William,
When I upgrade freeswitch-config package, all my customized config
files are silently overwritten with the default ones from .deb
package. This bug is quite annoying.
- Dmitry Bely
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I'm still having this issue with random EXCHANGE_ROUTING_ERROR disconnects
for users utilizing my inbound DID to connect to my FreeSWITCH server. It's
a predictive dialing application, with one agent session being bridged with
multiple calls and transfered back and forth between extensions in my
Hi,
Is there anyway to configure freeswitch so that it won't retry the SIP
message when 200 OK is not received?
thanks,
jb
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Hi Max,
Some operators (e.g. Orange in the UK) will allow you to have two
numbers attached to one SIM, which might get you 4/10ths of the way to
your goal.
I'm struggling to see why purchasing DIDs won't help, unless your FS box
has no internet connectivity. A bit more background on the problem
On Sun, Oct 18, Dmitry Bely wrote:
William,
When I upgrade freeswitch-config package, all my customized config
files are silently overwritten with the default ones from .deb
package. This bug is quite annoying.
Very sorry. I can try to include some code that detects if your moving from
Thanks Dave. I'm essentially trying to build a service in one country where
there is no DID provider. If DIDs were available for such countries, that
would have done the job.
Using the two-numbers-per-SIM approach would help significantly, if I could
only figure out, how to detect the Called
Nope its part of SIP it'll retry if a response isn't received in a
certain time.
/b
On Oct 18, 2009, at 10:29 AM, Juan Backson wrote:
Hi,
Is there anyway to configure freeswitch so that it won't retry the
SIP message when 200 OK is not received?
thanks,
jb
On Sun, Oct 18, 2009 at 8:40 PM, Frank Carmickle fr...@carmickle.com wrote:
On Sun, Oct 18, Dmitry Bely wrote:
William,
When I upgrade freeswitch-config package, all my customized config
files are silently overwritten with the default ones from .deb
package. This bug is quite annoying.
You can also try this
http://wiki.freeswitch.org/wiki/Channel_Variables#progress_timeout
/b
On Oct 18, 2009, at 10:29 AM, Juan Backson wrote:
Hi,
Is there anyway to configure freeswitch so that it won't retry the
SIP message when 200 OK is not received?
thanks,
jb
long time friend, diegoviola i going to help about the freeswitch
wiki markup
we are waiting for french users to join, and welcome
thank toAnthony Minessale for accepting the community
welcome!
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r...@rupa.com (Rupa Schomaker) writes:
On Sat, Oct 17, 2009 at 1:44 AM, Simon J Mudd sjm...@pobox.com wrote:
I've other plans like:
1. choose the cheapest gateway based on some scripts which download
rates from the various gateways web pages and let me compare rates.
Hard partly as the
!
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FreeSWITCH-users@lists.freeswitch.org
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FreeSWITCH-users
thank toAnthony Minessale for accepting the community
welcome!
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Diego Viola diego.vi...@gmail.com wrote:
I propose that we start a new documentation from the ground up, we really
need a better wiki engine that supports inline translation and a better
search engine. I'm open to do all the hard work like porting the entire
documentation to it.
Is there a
No, the checkbox is unchecked.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-
boun...@lists.freeswitch.org] On Behalf Of russell.mosem...@cune.org
Sent: Friday, October 16, 2009 10:30 AM
To: freeswitch-users@lists.freeswitch.org
On 10/18/2009 11:49 AM, Dmitry Bely wrote:
William,
When I upgrade freeswitch-config package, all my customized config
files are silently overwritten with the default ones from .deb
package. This bug is quite annoying.
Hope you have a recent backup. Years ago when I created rpm packages for
FreeSWITCH debug level logs should help tell you exactly what is
killing the call.
On Oct 18, 2009, at 10:25 AM, Matthew Fong wrote:
I'm still having this issue with random EXCHANGE_ROUTING_ERROR
disconnects for users utilizing my inbound DID to connect to my
FreeSWITCH server. It's a
Thanks, Russell, that did it. Works perfectly!
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-
boun...@lists.freeswitch.org] On Behalf Of Russell Mosemann
Sent: Sunday, October 18, 2009 5:40 PM
To: freeswitch-users@lists.freeswitch.org
users to join, and welcome
thank toAnthony Minessale for accepting the community
welcome!
__ Information from ESET NOD32 Antivirus, version of virus
signature database 4520 (20091018) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
I've tried all sorts of debug and parameter changes over the weekend,
but still can't figure out the correct solution.
If I disable timers on the sip profile then all works fine.
param name=enable-timer value=false/
But that seems like a hack; not a correct solution. With the build
13168M
The debug level logs to the console, right? The pastebin, had log level
debug, sofia trace on for external and default, and sofia loglevel all 9. Is
there another log enable command I'm missing? It seems loglevel all 9
outputs enter and exit functions, but at least to my novice eye, it's not
too
Hi!
I simply want to validate the dtmf digits I read from a user.From
the wiki, it appears I need to use inline=true when setting the
variable so it can be used directly within the same extension.
What have I done wrong below? I have tried many different
alternatives, but the second
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