Hi,
If there is no build for the tls with freeswitch on windows, can you
please guide how to test gtalk integration on windows.
I have successfully compiled the tls with iksemel, there were couple
of errors but I managed to compile it.
Now, I am also getting the error libdingaling.c:1545 xmpp_co
Thank you Ken. Do you know if mod_opal supports both H323 voice and video?
2009/3/30 Ken Rice
> Yes its possible... Look at mod_opal
>
>
>
> --
> *From: *Christian Bourke
> *Reply-To: *
> *Date: *Mon, 30 Mar 2009 10:55:14 +1000
> *To: *
> *Subject: *[Freeswitch-users
Thanks for your quick&clear answers.
On Fri, 2009-03-27 at 10:55 -0500, Anthony Minessale wrote:
> If you are using on-hook agents, it will place as many outbound calls
> as there are people waiting.
> If you are using off-hook agents it will just connect the first free
> agent.
By "people wait
How is the performance for mod_opal ? is the performance same as SIP ?
2009/3/30 Christian Bourke
> Thank you Ken. Do you know if mod_opal supports both H323 voice and video?
>
> 2009/3/30 Ken Rice
>
>> Yes its possible... Look at mod_opal
>>
>>
>>
>> --
>> *From: *
Hi,
I would like to implement freeswitch as an H323 gatekeeper. I would like to
know if its possible for incoming H323 calls to go into a hunt group, if no
'operators' are logged into the hunt group then the call will bounce to an
auto attendant and the H323 caller will be played a short video mes
so what i make of it is that FS cant do upper registration or lets say it
wont proxy everything and wont use the registration details it receives from
clients registering to it and use it to register clients to a different
server for which its proxying, or mayb its not meant to do so but then i as
Fadil Berisha wrote:
>
> Sure you can get reliable DTMF detection on 70%-80% of call paths with
> no echo cancellation,
>
>
> Fair enough. You say "sure you can get reliable DTMF detection on
> 70%-80% of call paths with no echo cancellation". OK, you forgot to
> mention that this is ach
FS does not proxy registrations, really it doesn't proxy anything very much
besides calls
in a limited capacity, you would want to use a SIP proxy for that.
On Sun, Mar 29, 2009 at 1:07 AM, xbipin wrote:
>
> i have been trying to do this but seem to have been lost, actually i want
> to
> use fr
I¹m not sure if the video support is complete but both freeswitch and opal
support video...
Now I also see another email you send asking about freeswitch being a
gatekeeper... FreeSWITCH is just another h323 end point, it does not do all
gatekeeper functions like keep up with what IP what h323 use
You should press f8 to get more detailed output from FS.
you also should capture more of the call,
starting at line 192 you seem to be sending yourself a notify, not sure how
you did that.
you are not by any chance trying to call a registered endpoint using the FS
ip together with @ are you?
say y
libdingaling.c:1545 xmpp_connect() io error 2 7
It means there was an I/O error trying to connect
it's probably because iksemel does not have srv support, if you are trying
to connect to gtalk
you have to manually specify the server as talk.google.com, the default
client.xml is self-explanatory f
the h323 support in FS is brand new. It's lacking video support at all.
You would have to provide funding to round up some developers to implement
what you need.
2009/3/30 Christian Bourke
> Hi,
>
> I would like to implement freeswitch as an H323 gatekeeper. I would like to
> know if its possi
On Mon, Mar 30, 2009 at 3:34 AM, François Delawarde <
fdelawa...@wirelessmundi.com> wrote:
> Thanks for your quick&clear answers.
>
>
> On Fri, 2009-03-27 at 10:55 -0500, Anthony Minessale wrote:
>
> > If you are using on-hook agents, it will place as many outbound calls
> > as there are people wa
is this deja-vu, this is the 2nd thread with the same subject, i seem to be
seeing multiple of every email you send
you may want to check your mail client.
like ken said, and me too in the other thread with the same exact email in
it, use a proxy
http://www.kamailio.org/
On Mon, Mar 30, 2009 at
perhaps we should stick to one thread per topic within the same 1 week span.
2009/3/30 Ken Rice
> I’m not sure if the video support is complete but both freeswitch and
> opal support video...
> Now I also see another email you send asking about freeswitch being a
> gatekeeper... FreeSWITCH is
I'm really starting to feel like we're playing musical threads here.
On Sun, Mar 29, 2009 at 12:50 AM, Bipin Patel wrote:
> hi,
>
> what are the other steps if done on windows platform with FS as SBC and
> Voipswitch as the server.
> secondly how to make the clients registering to voipswitch go
Anthony Minessale wrote:
> I'm really starting to feel like we're playing musical threads here.
Just avoid playing them through low bit rate codecs. :-)
Steve
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswit
Steve Underwood wrote:
Anthony Minessale wrote:
I'm really starting to feel like we're playing musical threads here.
Just avoid playing them through low bit rate codecs. :-)
I think we need an echo canceller ;-)
--Dave
___
Freeswitch-user
GHH
/me gouges eyes out with EC card
2009/3/30 David Knell :
> Steve Underwood wrote:
>
> Anthony Minessale wrote:
>
>
> I'm really starting to feel like we're playing musical threads here.
>
>
> Just avoid playing them through low bit rate codecs. :-)
>
>
> I think we need an echo cancelle
can any1 tell me where can i find a live cd image with the basic stuff to run
FS and FS with all it tools installed and WITH A GUI, something like a pbx
in a flash iso image so windows users like me find it easier to get testing
with FS as the support for windows SIP proxy or any SIP related tool
On Mar 30, 2009, at 9:33 AM, xbipin wrote:
can any1 tell me where can i find a live cd image with the basic
stuff to run
FS and FS with all it tools installed and WITH A GUI, something like
a pbx
in a flash iso image so windows users like me find it easier to get
testing
with FS as the supp
There is none yet available.
If you have patience, I suspect that one will be out in the next weeks, tough.
Watch the website and the mailing list for announcement.
Sincerely,
Giovanni Maruzzelli
=
www.celliax.org
via Pierlombardo 9, 20135 Milano
Italy
g
On Mar 30, 2009, at 10:33 AM, xbipin wrote:
>
> can any1 tell me where can i find a live cd image with the basic
> stuff to run
> FS and FS with all it tools installed and WITH A GUI, something like
> a pbx
> in a flash iso image so windows users like me find it easier to get
> testing
> wi
On Mar 30, 2009, at 1:34 AM, François Delawarde wrote:
> Thanks for your quick&clear answers.
>
>
> On Fri, 2009-03-27 at 10:55 -0500, Anthony Minessale wrote:
>
>> If you are using on-hook agents, it will place as many outbound calls
>> as there are people waiting.
>> If you are using off-hook a
hi,
im really sorry if it sounded as if i was demanding, its a place where
every1 helps so i did the same thing, asked if any1 knew of any such live cd
image coz its been quiet some time and i havent reached anywhere with the
testing of FS coz the reason being, i can test it offline at my home bu
Just following up... did you get these questions ironed out?
-MC
On Wed, Mar 25, 2009 at 2:51 AM, Gilles wrote:
> Hello,
>
> I have a couple of questions related to having SIP users connecting
> from the Net to a Freeswitch server through NAT routers on both ends:
>
> 1. How must I configure rout
Here is a really good start --> http://wiki.freeswitch.org/wiki/SBC_Setup
Overall I think you're going to just have to install Linux on a box and get
after it. It may be more painful in the short term ... but in the long term
your life will be much better.
The devs / community here are pretty in
> living in this region and being able to test and try open source softwares
> is not easy coz never 2 guys meet and end up talking something about linux,
> pbx, softswitches etc coz no1 has ever heard of such things other than the
> stuff whats running in the economy.
>
Out of curiosity, what par
sers
> UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-users
> http://www.freeswitch.org
>
>
> __ NOD32 3975 (20090330) Information __
>
> This message was checked by NOD32 antivirus system.
> http://www.eset.com
>
>
>
_
That is one interesting city I wouldn't mind paying a visit but
its a bit rich for my blood!
;)
On Mar 30, 2009, at 12:26 PM, Bipin Patel wrote:
hi,
i currently live in a country called UAE - united arab emirates and a
city called Dubai.
Regards,
Bipin
Brian West
br...@freeswitch.o
On Mon, Mar 30, 2009 at 10:26 AM, Bipin Patel wrote:
> hi,
>
> i currently live in a country called UAE - united arab emirates and a
> city called Dubai.
>
Hehe, Dubai is quite a popular place - even a number of us ignorant
Americans have heard of it! :) We would love to see FreeSWITCH become mo
The AstLinux ISO with FreeSWITCH is a live cd.
http://mirror.astlinux.org/freeswitch/
This one is a little old but I could easily compile a new one...
2009/3/30 Brian West :
>
> On Mar 30, 2009, at 9:33 AM, xbipin wrote:
>
> can any1 tell me where can i find a live cd image with the basic stuff
Hi all,
What I would like to stress is:
1) FreeSwitch is working on Windows, natively, without hacks
2) This is a huge advantage for a free software that want to be
*really* popular (eg: be capable of running on an already working
office machine, without dedicated hardware/expertise)
3) This is
Yes Kristian, please!
On Mon, Mar 30, 2009 at 7:47 PM, Kristian Kielhofner
wrote:
> The AstLinux ISO with FreeSWITCH is a live cd.
>
> http://mirror.astlinux.org/freeswitch/
>
> This one is a little old but I could easily compile a new one...
>
> 2009/3/30 Brian West :
>>
>> On Mar 30, 2009, at
Except for fields in a dial plan extension, I can't get tone_detect (or
stop_tone_detect),
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_tone_detect
http://wiki.freeswitch.org/wiki/Misc._Dialplan_Tools_stop_tone_detect
to work in JavaScript.
If there is one, what's the correct syntax f
s 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
>>
>>
>> __ NOD32 3975 (20090330)
Really hard to inspect packets when they run on port 443 and are
encrypted :P
/b
On Mar 30, 2009, at 1:31 PM, Gabriel Kuri wrote:
eh, how are poeple doing VoIP over there, given use of it outside the
UAE is "officially" outlawed by the TRA ? I've heard Etisalat is
pretty
strict with making
Hi Guys,
I've been following the "idea" from
http://wiki.freeswitch.org/wiki/Proxy_Media to match possible Nated
endpoints through dialplan and it works just fine for leg A but not for
leg B.
For example:
1. leg A private (172.16.0.2) -> NAT/ROUTER public (1.0.0.1) ->
FS public (2.0.0.1) -> l
Are you saying they should configure SIP TLS to run on port 443? :)
2009/3/30 Brian West :
> Really hard to inspect packets when they run on port 443 and are encrypted
> :P
> /b
--
Kristian Kielhofner
http://blog.krisk.org
http://www.submityoursip.com
http://www.astlinux.org
http://www.star2star
yeah, obviously they'd have to enable TLS, but my question is more in
terms of trying to use the mainstream providers outside the country,
particularly for call termination/origination, which don't support TLS.
~Gabe
Brian West wrote:
> Really hard to inspect packets when they run on port 443 and
no one would do that!
On Mon, Mar 30, 2009 at 8:41 PM, Kristian Kielhofner
wrote:
> Are you saying they should configure SIP TLS to run on port 443? :)
>
> 2009/3/30 Brian West :
>> Really hard to inspect packets when they run on port 443 and are encrypted
>> :P
>> /b
>
> --
> Kristian Kielhofne
Even better run OpenVPN on port 53 :P
/b
On Mar 30, 2009, at 1:41 PM, Kristian Kielhofner wrote:
Are you saying they should configure SIP TLS to run on port 443? :)
2009/3/30 Brian West :
Really hard to inspect packets when they run on port 443 and are
encrypted
:P
/b
Brian West
br...@fr
You sure could to get around some oppression! :P
/b
On Mar 30, 2009, at 1:47 PM, Giovanni Maruzzelli wrote:
> no one would do that!
>
>
> On Mon, Mar 30, 2009 at 8:41 PM, Kristian Kielhofner
> wrote:
>> Are you saying they should configure SIP TLS to run on port 443? :)
>>
>> 2009/3/30 Brian We
;-)
On Mon, Mar 30, 2009 at 8:49 PM, Brian West wrote:
> You sure could to get around some oppression! :P
>
> /b
>
> On Mar 30, 2009, at 1:47 PM, Giovanni Maruzzelli wrote:
>
>> no one would do that!
>>
>>
>> On Mon, Mar 30, 2009 at 8:41 PM, Kristian Kielhofner
>> wrote:
>>> Are you saying they
Hallo,
thank you for your answer Anthony.
>
> starting at line 192 you seem to be sending yourself a notify, not sure
> how you did that.
That is indeed strange, I have looked at the MjSip code but haven't found the
cause yet.
> you are not by any chance trying to call a registered endpoint u
maybe that phone does not support early media
try adding the answer application to your dialplan
On Mon, Mar 30, 2009 at 3:33 PM, wrote:
> Hallo,
>
> thank you for your answer Anthony.
>
> >
> > starting at line 192 you seem to be sending yourself a notify, not sure
> > how you did that.
>
> T
Hi,
I have an application where my Javascript hanguphook code calculates a
value (e.g. the cost of the call which can only be calculated post
hangup) and I need to have that value appear as a field in the cdrs.
As the 'session' object is no longer available for javascript logic post
hangup,
Are logging both a- and b-legs? Just curious what your setup is.
-MC
2009/3/30 Keith Laaks
> Hi,
>
>
>
> I have an application where my Javascript hanguphook code calculates a
> value (e.g. the cost of the call which can only be calculated post hangup)
> and I need to have that value appear as
in your script called via api_hangup_hook:
var env = request.dumpENV("text");
consoleLog("info", env);
all those vars are there for you, you can get the individually with
var hval = request.getHeader("some_header");
2009/3/30 Keith Laaks
> Hi,
>
>
>
> I have an application where my Javascri
I get similar behavior as Peter when trying to enter a locked conference.
If I am just dialing from a phone to a conference (on a dialplan), it will
properly lock me out. But if I do an originate command (originate
sofia/internal/1001 &conference(3000)), it will drop me into the conference,
even th
Update again to svn trunk... btw 1.0.4 pre3 is out on
files.freeswitch.org
/b
On Mar 30, 2009, at 6:44 PM, Dan Le wrote:
I get similar behavior as Peter when trying to enter a locked
conference.
If I am just dialing from a phone to a conference (on a dialplan),
it will properly lock me
hi,
i think this thread has got a bit too much of attention but in all
directions, bytheway sorry for that double posting but i think its something
to do with nabble as i get a mail saying ur post is rejected so i resend and
then i find both the posts get accepted.
UAE is very strict in terms of
This I/O error occurred once. And I am able to communicate google talk
to SIP softphone through FS.
Still does not know why this error happened as I did not change anything!
2009/3/30 Anthony Minessale :
> libdingaling.c:1545 xmpp_connect() io error 2 7
>
> It means there was an I/O error trying
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