it can be a codec issue , make sure to use g711 at both ends /updating
latest firmware can help
Original Message:
-
From: Gopal krishnan [EMAIL PROTECTED]
Date: Tue, 23 Sep 2008 19:31:07 +0530
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-users] Freeswitch with
distortion
From: "Brian West" <[EMAIL PROTECTED]>
Date: 26/09/2008 9:30 pm
What kind of load and how many channels?
/b
On Sep 26, 2008, at 1:13 PM, Alex Kinch wrote:
> Hi,
>
> Just setup an IAX trunk from FS to a SIP provider who runs Asterisk,
> but getting a truckload of
trunk and distortion
From: "Brian West" <[EMAIL PROTECTED]>
Date: 26/09/2008 10:06 pm
if the recorded wav exhibits the issue sure.
/b
On Sep 26, 2008, at 3:39 PM, [EMAIL PROTECTED] wrote:
> No load as no other calls. Just one channel active. Sip to the same
> provider has wor
comes back up. Again, all of this is on the same box, so it
isn't a lan/wan issue or anything like that. I suspect I will get a
"update to the latest", but I wanted to check if there are any known
issues before I do that.
[EMAIL PROTECTED]> version
FreeSWITCH Version 1.0.tr
t the email with all that data which is expanded on
> the fly per message.
>
> by the time we finished adding what you want we will have recreated SMTP
> from scratch ;)
>
>
> On Wed, Oct 8, 2008 at 7:34 PM, Michael Jerris <[EMAIL PROTECTED]
> <mailto:[EMA
Hi,
I saw in the wiki that the mod_cdr module is now unsupported. There
is also a note
about a revival of the module. I would like to ask the following :
What is the current state of the revival process? (should we expect
something in the near future?)
Will it have the same functionality a
wrote:
> Unsure at this time. There has been some work on mod_cdr_odbc. We
> generally advise against direct to db cdr methods without a very
> robust backup method for when the db is down.
>
> On Oct 29, 2008, at 9:57 AM, "[EMAIL PROTECTED]" <[EMAIL PROTECTED]>
wrote:
> Unsure at this time. There has been some work on mod_cdr_odbc. We
> generally advise against direct to db cdr methods without a very
> robust backup method for when the db is down.
>
> On Oct 29, 2008, at 9:57 AM, "[EMAIL PROTECTED]" <[EMAIL PROTECTED]>
Yes, the xml files give you tons of info... but isn't it a little
insufficient - performance wise -
to open and close so many files in such a little time. In a PBX
environment that wouldn't be an
issue but if we get to the small-voip-carrier level (some thousand cdrs
per hour)
that could slow th
That's very good news. :)
Shawn Lewis wrote:
In regards to auto log rotation - YES YES
ANTHM just completed that item for me, where by you can set the time in
minutes i believe it was.
I have not tested it yet, hope to this week.
Shawn
Michael Collins wrote:
Yes, I agree. But one coul
Good point. I have got this kind of behavior (cdrs push model) in my
current system (using radius servers).
The only drawback of this method is that if you want to be absolutely
sure that all the cdrs were handled by
the web server (or radius server) you have to check at certain intervals
every
situations)
David Knell wrote:
[EMAIL PROTECTED] wrote:
Yes, the xml files give you tons of info... but isn't it a little
insufficient - performance wise -
to open and close so many files in such a little time. In a PBX
environment that wouldn't be an
issue but if we get to th
d only insert 1800 cdrs per hour...
If I was to insert 36000 cdrs per hour this means that I have to
open
parse
close
10 files per second. Imagine the I/O penalty just for opening - closing the
file.
(the persing is the same for both situations)
David Knell wrote:
[EMAIL PROTECTED] wro
m merely suggesting
extending the flexibility
of the already existing ones. Put some more lego tiles in your box set :)
Michael Collins wrote:
>
> /me sends Anthony’s post to the printer to be laminated and framed… J
>
> -------
Hi,
How can my FS accept inbound SIP calls from other gateways
without the need of a registration from their part? I only need to be able
to accept inbound calls from specific gateway IPs. I tried creating my
own profile
and gateway but it fails : "Error Creating SIP UA for profile: myprofile
nd.
Hope this helps.
Cheers, Birgit
On 04/11/08 16:24, [EMAIL PROTECTED] wrote:
Hi,
How can my FS accept inbound SIP calls from other gateways
without the need of a registration from their part? I only need to be able
to accept inbound calls from specific gateway IPs. I tried creati
llows:
> ==
> U xxx.xx.xx.186:2054 -> xxx.xx.xxx.xx:5060
> INVITE sip:[EMAIL PROTECTED];user=phone SIP/2.0.
> Via: SIP/2.0/UDP xxx.xx.xx.186:2054;branch=z9hG4bK-7qnms1c9aoqt;rport.
> From: "Company1000" ;tag=pslbxvhxjo.
> To: .
> Call-ID: 3c267
What does your dialplan look like? I see the error, but I can't quite
tell what is wrong. It looks like there is some sort of strange
variable assignment going on "variable string 0 =
[EMAIL PROTECTED]".
> 2008-11-10 18:12:57 [DEBUG] switch_c
It appears to have been cutoff. The last line that I see is:
"2008-11-11 23:38:27 [NOTICE] switch_loadable_module.c:281
switch_loadable_module_process() Adding F"
Peter P GMX wrote:
> Aaargh, being able to read can be a real advantage sometimes.
> I have now put the log to
> http://pastebin.fr
I am using acls (cidr) to accept incoming calls from a gateway that
I do not want to register in my FS box.
I have this gateway configured in a xml file :
freeswitch/conf/directory/default/gateway1.xml
I have the corresponding cidr
plate
can include any variables from the session.
/b
On Nov 18, 2008, at 12:50 PM, [EMAIL PROTECTED] wrote:
Any help on how to define an endpoint (originating) and use some
attribute (like account_code or user id)
for billing pur
Then what's the point of having it in the directory configuration file in
the first place, if you don't mind me asking?
I am really confused... :)
Anthony Minessale wrote:
you have to manually set the var on the channel in your dialplan.
On Tue, Nov 18, 2008 at 1:40 PM, [EMAIL
at 1:40 PM, [EMAIL PROTECTED] wrote:
The ${accountcode} variable IS set in the cdr_csv.xml conf file yet
the field is empty after the call.
Shouldn't it also show in the xml cdr? I thought the XML CDRs
included all of the se
Is there an equivalent to asterisk's NoOp() so that I can print (within
the XML dialplan) the stuff I want
on the FS console? Like variables and stuff?
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iki.freeswitch.org/wiki/Misc._Dialplan_Tools_eval
-MC
-Original Message-
From: [EMAIL PROTECTED]
[mailto:freeswitch-
[EMAIL PROTECTED] On Behalf Of [EMAIL PROTECTED]
Sent: Wednesday, November 19, 2008 10:56 AM
To: freeswitch-users@lists.freeswitch.org
Subject: [Freeswitch-u
Figured that out by myself. One has to raise the console debug output to
"debug".
[EMAIL PROTECTED] wrote:
Tried that, but the output of a simple data="hello" /> does not appear in my console.
I verified that the context that the eval is in gets executed.
I have the
Hi,
Is there a way to declare more than one script with its binding in
perl.conf.xml?
Because from what I understood by reading the documentation, is that
there are
no different sections to define different perl scripts with bindings
like for example in the
xml_curl.conf.xml :
ipt is called from the xml dialplan? e.g. :
data="/root/test_perl2.pl" />
Anthony Minessale wrote:
no the languages only have one binding.
Do you really need more than one binding?
On Thu, Nov 20, 2008 at 6:20 AM, [EMAIL PROTECTED]
<mailto:[EMAIL
lize();
print $env->serialize("xml");
$info = $env->getHeader("info");
On Thu, Nov 20, 2008 at 8:34 AM, [EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]> <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:
What if I want to use one binding for "
ot;);
$info = $env->getHeader("info");
On Thu, Nov 20, 2008 at 8:34 AM, [EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]> <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:
What if I want to use one binding for "directory", one for
"configurat
I have a perl script (for dialplan generation) that works fine.
When I try to use the DBI module I get a segmentation fault. My OS is
Linux CentOS 5.2
and I am using freeswitch-1.0.1.
If I can recall correctly, some other guy had the same problem a few
months ago but I cannot
find the mailing lis
All the variables that I set show up only in the a-leg CDR.
How can I set a variable that can be used during the b-leg CDR generation?
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Hi,
I am making a simple bridge between two call legs :
Client --(a-leg)--> FS --(b-leg)-->Provider
How can I get information like network-address of the Provider,
media-address,
port used, media port used etc. from the second leg (b-leg)?
Is all the information provided by the a-leg avail
_caller_id_name}
On Wed, Dec 3, 2008 at 7:30 AM, [EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]> <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:
Hi,
I am making a simple bridge between two call legs :
Client --(a-leg)--> FS --(b-leg)-->Provider
r you need to set it as a
custom variable and insert it
into your template for csv cdr or it will just be there in xml cdr
On Wed, Dec 3, 2008 at 8:18 AM, [EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]> <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wrote:
b-leg logging is en
t reflects the port being used between that
> leg and it's remote connection eg the ip and port that the rtp stack
> was asked to use.
>
>
> On Wed, Dec 3, 2008 at 9:48 AM, [EMAIL PROTECTED]
> <mailto:[EMAIL PROTECTED]> <[EMAIL PROTECTED] <mailto:[EMAIL PROTECTE
. I guess nobody ever looked at that
field before.
it should be fixed in r10582
--
Anthony Minessale II
FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
AIM: anthm
MSN:[EMAIL PROTECTED]
<mailto:[EMAIL PROTECTED]>
GTALK/JABBER/PAYPAL:[EMAIL PROTECTED]
<mail
very broad set of tools to debug).
Measuring things like ASR, PDD, etc in my opinion is much easier from
the raw messaging than trying to do something with FS CDR records.
On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos
<[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wr
very broad set of tools to debug).
Measuring things like ASR, PDD, etc in my opinion is much easier from
the raw messaging than trying to do something with FS CDR records.
On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos
<[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wr
very broad set of tools to debug).
Measuring things like ASR, PDD, etc in my opinion is much easier from
the raw messaging than trying to do something with FS CDR records.
On Tue, Dec 9, 2008 at 9:19 AM, Apostolos Pantsiopoulos
<[EMAIL PROTECTED] <mailto:[EMAIL PROTECTED]>> wr
HI
We want to try generate 5000 simultanious Voice broadcast calls .
can the below config will work?
SS7 Links< > Sangoma SMG<--->FreeSwitch<>Broadcasting
Application ( SIP based)
Thank you
Imthiyaz
mail2web.
ation will be appreciated.
Thank you
Imthiyaz
Original Message:
-
From: Michael Collins [EMAIL PROTECTED]
Date: Sun, 29 Jun 2008 19:46:55 -0700
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] [Freeswitch-dev] YAML support
asanalternative of XML for configur
Hi All,
would FreeSWITCH 'transcode' H.245 alphanumeric DTMFs
to an H.245 signal / rfc2833 H.323 device over G.729 codec ?
Thanks for supporting,
.TF
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There is no SCCP module for FS. CM only uses SCCP to talk to phones, it
uses either MGCP or SIP to talk to gateways. So if you have a version
that has SIP support (I believe > 4.0), then you could connect CM to FS.
Cavalera Claudio Luigi wrote:
> Hello,
> is there a way to interconnect fs to a
hiyaz
Original Message:
-
From: Martin Joseph [EMAIL PROTECTED]
Date: Fri, 19 Sep 2008 09:20:45 -0700
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Asiterisk Dialplan for Freeswitch
On Sep 19, 2008, at 6:23 AM, Gopal krishnan wrote:
> Hi,
>
>
following error:
2008-09-20 15:49:32 [NOTICE] switch_channel.c:534
switch_channel_set_name() New Channel sofia/internal/[EMAIL PROTECTED]
[d1987c09-ebd2-4590-b072-ce851e4b5794]
2008-09-20 15:49:32 [NOTICE] switch_channel.c:534 switch_channel_set_name()
New Channel sofia/internal/[EMAIL PROTECTED
Hi brain
pls check this link http://wiki.freeswitch.org/wiki/Examples_calltest_js
thanks
Imthiyaz
Original Message:
-
From: Brian West [EMAIL PROTECTED]
Date: Sat, 20 Sep 2008 11:32:36 -0500
To: freeswitch-users@lists.freeswitch.org
Subject: Re: [Freeswitch-users] Error when
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