t-aliasing]
> && (*(char **)(group_inst->vars + var->offset) != NULL)
> ^
> core/cfg/cfg_ctx.c:1707:6: warning: dereferencing type-punned pointer will
> break strict-aliasing rules [-Wstrict-aliasing]
> replaced[num] = *(char **)(group_inst->vars + v
Hello list,
Where can I found any information to completely understand what do values
returned by 'kamctl stats' represent?
Cheers,
Alex
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http://li
utput of 'bt full' with gdb and
> send it over here.
>
> Cheers,
> Daniel
>
>
> On 15/11/16 22:35, Alexandru Covalschi wrote:
>> Hello list,
>>
>> We’re using dev version of Kamailio:
>> version: kamailio 5.0.0-dev4 (x86_64/linux) ff63e5
&g
Hello list,
We’re using dev version of Kamailio:
version: kamailio 5.0.0-dev4 (x86_64/linux) ff63e5
flags: STATS: Off, USE_TCP, USE_TLS, USE_SCTP, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, Q_MALLOC,
F_MALLOC, TLSF_MALLOC, DBG_SR_MEMORY, USE_F
(SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-user
Thanks for the info guys, I've fixed my config to suit the correct logic.
Thanks again!
2016-06-23 8:40 GMT+03:00 Daniel-Constantin Mierla :
> Hello,
>
> On 19/06/16 19:41, Alexandru Covalschi wrote:
>
> Hello list,
>
> I need to send to an external API events when us
The problem may be with record_route header.
Did you set
*advertised_address?*
2016-06-21 12:59 GMT+03:00 Amit Patkar :
> Hi
>
> I am using Kamailio as Websocket proxy.
>
> User 1 & User 2 are registered on Kamailio over WebSocket.
> When User 1 calls User 2, User 2 gets ring and answers the call
sends REGISTER he is de-register.
(Please correct me if I'm wrong.)
How can I catch that?
Can I use event_route[usrloc:contact-expired]?
Thanks in advance!
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
tel: +37367398493
web: http://abriss.solutions/ <ht
blem, or if you know
> how to solve it?
>
>
> Best regards.
>
> Sirvan Parasteh
>
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> http://lists.sip-router.org
Thanks everyone
2016-02-25 19:41 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:
> Ok guys. The issue was in my misunderstanding of RFC and
> advertised_address variable.
> Removing advertised_address solved the issue.
>
> 2016-02-25 17:49 GMT+02:00 Alberto Sagredo
>
> And ACKS will go to right place..
>
>
> 2016-02-25 16:43 GMT+01:00 Alexandru Covalschi <568...@gmail.com>:
>
>> force_send_socket is a good idea - thanx!
>> traces are in initial message
>>
>> 2016-02-25 17:02 GMT+02:00 Alberto Sagredo > >:
>&g
t;}
>
> Hope this helps you
> Use record_route() as well.
>
> Anyway show me a trace that goes to FreeSwitch from Kamailio.
>
>
>
> 2016-02-25 10:55 GMT+01:00 Alexandru Covalschi <568...@gmail.com>:
>
>> No other rr params defined so double rr is defau
No other rr params defined so double rr is default - enabled.
What do you mean by "force traffic" - how to do that? Every other request
(excep BYE - same problem with it) flows OK.
2016-02-25 11:49 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:
> Hi, thanks for answer
>
te IP?
>
> Do you use advertise?
>
> Maybe you need to force Outbound traffic to Public IP Socket and inside
> traffic to Private IP .
>
> Do you have double record routing?
>
> BR
>
> 2016-02-25 1:24 GMT+01:00 Alexandru Covalschi <568...@gmail.com>:
>
>>
aw/5LpwSigF
I'm seeking help with that - what parameter I need to change/add to solve
that?
Maybe it's a networking problem - but why then ACK reaches Freeswitch and
all other requests flow OK?
Thanks in advance, Alex
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and syste
uot;PUBLIC-IP", "$Rp", "udp") in
> order to received new transactions or should I follow a different
> procedure???
>
> Thank you
>
>
>
>
> ___
> SIP Express Router (SER) and Kamailio (OpenSER) - sr-
U != "chat-3500")
> {
> if(imc_manager())
> sl_send_reply("200", "ok");
> else
> sl_send_reply("500", "command error");
> exit;
> }
> This allows me to receive system messages - but I can't get any mess
body "111" from user 1001 and
imc_manager catches it - I receive 500 command error. Why? :/
All that is working on top of sipjs-demo.
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://
_send_reply("500", "command error");
exit;
}
This allows me to receive system messages - but I can't get any messages
from clients.
2015-12-15 18:43 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:
> Hello again
> First of all I wanted to ask if someone e
med out (in this
> case 408 is selected against 486). How many INVITE requests do you see
> being sent out? Or you can eventually make the sip trace available for
> viewing on this mailing list or some web site/pastebin out there.
>
> Cheers,
> Daniel
>
>
> On 15/12/15 12
pipermail/sr-users/2010-November/066382.html
> is recommended just to exit failure_route, but that didn't work for me. I
> need that to let Freeswitch know which cause has ended the call. Now
> Kamailio just sends ACK to endpoint on receiving 486 BUSY. Would you kindly
> tell me ho
e has ended the call. Now Kamailio
just sends ACK to endpoint on receiving 486 BUSY. Would you kindly tell me
how to achieve that?
Thanks in advance
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-teleco
Well I tried but didn't work for me :( however problem is solved using
other voip provider. Thanks for help!
27 нояб. 2015 г. 14:09 пользователь "Daniel Tryba"
написал:
> On Friday 27 November 2015 13:50:36 Alexandru Covalschi wrote:
> > I saw that, but
> > 1. It
I saw that, but
1. It doesn't work in failure_route (MANAGE_FAILURE from std. config) either
2. My question was more general - is it even possible to do what I need
with Kamailio
2015-11-27 13:24 GMT+02:00 Daniel Tryba :
> On Friday 27 November 2015 13:01:19 Alexandru Covalschi wrote:
, "Remote asked for authentication");
uac_auth();
}
to MANAGE_FAILURE or MANAGE_REPLY route Kamailio can't start.
Is that even possible?
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
__
x27;ll need to implement that in my script, or maybe I can use
some built-it functions?
2015-11-13 19:52 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:
> Many thanks for you help Sebastian!
>
> 2015-11-13 19:13 GMT+02:00 Sebastian Damm :
>
>>
>> On Fri, Nov 13
th, there can be many auth headers, each with different
> realm. In sip is very likely to be only one.
>
> Cheers,
> Daniel
>
>
> On 16/11/15 15:26, Alexandru Covalschi wrote:
>
> UPD: proxy_auth doesn't work either, however I'm sure I have WWW-Auth, not
> Pr
UPD: proxy_auth doesn't work either, however I'm sure I have WWW-Auth, not
Proxy-Auth :)
2015-11-16 16:23 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:
> Hello everyone!
>
> I need to extract values from authentication header, but
> 408. @authorization["strin
DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE,
USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES
ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16,
MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 8MB
poll method support: poll, epoll_lt, epoll_et, sigio_rt, selec
Many thanks for you help Sebastian!
2015-11-13 19:13 GMT+02:00 Sebastian Damm :
>
> On Fri, Nov 13, 2015 at 3:43 PM, Alexandru Covalschi <568...@gmail.com>
> wrote:
>
>> What if I don't need a plaintext password on Kamailio? I mean, I don't
>> want t
> Best Regards,
> Sebastian
>
>
>
> On Fri, Nov 13, 2015 at 3:13 PM, Alexandru Covalschi <568...@gmail.com>
> wrote:
>
>> simple send_reply("200", "OK");, sorry
>>
>> 2015-11-13 16:02 GMT+02:00 Alexandru Covalschi <568...@gmail
So it should be like
...
if (!has_credentials("myrealm")) {
www_challenge("$td", "1");
}
else {
if (!my_script()){
sl_send_reply("401", "Not Authorized");
}
}
...
2015-11-13 16:13 GMT+02:00 Alexandru Coval
simple send_reply("200", "OK");, sorry
2015-11-13 16:02 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:
> Thanks for your reply! But the problem is - I need to provide to API
> user's login and password. Kamailio doesn't know them. So my idea was t
riable.
>
>
> http://www.kamailio.net/docs/modules/4.3.x/modules/auth.html#auth.f.pv_www_authenticate
>
> Best Regards,
> Sebastian
>
> On Fri, Nov 13, 2015 at 2:14 PM, Alexandru Covalschi <568...@gmail.com>
> wrote:
>
>> UPD: If upper method is possible -
UPD: If upper method is possible - I assume I can check if message has Auth
header using
if (has_credentials("myrealm")) {
...
}
Can you please specify how to grab it?
2015-11-13 15:08 GMT+02:00 Alexandru Covalschi <568...@gmail.com>:
> Hello!
> My problem
;);
}
The main problem is - how can I grab or compare users password? I know
nonce, which I understand is MD5 salt. Can I, for example, grab users
password from API, then grab the MD5 string and the nonce user sent me,
calculate MD5 on base of API password and nonce - and then compare MD5
strings
Just wanted to ask when will http://www.asipto.com/sw/kamailio-admin-book/
become available in final version? Sorry for offtop, didn't know where to
ask about that :)
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-teleco
, but quality is sometimes bad and the delay is too big.
Anyone has any ideas? Kamailio 4.3, FS 1.4
<http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc>
<http://kb.asipto.com/freeswitch:kamailio-3.3.x-freeswitch-1.2.x-sbc>
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engin
t;
> Your assistance in this matter is greatly appreciated
>
> Thanks,
> Al
>
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> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinf
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> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
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ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://
s (FS#456) and
> seemed to be fixed in branch 4.1 (part of 4.1.5).
>
> Thanks.
>
>
> ___
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> http://lists.sip-router.org/cgi-
And server is under Amazon EC2, but that shouldn't really make any sense
2015-08-29 0:11 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
> Forgot to add
> cat /etc/issue
> Debian GNU/Linux 8 \n \l
>
>
> kamailio -V
> version: kamailio 4.3.1 (x86_64/linux)
>
compiled with gcc 4.9.2
openssl version
OpenSSL 1.0.1k 8 Jan 2015
2015-08-28 20:01 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
> Hello!
>
> I'm having problems with Kamailio configuration with TLS. Or, maybe,
> that's my misunderstanding about how it should work.
&
ire_certificate = no
Make everything work.
Cross-domain calling is essential and I'm just trying to figure out -
what's the problem? Is that my certificate, is that ostel.co certificate or
it is just the way it should be?
Thanks!
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and s
Also, take a look at kamailio-advanced.cfg, there is PSTN GW route already
included. Also you can use LCR for routing calls to different providers, a
simple guide can be found here
http://dopensource.com/least-cost-routing-with-kamailio-v4-1/
2015-08-12 0:41 GMT+03:00 Alexandru Covalschi <
;
> For security reasons, i would like to force the RTP through RTPProxy.
>
> I'm missing something, and need your help me to understand my errors.
>
> Best Regards,
> Bruno
>
>
>
> ___
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(SER) and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
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Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
__
Hmf... I saw the advice to put them on /tmp/ somewhere on mailing lists and
had same thoughts. Thanks, will fix that on my servers!
2015-08-10 14:16 GMT+03:00 Daniel Tryba :
> On Monday 10 August 2015 13:12:12 Alexandru Covalschi wrote:
> > Shouldn't they be /tmp/kamailio
le.
>
> DanB
>
>
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> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
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AB
; /var/run/kamailio/kamailio_(ctl|fifo)
>
> ___
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> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
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Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system admin
? is there any avaliable way to automate or we have
> to rewrite/modify the lcr/drouting module for rate selection ,
>
>
>
> On Sun, Aug 9, 2015 at 9:22 PM, Alexandru Covalschi <568...@gmail.com>
> wrote:
>
>> however you can try building LCR based on prefix and
however you can try building LCR based on prefix and weight, why not?
2015-08-09 18:52 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
> I know CGRateS allow cost-based LCR for Kamailio. Maybe there are some
> internal Kamailio modules, but I don't know about them
>
>
On Sun, Aug 9, 2015 at 3:09 PM, Alexandru Covalschi <568...@gmail.com>
> wrote:
>
>> Or, well, see that guide
>> http://dopensource.com/least-cost-routing-with-kamailio-v4-1/ - we have
>> priority and weight on LCR module
>>
>> 2015-08-09 12:37 GMT+03:00 Al
Or, well, see that guide
http://dopensource.com/least-cost-routing-with-kamailio-v4-1/ - we have
priority and weight on LCR module
2015-08-09 12:37 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
> try using CGRateS
>
> 2015-08-09 12:06 GMT+03:00 Arun Kumar :
>
>> Hi
rs@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
___
SIP Express
thanks!
2015-08-06 22:29 GMT+03:00 Frank Carmickle :
> Zrtp passes through rtpengine just fine.
>
> --FC
> Sent from my 6 plus
>
> On Aug 6, 2015, at 14:12, Alexandru Covalschi <568...@gmail.com> wrote:
>
> Sorry if writing to wrong mailing list, I am very limited t
Sorry if writing to wrong mailing list, I am very limited to traffic now
amd don't know if there is any for rtpproxy/rtpengine.
My question is - can they support ZRTP at least in pass-through mode? Will
rtpengine fail on trying to recognize unknown SDP fields?
__
I got bridging working well on internal interfaces in case of simple SIP
calls on a bit other configuration. But editing this config to support
WebRTC causes same problems. I need internal interfaces on asterisk to
completely close external ones (Security etc.).
Asterisk localip=10.0.0.87, sorry
2015-06-24 16:24 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
> Ok, so my scheme.
> Kamailio and Asterisk are in Amazon EC2
> Kamailio externip=54.197.230.121 localip=10.145.45.103
> Asterisk localip=10.145.45.103, externip doesn't
; Can you specify exactly which side received what IP and what you would
> expect there? It is not easy to digests lots of logs and also guess what
> would you expect to happen...
>
> Cheers,
> Daniel
>
>
> On 24/06/15 15:14, Alexandru Covalschi wrote:
>
> Heh...
&
to)=="ws")
{
xlog("L_NOTICE","= $fU has WEBSOCKETS");
rtpengine_manage("trust-address replace-origin
replace-session-connection ICE=force RTP/AVP");
}
else
{
/jZceP2Rn
Javascript log http://pastebin.com/4ZLePyKz
2015-06-24 1:27 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
> Well.. Guys, sorry, it was totally my fault. I just used VPN.
>
> 2015-06-24 0:59 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
>
>> I used
Well.. Guys, sorry, it was totally my fault. I just used VPN.
2015-06-24 0:59 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
> I used https://github.com/caruizdiaz/kamailio-ws configuration that 100%
> works on other then Amazon EC2 environment and I still get this error.
&
GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
> Here is it
> http://pastebin.com/JkkM4M5m
>
> 2015-06-23 18:53 GMT+03:00 Daniel-Constantin Mierla :
>
>> There are no major changes in 4.3 comparing with 4.2 in regards to
>> websocket -- the implementatio
ble. Can you look at
> javascript debug console in the browser to see what is printing?
>
> Daniel
>
>
> On 23/06/15 17:23, Alexandru Covalschi wrote:
>
> without fix_nated_contact error behaviour is the same
> maybe I should upgrade to 4.3 ?
>
> 2015-06-23 14:08 GM
without fix_nated_contact error behaviour is the same
maybe I should upgrade to 4.3 ?
2015-06-23 14:08 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
> Here's the trace on port which I use for ws server. Don't look at
> fix_nated_contact, I'll fix later - now t
Here's the trace on port which I use for ws server. Don't look at
fix_nated_contact, I'll fix later - now the trouble is that Kamailio can't
establish a ws connection properly. Client is SIPML5 demo phone
http://pastebin.com/LvAk2HkP
2015-06-23 14:03 GMT+03:00 Alex
I solved the SIP voice trouble, but WebRTC problem still exists. What kind
of trace I must do to make my post more informative?
2015-06-23 10:46 GMT+03:00 Daniel-Constantin Mierla :
> Hello,
>
> On 23/06/15 04:10, Alexandru Covalschi wrote:
>
> Hello. I'm trying to set u
P");
For WS: rtpengine_manage("trust-address replace-origin
replace-session-connection ICE=force RTP/AVP");
Do you have any ideas how ti fix that? I also make REGFWD's to Asterisk
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system admi
doing with Kamailio, then as Fred suggested, it is better to consult a
> lawyer.
>
> Cheers,
> Daniel
>
> On 15/06/15 18:09, Alexandru Covalschi wrote:
>
> Maybe it may be an offtopic, but I'm not really into legal issues - so
> I'm sorry if this message is not fu
Maybe it may be an offtopic, but I'm not really into legal issues - so I'm
sorry if this message is not fully related to this mailing list.
Can I use Kamailio to provide VoIP backend for kind of CRM system in case
of SaaS?
---
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer
thanks, will try that
2015-06-15 14:07 GMT+03:00 Juha Heinanen :
> Alexandru Covalschi writes:
>
> > > sorry, i thought you use registrar/usrloc modules
> > Well, I do use them - so if you could explain in which table does
> Kamailio
> > write the user's pro
xlog("L_NOTICE","= $fU is websocket user\n");
rtpengine_manage("direction=external direction=internal force
trust-address replace-origin replace-session-connection ICE=force RTP/AVP");
}
else
{
xlog("L_NOTICE&qu
and Kamailio (OpenSER) - sr-users mailing list
> sr-users@lists.sip-router.org
> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
--
Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
web: http://abs-telecom.com/
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>
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Alexandru Covalschi
ABRISS-Solutions
VoIP engineer and system administrator
phone: +37367398493
ivr and handle callcenter
> (freeswitch)
>
>so here i try to kamailiio act proxy server
>
> Any idea how i can achieve thid
>
>
>
>
>
> On Sun, Jun 14, 2015 at 12:24 AM, Alexandru Covalschi <568...@gmail.com>
> wrote:
>
>> Well, I perf
Sorry, a mistake: on outgoing webrtc it MUST have RTP/SAVP or RTP/SAVPF
2015-06-13 21:54 GMT+03:00 Alexandru Covalschi <568...@gmail.com>:
> Well, I performed that by creating a media relay consisting of 2
> freeswitches and using rtpengine.
>
> You just need to handle WebRTC
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>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>>
>>
>
> ___
>
creating a new user on subscriber table? Becase I see 2 fields with
hashes.
Thank you
--
Alexandru Covalschi
VoIP engineer and system administrator
phone: +37367398493
<http://abs-telecom.com/>
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SIP Express Router (SER) and Kamailio (OpenSER
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