Hi,
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?
Thanks!
--- On Tue, 2/24/09, freeswitch-users-requ...@lists.freeswitch.org
<freeswitch-users-requ...@lists.freeswitch.org> wrote:
From: freeswitch-users-requ...@lists.freeswitch.org
<freeswitch-users-requ...@lists.freeswitch.org>
Subject: Freeswitch-users Digest, Vol 32, Issue 181
To: freeswitch-users@lists.freeswitch.org
Date: Tuesday, February 24, 2009, 3:34 AM
Send Freeswitch-users mailing list submissions to
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When replying, please edit your Subject line so it is more specific
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Today's Topics:
1. Re: SIP dump to DB (kokoska.rokoska)
2. FREESwitch on Windows Server 2003 (Stephen Walker)
3. Re: mod_erlang_event compile problem (Andrew Thompson)
4. Re: FREESwitch on Windows Server 2003 (Carlos Talbot)
5. Re: SIP dump to DB (Joseph Bajin)
6. Re: SIP dump to DB (kokoska.rokoska)
7. Re: mod_portaudio: Do not accept next call after Hangup
(Rene Pankratz)
8. Patch for openzap concerning finding a free channel.
(Helmut Kuper)
----------------------------------------------------------------------
Message: 1
Date: Mon, 23 Feb 2009 23:32:26 +0100
From: "kokoska.rokoska" <kokoska.roko...@post.cz>
Subject: Re: [Freeswitch-users] SIP dump to DB
To: freeswitch-users@lists.freeswitch.org
Message-ID: <49a323fa.8000...@post.cz>
Content-Type: text/plain; charset=ISO-8859-1
Joseph Bajin napsal(a):
> Basically, you are trying to build what Empirix has with their Hammer
tool.
>
Thank you very much, Joseph, for your interest!
I have never heard about Empirix (I'll look at it), but what I'm trying
to build is something like SER/Kamailio/OpenSIPS sip_trace module.
> You can create an application that is basically a mix of tshark and a
> database feeder.
> You sniff with tshark and going to basically pipe it to another
> application that will read the pcap file, parse it, and load it into the
> db for you. There are plenty of modules out there that will read pcap
> for you.
>
Thank you once more, Joseph, for suggestion!
I think about it - it will be challenge for me to write robust and still
fast enough (thousands messages per second) SIP parser + DB feeder :-)
Best regards,
kokoska.rokoska
------------------------------
Message: 2
Date: Mon, 23 Feb 2009 14:47:13 -0800
From: "Stephen Walker" <swal...@sonasearch.com>
Subject: [Freeswitch-users] FREESwitch on Windows Server 2003
To: <freeswitch-users@lists.freeswitch.org>
Message-ID:
<3b93e0500b57d04cbae85520b750cff04ca...@exchange.sonasearch.com>
Content-Type: text/plain; charset="us-ascii"
Hello:
I have successfully loaded the Windows implementation (SVN 11602 -
02/02/09) from your site and it runs fine. I configured a Linksys SPA
2102 and have acquired dial tone and the '999X' tests work. I have not
been able to establish connection with either FreeWorldDialup or
Broadvoice as of yet.
Which files do I need to edit and what are the proper entries to enable
connection to FreeWorldDialup and Broadvoice? Example files and where
they reside in the file structure would be very much appreciated.
Thank you
All the Best,
Steve
Steve Walker
President
SONASEARCH, INC
425/883-1984
NOTICE: The information contained in this document is intended by
Sonasearch, Inc. or one of its subsidiaries for the use of the named
individuals or entities to which it is addressed and may contain
information that is privileged or otherwise confidential. It is not
intended for transmission to, or receipt by, any individual or entity
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without copying or forwarding it, and notify the sender of the error by
calling Sonasearch at (425) 883-1984.
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------------------------------
Message: 3
Date: Mon, 23 Feb 2009 19:22:08 -0500
From: Andrew Thompson <and...@hijacked.us>
Subject: Re: [Freeswitch-users] mod_erlang_event compile problem
To: freeswitch-users@lists.freeswitch.org
Message-ID: <20090224002207.gf13...@hijacked.us>
Content-Type: text/plain; charset=us-ascii
Leon,
I think I found the problem. I shouldn't have been defaulting to binding
to 127.0.0.1, instead the default should be 0.0.0.0. I've patched the
module to actually bind to 0.0.0.0 correctly and made it the default in
the config file. Erlang nodes by default bind to 0.0.0.0, so I decided
to make mod_erlang_event follow suit.
Please give that a shot and see if it fixes things.
Andrew
------------------------------
Message: 4
Date: Mon, 23 Feb 2009 20:20:20 -0600
From: Carlos Talbot <carlos.tal...@gmail.com>
Subject: Re: [Freeswitch-users] FREESwitch on Windows Server 2003
To: freeswitch-users@lists.freeswitch.org
Message-ID:
<5800526b0902231820u468908c6ia11191ccf8e37...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
On Mon, Feb 23, 2009 at 4:47 PM, Stephen Walker
<swal...@sonasearch.com>wrote:
>
> Which files do I need to edit and what are the proper entries to enable
> connection to FreeWorldDialup and Broadvoice? Example files and where
they
> reside in the file structure would be very much appreciated.
>
You'll need to place a gateway configuration for Broadvoice in
conf/sip_profiles/external similar to this example:
http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Broadvoice
The same applies to FWD.
http://wiki.freeswitch.org/wiki/SIP_Provider_Examples#Free_World_Dialup_.28FWD.29
Once the gateways are configured you'll need to modify the default dial
plan
to recognize these gateways:
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Dialing_out_via_Gatewayfor
dialing out and
http://wiki.freeswitch.org/wiki/Getting_Started_Guide#Receiving_an_inbound_call_from_a_Gatewayfor
incoming.
Most of this is actually covered here:
http://wiki.freeswitch.org/wiki/Installation_Guide#Windows_quick_start
regards,
Carlos
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------------------------------
Message: 5
Date: Mon, 23 Feb 2009 23:44:04 -0500
From: Joseph Bajin <josephba...@gmail.com>
Subject: Re: [Freeswitch-users] SIP dump to DB
To: freeswitch-users@lists.freeswitch.org
Message-ID:
<1dce11f20902232044u85259f4hf369da49ce00b...@mail.gmail.com>
Content-Type: text/plain; charset=ISO-8859-1
If you write it correctly it will work just fine. That is how most of
all the other correlation engines work. Your setup is not going to be
bigger than some of the large telecoms that use these systems today.
On 2/23/09, kokoska.rokoska <kokoska.roko...@post.cz> wrote:
> Joseph Bajin napsal(a):
>> Basically, you are trying to build what Empirix has with their Hammer
>> tool.
>>
>
> Thank you very much, Joseph, for your interest!
>
> I have never heard about Empirix (I'll look at it), but what I'm
trying
> to build is something like SER/Kamailio/OpenSIPS sip_trace module.
>
>> You can create an application that is basically a mix of tshark and a
>> database feeder.
>> You sniff with tshark and going to basically pipe it to another
>> application that will read the pcap file, parse it, and load it into
the
>> db for you. There are plenty of modules out there that will read pcap
>> for you.
>>
>
> Thank you once more, Joseph, for suggestion!
> I think about it - it will be challenge for me to write robust and still
> fast enough (thousands messages per second) SIP parser + DB feeder :-)
>
> Best regards,
>
> kokoska.rokoska
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
--
Sent from my mobile device
--Joe
------------------------------
Message: 6
Date: Tue, 24 Feb 2009 07:13:52 +0100
From: "kokoska.rokoska" <kokoska.roko...@post.cz>
Subject: Re: [Freeswitch-users] SIP dump to DB
To: freeswitch-users@lists.freeswitch.org
Message-ID: <49a39020.3020...@post.cz>
Content-Type: text/plain; charset=ISO-8859-1
Joseph Bajin napsal(a):
> If you write it correctly it will work just fine.
Yes, this is challenge I have talked about :-)
> That is how most of
> all the other correlation engines work.
I don't have enough informations but from what I heard from friendly
"competitors" they are usualy log (SIP|ISUP) messages after they are
parsed by their "routing" servers and not run separate
tshark+parser+logger. Or they duplicate (just) SIP messages to separate
machine and parse and log them there (SERlike server + sip_trace).
> Your setup is not going to be
> bigger than some of the large telecoms that use these systems today.
>
I hope so :-)
Thanks once more, Joseph, for your info!
Best regards,
kokoska.rokoska
------------------------------
Message: 7
Date: Tue, 24 Feb 2009 08:27:02 +0100
From: Rene Pankratz <r.pankr...@fh-wolfenbuettel.de>
Subject: Re: [Freeswitch-users] mod_portaudio: Do not accept next call
after Hangup
To: freeswitch-users@lists.freeswitch.org
Message-ID: <49a3a146.8050...@fh-wolfenbuettel.de>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
No, unfortunately the problem still persists. Portaudio still
automatically accepts/takes the next call.
Ren?
> On Thu, Feb 19, 2009 at 5:04 AM, Rene Pankratz
> <r.pankr...@fh-wolfenbuettel.de> wrote:
>
>> Hello,
>> when hanging up a call with portaudio automatically the next call that
>> is incoming or held is accepted.
>> Is it possible to configure PA that way, that after hanging up
(doesn't
>> matter whether caller or callee) no call is activated automatically? I
>> want to choose if I accept the next call or not.
>>
>> Thanks in advance
>> Ren?
>>
>>
> Just following up - did this get resolved?
> -MC
>
> _______________________________________________
> Freeswitch-users mailing list
> Freeswitch-users@lists.freeswitch.org
> http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
------------------------------
Message: 8
Date: Tue, 24 Feb 2009 09:33:42 +0100
From: Helmut Kuper <helmut.ku...@ewetel.de>
Subject: [Freeswitch-users] Patch for openzap concerning finding a
free channel.
To: freeswitch-users@lists.freeswitch.org
Message-ID: <49a3b0e6.80...@ewetel.de>
Content-Type: text/plain; charset=ISO-8859-1
Hello,
today I uploaded a little patch for openzap into trunk (r667). It marks
now inbound channels as "inUse" which is conform with outbound
channel
handling. This should solve some problems finding a free channel in
ozmod_isdn.c for inbound and outbound calls.
regards
Helmut
------------------------------
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