<table cellspacing="0" cellpadding="0" border="0" ><tr><td valign="top" style="font: inherit;">Hi,<br><br>The file directory.conf.xml had been mentioned in the documentation many times but there is not such file in the conf folder. Do you mean default.xml in directory folder?<br><br>Thanks!<br><br><br>--- On <b>Tue, 2/24/09, freeswitch-users-request@lists.freeswitch.org <i><freeswitch-users-request@lists.freeswitch.org></i></b> wrote:<br><blockquote style="border-left: 2px solid rgb(16, 16, 255); margin-left: 5px; padding-left: 5px;">From: freeswitch-users-request@lists.freeswitch.org <freeswitch-users-request@lists.freeswitch.org><br>Subject: Freeswitch-users Digest, Vol 32, Issue 181<br>To: freeswitch-users@lists.freeswitch.org<br>Date: Tuesday, February 24, 2009, 3:34 AM<br><br><pre>Send Freeswitch-users mailing list submissions to<br> freeswitch-users@lists.freeswitch.org<br><br>To subscribe or unsubscribe via the World Wide Web,
visit<br> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>or, via email, send a message with subject or body 'help' to<br> freeswitch-users-request@lists.freeswitch.org<br><br>You can reach the person managing the list at<br> freeswitch-users-owner@lists.freeswitch.org<br><br>When replying, please edit your Subject line so it is more specific<br>than "Re: Contents of Freeswitch-users digest..."<br><br><br>Today's Topics:<br><br> 1. Re: SIP dump to DB (kokoska.rokoska)<br> 2. FREESwitch on Windows Server 2003 (Stephen Walker)<br> 3. Re: mod_erlang_event compile problem (Andrew Thompson)<br> 4. Re: FREESwitch on Windows Server 2003 (Carlos Talbot)<br> 5. Re: SIP dump to DB (Joseph Bajin)<br> 6. Re: SIP dump to DB (kokoska.rokoska)<br> 7. Re: mod_portaudio: Do not accept next call after Hangup<br> (Rene Pankratz)<br> 8. Patch for openzap concerning finding a free channel.<br> (Helmut
Kuper)<br><br><br>----------------------------------------------------------------------<br><br>Message: 1<br>Date: Mon, 23 Feb 2009 23:32:26 +0100<br>From: "kokoska.rokoska" <kokoska.rokoska@post.cz><br>Subject: Re: [Freeswitch-users] SIP dump to DB<br>To: freeswitch-users@lists.freeswitch.org<br>Message-ID: <49A323FA.8000802@post.cz><br>Content-Type: text/plain; charset=ISO-8859-1<br><br>Joseph Bajin napsal(a):<br>> Basically, you are trying to build what Empirix has with their Hammer<br>tool.<br>> <br><br>Thank you very much, Joseph, for your interest!<br><br>I have never heard about Empirix (I'll look at it), but what I'm trying<br>to build is something like SER/Kamailio/OpenSIPS sip_trace module.<br><br>> You can create an application that is basically a mix of tshark and a<br>> database feeder. <br>> You sniff with tshark and going to basically pipe it to another<br>> application that will read the pcap file, parse
it, and load it into the<br>> db for you. There are plenty of modules out there that will read pcap<br>> for you. <br>> <br><br>Thank you once more, Joseph, for suggestion!<br>I think about it - it will be challenge for me to write robust and still<br>fast enough (thousands messages per second) SIP parser + DB feeder :-)<br><br>Best regards,<br><br>kokoska.rokoska<br><br><br><br>------------------------------<br><br>Message: 2<br>Date: Mon, 23 Feb 2009 14:47:13 -0800<br>From: "Stephen Walker" <swalker@SONASEARCH.com><br>Subject: [Freeswitch-users] FREESwitch on Windows Server 2003<br>To: <freeswitch-users@lists.freeswitch.org><br>Message-ID:<br> <3B93E0500B57D04CBAE85520B750CFF04CA6CE@exchange.sonasearch.com><br>Content-Type: text/plain; charset="us-ascii"<br><br>Hello:<br><br> <br><br>I have successfully loaded the Windows implementation (SVN 11602 -<br>02/02/09) from your site and it runs fine. I configured a Linksys
SPA<br>2102 and have acquired dial tone and the '999X' tests work. I have not<br>been able to establish connection with either FreeWorldDialup or<br>Broadvoice as of yet.<br><br> <br><br>Which files do I need to edit and what are the proper entries to enable<br>connection to FreeWorldDialup and Broadvoice? Example files and where<br>they reside in the file structure would be very much appreciated. <br><br> <br><br>Thank you <br><br> <br><br> <br><br>All the Best,<br><br>Steve<br><br> <br><br>Steve Walker<br><br>President<br><br>SONASEARCH, INC<br><br>425/883-1984<br><br> <br><br>NOTICE: The information contained in this document is intended by<br>Sonasearch, Inc. or one of its subsidiaries for the use of the named<br>individuals or entities to which it is addressed and may contain<br>information that is privileged or otherwise confidential. It is not<br>intended for transmission to, or receipt by, any individual or entity<br>other than the named
addressee (or a person authorized to deliver it to<br>the named addressee) except as otherwise expressly permitted in this<br>document. If you have received this document in error, please destroy it<br>without copying or forwarding it, and notify the sender of the error by<br>calling Sonasearch at (425) 883-1984.<br><br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br>URL:<br>http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/7d514817/attachment-0001.html<br><br><br>------------------------------<br><br>Message: 3<br>Date: Mon, 23 Feb 2009 19:22:08 -0500<br>From: Andrew Thompson <andrew@hijacked.us><br>Subject: Re: [Freeswitch-users] mod_erlang_event compile problem<br>To: freeswitch-users@lists.freeswitch.org<br>Message-ID: <20090224002207.GF13957@hijacked.us><br>Content-Type: text/plain; charset=us-ascii<br><br>Leon,<br><br>I think I found the problem. I shouldn't have been
defaulting to binding<br>to 127.0.0.1, instead the default should be 0.0.0.0. I've patched the<br>module to actually bind to 0.0.0.0 correctly and made it the default in<br>the config file. Erlang nodes by default bind to 0.0.0.0, so I decided<br>to make mod_erlang_event follow suit.<br><br>Please give that a shot and see if it fixes things.<br><br>Andrew<br><br><br><br>------------------------------<br><br>Message: 4<br>Date: Mon, 23 Feb 2009 20:20:20 -0600<br>From: Carlos Talbot <carlos.talbot@gmail.com><br>Subject: Re: [Freeswitch-users] FREESwitch on Windows Server 2003<br>To: freeswitch-users@lists.freeswitch.org<br>Message-ID:<br> <5800526b0902231820u468908c6ia11191ccf8e37767@mail.gmail.com><br>Content-Type: text/plain; charset="iso-8859-1"<br><br>On Mon, Feb 23, 2009 at 4:47 PM, Stephen Walker<br><swalker@sonasearch.com>wrote:<br><br>><br>> Which files do I need to edit and what are the proper entries to enable<br>>
connection to FreeWorldDialup and Broadvoice? Example files and where<br>they<br>> reside in the file structure would be very much appreciated.<br>><br><br>You'll need to place a gateway configuration for Broadvoice in<br>conf/sip_profiles/external similar to this example:<br>http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Broadvoice<br><br>The same applies to FWD.<br>http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Free_World_Dialup_.28FWD.29<br><br>Once the gateways are configured you'll need to modify the default dial<br>plan<br>to recognize these gateways:<br>http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Dialing_out_via_Gatewayfor<br>dialing out and<br>http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_call_from_a_Gatewayfor<br>incoming.<br><br>Most of this is actually covered
here:<br>http://wiki.freeswitch.org/wiki/Installation_Guide#Windows_quick_start<br><br>regards,<br><br>Carlos<br>-------------- next part --------------<br>An HTML attachment was scrubbed...<br>URL:<br>http://lists.freeswitch.org/pipermail/freeswitch-users/attachments/20090223/9bef760f/attachment-0001.html<br><br><br>------------------------------<br><br>Message: 5<br>Date: Mon, 23 Feb 2009 23:44:04 -0500<br>From: Joseph Bajin <josephbajin@gmail.com><br>Subject: Re: [Freeswitch-users] SIP dump to DB<br>To: freeswitch-users@lists.freeswitch.org<br>Message-ID:<br> <1dce11f20902232044u85259f4hf369da49ce00b46b@mail.gmail.com><br>Content-Type: text/plain; charset=ISO-8859-1<br><br>If you write it correctly it will work just fine. That is how most of<br>all the other correlation engines work. Your setup is not going to be<br>bigger than some of the large telecoms that use these systems today.<br><br><br><br>On 2/23/09, kokoska.rokoska
<kokoska.rokoska@post.cz> wrote:<br>> Joseph Bajin napsal(a):<br>>> Basically, you are trying to build what Empirix has with their Hammer<br>>> tool.<br>>><br>><br>> Thank you very much, Joseph, for your interest!<br>><br>> I have never heard about Empirix (I'll look at it), but what I'm<br>trying<br>> to build is something like SER/Kamailio/OpenSIPS sip_trace module.<br>><br>>> You can create an application that is basically a mix of tshark and a<br>>> database feeder.<br>>> You sniff with tshark and going to basically pipe it to another<br>>> application that will read the pcap file, parse it, and load it into<br>the<br>>> db for you. There are plenty of modules out there that will read pcap<br>>> for you.<br>>><br>><br>> Thank you once more, Joseph, for suggestion!<br>> I think about it - it will be challenge for me to write robust and still<br>> fast
enough (thousands messages per second) SIP parser + DB feeder :-)<br>><br>> Best regards,<br>><br>> kokoska.rokoska<br>><br>> _______________________________________________<br>> Freeswitch-users mailing list<br>> Freeswitch-users@lists.freeswitch.org<br>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> http://www.freeswitch.org<br>><br><br>-- <br>Sent from my mobile device<br><br>--Joe<br><br><br><br>------------------------------<br><br>Message: 6<br>Date: Tue, 24 Feb 2009 07:13:52 +0100<br>From: "kokoska.rokoska" <kokoska.rokoska@post.cz><br>Subject: Re: [Freeswitch-users] SIP dump to DB<br>To: freeswitch-users@lists.freeswitch.org<br>Message-ID: <49A39020.3020808@post.cz><br>Content-Type: text/plain; charset=ISO-8859-1<br><br>Joseph Bajin napsal(a):<br>> If you write it correctly it will work just
fine.<br><br>Yes, this is challenge I have talked about :-)<br><br>> That is how most of<br>> all the other correlation engines work.<br><br>I don't have enough informations but from what I heard from friendly<br>"competitors" they are usualy log (SIP|ISUP) messages after they are<br>parsed by their "routing" servers and not run separate<br>tshark+parser+logger. Or they duplicate (just) SIP messages to separate<br>machine and parse and log them there (SERlike server + sip_trace).<br><br>> Your setup is not going to be<br>> bigger than some of the large telecoms that use these systems today.<br>> <br><br>I hope so :-)<br><br><br>Thanks once more, Joseph, for your info!<br><br>Best regards,<br><br>kokoska.rokoska<br><br><br><br>------------------------------<br><br>Message: 7<br>Date: Tue, 24 Feb 2009 08:27:02 +0100<br>From: Rene Pankratz <r.pankratz@fh-wolfenbuettel.de><br>Subject: Re: [Freeswitch-users] mod_portaudio: Do not accept
next call<br> after Hangup<br>To: freeswitch-users@lists.freeswitch.org<br>Message-ID: <49A3A146.8050001@fh-wolfenbuettel.de><br>Content-Type: text/plain; charset=ISO-8859-1; format=flowed<br><br>No, unfortunately the problem still persists. Portaudio still <br>automatically accepts/takes the next call.<br><br>Ren?<br>> On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz<br>> <r.pankratz@fh-wolfenbuettel.de> wrote:<br>> <br>>> Hello,<br>>> when hanging up a call with portaudio automatically the next call that<br>>> is incoming or held is accepted.<br>>> Is it possible to configure PA that way, that after hanging up<br>(doesn't<br>>> matter whether caller or callee) no call is activated automatically? I<br>>> want to choose if I accept the next call or not.<br>>><br>>> Thanks in advance<br>>> Ren?<br>>><br>>> <br>> Just following up - did this get
resolved?<br>> -MC<br>><br>> _______________________________________________<br>> Freeswitch-users mailing list<br>> Freeswitch-users@lists.freeswitch.org<br>> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>> http://www.freeswitch.org<br>><br>> <br><br><br><br><br><br>------------------------------<br><br>Message: 8<br>Date: Tue, 24 Feb 2009 09:33:42 +0100<br>From: Helmut Kuper <helmut.kuper@ewetel.de><br>Subject: [Freeswitch-users] Patch for openzap concerning finding a<br> free channel.<br>To: freeswitch-users@lists.freeswitch.org<br>Message-ID: <49A3B0E6.80408@ewetel.de><br>Content-Type: text/plain; charset=ISO-8859-1<br><br>Hello,<br><br>today I uploaded a little patch for openzap into trunk (r667). It marks<br>now inbound channels as "inUse" which is conform with outbound<br>channel<br>handling. This should
solve some problems finding a free channel in<br>ozmod_isdn.c for inbound and outbound calls.<br><br>regards<br>Helmut<br><br><br> <br><br><br><br>------------------------------<br><br>_______________________________________________<br>Freeswitch-users mailing list<br>Freeswitch-users@lists.freeswitch.org<br>http://lists.freeswitch.org/mailman/listinfo/freeswitch-users<br>UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users<br>http://www.freeswitch.org<br><br><br>End of Freeswitch-users Digest, Vol 32, Issue 181<br>*************************************************<br></pre></blockquote></td></tr></table><br>