Can someone recommend a good stress tool for testing kamailio? Or can
point through the right path?
Thanks in advance.
Lucas
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(cfg_get.box02.gw_ip) + ":" +
$sel(cfg_get.box02.gw_port);
}
The problem I'm having is I'm not being able to do blind tranfers. I think
the cause is the prefix that remains in the TO field. After rewriting the
TO field nothing change. I would appreciate if someone could point me to
I want rewrite $tU but I'm not being able, I'm doing the following:
remove_hf("To");
insert_hf("To: sip:$rU@$rd\r\n", "From");
Then I'm printing $tU and it is still having the previous value, any help
will be a
I want rewrite $tU but I'm not being able, I'm doing the following:
remove_hf("To");
insert_hf("To: sip:$rU@$rd\r\n", "From");
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I have Kamailio 3.2.0 between two asterisk servers, after the call set, one
of the kamailio send the OK from the INVITE and the return ACK of that
message was discarded. This makes asterisk hangup the call after 5 secs.
It's that right?
OK message:
U 172.25.249.15:5060 -> 172.25.249.14:5060
SIP/2
t
the video media would be peer to peer and the audio follows the path
through asterisk.
Thanks in advance,
Lucas Alvarez
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at 5:04 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
>
> On 11/10/11 4:45 PM, Lucas Alvarez wrote:
>
> Hi, sorry for spamming again but I'm not being able to register to a SIP
> provider populating the uacreg table, I followed the steps of this guide:
> http://by-mico
to the sip
provider, is something missing? How can I know if the values from the table
are being loaded? What are the fields: "l_uuid" and "l_username" for? Is
for routing incoming calls? I would appreciate any kind of help.
Regards,
Lucas Alvarez
_
ow if the values from the table are being loaded? I
would appreciate any kind of help.
Regards,
Lucas Alvarez
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After restarting kamailio, there isn't any packet sent to the voipprovider,
how can I debug this? I'm stuck here.
Thanks in advance,
Lucas Alvarez
On Tue, Nov 8, 2011 at 6:11 PM, Lucas Alvarez wrote:
> Thanks Alex for you quick answer, I'm using kamailio 3.1.3 and I'
: +1-404-961-1892
> Web: <http://www.evaristesys.com/>http://www.evaristesys.com/
>
> On Nov 8, 2011, at 4:06 PM, Lucas Alvarez wrote:
>
> Hi, I have properly populate the table "uacreg" as says here:
> <http://by-miconda.blogspot.com/2010/10/best-of-new-in-kama
r.
Thanks in advance,
Lucas Alvarez
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}
exit;
break;
}
}
# http ops
xhttp_reply("200", "ok", "text/html",
"OK:
$si:$sp");
exit;
}
#!en
Hi, I've enabled xcap server in kamailio 3.1.3 and I'm stuck with this
error:
0(5073) ERROR: [db.c:408]: invalid version 0 for table subscriber
found, expected 6 (check table structure and table "version")
I've already checked the version table and the table_name = 'subscriber'
has table_versio
Hi, how can I do to filter the unregister messages? method == "REGISTER" &&
???
Thanks ins advance.
Lucas
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Hi, how may I catch the reply of an invite? I did this but it seems is not
working:
if (method=="INVITE") {
$avp(sdp_ret) = fix_nated_sdp("11");
xlog("L_ALERT","### SDP
return: $avp(sdp_ret)\n");
Hi, I want to modify the following sip packet, I want to change the SDP IP
address by $di, which rule could filter this packet? Something like: [ if
method("OK") and $di is RFC1918 ] ??? And with which function could I change
the Ip address in the SDP content? In this case I would like to change
1
address of the
sip phone. I have been with this since a couple of days and I would
appreciate a little help.
Thanks ons advance,
Lucas
On Tue, May 31, 2011 at 2:45 PM, Alex Balashov wrote:
> On 05/31/2011 01:40 PM, Lucas Alvarez wrote:
>
> Hi, I'm having a problem with RTP traffic
Hi, I'm having a problem with RTP traffic, I have softphone connecting from
the outside, I can register but Kamailio is sending all RTP traffic to the
private IP address that the softphone has in it own local network. How can I
fix this? Any help will be appreciated.
Lucas
92.168.15.231:13618
;branch=z9hG4bK-d8754z-ad5e03279628ca35-1---d8754z-;rport=13674;received=190.244.125.41.
To: "Lucas";tag=065c4e5ee3769aabfc84958a193ed91c-3b57.
From: "Lucas";tag=bd13120b.
Call-ID: Y2M0MTQ4YjFhNmM0NTZjMzAzYWE1OGQwZjViYzJlNDc..
CSeq: 2 REGISTER.
Server: kamailio
Hi, if I want to register clients from outside of my local network, do I
have to enable the domain module and add the external IP as a new domain?
Thanks in advance,
Lucas Alvarez
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. Should
Kamailio has to listen on port 5060 also for UAC module functions? If I
enable the port 5060, how do I prevent common SIP - UDP registrations? I
mean, I want the registrations to be over TLS only. I would appreciate if
someone can put me on the right path.
Thanks in advance.
Lucas Alvarez
be made to the
route configuration nor which parameters should be configured. I would
appreciate if anyone can enlighten me or could point me to the right
documentation page.
Thanks in advance.
Lucas Alvarez
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a.c:298: error: expected ')' before '*' token
abyss_data.c:311: error: expected ')' before '*' token
abyss_data.c:325: error: expected ')' before '*' token
abyss_data.c:331: error: expected ')' before '*' token
make: *** [abyss_
Wed, Apr 20, 2011 at 12:40 PM, Andrew Pogrebennyk <
andrew.pogreben...@portaone.com> wrote:
> Lucas,
> It's not possible to answer your question without at least a corresponding
> SUBSCRIBE message from Aastra.
>
>
> On 20.04.2011 17:17, Lucas Alvarez wrote:
>
&g
Hi, I have implemented kamailio with asterisk, I want to enable BLF using
kamailio instead of asterisk. I would appreciate any kind of help, below I'm
pasting my configuration.
*I'm testing with an Aastra phone, when phone sends the subscription
kamailio answers with:*
U 192.168.15.22:5060 -> 19
Hi, is it possible to change the name of the table voicemessages for
voicemail profile in a kamailio-asterisk integration? I mean of the kamailio
side.
Thanks in advance.
Lucas
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Hi, I have an integration kamailio-asterisk. I have many calls that don't
pass through asterisk, kamailio route that calls to other media gateways. I
would like to log those calls in asterisk CDR, is that possible? Could
someone give me any direction? Thanks in advance.
Lucas
_
Hi, I have compiled kamailio 3.1.0 without any error and I having this error
when running kamailio:
ERROR: [sr_module.c:523]: ERROR: load_module: could not open module
:
/usr/local/lib/kamailio/modules_k/sl.so: undefined symbol: fm_malloc
Any will be appreciated.
Regards,
Lucas
Hi, is there a way "database driven" to know who is using the phone? Thanks
in advance.
Lucas
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Take care if you are using realtime, I saw that asterisk erase for some
reason the "username" field in the sipusers table when you do sip reload.
Regards,
Lucas
On Fri, Oct 22, 2010 at 11:59 AM, Ovidiu Sas wrote:
> If you use that particular config, you need to disable authentication
> on the
Hello, here is How-To written by Daniel-Constantin Mierla of a
Kamailio-Asterisk integration:
http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb,
check the .cfg file posted there. Actually I followed up that tutorial and I
have the extensions registered both on Asterisk and
t: Asterisk PBX
Max-Forwards: 70
Date: Tue, 12 Oct 2010 14:44:26 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO
Supported: replaces
Content-Type: application/sdp
Content-Length: 287
On Mon, Oct 11, 2010 at 7:36 PM, Daniel-Constantin Mierla wrote:
> Hello,
>
&
Hi, I'm having a problem with the caller ID, I have implemented an
integration between asterisk and kamailio following this tutorial:
http://kb.asipto.com/asterisk:realtime:kamailio-3.0.x-asterisk-1.6.2-astdb
and the problem is that when I call from extension, let's say 1000, to
another extension,
and now?
> Regards
>
> Lucas
>
> On Tue, Jul 6, 2010 at 6:51 PM, Iñaki Baz Castillo wrote:
>> 2010/7/6 Lucas Alvarez :
>>> Hi, I new with kamailio, I've been able to integrate kamailio 3.02
>>> with asterisk 1.6. The only issue I'm having is if I have to
in? Do you understand now?
Regards
Lucas
On Tue, Jul 6, 2010 at 6:51 PM, Iñaki Baz Castillo wrote:
> 2010/7/6 Lucas Alvarez :
>> Hi, I new with kamailio, I've been able to integrate kamailio 3.02
>> with asterisk 1.6. The only issue I'm having is if I have to restart
>
Hi, I new with kamailio, I've been able to integrate kamailio 3.02
with asterisk 1.6. The only issue I'm having is if I have to restart
asterisk( for some config update) I loose all the sip registration in
asterisk, is there any way of fixing this?
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