Seems like my dialplan was a bit problematic, it works now.
Thanks.
On Thu, 18 Dec 2008 15:19:22 +0100, Anthony Minessale
anthony.miness...@gmail.com wrote:
did you turn up your console log level high enough to see it? The default
level is INFO
Hello!
I was wondering if it would be possible to catch messages from dingaling.
I saw it can print out messages into console when a user types in a
message,
but it doesn't understand it. I would like to catch that and do something
(like initiate a call).
I know you have to call you program
On Fri, Dec 19, 2008 at 10:12:49AM +0100, kriko wrote:
I was wondering if it would be possible to catch messages from dingaling.
I saw it can print out messages into console when a user types in a
message,
but it doesn't understand it. I would like to catch that and do something
(like
On Thu, Dec 18, 2008 at 09:46:39PM -0800, Marc Orenberg wrote:
Thanks for the response Brian. I don't understand what bind_meta does, or
how it can help me. Is it something I can use from my Python script? I
searched for a description of it, but I was unable to find one. Could you
please
Hello,
I get more warnings yet:
[WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for
channel 1:1( its going to 1:31)
[WARNING] zap_isdn.c:803 process_event() channel 1:1 (1:2) (its going to
1:31 (1:16))
I dont know what to do? Can you help me?
I have a
sendmsg redirect to an ip-adress of one of our fs server works great.
thanks for your help.
dannis
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Hello guys,
I'm playing with fs fax capabilities following these guidelines:
http://wiki.freeswitch.org/wiki/Examples_faxlib.jm
I've compiled mod_fax
with make mod_fax-install
and that should have taken care also of spandsp.
When I issue make in scripts/socket2me I get this error:
That's seems the right this, thanks.
But the dingaling is only returning this events:
dingaling::login_success
dingaling::login_failure
dingaling::connected
Is it possible in any way to catch text messages?
On Fri, 19 Dec 2008 10:22:07 +0100, Jason White ja...@jasonjgw.net wrote:
On
Which dialect are you running and what is on the other end of the PRI?
-MC
On Fri, Dec 19, 2008 at 2:10 AM, fidibus83 fidibu...@aol.com wrote:
Hello,
I get more warnings yet:
[WARNING] mod_openzap.c:1405 on_clear_channel_signal() Unhandled type for
channel 1:1( it's going to 1:31)
It gives me the impression there is something wrong with your firewall
running on the box.
Mike
On Dec 19, 2008, at 3:03 AM, mszla...@aol.com wrote:
I find it strange that I can have to endpoints get audio went using
bypass media mode but the audio fails to go across endpoints if I
use
mod_fax replaces socket2me, you don't need it anymore.
Mike
On Dec 19, 2008, at 7:36 AM, Cavalera Claudio Luigi wrote:
Hello guys,
I'm playing with fs fax capabilities following these guidelines:
http://wiki.freeswitch.org/wiki/Examples_faxlib.jm
I've compiled mod_fax
with make
it's me again about mod fax... it is short before christmas and my
whish is, to get mod fax working quite reliable. is this possible
under optimal conditions?
all our tests lead by far to more failed faxes than received faxes. i
really like the fax feature and would like to see it beeing usable.
You don't know where the audio goes after that switch in the same room until
it gets to the guy
with the fax machine.
No it will not be improved by Christmas. Not a chance.
Yes it will probably be much more reliable once it can do T38.
Be happy with what you have for the holiday season.
On
hi anthony,
thanks a lot for the clear answer. that is something i can work with :-)
i also want to thank you for the great support you gave us within the
last months and the great freeswitch. our fs servers are up and
running and everything works great (only fax is not working).
have a nice
With the firewall ON or OFF the problem still remains.
I've tried 3 different set-ups in a dial plan extension.
1. With only action application=set data=proxy_media=true/ before
bridging.
2. With only action application=set data=bypass_media=true/ before
bridging.
3. Neither
On Thu, Dec 18, 2008 at 11:20:57AM -0600, Anthony Minessale wrote:
It seems to be related to 20ms vs 30ms ptime.
What are the 2 devices and what rev of FS are you on?
There was more code added in the last few weeks to smooth out this
occurrence.
This might not be the same issue, but as an
Riddle me this... what firmware are you running?
/b
On Dec 19, 2008, at 5:13 PM, Jason White wrote:
On a Snom 320 SIP phone, select G.722 as the first codec.
Call extension 3001 in the default context of the supplied FreeSWITCH
configuration, with nobody else calling into the conference,
On Fri, Dec 19, 2008 at 05:28:44PM -0600, Brian West wrote:
Riddle me this... what firmware are you running?
My apologies - 7.1.30.
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Please update to 7.1.33 or higher, also I need a pcap of your
situation email me a link where I can wget it if off list... I need
the rtp and sip traffic. You can do it from the phone or from
FreeSWITCH.
/b
On Dec 19, 2008, at 5:36 PM, Jason White wrote:
My apologies - 7.1.30.
Btw I have a 300, 320, 360, m3 and an 820 on the way now. (I don't
see this problem you're describing at all)
/b
On Dec 19, 2008, at 5:36 PM, Jason White wrote:
My apologies - 7.1.30.
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I just tested again, and I myself am now having trouble reproducing it. It
happened on multiple occasions yesterday evening, though.
It is also easier to reproduce with an actual connection to a remote
end-point, but that obviously complicates the situation with potential network
issues. I
On Fri, Dec 19, 2008 at 05:41:02PM -0600, Brian West wrote:
Please update to 7.1.33 or higher, also I need a pcap of your
situation email me a link where I can wget it if off list... I need
the rtp and sip traffic. You can do it from the phone or from
FreeSWITCH.
I've just upgraded
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