Hi Guys,
Is it possible to restrain the call-out to be one-way, meaning the callee
can only listen, but not speak? If so, is it possible to switch off the
constraint dynamically during the call and allow the callee to speak?
Thanks,
-Jingwei
___
Jingwei,
the dialplan command eavesdrop does this. The person barging in can use key
presses to dynamically turn on/off voice.
--matt
Voice Broadcasting http://www.hellohunter.com/voice_broadcast.php
Predictive
Dialer http://www.hellohunter.com
On Mon, Sep 28, 2009 at 3:38 PM, Jingwei Yang
Hi Matt,
This looks exactly what I need. Thanks a lot for the direction :)
Regards,
-Jingwei
On Mon, Sep 28, 2009 at 4:55 PM, Matthew Fong mattdf...@gmail.com wrote:
Jingwei,
the dialplan command eavesdrop does this. The person barging in can use key
presses to dynamically turn on/off
Aloysius Thevarajah Lloyd ha scritto:
Hi All,
I am trying to setup FreeSwitch on a Ubuntu Server.
Where can I find the start up(boot time) script for FreeSwitch on a
Ubuntu Server?
Thank you .
http://fisheye.freeswitch.org/browse/FreeSWITCH/debian
See freeswitch.init and
On Sun, Sep 27, 2009 at 5:42 PM, Aloysius Thevarajah Lloyd
lloyd.aloys...@gmail.com wrote:
Hi All,
I am trying to setup FreeSwitch on a Ubuntu Server.
Where can I find the start up(boot time) script for FreeSwitch on a Ubuntu
Server?
Why then not to use ubuntu .deb package? Init script is
Hi,
Does Freeswitch support OCS? We are interested in having our desktop PC
control our in-house desktop phones (e.g. initiate call, answer call, hold
call, etc.) using the uaCSTA protocol.
Best Regards,
Jerry
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FreeSWITCH-users mailing list
On Sat, Sep 26, 2009 at 8:45 PM, Seven Du dujinf...@gmail.com wrote:
Hi, is this a bug?
freeswi...@internal regex 10|09|10
false
freeswi...@internal regex 10|10
true
freeswi...@internal regex 10|(09|10)
false
freeswi...@internal 2009-09-27 11:47:00.815355 [ERR] switch_regex.c:101
Yep escape it.
/b
On Sep 28, 2009, at 10:47 AM, Michael Collins wrote:
regex 10|09\|10
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FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Like jmesquita said, you shouldn't need to do anything special. Just make
sure that the login credentials that you set up for FS match those that you
have in sip.conf so that you don't have to change auth name and/or password
for your SIP phones. FYI, FS does not currently support IAX2 auth.
-MC
On Wed, Sep 23, 2009 at 5:33 AM, Rudá Cunha r...@ruda.com.br wrote:
I upgraded to version 1.0.trunk.
And still with the problem.
I am using the soft phone X-Lite.
I set it (1000) and I connect. Once connected to create the User 1001. Ai
in another X-Lite I connect with the (1001) and create
It was only set up the X-Lite
2009/9/28 Michael Collins m...@freeswitch.org
On Wed, Sep 23, 2009 at 5:33 AM, Rudá Cunha r...@ruda.com.br wrote:
I upgraded to version 1.0.trunk.
And still with the problem.
I am using the soft phone X-Lite.
I set it (1000) and I connect. Once connected to
To confirm links since I got a few questions. My git repo has two
interfaces. One web port that you go to with the browser, and another
that you can clone if you send me your public ssh key.
Web:
http://git.home.quentustech.com:9191/cgit.cgi/freeswitch-ubuntu/
git+ssh:
Brian,
I could not find anything wrong in the way it works for me.
I am new with FreeSwitch, so probably I am missing something.
I am curious to find out what s going.
Do you want me to send snapshots from my logfile for outcoming and
incoming calls?
Igor
I suspect its because its a 180 and not real media inline.
/b
On Sep 28, 2009, at 12:31 PM, Svetik wrote:
Brian,
I could not find anything wrong in the way it works for me.
I am new with FreeSwitch, so probably I am missing something.
I am curious to find out what s going.
Do you want me
Hello!
I followed a tutorial that describes load-balancing Asterisk with
Ultramonkey, but cannot get it to work with FreeSWITCH:
http://www.danielaliaman.com/blog/files/ultramonkeyasterisk.pdf
My X-Lite client fails to register with the server. I have looked at the
packets with wireshark, and
I have been working with a similar setup myself, but for some reason I
ended up ditching the
UltraMonkey setup because I just couldn't get it to work right.
It's been quite a while since my effort, so I don't remember what the
exact issue was.
I got registrations to work, but had some other
Yes, with OpenSER/Kamailio/OpenSIPs/SER (you pick a name).
On Mon, Sep 28, 2009 at 15:12, Mike van Lammeren m...@van.lammeren.net wrote:
Hello!
I followed a tutorial that describes load-balancing Asterisk with
Ultramonkey, but cannot get it to work with FreeSWITCH:
SILK will NEVER take off if they don't stop jerking developers
around. I have a silk binary but its 32bit only... You CAN NOT link a
32bit .a file into a 64bit .so just won't work. And I emailed about
this fact and I got brushed off .. it takes only a few seconds to
compile a 64bit
Brian West br...@freeswitch.org wrote:
SILK will NEVER take off if they don't stop jerking developers
around. I have a silk binary but its 32bit only... You CAN NOT link a
32bit .a file into a 64bit .so just won't work. And I emailed about
this fact and I got brushed off .. it takes
On 09/29/2009 09:05 AM, Jason White wrote:
Brian Westbr...@freeswitch.org wrote:
SILK will NEVER take off if they don't stop jerking developers
around. I have a silk binary but its 32bit only... You CAN NOT link a
32bit .a file into a 64bit .so just won't work. And I emailed about
Thanks. But I think it would be nicer if the regex looks the same as in
dialplan. can we add a optional separator arg on this case?
regex data|pattern [seperator]
regex 10:09|10 :
2009/9/29 Brian West br...@freeswitch.org
Yep escape it.
/b
On Sep 28, 2009, at 10:47 AM, Michael Collins
I'm not one that usually likes to sit around waiting on stuff... but I
waited on SILK and was let down.
/b
On Sep 28, 2009, at 8:22 PM, Steve Underwood wrote:
Its early days for the IETF work, but its possible SILK and CELT might
merge into one codec. The authors are certainly on amicable
On 09/29/2009 09:31 AM, Brian West wrote:
I'm not one that usually likes to sit around waiting on stuff... but I
waited on SILK and was let down.
/b
On Sep 28, 2009, at 8:22 PM, Steve Underwood wrote:
Its early days for the IETF work, but its possible SILK and CELT might
merge into
Hi,
I'm having a strange behavior with the FS when I'm using it with
inboud-late-negotiation=true and with the both scenarios proxy-
media=true or bypass-media=true. The FS is acting as a pseudo
proxy (I know that it is not intend for that).
The configuration is similar to this:
Doesn't SILK scale down on bandwidth much lower than CELT can?
-Michael
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jason White
Sent: Monday, September 28, 2009 6:26 PM
To: FreeSWITCH-users
That would be I suspect because your AddPac gateway is BROKEN. I have
no idea why it would be saying PCMU/8000/3 unless its horribly
broken. The SOA in sofia is moving the answer to 96 because that SDP
is not valid for 0 which is a single channel of ulaw. I don't know
about you but I
That all depends on how you want to configure it :P
/b
On Sep 28, 2009, at 11:33 PM, Michael Giagnocavo wrote:
Doesn't SILK scale down on bandwidth much lower than CELT can?
-Michael
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Thanks for the response.
I've already seen what you are referring to, however when FS is
handling the media and the AddPac does exactly the same, everything
works fine, so I assumed that sofia is ignoring the las part of the
map, and that is why I said that theoretically the capture was
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