Hi
I am using homer and managing multiple SIP domains. My problem is that in
the sip_capture table, the FROM and TO domains are not logged in the FROM
and TO fields. The MSG field contains the domains, but the queries
involving domains become tricky and heavy to execute.
Is there a way I can have
Thanks Richard for the tip :)
Reda
On Sun, Jun 24, 2012 at 7:28 PM, Richard Brady wrote:
> Hi Reda
>
> A bit late for a reply but I found your post recently and it helped me
> to solve a similar problem, so I wanted to share one possible
> solution.
>
> On 21 January 2
I think in such cases you get a "parse error" in the log. Do you?
If yes, then Kamailio stops processing the packet and no reply is sent.
Reda
On Tue, Jun 19, 2012 at 7:10 PM, Uri Shacked wrote:
> OK understood.
> I tried with no "TO" header as well, and still no error reply
> Is there su
I second that.
Siptrace doesn't log everything, but workarounds can be done, such as using
a combination of setting a flag and using the sip_trace( ) function, and
using them as well from the onsend_route for some outgoing messages, with
careful handling to prevent duplicate logging.
Reda
On T
DBRWUSER and DBROUSER are users created by kamdbctl to access the kamailio
database. They should be like kamalioro and kamailiorw or whatever you
want, but not root. Since you set them to root, it seems the root password
is overwritten.
Reda
On Tue, May 15, 2012 at 7:20 PM, Tim King wrote:
>
od perfectly in kamailio.cfg, now I need to add it in Siremis..
>
> Any idea , how I can add it in?
>
> Thanks
>
> Best regards
>
> 2012/5/15 Reda Aouad
>
>> You can check on domain or IP which you can find in the following pseudo
>> variables
>>
>>
stion
> before that, about sdpops:
>
> Is it possible to remove codecs ( yeah working well on 3.2 ) but
> specifically from an domain or IP .. ?
>
> And how?
>
> Thanks in advance
>
> Best regards
>
> 2012/5/15 Reda Aouad
>
>> Can't help anymore on t
r way to use it with the 3.2.3
>>> version of Kamailio? Cause I'm doing my final project of my bachloor, and I
>>> really need to use it for.
>>>
>>> Is it possible to change something in the code or.. to make it working
>>> well with my versio
o make it working
> well with my version?
>
> Best regards
>
>
> 2012/5/15 Reda Aouad
>
>> Hi,
>>
>> This function is available in Kamailio 3.3 (currently in dev stage) and
>> not 3.2.
>>
>> Reda
>>
>>
>>
>> On Tue, Ma
Hi,
This function is available in Kamailio 3.3 (currently in dev stage) and not
3.2.
Reda
On Tue, May 15, 2012 at 11:49 AM, Grégoire Vandendeurpel <
g.vandendeur...@gmail.com> wrote:
> Hello,
>
> I use Kamailio 3.2.3 and I used the command from the sdpops module "
> sdp_remove_media("video");
Hi,
Try this function from nathelper module: fix_nated_sdp
http://kamailio.org/docs/modules/stable/modules_k/nathelper.html#id2533354
Reda
On Tue, May 15, 2012 at 10:54 AM, Openser Kamailio <
kamailioopen...@gmail.com> wrote:
> hi,
>
> I try to change the IP address in connection information
> On 5/14/12 9:42 AM, Klaus Darilion wrote:
>
>> maybe tm module is not loaded before dialog module?
>>
>> On 13.05.2012 12:35, Reda Aouad wrote:
>>
>>> Hi,
>>>
>>> After upgrading from 3.2 to 3.3, I am getting the following error:
>>&
Hi,
After upgrading from 3.2 to 3.3, I am getting the following error:
ERROR: dialog [dlg_handlers.c:937]: registering TMCB to wait for
negative ACK
This is my version: kamailio -V
version: kamailio 3.3.0-pre1 (i386/linux) 275c8a
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_
which instantly drops my connection is
> 8033 !! even if netstat shows its listening to it on all interfaces.
>
>
> On Thu, May 10, 2012 at 11:17 AM, Reda Aouad wrote:
>
>> In that case it maybe a network or security problem.
>> Can you ping the remote server? Have yo
done
> that.,
> When I do "netstat -pln | grep kamailio" I can see kamailio listening on
> 0.0.0.0:8033 but still nothing happening :(
>
> Please let me know what else should I provide in order to resolve this
> issue.
>
> Regards,
> Sammy.
>
> On W
Hi,
The Kamailio server you wish to control remotely should have the
mi_datagram module listening on the correct interface and not on the
loopback one (127.0.0.1).
loadmodule "mi_datagram.so"
*modparam("mi_datagram", "socket_name", "udp:192.168.2.156:8033")*
Reda
On Wed, May 9, 2012 at 1:02
I had the same problem when calling mediaproxy twice by mistake.
rtpproxy_manage( ) calls implicitely rtpproxy_offer( ). This is the problem.
Either you use only rtpproxy_manage once on the INVITE and let it start and
terminate the session, or you use rtpproxy_offer, rtpproxy_answer and
rtpproxy_
>
> SIP user > SIP server > external auth script > OpenSSO server
>
> Thank you
>
>
> On Fri, May 4, 2012 at 5:56 PM, Reda Aouad wrote:
>
>> Sorry didn't reply to mailing list before. Emails are below.
>>
>> SHA1 encryption ma
FLD_TRACE)
modparam("siptrace", "duplicate_uri","sip:127.0.0.1:9060")
modparam("siptrace", "hep_mode_on",0)
modparam("siptrace", "trace_to_database", 0)
I get the following error:
ERROR: sipcapture [sipcapture.c:642]: ERROR: sip
nts), it gives the error above.
Is there a way to tell sipcapture to bind only on one interface using HEP
mode? It seems impossible for now to run NODE and AGENT on the same server
using HEP mode...
Thanks for helping.
Reda
On Mon, Apr 16, 2012 at 12:00 AM, Alexandr Dubovikov wrote:
> Hi
Sorry didn't reply to mailing list before. Emails are below.
SHA1 encryption may not encrypt the same way as HA1 (HA1 = MD5 of realm +
username + password), so the problem may be here.
I suggest you store your passwords as clear text in LDAP for testing first.
Reda
On Fri, May 4, 2012 at 11:14
in the line
if (!pv_www_authenticate("$td", "$avp(password)", "0")) {
write avp(s:password) instead of avp(password)
not sure it will solve it though.. if it doesn't, maybe others can help you
more on this.
Reda
On Fri, May 4, 2012 at 5:50 PM, Saul Waizer wrote:
> Hello Reda,
>
> Thank you f
Hi Saul,
username_avp_spec was previously a AUTH module parameter to specify a
variable that was passed to pv_www_authorize implicitly (the function
doesn't take arguments). Now you should use the new pv_www_authenticate and
pass to it explicitly the credentials as arguments.
So forget about user
Well 5xx error codes are processed in failure_route and not onreply_route.
Not sure though if you can append_hf there.. Give it a try.
FYI, for using a function in event_route (seems not possible for append_hf):
event_route[tm:local-request] {
...
append_hf(...);
...
}
Reda
On 26 avr. 20
t; *From:* sr-users-boun...@lists.sip-router.org [mailto:
> sr-users-boun...@lists.sip-router.org] *On Behalf Of *Reda Aouad
> *Sent:* 26 April 2012 14:34
>
> *To:* **SIP Router - Kamailio** (OpenSER) and SIP Express Router (SER) -
> UsersMailing List
> *Subject:* Re: [SR-Users] di
Hi,
In which route did you try append_hf ?
"event_route[tm:local-request]" is the route executed for locally generated
requests, I think you should be trying to modify it there.
Reda
On Thu, Apr 26, 2012 at 16:10, Vitaliy Aleksandrov
wrote:
> Hi all,
>
> I have Kamailio connected to PSTN gate
Hi,
@Carsten
Dispatcher algorithm 0 based on call-id should do it in your case of
re-invite within dialog with same call-id.
@Charles
In the case of attended transfers, shouldn't both media servers be relaying
media between them? I didn't understand why your are obliged to dispatch to
the same me
Hi,
You can get inspired by this
http://www.kamailio.org/docs/openser-performance-tests/
Reda
On Wed, Apr 18, 2012 at 20:39, Lucas Alvarez wrote:
> Can someone recommend a good stress tool for testing kamailio? Or can
> point through the right path?
> Thanks in advance.
>
>
> Lucas
>
>
Hi,
Do you have any client that is sending a corrupt request to the "AddPac SIP
Gateway" at 190.22.140.170, so that this gateway is replying "400 bad
request" ? Maybe you could resolve this problem at the source..
If it's not the case, you can send an email to the owner of the IP address.
A quick
Well, 30 seconds seems to be the timeout of your SIP client: I suspect your
client sends a 200OK but doesn't receive the ACK for it, and hence times
out 30 seconds after and terminates the call. You can run ngrep to trace
the call and see where the ACK is routed, and if it is routed to the
correct
Can you try another algorithm?
Can you set debug=3 and see what's happening?
Reda
On Mon, Apr 16, 2012 at 14:54, Karsten Horsmann wrote:
> Hi Reda,
>
>
> there is no error in the log file. Its hard to debug this function.
>
> 2012/4/16 Reda Aouad :
> > Hi,
>
Hi,
When you use forward( ) it loops itself since the dispatcher function is
not selecting a destination, and the SIP request is destined to your
server. Do you have any error in your log file?
Reda
On Mon, Apr 16, 2012 at 14:31, Karsten Horsmann wrote:
> Hi Carsten,
>
>
> its still the same
Hi,
I'm trying to setup Kamailio as a Homer capture server itself.
Kamailio listens on 2 ports : 53 for SIP users, and 9060 for homer SIP
capture.
siptrace module duplicates packets to sipcapture module on port 9060.
I'm getting the following error :
ERROR: siptrace [siptrace.c:1669]: invalid por
Thanks :D
It looks like a good interface. I just installed it and will start testing
it.
Reda
On Sun, Apr 15, 2012 at 18:53, Norm wrote:
> Take a look at Homer.
>
> Reda Aouad wrote:
>
> >Hello,
> >
> >Anyone knows a tool/software (NOT CDRTool) to graphically
Hello,
Anyone knows a tool/software (NOT CDRTool) to graphically visualize the SIP
traces recorded in database by Kamailio's siptrace module?
Thanks
Reda
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Hi.
I don't know what your problem is, but here is a more elegant way to do it.
Instead of manually rewriting $ru, try the following function which
achieves the same result:
rewritehostport("ip:port");
Then route(relay).
Reda
On 14 avr. 2012, at 02:56, Akan Technology wrote:
> hello,
>
> I
Hi,
Look at this function from the textopsx module
http://kamailio.org/docs/modules/stable/modules/textopsx.html#msg_apply_changes
Try to apply it after your sanitize and before you challenge the REGISTER.
Reda
On Thu, Apr 12, 2012 at 23:45, Fabian Borot wrote:
> greetings,
>
> I have a c
; Cheers,
> Daniel
>
>
> On 3/12/12 11:30 AM, Reda Aouad wrote:
>
> Hi Daniel,
>
> Any plans to backport this to 3.2 ?
> I could still do the changes manually before compilation if you don't have
> time to do it.
>
> Thank you again
> Reda
>
>
>
&g
Are you searching for examples of serial / parallel forking config found in
the TM module documentation ?
Reda
On Tue, Mar 27, 2012 at 16:55, Carsten Bock wrote:
> Hi Uri,
>
> you're wrong.
> "t_on_branch" works before the request is sent out
> The branches work in the manner, you program
How do you reload your htable? Is it using the exec( ) function to call the
mi command sht_reload? Or do you have another method?
Thanks
Reda
On Mon, Mar 26, 2012 at 14:56, Uri Shacked wrote:
> Hi,
>
>
>
> I made these kinds of tests before. I have two tips for you to pay
> attention to:
>
>
Congratulations Daniel and all the Kamailio team and developers !
Thank you all for this great community :D
Reda
On Fri, Mar 23, 2012 at 15:33, Daniel-Constantin Mierla
wrote:
> Hello,
>
> ITSPA UK has unveiled the winners of its 4th annual Awards, an event
> designed to celebrate innovation
One reason I didn't use sems is that it doesn't support G729 as a
media server. Did it change?
Reda
On 20 mars 2012, at 10:36, Juha Heinanen wrote:
> Reda Aouad writes:
>
>> But servers cost money to buy, host, and maintain.
>> This is why I would really love
Thanks Juha for the idea.
But servers cost money to buy, host, and maintain.
This is why I would really love to see such functionality in Kamailio.
May I suggest that for the next release ?
Reda
On Tue, Mar 20, 2012 at 10:12, Juha Heinanen wrote:
> Reda Aouad writes:
>
> > One
party will most likely hang up, the dialog is
terminated and finally we loose only a few seconds of billing.
Reda
On Tue, Mar 20, 2012 at 10:02, Reda Aouad wrote:
> Hello,
>
> In SIP, session timers can be used to periodically ping the UAC (using a
> re-INVITE or UPDATE) to know if
Hello,
In SIP, session timers can be used to periodically ping the UAC (using a
re-INVITE or UPDATE) to know if it's alive or not. Then action can be taken
- terminating the call.
Kamailio has the SST (SIP Session Timer) module which only enforces a
minimum session timer value for UACs, but not a
llo,
>
>
> On 3/13/12 7:08 PM, Reda Aouad wrote:
>
> Hi,
>
> Can we call MI commands from the script ?
> I'm thinking about reloading dispatcher or hash table from database for
> example..
>
> you can use exec module to run kamctl commands.
>
> Also,
prevent this from happening??
> I really don't know from where ekiga is getting that ip!! 0.0.0.0
>
> Regards,
>
> Vineet Menon
>
>
>
>
>
> On 14 March 2012 15:20, Reda Aouad wrote:
>
>> Your other UA at 33.236 is also broadcasting a SIP NOTIFY
>> ne
Your other UA at 33.236 is also broadcasting a SIP NOTIFY
neighbor@0.0.0.0in packets 1-4.
It seems Ekiga is not binding to the correct interface...
I suspect more a network problem than a SIP one.
Reda
On Wed, Mar 14, 2012 at 10:24, Vineet Menon wrote:
> Hi,
>
> I just installed kamailio for
e call hang up. Do you know if it's
> possible and how ?
>
> thank you again for your help
>
> Ravi
>
>
> Le 13/03/2012 23:56, Reda Aouad a écrit :
>
> and this also :
>
> http://kamailio.org/docs/modules/stable/modules/tm.html#max_inv_lifetime
>
> R
and this also :
http://kamailio.org/docs/modules/stable/modules/tm.html#max_inv_lifetime
Reda
On Tue, Mar 13, 2012 at 20:54, Reda Aouad wrote:
> You have two options :
>
> - http://kamailio.org/docs/modules/stable/modules/tm.html#fr_inv_timer
> -
> http://kamailio.org/docs
You have two options :
- http://kamailio.org/docs/modules/stable/modules/tm.html#fr_inv_timer
-
http://kamailio.org/docs/modules/stable/modules/tm.html#restart_fr_on_each_reply
Hope it helps if you didn't already tried them.
Reda
On Tue, Mar 13, 2012 at 19:14, Ravi DHIRAJLAL wrote:
>
>
>
Hi,
Can we call MI commands from the script ?
I'm thinking about reloading dispatcher or hash table from database for
example..
Thanks
Reda
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http://
27;t replace the old registration. and save("location",
0x04) doesn't take into account the user agent.
I think I should do it manually...
Reda
On Sat, Mar 3, 2012 at 12:20, Daniel-Constantin Mierla wrote:
>
>
> On 3/2/12 1:53 PM, Reda Aouad wrote:
>
> Thank you al
amailio and throw out the OpenSIPs
stuff…
From: Reda Aouad
Reply-To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -
Users Mailing List"
Date: Fri, 9 Mar 2012 16:41:57 +0100
To: "SIP Router - Kamailio (OpenSER) and SIP Express Router (SER) -Users
Mailing List"
horized.
The Aastra / X-Lite are NAT… It's possible that either I need to look at
nathlper closer – or that I misconfigured something somewhere.
I guess I'm not fluent enough in the configs and options to get a good
config.
From: Reda Aouad
Reply-To: "SIP Router - Kamailio (O
Hi Robert,
I had the same feeling about O***sips community.
I have a similar setup. A proxy that listens on port other than 5060,
forwarding to other servers on 5060.
If you can better define better your problem with your setup, maybe we can
help you better.
Is your server listening only on
ar modules for that.
Reda
On Fri, Mar 2, 2012 at 13:16, Daniel-Constantin Mierla
wrote:
> Hello,
>
>
> On 3/2/12 1:01 PM, Marius Zbihlei wrote:
>
> On 03/02/2012 12:44 PM, Reda Aouad wrote:
>
> Hi,
>
> Is there a way to ensure single-registration per user-agent
Hi,
Is there a way to ensure single-registration per user-agent for a user,
which overwrites previous registration ?
Or is there a way to limit the number of registrations per user, but
overwriting the earliest registration for each new one ?
Thanks,
Reda
_
Hi,
insert_hf doesn't work for CANCEL messages. It doesn't work either on 487
replies in failure_route..
Why? Is it because they are re-generated on every hop?
Is there a workaround?
Reda
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SIP Express Router (SER) and Kamailio (OpenSER) - sr-users m
:)
Reda
On Wed, Feb 29, 2012 at 09:58, Reda Aouad wrote:
> A quick grep on flags FL_* in the sources shows the following :
> Flag 29 is used by acc module, 31 by nat_traversal, 0 to 12 in the parser.
> Thus I assume that it is safe to test using flag 28.
> I'll keep you posted o
to (1<<29) and see if all works
> fine.
>
> On another hand, defining and using core msg flags in a module is a risk,
> a different solution has to be done, a simple one is to move the definition
> of these flags in the core, so there will be no overlap in the futur
The mediaproxy module is loaded, but no function from the module is called
at all.
Thanks
Reda
On Sat, Feb 25, 2012 at 18:58, Reda Aouad wrote:
> I looked in detail in the source of the call control module and didn't
> find anything related to the mediaproxy module neither..
> I
ources didn't reveal a point where
> mediaproxy is engaged.
>
> Can you load debugger module and enable cfgtrace, then run such a scenario
> and send out the output from cfgtrace to see all the config actions
> executed?
>
> Cheers,
> Daniel
>
>
> On 2/24/12 7:56
ot be big deal if it is all you are looking for -- I can look over
> it and send a patch if you are going to help testing it. I cannot do it
> these days, though, being out of the office.
>
> Cheers,
> Daniel
>
>
> On 2/23/12 8:59 PM, Reda Aouad wrote:
>
> First, I a
d wrote:
> >
> >> Reading from the module docs its clear why it needs to engage media/rtp
> >> proxy to start,stop billing or timer of a call. so what happens when it
> >> engages mediaproxy on unwanted calls !? audio-issues?
> >>
> >&
> engages mediaproxy on unwanted calls !? audio-issues?
>
>
> On Thu, Feb 23, 2012 at 1:21 PM, Reda Aouad wrote:
>
>> Thanks Sammy. I didn't get any reply yet.
>>
>> CallControl is an application used with CDRTool for prepaid calls. It
>> calculate
your problem is complicated for me atleast. I hope somebody could answer
you accurately and precisely.
btw, what are you using in real? opensips or kamailio, which version? and
in what context you need to use the call_control function?
Thanks,
Sammy
On Thu, Feb 23, 2012 at 12:45 AM, Reda Aouad wrot
Hi,
When I use the function call_control( ) of the call_control module, it
automatically engages mediaproxy if it finds the mediaproxy module loaded.
If the mediaproxy module is not loaded, call_control doesn't even try to
engage it.
I need mediaproxy for NAT traversal in some cases, but don't wa
Hi,
I noticed that calling call_control( ) function forces engage_media_proxy(
).
Is there a way to disable this behavior ?
Thanks
Reda
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Is this normal on restarting Kamailio ?
ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket
/tmp/kamailio_ctl: Operation not permitted (1)
RA
___
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Problem solved.
I included /etc/freeradius/radiusclient-ng/dictionary.sip in freeradius'
radiusd.conf file:
$INCLUDE/etc/freeradius/radiusclient-ng/dictionary.sip
The missing attributes in dictionary.kamailio were defined in
dictionary.sip.
Reda
On Sat, Feb 4, 2012 at 23:27, Reda
Hi,
I am trying to get Kamailio to work with CDRTool. In the step of installing
freeradius and adding kamailio's dictionary dictionary.kamailio to
freeradius, I get the following error when starting freeradius -X, which
fails to start:
Error: Errors reading dictionary: dict_init: No ATTRIBUTE
Hi,
My question concerns Call Control.
When a SIP call's timer ends, Call Control uses dlg_end_dlg MI command of
the dialog module to send a BYE to the UA.
Can I intercept the BYE message, drop it, change it and send instead an
INVITE or a REDIRECT using (maybe) the UAC module?
The INVITE/REDIREC
Hello,
Add an alias to your config file
alias=domain.com:5060
http://kamailio.org/dokuwiki/doku.php/core-cookbook:3.0.x#alias
RA
On Mon, Jan 23, 2012 at 16:06, Stoyan Mihaylov
wrote:
> Hi,
> If I am using IP address, I have no problems. If I use domain name
> (pointing to same IP address) th
y one suffering from this
type of configuration?
(Part of the problem is also tied to dumb ALG NAT routers which try to
out-smart SIP servers, without which I wouldn't run Kamailio on multiple
ports, and life would be much easier)
RA
On Thu, Jan 19, 2012 at 01:00, Reda Aouad wrote:
> O
ki/cookbooks/3.2.x/core#branch_route
> and set the Record-Route header there, for each individual branch.
>
> Regards,
> Ovidiu Sas
>
> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com
>
>
> On Wed, Jan 18, 2012 at 5:37 PM, Reda Aouad wrote:
> > Thank you
per
> Record-Route header.
>
> Regards,
> Ovidiu Sas
>
> --
> VoIP Embedded, Inc.
> http://www.voipembedded.com
>
>
> On Tue, Jan 17, 2012 at 3:01 PM, Reda Aouad wrote:
> > I just tried the record_route_advertised_address("public_ip").
> > It
Hi,
I am using the nat_keepalive function of the nat_traversal module, using
the OPTIONS message, and my code is the following :
if (is_method("REGISTER"))
nat_keepalive();
I see that Kamailio still sends OPTIONS packets even if the client
unregisters itself.
It stops later, but I don't know
I just tried the record_route_advertised_address("public_ip").
It doesn't add the port number of the outgoing socket.
Any suggestions?
RA
On Mon, Jan 16, 2012 at 15:57, Reda Aouad wrote:
> I know about record_route_advertised_address("ip:port") function. If
> Hi,
>
> On 01/16/2012 03:41 PM, Reda Aouad wrote:
> > I suggest that the function record_route( ) takes a public IP address as
> > a parameter, still doing what it does (correct record routing and cookie
> > addition did=xxx and loose route lr=on), but only replaci
Hello,
I am trying to implement the following configuration :
- Kamailio as a SIP proxy/registrar behind a one-to-one NAT (port number is
not modified) listening on ports 5060 and 53 (and more ports in the future)
- aliases correctly configured :
alias= udp: public_ip:53
alias= udp: publi
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