June 2010 - FreeSWITCH-users - FreeSWITCH Mailing Lists

Erlang Examples
by David Swardstrom 27 Jun '10

27 Jun '10

I have been using JavaScript to handle a Conferencing application that started with the conf-ivr.js example program but is significantly more complex. This has been fun even though I had never used JavaScript before this year. However, there are things that seem to not be possible using JavaScript. I need to interact with several web based applications for several reasons and also need to provide some time based interactions with FreeSwitch and/or artifacts (Database entries, files of recorded conferences, etc). We (RemoteLink) have decided that the best solution for this support is to use an Erlang program and mod_erlang_event. So now I need to learn another language. But one thing that I do not find one the FreeSWITCH site is any Erlang examples. Are there some sample programs available such as one that would look for a certain type of event and print it out? I have found some semi-samples in the freeswitch-users archives but am somewhat ambivalent about using any of these without permission. Regards, David Swardstrom (profile)http://wiki.freeswitch.org/wiki/User:Dswardstrom

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ringback answer vs preanswer
by Stephen Cattaneo 26 Jun '10

26 Jun '10

i am trying to understand why when i do session.preanswer ringback works when i bridge the call but when i do session.answer it does not. Thanks, Stephen C -All of my email addresses go to the same place -Save Paper, think before you print

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event-lock in mod_erlang_event
by Seven Du 25 Jun '10

25 Jun '10

Hi, As I can see, You cannot do this in inbound erlang sendmsg(FS, uuid, playback, "1.wav"); sendmsg(FS, uuid, playback, "2.wav"); sendmsg(FS, uuid, transfer, "xxxxx because it's async, and it will play 2.wav immediately. 1) Sure if I know the length of 1.wav I can sendmsg(FS, uuid, playback, 1.wav sleep(3000 sendmsg(FS, uuid, playback, 2.wav 2) Or I could wait the execute_complete event which will be a little complicated According to http://wiki.freeswitch.org/wiki/Event_socket_outbound Is it possible to send a event-lock param to lock the message temporarily? like sendmsg(FS, UUID, App, Args, [{"event-lock", "true"}]). -- Blog: http://www.dujinfang.com Proj: http://www.freeswitch.org.cn

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try to run mod_java but loading issue
by Ravi Kuru 25 Jun '10

25 Jun '10

Hi, I try to run PhoneTest on freeswitch and I followed the instruction but I got this error on freeswitch.log file when i start the freeswitch. 2010-06-24 15:01:47.096906 [ERR] modjava.c:124 Error loading /usr/telcan/jdk1.6.0_10/jre/lib/i386/server/libjvm.so 2010-06-24 15:01:47.096942 [CRIT] switch_loadable_module.c:882 Error Loading module /usr/local/freeswitch/mod/mod_java.so **Module load routine returned an error** what was the issue? Ravi

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Voice mail
by mattias jonsson 25 Jun '10

25 Jun '10

Have freeswitch integrated voicemail?

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Looking for FreeSWITCH user in Öresund-area (Copenhagen/Malmö)
by Harry Vangberg 25 Jun '10

25 Jun '10

Hello I am looking for an experienced FreeSWITCH user/administrator in the greater Öresund area (Copenhagen/Malmö) to join an interesting project. Please let me know if you are interested or know anybody that are. Best regards, Harry Vangberg <harry(a)vangberg.name> http://harry.vangberg.name

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Using mod_unimrcp
by Mayank Shrivastava 25 Jun '10

25 Jun '10

Hi, I am an undergraduate student, presently working on an academic project as part of my intern. I need to run ASR using our Nuance Speech Server, on a number of .wav files that I have. I have tried using UniMRCP, but am running into problems, as i persistently get a Completion Cause Error number 006 on the files that I run the ASR on. I know that FreeSWITCH has the mod_unimrcp module built in, and I was wondering how I could use FreeSWITCH for the same. >From http://lists.freeswitch.org/pipermail/freeswitch-users/2010-June/059335.html I have understood how to build the mod_unimrcp mod, but how do I proceed? Thanks, Mayank

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Re: [Freeswitch-users] Internal Profile Not Valid
by Steve Butterfield 25 Jun '10

25 Jun '10

I was wondering where to change the /default/default/? The really strange thing is that the system has corrected itself as of this evening without my intervention. Could I have a bad build? Is there a way to update freeswitch instead of a full re-install? I am using Ubuntu 10.04 Would it help to post logs? If so which ones. Thank you, Steve Steve Butterfield - steve(a)barrettsystems.com ---------- Forwarded message ---------- > From: Steven Ayre <steveayre(a)gmail.com> > To: freeswitch-users(a)lists.freeswitch.org > Date: Thu, 24 Jun 2010 09:59:19 +0100 > Subject: Re: [Freeswitch-users] Internal Profile Not Valid > I think /default/default/ should just be /default/ > > On 24 June 2010 05:49, Steve Butterfield <steve(a)barrettsystems.com> wrote: > >> I am new to using freeswitch but somehow, overnight my internal sip >> profile became invalid and freeswitch occupies 90+ % of my memory. I have >> looked over my install and everything seems to be in order. One thing that >> stands out during boot up is : >> >> 2010-06-23 21:48:15.225851 [DEBUG] sofia.c:1317 Creating agent for >> internal-ipv6 >> 2010-06-23 21:48:17.766457 [ERR] switch_xml.c:1297 Couldnt open >> /usr/local/freeswitch/conf/dialplan/public/default/default/*.xml (No such >> file or directory) >> 2010-06-23 21:48:17.817992 [ERR] switch_xml.c:1297 Couldnt open >> /usr/local/freeswitch/conf/dialplan/public/default/default/*.xml (No such >> file or directory) >> 2010-06-23 21:48:17.906682 [ERR] switch_xml.c:1297 Couldnt open >> /usr/local/freeswitch/conf/dialplan/public/default/default/*.xml (No such >> file or directory) >> 2010-06-23 21:48:17.967737 [ERR] switch_xml.c:1297 Couldnt open >> /usr/local/freeswitch/conf/dialplan/public/default/default/*.xml (No such >> file or directory) >> 2010-06-23 21:48:18.629030 [DEBUG] sofia.c:1353 Created agent for >> internal-ipv6 >> >> Any help would be appreciated. >> Steve >> >> >>

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ESL Conference
by Samuel Macedo 25 Jun '10

25 Jun '10

Hi, I'm using an Outbound ESL connection. In this scenario, how can I send a call into a conference room? Thanks, -- Samuel Macedo

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HOW SHOW CALLER_ID_NUMBER WITHOUT IP
by Kozak Vladimir 24 Jun '10

24 Jun '10

hi everybody. I have one problem. I need to show in phone caller_id_number without IP address or caller_id_number + domain without IP address. I use api originate command with origination_caller_id_name and origination_caller_id_number parameters. bgapi originate {origination_caller_id_name=125@123.12.13.14,origination_caller_id_number=123@123.12.13.14}[origination_uuid=6daa7b7e-97e4-4790-827d-44ff4f40fd18]sofia/internal/sip:1009@172.26.10.65:61802;rinstance=437cf350c3a4546f &park() FS cuts value of origination_caller_id_name parameter (delete specified ip) - it's ok. And FS deletes specified ip from value of origination_caller_id_number parameter and adds FS-IP - it's bad. how else can I show caller info? how can I show caller domain?

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