Of course, it will be useful to try it.
Artem
Hi Artem,
Please to see that some of the stuff I wrote is useful to someone..!
I've written an FS module which will send the audio over - it's more
efficient than using unicast. Let me know if you'd like a copy.
Cheers --
Dave
Cool, looking forward to test this :)
Even André
On 27. mai. 2009, at 04.51, Diego Viola wrote:
Sorry, this is the link.
http://wiki.freeswitch.org/wiki/Mod_nibblebill#Hangup_the_call_when_the_balance_is_depleted
Diego
On Tue, May 26, 2009 at 9:06 PM, Diego Viola diego.vi...@gmail.com
I want to do the following:
Originate a call via event_socket, I get back a job_uuid. Then I want to
control the call when it's established (2 call legs).
Scanning the variables of the 2 call legs I currentyl cannot see any
relation between the job_uuid and the uuid of the resulting call legs.
I
Peter P GMX pisze:
I want to do the following:
Originate a call via event_socket, I get back a job_uuid. Then I want to
control the call when it's established (2 call legs).
Scanning the variables of the 2 call legs I currentyl cannot see any
relation between the job_uuid and the uuid of the
Hi Anthony,
thanks, it seems to be working, but the extension is not hanging up once I
transfered the call to another mobile or to a conference. Some where I am
wrong?
On Tue, May 26, 2009 at 6:28 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
This one happens to every new guy
Any how let me try with uuid_kill to kill the extension uuid.
On Tue, May 26, 2009 at 6:28 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
This one happens to every new guy trying to make FS into a dialer app using
JS.
for every sessionX you create in js with the new Session
I am not sure what you mean at this point.
On Wed, May 27, 2009 at 5:53 AM, Gopalakrishnan A.N sai...@gmail.comwrote:
Hi Anthony,
thanks, it seems to be working, but the extension is not hanging up once
I transfered the call to another mobile or to a conference. Some where I am
wrong?
Here are 3 ways:
1) subscribe to the BACKGROUND_JOB event and find the one with the same
job-uuid
then the body of that message is the output from your backgrounded FSAPI
call which in the case
of an originate will contain the uuid of the actual channel.
2) You can do as suggested and
next time try this:
1) Read the error.
Error [unterminated ${var}] in line
/usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xml
line 12
ok, so there appears to be a problem in
/usr/local/freeswitch/conf/autoload_configs/../jingle_profiles/client.xml
line 12
and the
Hey Damin!
Glad to see you are still out there in the shadows.
You coming back to ClueCon this year?
On Tue, May 26, 2009 at 11:31 AM, Gregory Boehnlein da...@nacs.net wrote:
I can say, from having met with and talked to the CEO and founder of
Applogic that these guys are really
I faced a problem when I want to compile freeswitch with mod_opal. It gives me
a compilation error like bellow
I used ptlib-2.6.2 and opal-3.6.2.
Would someone please help me?
Sincerely,
Sadjad
making all mod_opal
make[5]: Entering directory
Hi everyone!
In my environment I use FreeSwitch as media server and session border
controller. SIP routing is mostly done with my private B2BUA. The problem
itself is in my hold functionality. In details: A is calling to B:
!--
if the calling party is the called party, go to their VM
You have to use the SVN version of both ptlib and OPAL and it will
compile.
/b
On May 27, 2009, at 3:26 AM, Sadjad Seyed-Ahmadian wrote:
I faced a problem when I want to compile freeswitch with mod_opal.
It gives me a compilation error like bellow
I used ptlib-2.6.2 and opal-3.6.2.
Wow, that's cool. Can you give us some figures? How many users/calls per
day, what is the AMI setup, an average cost per month? Do you think it would
be a feasible solution for a call center?
On Tue, May 26, 2009 at 18:21, Erik Wickstrom e...@erikwickstrom.comwrote:
I've been running a
Check the freeswitch log when the dtmf are being received, most likely your
device is sending them both as Dave says, maybe the telephone you're dialing
with sends it both inband and rtp. What does the log says?
2009/5/26 Drew Ozier drew.oz...@gmail.com
I've got a configuration where I receive
Try not using RFC2543 HOLD since we do not support it.
/b
On May 27, 2009, at 7:38 AM, Artem Shiyanov wrote:
Hi everyone!
In my environment I use FreeSwitch as media server and session border
controller. SIP routing is mostly done with my private B2BUA. The
problem
itself is in my hold
Thanks for the answer!
Maybe you can advise me another scheme how to accomlish mute or hold
functionality? I just wonder if mute/hold in X-Lite works with FreeSwitch..
I'll try and send the results.
On Wed, May 27, 2009 at 6:22 PM, Brian West br...@freeswitch.org wrote:
Try not using RFC2543
Yes, but if the stream says inactive we might have an issue but I
can't recall off the top of my head... but Last I tested x-lite it
works fine.
/b
On May 27, 2009, at 9:34 AM, Artem Shiyanov wrote:
Thanks for the answer!
Maybe you can advise me another scheme how to accomlish mute or
To clarify, I'm not running the IVR. I have a TDM T1 that comes in to an
AudioCodes Mediant 1000 (SIP Gateway) which goes to a FreeSWITCH machine. I
receive calls coming in off the T1 which goes through my Mediant 1000 which
goes to my FreeSWITCH machine which makes an outbound call back through
You should try to always use out of band... what is the call path
because it looks like a bridged call and the far end gets the inband
and the 2833
/b
On May 27, 2009, at 10:02 AM, Drew Ozier wrote:
To clarify, I'm not running the IVR. I have a TDM T1 that comes in
to an AudioCodes
Michael,
Thanks for the advice. Where would you place this extension in my current
conf/dialplan/default.xml to avoid the integration issue?
extension name=Local_Extension
condition field=destination_number expression=^(10[01][0-9])$
action application=set
Lars,
Please register and correct anything you see wrong on the wiki if
possible.
Thanks,
Brian
On May 27, 2009, at 10:16 AM, Lars Zeb wrote:
Also, on http://wiki.freeswitch.org/wiki/Mod_cdr_csv, near the
bottom of the page under ‘uuid’, the two hyperlinks (‘ref’) are
broken.
Brian
I'll get the docs section fixed this has changed and will require us
to fix it.
Thanks,
On May 27, 2009, at 10:16 AM, Lars Zeb wrote:
Also, on http://wiki.freeswitch.org/wiki/Mod_cdr_csv, near the
bottom of the page under ‘uuid’, the two hyperlinks (‘ref’) are
broken.
Brian West
Hey Brian,
It is a bridged call. Here's the majority of my dialplan:
condition field=source expression=mod_sofia
action application=ring_ready/
action application=pre_answer/
action application=start_dtmf/
action application=set
Still here.. working.. making a living.. you know the drill. Not sure about
ClueCon this year.. I've curtailed a lot of my traveling and talking this
year as I've been doing a lot more consulting work. I'll look into it.. I
may be able to drive up if I can spare the time..
From:
This is what starts the inband detector.. what is the far side?
/b
On May 27, 2009, at 10:25 AM, Drew Ozier wrote:
action application=start_dtmf/
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
___
I'm sorry, far side meaning what exactly? The call flow is as follows:
TDM T1 - AudioCodes Mediant 1000 - FreeSWITCH - AudioCodes Mediant 1000
- TDM T1
FreeSWITCH gets an inbound SIP call and makes an outbound SIP call back to
the Mediant 1000, bridging the two calls. The Mediant 1000 sends the
I would configure both mediant 1000's to do rfc2833, or disable 2833
on your sofia profile and do 100% inband.. right now I have a feeling
the mediant is getting the inband and the 2833 causing your problem.
/b
On May 27, 2009, at 10:45 AM, Drew Ozier wrote:
I'm sorry, far side meaning
Are the configurations of the audiocodes set to send only in-audio dtmf?
make sure everything is sending only in-audio
2009/5/27 Drew Ozier drew.oz...@gmail.com
To clarify, I'm not running the IVR. I have a TDM T1 that comes in to an
AudioCodes Mediant 1000 (SIP Gateway) which goes to a
I highly recommend you DO NOT use inband AT ALL if possible. out of
band is king.
/b
On May 27, 2009, at 10:49 AM, Milena wrote:
Are the configurations of the audiocodes set to send only in-audio
dtmf? make sure everything is sending only in-audio
Brian West
br...@freeswitch.org
--
The AudioCodes machines are always a bit of a nuisance to get acting the way
I want, but I think I have them sending all DTMF in-band. How do I disable
2833 in my sofia profile?
-Drew
On Wed, May 27, 2009 at 11:52 AM, Brian West br...@freeswitch.org wrote:
I highly recommend you DO NOT use
Hello Thanks for your hints,
I now added {initiator_uuid=my_uuid} prefix to the dialstring.
Then I catch the channel_answer event, get this variable_initiator_uuid
and pass it to the application. This works like a charm.
Thanks to all.
Best regards
Peter
Anthony Minessale schrieb:
Here are 3
It's been under pretty light use. About 20 users. A bunch of DIDs coming
in and some outbound campaigns. A couple hundred calls a day. (we also did
a test for an outbound campaign with 8 telemarketers making 1000s of calls
in a day -- worked great!)
The AMI is based on Ubuntu 8.04. We're
Thanks j3flight, I have used that method, but the profiles seem to be for
conferences not users. So I give a conference a profile, and everybody in
the conference shares the profile settings. If I'm wrong, let me know,
otherwise I'll stop hijacking this thread.
--Stephen
On Wed, May 27, 2009 at
Is transcoding a wav file something that I should try to avoid? I have
my files encoded as 16kbit, 8000hz, mono files, yet still in the logs
I see:
2009-05-27 11:55:18 [DEBUG] switch_ivr_play_say.c:1084
switch_ivr_play_file() Codec Activated l...@8000hz 1 channels 20ms
2009-05-27 11:55:18
Thanks Anthony,
Will do that next time :)
On Wed, May 27, 2009 at 9:04 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
next time try this:
1) Read the error.
Error [unterminated ${var}] in line
/usr/local/freeswitch/conf/
autoload_configs/../jingle_profiles/client.xml
line
Chances are those wav files are L16 either way its not an issue to
transcode those... now if you were transcoding 48k to 8k all day long
then I would worry but I think you're looking for savings in places
where it doesn't matter in FreeSWITCH.
/b
On May 27, 2009, at 11:25 AM, Greg Thoen
Hello
I have about 100 phones in lan. Every phone have set rtp port to 2
(very simple phone 8Level you can set one rtp port not range like
linsys can)
my logs in freeswithc looks:
2009-05-27 17:51:10 [INFO] mod_dialplan_xml.c:252 dialplan_hunt()
Processing 101-166 in context default
I've just tried native hold X-Lite hold with FreeSwitch - it works.
When you hold call, X-Lite (eyeBeam) sends re-INVITE with SDP
v=0
o=- 8 3 IN IP4 172.16.0.6
s=CounterPath eyeBeam 1.5
c=IN IP4 0.0.0.0
t=0 0
m=audio 43362 RTP/AVP 3 101
a=fmtp:101 0-15
a=rtpmap:101 telephone-event/8000
a=sendonly
Nope, it seems you are absolutley correct. I had setup my conferences this
way, but hadn't experimented yet. I tried putting two users into the same
conference using different profiles, but they both had the same caller
controls. Bummer.
I believe we can get around this though, assuming I can
Yes you won't hear anything if you press HOLD... the other caller you
were talking to will hear music.
/b
On May 27, 2009, at 11:45 AM, Artem Shiyanov wrote:
Probably mentioned sopftphones simply do not play incomming media
when the call is holded?
In general, is how should I hold a call
On Wed, May 27, 2009 at 8:20 AM, Brian West br...@freeswitch.org wrote:
I'll get the docs section fixed this has changed and will require us to fix
it.
Thanks,
On May 27, 2009, at 10:16 AM, Lars Zeb wrote:
Also, on http://wiki.freeswitch.org/wiki/Mod_cdr_csv, near the bottom of
the page
I've checked again (thanks for your hint) and, really, FreeSwitch is so wise
that he plays MOH for the holded person and and silence for the hold
initiator!
I'm going to is this work_flow for my B2BUA.
Thanks!
On Wed, May 27, 2009 at 8:47 PM, Brian West br...@freeswitch.org wrote:
Yes you
Michael, thanks for fixing the links.
And what about your advice on defining an extension that pertains all
internal extensions (1000-1019) which I have currently put in
conf/dialplan/default.xml? Is there an alternative file I should be putting
this into?
Lars
From:
On Wed, May 27, 2009 at 11:28 AM, Lars Zeb larc...@yahoo.com wrote:
Michael, thanks for fixing the links.
And what about your advice on defining an extension that pertains all
internal extensions (1000-1019) which I have currently put in
conf/dialplan/default.xml? Is there an alternative
Well actually I have an average 15 telemarketers running on a small (650
euros) server with the same load (an average 1%). Of course availability and
scalability are on a different level but it's no easy to build a case -
which is the more cost effective scenario with this and a growing (50+
Did a few TCP dumps, and before, I was seeing explicit DTMF packets coming
out of my FreeSWITCH machine. Removing the only things I could find that had
to do with 2833 in my internal and external sip_profiles (namely
rfc2833-pt), I find that I don't see any mention of switch_rtp.c sending any
DTMF
It would be an improvement to move the caller controls to the member so it
works
the way you expect but it will have to wait for the right time and
motivation level.
From there it would also be possible to make up some new caller controls
that were moderator inspired like (mute all besides
Hi,
We're considering deploying FS instead of a traditional PABX/Key-System in
a small office environment (i.e. primarily non-technical users, 15-20
handsets).
Anyone have any experiences (good/bad/whatever) in this sort of scenario?
Thanks,
Neale.
When I create an xml file in conf/dialplan/default, I assume that it's
'inheriting' its context, right?
If that's so, do I use this structure, for example
conf/dialplan/default/00_MyExtension.xml:
include
extension name=Local_Extension
.
/extension
/include
Thanks, Lars
Order matters if something is loaded higher that matches which the
debug log shows you this if you press F8
/b
On May 27, 2009, at 5:56 PM, Lars Zeb wrote:
When I create an xml file in conf/dialplan/default, I assume that
it’s ‘inheriting’ its context, right?
If that’s so, do I use this
On Wed, May 27, 2009 at 3:56 PM, Lars Zeb larc...@yahoo.com wrote:
When I create an xml file in conf/dialplan/default, I assume that it’s
‘inheriting’ its context, right?
Correct. Files in conf/dialplan/default/*xml are included in the default
context while conf/dialplan/public/*xml are
On Wed, May 27, 2009 at 3:51 PM, Neale Banks ne...@lowendale.com.au wrote:
Hi,
We're considering deploying FS instead of a traditional PABX/Key-System in
a small office environment (i.e. primarily non-technical users, 15-20
handsets).
Anyone have any experiences (good/bad/whatever) in this
Hey Brian,
I dug it!
Regards,
On Wed, May 27, 2009 at 5:47 AM, Brian West br...@freeswitch.org wrote:
Dear FreeSWITCHers,
Now I'm gonna take a moment here to guilt each and everyone of you into
checking out the story about Pre8 on Digg. We have all worked long and hard
to get to 1.0.4 and
Thank you... now let me explain how this works. The more DIGG's you
get in a short period of time... the more the chance you get on the
front page. If you don't get that in the first 5 hours you're pretty
much SOL. :P So there was a bit of urgency to that... maybe we'll be
ready for
What can I say, first time digger :(
On Thu, May 28, 2009 at 9:35 AM, Brian West br...@freeswitch.org wrote:
Thank you... now let me explain how this works. The more DIGG's you get in
a short period of time... the more the chance you get on the front page.
If you don't get that in the first 5
IMHO, you have tons of features w/ FS. i've setup FS on a low-power
consumption Intel D945GCLF2 motherboard (Atom dual-core CPU) ideal for 24/7
operation on a 10-seat contact center w/ default conversation recording. no
problem.
another cool feature. you can route the call based on the Caller ID.
I use it on a 12 extension office. Works like a charm. Specially because I
host it on a cheap dedicated server (iWEB).
The only thing I would say is to be careful not to loose focus on your
primary business and start developing your own GUI for the pbx. I have seen
that happen with lots of
Anthony,
Thank you for your suggestions! We are working on 1), but need to
re-integrate code we've changed, and do regression testing. That's in
progress, and we expect to be able to upgrade by the end of next week.
We did manage to do 3) and 4), and we now have SIP logs (attached). Are you
Keep that mouse clicker ready! :)
/b
On May 27, 2009, at 6:52 PM, Jim Burke wrote:
What can I say, first time digger :(
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
___
Freeswitch-users mailing list
Quoting Mr. Anthony Minessale:
thThe easiest way would be the new feature I added to 13442
in the conference profile add
param name=conference-flags value=wait-mod/
to your profile
and in your dialplan
action application=set data=conference_member_flags=*moderator*/
action
Wiki Tax paid...
That was my first contribution to the freeswitch wiki!
MC, you're welcome to have a look over it and see if i made things clear
enough. Feel free to edit.
On Tue, May 26, 2009 at 4:56 PM, Anthony Minessale
And the wiki tax if you feel comfortable adding this to the wiki. If
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