Yes
Brian West-3 wrote:
>
> Are you using Javascript?
>
> /b
>
> On Sep 17, 2008, at 5:19 AM, msp wrote:
>
>> Hi all,
>>
>> I bridge the call using
>> bridge(session, newsession);
>>
>> After hangup i use the following line to get start time of the call:
>> starttime=session.getVariable("sta
Hi all,
I have enabled proxy media from dialplan.
After that, I can make calls same as it done before without enabling proxy
media.
So, how can i test that my calls are in proxy media mode after enabling
"proxy-media" mode ?
Thanks,
MShehzad
___
Freesw
I give up :) You're right.
I've just started using FS and after reading so many stories about how
other products are not performing under stress I'm trying to think of
what else can slow things down... In any case, so far I'm impressed
with it!
Anthony Minessale wrote:
> if you say inhale the
if you say inhale the xml into memory and the sever goes haywire and sends
you 2 gigs out output you are in for a treat.
if you can get enough call volume on one box where the disk i/o of xml_curl
even shows up on the map in relation to all the rtp etc, we've won.
On Wed, Sep 17, 2008 at 6:00 PM,
Their are plans to do so for the static sound files and you can get
the spanish voice from Cepstral already.
And next time you email the list please DO NOT hijack a thread.
Please start a new email and input the address yourself. By pressing
reply... changing the subject and deleting the b
Greetings to FreeSWITCH team for their great job.
My query is whether there voices in Spanish for FreeSWITCH.
Thanks for your support.
Greetings
Fredy Gonzales P.
Lima - Peru
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less tickin keeps you breathin :) It'd be nice though if you could just
use xml_rpc to tell FS:
/xml_update...,/
similar to xml_locate
Cristian
Brian West wrote:
> If you're that concerned with it.. move the tmp to a ramdisk ;) I
> thought modern hard drives could take a lickin and keep
If you're that concerned with it.. move the tmp to a ramdisk ;) I
thought modern hard drives could take a lickin and keep on tickin
/b
On Sep 17, 2008, at 5:51 PM, Cristian Talle wrote:
> Oh, they are but it's still HDD... I wouldn't like to see the server
> die
> because of too much disk I
Oh, they are but it's still HDD... I wouldn't like to see the server die
because of too much disk IO
I'm trying to figure out what's the most efficient way to handle changes
in user profiles (and possibly dialplan, etc...) if order handle
thousands of users per server.
Cristian
Brian West wrot
Are those temp files not going away?
/b
On Sep 17, 2008, at 5:43 PM, Cristian Talle wrote:
> Just wondering...
>
> I noticed that with xml_curl temp files are being created for each
> request - this doesn't really help for higher request volumes.
> Do you know of any way of updating the FS xml t
Just wondering...
I noticed that with xml_curl temp files are being created for each
request - this doesn't really help for higher request volumes.
Do you know of any way of updating the FS xml tree (let's say the
directory node for one domain) without curl?
Cristian
Peter P GMX wrote:
> Just
Its 5.1.4
/b
On Sep 17, 2008, at 5:03 PM, Robert Clayton wrote:
> All,
>
> Is Lua in FreeSwitch derived from 4.0, 5.0, 5.1, etc.
>
> I am hoping to implement LuaCOM and it differs with versions.
>
> Bob
>
> ___
> Freeswitch-users mailing list
> Freeswi
All,
Is Lua in FreeSwitch derived from 4.0, 5.0, 5.1, etc.
I am hoping to implement LuaCOM and it differs with versions.
Bob
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I have found the problem - the user id="" in my document did not
match the one sent by the SIP client... duh :) I had an extra character
somewhere in the middle of the id attribute. It works nicely
now. (for now I'm using only directory in bindings)
Thank you again for your prompt reply!
Just to mention:
I figured out that, if you do an xml_curl request it tries to reply
based on the dynamic XML answer. If the answer is wrong if takes the
content of the config files.
This may be the reason why your standard numbers from the config files
1000, 1001, ... work and carl doesn't.
Be
According to http://wiki.freeswitch.org/wiki/Mod_xml_curl I'v tried to
set up gateways dynamically, but I do net get it to work:
I always get : "Invalid profile"
My assumptions for a right xml answer back to FS are as follows, but I
think at least one of it is false:
1. ) I start with
2
Thank you veeery much, I'll give it a try!
Best,
Cristian
Peter P GMX wrote:
Hello,
I have done it the following way:
xml_curl.conf.xml:
"http://192.168.0.35:3000/xml_curls/directory"
bindings="configuration|dialplan|directory"/>
My Ruby on Rails server is listening on Port 3000.
Hello,
I have done it the following way:
xml_curl.conf.xml:
http://192.168.0.35:3000/xml_curls/directory";
bindings="configuration|dialplan|directory"/>
My Ruby on Rails server is listening on Port 3000. But https works also
(but very very slow).
If a phone (e.g. 1002) tries to register
Did you notice -
in your xml you have
but you're looking for
carl at 172.16.26.10
the 16 and 26 are reversed in your xml. I think this causes your problem!
Tony
On Wed, Sep 17, 2008 at 4:58 PM, Cristian Talle <[EMAIL PROTECTED]> wrote:
> Hi Carl,
>
> I am experiencing a similar probl
Hi Carl,
I am experiencing a similar problem, have you found any solution so far?
Thank you,
Cristian Talle
> I wonder if anybody could provide a complete set of configuration files for
> a working xml_curl user directory lookup.
>
> I have been trying using the default set of configuration file
Make sure you're using the latest mod_pocketsphinx, and reinstall
speechtools and ps_pizza from svn. There was a time when it did this.
/b
On Sep 17, 2008, at 3:37 PM, Greg Thoen wrote:
> Hi, I have the pizza demo running on my test box, but the debug
> shows that while it seems to recogniz
Hi, I have the pizza demo running on my test box, but the debug shows
that while it seems to recognize the words, the score is always zero
so it never progresses. Any ideas?
2008-09-17 13:51:25 [DEBUG] mod_pocketsphinx.c:389
pocketsphinx_asr_get_results() Recognized: LARGE, Score: 0
2008-0
Hi,
I'm new to FS. Can anyone shed some more light or point me to a place to
read on how one can use mod_xml_curl for dynamic directory lookup?
My scenario is: I define a new contact somewhere outside of FS, I have a
SIP client attempting to register with FS using the new contact info ->
that in
On Fri, Jul 25, 2008 at 8:51 PM, UV <[EMAIL PROTECTED]> wrote:
> Yes I did, but you might not even need that.
> Try adding in your external SIP
> profile and see if it solves the problem.
>
I am still trying to get the DTMF 100%, I added the value but get this
message in the debug log:
[WARNING]
The lua modules are not part of the FreeSWITCH build system. They
have their own stand alone build system.
Mike
On Sep 17, 2008, at 3:41 PM, Robert Clayton wrote:
> All,
>
> What would be the process for adding LuaCOM (a windows COM library
> used by Lua) to the FreeSwitch build?
>
> Bob
__
All,
What would be the process for adding LuaCOM (a windows COM library
used by Lua) to the FreeSwitch build?
Bob
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UNSU
don't forget the bounty amount willing to pay for us to implement something
to make it work.
On Wed, Sep 17, 2008 at 12:39 PM, Michael Jerris <[EMAIL PROTECTED]> wrote:
> Can you open up a bug on http://jira.freeswitch.org with full traces both
> of freeswitch and how broadworks does it, along w
There should be no To tag with the initial INVITE. The tag is added by the
callee per RFC3261.
On Wednesday 17 September 2008, Jon Bruel wrote:
> Mike, in my test setup, no tags are added to the To-header by the
> FreeSWITCH. The reason why I want to control it? Well, many phones uses
> the tag t
Can you open up a bug on http://jira.freeswitch.org with full traces
both of freeswitch and how broadworks does it, along with if possible
the language about to tags from the rfc. There are a bunch of interop
qwirks related to those tags and when they should or shouldn't be in
there so we
Awesome - You Rock!
I shoulda looked at the wiki :)
On Wed, Sep 17, 2008 at 9:19 AM, UV <[EMAIL PROTECTED]> wrote:
> I believe what you're looking for is here:
> http://wiki.freeswitch.org/wiki/Sofia#Debugging_SOFIA_SIP
>
> In windows, you need to set those environment variables using "set" inst
Mike, in my test setup, no tags are added to the To-header by the
FreeSWITCH. The reason why I want to control it? Well, many phones uses
the tag to relate an existing call to a NOTIFY message (send after the
INVITE has been sent) with the same headers (and the same to-tag) as the
INVITE. I'm not a
I believe what you're looking for is here:
http://wiki.freeswitch.org/wiki/Sofia#Debugging_SOFIA_SIP
In windows, you need to set those environment variables using "set" instead
of "export". Like this:
set SOFIA_DEBUG=9
set NUA_DEBUG=9
set SOA_DEBUG=9
set NEA_DEBUG=9
set IPTSEC_DEBUG=9
set NTA_DEBU
Hello
Can anybody explain how to activate the voice mail ?
When calling one of the extensions it times out and send a busy signal.
2008-09-17 08:55:45 [INFO] mod_dptools.c:1789 audio_bridge_function()
Originate Failed. Cause: NO_ANSWER
thanks
Jair Santos
Good point. Is there anything inside Freeswitch that can provide more
context?
On Sep 17, 2008, at 8:48 AM, David Knell <[EMAIL PROTECTED]> wrote:
> Hi Christian,
>
> I'd download Wireshark if I were you - lets you see exactly what's
> going
> on on the
> wire.
>
> Cheers --
>
> Dave
>
>> I
Hi Christian,
I'd download Wireshark if I were you - lets you see exactly what's going
on on the
wire.
Cheers --
Dave
> I am about to embark on a sip debugging mission (did not coming in
> correctly).
>
> I am on windows. Other than setting "sofia loglevel 9" what other
> switches can I th
I am about to embark on a sip debugging mission (did not coming in
correctly).
I am on windows. Other than setting "sofia loglevel 9" what other
switches can I throw to get a butload of debut info?
I can hop over to ubuntu if need be.
Thanks!
__
if you mean tag= param on the to header, there should be one added
automatically as defined in rfc3261. To send the notify might be a
bit trickier, I think we added a way to use an event to send a notify,
are you trying to control as iff the user hit the hold button on their
phone? In res
On Sep 17, 2008, at 7:24 AM, Gopal krishnan wrote:
Hi,
I am using Freeswitch with Sangoma A102 and Openzap. I have
configured the extension in default.xml as
default.xml
This regular expression won't do what you think it will.
Based on what you have don
Hi,
I am using Freeswitch with Sangoma A102 and Openzap. I have configured the
extension in default.xml as
*default.xml*
*
openzap.conf*
[span wanpipe]
trunk_type => e1
b-channel => 1:1-15
d-channel=> 1:16
b-channel => 1:17-31
*openzap.conf.xml*
Are you using Javascript?
/b
On Sep 17, 2008, at 5:19 AM, msp wrote:
> Hi all,
>
> I bridge the call using
> bridge(session, newsession);
>
> After hangup i use the following line to get start time of the call:
> starttime=session.getVariable("start_stamp") ;
>
> This works fine in all case, exc
removing the second ^ i can dial the screeming monkeys numbe which is
*266300 but not the test sipbroker number which is *01118
Ivan C Myrvold wrote:
>
> Remove the second ^ and see if that helps.
>
> Ivan
>
> Den 17. sep.. 2008 kl. 12:52 skrev xbipin:
>
>>
>> with the following expres
Well, let me be more precise: In general, I want to be able to modify
the SIP headers to my liking. I would like to add a tag to the To-header
sent to the B-phone. This tag information can be used at a later stage
to put the phone on/off hold from the switch by sending a NOTIFY with an
header: Even
Remove the second ^ and see if that helps.
Ivan
Den 17. sep.. 2008 kl. 12:52 skrev xbipin:
>
> with the following expression i cant dial any sipbroker number, eg:
> *01118
>
> "^(^\*\d+)$"
> --
> View this message in context:
> http://www.nabble.com/regexp-help-tp19529525p19529525.html
> S
with the following expression i cant dial any sipbroker number, eg:
*01118
"^(^\*\d+)$"
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Sent from the Freeswitch-users mailing list archive at Nabble.com.
__
Hi all,
I bridge the call using
bridge(session, newsession);
After hangup i use the following line to get start time of the call:
starttime=session.getVariable("start_stamp") ;
This works fine in all case, except one,
When called party hangup, "start_stamp" becomes false.
is there any other met
Hello
I got the following setup:
Phone1 and FS behind NAT1.
Phone2 behind NAT2
Phone1 -> FS -> NAT1 -> INTERNET <- NAT2 <- Phone2
Both phone1 and phone2 registers to FS. Phone1 on the internal profile
and Phone2 on a profile that is identical to internal, except that
external sip/rtp ips are s
Hello Dave,
thank you. That was the solution. I loaded xml_curl at the end of the
configuration. Now I put it at the top of modules.conf and now the
missing requests are there.
Thanks to everybody for your help.
Best regards
Peter
David Knell schrieb:
> Hi Peter,
>
> Where do you load mod_xml
Hi Peter,
Where do you load mod_xml_curl - it needs to be close to the top of
modules.xml.conf and
certainly before you load mod_sofia.
Cheers --
Dave
> Hello Raymond,
>
> wow, what a number of requests I do not see.
>
> I have the same modules enabled except dingaling and iax but for
> Sep
Hello Raymond,
wow, what a number of requests I do not see.
I have the same modules enabled except dingaling and iax but for
Sep 16 15:10:31 lmdt fs_curl[3450]: [key_value] => 'console.conf'
Sep 16 15:10:31 lmdt fs_curl[3449]: [key_value] => 'conference.conf'
Sep 16 15:10:31 lmdt f
Hello Michael,
yes, mod_sofia is loaded. I would not be able to route calls to external
gateways defined under sofia.conf in the xml file if it was not loaded
Best regards Peter
Michael Jerris schrieb:
> On Sep 16, 2008, at 4:02 PM, Peter P GMX wrote:
>
>
>> Hello,
>>
>> as explained everyth
Do you want the SDP or the SIP Headers? Your question is ambiguous!
SDP should be in ${switch_r_sdp}
X-Header should be in ${sip_h_X-Header}
The "info" app helps show all the variables.
/b
On Sep 17, 2008, at 2:11 AM, Jon Bruel wrote:
> How do I read the contents of the SIP headers sdp in a
How do I read the contents of the SIP headers sdp in a dialplan?
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