Hi,
Michael Collins wrote:
Hello all!
There's been some discussion lately on how to handle multiple
languages, specifically with the *say* application. We would like some
input from the community on how to handle multiple languages and sound
files. Anthony notes that the say application
Hello,
I have the following problem: Every call stops after 30 seconds when TLS
is enabled. SIP/RTP and SIP/SRTP works but not TLS/SRTP.
The phones are behind NAT. So I expect, that every 30 seconds an Options
request is sent.
Wiresharking the traffic I can see
* that there are ongoing UDP
Some additions:
TLS/RTP instead of SRTP does also not work.
There are no logs on the debug console except the message that the call
is being terminated
2009-07-02 12:06:45.252177 [DEBUG] sofia.c:3100 Channel
sofia/internal/835...@sip.mydomain.de entering state [terminating][0]
and later cause:
, version of virus signature
database 4209 (20090702) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo
If its TLS you don't need options packets in the first place. Your
client should do the keep alive NOT FreeSWITCH. TLS is over TCP and
Options over UDP... doesn't make much sense.
/b
On Jul 2, 2009, at 6:11 AM, Peter P GMX wrote:
Hello,
I have the following problem: Every call stops
set the variable sip_invite_domain
/b
On Jul 2, 2009, at 12:56 AM, Jason White wrote:
I suspect the other end (whatever device you are calling from
FreeSWITCH) is
adding the IP address to the caller id. However, I am no SIP expert
and may be
wrong, but you can confirm this by doing a
/ Online
now i need to call / recev call from / to GTalk (Jingle)
cool anyone give me a sample Dialplan Extention?
thanks
__ Information from ESET NOD32 Antivirus, version of virus
signature database 4209 (20090702) __
The message was checked by ESET NOD32 Antivirus.
http
Thanks for that.
That seems to successfully re-invite and re-route the the B leg - but does
not reinvite the A leg and then immediately issues a bye on both legs.
Do I have to do something to reinvite that A leg?
On Wed, Jul 1, 2009 at 7:06 PM, Anthony Minessale
anthony.miness...@gmail.com
Antivirus, version of virus
signature database 4209 (20090702) __
The message was checked by ESET NOD32 Antivirus.
http://www.eset.com
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
http://lists.freeswitch.org
Thanks for responding and for your help.
The xml and confirm.js are attached below. Basically trying to bypass_media
after the leg B presses 1 to accept the call. I tried,
using bypass_media_after_bridge=true, but the re-invite appears to be done
before the confirm.js, So the media is
On Thu, Jul 2, 2009 at 3:01 AM, Igor Neves i...@3gnt.net wrote:
Hi,
Michael Collins wrote:
Hello all!
There's been some discussion lately on how to handle multiple languages,
specifically with the *say* application. We would like some input from the
community on how to handle multiple
Michael Collins wrote:
On Thu, Jul 2, 2009 at 3:01 AM, Igor Neves i...@3gnt.net
mailto:i...@3gnt.net wrote:
Hi,
Michael Collins wrote:
Hello all!
There's been some discussion lately on how to handle multiple
languages, specifically with the *say* application. We
Used:
session.execute(set,bypass_media_after_bridge=true);
in the confirm.js script and that works perfectly!
Thank you for you help!
On Thu, Jul 2, 2009 at 1:05 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
try setting bypass_media_after_bridge=true on the session in your confirm
sound prefix should not be used for lang just language
the sound_prefix is automatically built as
${base}/sounds/${language}/
each time you execute say.
and restored to its previous val when the say is over.
On Thu, Jul 2, 2009 at 12:07 PM, Michael Collins m...@freeswitch.org wrote:
On
no problem
On Thu, Jul 2, 2009 at 12:27 PM, Phillip Jones pjinthe...@gmail.com wrote:
Used:
session.execute(set,bypass_media_after_bridge=true);
in the confirm.js script and that works perfectly!
Thank you for you help!
On Thu, Jul 2, 2009 at 1:05 PM, Anthony Minessale
Hello Brian,
ok, I got it. Any other idea why the UDP port is closed after the TLS
packet?
Best regards
Peter
Brian West schrieb:
If its TLS you don't need options packets in the first place. Your
client should do the keep alive NOT FreeSWITCH. TLS is over TCP and
Options over UDP...
On Fri, 2009-07-03 at 01:29 +0800, Steve Underwood wrote:
Michael Collins wrote:
On Thu, Jul 2, 2009 at 3:01 AM, Igor Neves i...@3gnt.net
mailto:i...@3gnt.net wrote:
Hi,
Michael Collins wrote:
Hello all!
There's been some discussion lately on how to handle
On Thu, 2009-07-02 at 15:47 -0300, Raul Fragoso wrote:
On Fri, 2009-07-03 at 01:29 +0800, Steve Underwood wrote:
Michael Collins wrote:
On Thu, Jul 2, 2009 at 3:01 AM, Igor Neves i...@3gnt.net
mailto:i...@3gnt.net wrote:
Hi,
Michael Collins wrote:
Thanks Brian, that worked like a charm. :)
On Thu, Jul 2, 2009 at 6:37 AM, Brian West br...@freeswitch.org wrote:
set the variable sip_invite_domain
/b
On Jul 2, 2009, at 12:56 AM, Jason White wrote:
I suspect the other end (whatever device you are calling from
FreeSWITCH) is
adding
Raul Fragoso wrote:
On Fri, 2009-07-03 at 01:29 +0800, Steve Underwood wrote:
Michael Collins wrote:
On Thu, Jul 2, 2009 at 3:01 AM, Igor Neves i...@3gnt.net
mailto:i...@3gnt.net wrote:
Hi,
Michael Collins wrote:
Hello all!
There's been some
On Jul 2, 2009, at 7:50 PM, Steve Underwood wrote:
If by the usual way you mean the standard 2 + 2 letter codes we are
used to on computers, that just doesn't work. As I said before, those
are for written languages, not spoken languages. There are no standard
codes for many spoken
Guys, I don't know if I really get the problem here. I mean, I do get that
the 2+2 model does not work not even for where I live.
I really hate the fact that all spanish south american dialects (some within
the same country) are put in the same bag as it wouldn't matter to ppl so I
am with you
I have a GSM gateway. The issue is sometimes the calls failed what is the
cause of the error this is my logs?
This is on my freeswitch logs...
09-06-25 10:21:50 [DEBUG] sofia_glue.c:1913 sofia_glue_tech_set_codec() Set
Codec sofia/internal/1...@116.541.23.11 PCMU/8000 20 ms 160 samples
Try enabling 3pcc in the sip profile.
On Jul 2, 2009, at 11:12 PM, Edmar Cruz darklio...@yahoo.com wrote:
I have a GSM gateway. The issue is sometimes the calls failed what
is the
cause of the error this is my logs?
This is on my freeswitch logs...
09-06-25 10:21:50 [DEBUG]
The same issue...
Michael Jerris wrote:
Try enabling 3pcc in the sip profile.
On Jul 2, 2009, at 11:12 PM, Edmar Cruz darklio...@yahoo.com wrote:
I have a GSM gateway. The issue is sometimes the calls failed what
is the
cause of the error this is my logs?
This is on my
25 matches
Mail list logo
