On Wednesday 24 June 2015 17:14:03 Javier Aristizábal wrote:
> I think I understand now.. thanks a lot for your help and input :-)
I'm really wondering what tutorial you followed?
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb
has the same route[REGFWD]. But also menti
Hello Daniel,
I think I understand now.. thanks a lot for your help and input :-)
On Wed, Jun 24, 2015 at 5:03 PM, Javier Aristizábal <
javieraristiza...@gmail.com> wrote:
> Well, but in this behavior, anybody can register because the secret field
> is empty, how can manage a password for each e
Well, but in this behavior, anybody can register because the secret field
is empty, how can manage a password for each extension?
On Wed, Jun 24, 2015 at 4:59 PM, Daniel Tryba wrote:
> On Wednesday 24 June 2015 16:46:01 Javier Aristizábal wrote:
> > I just put NULL on the secret field and I got
On Wednesday 24 June 2015 16:46:01 Javier Aristizábal wrote:
> I just put NULL on the secret field and I got the status OK.. strange isn't?
This is intended behavior.
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@l
Hola Daniel,
I just put NULL on the secret field and I got the status OK.. strange isn't?
Name/username HostDyn
Forcerport ComediaACL Port Status Description
Realtime
101/101 192.168.1.156
On Wednesday 24 June 2015 16:31:31 Javier Aristizábal wrote:
> So, if I leave the secret field with NULL, I assume that I can manage the
> authentication with uac_auth(), right?
>
> Or I can do it with ACL?
It is an or, no secret -> no need for uac_auth,
Yes, I think is with NULL in that field.
So, if I leave the secret field with NULL, I assume that I can manage the
authentication with uac_auth(), right?
Or I can do it with ACL?
Thanks!
On Wed, Jun 24, 2015 at 4:20 PM, Daniel Tryba wrote:
> On Wednesday 24 June 2015 16:02:58 Javier Aristizá
Thanks for all the suggestions.
Have taken pcap dumps with commands
tcpdump -i any -w /tmp/sip_tcp.pcap host
http://whdd.org/sip_tcp_server.pcap
http://whdd.org/sip_tcp_client.pcap
The only obviously bad thing I see is "[TCP Previous segment not
captured]" mark on the packet client side dump whi
On Wednesday 24 June 2015 16:02:58 Javier Aristizábal wrote:
> Daniel, is this using proxy_challenge? from the auth module?
You need uac_auth():
http://kamailio.org/docs/modules/stable/modules/uac.html#uac.f.uac_auth%28%29
How you disable authentication for realtime users in asterisk, I don't kno
Daniel, is this using proxy_challenge? from the auth module?
On Wed, Jun 24, 2015 at 3:55 PM, Javier Aristizábal <
javieraristiza...@gmail.com> wrote:
> Hola Daniel!
>
> Could you please help me with your idea? :)
>
> this is my Routing logic: http://pastebin.com/bdWK1JW3
>
> Thanks!
>
> On Wed,
Hola Daniel!
Could you please help me with your idea? :)
this is my Routing logic: http://pastebin.com/bdWK1JW3
Thanks!
On Wed, Jun 24, 2015 at 3:48 PM, Daniel Tryba wrote:
> On Wednesday 24 June 2015 15:26:17 Javier Aristizábal wrote:
> > this is the trace: http://pastebin.com/ncwsptz0
>
> O
On Wednesday 24 June 2015 15:26:17 Javier Aristizábal wrote:
> this is the trace: http://pastebin.com/ncwsptz0
Okee, I think I understand the solution.
Endpoint registers on kamailio via challenge/response. When register is
succesful a new register is created for asterisk, but kamailio failes t
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.).
Do you have proper routing rules between the local ips of kamailio and
asterisk? Why aren't you use only external IPs if they are on different
servers? Asterisk has also the option to set external ip. It can reduce
the complexity of doing bridging of signaling and rtp. Once you get that
working you
Hi Daniel,
this is the trace: http://pastebin.com/ncwsptz0
Thanks!
On Wed, Jun 24, 2015 at 3:05 PM, Daniel Tryba wrote:
> On Wednesday 24 June 2015 13:12:48 Javier Aristizábal wrote:
> > Hi Daniel, I think that in fact I am doing the forwarding register to
> > asterisk with the uac_req_send();
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 matter
>
> Call should flo
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 matter
Call should flow like that:
webrtc <--> kamailio-externip <--> kamailio-localip <--> asterisk-localip
but now it's webrtc --> kam
Hello,
what are the errors you get? Maybe it is more that libm that needs to be
linked.
Cheers,
Daniel
On 24/06/15 10:46, Shouvanik Chakrabarti wrote:
> Hello,
> I have developed a Kamailio module to handle group messaging and
> am using certain functions from the libm.so library. Compilatio
Also, an interesting thing - if you can see in Kamailio log, a check of the
proto of user "300" is being made. But 300 is $tU, and $tU proto is being
checked only if source IP is asterisks IP.
Here's the part of config where rtpengine is engaged (in NATmanage route)
if((src_ip==10.0.0.87)
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...
> Well, I still have troubles with my configuration.
Heh...
Well, I still have troubles with my configuration. And in SDP media adress
is Amazon public interface - but rtpengine has replace-origin
replace-session-connection session, so it must be local address.
Any ideas?
Asterisk log http://pastebin.com/MFt9V9qK
Kamailio log http://pastebin.com/jZce
On Wednesday 24 June 2015 13:12:48 Javier Aristizábal wrote:
> Hi Daniel, I think that in fact I am doing the forwarding register to
> asterisk with the uac_req_send();
You generate a new REGISTER to the asterisk server, but:
-what will you do with the response from asterisk?
-what response will g
On 23/06/15 19:06, Andrey Utkin wrote:
> 2015-06-23 18:49 GMT+03:00 Daniel-Constantin Mierla :
>> Have you grabbed the sip trace on client side to see what it is
>> receiving? Are the clients reporting errors?
> Yes, see this https://gist.github.com/krieger-od/c9fe6ea4bb64fac82cda
> this is taken
Hi Daniel, I think that in fact I am doing the forwarding register to
asterisk with the uac_req_send();
On Wed, Jun 24, 2015 at 10:58 AM, Javier Aristizábal <
javieraristiza...@gmail.com> wrote:
> I see. How can forwarding the REGISTER to asterisk? Apparently I don't
> handling the uac_send
>
> O
Hi,
I would also add that if you see partial packets you can try to remove
any transport protocol (udp/tcp) and port filters. It will help if you
are dealing with IP fragmentation. Otherwise sniffer won't catch IP
fragments since they don't have transport level headers.
Best Regards,
Vitaliy
I see. How can forwarding the REGISTER to asterisk? Apparently I don't
handling the uac_send
On Wed, Jun 24, 2015 at 10:15 AM, Daniel Tryba wrote:
> On Tuesday 23 June 2015 21:22:26 Javier Aristizábal wrote:
> > On my cfg file I have this for the asterisk registration:
> >
> > route[REGFWD] {
>
Hello,
I have developed a Kamailio module to handle group messaging and am
using certain functions from the libm.so library. Compilation happens fine,
but Kamailio subsequently fails to start because the math.h symbols are
unlinked/undefined. Could someone please suggest a workaround for this.
On Tuesday 23 June 2015 21:22:26 Javier Aristizábal wrote:
> On my cfg file I have this for the asterisk registration:
>
> route[REGFWD] {
> if(!is_method("REGISTER"))
> {
> return;
> }
> $var(rip) = $sel(cfg_get.asterisk.bindip);
> $uac_req(method)="REGISTER";
...
> uac_req_send()
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