-Specific_Dialstrings
If you're calling a locally registered user... you need to use
user/u...@domain which uses dial-string from the params on the user or
directory. Or sofia/profile/user%domain
/b
On Jun 10, 2009, at 9:54 AM, Max Bridgewater wrote:
Thanks,
the first variant doesn't work for me
clue?
On Wed, Jun 10, 2009 at 12:52 PM, Michael Jerris m...@jerris.com wrote:
http://wiki.freeswitch.org/wiki/Dialplan_XML
break=on-true ?
On Jun 10, 2009, at 12:18 PM, Max Bridgewater wrote:
Thanks Folks; I'm making progress. The following origination string does
make my non-registered
Any help our there?
I'm still trying to get this piece working. Essentially what i wan to do is,
when a call comes in (from registered devices as well as unregistered
devices), notify the my server socket. Somehow it's not working. The change
i made compared to the standard Freeswitch settings
, Jun 19, 2009 at 10:14 AM, Max Bridgewater
max.bridgewa...@gmail.com wrote:
Any help our there?
I'm still trying to get this piece working. Essentially what i wan to do
is, when a call comes in (from registered devices as well as unregistered
devices), notify the my server socket. Somehow it's
site?
-MC
On Fri, Jun 19, 2009 at 12:58 PM, Max Bridgewater
max.bridgewa...@gmail.com wrote:
Hi Mike,
It's pasted here: http://pastebin.ca/1466521
Thanks,
Max.
On Fri, Jun 19, 2009 at 11:43 AM, Michael Collins m...@freeswitch.orgwrote:
Can you turn on debugging (F8) and capture
the quickest way to handle this is to create a
copy of your mysocket.xml file and put it in conf/dialplan/default/ and be
done with it.
-MC
On Fri, Jun 19, 2009 at 1:19 PM, Max Bridgewater
max.bridgewa...@gmail.com wrote:
Mike,
Here we go: http://pastebin.freeswitch.org/9447. Doesn't look very
, Max Bridgewater
max.bridgewa...@gmail.comwrote:
I don't have my settings to try it right now. Still i have a question. If
it's the way you describe it, why wouldn't sofia/extenal/f...@bar solve the
problem? I think i even copied the extension both to the default directory.
But i will confirm
%7Borigination_caller_id_number=120003%7Dsofia/internal/242...@192.168.50.67socket(
192.168.50.67:1 full)
/b
On Jun 22, 2009, at 11:57 AM, Max Bridgewater wrote:
Hi Mike,
Unfortunately this doesn't seem to solve my problem. Here is my extension
again:
extension name=mysocket
condition
Hi Michael,
Using loopback solves my problem. Thanks a lot.
There is a strange thing i observed though. I need to paste my extension in
the default.xml file. Having them in the default directory isn't enough. Is
that normal?
Max.
___
/
/condition
/extension
/include
Max.
On Tue, Jun 23, 2009 at 10:44 AM, Michael S Collins m...@freeswitch.orgwrote:
On Jun 23, 2009, at 7:04 AM, Max Bridgewater
max.bridgewa...@gmail.com wrote:
Hi Michael,
Using loopback solves my problem. Thanks a lot
Hi,
I've got some news on this. When i move my extension to a different
directory (/usr/local/freeswitch/conf/dialplan/sockets/) and add an include
element at the very sample place where the default is included, things work
just as expected. That is, my default.xml now include following:
Hi,
I have a server that i log into using SSH. Then in my local SSH terminal, i
start Freeswitch with:
/usr/local/freeswitch/bin/freeswitch -nonat
Yet, when i close the terminal window, Freeswitch also dies. I was hoping
that the ampersand would make it run as a dameon process that would live
Cool!
Thanks.
On Mon, Jul 6, 2009 at 7:27 PM, Brian West br...@freeswitch.org wrote:
add a -nc in there too.
/usr/local/freeswitch/bin/freeswitch -nonat -nc
/b
On Jul 6, 2009, at 6:24 PM, Max Bridgewater wrote:
Hi,
I have a server that i log into using SSH. Then in my local SSH
Hi Guys,
I keep getting this message printed in red on my consolde; and whenevr i get
it, DTMF will stop being trasmitting in that session.
[ERR] switch_rtp.c:2013 Failed DTMF sanity check.
What does that mean and how can i prevent this from occurring? I'm using
the socket API to send DTMF
, Brian West br...@freeswitch.org wrote:
That usually means your device you're using is broken for sending
rfc2833 can you tell me what device are you using?
/b
On Jul 7, 2009, at 12:47 PM, Max Bridgewater wrote:
Hi Guys,
I keep getting this message printed in red on my consolde
Can you get me an RTP trace bet they might be doing it wrong... seems
to be common.
Hmm Sorry. Can i activate RTP traces in Freeswitch somehow or do i need to
run Something like Wireshark?
Pascal.
/b
On Jul 7, 2009, at 1:07 PM, Max Bridgewater wrote:
Actually I'm using Voxeo
Hi,
Say i originate a call to a mobile phone and the call fails. There are many
possible reasons: congestion, user busy, call rejected by user, etc. Is
there a way i can get the failure code from Javascript?
Thanks,
Max.
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Hi,
using javascript, i do originate the call this way:
Session s= new Session(originateStr);
From this point, is it possible to know what states the call is going
through? In a previous message it was suggested that
variable_originate_disposition would give me the response code. Now, how to
i
Hi get the following messages while running make on CentOS 5.3 (Of course
after bootstrap and configure).
os_Linux_x86_64.s: Assembler messages:
os_Linux_x86_64.s:46: Error: bad register name `%rdi)'
os_Linux_x86_64.s:61: Error: bad register name `%rdi)'
os_Linux_x86_64.s:75: Error: bad register
on the first build.
/b
On Aug 10, 2009, at 1:51 PM, Max Bridgewater wrote:
Hi get the following messages while running make on CentOS 5.3 (Of course
after bootstrap and configure).
os_Linux_x86_64.s: Assembler messages:
os_Linux_x86_64.s:46: Error: bad register name `%rdi)'
os_Linux_x86_64.s:61
and uname -a says?
Linux miriam 2.6.18-128.1.10.el5xen #1 SMP Thu May 7 11:07:18 EDT 2009
x86_64 x86_64 x86_64 GNU/Linux
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sure they know they have something broken.
/b
On Aug 10, 2009, at 2:31 PM, Max Bridgewater wrote:
and uname -a says?
Linux miriam 2.6.18-128.1.10.el5xen #1 SMP Thu May 7 11:07:18 EDT
2009 x86_64 x86_64 x86_64 GNU/Linux
10, 2009 at 4:01 PM, Michael Jerris m...@jerris.com wrote:
Is this a 32 bit default compiler? Try CFLAGS=-m64 on the configure line.
On Aug 10, 2009, at 12:49 PM, Max Bridgewater max.bridgewa...@gmail.com
wrote:
I have no idea what is wrong and therefore what to tell them. Without
telling
CFLAGS=-m64 on the configure line.
On Mon, Aug 10, 2009 at 8:06 PM, Brian West br...@freeswitch.org wrote:
What flags are you using?
/b
On Aug 10, 2009, at 7:01 PM, Max Bridgewater wrote:
I suppose it is a 32 bit default compiler. When i compile with the
suggested flag it fails right
Hi,
I'm trying to have an audio as ringback (WAV, 8khz, mono) when originating a
call. Unfortunately, it doesn't seem to work; the RINGING teletone is being
used instead of my audio. I have debug level enabled on Freeswitch but i
don't find anything suspicious.
is there anything specific i
Hi,
I'm trying to use les.net for termination. When i place a call from the
command line, i get the following error message: No Dialplan on answered
channel, changing state to HANGUP. What does it mean?
My dialing string is as simple as it could be: sofia/gateway/
did.voip.les.net/13028838864
Hi,
Whenever i stop freeswich, it creates a core dump. How can i disable that?
thanks,
Max.
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Hi,
Assuming an inbound call, I have trouble understanding what the supposed
difference between the following two set of instructions is:
session.execute(ring_ready);
session.set(set, ringback=/home/ring.wav);
and
session.preAnswer();
session.streamFile(/home/ring.wav)
In practice, however,
Guys,
This question might be a bit off-topic, so please bear with me as i could
hardly find a better place to solicit some help.
I suspect caller authentication can be challenging in the broader context of
VoIP/SIP. Now, if we restrict the set of calls to those coming from the GSM
network, is
Hi,
I was trying to build freeswitch with cepstral enabled. Here is the procdure
i am following:
http://wiki.freeswitch.org/wiki/Mod_cepstral#Can_I_use_a_16khz_.22desktop_voice.22.3F
Unfortunately, the build fails as show in the log message hereunder. In a
previous thread, Brian says that the
.
Will let ya know.
On Sun, Aug 30, 2009 at 1:58 PM, Brian West br...@freeswitch.org wrote:
You still have to have the voices installed... the SDK comes with each
voice for Linux ... for windows you have to have the SDK...
/b
On Aug 30, 2009, at 12:47 PM, Max Bridgewater wrote:
Hi,
I
Please bear with me. Here is the content of my /etc/ld.so.conf:
include ld.so.conf.d/*.conf
/opt/swift/lib
And when i do ldconfig -p | grep swift , i get the following:
libswift.so.5 (libc6) = /opt/swift/lib/libswift.so.5
libswift.so (libc6) = /opt/swift/lib/libswift.so
Nop, the 64bit cepstral doesn't work either:
Creating mod_cepstral.so...
/usr/bin/ld: skipping incompatible /opt/swift/lib/libswift.so when searching
for -lswift
/usr/bin/ld: cannot find -lswift
collect2: ld returned 1 exit status
gcc -w -I/home/installs/freeswitch/src/include
I thought i did that before trying. But i tried again and everything works.
Thanks for you unyielding patience.
Max.
On Sun, Aug 30, 2009 at 5:53 PM, Brian West br...@freeswitch.org wrote:
Make sure you rm -rf /opt/swift then reinstall
/b
On Aug 30, 2009, at 4:49 PM, Max Bridgewater wrote
Hey Guys,
Pleas help me answer the following question. Sorry for being a bit
off-topic. I have a GSM-VoiP modem with just 4 SIM Cards. Is there a known
mechanism that allows me to use this modem for more than 4 (albeit not at
the same time) GSM numbers? Let's say the phone numbers of the SIM
Thanks Dave. I'm essentially trying to build a service in one country where
there is no DID provider. If DIDs were available for such countries, that
would have done the job.
Using the two-numbers-per-SIM approach would help significantly, if I could
only figure out, how to detect the Called
Hi,
I just did a fresh checkout of Freeswitch from trunk but it seems
mod_gsmopen is not there. Am I doing something weird or it has been removed
from trunk. Please advise!
Thanks,
Max.
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