October 2009 - FreeSWITCH-users - FreeSWITCH Mailing Lists

how to make more profiles listen on port 5060 ?
by Ryanny Lin 14 Oct '09

14 Oct '09

Hi, all, I read the article "Performance testing and configurations" on the wiki. Here is a Recommended from wiki: "libsofia only handles 1 thread per profile, so if that is your bottle neck use more profiles" How can I use more profiles and listen on port 5060? Is it possible? I used sipp to tested SIP REGISTER performance which is about 165 cps. It is about 260 cps when turned lots of modules. Does anyone test higher than this? -- Sincerely regards, Ryanny

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how can i get the caller-number when the caller in the conference
by qian ma 14 Oct '09

14 Oct '09

hi all: i can get the member_id in the conference,but how can can i get the number?? thx!

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Some help with my post-paid billing project
by Diego Viola 14 Oct '09

14 Oct '09

Hello, I'm trying to write a post-paid billing script, I have the CDR on my database and also a "rates" table, the CDR contains fields like caller_destination_number, variable_duration, etc. and the rates table contains: destination, prefix, rate (cost). The problem is that I can't just strip the destination number to take the prefix from it because I have to deal with destination numbers from different countries and they all have different prefix lengths... so I need to find another way to take the prefix from the destination number. Any ideas how to do this? Thanks, Diego

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Groups information in sqllite
by srinivasula reddy 14 Oct '09

14 Oct '09

Hi, can any know where group information is exactly stored in sqllite database, i have seen sip_registration here i can find the registered users, in the same way how i can i find the group information, and which user belongs to which user? any help would be great. thanks -- Srinivasula Reddy K

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mod_socket: custom timeout for HEARTBEAT event
by Artem Shiyanov 14 Oct '09

14 Oct '09

Hi all! As it stays in wiki: ... HEARTBEAT Status information for freeswitch trigerred by freeswitch's heartbeat every 20 seconds. ... Is there any way to customize timeout of HEARTBEAT events? Thanks in advance, Artem

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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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