Hello,
it works now. I'm not really sure what it was, but I know what I did in
what order:
1. update gcc to gcc version 4.2.4 (Ubuntu 4.2.4-1ubuntu3)
2. configure
3. make sure
4. make install
5 Tested it: FS still crash
6 did a gdb backtrace, last function call was mpg123_delete ...
7. delete
Hi all,
I'm using originate to initiate calls, and streamFile to play audio
files on the answered sessions. All the logic was encapsulated in a
Javascript file.
The problem with this setup is that origination_caller_id_number doesn't
work from inside the JS file (when calling session.originate).
On Fri, 2009-02-13 at 13:33 +0200, Alexandru Nedelcu wrote:
The problem with this setup is that origination_caller_id_number doesn't
work from inside the JS file (when calling session.originate).
I just discovered something interesting.
When originating the call like this ...
session = new
Use this method in js
var session = new
Session('{absolute_codec_string=PCMA,accountcode=54321,ignore_early_medi
a=true,origination_caller_id_number=4071122,originate_timeout=25}sof
ia/gateway/myprovider/87304071122);
-Original Message-
From:
Hello!
I'm trying to make such scheme:
--- FS_A -- FS_B -- record
Incoming calls to FS_A are redirected to FS_B with the help of
The first way is deprecated and will be removed.
The 2nd way is the correct way.
On Fri, Feb 13, 2009 at 5:48 AM, Alexandru Nedelcu a...@sinapticode.rowrote:
On Fri, 2009-02-13 at 13:33 +0200, Alexandru Nedelcu wrote:
The problem with this setup is that origination_caller_id_number doesn't
1) session.originate is depricated.
2) the first arg to session.originate is *another* session (not the same
one) *or* undefined.
session.originate(undefined, dial string);
session.originate(a_leg_session, dial string);
session.originate(session, dial string) is asking the session to
I think this page (external) is the source
http://alexn.org/docs/dialer.html
Regards,
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
Anthony Minessale
Sent: 13 February 2009 14:06
I wrote that document ... I can't remember from where I got the idea
that you should specify the a-leg as being the same session.
That document is a draft, but it got indexed by Google unfortunately :(
On Fri, 2009-02-13 at 14:12 +, Nik Middleton wrote:
I think this page (external) is the
Hi all,
I've created a simple framework in Ruby that you can use to talk to
Freeswitch via even socket outbound. It won't suite your needs
perfectly if you are doing anything non-trivial, but it might be a
nice starting point.
Check it out at http://github.com/jankubr/freec
Cheers,
Jan Kubr
My mistake, they do seem to be microsecs.
But still I cannot correlate times from the A-leg with the B-legs.
I have included below the xml_cdr files generated for the test call.
The test call was made using three registered extensions. Basically, Ext 201
calls ext 203 and they talk, then 203
I'm assuming that you are saying these are 2 boxes if the protocol is a
sip you can append a sip header ... _sip_h_X- This should be available
as a channel variable on FS A.
SDR
On Fri, Feb 13, 2009 at 6:10 AM, Evgeniy Zolotov zolo...@altron.ua wrote:
Hello!
I'm trying to
Btw ... I fixed the document.
Sorry about that guys, I'm a rookie and I thought other people would
find my setup useful.
Can you guys read it and tell me if it contains other mistakes? My
intention was to publish it on the wiki once it was ready, but I
temporarily moved on to another project.
Hi Anthony,
Excuse me if I'm wrong but inbound-late-negotiation must be used proxy_media
as I see in documentation. I don't want to proxy media because of some
issues with MOH or 3-way conferencing. Also I want to exclude media codecs
that are supported only in pass-trough mode. Let mi give you
Hi,
yes, I'm using gateway, but it ignores softswitch side challenges for
BYE messages coming from FS.
My dialplan:
extension name=outgoing-ssw-ddi
condition field=destination_number
expression=^0([0-9]+|^940[0-9]+) break=on-false
action
I made it an error now to do it this way which should clear things up.
Doing it that way probably led to instability in js.
On Fri, Feb 13, 2009 at 8:25 AM, Alexandru Nedelcu a...@sinapticode.rowrote:
I wrote that document ... I can't remember from where I got the idea
that you should specify
Thanks
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Michael
Collins
Sent: Thursday, February 12, 2009 9:02 PM
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] FS + Call
Turn up debug and look harder are you sure it does not say no matching
gateway when it gets the challenge to bye?
On Fri, Feb 13, 2009 at 9:00 AM, Helmut Kuper helmut.ku...@ewetel.dewrote:
Hi,
yes, I'm using gateway, but it ignores softswitch side challenges for
BYE messages coming from
yes you are wrong.
inbound-late-negotiation setting delays the codec negotiation until the
instant audio is needed.
It is not tied to inbound-proxy-media.
This allows the call to come into the dialplan before any codec negotiation
is done giving you a chance to look at the SDP before the
each b leg call on the a leg shows up in a callflow tag
At the bottom of
http://pastebin.freeswitch.org/7206
times
created_time1233942283835696/created_time
profile_created_time1233942283835696/profile_created_time
progress_time1233942283999716/progress_time
Hi Anthony,
yes you are right, sorry.
2009-02-13 16:56:20 [ERR] sofia_reg.c:1358
sofia_reg_handle_sip_r_challenge() No Matching gateway found
works now :) thx
regards
helmut
On 13.02.2009 16:40, Anthony Minessale wrote:
Turn up debug and look harder are you sure it does not say no
matching
Hi Anthony,
I'm not sure that you understood the problem. As it shown bellow the offered
codec in leg B contains only one codec (first matched in codec preference
list for this profile). Is there way to offer in leg B not only first codec
but all codecs that exists in INVITE in leg A that matches
On Fri, Feb 13, 2009 at 6:27 AM, Jan Kubr jan.k...@gmail.com wrote:
Hi all,
I've created a simple framework in Ruby that you can use to talk to
Freeswitch via even socket outbound. It won't suite your needs
perfectly if you are doing anything non-trivial, but it might be a
nice starting
On Fri, Feb 13, 2009 at 6:53 AM, Alexandru Nedelcu a...@sinapticode.ro wrote:
Btw ... I fixed the document.
Sorry about that guys, I'm a rookie and I thought other people would
find my setup useful.
Can you guys read it and tell me if it contains other mistakes? My
intention was to publish
As i have already answered, no, it does not do what you want automaticly,
the only way to influence codec negotiation is the way i have described.
parsing the sdp string allows you to set absolute_codec_string going both
ways.
if you set it before you answer the channel with late negotiation
This is perhaps the 4th time i have seen someone do this, can you
point out where this is incorrectly documented?
FYI, I've updated this html file for the orig author and sent it to
him for review.
-MC
___
Freeswitch-users mailing list
Hi Antony,
Can you tell me why you do codec negotiation like that. I'm just curious. If
you do not have time do not reply me.
Thanks a lot for your help.
2009/2/13 Anthony Minessale anthony.miness...@gmail.com
As i have already answered, no, it does not do what you want automaticly,
the only
Hello FreeSWITCHers,
mod_skypiax is available for testing, feature requests, bug hunting.
I would like to ask the help of you all to make Skypiax robust and
feature full on FreeSWITCH, and particularly of Massimo Cetra (CtRiX
on IRC), that has developed mod_airpe (another Skype endpoint).
I've
I updated the version of the speex library we use in tree last night
and it may cause some build issues for those with current working
copies. To fix this issue you can type make speex-reconf
MIke
___
Freeswitch-users mailing list
Do it like what?
you are using FS as a b2bua, the default behavior is to make a list of
codecs that are parsed on every inbound call as soon as the invite is
received.
if you bridge that inbound leg to an outbound leg it will again use that
same list with the one used by the inbound leg as the
Yay for the new speex with good Acoustic Echo Cancellation.
I'll put it to work when I'll port Celliax, the GSM endpoint, for
cancelling the sidetone that certain interfaces give back. :-)
Thanks MikeJ !
Sincerely,
Giovanni Maruzzelli
=
www.celliax.org
FYI,
There are a few new items on the main page: www.freeswitch.org, just
in case you haven't been there lately. :)
-MC
___
Freeswitch-users mailing list
Freeswitch-users@lists.freeswitch.org
Can't figure this one out.
I've enabled a hang-up hook in js to do some cleanup.
I've followed the example on the wiki, but it would appear it's never
called.
http://wiki.freeswitch.org/wiki/Example_Hangup_hook
Is the code in error?
Regards
http://wiki.freeswitch.org/wiki/Example_Hangup_hook
Is the code in error?
It might just be. I think you are better off using api_hangup_hook.
What are you trying to do on hangup? The api_hangup_hook lets you call
any API, including running a script. Here's an example that we played
with
groan Guess I'll have to dust off KR. Having made the mindset leap
to c++ I find C very procedural. Still would be like old times. Code
I've looked at so far is very neat, but boy is there a lack of in-line
comments. Haven't looked at the main source yet though. I always used
to work on 3
I'm trying to capture the hang-up reason and write it to the db (Was it
busy etc). I also close the db in that function. That way I know I
don't have any open connections. This is in JavaScript BTW
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
modules can be c++ too.
See mod_opal , mod_python, mod_java, mod_soundtouch, mod_managed and
mod_perl
all use switch_cpp.cpp a wrapper used to bridge into scripting langs.
On Fri, Feb 13, 2009 at 5:04 PM, Nik Middleton
nik.middle...@noblesolutions.co.uk wrote:
groan Guess I'll have to
Nik Middleton nik.middle...@noblesolutions.co.uk wrote:
Code
I've looked at so far is very neat, but boy is there a lack of in-line
comments. Haven't looked at the main source yet though. I always used
to work on 3 lines of comments to 1 major line of code. Call me
pedantic, but it aids
That would assume that the underlying code is perfect, which it probably
isn't. Not knocking the efforts, but in my view, you can't have too
much in line documentation. I hope to make a contribution shortly.
Right now I'm updating the WIKI where appropriate. Top level examples
should work with a
Right now I'm updating the WIKI where appropriate. Top level examples
should work with a cut and paste, if they don't you're going to alienate
new entrants.
This is a valid point. I will be happy to help with the wiki since
documentation is kind of my bailiwick. My challenge is just being in a
Hello,
I am using a SNOM 320 and Windows Mobile 6 VoIP softphone on two separate
extensions. When dialing from the SNOM to the WM6 device I get ringback on
the SNOM but when calling the SNOM from the WM6 device I don't get ringback
though the call does complete and I get voice after the
Would need a sip trace to know. TPORT_LOG=1 ./freeswitch
/b
On Feb 13, 2009, at 8:17 PM, Maxim Karp wrote:
Hello,
I am using a SNOM 320 and Windows Mobile 6 VoIP softphone on two
separate extensions. When dialing from the SNOM to the WM6 device I
get ringback on the SNOM but when
Hi,
I tried to configure opensips as sip proxy and sip registrars and
freeswitch as B2BUA. Everything works until I start to connect sip
clients that are behind ADSL.
Both freeswitch and opensips are on public IP and I am using external
profile as well.
Does anyone have experience in setting
You have let the names of the profiles confuse you. Chances are
you're trying to hair pin the calls out and back into the same nat.
That usually doesn't work. You will need to give me more details
about your setup.
/b
On Feb 13, 2009, at 9:41 PM, Woody Dickson wrote:
Hi,
I tried to
I'm trying to use custom events for a conference call in a Python script. I
set-up the events in the conference.conf.xml file, and I send bgapi event
plain CUSTOM conference::maintenance to enable them. But I don't know how
to look for these events in my script. Does anybody have some example
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