Hello,
As it happens we run Kamailio 3.2.0 on Debian Lenny which for some
reason does not support uac_replace_to()!
Anyhoo, we are trying to change the domain part in the To URI. What is
the best way to do it ?
Thank you!
___
SIP Express
Hello,
On 3/31/13 4:27 PM, Ramaseshi reddy kolli wrote:
Hi,
I am using sip_trace function after next_branches(), it is not
capturing outbound SIP INVITE request can anyone please help me.
iirc, you have to set a flag to trace outgoing messages. the sip_trace()
function just takes the
Hello,
On 4/4/13 10:43 PM, Krishna Kurapati wrote:
Hi,
With TCP, there is no way Server can establish a TCP connection
through client's Firewall/NAT.
So, when Client connection is disconnected from the server due to
Network issues, the Registration should be removed.
I tried to handle this
Hello,
On 4/4/13 9:15 PM, Richard Fuchs wrote:
Hi,
On 04/04/13 14:58, Daniel-Constantin Mierla wrote:
quite interesting, I didn't know it has two operations modes: user space
forwarding and kernel forwarding.
Is there any plan in supporting more one mode (or dropping the other) in
the
Hello,
On 4/4/13 2:32 PM, qm...@polarismail.com wrote:
Hello,
As it happens we run Kamailio 3.2.0 on Debian Lenny which for some
reason does not support uac_replace_to()!
Anyhoo, we are trying to change the domain part in the To URI. What is
the best way to do it ?
if you don't need to
On 04/05/13 03:53, Daniel-Constantin Mierla wrote:
She fallback to user space can happen even during a call? Or is just
about when the call is initialized, the application detects is some
problem when setting up forwarding rules in the kernel and goes for user
space.
It can happen any time.
Sorry if I repeat myself - I just subscribed properly to the list:
if you don't need to change it back for replies and next requests, try:
$td = xyz.com;
IIRC, that is available on 3.2.x. If not, either you upgrade or use
replace functions from textops.
Cheers,
Daniel
Thank you for
Let me also explain a bit what I'm trying to fix through this method.
We have an Asterisk box that forwards call through kamailio to one of
our carriers. That works great. The calls are being forwarded with a
To of: 5551112...@kamail1.domain.com towards our carrier
We also added another
check sdp's at all pit stops
On Apr 1, 2013 10:24 PM, indef...@yahoo.gr wrote:
Hello,
I have the following topology.
Kamailio as SIP Proxy and Asterisk as B2BUA. In Kamailio I use rtpproxy
for NAT users.
The problem that I have is the following:
-When the CSipSimple is registered
Hi,
PID_user = P-Asserted-Identity, user part
Proto = type of protocol (UDP, TCP)
Family = protocol family (IPv6, IPv4)
RTP_stat = X-Rtp-Stat / P-Rtp-stat header
Node = capture node ID
Wbr,
Alexandr
From: sr-users-boun...@lists.sip-router.org
Changes are not immediately visible, this is the internal design from
the time of SER back in 2001 -- I made a FAQ entry with more details:
-
http://www.kamailio.org/wiki/tutorials/faq/main#why_changes_made_to_headers_or
In short, you have to use msg_apply_changes() function after changing
Mersi mult Daniel! It worked but unfortunately it didn't fix the issue.
As I mentioned previously the original setup is like this:
Asterisk - Kamailio gateway - carrier. All calls go through just fine
The new setup we are trying is:
Asterisk - Kamailio load balancer ( dispatch module ) -
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