Greetings list,
I am trying to get profile size with jsonrpc-s module. Below is jsonrpc-s
configuration and a curl command to get the profile size.
listen=tcp:0.0.0.0:5060
loadmodule "xhttp"
loadmodule "jsonrpc-s"
modparam("xhttp", "url_match", "^/rpc_path/")
modparam("jsonrpc-s", "pretty_form
Hello,
can you get the log messages with debug=3 in kamailio.cfg for the
execution of the rpc command?
Cheers,
Daniel
On 10/11/16 09:35, Aqs Younas wrote:
> Greetings list,
>
> I am trying to get profile size with jsonrpc-s module. Below is
> jsonrpc-s configuration and a curl command to get t
Hello,
ok -- that's clear an issue due to new rules with latest versions of
mysql, definitely we need to investigate a bit more in this direction.
Cheers,
Daniel
On 10/11/16 08:56, Grant Bagdasarian wrote:
>
> Hi Daniel,
>
>
>
> Thank you! Changing to bigint in the DB fixed it. Both expires a
This is only the route block for dispatcher, but where is executed --
full content with routing blocks is more helpful.
I would suggest that you look also at the example from dispatcher docs:
-
https://www.kamailio.org/docs/modules/stable/modules/dispatcher.html#dispatcher.ex.config
It offers
Hello,
as I said before, the registrations have little to do with calls in sip,
unless there is gruu in use.
Cheers,
Daniel
On 09/11/16 18:07, Slava Bendersky wrote:
> Hello Everyone,
> I cleared registrations and tried again and issue still present.
> Client reply with 481.
>
> IP (tos 0x0, tt
If it's the client that sends the 481, then routing went ok, but the
client didn't match the dialog. Can be because it already terminated it
or callid/from-tag/to-tag mismatch.
Cheers,
Daniel
On 09/11/16 18:17, Slava Bendersky wrote:
> Based on this out put Freeswitch send BYE to kamailio and R
According to the trace, you don't route the BYE based on loose routing
rules:
1.
2016/11/09 16:55:00.788067 10.18.130.27:5060 -> 10.18.130.24:5160
2.
BYE sip:mod_sofia@10.18.130.26:5160 SIP/2.0
3.
Via: SIP/2.0/UDP
10.18.130.27;branch=z9hG4bKca09.3439664767a2d9212561e9758e87ea79.
Hello,
if by CLID you mean caller id, then you need to replace From, not To --
there is another function for it in the uac module.
Do the change in a branch_route, in that way the change is done only for
that specific outgoing request. The brach route can be re-armed from
failure_route before sen
Hello,
I guess you refer to sip_trace() -- iirc, sip_capture() only saves
locally what sip_trace() is sending over hep. If yes, then sip_trace()
is using some callbacks internally to get data at various levels of
transaction processing.
Also, if there is no port in a URI, then it is considered to
Hello,
siptrace table is completely unrelated to homer. It is the initial
version of saving sip traffic in a database.
Are you using only sip_trace() function or do you set the sip trace flag
as well?
Cheers,
Daniel
On 09/11/16 15:09, Ján Füri wrote:
> Hi guys,
>
> I'm using captagent on my al
I have set only sip_trace() without FLAG
JF
On 11/10/2016 10:23 AM, Daniel-Constantin Mierla wrote:
Hello,
siptrace table is completely unrelated to homer. It is the initial
version of saving sip traffic in a database.
Are you using only sip_trace() function or do you set the sip trace flag
a
Many thanks for the prompt reply. Below are requested logs.
root@debian:/usr/local/kamailio/sbin# Nov 10 04:56:34 debian
./kamailio[5527]: DEBUG: [ip_addr.c:229]: print_ip(): tcpconn_new:
new tcp connection: 127.0.0.1
Nov 10 04:56:34 debian ./kamailio[5527]: DEBUG: [tcp_main.c:985]:
tcpconn_new(
Hello Daniel,
My setup is proxy all requests to freeswitch via dispatcher.
Slava.
From: "Daniel-Constantin Mierla"
To: "volga629" , "sr-users"
Sent: Thursday, 10 November, 2016 04:56:53
Subject: Re: [SR-Users] BYE dispatcher
Hello,
as I said before, the registrations have little t
On Sun, Nov 06, 2016 at 02:22:06AM -0800, Gonzalo Gasca Meza wrote:
> Currently in sample configuration script, seems to be that value: $avp(oexten)
> is used to redirect to VM, but in my case this value is null.
> I didnt find any documentation for this.
>
> *Questions:*
> a) What is $avp(oexten)
Hello
I have some UAC like Panasonic PBX, that send traffic to port 6060 that
I am listening on,
but that port isn't included to R-URI.
I can see that only at tcpdump, or sip_trace from sip_capture module.
Kamailio variables like $dp or $rp have default value of 5060.
Thank You,
_
Hello,
this logic is definitely wrong -- FreeSwitch can send also a request, it
means that you send it back to it.
Only the initial request of a dialog should be routed with rules like
dispatcher/load balancer/least cost routing/... The requests within
dialog should be routed based on loose routi
Thanks Daniel
On Thu, Nov 10, 2016 at 8:04 AM, Daniel Tryba wrote:
> On Sun, Nov 06, 2016 at 02:22:06AM -0800, Gonzalo Gasca Meza wrote:
> > Currently in sample configuration script, seems to be that value:
> $avp(oexten)
> > is used to redirect to VM, but in my case this value is null.
> > I di
Hello Daniel,
I really ask for help, here are configuration file
https://paste.fedoraproject.org/477652/88413891/
I spent quite a lot of time trying understand loose_route() /record_route()
mix.
I can get signalling working, call is not disconnects, but no RTP. Or I can get
rtp and signalli
Any thoughts?
On 10 November 2016 at 15:02, Aqs Younas wrote:
> Many thanks for the prompt reply. Below are requested logs.
>
> root@debian:/usr/local/kamailio/sbin# Nov 10 04:56:34 debian
> ./kamailio[5527]: DEBUG: [ip_addr.c:229]: print_ip(): tcpconn_new:
> new tcp connection: 127.0.0.1
> Nov
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