Hi Seven,
Thanks, however there is no early_media in this call.
xliteFS---eyebeam. bypass_media is set and the response to the
invite is 180 ringing.
Regards,
Jim
On Wed, May 20, 2009 at 3:51 PM, seven dujinf...@gmail.com wrote:
try ignore_early_media
On May 20, 2009, at 12:45 PM, Jim
I have installed Freeswitch and configured to use SIP Proxy Server for
making calls.
I tested Originate and Bridge commands and both sides of the call are
getting connected.
Now the problem is :-
--I am not recieving any voice/audio on either endpoint of the call ( both
endpoints are mobile
ravi hum ravi_...@yahoo.co.in wrote:
*call request is send from FreeSwitch to SIP Proxy server ( FreeSwitch ---
SIP Proxy Server)
please let me know how to solve this issue.
If there is a NAT device involved anywhere in your scenario, it's probably the
cause.
I briefly looked at the code, so I will then answer my own questions
from what i understood (and I didn't understand much). Please correct me
if I'm wrong (see inline answers).
A new question would then be: Are any of those features planned for the
future of OpenZAP?
On Mon, 2009-05-18 at 10:29
the timer starts as soon as you make the call to sched_hangup
if you want to start it as soon as the call is answered run the command via
the execute_on_answer mechanism
On Wed, May 20, 2009 at 1:15 AM, Jim Burke j...@evolutiontel.net wrote:
Hi Seven,
Thanks, however there is no early_media
The most difficult one would be the cadence detection.
The rest are just based on events from the driver. We have never tried it
in Spain so we unfortunately have not had
a working environment to test it in. I am sure we could strive to support
your requests but it will take time and resources
Thanks Anthony,
I will give it a go.
Cheers,
On Wed, May 20, 2009 at 10:49 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
the timer starts as soon as you make the call to sched_hangup
if you want to start it as soon as the call is answered run the command via
the execute_on_answer
Hello
I have a problem with core dumped with signal 6 Aborted
in gdb looks:
Core was generated by `/usr/local/freeswitch//bin/freeswitch'.
Program terminated with signal 6, Aborted.
#0 0xb7f4f410 in __kernel_vsyscall ()
#1 0xb7c11085 in raise () from
I would update to SVN trunk and try again chances are this has already
been fixed. If it happens after you update please report bugs to http://jira.freeswitch.org
Thanks,
Brian
On May 20, 2009, at 7:48 AM, Mariusz Kołodziejczyk WP wrote:
Hello
I have a problem with core dumped with signal
Ok, feature request for progress detection opened as OPENZAP-70.
Meanwhile, I humbly offer a basic working environment in Spain (a TDM400
with 1xFXO) if you need it for testing.
François.
On Wed, 2009-05-20 at 08:08 -0500, Anthony Minessale wrote:
The most difficult one would be the cadence
In the http://wiki.freeswitch.org/wiki/Getting_Started_Guide under Basic
Calling, it states Call your own extension number to login to your
voicemail box. When I dial my extension (1000) from an internal softphone
at the same extension, it just asks me to leave message. It does not give me
the
Hi N.Baskar,
Create an extension mapping in your dialplan for your conference room as
follows: (make sure this is accessible for both your local extensions
and your public numbers)
extension name=TestConf
condition field=destination_number expression=^testConf(\d+)$
action
Lars,
Thanks for pointing this out. I will update the wiki. The new way to check
voicemail is to dial 4000 and then enter your extension.
-MC
On Wed, May 20, 2009 at 10:42 AM, Larry Marshall l...@marshap.com wrote:
In the http://wiki.freeswitch.org/wiki/Getting_Started_Guide under Basic
*Hi,
I have an issue in transfer the call through JavaScript session.
In JavaScript Session i have dialed 2 numbers
One is mobile number and another one is extension number
I want both the call to transfer in the conference room using JavaScript
session
Whether transfer is possible in
Hello, all. Being new to FS, I was curious if there are any logs/cdrs which
could be generated to gather statistics about a conference call? I'm mainly
looking for call duration and user count. So far, my CDR's only have
individual user CDR's, but nothing for a conference bridge.
Thanks!
Thanks, Michael. I should have looked at conf/dialplan/default.xml.
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Wednesday, May 20, 2009 12:04 PM
To: freeswitch-users@lists.freeswitch.org
Subject:
When I issue a fsctl shutdown via xmlrpc I get a segmentation fault on
Ubuntu server 9.
It a 32 bit version with all packages fully up to date running freeswitch
1.0.3
Any ideas?
Lon
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I would update to svn trunk and try again! ;)
/b
On May 20, 2009, at 4:47 PM, Lon Baker wrote:
When I issue a fsctl shutdown via xmlrpc I get a segmentation fault
on Ubuntu server 9.
It a 32 bit version with all packages fully up to date running
freeswitch 1.0.3
Any ideas?
Lon
Lon Baker l...@kickasspixels.com wrote:
When I issue a fsctl shutdown via xmlrpc I get a segmentation fault on
Ubuntu server 9.
I think there was a fix to fsctl to eliminate segfaults recently. If you
upgrade to trunk it might work now.
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