I was on this conference and there were 4 of us that got dropped at the same
time. By the way, these were all regular PSTN lines calling through our SIP
provider and being routed to our freeswitch instance which is being hosted
on a VPS.
When you say a device on my end terminated the call, do you
Hello,
I tried conferencing for FS und tried to lock/unlock conferences.
While conf-is-locked.wav was played, conf-is-unlocked.wav was
missing in the file system.
Any idea where I can download this?
Best regards
Peter
___
Freeswitch-users mailing
It appears to be a broken client. Your client doesn't ack with the to
tag like zoiper does.
/b
On May 5, 2009, at 5:10 AM, seven wrote:
My problem is why the first one keep sending 404s even got the ACK?
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
First off you're not on SVN trunk secondly Are you executing the
conference app inside your js file? If so then there could be the
problem! You have also forgotten to include anything about Distro,
OS, CPU and Memory.
/b
On May 5, 2009, at 12:12 AM, Stephen Crosby wrote:
We had our
Thank you. Even the client is broken, we cannot fix that as we don't
own the code. But we need that client, is that possible to make FS
work around that?
On May 5, 2009, at 6:41 PM, Brian West wrote:
It appears to be a broken client. Your client doesn't ack with the
to tag like zoiper
Forgot to add that my OS is Ubuntu 8.04LTS (hardy heron).
--Stephen
On Tue, May 5, 2009 at 7:42 AM, Stephen Crosby stevecr...@gmail.com wrote:
I know I'm not on svn trunk, but this is a production server and it's just
not feasible to update it constantly. I can update it though if you think I
You should rule out the network problems first, which sound more likely.
you can reduce the overuse of JS if you transfer the call to a regular
extension with a dynamic regex.
session.execute(transfer, conf-xyz);
then make a regex in your xml dialplan to pick up ^conf-(.*) and execute
Hi All,
In an inbound call center scenario is it possible that customers calls in
and calls are distributed between online (who are registered on FS and in
idle state) agents. I saw some on going discussion on list where it looks
that currently it's not possible but I am newbie so maybe I didn't
Hi Michael,
Thanks for a quick reply.
I would definitely create a test environment, but my question is that will
it work in required way?
I read that in Mod_fifo agent has to call in queue but I need that all
incoming calls are automatically distributed between available agents or if
all are
I want to invite another party into a conference with TLS and SRTP enabled.
Internal phones are invited by the following dialstring:
{originate_timeout=30,sip_secure_media=true,context=default}sofia/default/723...@sip2.mydomain.de
72332200 Conference'.
This enables SRTP but no TLS.
Is there any
now append transport=tls
{originate_timeout=30,sip_secure_media=true,context=default}sofia/default/723...@sip2.mydomain.de
;transport=tls
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
___
Freeswitch-users
When I append transport=tls I recieve the following and the call is not
initiated:
2009-05-06 00:01:37 [DEBUG] mod_sofia.c:83 sofia_on_init()
sofia/internal/723...@sip2.mydomain.de;transport=tls SOFIA INIT
2009-05-06 00:01:37 [DEBUG] sofia_glue.c:1972 sofia_glue_build_crypto()
Set Local Key [1
The far end challenged you and it looks like you couldn't answer said
challenge.
/b
On May 5, 2009, at 5:06 PM, Peter P GMX wrote:
Cannot create outgoing channel, cause: MANDATORY_IE_MISSING
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
The far end is a Snom phone which I can dial the normal way (Snom - FS
- Snom) via TLS.
So I have no clue what to do now. Any hint?
Best regards
Peter
Brian West schrieb:
The far end challenged you and it looks like you couldn't answer said
challenge.
/b
On May 5, 2009, at 5:06 PM, Peter
FYI,
I just wanted to let the community know that we maintain a
bloghttp://cluecon.com/blog/1on the ClueCon
website http://www.cluecon.com. Please check it out. The latest entry
mentions Moises Silva's recent blog entry about his speaking at ClueCon this
year.
Please check the ClueCon blog
On May 6, 2009, at 1:50 AM, Saeed Ahmed wrote:
Hi Michael,
Thanks for a quick reply.
I would definitely create a test environment, but my question is
that will it work in required way?
I read that in Mod_fifo agent has to call in queue but I need that
all incoming calls are
Hi all
I'm not familiar with the configuration for freeswitch.
Anyone who could help me sovle the following two problem:
1. user A is in conversation with user B, and at this time, a incoming call
from user C comes to A, in this case, I want freeswitch to play busytone to C,
how to
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