Brian,
Does this mean that Bria should be sending the invite directly to
sip:8...@conference.freeswitch.org and the From address should be the address
of the Bria itself rather than FS?
Lars
From: freeswitch-users-boun...@lists.freeswitch.org
once you have 123456 won't you still be unsure if he will type the next 1 or
not and be forced to refuse it and wait anyway?
basically you are right. BUT, we know, that a basic phone number has 6
digits - so, we do not have to check anything before. as soon as we
have 6 digits, we look in our
Thanks for answer.
As I've understood eavesdrop allows C to hear A and B, but A and B will not
hear C. But in my case B SHOULD hear C. In my case A - client, B - agent
worked with client and C - coach/supervisor for B. Can I make such
conference in FS using eavesdrop or other API command?
hi all
how can i get the digits when users in the conference??
and,in conference.conf.xml
control action=mute digits=0/ the action will set another
value?e.g:transfer?
thanks
2009-10-15
god.nirvana
___
FreeSWITCH-users
The wiki has a pretty good explanation of how eavesdrop works.
Enabling a talk path to A or B or both A and B requires dtmf.
So, if C hits the 1 button on the phone they can talk to the UUID you
bound the eavesdrop to. 2 talks to the other party. 3 talks to both
parties.
no clue it depends on how you have it configured. If you have an
outbound proxy set it'll send all calls to the proxy. If not then
most likely it will go direct to the destination.
/b
On Oct 15, 2009, at 12:00 AM, Lars Zeb wrote:
Brian,
Does this mean that Bria should be sending the
right you can reply 484 in your dp at any time
action application=respond data=484 Address Incomplete/
then it should try again.
The bit i can't remember is if we committed a certain 1 line patch that
makes sofia parse the next invite to the same call properly, the patch was
to the sofia lib
This is a known limitation until we add actual t38 support to the project.
On Wed, Oct 14, 2009 at 6:56 PM, Klaus Hochlehnert maili...@kh-dev.dewrote:
Hi,
sometimes I have the problem that after doing an attended transfer the
media got stuck in FS.
Meaning the call goes through, but I
Thanks, now it's working, but in my user directory I have something like
this :
user id=john number-alias=1368
and in my dialplan.xml I have to put the two entry for working :
condition field=destination_number expression=^(toto|john|8400|1368)$
If I only put john, when I try to call the
ok, we will try this with the cirpack of our carrier. this will take
some days, till everything is set up.
after the tests i will come back to report.
2009/10/15 Anthony Minessale anthony.miness...@gmail.com:
right you can reply 484 in your dp at any time
action application=respond data=484
Hey Orien,
OK, let's have a look here. How about going to the console of your FS and
posting the output of the following commands:
sofia status profile internal
sofia status profile external
We'll take it from there. Your problems are just like what I ran into
initially, so I'm hopeful we get
Thanks, Michael, using make current resolved the issue. I will use only
that command in the future.
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Tuesday, October 13, 2009 1:12 PM
To:
Hi Michael,
m...@freeswitch.org (Michael Collins) writes:
On Wed, Oct 14, 2009 at 3:01 PM, Simon J Mudd sjm...@pobox.com wrote:
I'm trying to build up a soho dialplan to replicate an existing Asterisk
setup I want to replace.
...
My question is how do I avoid repeatedly using these
I don't think I have a proxy. It's pretty standard installation.
Name Type Data
State
=
internal profile
On Oct 15, 2009, at 11:03 AM, Simon J Mudd wrote:
OK. That looks clear. However it doesn't look as if it's possible to
pass parameters this way which means that data can only be passed to
the extension via variables. That works but isn't so clean. I'll
check this out further shortly. I'm
Lars Zeb larc...@yahoo.com said:
I don't think I have a proxy. It's pretty standard installation.
Is the phone itself configured to use a proxy?
--
Russell Mosemann
Concordia University, Nebraska
See http://www.cune.edu/ for the
Hi list,
I'm playing around with this wonderful, mighty solution.
Setup so far:
I created a gateway pointing to my ITSP (sip.qsc.de). Outbound calls getting in
an depending on the channel variable
sip_to_user I want to rewrite the sip_from_user (=text which get's displayed in
the phones):
Hi,
just found the answer:
just use
effective_caller_id_name
and it work's :)
Cheers
Stefan
Stefan Weigel
Advanced IT-Professional
Tel.: +49 89 2000 48 975
Fax: +49 89 2000 48 566
eMail:
stefan.wei...@allianz-warranty.commailto:stefan.wei...@allianz-warranty.com
Allianz Automotive
I don't think Bria is set to use a proxy.
In the SIP Account/Account tab, it is set to Register with domain and
receive calls and Send outbound via is set to use Domain rather than
Proxy Address.
The Account ID is set to 1...@192.168.10.29, where the IP is the address of
FreeSwitch.
Not sure at all what you are asking. apr_queue is simply a data
management container that is threadsafe.
--
Yossi Neiman
Cartis Solutions, Inc. - http://www.cartissolutions.com
srinivasula reddy wrote:
Hi all,
does any know about How apr_queue is maintaing and retriving all
registered
Mike,
I copied the status screen from the freeswitch package in the web
setup. I you need a console view, I will need instructions on how to
console into pfsense, Since I have never done that.
Orien L.
Name Type
DataState
Lars Zeb larc...@yahoo.com said:
I don't think Bria is set to use a proxy.
In the SIP Account/Account tab, it is set to Register with domain and
receive calls and Send outbound via is set to use Domain rather than
Proxy Address.
The Account ID is set to 1...@192.168.10.29, where the IP
On Thu, Oct 15, 2009 at 7:56 AM, Lars Zeb larc...@yahoo.com wrote:
Thanks, Michael, using “make current” resolved the issue. I will use only
that command in the future.
Excellent! Thanks for the feedback.
-MC
___
FreeSWITCH-users mailing list
On Thu, Oct 15, 2009 at 7:10 AM, Jonathan Barou jba...@sqli.com wrote:
Thanks, now it's working, but in my user directory I have something like
this :
user id=john number-alias=1368
and in my dialplan.xml I have to put the two entry for working :
condition field=destination_number
Hi,
I tried to originate a call from x-lite and hope to terminate to SIPp
running in another machine (192.168.1.36:5069)
through FS. My x-lite received service no available and the FS report
NORMAL_TEMPOARY_FAILURE
(http://pastebin.freeswitch.org/10720). My purpose for FS is to simply
forward the
hi, any clue when can t38 be added?
T.
On Thu, Oct 15, 2009 at 3:57 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
This is a known limitation until we add actual t38 support to the project.
On Wed, Oct 14, 2009 at 6:56 PM, Klaus Hochlehnert maili...@kh-dev.dewrote:
Hi,
Russell,
My settings in Bria are just like yours.
I captured the SIP trace from the Bria and it says it is using the
FreeSwitch as the Outbound Proxy. I am very confused.
I tried dial sip:e...@iptel.org. In the pastebin you can see that this
address came over to the cli as echo. Is it Bria
Hi,
Does Freeswitch support TLS Client-Authenticated handshake. Openssl does,
but it has to be enabled in order to send the certificate request to the
client.
I tested simple TLS hanshaking and it wotks well.
Thanks.
___
FreeSWITCH-users mailing list
Hi!
I was wondering if the Dingaling implementation in FS supports DTMF?
This is now supported in the jingle specs (
http://www.jabberforum.org/showthread.php?t=2709 ), even though Google
Talk client does not currently support DTMF.
If not, are there plans to implement this?
Thanks!
Erwin Davis davis.er...@gmail.com wrote:
action application=bridge data=sofia/internal/profile1/$
0...@192.168.1.36/
The above syntax won't work.
data=user/$...@192.168.1.36 (assuming that your FreeSWITCH domain is
192.168.1.36) would be better.
If the other end doesn't require
On Thu, Oct 15, 2009 at 4:22 PM, Jason White ja...@jasonjgw.net wrote:
Erwin Davis davis.er...@gmail.com wrote:
action application=bridge data=sofia/internal/profile1/$
0...@192.168.1.36/
The above syntax won't work.
data=user/$...@192.168.1.36 (assuming that your FreeSWITCH domain
I prefer the second case. How to define the domain? Thanks,
e
On Thu, Oct 15, 2009 at 7:22 PM, Jason White ja...@jasonjgw.net wrote:
Erwin Davis davis.er...@gmail.com wrote:
action application=bridge data=sofia/internal/profile1/$
0...@192.168.1.36/
The above syntax won't work.
On Thu, Oct 15, 2009 at 11:54 AM, Tihomir Culjaga tculj...@gmail.comwrote:
hi, any clue when can t38 be added?
Eventually. :) Of course, if we could get more to add to the bounty it
might grease the wheels of innovation.
Erwin Davis davis.er...@gmail.com wrote:
I prefer the second case. How to define the domain? Thanks,
It's the domain of FreeSWITCH itself. You could just specify $${domain} if you
aren't using multiple domains.
user/6...@$${domain}
By default, the domain is the IPv4 address of the machine
It updates the display on a phone if the phone supports this. This
works on some sip phones right now including polycom and snom.
Mike
On Oct 12, 2009, at 2:11 AM, Matthew Fong wrote:
Hi Mike,
I'm just trying to send it an event with some custom event headers,
just so an external
Thanks Brian. Is this something that is planned to be implemented?
The workaround is to set the stun server also in the dingaling
configuration, but as I said, for some reason the stun times for me
out occasionally with dingaling.
Thanks!
On Wed, Oct 14, 2009 at 11:33 AM, Brian West
Hey Orien,
pfsense has a nice interface, but I only use it for adding the FS package,
maybe messing with LAN and WAN addresses. Using the command line helped me
to better understand what FS was doing and how. Pfsense just made it easier
to have an OS and really good firewall married to FS without
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