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
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
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,
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
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,
_
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 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
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(
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
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
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,
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
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.
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
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
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,
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
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
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
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