- FreeSWITCH-users - FreeSWITCH Mailing Lists

How to start with js
by Tamas 21 Aug '06

21 Aug '06

Hi, I'm experimenting with freeswitch and want to build simple IVR in js, however I don't know how to set up freeswitch for it. I have mod_spidermonkey compiled already. Could anybody tell me what to do next? [I know, there are some examples scripts at http://www.freeswitch.org/eg/js/] Thanks in advance, Tamas

3 3

IAX2 'redirects'
by JP Hindin 11 Aug '06

11 Aug '06

Greetings; I was wondering if FreeSwitch (or, rather, libiax I suppose) had the ability to do a redirect in such a way that FreeSwitch would drop out of the media stream. In other words, on a bridge, will FreeSwitch let the source send the media stream direct to the destination service, without having to proxy the packets - much in the same way a SIP 302 Redirect works. Thanks; JP

3 3

Re: [Freeswitch-users] How to interface with google talk
by danish.samad＠vocalseeds.com 10 Aug '06

10 Aug '06

Hi, Mike, thanks for showing your concern. I was able to successfully bridge a call between a google talk client and asterisk(SIP). Although there were times when asterisk received and accepted the call while the jingle session was dropped. I did not really delve into the issue, so I am not really sure what was causing the issue. There are certain things I noticed, that I would like to know more about. Firstly, when calling from jinlge to sip,I can only map a single jabber account with a SIP account. Is there any way I can bridge multiple calls between Jingle and SIP endpoints, using the current configuration. If not, are you guys planning to implement this feature. Secondly, how can I make calls from sip endpoints to Jingle endpoints. Any configuration tips will be helpful. Regards, Danish > Danish, did you ever get a response to this message that resolved your > issue? We were all at our conference last week, so this may have been > missed. > > Mike > > -----Original Message----- > From: danish.samad(a)vocalseeds.com [mailto:danish.samad@vocalseeds.com] > Sent: Tuesday, August 01, 2006 6:37 AM > To: freeswitch-users(a)lists.freeswitch.org > Subject: [Freeswitch-users] How to interface with google talk > > Hi, > > I am trying to interface with google talk using the following setup > > Freeswitch<->JabberServer<->GoogleTalkServer<->Googletalk client > > Freeswitch will register with Jabberserver using a valid jabber account > on > the Jabber server. Using this account I should be able to login and add > any valid gmail id in its userlist and perform standard operations > (using > the jingle protocol) such as chat, voice, etc. Eventually I intend to > use > Freeswitch as a gateway between SIP and google talk clients, but I first > need to validate the above setup. > > To test the setup out, I created an account on jabber.anywise.com. I > was > able to successfully log into the server using PSI and adding a valid > gmail accounts in my roster. Now when I specify the account in > freeswitch.xml I am not able to register as I donot see the user coming > online in my google talk user list. I just see the following XML > messages > being dumped on the console. > > freeswitch> SEND[<?xml version='1.0'?><stream:stream xmlns:stream='h > ttp://etherx.jabber.org/streams' xmlns='jabber:client' > to='jabber.anywise.com' version='1.0'>] > > RECV[<?xml version='1.0'?><stream:stream > xmlns:stream='http://etherx.jabber.org/streams' id='44CF2BF7' > xmlns='jabber:client' from='jabber.anywise.com' version='1.0'>] > > RECV[<stream:features><register > xmlns='http://jabber.org/features/iq-register'/><auth > xmlns='http://jabber.org/features/iq-auth'/></stream:features>] > > For reference I am attaching related excerpts from the freeswitch.xml > file > > <load module="mod_dingaling"/> > .... > <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> > <settings> > <param name="debug" value="1"/> > <param name="codec-prefs" value="PCMU"/> > </settings> > <interface> > <param name="name" value="jingle"/> > <param name="login" value="user(a)jabber.anywise.com/talk"/> > <param name="password" value="pwd"/> > <param name="dialplan" value="XML"/> > <param name="message" value="Jingle all the way"/> > <param name="rtp-ip" value="vostroip"/> > <param name="auto-login" value="true"/> > <param name="use-rtp-timer" value="true"/> > <param name="exten" value="7777"/> > <param name="vad" value="both"/> > </interface> > </configuration> > ... > > <extension name="7777"> > <condition field="destination_number" expression="^7777$"> > <action application="bridge" data="playback > /var/lib/asterisk/sounds/test.wav"/> > </condition> > </extension> > > Please suggest what I need to do to achieve the required setup. Any help > will be appreciated. > > Regards, > Danish > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org > > > _______________________________________________ > Freeswitch-users mailing list > Freeswitch-users(a)lists.freeswitch.org > http://lists.freeswitch.org/mailman/listinfo/freeswitch-users > UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users > http://www.freeswitch.org >

2 1

the Compile Failure for FreeSWITCH
by Mizuki Tashiro 10 Aug '06

10 Aug '06

Dear FreeSWITCH guys! I tried to compile the FreeSWITCH, but failed. I compiled as the below URL instruction. http://www.freeswitch.org/gettingstarted.html At the end of compiling, the following message has been displayed. ********************************************************* creating freeswitch creating fsxs cd . && /bin/sh /usr/local/snapshot_freeswitch/build/config/missing --run autoheader configure.in:7: error: Autoconf version 2.58 or higher is required aclocal.m4:6426: AM_INIT_AUTOMAKE is expanded from... configure.in:7: the top level autom4te: /usr/bin/m4 failed with exit status: 1 autoheader: /usr/bin/autom4te failed with exit status: 1 gmake[2]: *** [src/include/switch_am_config.h.in] Error 1 gmake[2]: Exit Directory `/usr/local/snapshot_freeswitch' gmake[1]: *** [install-recursive] Error 1 gmake[1]: Exit Directory `/usr/local/snapshot_freeswitch' make: *** [install] Error 2 ********************************************************* I guess that the very primary error has been occurred on the final step for the creating freeswitch binary. Could you please advise me for fixing it? Thanks in advance and Best Regards, Mizuki Tashiro

3 4

Call recording
by Bruno Impens 09 Aug '06

09 Aug '06

For telephone training, I need to create a fairly simple system that allows a call between two (sip) phones to be recorded and maybe to be transferred to a different sip number. At the customers request, this has has to be done on windows do no asterisk. Can freeswitch already do this ? Before I spend a lot of time trying to get it working only to find out that you cannot record. Suggestions on how to do it are welcome of course. Bruno Impens

2 2

How to interface with google talk
by danish.samad＠vocalseeds.com 03 Aug '06

03 Aug '06

Hi, I am trying to interface with google talk using the following setup Freeswitch<->JabberServer<->GoogleTalkServer<->Googletalk client Freeswitch will register with Jabberserver using a valid jabber account on the Jabber server. Using this account I should be able to login and add any valid gmail id in its userlist and perform standard operations (using the jingle protocol) such as chat, voice, etc. Eventually I intend to use Freeswitch as a gateway between SIP and google talk clients, but I first need to validate the above setup. To test the setup out, I created an account on jabber.anywise.com. I was able to successfully log into the server using PSI and adding a valid gmail accounts in my roster. Now when I specify the account in freeswitch.xml I am not able to register as I donot see the user coming online in my google talk user list. I just see the following XML messages being dumped on the console. freeswitch> SEND[<?xml version='1.0'?><stream:stream xmlns:stream='h ttp://etherx.jabber.org/streams' xmlns='jabber:client' to='jabber.anywise.com' version='1.0'>] RECV[<?xml version='1.0'?><stream:stream xmlns:stream='http://etherx.jabber.org/streams' id='44CF2BF7' xmlns='jabber:client' from='jabber.anywise.com' version='1.0'>] RECV[<stream:features><register xmlns='http://jabber.org/features/iq-register'/><auth xmlns='http://jabber.org/features/iq-auth'/></stream:features>] For reference I am attaching related excerpts from the freeswitch.xml file <load module="mod_dingaling"/> .... <configuration name="dingaling.conf" description="XMPP Jingle Endpoint"> <settings> <param name="debug" value="1"/> <param name="codec-prefs" value="PCMU"/> </settings> <interface> <param name="name" value="jingle"/> <param name="login" value="user(a)jabber.anywise.com/talk"/> <param name="password" value="pwd"/> <param name="dialplan" value="XML"/> <param name="message" value="Jingle all the way"/> <param name="rtp-ip" value="vostroip"/> <param name="auto-login" value="true"/> <param name="use-rtp-timer" value="true"/> <param name="exten" value="7777"/> <param name="vad" value="both"/> </interface> </configuration> ... <extension name="7777"> <condition field="destination_number" expression="^7777$"> <action application="bridge" data="playback /var/lib/asterisk/sounds/test.wav"/> </condition> </extension> Please suggest what I need to do to achieve the required setup. Any help will be appreciated. Regards, Danish

2 1

Conference Latency
by Tom Chez 28 Jul '06

28 Jul '06

I installed FreeSwitch on an Intel Dual Xeon. I had a conference call (total 2 people) using idefisk (iax) softphone and the uLaw codec.The server was on a seperatre network with 130ms ping times from each client. There was excessive delay (> 3sec round trip). Any ideas what might be causing this - or what I can do to troubble shoot this? Anyone else experiencing excessive delays? Tom --------------------------------- Groups are talking. We´re listening. Check out the handy changes to Yahoo! Groups.

2 3

CDR Status Update
by Yossi Neiman 27 Jul '06

27 Jul '06

Hey everybody, Not only has mod_cdr been included in trunk now for a couple days, but I have made some additional enhancements to it. 1. From BaseCDR I have cleaned up and removed some of the earlier used debugging output statements. 2. BaseCDR properly initializes the errorstate upon construction. 3. MysqlCDR now checks the status of the connection to the database, and in the event that the connection is lost, it will try up to 5 times to reconnect per instance of a MysqlCDR object. There is no failover handling yet, so you might want to use at least 1 other logging routine if you have less than reliable connection to your Mysql server. Please remember, you must use MySQL 4.1.x or greater, as this uses prepared statements. Future planned enhancements: 1. Adding in exception handling (try/throw/catch) to detect when a failure to write out a record has occured. 2. Adding in failover handling (queuing to disk) of records so that data is not lost if a record fails to write out the data. (Note: not all loggers will implement this.) 3. Additional logging classes, such as Postgres and possibly ASN1 format. Enjoy! -- Yossi Neiman Cartis Solutions, Inc. - http://www.cartissolutions.com

1 0

xcode modules universal binary
by Brian West 27 Jul '06

27 Jul '06

Hey guys, I have the xcode build working on my intel mac building universal binaries but it seems that the modules do not load on the PPC mac but the binary will execute... It complains about dyld_stub_binding_helper which makes me think I have missed something on my Loadable Bundle template. Any idea? Thanks, Brian

2 5

IAX and log
by Ivan C Myrvold 27 Jul '06

27 Jul '06

The default iax.conf section of freeswitch: <configuration name="iax.conf" description="IAX Configuration"> <settings> <param name="debug" value="0"/>  <param name="port" value="4569"/> <param name="dialplan" value="XML"/> <param name="codec-prefs" value="PCMU@20i,PCMA,speex,L16"/> <param name="codec-master" value="us"/> <param name="codec-rates" value="8"/> </settings> </configuration> I want to test freeswitch with an IAX client. What values do I put in here for a user e.g. "ivan" and password "mypassword"? Where do freeswitch write its output? I couldn't find anything from freeswitch in system.log nor console.log. Ivan

1 0