s way.
When I tried:
Everything went fine.
If others could check that absolute_codec_string doesn't work as
expected when used with application bridge 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
illing to use G711 and FS is unable 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
G729 <--> G711.
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
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:
Hristo Benev a écrit :
> Hello Rod,
>
> I did the change.
>
> Here is ext
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 the change.
>
> Here i
mber of concurrent calls per AREA,
PEER...
Thanks for "the good tutorial", but don't forget the dev team who did
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 intern
0' port 5060 -W byline
I will check what's happening.
Do you have an entry for 1000 in your mysql database ?
regards,
rod
Hristo Benev a écrit :
>
> It seems that the problem is on kamailio configuration.
> Will ask on their list.
>
> But to test i try to connect to my
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,
Hi Brian,
don't understand very well your advice:
--> Can't use proxy media in this case. (I highly recommend you not use
Proxy Media mode)
If i want to hide my topology network and deal with G729, I must use
proxy media ?
Why is Proxy media mode not recommended ??
regards.
rod
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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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 q
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
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
al 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,
>
> we had the same problem. we couldn't test r12581 for sure yet, but we will.
> This fix is only for the limit (db ve
m using latest svn: 12580.
regards,
rod
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from url
>
>
> On Wed, Mar 11, 2009 at 10:14 AM, Anthony Minessale
> mailto:anthony.miness...@gmail.com>> wrote:
>
> try now
>
>
> On Wed, Mar 11, 2009 at 9:09 AM, rod <mailto:kawa...@laposte.net>> wrote:
>
>
ed 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 <mailto:kawa...@laposte.net>> wrote:
>
> thanks a lot Anthony,
>
> I consider the work is done
: "test"
;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 date you should be able to add
>
> {sip_fr
.@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 caller id info:
> {sip_cid_type=none}sofia/default/u...@exa
s 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
>>
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
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:
> application="set">
>
>
> --Dave
&
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: "Anonymous"
it has been rewritten like this:
From: "Anonymous"
rod
rod wrote:
> Hi Brian,
>
> if I use the function effective_caller_id_number with my INVITE, I get this:
>
> From: "Anonymous" ;tag=17geyFjX5p
Hi Brian,
if I use the function effective_caller_id_number with my INVITE, I get this:
From: "Anonymous" ;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
> b
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: Anonymous;tag=1208367
Remote-Party-ID:
;privacy=full;screen=yes;party=cal
too:
>
> list_sessions
>
> hash_it_str_x 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
g GDB.
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,
>>
>>
show calls", I see nothing. 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%)
rega
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,
s time, if I'm successful 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:
>
>
>
>
>
>
>
>
>
r the txfax ??
regards,
rod
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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
your FS servers, take a look at the dispatcher module
Using that, the FS server receiving the Invite, just need to parse the
X-ROUTE header and route the call, without having to resend the call to
a Kamailio server.
I think you can adapt this scenario to your perl script using variable
exportatio
can't 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, Th
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 versio
new LCR module, and I
think that Ken Rice on this list can 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.
reg
the linux 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
x27;m a very bad programmer, so that's the only way for me to contribute
to FS, and as I see many people interested for an SBC setup, I think it
could be great if we share our work/knowlegde.
The wiki page is there:
http://wiki.freeswitch.org/wiki/SBC_Setup
regards,
rod.
Adam Long wrot
larly based on what I did.
Saeed, why are you replacing your Nextone, it's said to be one of the
best commercial SBC on the market.
regards.
Saeed Ahmed wrote:
> Thanks rod for a quick answer,
>
> FS is installed on Ubuntu Server.
>
> I am planning to replace Nextone SBC wit
_XML
maybe you could tell more about the linux distribution you're using so
that I can give you some pointers for sipp...
regards.
rod.
Saeed Ahmed wrote:
> Hi Rod,
>
> Could you please share how you configured Sipp & FS to create a test
> environment? Especially the dial pla
:54 [ERR] switch_core_io.c:117
switch_core_session_read_frame() sofia/external/9...@10.10.20.100 has no
read codec.
regards,
rodrigue
rod wrote:
> Hi Ken, Jay,
>
> thanks for pointing to proxy media, I will test.
>
> Ken, you are right, I was brain damaged (a stupid mistake) when sett
Hi Ken, Jay,
thanks for pointing to proxy media, I will test.
Ken, you are right, I was brain damaged (a stupid mistake) when setting
INFO cause this kind of level could be very verbose. I'm switching to
CRIT or ERR.
Thanks guys,
rod.
thanks for
Ken Rice wrote:
> If you don&#
I can switch off this kind of logging. I
thought it should be in /dialplan/internal.xml, but I see that in
internal.xml ->
thanks a lot for your suggestion.
regards,
rod
Ken Rice wrote:
> Dont forget there are several things you can do to increase performance...
>
> 1) where pos
S
process).
I understand that FS1 server is not the best hardware platform, and I'm
waiting for new 4 cores server for testing.
I will update those numbers when testing with the new hardware.
regards,
rod.
Anthony Minessale wrote:
> Which of the 2 machines has the load issue? You s
67,330,000,996,93
0,000,00 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
,750,00 67,330,000,996,93
0,000,00 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
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, ro
y for failover).
regards,
rod
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Thanks guys, it works.
Brian West wrote:
>
>
> /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 to use it with kamailio workin
Dear All,
I've been playing with the freeswitch options for one month now, and
I've been able to use it with kamailio working as a registrar.
What I'd like to do is to add a diversion header using the following
action in the dialplan:
Please note, that I'd like to put the word "unconditio
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