All,
The Kamailio-Asterisk integration that is described here
asterisk:realtime:kamailio-4.0.x-asterisk-11.3.0-astdb [Asipto - SIP and VoIP Knowledge Base Site]
is limited to Asterisk servers with a fixed IP address. The reason is that Kamailio identifies and authorizes traffic from Asterisk based on the source IP address. Attached is a patch to kamailio.cfg that adds support for dynamic Asterisk hosts. It introduces the option to define a fully-qualified domain name instead of the IP address to identify the Asterisk server. The host name can be dynamically updated to always point at the (changing) IP address of the Asterisk server. Defining WITH_ASTERISK_FQDN enables the new option.
When WITH_ASTERISK_FQDN is set, Kamailio will perform a DNS search on the host name in the asterisk.bindhost variable. It will authorize incoming traffic if it originates from one of the IP addresses the DNS lookup returns (and matches the asterisk.bindport variable as before).
Furthermore, the patch introduce the option to automatically set the IP address of the server that Kamailio is running on. This does away with the need to properly define kamailio.bindip on every server that the configuration is running on. This option is enabled by defining WITH_AUTOIP.
With the suggested changes I have been able to successfully run an Asterisk server in my home, having a dynamic public IP address and connecting to the Kamailio proxy on a VPS server with a static IP addres.
Please let me know if you have any comments or suggestions.
Kind regards,Rudy Eschauzier.
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I have Kamailio running and connected to a PSTN gateway.
My subscribers are named ex. +442071234567 - same as their real GSM number
from my PSTN gateway.
I'm using the standard kamailio.cfg which ships with version 4.3.
When I'm trying to dial SIP client to SIP client I would like to have
Kamailio route the call internally if a subscriber exists with ex.
+442071234567.
If no subscriber exists with ex. +442071234567 it should send it to my PSTN
gateway.
As it is now it seems as if it are trying to both call "internally" and via
the PSTN gateway.
How should one fix this issue the best way?
Hello,
I set some logs in onsend_route and it works fine for requests.
With setting core parameter onsend_route_reply=1 ( yes, on), i expect to
see logs for replays also, but it not occurs.
Is this a bug or I miss something?
version: kamailio 4.2.6 (x86_64/linux) db77ac
BR,
Julia
Good morning.
Has anyone seen this too?
Looking forward to hear from you.
BR,
--
*Nuno Miguel Reis* | *Unified Communication** Systems*
M. +351 913907481 | nreis(a)wavecom.pt
WAVECOM-Soluções Rádio, S.A.
Cacia Park | Rua do Progresso, Lote 15
3800-639 AVEIRO | Portugal
T. +351 309 700 225 | F. +351 234 919 191
*GPS
<http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…>
| www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>*
<http://www.wavecom.pt/>
<http://www.wavecom.pt/pt/>
On Fri, Aug 14, 2015 at 1:41 AM, Nuno Reis <nreis(a)wavecom.pt> wrote:
> Hi guys.
>
> I've noticed this CRITICAL log line on kamailio startup:
>
> CRITICAL: dialog [dlg_cb.c:137]: register_dlgcb(): DLGCB_CREATED type
> registered after shutdown!?!
>
>
> This line started to show up after upgrade from 4.2.x to 4.3 and now to
> 4.3.1.
> Everything seems to be working fine and a really don't see any harm
> anywhere.
> Could you please comment on this?
>
> Cheers,
>
> --
>
> *Nuno Miguel Reis* | *Unified Communication** Systems*
> M. +351 913907481 | nreis(a)wavecom.pt
> WAVECOM-Soluções Rádio, S.A.
> Cacia Park | Rua do Progresso, Lote 15
> 3800-639 AVEIRO | Portugal
> T. +351 309 700 225 | F. +351 234 919 191
> *GPS
> <http://maps.google.com/maps/ms?msa=0&msid=202333747613191340808.0004b4b227a…>
> | www.wavecom.pt <http://www.wavecom.pt/>** <http://www.wavecom.pt/>*
>
> <http://www.wavecom.pt/>
> <http://www.wavecom.pt/pt/>
>
>
>
Hi everyone,
I am using Kamailio-4.1.5 and Kamailio-4.2.5 but with this both version I
had a problem:
Connection diagrams:
SIP client ---- SBC --- Kamailio --- MGC/MGW --- GSM/PSTN subscriber
Scenario:
- Caller (SIP) make a new call to GSM (PSTN) subscriber
- When GSM/PSTN is ringing or not ringing, Caller cancel call (Caller send
CANCEL message)
- But MGC does not send response message for CANCEL message
- After about 120s (fr_inv_timer), Kamailio send 487 Request Terminated to
SBC.
>From this time, Kamailio have memory leak (tm does not free transactions)
I reproduced this bugs 4-5 times and same problem is happen
I attached capture file and log when I show shared memory (using kamcmd
core.shmmem) and tm statistics (using kamcmd tm.stats). And time Kamailio
start memory leak is 08-30-15-10:18:53
Could you help me solve this problem?
Thank you so much!
--
---------------------------
*Bui Thi Minh Yen*
Hi list!
(sorry for sending first mail to wrong sub forum)
Have a following issue:
Server with Kamailio restart with wrong time (why - this is separate part of
my investigations), timeshift e.g. for 2 hours in feature.
After ntp daemon adjust correct time (move time backwards), Kamailio do not
accept REGISTERs, pv_www_authenticate answer with code -4 (nonce expired)
I've reade the description of:
http://kamailio.org/docs/modules/4.2.x/modules/auth.html#auth.p.nonce_expire
and
http://kamailio.org/docs/modules/4.2.x/modules/auth.html#auth.p.nonce_auth_…
This descriptions mention situation with backwards timeshift from future.
As I understand, Kamailio do not authorise the registration, whose nonces
where generated in future (before ntp shifted the time). But in my tests,
Kamailio do not accept even REGISTER from UA, sent after time shift (with
new nonces, that as I expected has to use correct time for nonce generation,
and be authorised with no problem).
Only Kamailio process restart help to cope with the issue.
It looks like for nonce generating Kamailio use old time (in feature, that
was before timeshift).
I wander, does the Kamailio behave in described case - as is expected? If I
am wrong in some of my assumptions, please point out.
version: kamailio 4.2.5
Thank in advance!
Cheers!
--
View this message in context: http://sip-router.1086192.n5.nabble.com/during-registration-nonce-expired-a…
Sent from the Users mailing list archive at Nabble.com.
Hi All,
We are trying to make 10000 tls connection with Kamailio server for SIP
signaling.When I try beyond 6500 TLS connection then TLS handshake is
failing.We are running the kamailio by using below command
- kamailio -f kamailio.cfg -N 40 -m 1024 -M 16
NOTE: In kamailio.cfg, we have mentioned tls_max_connections=60000
Could you please guide me, how can I improve the tls connections with
kamailio ?
Thanks
Priyaranjan
Hello
I'm looking for information on setting SER up on Ubuntu. I have it installed along with rtpproxy but cannot get any info from this site as all documentation links are not working. What I want to do is to receive a input rtp live stream and restream it out to various TV's and decoders. Is this possible with SER ?
Thanks
Reaz