and that it's not related to
my setup (I don't think so, but...), I'll open a jira ticket for the devs.
regards,
rod
Dmitry Bely a écrit :
I had a similar problem when I needed to talk to a gateway using g729
while g711 was used by default. The following works for me:
vars.xml
(...)
X-PRE-PROCESS
to
transcode G729 --- G711.
I was wondering if there is a way for FS to force the codec order on Leg
A with some knowledge of the preferred codec on Leg B, ie I know that
Leg B will always use G711 so that I want to biase the SDP answer on Leg
A based on this fact.
regards,
rod
Nandy Dagondon
.
Tried a lot of things (greedy for codec-negociation, late_codec,
disable_transcoding, codec-prefs) without success.
If you have some clue.
regards,
rod
Michael Collins a écrit :
Check out this page:
http://wiki.freeswitch.org/wiki/Codec_negotiation
Late negotiation will probably let you handle
are
not working.
So 2 questions:
- does application start_dtmf_generate requires transcoding
- if yes, can I set the variable disable-transcoding in my dialplan
regards,
rod
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you are right for the regex. This is part of an old setup, correct with:
extension name=PEER_01
condition field=${ROUTE_GW} expression=PEER_01
action application=set data=hangup_after_bridge=true/
Hristo Benev a écrit :
Hello Rod,
I did the change.
Here is extract
this great product ;-)
rod.
Hristo Benev a écrit :
I assume you asked for port 5062 since I do not have any traffic on 5060 (I
have one IP and my internal sip port is 5090 and external 5080).
If you need additional info I'll provide it.
Here is trace:
ngrep -d any -nn -i '1000' port
=${sip_redirect_contact_user_0}/
please correct with:
action application=set data=ROUTE_GW=${sip_redirect_contact_host_0}/
This should match PEER_01 in dialplan instead of trying matching
fra...@peer_01.
Let me know if this is right now.
rod
Hristo Benev a écrit :
Hello Rod,
I did
to deal with DTMF even if not
analyzing RTP for transcoding. My commercial SBC is doing this, but it
sucks and that's the last step before final migration to FS.
regards,
rod
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there is an option to check. Any pointers.
regards.
Anthony Minessale wrote:
if you enable mod_g729 you can use freeswitch normally with that g729
codec as long
as no transcoding is enabled (same passthru concept as proxy_media_mode)
On Fri, Mar 27, 2009 at 10:07 AM, rod kawa...@laposte.net
mailto:kawa
thanks Mathieu.
I setup an IRC account to give it a try.
Comme ça je pourrais t'embeter avec mes pbms :p
rod
Mathieu Rene wrote:
limit_hash uses a faster data structure then limit but works the same
way for tne end-user.
viens sur IRC si t'as des questions en francais =)
Math
On 17
Hi,
not too hard :p
but it's just a bad habit when I write in my native language (french). I
guess that this spelling is not too common for english speaker.
I'll do my best next time to write it correctly.
@tamas
you are right, we could use limit_hash the same way as limit when not
specifying
Hi,
running SVN r12638, I don't have access anymore to these 2 variables
after a SIP 302 message, using info application:
variable_sip_redirect_contact_user_0
variable_sip_redirect_contact_host_0
It was okay with SVN r12611.
regards,
rod
for PSTN without relying
If you still have pbm with limit and svn, pay attention to you dialplan :p
@Mathieu
I hope you didn't work on a virtual pbm, cause it seems to be a dialplan
misconfiguration. I'll let you know if I still have pbm. Thanks for your
help.
regards.
rod
Tamas Cseke wrote:
Hello
, 2009 at 10:14 AM, Anthony Minessale
anthony.miness...@gmail.com mailto:anthony.miness...@gmail.com wrote:
try now
On Wed, Mar 11, 2009 at 9:09 AM, rod kawa...@laposte.net
mailto:kawa...@laposte.net wrote:
ok,
almost perfect :p
using this in dialplan
and decrease depending on load, and the pbm doesn't appear.
Does anybody is using mod_limit and have encountered the same pbm.
I'm using latest svn: 12580.
regards,
rod
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...@anonymous.invalid;tag=tQ52gmv6NyN0D.
Remote-Party-ID: test
sip:00123456...@172.29.0.5;party=calling;screen=yes;privacy=off.
where 172.29.0.5 is the external IP used by FS to bridge the call in
this example (but I'm sure you already know this :p)
regards,
rod
Anthony Minessale wrote:
ok if you are up
to circumvent this.
Anthony, I tried many settings for origination_privacy and it seems to
do nothing on the RPID header. Any clue?
regards.
Anthony Minessale wrote:
On Wed, Mar 11, 2009 at 3:33 AM, rod kawa...@laposte.net
mailto:kawa...@laposte.net wrote:
thanks a lot Anthony,
I
bounty for 50$: Make RPID SIP
header optional
I'll add 150$ for this if I could manage RPID as described above.
Sorry to use mailing list for this, I'm unable to add a note on jira for
this bounty.
regards,
rod
rod wrote:
Hi David,
already tried this :p
the pbm is that this doesn' modify
for supporting this request :p
kokoska rokoska wrote:
rod napsal(a):
Hi all,
it seems there is no way to do this :(
It could be great to be able to:
- decide if RPID should be present or not in the B leg for an
outbound call
- make RPID header fully customizable with variables
and I'd like B-leg to match A-leg for anonymous call as stated in RFC,
anonym...@anonymous.invalid is the proposed way to handle anonymous
call, I'll add 50$ more for support of this last request.
thanks for your reactivity.
regards,
rod
Anthony Minessale wrote:
Latest SVN:
Send no extra
Hi David,
already tried this :p
the pbm is that this doesn' modify the RPID header, but it adds a new
one so that I have 2 RPID header in the SIP INVITE :(
rod
David Knell wrote:
Hi Rod,
You can set it directly:
action
application=set![CDATA[sip_h_Remote-Party-Identity=${caller_id_number
Dear list,
I'd like to rewrite the number in the Remote Party ID header and only in
this header.
ex: I'd like to prefix the caller ID with a prefix code (000 in this
example) in the RPID header :
From: Anonymoussip:anonym...@anonymous.invalid;tag=1208367
Remote-Party-ID:
Hi Brian,
if I use the function effective_caller_id_number with my INVITE, I get this:
From: Anonymous sip:00anonym...@172.29.0.5;tag=17geyFjX5p0gS.
this is not exactly what I'm looking for :p
rod
Brian West wrote:
Well this depends on how you're placing the call.. if its a standard
the A leg invite looks like this:
From: Anonymoussip:anonym...@anonymous.invalid;user=phone
it has been rewritten like this:
From: Anonymous sip:00anonym...@172.29.0.5
rod
rod wrote:
Hi Brian,
if I use the function effective_caller_id_number with my INVITE, I get this:
From: Anonymous
using these functions like this did nothing on the SIP INVITE packet :'(
seven wrote:
try
bridge
({effective_caller_id_name
=your_name,effective_caller_id_number=}sofia/b-leg)
On Mar 5, 2009, at 9:00 PM, rod wrote:
the A leg invite looks like this:
From: Anonymoussip:anonym
session_manager.session_table switch_core_session_t
channel-state
Its important to know that what you see in show channels and show
calls is just a DB query to sqlite, Those commands will go directly
in the core and list those sessions.
Math
On 3-Mar-09, at 2:07 AM, rod wrote:
Hi
.
regards,
rod
Michael Jerris wrote:
Could you please post this to jira along with a thread apply all bt of
a core file taken from the process with the stuck sessions.
Mike
On Mar 2, 2009, at 2:06 AM, rod wrote:
Hi All,
I ran some longer tests with FS 1.0.3 acting as an SBC
. So I'm
wondering where are these 22 sessions ?
FYI, FS has run flawlessly with 750 sim. calls with 25-30% free CPUs.
Successful call -- 5271434
Failed call --- 1554 (less than 0.03%)
regards,
rod.
complete SIPP summary:
-- Scenario
Hi,
the clue for sending fax is to use the originate command in the CLI:
originate sofia/example/1...@10.10.10.10 txfax(/path_to_fax_file)
this command will send the fax file via profile example to fax machine
100 reachable via 10.10.10.10
Hope this could help others :p
regards,
rod
I will update the wiki.
regards,
rod.
rod wrote:
Hi all,
I don't understand how to use the fax commands for sending a fax. In the
wiki I saw this:
extension name=test_txfax_stream
condition field=destination_number expression=^\*90012$
action application=txfax data
/
/condition
/extension
my question is how to specify the gateway/profile that will handle the call.
For a call I can use the bridge application like this, but for the txfax ??
action application=bridge
data=sofia/external/${destination_numb...@10.10.10.10/
regards,
rod
Hi,
how many static xml files did you create for your test ?
rod.
kokoska rokoska wrote:
Anthony Minessale napsal(a):
What does it look like if you serve the directory from the static xml
file out of curiosity.
Well, I write all user infos into static xml files loaded at startup
commands.
Maybe some things are missing and I will update as soon as I get my new
servers for reinstallation.
I have to cleanup the way it is displayed, cause it lacks some wiki rules.
If some would like to contribute, they are welcome.
http://wiki.freeswitch.org/wiki/SBC_Setup
regards,
rod
jay
provide advices for a high
performance LCR setup.
I subscribed to this list a long time ago, and my feeling is that FS is
a great piece of software with a great community, so that I decided that
it could be great to contribute.
regards,
rod
Saeed Ahmed wrote:
Hi rod,
It's really amazing
I did the test when I start looking at FS.
With 10 000 files in conf/directory/default mounted as a ramdisk (if not
in Ramdisk, the IO are too high) and an intel quad core q9550 (2.83Ghz)
with 4GB RAM and the db also in Ramdisk, I was stuck at approx 150cps
with a very high CPU usage. The
define metrics/properties
for a route (quality, cost, fax compliance...) in realtime, and this is
where using/enhancing the native FS module mod_lcr could be better (I
have no idea on how mod_lcr performs, I will give it a try).
rod
Adam Long wrote:
Hi Rod,
Great info, Thanks!
Glad to see
to resend the call to
a Kamailio server.
I think you can adapt this scenario to your perl script using variable
exportation and append_hf function.
rod.
Adam Long wrote:
Hi Rod,
Great info, Thanks!
Glad to see others are interested in the same concept.
My reasons for SER as routing core
/knowlegde.
The wiki page is there:
http://wiki.freeswitch.org/wiki/SBC_Setup
regards,
rod.
Adam Long wrote:
Hi Guys,
I’ve been working at setting up a couple of FreeSwitch nodes as a
topology hiding SBCs that handles both ingress traffic from my
providers/peers and pass traffic up
5483,17
The CPU is going from 89% idle to 0% in less than 2 seconds.
I know that I don't have to expect too much from this kind of hardware,
but it seems strange that the CPU power vanished so suddenly.
Thanks a lot for the guys that have read this long mail :p
kind regards,
rod
is going from 89% idle to 0% in less than 2 seconds.
I know that I don't have to expect too much from this kind of hardware,
but it seems strange that the CPU power vanished so suddenly.
Thanks a lot for the guys that have read this long mail :p
kind regards,
rod
,
rod
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Hi,
I upgraded today to 10999 with same results.
rod.
Michael Jerris wrote:
What revision of FreeSWITCH are you trying with? I would try with
current trunk, I have a suspicion we fixed the main issue your running
into.
Mike
On Dec 30, 2008, at 7:21 AM, rod wrote:
Hi all,
I
Thanks guys, it works.
Brian West wrote:
action application=export![CDATA[sip_h_Diversion=123456...@10.10.10.254
;reason=unconditional]]/action
/b
On Dec 22, 2008, at 9:55 AM, rod wrote:
Dear All,
I've been playing with the freeswitch options for one month now, and
I've been able
character FS will consider
this instruction instead:
action application=export
data=sip_h_Diversion=123456...@10.10.10.254;reason=
cause the quote after reason= is considered as a closing quote for data=.
Is there a way to achieve this.
Thanks.
rod
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