Another question:
When I try routing calls through the public context to the default context,
global variables (set in vars.xml) seem to be “forgotten” and appear blank.
I’m trying a very simple scenario of an incoming call on the public context
routed to a phone number which is rightfully
I’m trying to set up a default provider via the example.com.xml using the
variables set in vars.xml.
The provider has a registration expiry of 120 seconds and I’m trying to set
it up to register every 60 seconds but when I change the “expire-seconds”
variable (in
You would have to reload and restart the profile for that to take
effect. You can't change the global and have it magically start using
the new value.
/b
On Feb 7, 2009, at 3:39 AM, UV wrote:
I’m trying to set up a default provider via the example.com.xml
using the variables set in
No the vars are there can you provide more detail on what exactly
you're doing? The default config uses the call_debug variable and its
a global set in vars.xml.
/b
On Feb 7, 2009, at 3:40 AM, UV wrote:
Another question:
When I try routing calls through the public context to the default
Great, thanks for that.
One of the big issues with Asterisk's way of billing is that if let's
say a remote phone diverts a call to another number, say a mobile,
because a local channel is created for the redirect, Asterisk loses
critical information such as the account code and therefore cannot
Hi Guys,
Is there any form of Answer phone detection in FS? A search hasn't
really brought up anything
Regards,
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
Obviously. XML was reloaded, sofia profiles were reloaded, freeswitch app
was shutdown and restarted and last but not least, I’ve rebooted the
computer several times just to make sure :-)
No, seriously, I’ve done everything to verify the settings are there – it’s
just not re-registering. My
I took the out-of-the-box public context dialplan, added an entry to dial a
10 digit number through the default context and when it ran, I noticed in
the console log that all the values below are either null or empty.
To be more specific:
In the public context I’ve added this extension in
turn on the sofia debug. param name=debug value=1/ on the
profile. I will suspect that the far end proxy is forcing your expire
to a higher number.
/b
On Feb 7, 2009, at 7:14 AM, UV wrote:
Obviously. XML was reloaded, sofia profiles were reloaded,
freeswitch app was shutdown and
Please do... also make sure you have a context param on your gateway.
/b
On Feb 7, 2009, at 7:30 AM, UV wrote:
I took the out-of-the-box public context dialplan, added an entry to
dial a 10 digit number through the default context and when it ran,
I noticed in the console log that all the
Nik,
Right now there is mod_vmd. It sets the channel variable vmd_detect if it
detects a beep. If a beep is detected it will set vmd_detect=TRUE. If no
beep is detected then it won't do anything.
Example of usage as follows (with the outcome being hangup if answering
machine is detected):
Will do.
Now I have a little problem delaying me as the latest build changed
something with the sound file playing and now the FS can’t find any local
file to play (adds mysterious \16000\ to the file location…). I’ll try to
isolate this first then get back to this issue.
_
From:
…. And correct you are! Far end proxy does force to a higher number… Thanks!
_
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Brian
West
Sent: Sunday, February 08, 2009 12:32 AM
To:
Hi,
I notice that there is a newly-developed mod_easyroute model available. Has
anyone used it with large amount of routes ( ex 1M ) on a high traffic
scenario? For that kind of scenario, would it be better to consider using
out-going event socket to serve that purpose? I would greatly
Mod_easyroute can handle millions of numbers... It is NOT however an LCR
module... There are other things for that... Look at mod_lcr or if you need
an extremely high performance LCR contact me off list
From: Juan Backson juanback...@gmail.com
Reply-To: freeswitch-users@lists.freeswitch.org
Its trying to open the file to match the current channel rate... you
can install the 16k files via make hd-sounds-install hd-moh-install
/b
On Feb 7, 2009, at 9:26 AM, UV wrote:
Will do.
Now I have a little problem delaying me as the latest build changed
something with the sound file
tamas,
the opposite. The default is to not do one by one and setting the var to
false makes it more i/o intensive but it would
provide more real time recording when recording to streams.
BTW the reversed logic is fixed in tree
On Fri, Feb 6, 2009 at 3:36 PM, Tamas jal...@gmail.com wrote:
Yeah, I have all the sounds installed. I don’t think it’s that.
I’m getting error messages such as “[ERR] mod_sndfile.c:185
sndfile_file_open() Error Opening File
[E:\FS/sounds/en/us/callie\voicemail/8000\16000\vm-goodbye.w] [System error
: The system cannot find the path specified.]” all across
You have a \ somewhere in your path... which doesn't make sense...
you're on windows.
Can you open a jira... I think this was the cause
http://fisheye.freeswitch.org/browse/FreeSWITCH/src/mod/formats/mod_sndfile/mod_sndfile.c?r1=11090r2=11601
/b
On Feb 7, 2009, at 6:46 PM, UV wrote:
Hi,
In my deployment scenario, I plan to have two redundant freeswitch
servers running on two different boxes. Two key features I am
leveraging on freeswitch are voicemail and call recording and
playback., and as a result of that, a shared storage for playback of
the recorded wav files is
Whats wrong with NFS?
As long as you put a reasonable disk subsystem you'll be fine... GFS sucks
for voice anyway... It can take several seconds to get a lock...
No matter what you use, you have to remember that you *MUST* have a cluster
aware file system, simply mounting the same iscsi or SAN
Hi,
I am a real newbie.
I have been building Asterisk based applications for a couple of years
now.
I am looking at migrating these apps to FreeSwitch - eventually.
I want to do this gradually - I need to keep things running in the
meantime.
I have two Asterisk boxes, A1 A2, each running
Followed Wiki to install and configure mod_cepstral. The problem is FS
always defaults to one voice, which I installed first, and ignores others.
I did define SWIFT_HOME and added swift lib path to /etc/ld.so.conf.
After I restart FS I see on FS console after dialing my test extension:
Failed
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