Another idea would be to write simple rsync method, and post a page
on the same on the Wiki so all those people who have their own server
and willing to spare some bandwidth can mirror the entire Wiki
locally.
I am willing to provide a mirror for the existing wiki on a rsync
update daily for the
The rsync idea sounds better.
Regards,
Diego
On Fri, Apr 17, 2009 at 4:03 AM, Mitul Limbani mi...@enterux.com wrote:
Another idea would be to write simple rsync method, and post a page on the
same on the Wiki so all those people who have their own server and willing
to spare some bandwidth
On Fri, Apr 17, 2009 at 12:30 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
are you planning on just signaling on TCP or both audio and signalling
cos realtime audio over TCP kinda stinks.
you may find that just running FS as the farm and calling to it with sip is
more or less the
On Fri, Apr 17, 2009 at 12:58 AM, UV u...@yuvalhertzog.com wrote:
Ok, I think I know where's the confusion here. Let me clarify:
1. FS run beautifully as a service - that's why I assumed it should work.
2. Skype client runs as a service very well too.
3. When running FS as a service with
Special:Export will export a page to an XML format that can, in turn, be
imported.
There must be a way to automate that process (export extire wiki to XML, rsync
and import and mirrors).
--
Regards,
Will Boyce m...@willboyce.com
tel: 07933 515 987
url: http://willboyce.com
- Jason
mercutioviz wrote:
definitely stop by IRC and talk real-time with others who've dealt with
this
kind of thing.
Thanks for the tip, but since this firewalls are a frequent issue with SIP,
I think it'd be very useful if someone who has good experience with this
wrote an article in the FS
hi,
i was trying to do the stuff i had written above but seems a bit too tough
so on giving it a second though, the reason i need to do the above was each
client id is basically billed on my voipswitch so a way around it could be
was the client id and pass on voipswitch for all clients, i can
It has not been used much but there was a way to add gateways to a user tag
in the directory
so if you register locally to FS to that user then it will register that
gateway and unreg it when he unregisters.
I don't have time to dig it up but the params are in there somewhere.
On Fri, Apr 17,
Hi Dave,
didn't fully get your explanation. Maybe I should explain how I do it right now.
I am using the http://www.codeplex.com/eventsocket library. I take my x-lite
phone with number 1000 and call into FS with number 301234. Under this number I
launch my app as async full.
extension
Are you calling a registered user? If so then please replace the @
with a % or user user/1...@192.168.1.102
/b
PS I do get called brain a lot!
On Apr 16, 2009, at 11:45 PM, Prabhuram Mohan wrote:
Thanks Brain/ Dave,
I ran the modified command as follows
originate
does it only work for registrations only or does it limit the user to calling
through that gateway only?
Anthony Minessale-2 wrote:
It has not been used much but there was a way to add gateways to a user
tag
in the directory
so if you register locally to FS to that user then it will
You're suggesting running two instances of FS on same machine, have them
communicate first over TCP then UDP via SIP over socket (Skype -- Skypiax
-- FS1 -- FS2) on a production system? Hmmm... nah... :-)
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
Give a shout if you get Skypiax working as a service.
I'll be happy to contribute to its wiki about it once you get it working.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Giovanni
Maruzzelli
On Fri, Apr 17, 2009 at 4:02 PM, UV u...@yuvalhertzog.com wrote:
Give a shout if you get Skypiax working as a service.
I'll be happy to contribute to its wiki about it once you get it working.
Yes, definitely!
gm
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If someone has a way to make true mirrors that support read/write this
would be interesting.
Mike
On Apr 17, 2009, at 4:06 AM, Will Boyce wrote:
Special:Export will export a page to an XML format that can, in
turn, be imported.
There must be a way to automate that process (export extire
Hi Brian,
The follower command worked per your suggestion
originate user/1...@192.168.1.102 conference(3000-192.168.1.102)
Thanks a lot,
Prabhu
On Fri, Apr 17, 2009 at 6:43 AM, Brian West br...@freeswitch.org wrote:
Are you calling a registered user? If so then please replace the @ with
If someone has a way to make true mirrors that support read/write
this
would be interesting.
Do it robustly, transparently and in real time and that's the problem of
distributed source code revision control mostly sorted as well.
Although I'm not sure I'd really want to use the kernel anyone
Hi,
I am using xml_curl, and what I'm trying to do is dynamically register
the users to the platform, instead of bulking all the users on the FS
boot time. So far I have successfully register the users and also make
calls through gateways or to static users (those users that are in the
On Apr 17, 2009, at 12:07 PM, JuanMa wrote:
Hi,
I am using xml_curl, and what I'm trying to do is dynamically register
the users to the platform, instead of bulking all the users on the FS
boot time. So far I have successfully register the users and also make
calls through gateways or to
Hello,
please anyone here a implemented CURL for freeswitch using any Language
and any
RDBNMS ?
i have a freeswitch contributed one implementation using PHP / MySQL but
is not working.
any help is werlcome
thanks!
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Meftah Tayeb wrote:
Hello,
please anyone here a implemented CURL for freeswitch using any Language
and any
RDBNMS ?
i have a freeswitch contributed one implementation using PHP / MySQL but
is not working.
not working is relatively loose. Do you have any more information than
that?
2009/4/15 Ron McCarthy ronmc...@gmail.com
Hi,
We have several context setup with different users, all in the directory
with IP based auth.
Anyway we can turn on debug for just one context/user in the directory,
when lots of people are making calls it's hard to see traffic to just for
one
below is the lua script that is running when the caller dials in. As mentioned before, swaping the streamFile and sleep make the problem go away. I am experiencing different issues all related to DTMF tones, again, I don't think FS is the problem. But I'm wondering how to find the
This isn't anything we can do anything about... its the gsm to PSTN
conversion of the digits... it'll take some time.. I have noticed this
on ATT also while dialing digits on a remove IVR such as BofA.
/b
On Apr 17, 2009, at 1:47 PM, p...@privateconnect.com wrote:
I believe the isuse
Quick question: do you do anything interesting in the dialplan prior to
calling the script? For posterity's sake could you paste the extension info
here as well?
Thanks,
MC
On Fri, Apr 17, 2009 at 11:47 AM, p...@privateconnect.com wrote:
below is the lua script that is running when the caller
Quick question ya'll:
Is there a way to detect DTMF during bridged conversation? Or do I have to
use a conference for that?
Scenario: Caller A dials in and is bridged with caller B (in my case via lua
- session:execute(bridge, some_route)) If at any time during the
conversation caller B
On Fri, Apr 17, 2009 at 4:02 PM, UV u...@yuvalhertzog.com wrote:
Give a shout if you get Skypiax working as a service.
I'll be happy to contribute to its wiki about it once you get it working.
shoutgot Skypiax working as a service/shout
I will document this better in the future, but following
On Fri, Apr 17, 2009 at 11:58 PM, Giovanni Maruzzelli
gmar...@celliax.org wrote:
Then you install FS as service (freeswitch.exe -install servicename),
start FS as a service (under local system), manually (again, from
the services applet).
make sure the FS service is owned by local system and
Thanks for insight in this, Ill look into doing that.
In the directory though, we have other PBXs connected to us (basically we
provide some dialtone for a few people), and we stuck them in the directory
as it was the only way we dould have each user have their own context, but
we can't do a nice
On 17 Apr 2009, at 18:42, Meftah Tayeb wrote:
Hello,
please anyone here a implemented CURL for freeswitch using any
Language
and any
RDBNMS ?
i have a freeswitch contributed one implementation using PHP / MySQL
but
is not working.
any help is werlcome
thanks!
Yeah, we have done one
Pete Mueller p...@privateconnect.com wrote:
Is there a way to detect DTMF during bridged conversation?
You can use bind_meta_app in your dial plan; see the wiki for details.
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