;
> I would stick with asterisk handling the mwi subscription and
> configure kamailio as a relay only for those events.
>
>
> Regards,
> Ovidiu Sas
>
> On Thu, Jun 30, 2011 at 9:30 AM, Spinov Evgeniy
> wrote:
> > This makes the main problem as in documents I
This makes the main problem as in documents I've found on asterisk is
that it doesn't support PUBLISH, i.e. there is no way to force sending
them on MWI event. It works only on SUBSCRIBE-NOTIFY scheme.
Did you succeeded with PUBLISH on Asterisk in order to inform Kamailio
about MWI? If yes, how? C
Hello.
Trying to understand how to work with the presence_mwi module. It seems
to me, that it's just for my purpose - inform subscribed peers about
waiting messages. But it do not export any functions, noting.
The scheme is simple: Peer->Kamailio->Asterisk.
Asterisk can accept voicemail and when
gt;
>> Thanks in advance.
>>
>>
>>
>> ___
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users@lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr
Hello, everyone.
I've accidentally started 2nd copy of Kamailio and killed it at once.
However, fifo file was deleted by killed Kamailio, while the first one
copy is running fine.
Is there any way to restore the fifo file to the first kamailio, unless
restarting it?
Thank you.
___
Hello, all.
Is there any possibility to set one of dispatcher destinations to
probing mode manually? I can set it to active/inactive, however cannot
set it to probing via MI.
This is really required as, for instance, when asterisk reload takes
like 1 minute to reload ( huge DB ), I want to set n
Hello, everyone.
I've got stuck with a small problem with SUBSCRIBES.
Let's say we have 2 separate customers with IDs: 1 and 2.
They have 2 extenstions each: 1_100, 1_101 and 2_100, 2_101.
The task is to subscribe 1_100 to 1_101. When I assign on DSS ( BLF )
key, phone number 1_101, then everyt
Looking good. I've rewrited variables and it parses well. Script
continues to run just after call of msg_apply_changes().
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
sr-users@lists.sip-router.org
http://lists.sip-router.o
On Thu, 2011-03-24 at 14:14 +0100, Daniel-Constantin Mierla wrote:
> Are you calling msg apply changes after record_route()? Try to do
> record_route() after since it tries to discover the outgoing local
> interface which might not be set at that point.
This has fixed the problem.
Also I'm wond
r.
If I've provided you not enough info to find out the reason, I can do
the full dumps, as problem is easily repeated.
Thank you.
>
> Cheers,
> Daniel
>
> On 3/24/11 6:48 AM, Spinov Evgeniy wrote:
> > On Wed, 2011-03-23 at 15:16 +0100, Daniel-Constantin Mierla wrot
the
> message and then call msg_apply_changes() only when the flag is set, e.g.,
>
> remove_hf("From");
> setflag(20);
>
> ...
>
> if(isflagset(20))
> msg_apply_changes();
>
> However, I will update the sources so such log messages will not appear
> unless th
Hi,
I've installed, as you advised Kamailio 3.1.2 and used
msg_apply_changes(). However, even when I'm not changing anything in the
packet and just calling msg_apply_changes() I get errors from K:
7(31755) : [msg_translator.c:519]: ERROR: lump_check_opt: null
send socket
7(31755) : [msg_trans
. As
result - complete silence.
Please advice, where to dig?
On Wed, 2011-03-02 at 09:53 +0100, Daniel-Constantin Mierla wrote:
>
> On 3/2/11 9:32 AM, Spinov Evgeniy wrote:
> > Unfortunately ngrep is unavailable right now, cause network was
> > configured to use public IPs. May be I
On Wed, 2011-03-02 at 09:53 +0100, Daniel-Constantin Mierla wrote:
>
> On 3/2/11 9:32 AM, Spinov Evgeniy wrote:
> > Unfortunately ngrep is unavailable right now, cause network was
> > configured to use public IPs. May be I'll can do that on development
> > network
eave
> the ports. Seeing SIP headers and SDP can indicate the presence of an
> ALG or something broken in config logic.
>
> Also, what is the parameter you give to force_rtp_proxy(...)?
>
> Cheers,
> Daniel
>
> On 3/2/11 8:38 AM, Spinov Evgeniy wrote:
> > M
May be I miss some important details? No suggestions?
Thank you.
> Hello, all.
> Using nathelper + rtpproxy for subj. Kamailio has public and private
> network interfaces. Asterisk is only private. RTP Proxy is working in
> bridge mode and relaying traffic from UAC to Asterisks.
> Everything is
Hello, all.
Using nathelper + rtpproxy for subj. Kamailio has public and private
network interfaces. Asterisk is only private. RTP Proxy is working in
bridge mode and relaying traffic from UAC to Asterisks.
Everything is working fine, except one configuration. When the client is
behind router ( a
do what you want, explicitly force the outgoing
>interface by manipulating the $fs pseudovariable prior to
>relaying/forwarding, e.g.
> $fs = "udp:xxx.xxx.xxx.xxx:5060";
> t_relay(); ...
> etc.
>Cheers,
>-- Alex
>On 02/18/2011 10:05 AM, Spinov Evgen
warding, e.g.
> $fs = "udp:xxx.xxx.xxx.xxx:5060";
> t_relay(); ...
> etc.
>Cheers,
>-- Alex
>On 02/18/2011 10:05 AM, Spinov Evgeniy wrote:
>> Hello, all.
>>
>> I'm having Kamailio 3.0.4. Server have 2 physical interfaces, one is
>
Hello, all.
I'm having Kamailio 3.0.4. Server have 2 physical interfaces, one is
private, second is public.
INVITE request is coming to Kamailio from public interface and being
routed by Kamailio to one of ther asterisk boxes, which is in private
network. Packet is routed to correct interface, bu
>Hello,
>with the latest version there are alternatives you can use:
>> On 12/10/09 5:06 PM, David wrote:
>> Hey,
>>
>> I won't pretend to be an expert in Kamailio, someone will probably
>> suggest a better way. But here is how I rewrote my SIP packet's TO
>> header before relaying it to the ne
+0100, Klaus Darilion wrote:
>
> Am 11.02.2011 07:57, schrieb Spinov Evgeniy:
> > Thank you for detailed explanation, now I've got it, but I'm confused
> > about workaround. Really nothing could be done to send phone offline or
> > get it online when it
tried something like that?
On Fri, 2011-02-11 at 01:28 +0100, Klaus Darilion wrote:
> Spinov Evgeniy wrote:
> > On Wed, 2011-02-09 at 16:24 +0100, Klaus Darilion wrote:
> >
> > Thank you for the links, especially this one
> > http://www.kamailio.org/docs/module
o never
come up in request.
Dialog processing is very simple:
if (is_method("INVITE|REGISTER")) {
dlg_manage();
}
Somewhy, pua_dialoginfo comes with INVITE request and never comes with
REGISTER.
Any ideas?
>
> Am 09.02.2011 14:56, schrieb Spinov Evgeniy:
&g
ialog are empty, cause there are
not PUBLISH event: dialog?
>
> Am 09.02.2011 14:16, schrieb Spinov Evgeniy:
> > Hello all.
> >
> > I'm having problem with sending NOTIFY packet for event dialog.
> >
> > The problem is, that all NOTIFY packets sent with t
Hello all.
I'm having problem with sending NOTIFY packet for event dialog.
The problem is, that all NOTIFY packets sent with this event is empty.
( Content-length: 0 ) and in fact, not passing actual status to the
phone. In my case this is SPA962.
NOTIFY packets for event presence are sent fine
Mierla
wrote:
> Hello,
>
> On 12/2/10 1:42 PM, Spinov Evgeniy wrote:
>> Hello.
>>
>> I have Kamailio ( K in further ) and 2x Asterisk boxes ( A1 and A2 in
>> further ) configured, so UAC registers at K and when it sends a call,
>> it's routed to A1 or A2, b
t dumps and they are correct. If you
like, I can put them here, but it senseless, cause UA is replying in same
way every time:
1. -> INVITE
2. <- 407 from K
3. -> ACK
4. -> DIGEST
Hope this will help.
On Fri, 03 Dec 2010 11:20:38 +0100, Daniel-Constantin Mierla
wrote:
> Hello,
&g
Hello.
I have Kamailio ( K in further ) and 2x Asterisk boxes ( A1 and A2 in
further ) configured, so UAC registers at K and when it sends a call,
it's routed to A1 or A2, balanced.
The problem is, that I cannot find how to authorize INVITE requests, so
unregistered UAC could not send INVITE requ
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