de to send traffic to another node.
Likely a bug or misconfiguration in the SBC.
Definitely looks like it. Thanks for taking a look Daniel!
Cheers,
Daniel
On 17/01/2017 19:40, Andres wrote:
Hi,
I am troubleshooting some strange call setup problems using an
Edgewater Networks SBC. A coup
Hi,
I am troubleshooting some strange call setup problems using an Edgewater
Networks SBC. A couple of Polycom phones sit behind the SBC and they
connect to a SIP Server in the cloud. The SBC is in passthrough mode.
What I see that I am not sure about is:
INVITE from SIP Server to SBC IP (n
Andres-27 [via SIP Router] yazdı:
On 4/8/16 5:27 AM, ycaner wrote:
Hello Daniel;
source and destination is different machines and ips. they are
sniffed by tshark and they are the same and truncated.
Maybe they are not truncated by rather fragmented. If you are
capturing packets by UDP Port
On 4/8/16 5:27 AM, ycaner wrote:
Hello Daniel;
source and destination is different machines and ips. they are
sniffed by tshark and they are the same and truncated.
Maybe they are not truncated by rather fragmented. If you are capturing
packets by UDP Port you will not see the fragments.
On 2/2/16 1:36 PM, Rene Montilva wrote:
Daniel
checking the frame size trace, i notice when kamailio receive packets
less than 1100, it response with packets more than 1300 , but when the
packets are more than 1100, kamailio send packets less 1000 and
connection to softphone fail
You are go
On 9/16/15 4:00 AM, Sebastian Damm wrote:
Hi,
we have a load balancer which is handling a lot of SIP traffic all
day. There's always 20-40 Mbit SIP traffic going through. From time to
time we see in our logs messages like these:
Sep 16 09:46:28 ecker /usr/sbin/kamailio[25505]: ERROR:
[udp_
On 4/30/15 9:28 AM, Alex Balashov wrote:
No, that's not correct. The provider needs to send DNIS in the RURI in these
cases, and providers should have a setting to enable this. It does require
overriding the Contact binding of the registrant (if applicable), which is not
RFC-compliant, but tha
On 4/30/15 7:35 AM, Alex Balashov wrote:
On 04/30/2015 07:31 AM, Andres wrote:
I am inclined to believe this is perfectly normal and compliant but let
me know what you think.
Yep, it's normal. Moreover, only the RURI value should be used for
routing purposes or for anything else t
I have a general question maybe somebody can help me out with. We have
a new SIP Trunk setup with a provider. The SIP Trunk has a username of
'jane' and it handles 400 DIDs. When the incoming INVITE from the
provider comes in, the URI in the Invite is the username of the trunk
while the To h
On 4/18/15 8:22 AM, Daniel-Constantin Mierla wrote:
Hello,
On 18/04/15 14:15, Andres wrote:
Hi,
I have a question about about 'myself' usage.
for example:
if(uri==myself) {
log("the request is for local processing\n");
};
I have always expected the ab
Hi,
I have a question about about 'myself' usage.
for example:
if(uri==myself) {
log("the request is for local processing\n");
};
I have always expected the above to hold true for any 'alias' definition
in the config file. But I never really though about the domain module
sin
In the documentation at:
http://kamailio.org/docs/modules/4.2.x/modules/rtpproxy.html
I see a variable called $rtpstat which "Returns the RTP-Statistics from
the RTP-Proxy. The RTP-Statistics from the RTP-Proxy are provided as a
string and it does contain several packet-counters."
But I cann
On 4/8/15 9:14 AM, Grant Bagdasarian wrote:
I misunderstood the reason for the domain module. I thought it was
used to store (remote) domains the proxy should handle and allow.
And by domains I mean remote addresses. For instance, when providing
customers with a SIP trunk and the Kamailio be
On 4/2/15 8:44 AM, Daniel-Constantin Mierla wrote:
On 02/04/15 14:29, Andres wrote:
On 4/1/15 11:10 PM, Alex Balashov wrote:
How complex is the SER configuration? SER couldn't do that much, by
the standards of the modern feature set, so there may not be much to
port. :-)
It is very s
On 4/2/15 8:44 AM, Daniel-Constantin Mierla wrote:
On 02/04/15 14:29, Andres wrote:
On 4/1/15 11:10 PM, Alex Balashov wrote:
How complex is the SER configuration? SER couldn't do that much, by
the standards of the modern feature set, so there may not be much to
port. :-)
It is very s
from one version to another. If I have to
write a script I will but before I do that I wanted to check in an see
if somebody has already done it even if for an older version of Kamailio.
Thanks.
On 1 April 2015 22:58:14 GMT-04:00, Andres wrote:
How would one even approach the daunting
How would one even approach the daunting task of upgrading a fully
operational Sip Express Router 0.9.6 installation with thousands of
users that has been running unmodified for over 10 years.
I would like some pointers on how to approach the mysql database
migration from the old schema to the
On 2/18/15 9:44 PM, Will Ferrer wrote:
Hi Alex
Thanks so much for the reply.
Is there anything that we could do perhaps that is a more creative
solution, for instance not passing the re-invite all the way to the
softphone and just responding from the kamailio box handling the call?
We tried
On 9/25/14, 9:17 AM, Igor Potjevlesch wrote:
I just identified that if "IP_KAMAILIO/ASTERISK" is set into domain table,
As I said before, what you need to use is IP_KAMAILIO/ASTERISK:5060.
If you use it without the port, it will match also when directed to your
Asterisk server residing on the
On 9/24/14, 12:41 PM, Igor Potjevlesch wrote:
Hi Klaus,
Sorry for the bad format.
I took a debug:
DEBUG: [parser/msg_parser.c:623]: parse_msg(): SIP Request:
DEBUG: [parser/msg_parser.c:625]: parse_msg(): method:
DEBUG: [parser/msg_parser.c:627]: parse_msg(): uri:
:4060>
DEBUG: [parser/
On 9/23/14, 8:47 AM, Igor Potjevlesch wrote:
Hello,
I'm still experienced this issue where the port looks to be rewritten.
It's like the ACK is not consider to be in loose route.
I loops inside because you probably do not have the port number in the
alias definition. Make sure you put like
On 8/29/14, 1:42 PM, Alex Villacís Lasso wrote:
Please consider the following SIP packet exchange, as seen by a
tcpdump running on 201.234.196.170. Here 198.58.101.75 initiates a
call to 201.234.196.170:
IP 198.58.101.75.5060 > 201.234.196.170.5060
INVITE sip:*43@201.234.196.170:5060 SIP/2.0
On 3/26/14, 2:40 PM, Rainer Piper wrote:
Hi Andres,
today I had a very funny one ... an amazon server tried to relay over
my server.
I see that. Its cheap and easy to use an Amazon server for this
purpose. Plus you can change its public IP by shutting down and
starting the instance again
On 3/26/14, 2:27 AM, Rainer Piper wrote:
Hi Aryn,
changing the standard Listen Port 5060 to something like 5871 will
keep approximately 50% of the bad boys away.
Log user agent client name like
if
($ua=~"friendly-scanner"||$ua=~"sipcli"||$ua=~"sundayddr"||$ua=~"sipsak"||$ua=~"sipvicious"||$
smime.p7s
Description: S/MIME Cryptographic Signature
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e rtpproxy will match the new stream from the
client immediatly.
--
Technical Support
http://www.cellroute.net
On 5/13/13, Andres wrote:
On 5/11/2013 4:29 PM, hiro wrote:
using kamailio-4.0.1_src.tar.gz with rtpproxy and a nokia e72 behind
NAT registered via UDP I get no voice.
The e72 st
On 5/11/2013 4:29 PM, hiro wrote:
using kamailio-4.0.1_src.tar.gz with rtpproxy and a nokia e72 behind
NAT registered via UDP I get no voice.
The e72 strangely sends a single udp packet from a wrong port (49152)
before the rtp stream should start.
This quirk of the e72 doesn't seem to work well w
great support.
Andres Collazos.
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There is just one registrar -- the one under discussion. It's the
main reason Kamailio is used here; it is the only OSS registrar I
know of that has enough throughput capacity to sustain thousands of
devices banging on it with relatively short re-registration intervals.
Hi Alex,
At this
On 11/17/2011 10:28 PM, Alex Balashov wrote:
I have a 1.5.3 installation functioning as a registrar that is
exhibiting a very curious, if only infrequent set of behaviours.
The host is CentOS 5.x, with PostgreSQL 9.0 backing usrloc, and
db_mode = 1 (immediate write-through).
I have a registrat
cribe. The solution was
to respond immediately to keep alive messages without any further
processing.
Andres
http://www.telesip.net
Thanks!
//Anders
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