Issues of this type must be reported to jira http://jira.freeswitch.org
On Mon, May 11, 2009 at 3:54 AM, Richard Lamkin
richard.lam...@mettoni.comwrote:
Build on fails @ trunk =13273
My last working build @ trunk = 13231
Looks like the addition of the directives _Out_, _In_, _In_opt_,
Folks;
Bear in mind that the frequency is (X)Hz * (num cores), hence saying
100Hz on a dual core winds up being 200Hz.
My setup is 250Hz on a Dual-Core and the quality is perfect.
Oh, btw folks, don't attempt to do anything involving QOS (be it TBF,
CBQ, HTB, or whatnot) on anything
I'm implementing an IVR solution for FreeSWITCH, but I have a little problem
with audio playback. I'm just calling into application park, and then handle
the flow using the event socket.
All my audio phrases are .PCMA (8KHz a-law), and I play lots of files after
eachother. Between each file I
Chances are if both legs are NOT already alaw you'll need to record it
with .wav or .al files. .PCMU or .PCMA are native file formats if any
transcoding is taking place you probably can't get way with .PCMA.
/b
On May 11, 2009, at 8:39 AM, Peter Olsson wrote:
Hello again,
I also have a
I record them to file.wav and they play perfectly. I think it's recorded
in a raw-format here. See:
http://www.nabble.com/Recording-ULAW-files-td21587474.html
Best regards
Peter
Peter Olsson schrieb:
Hello again,
I also have a problem when I try to record messages. I record to
Have you tried wav files?
/b
On May 11, 2009, at 8:34 AM, Peter Olsson wrote:
I’m implementing an IVR solution for FreeSWITCH, but I have a little
problem with audio playback. I’m just calling into application park,
and then handle the flow using the event socket.
All my audio phrases
In this case it is pure PCMA, all the way from the phone. I just dial in to
number 2100 (using SIP, codec PCMA), and then I have a event socket connected
that sees the ivr-test flag. I then play some files (PCMA), and then start a
recording.
extension name=ivr-test2
condition
Brian,
Thanks for the response. No, I didn't try wav files - and I'd prefer to keep
the current codec if that's possible. But I could give it a try and see what
happens.
Do you think it might only be related to the native files in FS?
//Peter
Från:
I'm not sure. Can you provide me a test file and a known case that
you can produce this issue with?
/b
On May 11, 2009, at 9:20 AM, Peter Olsson wrote:
Brian,
Thanks for the response. No, I didn’t try wav files – and I’d prefer
to keep the current codec if that’s possible. But I could
-BEGIN PGP SIGNED MESSAGE-
Hash: SHA1
Hello,
I can reproduce the error with Snom 370 and FW 7.3.11. An update to
7.3.20 (beta) seems to solve the problem - at least I can't reproduce it
with the metioned scenario anymore.
regards
helmut
-BEGIN PGP SIGNATURE-
Version: GnuPG
The problem is that it's not 100% reproducable. Sometimes I can play 2-3-4
files, and it sounds great, sometimes it clicks louder, somtimes not so much.
I could get a Wireshark dump for you, could that help?
Regards,
Peter
Från: freeswitch-users-boun...@lists.freeswitch.org
What phone are you using?
/b
On May 11, 2009, at 9:40 AM, Peter Olsson wrote:
The problem is that it’s not 100% reproducable. Sometimes I can play
2-3-4 files, and it sounds great, sometimes it “clicks” louder,
somtimes not so much. I could get a Wireshark dump for you, could
that help?
I've confirmed that recording to wav files works just fine. So it seems to be
somthing strange with the native files in FS?
/Peter
Från: Peter Olsson
Skickat: den 11 maj 2009 16:16
Till: 'freeswitch-users@lists.freeswitch.org'
Ämne: RE: [Freeswitch-users] Audi record using uuid_record
In this
In this case I'm using Avaya Phones connected to a Avaya CM PBX, which talks
SIP to FreeSWITCH. But I'll try to connect a SIP phone directly as well - to
see if it makes any difference. I Have a Polycom IP550 I can use for some
testing.
I've used the exact same setup using yate, and that
FYI, for those of you keeping up on ClueCon 2009 please visit the latest
blog entry:
http://cluecon.com/node/29
Thanks,
Michael
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I am having difficulty making an outbound call. I have read the wiki many
times, but I am missing something.
First, after setting up a gateway xml file, should that gateway show in the
FS console after issuing the 'reloadxml' command? It does not.
Can anyone give me a push? Thanks, Lars
Try this ^1?(\d{10})$
and this sofia/gateway/flowroute/flowrouteAccount#1$1
/b
On May 11, 2009, at 4:38 PM, Lars Zeb wrote:
in conf/dialplan/default/02_long_distance.xml:
!-- Dial any 10 digit number (222333) or 1+10 number
(1222333) here --
extension name=Long Distance -
Hi!
I need to enable debug mode in dingaling as I can't see that freeswitch is
coming online in gtalk.
I have changed the following:
changed the loglevel to debug in console.conf.xml
changed the debug level to 1 in dingaling.conf to 1
I do not see any xmpp logs in the console or in
Try dl_debug on at the CLI
/b
On May 11, 2009, at 5:11 PM, Mark Campbell-Smith wrote:
Where is the XMPP traces?
Thanks
Brian West
br...@freeswitch.org
-- Meet us at ClueCon! http://www.cluecon.com
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I'm sorry, but do not understand what it is that I should try. Are you
saying to change the data attribute in the action command of the dialplan?
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of
You want to collect 11 or 10 digits and send it out flowroute which
requires 1+ number. The 1? makes the one optional. So now we only
collect 10 digits and append the on in the bridge line.
extension name=Long Distance - flowroute
condition field=destination_number
Hi,
I need get call duration after bridge application using mod_managed, my code:
Session.Execute(bridge, sbNewOutBoundNum);
Debug(billsec : + _Session.GetVariable(billsec));
Debug(duration : + _Session.GetVariable(duration));
The bridge is ok, but the variable value duration and
On May 11, 2009, at 7:51 PM, Lars Zeb wrote:
Thanks for the clarification. It made sense, but the results remain
the same. The log still says ‘Invalid Gateway’ and ‘sofia status’ at
the console does not show flowroute.
It sounds like sofia hasn't picked up your new gateway yet. Have
Hi,
I can see the RDN in the log file, but don't know how to retrieve it
from a Lua script.
Regards,
Cliff
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I found a workaround, but it'd be nice to actually have the RDN easily
accessible from Lua:
calling_number = session:getVariable ( sip_h_Diversion )
_, _, calling_number = string.find ( calling_number, sip:(%d+)@ )
Cliff
On Mon, 2009-05-11 at 17:22 -0700, Cliff Wells wrote:
Hi,
I can see
The one you emailed anthony about the invite... 487 repeat stuff.. can
you give me more details on what might be going on?
/b
On May 11, 2009, at 5:23 PM, Lars Zeb wrote:
I'm sorry, but do not understand what it is that I should try. Are you
saying to change the data attribute in the action
Hehe its not really a work around... Its how you do it either way...
but I did add the patch from http://jira.freeswitch.org/browse/MODSOFIA-7
which would require you to do similar.
/b
On May 11, 2009, at 8:20 PM, Cliff Wells wrote:
I found a workaround, but it'd be nice to actually have
Hi,
I'm trying to use the Freeswitch as a proxy (I know that is not designed for
that, but I really need to do it in this way), here is my config:
Endpoint 1- FS A--FS B-FS A-Endpoint 2
* Both Endpoints are registered in FS A how is acting as a proxy and
registrar.
* FS B only sends back the
Juan,
Can you explain your situation a little better you seem to have
breezed over the critical details. Also you should enable STUN on
your endpoints and not depend on your Registrar to overcome nat issues
since its not its job.
/b
On May 11, 2009, at 10:03 PM, Juan Manuel Vicente
I believe you're talking about the FS sending out an ICMP in response to the
client's invite, which resulted in 'ICMP Destination unreachable'. He told
me to turn iptables off, which I did. Then the Eyebeam registered
successfully. I don't know what the '487 repeat stuff' was.
From:
Haha that's ok I sent that email to you by mistake I wondered
where that email went! /b
Sent from my iPhone
On May 11, 2009, at 11:29 PM, Lars Zeb larc...@yahoo.com wrote:
I believe you’re talking about the FS sending out an ICMP in respons
e to the client’s invite, which resulted in
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