On Friday 07 February 2014 18:18:50 jaflong jaflong wrote:
> $ru = "sip:"+$rU+"10.31.2.101:5090;transport=tls";
>
> if (!t_relay_to_tls()) {
> sl_reply_error();
> return;
> }
>
>
> My ip address dest is 10.31.2.101.
>
> Why has 130.31.2.101 been used?
Check the
I have this in my config
$ru = "sip:"+$rU+"10.31.2.101:5090;transport=tls";
if (!t_relay_to_tls()) {
sl_reply_error();
return;
}
My ip address dest is 10.31.2.101.
Why has 130.31.2.101 been used?
13(10882) DEBUG: [ip_addr.c:243]: print_ip(): tcpconn_new: new
How can I get the determine the total size of an outgoing INVITE?
The size of the body is $bs, but is there a way to get the size of all
headers? Based on the size of the total request I have to chose between UDP
and TCP, upstream provider requires IP packets>1300 to be transmitted over
TCP.
Hello,
Thanks for your reply
Here is the field
Reason: Q.850;cause=047
I've tried it in the failure_route block (using t_on_failure before
relaying a SIP INVITE).
If I print the From header using $hdr(From) ou @message.header.From, it
works, but it is not the case with @message.header.From.para
Any answer? =(
2014-02-06 Alexandr Usov :
> Integration - works.
> Problem - dialing peer to peer via Kamailio OK but with missing VARs and
> extension number, on dialing/transferring.
> Maybe you know other way to configure Asterisk dialplan for users,
> registered on kamailio and alowing dia
Dear All,
Could you please provide any possible info regarding this issue.
Best regards.
- Αρχικό μήνυμα -
Από: sipa...@in.gr
Ημ/νία: Τετάρτη, 05 Φεβρουαρίου 2014 12:11 μμ
Πρός: sr-users@lists.sip-router.org
Θέμα: [SR-Users] Msilo message body type
Dear All,
We would li
Hi,
We frequently run into the situation where a call is simultaneously
CANCEL'd by the caller and answered (2xx) by the callee. This results in
the caller not sending an e2e ACK, since it CANCEL'd the branch. It
results in retransmission of the 200 OK, since the proxy can't CANCEL
the branch
Daniel,
On an incoming call I need to get the call-id from a ringing call in a specific
dialog profile. I then add that called to a replaces header and send to a B2BUA
which connects the current call to the ringing one.
Thanks
John
From: Daniel-Constantin Mierla [mailto:mico...@gm
Hello,
in config you have access to SIP message that is currently processed and
there you can simply use $ci. Or is there a special event route where
you need the call-id?
Cheers,
Daniel
On 07/02/14 13:29, John Murray wrote:
Hi Daniel,
Yes I need it in the kamailio config.
Thanks
John
Hi Daniel,
Yes I need it in the kamailio config.
Thanks
John
From: sr-users-boun...@lists.sip-router.org
[mailto:sr-users-boun...@lists.sip-router.org] On Behalf Of Daniel-Constantin
Mierla
Sent: 07 February 2014 12:15
To: Kamailio (SER) - Users Mailing List
Subject: Re: [SR-Users
Hello,
can you paste the header here and how you used it in the config? Have
tried to print it with xlog or in some other expression? Also, in what
kind of routing block you try to get it?
Cheers,
Daniel
On 07/02/14 13:14, Tuan Viet Nguyen wrote:
Hello,
How to get ISUP cause in the 'Reason
Hello,
How to get ISUP cause in the 'Reason' header field of SIP error reply ?
I've tried a few ways but it did not work
@message.header.Reason.params["cause"]
or
$hdr(Reason) gives null
Thanks,
___
SIP Express Router (SER) and Kamailio (OpenSER) - sr-
Hello,
do you need it in kamailio config or from an external application? From
external application the mi/rpc command has to be used.
Cheers,
Daniel
On 06/02/14 21:45, John Murray wrote:
Hi,
I need to get call-id and from-tag from a call in ringing state (2).
If I use dlg_manage() and pu
Hello,
Hope you don't mind if I borrow this topic to place a question or request.
In past days I successfully setup Kamailio in my local network and made
successful WebRTC to WebRTC and SIP to SIP calls. The problem is with WebRTC
to SIP call. I also added mediaproxy-ng by following instructions f
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