one sip ua sent this kind of authorization header:
Authorization: Digest
username=foo,realm=bar,uri=xxx,response=e439fa7438452da7d50690f2268be16f,nonce=4d8060d241b2430e64fe7ec681d8fb5709e9c72b8a01,qop=auth,cnonce=0004ccb0,nc=60a2.
and sr auth module didn't like it but emitted error
below is my fix to this what i consider (unless proven otherwise) a
major bug. UAs that now fail with sr auth module, worked fine in k.
if there is no misunderstanding and this is a real bug, can someone
please check the fix and commit it asap.
-- juha
***
Next time please send only the trace of the relevant SIP dialog (between
provider and Kamailio/Asterisk). Ther seconds dialog started by Asterisk
is not relevant.
The problem is rather simple:
U 2011/03/15 15:43:48.237614 6.1.1.1:5060 - 5.1.1.1:5060
INVITE sip:1231...@domain.com SIP/2.0
On 15.03.2011 16:51, Sabatella, Michael wrote:
The problem with upgrading is we have a whole provisioning system
built for ser and if upgrading changes the databse I would have to
rewrite the whole front end to integrate the system and that could
take me months..
It depends on which tables
Klaus Darilion writes:
It depends on which tables you are really using. E.g. subscriber, avpops
changed only minimal. If the provisioning system uses proper SQL
queries, then the additional columns should cause much problems. IIRC
only LCR was heavily changed.
and there is a script that
This trace is more or less useless - at least I won't waste time reading
hex-code traces. next time use proper formating:
ngrep -Wbyline -t -q -P port 5060
Anyway, it seems that iptel's NAT traversal policy always assumes, that
the user agent is directly connected to iptel, without any
Hi Alex,
As I said I was doubting about this inference, but as calls are working
with other providers and I read the post I linked, I don't really know
in what side the solution is.
The scenario is as follows:
Softphone A - providerProxy - myProxy - Softphone B
Softphone A sends the invite to
Hey Derrick,
On 15.03.2011 20:39, Derrick Ding wrote:
I find there are several links that show installation or store directory
of Kamailio. But this link works:
http://www.kamailio.org/dokuwiki/doku.php/packages:debs
However it’s not complete.
The last step is: #apt-get install
Hi,
Sorry for confusion:
You will have to define a Timeout socket, when starting RTP-Proxy (-n,
may be invalid):
bock@bock-tde:~/ims/sr-rtpp/sip-router/modules/rtpproxy/test$ cat
./exec_rtpproxy.txt
./rtpproxy -T 10 -f -F -i -l 127.0.0.1 -s udp:*:2 -d DBUG -n tcp:127.0.0.1
Recent Versions
Hello Carsten,
If you use -n tcp:127.0.0.1 without port you get:
[root@devel ~]# rtpproxy -T 10 -F -i -l 192.168.200.90 -s udp:127.0.0.1 7722
-n tcp:127.0.0.1 -d DBUG
rtpproxy: can't parse host:port in TCP address
rtpproxy: can't start notification thread
As I wrote in my previous mail, I am
Am 16.03.2011 10:55, schrieb Dominguez Jover, Ricardo:
Thanks Klaus,
I just was wondering if there was something I could configure in my proxy.
I don't think so.
Of course you could do dirty hacks, e.g. storing the clients contact in
a record-route cookie and restoring the RURI from this
Hi Alexandre,
i don't have a clue, what is going wrong there. The timeout socket,
you provide, must not necessarily point to the Kamailio-XML-RPC-Port,
in my test-cases it just pointed anywhere... (i hope it does not mess
up the XML-RPC Process of Kamailio if you connect there and do
nothing).
I
Hi Carsten,
I just tested again changing the port 8000 to point to another port. Still
does not work. I am waiting your RTPPROXY tarball to test again.
Thanks,
Alexandre.
-Mensagem original-
De: kaiserbo...@googlemail.com [mailto:kaiserbo...@googlemail.com] Em nome
de Carsten Bock
Hello,
I have a Kamailio 3.0 server with two interfaces. 192.168.x.x and a
public interface.
I send PUBLISH out on the private interface to my presence server.
Everything else happens on the public address.
I did not use the listen= option in the config, so Kamailio was
automatically
Hi Carsten,
Even with your RTPPROXY tarball I was unable to get this working. Session
remains after RTPPROXY timeout.
I am using KAMAILIO 3.1 branch from GIT and as I told you, I moved the
rtpproxy/ from GIT-MASTER to the Kamailio branch (waiting your backport). Is
there anything else regarding
Hi Alexandre,
That is strange:
I run the RTP-Proxy like this (directly from the TAR-File, i sent you)
and Kamailio with attached config-file.
bock@bock-tde:~/ims/rtpproxy$ sudo ./rtpproxy -T 10 -f -F -i -l
127.0.0.1 -s udp:*:2 -d DBUG -n tcp:127.0.0.1
rtpproxy: Timer started.
INFO:main:
Hi Klaus,
On 16/03/2011 09:17, Klaus Darilion wrote:
Contact: sip:1231000@6.1.1.1:5060
Here, Kamailio changed the received contact. As there is another proxy
between the UAC and Kamailio, Kamailio must not modify the contact.
(remove fix_nated_contact()
Carsten,
Indeed. Very strange.
Are we running the same RTPPROXY version? How can you start using '-n
tcp:127.0.0.1' without specifying a port?
[root@devel rtpproxy-carsten]# ./rtpproxy -T 10 -f -F -i -l 192.168.200.90
-s udp:*:7722 -d DBUG ERR INFO -n tcp:127.0.0.1
rtpproxy: can't parse
Hi Alexandre,
My version of RTP-Proxy is following:
bock@bock-tde:~/ims/rtpproxy$ ./rtpproxy -v
Basic version: 20040107
Extension 20050322: Support for multiple RTP streams and MOH
Extension 20060704: Support for extra parameter in the V command
Extension 20071116: Support for RTP
Carsten,
Here it goes:
[root@devel rtpproxy-carsten]# ./rtpproxy -T 10 -f -F -i -l 192.168.200.90
-s udp:*:7722 -d DBUG ERR INFO -n tcp:127.0.0.1:7723
INFO:main: rtpproxy started, pid 22495
rtpproxy: Running Timeout-Process
DBUG:handle_command: received command 22428_8 Uc0,8,101
Am 16.03.2011 15:06, schrieb David:
Hello,
I have a Kamailio 3.0 server with two interfaces. 192.168.x.x and a
public interface.
I send PUBLISH out on the private interface to my presence server.
Everything else happens on the public address.
I did not use the listen= option in the
hi
I have two questions regarding kamailio 3.1
(1) It seems the following format for rtpproxy_sock not working in
Kamailio 3.1
modparam(rtpproxy, rtpproxy_sock,
udp:localhost:12221 udp:localhost:1)
the error is:
0(11881) ERROR: rtpproxy [rtpproxy.c:957]: Name or service not known
Hi, I have an integration kamailio-asterisk. I have many calls that don't
pass through asterisk, kamailio route that calls to other media gateways. I
would like to log those calls in asterisk CDR, is that possible? Could
someone give me any direction? Thanks in advance.
Lucas
On 3/16/11 4:02 PM, Lucas Alvarez wrote:
Hi, I have an integration kamailio-asterisk. I have many calls that
don't pass through asterisk, kamailio route that calls to other media
gateways. I would like to log those calls in asterisk CDR, is that
possible? Could someone give me any direction?
Have a look at the canreinvite parameter in asterisk and see if that helps
you in any way.
On 16 Mar 2011, at 20:02, Lucas Alvarez luca...@gmail.com wrote:
Hi, I have an integration kamailio-asterisk. I have many calls that don't
pass through asterisk, kamailio route that calls to other
25 matches
Mail list logo
