hi all
freeswitch support PCMU only?
i follow the http://wiki.freeswitch.org/wiki/Codecs#sofia.conf.xml,
but freeswitch still support PCMU only,
below is the trace:
2009-07-01 14:39:35.571364 [DEBUG] sofia_glue.c:3070 Audio Codec
Compare [PCMA:8:8000:0]/[PCMU:0:8000:20]
*Hi,
i have changed the openzap.conf file but still i get the same error
**[span wanpipe 1]
number => 1
trunk_type => e1
b-channel => 1:1-15
d-channel => 1:16
b-channel => 1:17-31*
*
freeswi...@localhost.localdomain> load mod_libpri
API CALL [load(mod_libpri)] output:
-ERR [module load file routi
On Jun 30, 2009, at 8:43 PM, Edmar Cruz wrote:
>
> I have 1 Fs and 1 Asterisk if G729 is available on Asterisk so i
> shall load
> to G729 for freeswitch that needs a license?
>
You need a license if you are transcoding to or from G729. If you are
just passing the media stream or you stay
Hello,
I'm experiencing a bug that I've been working on most of today. I can not
call between two SIP phones that register successfully. In order to diagnose
it, I have removed my FreeSWITCH server out of the NAT/firewall to try and
eliminate any such issues with these things.
Here is how I ran in
I have 1 Fs and 1 Asterisk if G729 is available on Asterisk so i shall load
to G729 for freeswitch that needs a license?
Steve Underwood wrote:
>
> Edmar Cruz wrote:
>> So what codec supports mobile phones?
>>
> The main codecs used by mobile phones are:
> GSM FRThe original GSM c
And unless you are directly connecting to the cell phone provider you are
going to be converted to ULAW/ALAW to traverse the PSTN. So there is no
advantage going to a native GSM codec only to have it expanded out to G711
and then back to GSM codec.
On Tue, Jun 30, 2009 at 10:05 PM, Steve Underwoo
Edmar Cruz wrote:
> So what codec supports mobile phones?
>
The main codecs used by mobile phones are:
GSM FRThe original GSM code, largely replaced by later
codecs (some VoIP stuff uses this)
GSM HRThe half rate codec for GSM
GSM EFR A later improved full rate
GSM
2009/7/1 Edmar Cruz :
>
> So what codec supports mobile phones?
>
--
Craig Askings
Network Engineer | Over the Wire Pty Ltd
cr...@overthewire.com.au | www.overthewire.com.au
Phone: 07 3847 9292 | Fax: 07 3847 9696 | Mobile: 0404 019 365
___
Freesw
So what codec supports mobile phones?
jfenton wrote:
>
>
> Hi Edmar,
>
>> I need actually G729 license if there's any for freeswitch to call
>> mobile
>> phones... I already load it but has an issue on it passthrough
>> mode... If i
>> install mod_dandhi_codec it overwrites the existing G7
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On Jun 30, 2009, at 12:45 PM, Raffaele P. Guidi wrote:
Ok, so tomorrow I'll find http://files.freeswitch.org/freeswitch-1.0.4pre9.msi?
Thanks a lot,
Raffaele
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Ok, so tomorrow I'll find
http://files.freeswitch.org/freeswitch-1.0.4pre9.msi?
Thanks a lot,
Raffaele
On Tue, Jun 30, 2009 at 17:29, Carlos Talbot wrote:
> I do make an effort to update the svn MSI every time a new release is
> announced. The current MSI was posted this morning (svn 14043) an
On Tue, Jun 30, 2009 at 9:03 AM, Michael Jerris wrote:
> the bridge app already does all this for you doesn't it (along with
> bind_meta) ?
>
> Mike
>
In other words, everything you want is available in the dialplan with no
overheard from launching a JS.
-MC
_
We currently support t.38 passthrough only using proxy_media mode. T.
38 gateway is on the roadmap but not yet close to complete.
Mike
On Jun 30, 2009, at 5:15 AM, François Delawarde wrote:
> Many issues on Asterisk's T.38 (or probably just on T.38?)...
>
> Could it convince those relying on t
you have a pointer somewhere in your directory for that user, hard to
see without seeing the whole config, but grep for 111 and see what
else you find.
Mike
On Jun 30, 2009, at 10:21 AM, Alexey Lubimov wrote:
> I sofia_reg.c:1765 have two user records - good #110 and bad #111.
>
> bad.xml:
http://wiki.freeswitch.org/wiki/Mod_commands#in_group
http://wiki.freeswitch.org/wiki/Mod_commands#user_exists
On Jun 30, 2009, at 6:09 AM, Christian Benke wrote:
> Hello!
>
> I have the following scenario:
>
> I want to check if a called extension is part of a group, or as an
> alternative, if
Thanks. I installed libtiff from source to /usr/local/ and now it works.
On Jun 30, 2009, at 11:49 PM, Michael Jerris wrote:
> I think that detection is not working on mac right due to it looking
> in default search paths. I am in process of fixing this to use in
> tree libtiff soon so this shou
If you don't need authentication, you don't need a gateway, if you do,
you will need to setup a user on the other box to register to.
On Jun 30, 2009, at 3:05 AM, Brad Tuan wrote:
> I know that i need to set the dialplan,
>
> my problem is when FS_B send a REGISTER to FS_A, FS_A will return a
the bridge app already does all this for you doesn't it (along with
bind_meta) ?
Mike
On Jun 27, 2009, at 2:45 AM, Dome Charoenyost wrote:
> Dear All,
>
> I try
>
> s = new Session("sofia/external/x...@xxx.xxx.xxx.xxx);
> if (s.ready()){
> s.setVariable("nibble_rate", "2.5");
> s.setVariab
I think that detection is not working on mac right due to it looking
in default search paths. I am in process of fixing this to use in
tree libtiff soon so this should fix this issue.
Mike
On Jun 29, 2009, at 11:08 PM, seven wrote:
> Hi,
>
> I'm on the latest svn 14041, and I have tiff inst
I do make an effort to update the svn MSI every time a new release is
announced. The current MSI was posted this morning (svn 14043) and should be
synced up by this evening (CST).
Carlos
http://wiki.freeswitch.org/wiki/Download_%26_Installation_Guide#Precompiled_Binaries
On Tue, Jun 30, 2009 at 6
If you're already on trunk then just do "make current"
-MC
Sent from my iPhone
On Jun 30, 2009, at 2:04 AM, Saeed Ahmad
wrote:
Hi,
What is the best way to update to latest version if we are already
running an older stable version?
I am using SVN trunk, sources are in /usr/src and its
> Well the 8XX series is much prettier than the 3XX series. Though I
> wonder if anyone has thought of porting FreeSWITCH to the Palm Pre?
Palm Pre is Linux-based, so shouldn't be that difficult so long
as you can get a working portaudio up and running (it looks like
pulseaudio has already been p
Hi Edmar,
> I need actually G729 license if there's any for freeswitch to call
> mobile
> phones... I already load it but has an issue on it passthrough
> mode... If i
> install mod_dandhi_codec it overwrites the existing G729 without
> license to
> a new G729 with license?
Are you sure yo
I sofia_reg.c:1765 have two user records - good #110 and bad #111.
bad.xml:
good.xml:
Good user 1
Hi, I would like to give a try to this and all other "pre" releases but
being tied to windows platforms (and not having a C compiler available) I
would need an MSI installer. Is there a way to add a windows build to the
(pre)release process (without doubling your work, of course ;)? I think one
of
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hi Drago,
sorry for correcting you, but Snom's touch scrren phone is the 870 ...
http://www.snom.com/en/products/snom-870/
cheers
Helmut
On 30.06.2009 04:32, Drago Totev wrote:
> Actually, Snom does have a version with color LCD touch screen - mode
Saeed Ahmad wrote:
> Can you give me more info with CentOS.
>
> I am more comfortable with SVN trunks, can i do the same with SVN trunks?
Yes. There is a spec file in the source tree for building packages. There
should be instructions on the wiki explaining how to use it - if not, someone
who is
Hello!
I have the following scenario:
I want to check if a called extension is part of a group, or as an
alternative, if it is a user in the directory.
My basic intention is to find out if a 3-digit extension leads to a
valid user - if it doesn't, some other action will happen.
Maybe there are be
Can you give me more info with CentOS.
I am more comfortable with SVN trunks, can i do the same with SVN trunks?
Thanks
On Tue, Jun 30, 2009 at 11:14 AM, Jason White wrote:
> Saeed Ahmad wrote:
> > What is the best way to update to latest version if we are already
> running
> > an older stabl
Replace [span wanpipe1] with [span wanpipe 1] in your openzap.conf file
- it's missing a space between wanpipe (the span type) and 1 (the span
ID).
Regards,
Raul
On Tue, 2009-06-30 at 14:57 +0530, Baskar wrote:
> Hi,
>
> I have installed the latest trunk with A102D Sangoma card when i load
> th
*Hi,
**I have installed the latest trunk with A102D Sangoma card when i load the
openzap i get this error
**
freeswi...@localhost.localdomain> load mod_openzap
2009-06-30 14:47:33 [NOTICE] zap_io.c:2626 zap_global_init() Modules
configured: 1
2009-06-30 14:47:33 [ERR] zap_io.c:2365 load_config()
Many issues on Asterisk's T.38 (or probably just on T.38?)...
Could it convince those relying on this "modern" version of a 50yo
technology to switch to and with FreeSwitch?
François.
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Saeed Ahmad wrote:
> What is the best way to update to latest version if we are already running
> an older stable version?
You did ask for the best way, which is to build packages for your operating
system, then use your operating system's package manager to install them and
keep track of differe
Hi,
What is the best way to update to latest version if we are already running
an older stable version?
I am using SVN trunk, sources are in /usr/src and its installed in
/usr/local/freeswitch
>From the same src directory, is it possible to install latest rev in
somewhere else (for example /opt/f
On Jun 30, 2009, at 2:54 PM, Baskar wrote:
Hi Michael Jerris,
Is there any other possible way to queue the inbound call in
JavaScript.
I am working on this process:
step 1: I want the inbound call to be in queue through JavaScript
step2: Then JavaScript will check most waiting agent and
I know that i need to set the dialplan,
my problem is when FS_B send a REGISTER to FS_A, FS_A will return a 403 to
FS_B
Like this:
2009-06-30 15:03:25 [NOTICE] sofia_reg.c:305 sofia_reg_check_gateway()
Registering FS_A
2009-06-30 15:03:25 [ERR] sofia_reg.c:1391 sofia_reg_handle_sip_r_register()
2009/6/30 Michael Collins :
> can you post your script and dialplan? Let's take a look.
> -MC
dialstr[i] is array like a sofia/external/1...@xxx.xxx.xxx.xxx
dial_option =
"{absolute_codec_string='GSM,G729',ignore_early_media=false,originate_timeout=20,origination_caller_id=x}
for (var
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