Hi all,
I call outbound gateway using following dialstring:
sofia/outbound/[EMAIL PROTECTED]
here 192.168.1.29 is any SIP provider (e.g. asterisk)
In case when the number 9122 is not registered on that provider, so provider
hangup with SIP response "503 Service Unavailable".
Now my query is that
Hi,
I received loads of useful hints on IRC yesterday regarding this
problem, and as a result was able to come up with a decent solution
which just involves using the API and 'runtime' functions to run my
post-processing job in a thread after hangup. I don't think it is
possible to pass objects
Concerning TLS and SRTP on S60 see
http://mosh.nokia.com/common/download/4452B13D5F854A8DE040050A45306C1B/original/Developing_3rd_party_VoIP_clients_on_S60_platform_v1_0_en.pdf
But I a not sure whether they use SDES or Mikey for key exchange.
Brian West schrieb:
> Eric,
> I wasn't aware tha
Hello,
I am using Ruby on Rails for managing endpoints(directory) and dialplans
in Freeswitch.
However I am wondering whether it is possble to dynamically generate
external SIP gateways. I suspect this is done using the configuration
bindings.
When I enable this I receive the following xml_cur
All,
Can a voice file be played to a user (streamFile) while that user is
recording (recordFile)?
As well can two simultaneous files (streamFile) be played?
Thanks,
Bob
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All,
Can recordFile be paused?
Bob
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Eric,
I wasn't aware that the s60 did TLS yet nor was I aware that it does
SRTP till very recently. I do know that the phone is very picky and
thats the extent of my knowledge on that. Still not 100% sure on
those facts.
/b
On Sep 12, 2008, at 1:17 PM, R. Eric Bennett wrote:
> No
Note that I was mostly just hoping to find someone who'd already had
experience trying to do SIP over TLS with an s60 device, not asking
you folks to do add developing that knowledge in order to have you do
my debugging for me.
That said, I did leave out a few things in the interest of note
On Sep 12, 2008, at 11:26 AM, Kin Quek wrote:
> Brian,
> Thanks for your advice. I will like to join FS on the IRC and to
> further learn from you and the FS team. What is the most effective
> way to set up an experimental FS environment so I can quickly move
> up the learning cureve? What
Hi,
Problem found, somebody had changed my setup and add activated the recording of
all calls, without notifying me. I was out of disk space.
PG
From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Patrick Grondin
Sent: September-12-08 11:05 AM
To: freeswitch-users@lists.freeswitch.o
Brian,
Thanks for your advice. I will like to join FS on the IRC and to further learn
from you and the FS team. What is the most effective way to set up an
experimental FS environment so I can quickly move up the learning cureve? What
Linux platform will you recommend? I also have access to Mac
Without one for me to test with I can only guess. I have tested TLS
on Polycom, Snom you might need to setup sslv3 instead of tls on your
profile for doing secure SIP. What are you trying https against?
/b
On Sep 12, 2008, at 10:01 AM, R. Eric Bennett wrote:
> folks,
>
> i'm wondering if
Hi,
Has anyone ever seen this error message ?
[CRIT] switch_core_sqldb.c:208 switch_core_sql_thread() SQL thread unable to
commit transaction, records lost!
Thanks !
PG
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folks,
i'm wondering if anyone here has actually managed to get SIP over TLS
working from a symbian s60 phone to FreeSWITCH. i've been trying for
some time with an e71 and while i've made some progress, progress !=
success.
in fact, i've progressed to the point where the failure is known t
Or should I calculate it with the help of mod_socket? I mean when
application received call pickup event, it calculates something like,
*new total time for call = (call pick up time + session start) + total time
for call*
and execute the sched_hangup with the new value.
On Fri, Sep 12, 2008 at
Have you ever thought about hanging out with us On IRC? #freeswitch
on irc.freenode.net it could be a lot of fun... quicker help... ;)
/b
On Sep 12, 2008, at 2:30 AM, Adeel Ansari wrote:
> It is working. It was a stupid trailing space.
> Thanks.
Brian West
sip:[EMAIL PROTECTED]
_
It is working. It was a stupid trailing space.
Thanks.
On Thu, Sep 11, 2008 at 8:39 PM, Adeel Ansari <[EMAIL PROTECTED]> wrote:
> No I was wrong, I suppose. Actually, I guess, these must be a trailing
> space in it. Will confirm tomorrow, hopefully. Thanks for your support.
>
>
> On Thu, Sep 11,
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