Sorry for the potentially stupid question...
I'm trying to make sense of the standard kamailio.cfg file. At the
top, it says to remove the #n# comments to enable nat processing.
Looking down in the code, I see that one of the sections to be enabled
is route[4]. But, no where else in the file do I see that route(4) is
ever called.
Is the cfg file missing something, or is route(4) called in some way
that is not apparent from reading the file?
Thanks!
--
Mark Sidell
Partner
Forte, Inc.
919-942-7068
fax 919-969-2844
www.forteinc.com
Hello Guys,
Going to say thanks for any input ahead of time. My question is
currently we may have a scenario where we have a client using several UAC's
registered to the same username. Now what I've seen is that sometimes one
endpoint of User A may be NATed, another may be public. This seems to be
causing some call flow control issues in a sense for when we handle for NAT
/ vs NOT NAT, at the time of a request and all of this is in parallel being
handled.
Example Scenario.
User A calls User B -- (both internal).
User B has end points X (NAT), Y(PUBLIC), and Z(NAT).
So a call is being branched to X, Y, and Z in parallel. Now we
fix_nated_contact for endpoint X, next we go to Y, however we've already
fixed_nated_contact, thus this call is deemed bad, and then Z continues.
Any thoughts / examples / suggestions for these types of scenarios? This is
just an example one, one particular case where issues we have seen is more
like follows:
PSTN Calls User B (external to internal)
User B has the same previous end points (X,Y,Z) + (PSTN End Points that we
are adding in parallel to fork to).
Same type issue, same type scenario.
The end result of the issues are lack of audio, mis routed contacts, etc.
So any help and input is appreciated, thanks!
hi all,
is it possible to run openser and asterisks in one box,
I have googled but all the result I got have rewritehostport("X.X.X.X:5060) with a diffrent IP than the server IP,
I am assuming either they have 2 network card or refering to another host .
but will it be possible to put them togather? if so how,
thanks,
please help
Hello,
I'm using the set_dlg_profile function from the dialog module (kamailio 1.4).
I use it to control limit of simultaneous calls to clients.
I'm noticing that after this function is called, even if the call
terminates immediately, it takes some 3 to 4 seconds for the profile
to be cleared. Is there any reason for this?
I can see even if I call unset_dlg_profile on failure_route, the
profile will take that time span to be cleared.
regards,
mayama
Hi,
I'm a newbie of kamailio. I'm an Asterisk expert, and I want to explore the
"*SER" world, because I need a real SIP proxy. I planned kamailio project to
begin. I want the radius auth.
I have a question about a compile message. If I
try to compile kamailio-
1.5.0-tls, I always receive the message "cfg.y:
conflicts: 1 shift/reduce"
Before build:
# make distclean
# make mantainer-
clean
and then:
# make all
bison -d -b cfg cfg.y
cfg.y: conflicts: 1
shift/reduce
flex cfg.lex
The rest of build is OK, without errors.
Is this a
problem or is normal?
And if is not normal, how to avoid this problem?
I have
a
Debian Lenny
# uname -a
Linux voicelab 2.6.26-1-686 #1 SMP Sat Jan 10 18:29:
31
UTC 2009 i686 GNU/Linux
Uncomment:
Makefile:
TLS=1
FREERADIUS=1
Makefile.
vars
MODS_MYSQL=on
and
MODS_RADIUS=on
/usr/local/src/kamailio/kamailio-1.5.0-
tls/modules/acc/Makefile
ENABLE_RADIUS_ACC=true
Many thanks
Priz
Hi!
I am new to opensips, and I am looking for a powerfull sip gateway sollution, because asterisk isn't able to handle such thing.
My case:
I have a Grandstream VoIP Device, at which a DECT base with 2 cordless
phones are connected at 1 FX1 Port. If a call is placed and made through
one cordless phone the other cordless phone appears as busy.
What I want:
I want to place and receive as many calls at the same time through 1
1 FXS Port with one single SIP Account, and through this where the base
station is connected through.
Question:
Does kamilio have the ability for such things?!
For any help and advise I would thank you kindly
Tamer
Hi,
I am having a difficult time setting up a config file that works properly
for a simple dispatcher with failover, database authentication and
persistence.
Could someone send me a more extensive example than the one
that comes by default?
Thanks,
Alberto Furtado
Hello all,
i got bitten by the lumps mess again. I can understand the lump concept makes
a bit of sense if requests are just passing though a proxy, but nowadays
Kamailio obtains more and more UA functionality where it is completely
unappropriate.
How do I store a modified PUBLISH body? I change it with subst_body() before
calling handle_publish() but ofcourse the original body is stored by the
presence UA.
Is there currently a method to store a modified body? (apart from looping the
message back to myself)
Would implementing an apply_lumps() function or something like that be very
difficult?
--
Greetings,
Alex Hermann
Hi All;
I need to use switch in my ser.cfg file .But ser give me some errors .so i
need the correct syntax to do it
here is my ser.cfg
if ( method=="REGISTER"){
i = %fu % 2;
switch(i){
case"1":
route(1);
break;
case"0":
route(2);
break;
}
}
thank tou for your help
best regards
Hi all,
I need to get the INVITE sdp audio port in a script variable to be used in
an external command.
Is this possible without modifying the source code? I think it can't be done
with the TEXTOP module, can be?
Thanks in advance
Regards
--
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