>>> How can I tell Kamailio to use TLS protocol ( instead of udp) after
>>> NAPTR lookup ?
Hello,
you can see it by doing some traces, e.g. by ngrep, or increase the debugging
level and then check the logs.
>> $du = "sip:__ip_or_host__;transport=tls";
>> t_relay();
>
>IIRC we do have NAPTR suppo
Maybe the database can apply the regex and give you the transformed
result? Databases are quite capable these days. I do this in
PostgreSQL all the time.
On 07/08/2010 01:53 PM, Uriel Rozenbaum wrote:
Hi Alex, :)
I was just trying to keep it simple, but let me explain a little bit.
I'm us
Hi Alex, :)
I was just trying to keep it simple, but let me explain a little bit.
I'm using a Proxy+RTPProxy in mhomed mode. I have a public and a
private IP on that Proxy and route calls in and out of a network.
As I have many carriers that want different ANI and DNIS patterns I
have to deal wi
Hi klaus,
Suppose I can't access to NAPTR settings.
I need to manage SIP URI, so , If I right understand, the only way to
use TLS protocol in kamailio 1.5 is to append ";transport=tls" in R-URI
before relay.
In other words I need to rewrite R-URI:
$ru = $ru + ";transport=tls" ;
# and the t_relay
Am 08.07.2010 18:10, schrieb Daniel-Constantin Mierla:
Hello,
On 7/8/10 5:59 PM, Matteo Campana wrote:
Hi all,
I'm using kamailio 1.5 with TLS module.
I need to make ENUM query and get NAPTR record.
> From NAPTR lookup, I'd like to relay my SIP Invite with tls protocol.
How can I tell Kamai
Hello,
On 7/8/10 5:59 PM, Matteo Campana wrote:
Hi all,
I'm using kamailio 1.5 with TLS module.
I need to make ENUM query and get NAPTR record.
> From NAPTR lookup, I'd like to relay my SIP Invite with tls protocol.
How can I tell Kamailio to use TLS protocol ( instead of udp) after NAPTR
loo
Hi all,
I'm using kamailio 1.5 with TLS module.
I need to make ENUM query and get NAPTR record.
>From NAPTR lookup, I'd like to relay my SIP Invite with tls protocol.
How can I tell Kamailio to use TLS protocol ( instead of udp) after NAPTR
lookup ?
I've try to set :
dns_tls_pref=1
dns_udp_
Very clear ! thanks.
Le jeudi 08 juillet 2010 à 13:00 +0200, Klaus Darilion a écrit :
>
> Am 07.07.2010 19:30, schrieb inge:
> > Yes, why not, but in a few moment ;)
> >
> > Does SER 3.0 released ? In the siprouter project ?
>
> There is no ser/sip-router 3.0 "release". You have to checkout the
There may be an easier and more elegant and more performant way to
accomplish what you are trying to accomplish, but we cannot know
without seeing the pattern and the regex. :)
--
Alex Balashov - Principal
Evariste Systems LLC
1170 Peachtree Street
12th Floor, Suite 1200
Atlanta, GA 30309
Te
Am 06.07.2010 20:48, schrieb 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
asterisk( for some config update) I loose all the sip registration in
asterisk, is there any way of fixing this?
Am 08.07.2010 15:09, schrieb David:
Hey,
OK, so after further research, I found that the trouble was not in fact
in the transaction, the trouble was with fix_nated_contact();
So I setup a spiral SIP path. I have a public server that is used by a
bunch of different phones, and I have a alg ser
Hi Daniel,
Actually I'm going to partially take your advice, implementing the following:
exec_avp("echo '$avp(s:ANI)' | sed '$avp(s:carrierAniRegex)'",
"$avp(s:ANIegress)");
I'll check for performance issues later but surely this simplifies a
lot the config.
Thanks!
Uriel
On Thu, Jul 8, 2010 a
Hey,
OK, so after further research, I found that the trouble was not in fact
in the transaction, the trouble was with fix_nated_contact();
So I setup a spiral SIP path. I have a public server that is used by a
bunch of different phones, and I have a alg server ( one public and one
private).
Even if it didn't, just add your own RR header.
--
Alex Balashov - Principal
Evariste Systems LLC
1170 Peachtree Street
12th Floor, Suite 1200
Atlanta, GA 30309
Tel: +1-678-954-0670
Fax: +1-404-961-1892
Web: http://www.evaristesys.com/
On Jul 8, 2010, at 7:33 AM, Iñaki Baz Castillo wrote:
201
2010/7/8 Daniel-Constantin Mierla :
>> But be caredull with in-dialog requests as by default they would use
>> the "default" source port ;)
>>
>
> if invite is received on one socket and forwarded using a different socket,
> then it should get 2 record-route headers so the within-dialog requests ar
Am 08.07.2010 00:53, schrieb David:
Hey,
I do not do anything IP level forwarding. All my forwarding is done
using Kamailio. It looks like what I am doing is called hairpin routing.
I think the correct term is "spiral" - at least the RFC uses this term.
How do you forward the request - t_re
On 7/8/10 12:57 PM, Iñaki Baz Castillo wrote:
2010/7/8 Alex Balashov:
On 07/08/2010 06:18 AM, Victor Pascual Avila wrote:
How could I force my sip-router to send from a given port number?
Given that sip-router listens on the address of the interface (IP:port) that
you want
Am 07.07.2010 19:30, schrieb inge:
Yes, why not, but in a few moment ;)
Does SER 3.0 released ? In the siprouter project ?
There is no ser/sip-router 3.0 "release". You have to checkout the
sip-router 3.0 branch from git if you want "ser" flavor of sip-router.
If you do not care about the
Hello,
2010/7/8 Daniel-Constantin Mierla
> Hello,
>
> thanks for troubleshooting further. I just committed the fix on GIT, try to
> see if now is working ok, then I will backport in 3.0.
>
> Cheers,
> Daniel
>
>
It's working ok. Thanks!
Best regards,
Santi
2010/7/8 Alex Balashov :
> On 07/08/2010 06:18 AM, Victor Pascual Avila wrote:
>
>> How could I force my sip-router to send from a given port number?
>
> Given that sip-router listens on the address of the interface (IP:port) that
> you want to send from, which is a requirement, you can use
> force
Hi
2010/7/8 marius zbihlei
> Hello,
>
> Can you please check that tm.so is loaded before tmx.so in the cfg file. I
> investigated and _tm_table is present in tm.so
>
Yes it is loaded before tmx.so
Best regards,
Santi
___
SIP Express Router (SER) an
Thanks Alex
On Thu, Jul 8, 2010 at 12:26 PM, Alex Balashov
wrote:
> On 07/08/2010 06:18 AM, Victor Pascual Avila wrote:
>
>> How could I force my sip-router to send from a given port number?
>
> Given that sip-router listens on the address of the interface (IP:port) that
> you want to send from,
Santiago Gimeno wrote:
Hi,
I've been digging into the code and I think this error happens when
TM_HASH_STATS is defined which happens when mode=debug. This makes
that the code in modules_k/tmx/t_mi.c calls get_tm_table function that
is defined in modules/tm/h_table.h as:
#define get_tm_tabl
Hello,
thanks for troubleshooting further. I just committed the fix on GIT, try
to see if now is working ok, then I will backport in 3.0.
Cheers,
Daniel
On 7/8/10 11:45 AM, Santiago Gimeno wrote:
Hi,
I've been digging into the code and I think this error happens when
TM_HASH_STATS is defi
On 07/08/2010 06:18 AM, Victor Pascual Avila wrote:
How could I force my sip-router to send from a given port number?
Given that sip-router listens on the address of the interface
(IP:port) that you want to send from, which is a requirement, you can
use force_send_socket():
http://www.kama
Hi Folks,
How could I force my sip-router to send from a given port number?
Many thanks in advance,
--
Victor Pascual Ávila
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-rout
Hi,
I've been digging into the code and I think this error happens when
TM_HASH_STATS is defined which happens when mode=debug. This makes that the
code in modules_k/tmx/t_mi.c calls get_tm_table function that is defined in
modules/tm/h_table.h as:
#define get_tm_table() (_tm_table)
where
exter
Hello,
On 7/7/10 11:06 PM, Uriel Rozenbaum wrote:
Hey guys,
I'm using Kamailio 1.5.3-notls and need to apply some re operation to an AVP.
The problem I have is that my re expression is held in an AVP, so the
parser is not recognizing it :(
Would it be very hard to change that behavior? Any id
Hello,
post the sip trace (e.g., ngrep) with such invites.
Cheers,
Daniel
On 7/8/10 12:53 AM, David wrote:
Hey,
I do not do anything IP level forwarding. All my forwarding is done
using Kamailio. It looks like what I am doing is called hairpin routing.
Thanks,
David
On 2010-07-07 18:48,
Hi,
sorry for my late reply.
I hardcoded the Contact header in dialog/dlg_transfer.c file. It is not a
very elegant way, but I had a short time to fix it :-)
Attached you can find a diff output with the changes I made. Besides the
Contact, I had to add more headers for compatibility reasons both
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