Hi,
We have FreesSWITCH runing on version 11066, which INVITE with
i...@30i number 102, however the recent version INVITE with i...@30i
number 97. Unfortunately I have a broken sip client only talks with
RTP payload 102. So here is the question:
1) What's the difference between 97 and 102?
Once you're in dynamic range you should compare the codec name not
number.
/b
On May 8, 2009, at 2:04 AM, seven wrote:
Hi,
We have FreesSWITCH runing on version 11066, which INVITE with
i...@30i number 102, however the recent version INVITE with i...@30i
number 97. Unfortunately I have a
I've narrowed this problem down.
When I call my ISP's DTMF test and issue DTMF from the Snom phone, do_2833()
from switch_rtp.c is never called, as evidenced by freeswitch.log.
However, if I call a friend's FreeSWITCH box from the phone (via my FreeSWITCH
instance), do_2833() is called. It is
On May 8, 2009, at 3:18 PM, Brian West wrote:
Once you're in dynamic range you should compare the codec name not
number.
/b
On May 8, 2009, at 2:04 AM, seven wrote:
Hi,
We have FreesSWITCH runing on version 11066, which INVITE with
i...@30i number 102, however the recent version INVITE
No thats not how the dynamic range works. You can change it in
mod_ilbc.c but again NOT the right way. Your phone is broken and
should be fixed.
/b
On May 8, 2009, at 2:50 AM, seven wrote:
Thank you. It must be changeable somewhere in the source code, can
you help me to find out? I
Jason White ja...@jasonjgw.net wrote:
It is also called if I use the voicemail
extension on my local FreeSWITCH.
Apologies for the nonsense - I meant that switch_rtp_dequeue_dtmf() is called
in that case, for DTMF detection.
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As a matter of interest, the other end (as reported in its SDP) is BroadWorks.
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Sorry for all the e-mail...
If I turn off the jitter buffer that I had set in the dialplan extension for
that provider, DTMF is correctly sent and detected by the other side.
I suspect a bug, but maybe this is the desired behaviour.
___
But there is only two 30ms variants,
SWITCH_CODEC_TYPE_AUDIO, 97, iLBC, mode=20,
SWITCH_CODEC_TYPE_AUDIO, 98, iLBC, mode=30,
SWITCH_CODEC_TYPE_AUDIO, 102, iLBC, mode=30,
Why FS INVITE with mode=30 and 97 but not 98/102 as I'm using i...@30i ?
v=0
o=FreeSWITCH 6662257026041736756
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Hash: SHA1
Hello Brian,
no, during conversations calls were desconnected.
regards
Helmut
On 07.05.2009 17:09, Brian West wrote:
Were they on hold for a bit?
/b
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Hello again,
I spent some more time investigating the problem. The scenario seems to
be always the same:
- -Phone A calls Phone B on PSTN (OpenZAP+Sangoma) via FS.
- -Call is established and Phone A has a (G722)SRTP connection to FS.
- -After a
Sound bugish to me - or at least not desired behavior.
I'd suggest opening up a jira (jira.freeswitch.org) with as much
documentation as you have so it can be researched and resolved.
On Fri, May 8, 2009 at 3:46 AM, Jason White ja...@jasonjgw.net wrote:
Sorry for all the e-mail...
If I turn
Hi everyone,
I'm currently developing a calling card application that uses event
socket and mod_nibblebill to bill calls. Well, the question is: can
mod_nibblebill disconnect a call when the balance is depleted, or when
it reaches 0 cash?
The wiki says:
Allow for disconnecting or re-routing
The mod modules are still not built. I tried with your suggestion.
making install mod_cluechoo
Compiling mod_cluechoo.c...
quiet_libtool: compile: gcc -I/usr/src/freeswitch/src/include
-I/usr/src/freeswitch/libs/libteletone/src -fPIC -Werror -fvisibility=hidden
-DSWITCH_API_VISIBILITY=1
Okay, time for a fresh checkout. If you are on Linux then I recommend that
you completely rm -fr your freeswitch source directory, then do the quick
and dirty install:
http://wiki.freeswitch.org/wiki/Quick_and_Dirty_Install
Let us know if that works...
-MC
On Fri, May 8, 2009 at 10:20 AM, Lars
That solved it! Is there any downside to this method of keeping the
nat binding alive?
--
Greg
On May 8, 2009, at 12:49 PM, Mike Tkachuk wrote:
Hello Greg,
It's a NAT box issue. Nat bindings expire if no activity.
Try adding a:
param name=ping value=30 /
to your gateway params.
But to be
Oh I see that it has it already :D
!--
By default, warn a caller when their balance is at $5.00. You can set
this to a negative number.
--
param name=lowbal_amt value=5/
param name=lowbal_action value=play ding/
-
!--
By default, terminate a caller when their balance hits $0.00. You can
set
I have set these actions:
param name=nobal_amt value=0/
param name=nobal_action value=hangup/
But when it reaches 0 cash it doesn't hangup :(.
On Fri, May 8, 2009 at 2:34 PM, Diego Viola diego.vi...@gmail.com wrote:
Oh I see that it has it already :D
!--
By default, warn a caller when
Hello,
I am trying to get a new freeswitch installation working and I have it
registering to a sip provider just fine and can receive inbound calls
but when I try and place a call out through the gateway the switch is
rejecting it due to the domain in the to of the INVITE.
Here is the
Sorry to report that the results are the same. I executed the 'make' command
after the 'wget' and captured stderr and stdout to a file. I uploaded it to
http://rapidshare.com/files/230726648/openSuse_11.1_Freeswitch-1.0.4pre7_log
.zip.html if it's of any use.
The mod_cluechoo errors persist.
Well, Tony likes to say that you can compile FreeSWITCH anywhere but he only
supports it if the name of the OS rhymes with Mentos. :)
We have had much success with CentOS 5.x installs. I can highly recommend
it.
-MC
On Fri, May 8, 2009 at 1:46 PM, Lars Zeb larc...@yahoo.com wrote:
Sorry to
Is there a way to compile and install without all the example
configuration information?
A truly clean install out of the gate.
Lon
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cd /usr/local/freeswitch/
rm -rf conf
but You'll have to do it all by hand its better off to strip down the
config once you understand how FreeSWITCH works.
/b
On May 8, 2009, at 3:22 PM, Lon Baker wrote:
Is there a way to compile and install without all the example
configuration
Woof!
On Fri, May 8, 2009 at 4:22 PM, Lon Baker l...@kickasspixels.com wrote:
A truly clean install out of the gate.
We took a different approach. Rather than change what FS comes with,
we created an alternate configuration area and point FreeSWITCH to it
when we start.
Here's the script:
Thanks for the tip!
-MC
On Fri, May 8, 2009 at 2:20 PM, Andy Spitzer w...@iwoof.org wrote:
Woof!
On Fri, May 8, 2009 at 4:22 PM, Lon Baker l...@kickasspixels.com wrote:
A truly clean install out of the gate.
We took a different approach. Rather than change what FS comes with,
we created
What does your dialplan entry look like? Also, a debug trace and even a sip
trace would be useful. You can use pastepin.freeswitch.org to paste a lot of
stuff in a place where everyone can view it and not overwhelm the email
list.
-MC
On Fri, May 8, 2009 at 12:36 PM, Dale fdh...@gmail.com wrote:
Thanks for the response. Of course after sending the email I find the
problem. The account was not properly setup on the other end to place
outbound calls. So everything is working now.
Thanks,
-Dale
On May 8, 2009, at 6:45 PM, Michael Collins wrote:
What does your dialplan entry look
Rupa Schomaker r...@rupa.com wrote:
Sound bugish to me - or at least not desired behavior.
I'd suggest opening up a jira (jira.freeswitch.org) with as much
documentation as you have so it can be researched and resolved.
If someone could add it to Jira, I'll detail the issue here. The Jira
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