sure you open a
bug on jira.freeswitch.org with as much detail to reproduce this as
possible.
Mike
On Sep 10, 2009, at 6:14 PM, Jan Kubr wrote:
Hi,
we have a Freeswitch server on a public IP and a few phones behind
NAT. The phones are configured to use STUN and can register and call
each
Yes, the phones are behind NAT. Freeswitch is on a public IP.
j
On Tue, Sep 29, 2009 at 3:10 PM, Brian West br...@freeswitch.org wrote:
NAT involved?
/b
On Sep 29, 2009, at 7:27 AM, Jan Kubr wrote:
I commented out the following in our internal profile:
param name=media-option value
Hi,
we have a Freeswitch server on a public IP and a few phones behind
NAT. The phones are configured to use STUN and can register and call
each other fine.
The problem is that after attended transfer (using the mechanism the
phones provide - REFER) is finished, the two parties can't hear each
You can use this setup to control calls, but I believe not to change
configuration. Dunno what you're after.
On Sunday, August 30, 2009, tom tomabr...@gmail.com wrote:
hi,
i was actually looking for something completely else, but reading wiki led
me to the tought:
if mod_event can
I have been experiencing this as well. It happens randomly and I
haven't been able to find out what the issue is. I think there is some
delay when the RTP ports are being negotiated/allocated. Or something.
What helped me a bit: I start with playing a file containing 1 second
of silence and only
We have a SIP gateway behind NAT which I haven't been able to set up
to work with Freeswitch. The configuration I thought would work is:
include
gateway name=nat
param name=username value=user/
param name=password value=pass/
param name=realm value=11.12.13.10/
param name=proxy
PM, Jan Kubr jan.k...@gmail.com wrote:
We have a SIP gateway behind NAT which I haven't been able to set up
to work with Freeswitch. The configuration I thought would work is:
include
gateway name=nat
param name=username value=user/
param name=password value=pass/
param name
Creating a separate sofia profile just for this gateway definitely works,
just wondering whether there is a cleaner solution. The register-proxy
params seems to do something very similar..
On Sun, Jun 21, 2009 at 1:39 PM, Jan Kubr jan.k...@gmail.com wrote:
I have found this: http
and/or from-user I suspect.
/b
On Jun 21, 2009, at 7:16 AM, Jan Kubr wrote:
Creating a separate sofia profile just for this gateway definitely
works, just wondering whether there is a cleaner solution. The
register-proxy params seems to do something very similar
stuff to consider.
Jay
2009/6/18 João Mesquita jmesqu...@gmail.com
Pricewise, is it worth it?
jmesquita
On Wed, Jun 17, 2009 at 6:31 PM, Jan Kubr jan.k...@gmail.com wrote:
We plan to buy one of these:
http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway
We plan to buy one of these:
http://www.2n.cz/products/gsm_gateways/voip_gsm_gateway/voiceblue_enterprise_voip_gsm_gateway.html
since you can use SMTP/POP3 to manage SMS.
Jan
2009/6/17 João Mesquita jmesqu...@gmail.com
Guys, I was looking at the advantages and disadvantages of having a GSM
Hi all,
I've created a simple framework in Ruby that you can use to talk to
Freeswitch via even socket outbound. It won't suite your needs
perfectly if you are doing anything non-trivial, but it might be a
nice starting point.
Check it out at http://github.com/jankubr/freec
Cheers,
Jan Kubr
approach? I'm on revision 10751. I've tried to set a few configuration
variables based on suggestions from this list, but it didn't make any
difference.
Thanks,
Jan Kubr
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http
pass
the uuid, I get -ERR invalid session id. I can always pass it
explicitly though, so no big deal.
Jan Kubr
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, 2008 at 4:48 AM, Jan Kubr [EMAIL PROTECTED] wrote:
OK my bad. The variable is set (I can see it in the Freeswitch console
when I use the info app), but they are only not send to me via the
socket interface. I get the variable_* variables only in the
beginning (after calling connect
the info app right after to see all the vars.
I'm not saying i don't believe you but it seems fishy.
On Sun, Dec 7, 2008 at 5:31 AM, Jan Kubr [EMAIL PROTECTED] wrote:
Hi,
I checked out the current trunk (rev 10641) and found out that the
read app ignores the varname parameter, it always puts
release.
Any ideas? Thanks,
Jan Kubr
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http
any
docs, either.
Any ideas would be appreciated.
Jan Kubr
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no solution. I have a similar problem, when calling Freeswitch from my
cell phone (via a SIP provider), sometimes DTMF is not recognized
The important thing to note is that when using
a SIP softphone (X-Lite) I have never had this problem, DTMF is
So i guess that using latest version with
, at 12:30 PM, Jan Kubr wrote:
I have try different format of files (from 8KHz mono wavs to MP3s, all
of which play fine via playback) and some caused the bridge to be
finished immediately (with NO_USER_RESPONSE), some make it generate
crazy beeping, but none is played while the phone is ringing
Hi Faisal,
the path is either an absolute path or a path relative to the
directory in the sound_prefix var in vars.xml.
So this
action application=playback
data=/usr/local/freeswitch/sounds/en/us/callie/misc/8000/call_secured.wav/
works fine on my box. You sure this one doesn't work for you?
Hi,
I would like to play a sound file while bridging the call. The wiki
says that the ringback variable can be set to a file name as well.
However, I haven't been able to get this to work (as I thought it
would).
action application=set data=ringback=ring.wav/
action application=bridge
I'll stick to what I'm doing now which is that I replace the at-sign
with a dot and take that as a username.
Sorry for the newbie question and thanks all for the answers.
Its better to use a dash... a dot is valid in the username part of an
email address... so what dot do you split on?
You can't have two @'s in a uri. You do know its already domain
based.. you have domains that have users inside them? so you can dial
user/[EMAIL PROTECTED]
already without any extra thought? You can have multiple domains in
the directory... with users in each domain.
These domains
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