thanks allot, this was my mistake.
/Tzury
It tells you to edit conf/directory/default.xml not dialplan/
default.xml and put
param name=dial-string value={sip_secure_media=${regex($
{sofia_contact(${dialed_us...@${dialed_domain})}|
transport=tls)},presence_id=${dialed_us...@${dialed_domain}}$
Dear All,
How to set ${local_ip_v4} to 0.0.0.0 ? i want to set up
external profile for All IP
Best regards.
Dome C.
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I think by default it maps to all IPs unless you mention one here (by
replacing ${local_ip_v4} to some ip address).
Thank you.
On Thu, Jul 16, 2009 at 7:30 PM, freeswitch-users@lists.freeswitch.orgwrote:
Dear All,
How to set ${local_ip_v4} to 0.0.0.0 ? i want to set up
external
Hi,
I usually find it very useful when I can retrieve a list of the currents
calls along with durations. I noticed that the 'show channels' format
does not include the duration (or the answered timestamp - so that one
can extract it from there). So, I made a patch that includes the
answered
I'm ok with the idea as long as it's thoroughly tested.
If there is any more info you want to save from those events you should
consider it now while we are modifying it.
On Thu, Jul 16, 2009 at 9:02 AM, freeswitch-users@lists.freeswitch.orgwrote:
Hi,
I usually find it very useful when I can
Well,
that's very useful for us in order to have this info in our FS Operator
panel.
Best regards
Peter
freeswitch-users@lists.freeswitch.org schrieb:
Hi,
I usually find it very useful when I can retrieve a list of the
currents calls along with durations. I noticed that the 'show
channels'
Hello again,
I wanted to first say thanks to Brain for helping me fix my from
domain issue the other day. It helped quite a bit.
Now with more testing and talking with the vendor (please don't shoot
the messenger :) )
They want the caller id info in the from and the charge number/
On Jul 16, 2009, at 9:30 AM, Dale wrote:
They want the caller id info in the from and the charge number/
screening number in the P-Asserted-ID.
set the sip_h_P-Asserted-ID=contents of header
its just a variable you need to set now.
-Dale
___
Now that I come to think of it...
It would be useful if we had the timestamp (and epoch)
of the events PROGRESS and PROGRESS WITH MEDIA so that we can extract
the PDD (Post Dial Delay) which is a very useful statistic.
Adding the user's (for the incoming) and the gateway's (for the
outbound)
That's just right down screwing with the standards...
PAID is the caller id... This particular definition is from the RFCs and
3GPP docs for IMS which is why we have standardized P- headers...
Can your vendor not look at the P-Charging-Vector field?
Also, From when used with PAID is more like
On Jul 15, 2009, at 5:40 PM, freeswitch-users@lists.freeswitch.org
wrote:
Could you please just tell me where to set it??
The menu actions are defined in conf/autoload_configs/ivr.conf.xml
The audio played for the menus is defined in conf/lang/en/demo/demo-
ivr.xml
-MC
Please try
I wasn't planning on implementing it but I was just mentioning that if you
were going to do your patch,
consider if there is any other info to store while the patient is on the
operating table.
The line we should not cross is to store all the info in the table since,
really, you could be
That works perfectly.
Thanks,
-Dale
On Jul 16, 2009, at 10:37 AM, Brian West wrote:
On Jul 16, 2009, at 9:30 AM, Dale wrote:
They want the caller id info in the from and the charge number/
screening number in the P-Asserted-ID.
set the sip_h_P-Asserted-ID=contents of header
its just
I wonder if it would make sense to create a separate sub-command like
show channels stats or something. That way we could put all sorts of
nifty info there without breaking the existing command.
Thoughts?
-MC
Sent from my iPhone
On Jul 16, 2009, at 7:37 AM, Apostolos Pantsiopoulos
Hi Dale,
You can set the header to anything you like by including something along
the lines of
action application=set data=sip_h_P-Asserted-Identity=sip:
${caller_id_numb...@1.2.3.4 /
in your dialplan.
Cheers --
Dave
Hello again,
I wanted to first say thanks to Brain for helping me fix my
it doesn't really break anything to add more fields besides the ability to
read it but it's already fairly wide as it is.
Thats why we have show channels as xml
On Thu, Jul 16, 2009 at 9:50 AM, Michael S Collins m...@freeswitch.orgwrote:
I wonder if it would make sense to create a separate
Kinda wrong there!
Gotta use CDATA because it has and in the data you're setting.
And you'll wanna export it I suspect.
action application=export![CDATA[sip_h_P-Asserted-Identity=sip:${caller_id_numb...@1.2.3.4
]]/action
/b
On Jul 16, 2009, at 9:51 AM, David Knell wrote:
Hi Dale,
OK I 'll start implementing the progress timestamp field.
The only reason I mentioned the user/gateway id field is that FS admins
gain a lot by looking at a row of show channels result and be able to
see who is the caller and who is the callee.
Anthony Minessale wrote:
it doesn't really break
If you want to use , within a { } block you can escape it with \,
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
Am 16-Jul-09 um 11:38 AM schrieb Saeed Ahmed:
Hi,
its very useful feature for monitoring, I am doing it with
Hi,
its very useful feature for monitoring, I am doing it with XML RPC and
getting the result on webpage.
There is one issue which is nothing to do with that patch but in general: if
we are using absolute_codec_string variable and codes are like G729,G723
then this *comma* between codec ruin the
I have eth0 and tap0 (vpn) FS bind only eth0 ip
Dome C.
2009/7/16 freeswitch-users@lists.freeswitch.org:
I think by default it maps to all IPs unless you mention one here (by
replacing ${local_ip_v4} to some ip address).
Thank you.
On Thu, Jul 16, 2009 at 7:30 PM,
On a second thought... I am starting to agree with Michael.
Nobody really collects stats about PDD from the 'show channels'
command. This is usually done by using the cdrs.
So, forget about the second request. The patch I sent earlier covers
my needs (and I hope everybody else's too.)
Apostolos
Who knows. I have asked why they are backwards when it comes to the
RFC's.
-Dale
On Jul 16, 2009, at 10:41 AM, Ken Rice wrote:
That's just right down screwing with the standards...
PAID is the caller id... This particular definition is from the RFCs
and
3GPP docs for IMS which is why
Hey, you happen to have the RFC number and section they are violating?
I'll dig it up. But figured I'd ask since you all seem to have them
memorized by now. :)
-Dale
On Jul 16, 2009, at 10:41 AM, Ken Rice wrote:
That's just right down screwing with the standards...
PAID is the caller
PAI is defined @ http://www.ietf.org/rfc/rfc3325.txt
For what they are trying to do Sonus suggested P-Charge-Info (which I think
it is still in the draft stage) --
http://www.ietf.org/internet-drafts/draft-york-sipping-p-charge-info-06.txt
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right away by calling 877.742.CLUE. This year's event is going to be awesome
- you don't want to miss it!
Some news and notes:
We have two more sponsors for our event this year: Nokia and Twilio! We are
happy to welcome
Really? That line (+/- the IP address) came directly out of a working
dialplan. To be fair, the box is running a faintly prehistoric
FreeSWITCH - you crazy cats haven't been chewing on the tail of my
cherished mouse of backwards compatibility again, have you?!
What has been incorrect in this
Not sure how you were able to set the variable with and in it at
all thats not been possible cuz the XML parser will barf on it usually.
/b
On Jul 16, 2009, at 2:20 PM, David Knell wrote:
Really? That line (+/- the IP address) came directly out of a working
dialplan. To be fair, the box
Will there be video recordings availble online?
On 7/16/09, Michael Collins market...@cluecon.com wrote:
Only 19 more days until ClueCon! If you haven't registered then please do so
right away by calling 877.742.CLUE. This year's event is going to be awesome
- you don't want to miss it!
Some
Yes their will be a couple of weeks after cluecon!
/b
On Jul 16, 2009, at 2:36 PM, Jens Vegeby wrote:
Will there be video recordings availble online?
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Hmm, it seems to work on a copy of trunk from a few days ago. :) Both
in the dialplan and the gateway config.
But I could see how that might cause a problem with the xml.
-Dale
On Jul 16, 2009, at 3:27 PM, Brian West wrote:
Not sure how you were able to set the variable with and in it at
I think the parser is probably supposed to barf on consecutive chars but
it doesn't. I learned that the hard way when I left the closing tag off of
an action in the dp. Consider this dp snippet:
action application=log data=INFO Now you see me
action application=log data=INFO Now you don't/
hello
please cool anyone give me a Speech ASR script in LUA ?
only recognise the speech and put it into a variable
thanks!
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Great!
On 7/16/09, Brian West br...@freeswitch.org wrote:
Yes their will be a couple of weeks after cluecon!
/b
On Jul 16, 2009, at 2:36 PM, Jens Vegeby wrote:
Will there be video recordings availble online?
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Hey Dale RFC4694 defines how to pass JIP LRN and CIC (and a few other
things) as part of the tel URI... This has been adapted for use in SIP as
the user part of the invite and from URIs... Theres also RFC3325 for PAID
and if you check out the wikipedia page for IMS (IP Multimedia Subsystem)
there
Hi there,
I am looking at the FreeSWITCH.Demo.AppDemo class. Have it running nicely
from a soft phone - all is well.
However I am not seeing the hangupHook() method fired, when I hangup.
Debug log for an example call is at: http://pastebin.freeswitch.org/9744
Reminder of demo code is here:
The debug log has this:
65.2009-07-16 16:48:41.432200 [DEBUG] switch_cpp.cpp:1124 AppFunction is in
hangupCallback.
66.2009-07-16 16:48:41.432200 [WARNING] switch_cpp.cpp:1124 Thread will not be
aborted because Hangup was called from the Run thread.
The problem is the Demo doesn't have code to
I am getting an error in a lua script which I don't understand. Why is it
returning nil in the script yet something in the cli?
lua snippet:
user_data = session:getVariable('user_data 1...@192.168.10.29 var
callgroup');
freeswitch.console_log(INFO, UserData group .. user_data .. \n)
You need to make an API call, not get a variable.
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
Am 16-Jul-09 um 5:36 PM schrieb Lars Zeb:
I am getting an error in a lua script which I don’t understand. Why
is it returning
Hey, if there are any mod_managed users on this list, I'd love it if you were
able to let me know. I'd like to get feedback, positive or negative, on what
worked, what didn't, and how mod_managed can improve for you. Feel free to
write on list or directly to me: mgg at giagnocavo.net
Thanks!
hello
please cool anyone give me a Speech ASR script in LUA ?
only recognise the speech and put it into a variable
thanks!
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You can not do that with a single profile. Each profile is bound to only
one local IP, so if you need to bind to more than one you will have to
create a new profile and set the specific sip-ip/rtp-ip params for them.
Regards,
RAul
On Thu, 2009-07-16 at 20:30 +0700,
Hey, I am here :)
I am working with mod_managed on Windows 2003 and Windows Vista with
sucessfull. I noted on user list the issue with LoadFile on Loader.cs when a
assembly had reference to others assemblies, I change LoadFile by LoadFrom and
the load is made fine.
I use c# application
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Perfect - thank you very much!
On Thu, Jul 16, 2009 at 5:32 PM, Michael Giagnocavo m...@giagnocavo.netwrote:
The debug log has this:
65.2009-07-16 16:48:41.432200 [DEBUG] switch_cpp.cpp:1124 AppFunction is in
hangupCallback.
66.2009-07-16 16:48:41.432200 [WARNING] switch_cpp.cpp:1124
Mathieu,
Thanks for the reply. I'm very new with FreeSWITCH and not familiar with api
calls. I tried:
user_data = apiExecute(user_data, 1...@192.168.10.29 var
callgroup);
But got a similar error:
2009-07-16 17:01:47.212904 [ERR] mod_lua.cpp:182
http://wiki.freeswitch.org/wiki/Mod_lua#For_making_API_calls
Please READ that page. These details are there!
/b
On Jul 16, 2009, at 7:07 PM, Lars Zeb wrote:
Mathieu,
Thanks for the reply. I’m very new with FreeSWITCH and not familiar
with api calls. I tried:
user_data =
2009/7/17 Raul Fragoso r...@etellicom.com:
You can not do that with a single profile. Each profile is bound to only
one local IP, so if you need to bind to more than one you will have to
create a new profile and set the specific sip-ip/rtp-ip params for them.
Thanks.
Regards,
RAul
On
I tried to install mod_managed on ubuntu-9.04, mono framework version 2.0.
It gave me a lots of errors in Loader.cs, which seems to be SWIG related.
Since i am not a expert in SWIG so i disabled this module. This happend long
ago, i think FS svn revision 136xx.
Let me try to compile it from
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