Hi,
Have you tried the acl function in freeswitch. For example, setup the DIDWW
domain as an acl so no need to use username/password from that domain.
In the default profile, you can specify the default context for that
incoming call to go to in the dialplan.
If my solution works for you,
http://wiki.freeswitch.org/wiki/Wiki_meet_2008_05_28
Please add anything to this list you think we need to talk about. As
usual we'll breeze thru jira, docs and various other things that come
up along the way. Please remember to DIGG the 1.0 release article at
HI
We want to try generate 5000 simultanious Voice broadcast calls .
can the below config will work?
SS7 Links Sangoma SMG---FreeSwitchBroadcasting
Application ( SIP based)
Thank you
Imthiyaz
mail2web.com
First of all- Amazing project. Tired of asterisk deadlocking all the
time we have been deploying asterisk with OpenSER as the registrar.
Freeswitch is a huge relief!
This is an extremely important feature we have been looking for.
Asterisk realtime ldap integration is very flaky. I found this
need billing for freeswitch [EMAIL PROTECTED]
On 5/27/08, [EMAIL PROTECTED] [EMAIL PROTECTED] wrote:
HI
We want to try generate 5000 simultanious Voice broadcast calls .
can the below config will work?
SS7 Links Sangoma SMG---FreeSwitchBroadcasting
Application ( SIP
Here is some info on it
http://wiki.freeswitch.org/wiki/Mod_spidermonkey_odbc
-E
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Currently the directory interface is only used for that dialplan, I
would like to enhance that in the future. The directory dialploan
uses a filter of exten=destination number, and then has name/value
pairs, I will see if I can find the schema we used back when we
developed it, short of
There is a prototype developed in OpenZAP to allow FreeSWITCH to act in
place of sangoma SMG in your model.
SS7 Links FreeSwitch(OpenZAP) Broadcasting Application (
SIP based)
OpenZAP has a new protocol plugin that can communicate with the other end of
Sangoma SMG using the same
Yes,
We have experimental support for 2 ways to do ss7, telco bridges (not in
tree but privately under development)
and also Sangoma SS7Boost using ss7box (in tree in the OpenZAP lib dist with
FreeSWITCH). Both are new and need testing.
We also have plans to add native isup support to OpenZAP
Awesome. Now if we could just get astgui/vicidial guys to port their
stuff over... :D
-MC
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ward
Mundy
Sent: Wednesday, May 28, 2008 5:33 AM
To: freeswitch-users@lists.freeswitch.org
Subject:
Oh hell no
From: Michael Collins [EMAIL PROTECTED]
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Wed, 28 May 2008 09:52:21 -0700
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] RESTful Bounty . Putting the
competitionto REST with RestFul VoIP2.x services
Heh. Maybe you could send me what you're using so I don't have to
reinvent the wheel all by myself! :P
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ken
Rice
Sent: Wednesday, May 28, 2008 9:50 AM
To: freeswitch-users@lists.freeswitch.org
if can run vicidial under FS that would be owesome !!!
On Wed, May 28, 2008 at 1:52 PM, Michael Collins [EMAIL PROTECTED] wrote:
Awesome. Now if we could just get astgui/vicidial guys to port their stuff
over… :D
-MC
From: [EMAIL PROTECTED]
We¹re actually in the process of building a new dialer based on
freeswitch... I don¹t see how people can actually deal with vicidial... I¹ve
used it and Iw as horrified by the meetme requirements and all kinds of
other fun stuff that just tremondously added to the the load... Not to
mention lack
i'work on a call center with 50 seats, i'can test any dialer based on
FS if you want.
On Wed, May 28, 2008 at 2:22 PM, Ken Rice [EMAIL PROTECTED] wrote:
We're actually in the process of building a new dialer based on
freeswitch... I don't see how people can actually deal with vicidial... I've
Brian West wrote:
What else is great is his phone does G722 wideband. w00t!
btw, what kind of G.722 wideband codec do you support?
I see that you are using G.722 implementation from the voipcodecs
library, and that you are using different samplerates, one signalled
in SDP (8000 Hz) and
At one point I was very interested in this...then I got a job. =[
I thought mod_ldap was more of a PoC than anything. It might work (I
couldn't get it working and unfortunately don't remember exactly why..)
but there really isn't much point. I would have to do at least 5 ldap
queries (if not
I've been looking around a little bit and I haven't seen anything on this yet
so I wanted to ask here.
I want to have a dialplan that captures certain SIP responses and takes
different actions based on the response received.
For example, if I place a call out to a peer and get back 486 Busy
Thanks a lot. I intend to use it mostly as a SIP user directory. For the
dial-plan I dont mind parsing and syncing XML file across servers (if
there were a small cluster). The main deal is AUTHENTICATION. The
authentication scheme I wish to keep is Kerberos (with SASL in Ldap for
binding). This
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