I'm running 1.01
-N
On Nov 9, 2008, at 3:10 PM, Brian West wrote:
That tells me you're running older code... can you tell me what rev
you're on?
/b
On Nov 9, 2008, at 5:06 PM, Noah Silverman wrote:
It seems to be happening when calls get dropped. When speaking to
one
of my users
It seems to be happening when calls get dropped. When speaking to one
of my users, calls get dropped after 2-4 minutes.
It seems like I get a 408 error followed by a bad frame error.
It only happens with this particular user. Works fine for everyone
else. Could this be a function of a
Hi,
I keep seeing the following error when watching the debug log in FS.
Can anyone give me a hint as to what this might be??
sofia.c:197 sofia_event_callback() event [nua_r_options] status [408]
[Request Timeout] session: n/a
Thanks,
-N
___
Hello,
I have a configuration (dialplan?) question.
One of my users used the call forward button on his sip phone. The
phone is directly registered to FS.
The forwarded calls fail. It looks like the phone is sending a
redirect message to FS, but then the call is then not getting routed
, Noah Silverman wrote:
Hello,
I have a configuration (dialplan?) question.
One of my users used the call forward button on his sip phone. The
phone is directly registered to FS.
The forwarded calls fail. It looks like the phone is sending a
redirect message to FS, but then the call
broken
clients... thats how we got into this SIP mess in the first place.
/b
On Oct 30, 2008, at 5:03 PM, Noah Silverman wrote:
Ok,
But don't I need the aggressive nat detection since most of my
clients
will be behind nat??
-N
___
Freeswitch
into this SIP mess in the first place.
/b
On Oct 30, 2008, at 5:03 PM, Noah Silverman wrote:
Ok,
But don't I need the aggressive nat detection since most of my
clients
will be behind nat??
-N
___
Freeswitch-users mailing list
Freeswitch
Hi,
Weird situation.
I am now running my office phone through freeswitch. It works great!
Most of the companies that I call have an IVR that requires a
keypress: for sales press 1, for support press 2, etc.
With most of these calls, my key presses pass the DTMF through
perfectly.
The odd
Hi,
I have a question about setting up speech in FS...
I'm far away from the default config, so I just want to add the
MINIMUM amount necessary to my config to enable speech.
My goal is to have some basic functions like You have 3 dollars and
27 cents left.
(Do I need to build the cepstral
Nice,
Do I have to build and/or install any additional modules?
It seems as if the demo relies on some macros that in turn rely on
specific language files. is there any more detail about working with
them.
For example, I see sound files in /usr/local/freeswitch/sounds/en/us/
Hi,
I've been testing some xml_rpc scripts to make calls. (For a click
to call application I want to write.)
I'm experiencing some strange behavior in regards to setting the
caller id.
If I DON't pass a caller id with the originate command, the calls
works perfectly and the caller id
Hi,
I'm writing a script to connect one of my SIP phones with an external
number via a web interface. (Like click to call.) Right now, I'm
using xml_rpc and after some help from the guys on IRC, I've got it
working nicely.
When I execute the script, first my local SIP phone rings. I
Hi,
I managed to write a really short perl script to insert cdr records
into mysql. I wanted to share it with the group...
Just call this from cron at whatever frequency you want to update your
records.
DISCLAIMER: THIS IS A QUICK AND DIRTY SCRIPT. IT HAS NO BUILT IN
ERROR CHECKING
Hi,
I'm looking at developing a dynamic dialplan for FS.
I'm curious as if the group has a suggestion as to which method would
be better, and why...
1) Use mod_xml_curl to fetch the dialplan from a script
2) use mod_rpc to fetch the dialplan from a script.
At first glance, it looks like
the directory don't use the ACL...
If you don't want FS to respond to SIP from unknown IP Addresses
that's a
more appropriate job for your firewall software (iptables?)
From: Noah Silverman [EMAIL PROTECTED]
Reply-To: freeswitch-users@lists.freeswitch.org
Date: Tue, 30 Sep 2008 00:41:17 -0700
Hello,
I'm having trouble getting the accountcode variable to show up in the
CDR.
Also, the effective_caller_id_number isn't getting passed through to
the actual call. (Separate but possible related problem)
Calls are working, the CDR looks great, I just don't get these two
variables..
]/
On Thu, Sep 25, 2008 at 1:40 PM, Noah Silverman
[EMAIL PROTECTED] wrote:
Hello,
I'm having trouble getting the accountcode variable to show up in the
CDR.
Also, the effective_caller_id_number isn't getting passed through to
the actual call. (Separate but possible related problem)
Calls
I thought I was.
If I do a sofia status profile internal I see myself registered
there...
How else can I be sure?
-N
On Sep 25, 2008, at 12:51 PM, Brian West wrote:
Yes that is if you're really authenticating ;)
/b
On Sep 25, 2008, at 12:46 PM, Noah Silverman wrote:
OK,
That makes
${domain} is the
domain that user lives in:
action application=set_user data=[EMAIL PROTECTED]/
On Thu, Sep 25, 2008 at 1:40 PM, Noah Silverman
[EMAIL PROTECTED] wrote:
Hello,
I'm having trouble getting the accountcode variable to show up in the
CDR.
Also, the effective_caller_id_number
.
So double check the profile for auth-calls and accept-blind-reg.
/b
On Sep 25, 2008, at 1:07 PM, Noah Silverman wrote:
I thought I was.
If I do a sofia status profile internal I see myself registered
there...
How else can I be sure?
-N
On Sep 22, 2008, at 7:00 PM, Brian West wrote:
What was the response and where did it go? I need to see the full
sip trace along with console output.
TPORT_LOG=1 ./freeswitch
/b
On Sep 22, 2008, at 4:01 PM, Noah Silverman wrote:
U 222.222.222.222:1024 - 111.111.111.111:5060
Hi,
I have a VERY simple FS setup.
1) Calls come in and go out via Vitelity. (I just route to their IP
address, don't even need a gateway.)
2) I have users registered remotely with anything from a single SIP
phone to an asterisk box.
3) Basically, I am a pass through of calls to and from
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