Hi all, I'm testing a FS server using sipp, I found that sipp only show the
retrans of sip packet, Does someone known is there a tool to test FS rtp
packet lost rate in high concurrent call env?
--
Lei.Tang
lei.tl...@gmail.com
___
FreeSWITCH-users
I found mod_distributor in SVN. I want to know how does it work ?
BG
Dome C.
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The simplest way I know of is to bring up another call from a local
phone and listen to the audio. At the same time run tcpdump/etc with
a strict filter to capture the rtp to/from that phone. You can then
run RTP stream analysis and the like in Wireshark to identify any lost
packets. While this
Hi,
I am using 1.0.4 version of freeswitch and I am doing proxy_media for all
calls. Basically, I just proxy all media from one gateway to another with
freeswitch serving as a middleman.
In the outgoing invite, I found that the owner line ( o= ) in SDP is showing
the originator's IP which I
It's also configurable on some phones...
As Brian said, welcome to VoIP! ;)
On Tue, Nov 10, 2009 at 5:23 PM, Brian West br...@freeswitch.org wrote:
That depends on the phone... some let you do it.. some don't...
WELCOME TO VOIP!!!
/b
--
Kristian Kielhofner
http://www.astlinux.org
You use OpenSER
/b
On Nov 11, 2009, at 11:08 AM, Juan Backson wrote:
Hi,
I am using 1.0.4 version of freeswitch and I am doing proxy_media
for all calls. Basically, I just proxy all media from one gateway
to another with freeswitch serving as a middleman.
In the outgoing invite, I
On Wed, Nov 11, 2009 at 7:49 AM, Dome Charoenyost d...@tel.co.th wrote:
I found mod_distributor in SVN. I want to know how does it work ?
It's brand new - I haven't even seen it yet. I will start documenting it
shortly. In the meantime if anyone else has started playing with it please
let me
see conf/autoload_configs/distributor.conf.xml
configuration name=distributor.conf description=Distributor
Configuration
lists
!-- every 10 calls to test you will get foo1 once and foo2 9
times...yes NINE TIMES! --
!-- this is not the same as 100 with 10 and 90 that would do foo1 10
hi
but this wouldn't work for larger volumens, g729 and t.38
or am i wrong on this?
br
On 2009-11-11 18:23, Kristian Kielhofner wrote:
This might be a bit too obvious but unless you have a specific reason
to use proxy_media (handling goofy codecs is a big one) you could just
set
Anthony,
Would this configuration work if we want to do load sharing 50/50:
#distributor.conf.xml
configuration name=distributor.conf description=Distributor Configuration
lists
list name=carrier1 total-weight=2
node name=gateway1 weight=1/
node name=gateway2 weight=1/
you got 2 out of 3,
the dialplan would look like this:
include
extension name=pstn
condition field=destination_number expression=^(.*)$
action application=bridge
data=sofia/gateway/${distributor(carrier1)}/$1/
/condition
/extension
/include
On Wed, Nov 11, 2009 at 1:19 PM, DJB
This is correct. The nathelper module and RTPProxy have an option to
rewrite o= as well as c=.
On Wed, Nov 11, 2009 at 12:29 PM, Brian West br...@freeswitch.org wrote:
You use OpenSER
/b
--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
Thank you. I will test with the latest SVN; however, can you please advise how
do I add it in modules.conf since I don't see the item in there so that I can
rebuild the source.
Regards,
Dorn B.
From: Anthony Minessale anthony.miness...@gmail.com
To:
Brian once told me (at ClueCon) that proxy_media isn't really that
much lighter on the CPU. At least that's what I think he said.
Anyone care to clarify/quantify?
Anyways it would work for G729 (pass through is no problem) but not
T.38 (it isn't a recognized codec at all).
2009/11/11 Christian
Perfect. I'll have it documented by the end of the day.
-MC
On Wed, Nov 11, 2009 at 10:32 AM, Anthony Minessale
anthony.miness...@gmail.com wrote:
see conf/autoload_configs/distributor.conf.xml
configuration name=distributor.conf description=Distributor
Configuration
lists
!-- every
Actually, I got it. I've added: applications/mod_distributor in the
modules.conf
I will start testing now.
Thank you,
Dorn B.
From: DJB djbin...@yahoo.com
To: freeswitch-users@lists.freeswitch.org
Sent: Wed, November 11, 2009 11:57:29 AM
Subject: Re:
Full log and trace here:
http://pastebin.freeswitch.org/11062
Pretty standard situation. User calls another user (same profile) and
tries to place the call on hold (RFC 3264/sendonly). FS places call
on hold and tries to start music but ends with:
[local_stream://moh] already
Hi!
I'm trying to register a SIP UA to my FreeSWITCH server and for some reason
I always get a 401 Unauthorized response. I've tried with other UA (X-Lite
and Ekiga) and they do work. The UA is: http://www.doddlephone.com
My user configuration is:
include
user id=doddle
params
param
From the trace:
#
2009-11-11 11:23:58.909804 [DEBUG] switch_ivr.c:540
sofia/pjsip/nob...@192.168.4.192 Command Execute
set(sip_h_X-voalte-call-id=9a072f8e-06cd-48e2-b7bd-2b2b8babb3ec)
#
EXECUTE sofia/pjsip/nob...@192.168.4.192
set(sip_h_X-voalte-call-id=9a072f8e-06cd-48e2-b7bd-2b2b8babb3ec)
#
Hello,
I have some problems with attended transfer and loopback
Scenario how id does work
- A calls B
- B enters *4 gets an announcement and enter digits for C (A get MOH)
- C is called
- As soon as C picks up the call, A and C are connected and B is dropped
How it should work until here:
- A
FYI, I added some docs here:
http://wiki.freeswitch.org/wiki/Mod_distributor
Please feel free to add to it if you are doing anything interesting or
creative that hasn't been covered.
-MC
On Wed, Nov 11, 2009 at 12:44 PM, Michael Collins m...@freeswitch.orgwrote:
Perfect. I'll have it
Also forgot to mention - this is trunk rev 15428 on CentOS 5 x86_64.
On Wed, Nov 11, 2009 at 5:20 PM, Kristian Kielhofner
kristian.kielhof...@gmail.com wrote:
From the trace:
..snip..
--
Kristian Kielhofner
http://www.astlinux.org
http://blog.krisk.org
http://www.star2star.com
Mathieu, thanks for the help. I got external oubound calls working.
The things are simpler then I expected.
This is my configuration:
extension name=domestic.test
condition field=${toll_allow} expression=domestic/
condition field=destination_number expression=^(\d{7,})$
action
$1 gives you the content of the first regex capture group, so the
first ( ) group.
^9(\d{7,})$ would put it in $1
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 11-Nov-09, at 2:33 PM, Sergey Kobzar wrote:
Mathieu,
condition field=destination_number expression=^(9\d{7,})$
...
But I see that 9 still exists.
condition field=destination_number expression=^9(\d{7,})$
Put the parentheses around the portion you want to capture.
http://wiki.freeswitch.org/wiki/Regular_Expression
--
Russell
Ah, right. I was inattentive :)
What about my 2nd question? Each user must have unique outbound number
which is mapped to his internal number.
How can I set ${outbound_caller_id_number} depending on calling
internal number?
Thursday, November 12, 2009, 12:41:22 AM, Mathieu wrote:
$1 gives
Set it in the user directory entry. All variables all loaded whenever
the user is authenticated (before the call hits the dialplan)
Mathieu Rene
Avant-Garde Solutions Inc
Office: + 1 (514) 664-1044 x100
Cell: +1 (514) 664-1044 x200
mr...@avgs.ca
On 11-Nov-09, at 3:04 PM, Sergey Kobzar
Your SIP UA needs to take the info in the 401 and use it to digest
authenticate. If you trace a SIP UA that supports authentication you will see
that they also get the 401/407 and only then are able to authenticate. This
is just a fact of how digest auth works in SIP ... see section 22.4 The
Hi everybody,
I have setup a FreeSWITCH IP-PBX for my office using a T1 and Redfone
foneBridge2, which uses Openzap, for my connection to the PSTN. I am trying
to figure out if it is possible to forward a call that comes in through the
T1/Openzap, back out to a PSTN number.
An example would be,
Adam Ford wrote:
I have also noted that I can simply bridge the call out another line on
the T1 through Openzap. However, that seems to tie up 2 lines just to
forward a call. This is not a desirable solution.
That's the way it has to work with any phone system, including your cell phone.
If
Alright. Thank you for your answer. I just had hoped there might be
something better that I didn't know about, after reading about deflect on
the wiki.
-Adam
-Original Message-
From: freeswitch-users-boun...@lists.freeswitch.org
[mailto:freeswitch-users-boun...@lists.freeswitch.org] On
deflect would work if the stack and your provider supported TBCT
/b
On Nov 11, 2009, at 7:43 PM, Adam Ford wrote:
Alright. Thank you for your answer. I just had hoped there might be
something better that I didn't know about, after reading about
deflect on
the wiki.
-Adam
FWIW: If you're getting your T1 from Qwest, they have an option called TnR
(Transfer and Release). You play a DTMF sequence followed by the destination
number. They take the call back and bridge it to the new number. Of course,
you pay to have the feature enabled and you pay per use. It's only
Wow. we can use FS for sip dispatcher :)
How to forward call in FS ? i mean 302 redirect not bridge ?
BG
Dome C.
2009/11/12 Michael Collins m...@freeswitch.org:
FYI, I added some docs here:
http://wiki.freeswitch.org/wiki/Mod_distributor
Please feel free to add to it if you are doing
Got it from wiki
action application=redirect data=sip:f...@bar.com /
Dome C.
2009/11/12 Dome Charoenyost d...@tel.co.th:
Wow. we can use FS for sip dispatcher :)
How to forward call in FS ? i mean 302 redirect not bridge ?
BG
Dome C.
2009/11/12 Michael Collins m...@freeswitch.org:
dont execute bridge that way, your bridge itself is the other thing already
broadcasting.
api uuid_transfer uuid of chan bridge:sofia/myprofile/f...@bar.com inline
if you want to do more after the bridge
set the variable park_after_bridge=true to make it go back to idle
On Wed, Nov 11, 2009
set/export the channel variable loopback_bowout=true so it's on the loopback
leg
On Wed, Nov 11, 2009 at 4:27 PM, Peter P GMX prometheus...@gmx.net wrote:
Hello,
I have some problems with attended transfer and loopback
Scenario how id does work
- A calls B
- B enters *4 gets an
hit send too soon
you want to set loopback_bowout=false
This keeps loopback from trying to destroy itself when it sees a chance to
cut out of the call path.
On Wed, Nov 11, 2009 at 10:11 PM, Anthony Minessale
anthony.miness...@gmail.com wrote:
set/export the channel variable
Hi, thanks Kristian for your answer, it make sense
2009/11/12 Kristian Kielhofner kristian.kielhof...@gmail.com
The simplest way I know of is to bring up another call from a local
phone and listen to the audio. At the same time run tcpdump/etc with
a strict filter to capture the rtp to/from
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