How do I read the contents of the SIP headers sdp in a dialplan?
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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 ${sip_h_X-Header}
The info app helps show all the variables.
/b
On Sep 17, 2008, at 2:11 AM, Jon Bruel wrote:
How do I read the contents of the SIP headers sdp in a
Hello Michael,
yes, mod_sofia is loaded. I would not be able to route calls to external
gateways defined under sofia.conf in the xml file if it was not loaded
Best regards Peter
Michael Jerris schrieb:
On Sep 16, 2008, at 4:02 PM, Peter P GMX wrote:
Hello,
as explained everything
Hello Raymond,
wow, what a number of requests I do not see.
I have the same modules enabled except dingaling and iax but for
Sep 16 15:10:31 lmdt fs_curl[3450]: [key_value] = 'console.conf'
Sep 16 15:10:31 lmdt fs_curl[3449]: [key_value] = 'conference.conf'
Sep 16 15:10:31 lmdt
Hi Peter,
Where do you load mod_xml_curl - it needs to be close to the top of
modules.xml.conf and
certainly before you load mod_sofia.
Cheers --
Dave
Hello Raymond,
wow, what a number of requests I do not see.
I have the same modules enabled except dingaling and iax but for
Sep 16
Hello Dave,
thank you. That was the solution. I loaded xml_curl at the end of the
configuration. Now I put it at the top of modules.conf and now the
missing requests are there.
Thanks to everybody for your help.
Best regards
Peter
David Knell schrieb:
Hi Peter,
Where do you load
Hello
I got the following setup:
Phone1 and FS behind NAT1.
Phone2 behind NAT2
Phone1 - FS - NAT1 - INTERNET - NAT2 - Phone2
Both phone1 and phone2 registers to FS. Phone1 on the internal profile
and Phone2 on a profile that is identical to internal, except that
external sip/rtp ips are set.
Hi all,
I bridge the call using
bridge(session, newsession);
After hangup i use the following line to get start time of the call:
starttime=session.getVariable(start_stamp) ;
This works fine in all case, except one,
When called party hangup, start_stamp becomes false.
is there any other method
with the following expression i cant dial any sipbroker number, eg:
*01118
^(^\*\d+)$
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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+)$
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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:
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 expression i
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, except one,
Hi,
I am using Freeswitch with Sangoma A102 and Openzap. I have configured the
extension in default.xml as
*default.xml*
extension name=Long Distance - wanpipe
condition field=destination_number expression=^0([0-9]+)$
action application=set data=dialed_ext=$1/
action
On Sep 17, 2008, at 7:24 AM, Gopal krishnan wrote:
Hi,
I am using Freeswitch with Sangoma A102 and Openzap. I have
configured the extension in default.xml as
default.xml
extension name=Long Distance - wanpipe
condition field=destination_number expression=^0([0-9]+)$
action
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
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!
___
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 throw to
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 am about to
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
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_DEBUG=9
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
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 instead
of
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
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 to
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 with
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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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
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 param name=pass-rfc2833 value=true/ 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
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
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
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 recognize
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 files,
Did you notice -
in your xml you have domain name=172.26.16.10
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
Hello,
I have done it the following way:
xml_curl.conf.xml:
configuration name=xml_curl.conf description=cURL XML Gateway
bindings
binding name=example
param name=gateway-url
value=http://192.168.0.35:3000/xml_curls/directory;
bindings=configuration|dialplan|directory/
/binding
/bindings
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:
configuration name="xml_curl.conf" description="cURL XML Gateway"
bindings
binding name="example"
param name="gateway-url"
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
document
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 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
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
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 IO
less tickin keeps you breathin :) It'd be nice though if you could just
use xml_rpc to tell FS:
/xml_update section tag tag_attr_name tag_attr_val...,/
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
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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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
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
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
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
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