Hi Daniel
Kamailio is for hard people and fun :)
Thanks Visily i finnaly got it working with your tip. You were right about
internal external options instead direction=...
Here its some code to someone could need it
route[RTPPROXY] {
if (is_method("INVITE")){
if(ds_is_from_list(1)){
I am a newbie trying out kamailio
*kamailio version: *4.3.0
*Operating system:* CentOS 7
Following the instructions provided at
http://www.kamailio.org/wiki/install/4.3.x/git
While I was trying to include the module "db_mysql", I get the following
error
[root@localhost kamailio]# make modules i
Hello,
I need to prepare two kamailio instances accessing the same mysql db.
They will write to different location tables so i use function calls like
these:
Server1:
save("k1_location");
lookup("k1_location");
Server2:
save("k2_location");
lookup("k2_location");
This is working fine.
H
Hello,
I have a setup with 2 Kamailio server and their own database. Is it possible
to connect these 2 Kamailio server, that the client from the one can make a
call to the other? If yes, how could this be realized?
Many Thanks.
Cheers,
Karl
---
Diese E-Mail wurde von Avast Antivirus-So
Sure but if you look into the dispatcher module there is a field called
'setid' or groupid. Use it wisely to differentiate between the Load
Balanced asterisk pool and the Telco IP.
The dispatcher module is exactly what you should use. You can find out if
incoming source IP belongs to a particular s
Hi,
Thanks for the immediate reply.
You are right ,using the dispatcher module , i am able to send the OPTIONS
packet to MSC Telco.
But as i describer in my earlier mail, i am using the same dispatcher
module to establish the sip trunk between my My Kamailio server and my
Asterisk server.
Ther
Hi,
You're right about using IP Auth in Kamailio. You'll need to use the
permissions module. However I believe permissions module wont send the
OPTIONS to the MSC SIP Server. For this you may alternatively use the
"dispatcher" module.
Take a look at the sample kamailio.cfg here:
http://kb.asipto.c
Hi,
We have a requirement with one of our telco
We are using asterisk in our servers and we are planning to implement SIP-I
protocol and we choosed kamailio for it.
In Kamailio website, i came to know that kamailio will be supporting both
SIP-I and SIP-T protocols
Below is what we need and pls co
On Tuesday 14 July 2015 18:19:02 Alberto Sagredo wrote:
> In my tests rtpproxy recording waste less resources than asterisk
>
> That was one of the reasons
How much time have you spend so far on a problem that asterisk can handle out
of the box? ;)
I'd love to do this with kamailio/rtpengine (I
Hi daniel
In my tests rtpproxy recording waste less resources than asterisk
That was one of the reasons
BR
ALBERTO
El martes, 14 de julio de 2015, Daniel Tryba escribió:
> On Tuesday 14 July 2015 17:01:38 Alberto Sagredo wrote:
> > What i try is to detect if i have SAVP from endpoint and tr
On Tuesday 14 July 2015 17:01:38 Alberto Sagredo wrote:
> What i try is to detect if i have SAVP from endpoint and translate to RTP
> to ASterisk an later RTP from ASterisk translate to SRTP using rtpengine
Why not just handle SRTP with asterisk and record there?
Thanks Vasily
direction parameter i understood it was as ie options on rtpproxy, maybe im
wrong.
About them , i was ngreping all trace and that was the option to get SDP
correct mapped using RTPengine..
WIll check with your comments and let you know back.
BR
2015-07-14 17:16 GMT+02:00 Vasiliy
Alberto Sagredo-2 wrote
> Thanks Vasily i have changed a little today using a RTPPROXY route.
>
> Thats what i have right now
>
> But its not working as expected
>
> What i try is to detect if i have SAVP from endpoint and translate to RTP
> to ASterisk an later RTP from ASterisk translate to SR
Hi,
I'm trying to restore a SIP session dialogue context and found dlg_get(callid,
ftag, ttag) function for that.
However, I cannot access the To tag. I noticed on the 180 RINGING there was
already an self-generated included on this response, but I also could not
access its value.
When recei
Thanks Vasily i have changed a little today using a RTPPROXY route.
Thats what i have right now
But its not working as expected
What i try is to detect if i have SAVP from endpoint and translate to RTP
to ASterisk an later RTP from ASterisk translate to SRTP using rtpengine
I had extrange behav
Hi Daniel
Here its Yealink one (Optional SRTP)
If you need anything more let me know
INVITE sip:1@192.168.0.181:5080 SIP/2.0.
Record-Route: .
Record-Route:
.
Via: SIP/2.0/UDP
x.x.x.x:8002;branch=z9hG4bK4928.64d27b89d3de27a54610b8ebb2aa9f43.0;i=2b2.
Via: SIP/2.0/TLS 10.0.1.111:11880
;received
Hi Daniel,topoh moduling and its working well for this scenario now, so thank
you.
However like you say enabling might introduce problems with call flow.
Are there any other methods to modify/regenerate call-id header?
Other than that yes I will look at dialplan :)
Thanks
Jon
> From: d.tr...@poc
Alberto Sagredo-2 wrote
> ...
> I have been able to make SRTP To RTP to Asterisk
>
> But im not able to call between SRTP extensions, i understand also SRTP to
> RTP would work as im doing with Asterisk (Only the speak SRTP as rtpengine
> trasncode)
>
>
> If you need any more info let me know.
>
Thanks Daniel!
2015-07-14 13:11 GMT+02:00 Daniel Tryba :
> >If i turn STUN off as Olle mentioned, all worked fine.. but if i have a
> >customer that for error it turns ON it does not work in my case..
>
> > Any way to force RTP in that cases?
>
> Just always start the rtpproxy, especially when
On Tuesday 14 July 2015 11:29:34 Jonathan Hunter wrote:
> No such extension/context *1022-MAGNORTH13@from-outside while calling
> Local channel[2015-07-14 10:49:38] NOTICE[15318]: app_dial.c:1003
> do_forward: Forwarding failed to dial
> 'Local/*1022-MAGNORTH13@from-outside'
This sounds l
Hi Daniel,
It fails the call, well doesn't respond;
2015-07-14 10:49:38] NOTICE[15318]: app_dial.c:901 do_forward: Not accepting
call completion offers from call-forward recipient
Local/*1022-MAGNORTH13@from-outside-0001;1[2015-07-14 10:49:38]
NOTICE[15318]: chan_local.c:973 local_call:
On Tuesday 14 July 2015 11:04:59 Jonathan Hunter wrote:
> The problem I have is when I send an INVITE from Asterisk Server, and then
> kamailio sends an INVITE back to the same Asterisk Server I get the
> following failure in asterisk console;
Is it just a warning or does the chanspy fail?
You
>If i turn STUN off as Olle mentioned, all worked fine.. but if i have a
>customer that for error it turns ON it does not work in my case..
> Any way to force RTP in that cases?
Just always start the rtpproxy, especially when you have a setup where some
rtp is handled with non public ips:
rout
Hi guys,
I have a kamailio load balancing requests across 2 Asterisk Servers, and all is
working well.
Only problem now is when utilizing feature codes to look to listen into calls
using the chanspy application I am having issues.
I am passing calls to * into Asterisk, where an IVR answers th
Hi Daniel,
this is the ACK received on Kamailio:
#
U 172.26.130.235:44435 -> 192.168.0.245:5060
ACK sip:7240F8EF-55A4D642000CBC22-8A135700@172.16.0.21;transport=udp SIP/2.0
Via: SIP/2.0/UDP
172.26.130.235:44435;branch=z9hG4bK-d8754z-1cd3a01fa9171649-1---d8754z-
Max-Forwards: 70
Route:
Route:
Yes you are right , but in my scenario a couple of Asterisk behind NAT
must not route RTP directly , all must go to RTP Proxy so i have found
that case that for me its not working.
Sorry if i haven explained so much..
If i turn STUN off as Olle mentioned, all worked fine.. but if i have a
custo
Hello,
what is the request URI? It might be better to just paste here the full
ACK message.
Cheers,
Daniel
On 14/07/15 12:25, Antonio Reale wrote:
> Sorry,
> here's the Route HF present in the ACK received from U1:
>
> Route:
> Route:
>
> Regards.
>
> Antonio
>
>
>
> Il 14/07/2015 12:09, Anton
Hi Carsten,
Thanks for the quick response, do you refer to this?
http://by-miconda.blogspot.nl/2014/10/kamailio-42-tips-10-lightweight.html
Currently there is no precheck, just the t_check_trans() on itself. I'll
add it before, just like the example. Thanks again for the quick
pointer. Hopefu
Sorry,
here's the Route HF present in the ACK received from U1:
Route:
Route:
Regards.
Antonio
Il 14/07/2015 12:09, Antonio Reale ha scritto:
Hi all,
I have the following scenario:
U1 (caller) ---> P1 (192.168.0.245, kamailio 4.3, loose-router) >
P2 > (192.168.0.101, strict rout
Hi Dirk,
i guess, this could be a retransmission.
Do you catch retransmissions? An alternative might be, just to send a
"100 Trying" in the beginning of the script, in order to stop the
retransmission.
I guess for further insight into your issue, you would have to sahre
your routing-logic.
Thank
Hello,
Running kamailio 4.2.5 for a few weeks now, coming from 4.2.3. Since the
upgrade we increased the logging substantially in our routing to get
more insight into what is going on.
We noticed that in a few cases the log showed a SIP message was being
threated by multiple forks, each cons
Hi all,
I have the following scenario:
U1 (caller) ---> P1 (192.168.0.245, kamailio 4.3, loose-router) > P2
> (192.168.0.101, strict router) > > U2 (called)
When U2 answers the call, at P1 arrives the 200 OK with:
Record-Route:
Record-Route:
The problem is that the AC
Hello,
the request has public IP address in Via and Contact, matching the
source IP. If you look at the tests you do in the readme of the
nathelper module, then the request appears as not being natted, see:
-
http://kamailio.org/docs/modules/stable/modules/nathelper.html#nathelper.f.nat_uac_test
On Tuesday 14 July 2015 11:56:38 Alberto Sagredo wrote:
> I have found an issue detecting NAT of GS 1.4.23 Phones when using Stun on
> this phones
>
> Usual nat_uac_test with numbers 19 and 3 does not seems to detect is behind
> nat so NATMANAGE is not called
Isn't that the point of using things
> On 14 Jul 2015, at 11:56, Alberto Sagredo
> wrote:
>
> I have found an issue detecting NAT of GS 1.4.23 Phones when using Stun on
> this phones
If they activate STUN they signal with a public IP. If they signal with a
public IP, they tell us they
can handle NAT by themself and require no h
I have found an issue detecting NAT of GS 1.4.23 Phones when using Stun on
this phones
Usual nat_uac_test with numbers 19 and 3 does not seems to detect is behind
nat so NATMANAGE is not called
Here its some trace. ANy clue how to handle this phones if they activate
STUN for example
U 80.26.x.x
Hello,
On 09/07/15 11:30, Djamel Bahamid wrote:
> Hi,
>
> I have terminals of video conference which are in public IP ( *DMZ*).
> I installed kamailio (4.2) which works correctly, it is also in IP
> pulique (*DMZ*). I wish configured the /kamailio.cfg/ file, so that
> kamailio works in *proxy SIP*
Hello,
On 09/07/15 12:54, Surendra Pullaiah wrote:
>
> Hello
>
>
>
> I am trying to implement code coverage tool in
> kamailio using gcov. I have added CFLAGS and LIBS in Makefile.defs and
> it got compiled successfully.
>
> My problem is when I am running kamaili
Hello,
I am considering to release the first patch version in 4.3 series by
beginning of next week (Monday or Tuesday, July 20 or 21). If you are
aware of issues not reported yet, then open an item on the tracker to be
taken care:
- https://github.com/kamailio/kamailio/issues
Cheers,
Daniel
-
On Monday 13 July 2015 13:19:32 kai.ohnac...@cbc.de wrote:
> To change the IP address which is used from the private server as the
> destination address, I added the following lines to the Kamailio.cfg.
>
> modparam("registrar", "use_path",1)
> modparam("registrar", "path_mode", 2)
> modparam("re
40 matches
Mail list logo
