Just a question,
do you use Freeswitch in bypass-media-mode in this scenario? Then media
negociation should be handled outside Freeswitch.
Best regards
Peter
Jerry Richards schrieb:
After establishing an audio call between two Bria softphones, and then
starting video at the caller phone, FS
Hello,
in a bigger installation with some thousand endpoints in the field we
see, that the endpoint equipment is always using INFO messages (standard
setting is auto, so the endpoint decides which method to use). I have 2
questions to that scenario:
1. Is there a way that Freeswitch
ubuntu-8.04.3-server-amd64.iso (update/upgrade)
FreeSWITCH Version 1.0.trunk (15787)
skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb
mod_skypiax
(POTS)--(PSTN)--(skypeIN)--(skype_client)--(skypiax)--(fs)
extension name=Indial_to_fs_via_skypeIN
condition field=destination_number expression=^501$
Can you repeat that same trace with latest trunk?
On Mon, Dec 21, 2009 at 6:44 PM, Jerry Richards
jerry.richa...@teotech.comwrote:
After establishing an audio call between two Bria softphones, and then
starting video at the caller phone, FS replies to the re-INVITE with a 200
OK with only
The force-register-domain and force-register-db-domain are set in the defaults
so you can only do one domain. Remove those and you'll be able to do multiple
domains.
/b
On Dec 21, 2009, at 6:15 PM, j...@acsol.net wrote:
I have Freeswitch setup and working as a single tenant
system mostly
add start_dtmf app to your dialplan before bridge to start the inband dtmf
detector.
On Tue, Dec 22, 2009 at 8:57 AM, Scott Torr
scott.torr...@letterboxes.orgwrote:
ubuntu-8.04.3-server-amd64.iso (update/upgrade)
FreeSWITCH Version 1.0.trunk (15787)
skype-ubuntu-intrepid_2.1.0.47-1_amd64.deb
do as anthm say :-)
On Tue, Dec 22, 2009 at 4:21 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
add start_dtmf app to your dialplan before bridge to start the inband dtmf
detector.
On Tue, Dec 22, 2009 at 8:57 AM, Scott Torr scott.torr...@letterboxes.org
wrote:
It is probably because mod_skypiax does not analize incoming audio
looking for dtmf, because the normal call from a Skype client peer
sends *both* inband and out of band (signaling) dtmf.
So, I choose to only detect out of band (signaling) dtmfs, and ignore
possible inband dtmfs (in the audio
I have a Freeswitch PBX server with an installed Sangoma A101D card
connected to a PRI. Most everything works okay, however when I get an
inbound call from the PSTN, if the call is not answered within about 12
seconds, the call ends (so it doesn't go to voice mail). If I make a call
from one
Thanks Brian. I did have both force-register-domain and
force-register-db-domain commented in both the internal.xml and
internal-ipv6.xml. The phones appear to register to the company1 domain,
as shown in sofia status profile company1; however I have noticed that
when I try to make a call to
No. The following lines is commented out (internal.xml):
!--param name=media-option value=bypass-media-after-att-xfer/--
!--param name=inbound-bypass-media value=true/--
Thanks,
Jerry
-Original Message-
From: Peter P GMX [mailto:prometheus...@gmx.net]
Sent: Tuesday, December 22,
I'm not too sure you can put an a1-hash on outbound auth.
/b
On Dec 22, 2009, at 11:26 AM, Babak Karvandi wrote:
Hi,
Does any body know or has test the a1-hash parameter with gateway
setting? I am not sure if it is even allowed. I have the following
gateway setting but when the
One point of clarification, currently all the phones are behind NAT, so
it appears that when the phones are in a Non-multitenant scenario, they
use SIP:dialed_num...@ip-address-of-their-gateway.
On 12/22/2009 9:16 AM, John wrote:
Thanks Brian. I did have both force-register-domain and
I have set up a second FreeSWITCH box on the same LAN. I have v16018
installed on it and have changed nothing.
I configured a Polycom phone to register one of its four lines to this
second box, but it does not register. When looking at the console, there is
no activity. However, there is SIP
On Tue, Dec 22, 2009 at 10:50 AM, Larry Marshall l...@marshap.com wrote:
I have set up a second FreeSWITCH box on the same LAN. I have v16018
installed on it and have changed nothing.
I configured a Polycom phone to register one of its four lines to this
second box, but it does not
Hello
I'm running 1.0.trunk (15841) on Linux CentOS with a the default settings.
After succesfully connecting a Windows PC running XLite (EyeBeam, really)
and a GrandStream IP phone to Freeswitch, I try to make calls, but am having
the following issues:
1. When calling XLite from GS, XLite
I found the cause for #2: The GS phone was still configured to use NAT, even
though both XLite and GS are located in the same, private LAN. Unchecking
this on the GS phone solved the issue.
But I'm still having issue #1, regardless of which phone is calling or being
called: When the phone
It is usually CODEC related. probably the SIP messages has the cause inside.
__Yehavi:
2009/12/22 Fred-145 codecompl...@free.fr
I found the cause for #2: The GS phone was still configured to use NAT,
even
though both XLite and GS are located in the same, private LAN.
My distro is fedora 10 with all the current patches.
SSLwatch fails to build and it seems more than a trivial change to make it
work; however, it seems that the error message from Freeswitch tells it
all...
Is there any special debug statement in Freeswitch to see more about its TLS
negotations?
On Tue, Dec 22, 2009 at 11:35 AM, Yehavi Bourvine yehavi.bourv...@gmail.com
wrote:
Try tracing the calls from both sides with TCPDUMP or enable siptrace on
FreeSwitch. I guess this will give you some clue.
__Yehavi:
Additionally, turn on debugging on the console
Yes, the internal profile exists.
Name Type Data
State
=
internal profile sip:mod_so...@192.168.10.25:5060
Yuriy,
The FS wiki has examples of how to control the BLF/MWI using events.
I had no problem getting to work with my GXP2020.
Let me know if you want some direct code examples.
-- MM.
On Thu, Dec 17, 2009 at 6:05 AM, Yuriy Ivzhenko yivzhe...@mksat.net wrote:
Hallo All!
I need information
Excellent work and answers.
Thanks gentlemen.
I'm firing off a new thread re: codecs et. al.
Have a great Christmas and a wonderful, prosperous New Year.
Regards,
Ahmed.
2009/12/21 Bill W freeswi...@aastral.net
I recently added an overview to this wiki page to help make things more
clear
Hello people,
Can someone please clear the following ambiguities with codecs:
1. Are we definitively able to run pass-through codecs (e.g. G.729) in
Proxy Media mode, or does FS need to be running in bypass-media ? the Wiki
is not clear in this regard
2. When an A-leg has negotiated
It's upgrade-and-test time! The new release announcement is on the main
FreeSWITCH page:
http://www.freeswitch.org/node/224
Please update, test, and report back bugs and questions.
Thanks!
-Michael
___
FreeSWITCH-users mailing list
On Tue, Dec 22, 2009 at 11:46 AM, Lars Zeb larc...@yahoo.com wrote:
Yes, the internal profile exists.
Name Type
Data State
=
internal
Hi,
I am playing a file to a landline number.
the format of the file is as follows:
[r...@static-host var]# file message.wav
message.wav: RIFF (little-endian) data, WAVE audio, ITU G.711 mu-law, mono
8000 Hz
In my vars.xml file I have used the following codec prefs:
X-PRE-PROCESS cmd=set
Hi all,
This weekend, I got the chance to buy a profoon IP-150 RJ11-to-USB
device for just 15 euro. This is a device which has on one side a
USB-connector and on the other side 2 RJ-11 connectors (one FXO and one
FSX). Internally, the device seams to contain a tigerjet 560C chipset.
(see here:
Hello Lars,
You can apply any acl to any profile. What you should do really depends
on what you want to accomplish.
But let's take a simple example. Let's say you want to allow any phone
on your internal network (192.168.0.0/24) to connect to your internal
profile and make calls without
Why? You don't have to avoid it... why bother?
/b
On Dec 22, 2009, at 4:28 PM, Vinuth Madinur wrote:
My basic intent is to avoid on-the-fly transcoding, while having a high
quality audio playing on PSTN.
___
FreeSWITCH-users mailing list
The audio quality is a lot different when it plays on the landline. And the
quality degrades a bit when the message played is lengthy 30s. So I thought
it would be better if I have the file in mu-law and play it as is..
Thanks,
Vinuth.
On Wed, Dec 23, 2009 at 4:09 AM, Brian West
If its degrading like that you have bigger issues... the sound files played
from wav files vs raw PCM files is NO different on a land line and I speak from
very many years of experience... your wav files are ulaw in wav containers thus
will never play native which might just be part of your
Hi all,
I just downloaded the newest trunk about 5 minutes ago and I got the following
make error on Ubuntu 8.04:
gcc -E /usr/src/freeswitch/src/include/switch_cpp.h -DSWITCH_DECLARE_CLASS=
-DSWITCH_DECLARE\(x\)=x -DSWITCH_DECLARE_CONSTRUCTOR=
-DSWITCH_DECLARE_NONSTD\(x\)=x 2/dev/null | grep
Klaus Hochlehnert maili...@kh-dev.de wrote:
src/switch_apr.c:899: warning: control reaches end of non-void function
Are you on rev. 16032?
As of 16032, this function shouldn't generate any such warning unless there's
a compiler bug.
___
I was on 16031.
Now I downloaded 16032 and currently the make is running.
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On Behalf Of Jason White
Sent: Wednesday, December 23, 2009 12:38 AM
To:
On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji a.alalo...@gmail.com wrote:
Hello people,
Can someone please clear the following ambiguities with codecs:
Are we definitively able to run pass-through codecs (e.g. G.729) in Proxy
Media mode, or does FS need to be running in bypass-media ? the Wiki
Interesting. It would have to do more than just dialtone/dtmf though.
Need call control, caller id, etc. What do they ship with it as far
as drivers go?
On Tue, Dec 22, 2009 at 4:06 PM, Kristoff Bonne
kristoff.bo...@skypro.be wrote:
Hi all,
This weekend, I got the chance to buy a profoon
On the other hand, a u-law WAV turned into L16 and then back to u-law to
be sent down the line shouldn't suffer any alteration at all - if it
does, the there's something wrong with the translation.
The quality dropping over time is almost certainly down to something
else. Vinuth -can you get a
Have you considered GIPS http://www.gipscorp.com/products/overview.php ?
-E
___
FreeSWITCH-users mailing list
FreeSWITCH-users@lists.freeswitch.org
http://lists.freeswitch.org/mailman/listinfo/freeswitch-users
Am I correct in presuming that Freeswitch will answer a fax from a local zap
based user
just like it does from an FXO port connected to a POTS line? What I hope to
do here is
catch any call made from that extension (the zap based fax machine/user) and
push its
call into the fax module.
If this is using prid it also requires the latest drivers from
sangoma. I am pretty sure these are just in dev snapshots not release
drivers yet. Something 3.5.8.6 or later iirc.
Mike
On Dec 21, 2009, at 7:52 PM, Brian West br...@freeswitch.org wrote:
You know that warning is
For the path in the dialplan I don't think we have any right now but
file a bug on jira and I can try to add them. As for something in the
script itself that is a bit more work but if anyone has a patch to
inject some vars into scripts like that it would be a nice addition.
Mike
On Dec
Not sure if we have an option to disable info. Even without this,
dtmf should go across the bridge fine. Please open up a bug on jira
about this
Mike
On Dec 22, 2009, at 6:40 AM, Peter P GMX prometheus...@gmx.net wrote:
Hello,
in a bigger installation with some thousand endpoints in
If your seeing the trafic in ngrep bit not in sip trace in Sofia when
enabled, your firewall is blocking the traffic
Mike
On Dec 22, 2009, at 5:20 PM, Michael Collins m...@freeswitch.org wrote:
On Tue, Dec 22, 2009 at 11:46 AM, Lars Zeb larc...@yahoo.com wrote:
Yes, the internal profile
We expect the g729 sometime very soon, weeks not months away.
Mike
On Dec 22, 2009, at 7:45 PM, Rupa Schomaker r...@rupa.com wrote:
On Tue, Dec 22, 2009 at 2:55 PM, Ahmed Naji a.alalo...@gmail.com
wrote:
Hello people,
Can someone please clear the following ambiguities with codecs:
Are
That being said, ulaw l16 alaw will cause degredation and any other
modifications such as volume adjustment in this path will make it
worse. Tha being said that does not sound like what you are
experiencing
Mike
On Dec 22, 2009, at 10:29 PM, David Knell d...@3c.co.uk wrote:
On the other
For the path in the dialplan I don't think we have any right now but
file a bug on jira and I can try to add them. As for something in the
script itself that is a bit more work but if anyone has a patch to
inject some vars into scripts like that it would be a nice addition.
Mike
Ok, signed up
Hello. I've written an application using mod_conference which often has two
parties speaking at once and one party listening.
When only one party is speaking, the sound quality is fine, but when a second
party starts speaking while the first party is still speaking, the second
party's
voice
Sounds right to me, just assign it to me if it lets you
Mike
On Dec 23, 2009, at 12:03 AM, Joseph L. Casale jcas...@activenetwerx.com
wrote:
For the path in the dialplan I don't think we have any right now but
file a bug on jira and I can try to add them. As for something in
the
Try setting your energy-level down, at 0 for instance. If it helps,
then increase until you find a happy medium.
On Dec 22, 2009, at 11:14 PM, Marc Orenberg wrote:
Hello. I've written an application using mod_conference which often
has two parties speaking at once and one party
Thanks Rob, thanks Jason.
I'm going to try this first thing tomorrow.
The energy-level paramter is described in the file as, Energy level required
for audio to be sent to the other users, so one would think that this would
have no effect if member-flags is set to waste, right?
No, from my understanding that's not how it works. Waste just means it'll
always send RTP packets, doesn't mean it will contain audio... so if you
have audio that's under your energy threshold, you still won't hear it.
Dan
On Wed, Dec 23, 2009 at 12:45 AM, Marc Orenberg m...@kasteris.com wrote:
On Tue, 2009-12-22 at 23:52 -0500, Michael Jerris wrote:
That being said, ulaw l16 alaw will cause degredation and any other
modifications such as volume adjustment in this path will make it
worse.
Indeed. Storing prompts
as 8k, 16-bit WAVs
makes a lot of sense.
[I am inordinately
My setup is as follows:
FreeSWITCH - SIP Trunk - PSTN.
From freeswitch, I'm making outbound calls using event socket via the
external profile. Except for the ext_rtp_ip and ext_sip_ip, everything is
default settings. Using playback application, I'm playing a mu-law audio.
I'm also starting the
On Wed, Dec 23, 2009 at 12:17 PM, David Knell d...@3c.co.uk wrote:
On Tue, 2009-12-22 at 23:52 -0500, Michael Jerris wrote:
That being said, ulaw l16 alaw will cause degredation and any other
modifications such as volume adjustment in this path will make it
worse.
Indeed. Storing
55 matches
Mail list logo
