmail.com Email: shaheryarkh(a)googlemail.com

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606 error
by srinivasula reddy 13 Oct '09

13 Oct '09

Hi, two users are registered in freeswitch, when i making call to another user i am getting 606 error, any help -- Srinivasula Reddy K

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