Hi Rupa,
None. That's exactly the point.
Everything has to be done over the usb HID interface.
I've been reading about HID yesterday. HID is a usb interface that can
be used for a large number of things, ranging from keyboard and
game-controllers up to water-cooling and PC-chassis and
mercutioviz wrote:
Additionally, turn on debugging on the console and capture that output. If
you use fs_cli it has debug output turned on by default.
Thanks for the tip. I launched fs_cli, typed sofia profile internal
siptrace on, and then made a call from XLite to the GS phone, with the
I guess I can limit the amount of debug data in the CLI by choosing the right
debug level:
http://wiki.freeswitch.org/wiki/Sofia#Debugging_Sofia-SIP
What is the recommended way to debug SIP connections like I'm having?
--
View this message in context:
Yes i'l be happy to see some working examples :)
I can't fully understand how freeswitch conceptually manage presence events.
And i not found any information about it in wiki.
With default configuration fs sends some notifications to subscribed phones
without use any external scripts, but this
Hi Rupa,
Thanks for your feedback.
I am currently running proxy mode, but whenever I try to force G.729 on
in-bound and out-bound calls, I get an error in my logs to the effect the
G.729 is only a pass-through codec. Both originator and reciepient have
G.729 codecs.
Have you seen this before ?
On 12/23/2009 11:29 AM, David Knell wrote:
On the other hand, a u-law WAV turned into L16 and then back to u-law to
be sent down the line shouldn't suffer any alteration at all - if it
does, the there's something wrong with the translation.
The quality dropping over time is almost certainly
FWIW, I downloaded and compiled the latest trunk (16041), and am still having
this issue.
--
View this message in context:
http://old.nabble.com/-Dialplan---Call-either-sent-to-VM-or-phone-doesn%27t-ring-at-all-tp26892767p26902800.html
Sent from the Freeswitch-users mailing list archive at
Hi There
Our Freeswitch cluster receives inbound calls via a SIP trunk from our
supplier. I currently have an issue where when a call is sent to voicemail
using session:execute(record), our supplier will terminate the call with a
BYE approximately 30 seconds into the recording.
They believe
Of course there is a way. Depending on the interface your looking at
either a freeswitch endpoiny module or an openzap module.
Mike
On Dec 23, 2009, at 4:54 AM, Kristoff Bonne kristoff.bo...@skypro.be
wrote:
Hi Rupa,
None. That's exactly the point.
Everything has to be done over the
On 12/23/2009 04:55 AM, Ahmed Naji wrote:
Hello people,
Can someone please clear the following ambiguities with codecs:
1. Are we definitively able to run pass-through codecs (e.g. G.729)
in Proxy Media mode, or does FS need to be running in
bypass-media ? the Wiki is not
There is no such thing as freeswitch 1.5. Have you tried latest svn
trunk to see if this behavior is the same?
Mike
On Dec 23, 2009, at 7:49 AM, Lei Tang lei.tl...@gmail.com wrote:
Hi all, I'm using FS 1.5, doesn't somebody known something about
this problem?
My scenario is :
VMD will force a transcode anyway too.
/b
On Dec 23, 2009, at 1:08 AM, Vinuth Madinur wrote:
My setup is as follows:
FreeSWITCH - SIP Trunk - PSTN.
From freeswitch, I'm making outbound calls using event socket via the
external profile. Except for the ext_rtp_ip and ext_sip_ip,
What does pretty much mean to you? Can you give me an exact rev?
/b
On Dec 23, 2009, at 8:26 AM, TTNC - Technical wrote:
Oh, I'm running pretty much the latest svn truck.
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FreeSWITCH-users@lists.freeswitch.org
http://wiki.freeswitch.org/wiki/Variable_record_waste_resources
On Wed, Dec 23, 2009 at 8:26 AM, TTNC - Technical techni...@ttnc.co.uk wrote:
Hi There
Our Freeswitch cluster receives inbound calls via a SIP trunk from our
supplier. I currently have an issue where when a call is sent to
It's more than highly likely you have some other problem like jitter or a
bad network connection.
Not many people would be able to tell the difference between the sound of an
8k PCM file and the same file encoded to G711 just by listening to it unless
there was a severe problem somewhere. Since
More information: I can dial the default extensions like just fine. It's
only when I call any of the IP phones (1001,1002,1003) that the call is
immediately forwarded to the callee's voice-mail when the phone goes off the
hook.
To only keep the SIP messages in the fs_cli screen, typing
Yes DNS is required for this to work properly.
/b
On Dec 23, 2009, at 9:43 AM, John wrote:
Still having this issue. Do separate domains need to be real fully
qualified domains, or can they just be added as in Company1, 2, 3, etc?
___
That usually means they are saying 30 but sending 10 which is broken.. you
can't say hey i'm sending 30 and then send 10... find a new provider or beat
them to death with a cluebat in hopes they fix their broken stuff.
/b
On Dec 23, 2009, at 9:48 AM, Matthew Fong wrote:
I use the SIP
I haven't played with any of these usb-to-rj11 in a long time but from what
I do recall it is picked up as an audio device. That way a regular telephone
can be used for mic/speaker for a softphone. Given that you should be able
to get audio to/from it using portaudio for starters. Pretty basic and
They don't operate their own voip gateways, just run an SBC in front of a
bunch of other providers. So someone they are reselling is using Sonus
gear. I use them to originate to some destinations but in the US I avoid
them due to the sonus stuff that pops up on certain routes.
On Wed, Dec 23,
If I only care about outbound audio, is there a way to force the audio
packets FreeSWITCH sends to be of a certain ptime (like 30ms)? Or is there
still this same issue?
--matt
On Wed, Dec 23, 2009 at 8:20 AM, Rupa Schomaker r...@rupa.com wrote:
They don't operate their own voip gateways, just
You can disable auto-adjust in the sip profile., but that might just
make it worse, no warranty:
param name=rtp-autofix-timing value=false /
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 23-Dec-09, at 11:41 AM, Matthew
You might also have to set the codec negotiation to scrooge
/b
On Dec 23, 2009, at 10:53 AM, Mathieu Rene wrote:
You can disable auto-adjust in the sip profile., but that might just make it
worse, no warranty:
param name=rtp-autofix-timing value=false /
Mathieu Rene
Avant-Garde
Yes,
I noticed the Jira for the situation where the where the fs controlled
skype client generates both an In Band audible DTMF tone and an API
signal causing potential confusion for devices down the line. If only
the skype client had an option not the generate the tone in the first
place that
Scott,
do as tony wrote,
=
add start_dtmf app to your dialplan before bridge to start the
inband dtmf detector.
=
-giovanni
On Wed, Dec 23, 2009 at 7:00 PM, Scott Torr
scott.torr...@letterboxes.org wrote:
Yes,
I noticed the Jira for the situation where the where the fs controlled
Ooops, Had not seen you got it in the dialplan...
try to move it after the answer and test again.
Other than this, only thing that comes in my mind is that the
conversion from the pstn to sip (skype partner that gives pstn access)
to skype is ruining the dtmfs beyond recognition... but you said
On Tue, Dec 22, 2009 at 7:58 PM, Joseph L. Casale jcas...@activenetwerx.com
wrote:
Am I correct in presuming that Freeswitch will answer a fax from a local
zap based user
just like it does from an FXO port connected to a POTS line? What I hope
to do here is
catch any call made from that
On Wed, Dec 23, 2009 at 7:39 AM, Fred-145 codecompl...@free.fr wrote:
More information: I can dial the default extensions like just fine.
It's
only when I call any of the IP phones (1001,1002,1003) that the call is
immediately forwarded to the callee's voice-mail when the phone goes off
Mike,
You were right. I turned iptables off and the phone registered.
Thanks so much, Lars
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Jerris
Sent: Tuesday, December 22, 2009 8:39 PM
To:
Hello all!
Because the holidays fall on consecutive Fridays this year we decided to
have a single conference call on Wednesday Dec 30th at the usual time of
11AM CST. The agenda is posted here:
http://wiki.freeswitch.org/wiki/FS_weekly_2009_12_30
Thanks for supporting the weekly calls. Don't
I am trying to setup a second FS box from scratch using v16048.
What can cause a local call (81002, or 9996) to use context public? It's a
standard vanilla install.
http://pastebin.freeswitch.org/11629
Thanks, Lars
___
FreeSWITCH-users
2009-12-23 15:00:01.955357 [DEBUG] sofia.c:5322 IP 192.168.10.105 Approved by
acl 192.168.10.0/24[]. Access Granted.
Because the context is set on the profile as public... and you really didn't
auth the user so user_context was never set.
/b
On Dec 23, 2009, at 7:49 PM, Lars Zeb wrote:
I am
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