Hi,
Is this just me who is having this problem? I can't compile the latest
freeswitch source code and here is the error:
checking for gcc option to accept ANSI C... none needed
checking for style of include used by make... GNU
checking dependency style of gcc... gcc3
checking whether gcc and cc
Yeah, on top of it would'nt it be nice if:
when they call the giant 3d unix '*' character with the cell phone
* switch inside the con of the giant 3d unix '*'
* People nervous and shouting about incoming (like in the fight szenes
when they call incoming if missiles are fired)
* One is
Woody Dickson woodydick...@gmail.com wrote:
Is this just me who is having this problem? I can't compile the latest
freeswitch source code and here is the error:
Try starting with a fresh checkout from the repository.
If the problem persists, please report the operating system and version
Hello!
I try to use ;fs_path in originate command, but this seems to not work:
bgapi originate
I also try to use proxy param in gateway, but this doesnt work too. INVITE
dont going to proxy pointed by me.
--
С уважением, Кривушин Михаил
г. Томск сот. +7 913 865 78 66
icq: 218 744 127
xmpp: krivushi...@jabber.ru
skype: mkrivushin
___
Hi,
I've installed FS on Ubuntu 9.04 and I want to run mod_nibbles on it. I
follow the steps to configure my ODBC connection with MySQL as explained in
wiki (mod_nibbles and mod_spidermonkey). But FS, unable to connect it. The
error I got is listed below when I restart FS,
2009-10-06
Hi,
are you configured correctly the nibblebill.conf.xml file?
BR
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Hi I'm using freeswitch1.0.4. This post is moreover similar to my previous
post.
When I make an outgoing call, it is saying INVALID_IE_CONTENTS.
Here are the details.
openzap.conf.xml
configuration name=openzap.conf description=OpenZAP Configuration
settings
param name=debug value=7/
/settings
maybe you can check this: http://www.gsmopen.org/
2009/10/6 Moiz Chinoy moizchi...@gmail.com
Hi,
Is it possible to connect a mobile phone (GSM phone) to Freeswitch and
use this as a GSM gateway?
--
Regards,
Moiz Chinoy.
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FreeSWITCH-users
Hi,
Can any one tell me how to add users dynamically to groups in default.xml,
with out restart the freeswitch.
Thanks
Srinivasula Reddy K
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Hi,
Can any one tell me how to add users dynamically to groups in default.xml,
with out restart the freeswitch.
Thanks
Srinivasula Reddy K
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change the xml and execute reloadxml in FS console or fs_cli
or you can check mod_xml_curl
2009/10/6 srinivasula reddy srinivas.ksvre...@gmail.com
Hi,
Can any one tell me how to add users dynamically to groups in default.xml,
with out restart the freeswitch.
Thanks
Srinivasula Reddy K
thx guys, that helped me alot!
On Tue, Oct 6, 2009 at 12:20 AM, Michael Gende mge...@gendesign.com wrote:
Michael,
Thanks for wiki-fying my text-only attempt at some user doc. I should
have done that for you. I actually have an updated version with many
corrections and the end tabs filled
I'm flattered that you consider my abilities so capable, but time nor
budget are available to afford such extravagance as outlined below.
Besides, bashing something doesn't really gain you any respect (but I
do think that _*_ does ever sooo much deserve bashing).
What I have done is added
gateway calls do not contain any uri data
sofia/gateway/mygw/1000
is all you can do
if you want all that other stuff you need to formulate a direct url
connection
On Tue, Oct 6, 2009 at 4:41 AM, Mikhail Krivushin m.krivus...@imarto.netwrote:
I also try to use proxy param in gateway, but this
This looks like you have an ALG messing with packets... notice it says
rport 5080 but we are sending to 5060.
/b
On Oct 5, 2009, at 11:42 PM, Nicolas Brenner wrote:
Ignore my previous email, the traces were incomplete, got much
better (and complete) traces with ngrep (found a suggestion
Thank u very much seven du.
Regards
Srinvas
On Tue, Oct 6, 2009 at 6:08 PM, Seven Du dujinf...@gmail.com wrote:
change the xml and execute reloadxml in FS console or fs_cli
or you can check mod_xml_curl
2009/10/6 srinivasula reddy srinivas.ksvre...@gmail.com
Hi,
Can any one tell me how
This web request goes to a server running IIS on Windows Server 2003.
From: Brian West [mailto:br...@freeswitch.org]
Sent: Monday, October 05, 2009 5:43 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] mod_xml_curl http POST is
inconsistent/bug
Are you using
Hi,
Is there anyway of using curl without having to setup a standalone http
service? Is it possible to generate curl xml using scripts?
woody
On Mon, Oct 5, 2009 at 11:05 PM, Michael Jerris m...@jerris.com wrote:
On Oct 5, 2009, at 10:49 AM, Woody Dickson wrote:
Is is possible to
My guess is that we configure the curl to support the full range of http
auth methods.
Some of them like Digest require a challenge and realm etc so it's probably
asking without auth header because it cannot create one until it gets that
data. In the case of Basic you can send the login and pass
Ahmed,
I believe you need to specify the database name as it is configured in the
odbc.ini
I am assuming you have something like this in your nibblebill.conf.xml
param name=db_dsn value=freeswitchdb/
Try changing it to this as it is named in the odbc.ini:
param name=db_dsn
As I said in the duplicate thread, the voip codecs issue has been
resolved in trunk, I had a change 1/2 done waiting for testing and it
is now complete.
Mike
On Oct 6, 2009, at 12:30 AM, David Clark wrote:
No I found the one header. I added it to the include list for the
project. It
I am not sure what you mean, do you think that fixes from today should
somehow go somewhere else before we do a release?
On Oct 6, 2009, at 3:21 AM, Vladimir Elizarov wrote:
Brian West пишет:
Because TRUNK is stable... its only fixes going in usually and if
things do break they don't stay
http://pastebin.freeswitch.org/10612
I having been running v14996 OK for a while. I have upgraded a couple of
times after, but every time, an inbound call is hung up on. The only thing
that has changed is the upgrade. This morning I upgraded to v15098 and the
problem persists.
I believe it
Pastebin your openzap.conf file. Also, is this Sangoma or zaptel-based
hardware? If it's Sangoma, pastebin your wanpipe1.conf file. If zaptel,
please paste your zaptel.conf file.
-MC
On Mon, Oct 5, 2009 at 9:06 PM, lakshmanan ganapathy
lakindi...@gmail.comwrote:
Openzap.conf.xml
configuration
On Mon, Oct 5, 2009 at 9:20 PM, Michael Gende mge...@gendesign.com wrote:
Michael,
Thanks for wiki-fying my text-only attempt at some user doc. I should
have done that for you. I actually have an updated version with many
corrections and the end tabs filled in. Can you point me to info on
On Tue, Oct 6, 2009 at 5:31 AM, srinivasula reddy
srinivas.ksvre...@gmail.com wrote:
Hi,
Can any one tell me how to add users dynamically to groups in default.xml,
with out restart the freeswitch.
Thanks
Srinivasula Reddy K
Changes to the dialplan xml files get updated with a simple
On Tue, Oct 6, 2009 at 3:30 AM, lakshmanan ganapathy
lakindi...@gmail.comwrote:
Hi I'm using freeswitch1.0.4. This post is moreover similar to my previous
post.
When I make an outgoing call, it is saying INVALID_IE_CONTENTS.
Here are the details.
openzap.conf.xml
configuration
On Tue, Oct 6, 2009 at 8:05 AM, Woody Dickson woodydick...@gmail.comwrote:
Hi,
Is there anyway of using curl without having to setup a standalone http
service? Is it possible to generate curl xml using scripts?
woody
Check out this page on the wiki:
I've tested this and making the change from ANY to BASIC worked. Thanks
for the help.
It no longer sends the initial post without auth.
From: Anthony Minessale [mailto:anthony.miness...@gmail.com]
Sent: Tuesday, October 06, 2009 11:02 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re:
hello guys,
i was playing with mod_opal to see if i can make it working ... well it
seems SIP-H323 interworking is not tuned at all.
I have a call from a registered sip user (1001) to PSTN via mod_opal
include
extension name=EMERGENCY
condition field=destination_number
What operating system?
-William King
Dome Charoenyost wrote:
Dear All,
why freeswitch use more memory after send and receuve
call. i attach htop capture screen.
you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
(start from 1.2%) after running about 4 hr.
We use memory pools and its not uncommon to use what you displayed.
/b
On Oct 6, 2009, at 12:51 PM, Dome Charoenyost wrote:
Dear All,
why freeswitch use more memory after send and receuve
call. i attach htop capture screen.
you can compare to asterisk it use 0.7% for long time.
On Tue, Oct 6, 2009 at 10:51 AM, Dome Charoenyost d...@tel.co.th wrote:
Dear All,
why freeswitch use more memory after send and receuve
call. i attach htop capture screen.
you can compare to asterisk it use 0.7% for long time. but FS use 7.7%
(start from 1.2%) after running
Debian Squeeze i386 32bit
And Debian Lenny are same
Dome C.
2009/10/7 William King quentus...@gmail.com:
What operating system?
-William King
Dome Charoenyost wrote:
Dear All,
why freeswitch use more memory after send and receuve
call. i attach htop capture screen.
you can
Which modules do you have loaded?
-William King
Dome Charoenyost wrote:
Debian Squeeze i386 32bit
And Debian Lenny are same
Dome C.
2009/10/7 William King quentus...@gmail.com:
What operating system?
-William King
Dome Charoenyost wrote:
Dear All,
why
2009/10/7 Michael Collins m...@freeswitch.org:
On Tue, Oct 6, 2009 at 10:51 AM, Dome Charoenyost d...@tel.co.th wrote:
Dear All,
why freeswitch use more memory after send and receuve
call. i attach htop capture screen.
you can compare to asterisk it use 0.7% for long time. but
2009/10/7 William King quentus...@gmail.com:
Which modules do you have loaded?
default config and nibllebill , lcr , odbcquery
-William King
Dome Charoenyost wrote:
Debian Squeeze i386 32bit
And Debian Lenny are same
Dome C.
2009/10/7 William King quentus...@gmail.com:
What
Hello.
I want to set up faxing via the gateway linksys spa-3102 (with support
for t38) via SIP.
SIP-client - linksys spa3102 - fs - provider
gateway name=callwithus
param name=inbound-late-negotiation value=true/
param name=stun-enabled value=true/
/gateway
It will use enough ram to load all of your modules, and applications.
That is the initial ram usage on startup.
So how much ram is FS using on startup? less than 250-300 MB of ram on
initial load with all the modules and applications isn't unreasonable. I
think by default it loads tons of stuff
pcap is not as useful as FS console log on debug with:
sofia profile internal siptrace on
you should be reporting issues to jira under mod_opal not to the mailing
list.
http://jira.freeswitch.org
FYI
There is little financial support from the community for h323 which prevents
the mod_opal from
hi Anthony,
it is somewhere here:
switch_status_t
FSConnection::receive_message(switch_core_session_message_t *msg)
anyhow, i will open an issue jira of course.
I understand your financial point of view, but anyhow while the entire world
is wants sip and trying to move to sip, the
That happens with both gateways though, one works and the other doesn't.
Would the rport have anything to do with the registration failing?
The big difference to me is that the working gateway replies a 401
Unauthorized containing:
WWW-Authenticate: Digest realm=pxextmy.redvoiss.net,
First off you have to fully understand how SIP authentication works
the two authorization line are different because one is for a
challenge and one is a response to a challenge.
http://en.wikipedia.org/wiki/Digest_access_authentication
On Oct 6, 2009, at 4:22 PM, Nicolas Brenner wrote:
btw My mistake it doesn't assume auth it just calculates the response
hash differently on this case where qop isn't present.
/b
On Oct 6, 2009, at 4:22 PM, Nicolas Brenner wrote:
What does the qop parameter stand for? Apparently because of that
parameter, FS sends a new REGISTER
Tihomir Culjaga tculj...@gmail.com wrote:
I understand your financial point of view, but anyhow while the entire world
is wants sip and trying to move to sip, the reality is quite different. The
majority of voice traffic exchanged via IP is still H323.
Is there any evidence in support of
Diego,
what i'm pointing here is the situation where you have a great product that
lacks in one of most common protocol. It is true H323 is going to disappear
(eventually), it is true that the community prefers SIP/IAX instead ... but
the reality still remains. H323 is going to be used for quite
thanks for your e-mail,
H323 is mainly used for trunking purpose, inter-carrier traffic exchange...
it is not used to control IP phones :P
well, believe me, I've heard enough of H323 that i'm sick of it :P
What i can tell you comes from my own experience on daily activities i'm
doing for
Yeah I understand your point of view, but saying I want a H.323 module or
I want a Ferrari wont magically make it happen.
We need to work on it ourselves or pay to the people that knows how to do it
to do it for us.
There is no other way I think.
Diego
On Tue, Oct 6, 2009 at 11:41 PM,
I didn't mean to start anything. I'm just saying we work very long hours
and barely get anybody asking about h.323.
I wanted to support it and that's why we took up a collection to get funding
for mod_opal but when only 1 donor showed any interest we were forced to
proceed in our spare time which
Thanks Michael,
I found this right after posting you and have already begun
adding/correcting the guide. Thanks again for getting it up there in the
first place.
Mike G.
On Tue, Oct 6, 2009 at 11:15 AM, Michael Collins m...@freeswitch.org wrote:
On Mon, Oct 5, 2009 at 9:20 PM, Michael Gende
Sounds good.
On Tue, Oct 6, 2009 at 7:44 AM, tom tomabr...@gmail.com wrote:
thx guys, that helped me alot!
On Tue, Oct 6, 2009 at 12:20 AM, Michael Gende mge...@gendesign.comwrote:
Michael,
Thanks for wiki-fying my text-only attempt at some user doc. I should
have done that for you. I
Could you open a bug on jira.freeswitch.org as a feature request to
make this a configurable param. (patches that do it even better)
Mike
On Oct 6, 2009, at 12:55 PM, Christian Damianidis wrote:
I’ve tested this and making the change from ANY to BASIC worked.
Thanks for the help.
It no
I got a lot of problem last week for making conference call. I was at home
(conference call starts at 2200hours PKST, my time) and unable to make SIP
call since the government has blocked it. So my only choice was Skype, but
unfortunately DTMF wasn't working, i get connected on Skypiax5 for about
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