Daniel,
Thanks for the info. I am already starting my server with -m 2048, so
I should be more than fine on the memory side :) My Kamailio
deployment is a 64bit server with 8G of RAM.
How does this look for an implementation of this check? I based it
off the following snippet of the RFC
8.2.2.
Asim Riaz writes:
> could you please send your configuration for the lcr so i can compare to
> with my config just to make sure nothing wrong in my kamailio.cfg
i first call load_gws on an uri that i want to use to make gw
selection. if that call succeeds, i call next_gw() and if that call
suc
On Sat, Feb 21, 2009 at 5:16 AM, Juha Heinanen wrote:
> Asim Riaz writes:
>
> > yes two entries are the full output of lcr_dump, userpart of ruri is
> > starting from 0 but it does not find gateway. tried to increased the
> debug
> > level but not getting anything related to lcr.
>
> asim,
>
2009/2/23 ram :
>> You can use DNS SRV records to direct the sip traffic to kamailio.
>> mydomain.org can point to something else (e.g., web server). However,
>> you phone must support DNS SRV queries.
>
>
> Can i map multiple Domain to same IP address
> and use multi domain setup .
>
> example
>
>
On Thu, Feb 19, 2009 at 3:37 PM, Daniel-Constantin Mierla wrote:
>
>
> On 02/18/2009 08:25 PM, Iñaki Baz Castillo wrote:
> > El Miércoles, 18 de Febrero de 2009, Ramu escribió:
> >
> >> Hi,
> >>
> >> Thanks for the response.
> >>
> >> I went thru' the module and I got below doubt
> >>
> >> 1) I w
I found the solution.
I just had to do do record-route!
I am sorry, I could think about that.
Thank you for answers
Cordialement,
BERGANZ François
Pensez à l'Environnement, n'imprimez ce mail que si nécessaire.
-Message d'origine-
De : users-boun...@lists.kamailio.org [mailto:users-bo
2009/2/23 BERGANZ François :
> Hello,
>
>
>
> I have Asterisk1---SER---Asterisk2.
>
>
>
> When I do INVITE from the left,
>
> --the asterisk2 send 200ok to the SER
>
> --the SER forward to the Asterisk1
>
> --but the asterisk1 directly send the ACK to Asterisk2
>
>
>
> Asterisk2 retransmit the 200o
Hello,
I have Asterisk1---SER---Asterisk2.
When I do INVITE from the left,
--the asterisk2 send 200ok to the SER
--the SER forward to the Asterisk1
--but the asterisk1 directly send the ACK to Asterisk2
Asterisk2 retransmit the 200ok… and error.
I think that it need that the ACK co
Opening a persistent MI connection to the datagram socket would achieve
the IPC effect you're looking for.
marion.de...@fr.thalesgroup.com wrote:
> Hi everybody,
>
> I'm novice in using Kamailio and I need to know if it's possible to create
> event from kamailio.cfg to another program.
> Actual
Hi everybody,
I'm novice in using Kamailio and I need to know if it's possible to create
event from kamailio.cfg to another program.
Actually, I use MI command but I would find a way to force kamailio to send
me messages and not just responses.
Some help?
thanks marion
_
On 02/23/2009 03:50 PM, Geoffrey Mina wrote:
> Sorry about the cross post. I wasn't sure how many people were on
> both the OpenSIPs and Kamailio mailing lists... and since this is a
> 'core' issue, I figured it would be good to get input from the most
> people.
ok, but separately is better, as
I didn't really capture the signaling for the issue, I'll get a trace when I
get the chance and send it to you.
Rgds,
Uriel
On Mon, Feb 23, 2009 at 12:02 PM, Daniel-Constantin Mierla <
mico...@gmail.com> wrote:
>
>
> On 02/23/2009 02:25 PM, Uriel Rozenbaum wrote:
>
>> Sure, I can turn of report
On 02/23/2009 02:25 PM, Uriel Rozenbaum wrote:
> Sure, I can turn of report ack to check; anyway i substracter $re from
> db_extra and now I'm not seeing any crashes. But maybe its useful for
> you to clear the bug in there.
yes, this is the goal. This issue has to be fixed. You can keep the
r
Sorry about the cross post. I wasn't sure how many people were on
both the OpenSIPs and Kamailio mailing lists... and since this is a
'core' issue, I figured it would be good to get input from the most
people. In the future I will only post to one.
I will go down the path of the htable, what kin
Sure, I can turn of report ack to check; anyway i substracter $re from
db_extra and now I'm not seeing any crashes. But maybe its useful for you to
clear the bug in there.
Let me know how you want to proceed and I'll send you everithing.
-- Uriel
On Mon, Feb 23, 2009 at 10:18 AM, Daniel-Constant
Hello,
thanks. Do you have report ack set for acc module?
The crash happens to an ACK and relates to building outgoing request and
processing lumps. I will investigate and try to reproduce. Just asking
for now, would be any chance to get the ACK and eventually your config
in case it is needed
Hello,
please do not cross-post on many mailing lists. Will create confusion
about available solutions.
Theoretically, this is valid in SIP (e.g., 2 invites with same call-id)
-- it is same scenario as parallel forking in upstream.
However, if you know that this shouldn't happen, you can try t
2009/2/23 BERGANZ François :
> Hello,
>
> I come back for NAT transversal.
> First, I am sorry if you think that I email a lot!
Well, you should understand that VoIp is complex, even more hen
dealing with NAT, so there is not a magic solution and some knowledge
is required.
> After some capture
Hello,
I come back for NAT transversal.
First, I am sorry if you think that I email a lot!
After some captures...
I could see that the Nated phone tell in the 200ok that its RTP port is
192.168...
"Peer audio RTP is at port 192.168.1.82:41000"
I just have to find why the phone say it.
And ho
On Sat, Feb 21, 2009 at 5:16 AM, Juha Heinanen wrote:
> Asim Riaz writes:
>
> > yes two entries are the full output of lcr_dump, userpart of ruri is
> > starting from 0 but it does not find gateway. tried to increased the
> debug
> > level but not getting anything related to lcr.
>
> asim,
>
Elena-Ramona Modroiu schrieb:
> Hi,
>
> looking at the default config file, we deliver something pretty useless
> and very insecure.
>
> I propose to add to it several features so it becomes really usable:
> - mysql support - even some use radius for aaa, mysql is for sure used
> for usrloc.
Hello,
On 02/23/2009 08:03 AM, Jinsong Hu wrote:
> Hi, Sir:
> In the US, assume a person from 1-408-1234567 is dialing phone number
> 1234568.
> how do I figure out the caller's country code and area code and prepend to
> the local number ?
>
> I tried
> prefix($(from.user{s.substr, 0, 4}));
Hello,
On 02/21/2009 09:58 AM, catalina oancea wrote:
> Hi
>
> Does anybody know if there is a way to use the set_advertised_address
> function with an avp or variable parameter instead of plain string? I
> need to change the IP in the Via header with the value of a variable.
>
it is not possib
Hello,
do you have the ldap library and devel files installed?
http://kamailio.org/docs/modules/1.5.x/ldap.html#id2469903
Cheers,
Daniel
On 02/23/2009 06:14 AM, Husna Bte Mad Baguri wrote:
>
> Dear Whom May Concern,
>
>
>
> When I try to compile ldap module in kamailio I get error,ldap.h :No
Dear Whom May Concern,
When I try to compile ldap module in kamailio I get error,ldap.h :No such file
or directory.
Where can I get that file.please someone help me.
Regards
Husna
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