Hello,
I'm a little confused with FreeSWITCH behavior.
At http://wiki.freeswitch.org/wiki/XML_User_Directory_Guide
we can read:
The directory section is used to add accounts for all users that should be
able to register in the pbx by using User Agents (SIP Phones).
So I suppose I should provide
On Wed, Aug 6, 2008 at 1:08 PM, Lito Manansala
[EMAIL PROTECTED]wrote:
hi,
sorry it's http://www.wikipbx.org/
thank you
I have installed CD version Trixswitch
on top of it can i install this GUI
if yes any one success on this ?
ram
___
Has anybody successfulkly tested a Softphone that works with TLS/SRTP
and Freeswitch?
- I tried Minisip (I think it works with MIKEY) - no success
- I tried Zoiper Biz - it did not like to connect via TLS (any hints?)
- more ??
Best regards
Peter
___
Peter P GMX wrote:
Has anybody successfulkly tested a Softphone that works with TLS/SRTP
and Freeswitch?
Not me but iirc bkw has successfully tested TLS/SRTP on Snom and Polycom.
- I tried Minisip (I think it works with MIKEY) - no success
Afaik FreeSWITCH does not support MIKEY.
- I
You all did well but you didn't pick the simplest method:
Add this to a user:
variable name=toll_allow value=local,ld,int/
Then in your dialplan for the extension you wish to protect from
various routes:
condition field=${toll_allow} expression=local/
/b
On Aug 6, 2008, at 12:33 AM,
Thank you. This worked. I added a second context to the dialplan allowing
PSTN and assigned to user 1010.
/b: I like the simplest method, just don't understand it yet. Where is
toll_allow defined?
(conf/dialplan/default.xml):
include
context name=default
/* extensions */
This will require you to duplicate most of your dialplan twice :P
/b
On Aug 6, 2008, at 8:28 AM, Roberto Hernandez wrote:
Thank you. This worked. I added a second context to the dialplan
allowing PSTN and assigned to user 1010.
___
Hi folks,
I am looking to download freeswitch.jar, but couldn't find it till now. May
be its hidden somewhere, or my mind is quite exhausted right now. Please
help.
Thanks.
--
Best,
Adeel Ansari
http://www.linkedin.com/in/adeelansari
___
Hi,
I'm tearing my hair out trying to get an inbound FXO call to bridge to a SIP
extension.
My setup is:
Zapmicro ZMA400P (TDM400P equivalent).
Cisco 7940 with SIP firmware
FS 1.0.1 + zaptel 1.4.11
I'm able to make calls out from various SIP extensions via the ZMA400P FXO,
however I can't get
Michael Collins wrote:
And thank you! We appreciate it when people make suggestions about
documentation. Everyone wants the program to do something but precious
few people offer feedback on getting the system documented. Please
continue offering suggestions.
I speak as someone new to
THe freeswitch on the iso is about 3 months old. I am working on updating this
but I can get the latest FreeSWITCH to build into an RPM under CentOS.
-Andrew
--- On Wed, 8/6/08, Patrick [EMAIL PROTECTED] wrote:
From: Patrick [EMAIL PROTECTED]
Subject: Re: [Freeswitch-users] GUI
To:
I have to agree with Roberto on not understanding what you’ve done, Brian.
Is toll_allow an undocumented channel variable?
Or did you just define/declare a generic variable on the spot?
If you did, can all user defined variables be in this array format?
_
From: [EMAIL
Hi,
I almost visit the docu wiki of FS daily to see if some documentation has
been changed/added or altered. So, i have to literally go through all the
sections and sub-sections of the wiki to find material of interest which
might have changed in last 24 hours.
I wondered if the home page of
He says yes! It would take less than a week to finalize BRI support.
/b
Sent from my iPhone
On Aug 6, 2008, at 11:01 AM, Wasim Baig [EMAIL PROTECTED] wrote:
On Wed, Aug 6, 2008 at 3:17 AM, Patrick [EMAIL PROTECTED]
wrote:
Wasim Baig wrote:
if i were at cluecon, i'd wish for ...
It would or it will ?
On Aug 6, 2008, at 8:14 PM, Brian West wrote:
He says yes! It would take less than a week to finalize BRI support.
/b
Sent from my iPhone
On Aug 6, 2008, at 11:01 AM, Wasim Baig [EMAIL PROTECTED]
wrote:
On Wed, Aug 6, 2008 at 3:17 AM, Patrick [EMAIL PROTECTED]
On the left side click recent changes. That should help greatly.
/b
Sent from my iPhone
On Aug 6, 2008, at 1:13 PM, Ashutosh [EMAIL PROTECTED] wrote:
Hi,
I almost visit the docu wiki of FS daily to see if some
documentation has been changed/added or altered. So, i have to
literally
Ohhh.. i see that. Thanks for pointing it. :)
Cheers!
-ashu
On Wed, Aug 6, 2008 at 6:24 PM, Brian West [EMAIL PROTECTED] wrote:
On the left side click recent changes. That should help greatly.
/b
Sent from my iPhone
On Aug 6, 2008, at 1:13 PM, Ashutosh [EMAIL PROTECTED] wrote:
Hi,
http://wiki.freeswitch.org/wiki/Special:Recentchanges
_
From: [EMAIL PROTECTED]
[mailto:[EMAIL PROTECTED] On Behalf Of Ashutosh
Sent: Thursday, August 07, 2008 4:14 AM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Documentation suggestion
Hi,
I
I'm using mod_eventsocket.
If I perform the API command
originate sofia/internal/5344 park()
and 5344 is not a valid destination, I receive the following:
2008-08-06 16:04:30 [WARNING] mod_sofia.c:1890 sofia_outgoing_channel()
Cannot l
ocate registered user [EMAIL PROTECTED]
2008-08-06
We're working on the billing solution for fs (astpp) fastpp?
And need some input on what is the best way to limit call length.
In asterisk we've been using L(x[:y][:z]) option to Dial.
Whats the recommended route for this in fs?
Does something like this exist? If not, can I add this to the if
I am trying to determine the capabilities of Freeswitch. I have also read
through the technical recommendations for SIPConnect from the SIP forum. The
reference architecture at the enterprise end shows IP-PBX - SIP proxy -
firewall - internet.
Using Freeswitch is the proxy necessary or is FS
On Thu, Aug 7, 2008 at 2:10 AM, Wasim Baig [EMAIL PROTECTED] wrote:
In asterisk we've been using L(x[:y][:z]) option to Dial.
to follow up, use sched_hangup for the x:
see
http://wiki.freeswitch.org/index.php?title=Misc._Dialplan_Tools_sched_hangup
and for :y and :z use sched_broadcast
see
As a token of your appreciation you could add this to the Rosetta Stone
page... :D
BTW, any time someone says, Here's how we do it in Asterisk and then
someone else says Well in FS you can do that like this... it would be
great to add that to the Rosetta Stone page. That page is kinda thin and
It's there and on the wiki look at the variable list.
Sent from my iPhone
On Aug 6, 2008, at 3:10 PM, Wasim Baig [EMAIL PROTECTED] wrote:
We're working on the billing solution for fs (astpp) fastpp?
And need some input on what is the best way to limit call length.
In asterisk we've been
I just grepped the entire source tree and toll_allow is not a predefined
chan var. In other words, Brian just demonstrated the power and flexibility
of FS by giving you just a few lines of config.
setting toll_allow to a comma-delimited value lets you have have multiple
values that can be
I am making a load test on FS.
I am using a FS box that calls an extension of remote FS box.
I am sending originate calls in background command at once and receiving
responses through mod_socket interface.
Almost 570 out of 1000 calls are having DESTINATION_OUT_OF_ORDER error. I am
sending this
I am making a load test on FS.
I am using a FS box that calls an extension of remote FS box.
I am sending originate calls in background command at once and receiving
responses through mod_socket interface.
Almost 570 out of 1000 calls are having DESTINATION_OUT_OF_ORDER error. I am
sending this
Ray,
I've just updated to rev 9232 and the problem is gone.
I could not find the issue at FS-JIRA (looked into FreeSwitch-Core and
Sofia-Sip).
Well, if someone has really corrected this issue, thank you.
On Thu, Aug 7, 2008 at 5:43 AM, Raymond Chandler [EMAIL PROTECTED]
wrote:
Could you
Hi buddies, any idea where to get this freeswitch.jar file?
Thanks.
On Wed, Aug 6, 2008 at 6:44 PM, Adeel Ansari [EMAIL PROTECTED] wrote:
Hi folks,
I am looking to download freeswitch.jar, but couldn't find it till now. May
be its hidden somewhere, or my mind is quite exhausted right now.
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