Hello,
On 7/30/12 10:56 AM, Richard Zheng wrote:
Hi,
We are trying to follow
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb
to setup Kamailio as front end and multiple PBXs as back end.
We need to forward sip messages to the respective servers based on the
conta
Hello,
do you have a backtrace that you can send just to compare with the one
from Bruno?
Also, send the output of 'kamailio -V' to look at compile flags. Bruno
had a custom compile time flag, a matter of that a different memory
manager was used comparing with the default one for 3.1 series.
Hello,
On 7/30/12 7:23 PM, Iñaki Baz Castillo wrote:
2012/7/30 Daniel-Constantin Mierla :
quick question to double check if what I understood when I read the specs
was ok -- in gruu/ob, it does not matter anymore the callid/cseq
combination, or there should still be some checks related to it?
Hello,
I am planning to package v3.3.1 out of the latest branch 3.3 this
Thursday, August 2. If you are aware of issues that are not listed on
the tracker (http://sip-router.org/tracker/), add them there to see what
can be sorted out before.
Cheers,
Daniel
--
Daniel-Constantin Mierla - http
2012/7/30 Daniel-Constantin Mierla :
> quick question to double check if what I understood when I read the specs
> was ok -- in gruu/ob, it does not matter anymore the callid/cseq
> combination, or there should still be some checks related to it?
In fact that depends on Outbound instead of GRUU, a
Hello,
On 7/30/12 6:51 PM, Peter Dunkley wrote:
Hi Iñaki,
I'll stop using the mangling and aliasing as soon as Kamailio supports
Outbound. But for now... it works :-)
for such scenario, the gruu support is enough -- the problem is
selecting the right destination address in case of natted co
Hi,
Bellow is output of 'kamailio -V'...
--
version: kamailio 3.1.2 (i386/linux) 4d9f90
flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS,
DISABLE_NAGLE, USE_MCAST
Hello,
On 7/30/12 5:21 PM, Iñaki Baz Castillo wrote:
2012/7/30 Daniel-Constantin Mierla :
After playing a bit more with it, I pushed some commits on master branch.
Can you try to see if they fixed the issue you reported?
Hi Daniel, we are testing it. It looks ok for now but we need further
tes
Hi Iñaki,
I'll stop using the mangling and aliasing as soon as Kamailio supports
Outbound. But for now... it works :-)
Peter
> 2012/7/30 Peter Dunkley
>
>> With these changes (and a client that supports it) should I now be able
>> to
>> use WebSockets without the aliasing from nathelper?
>
>
>
2012/7/30 Peter Dunkley
> With these changes (and a client that supports it) should I now be able to
> use WebSockets without the aliasing from nathelper?
Hi, there should be no difference with the case of SIP over TCP or TLS. The
main benefict of using Outobund and GRUU in clients (with reliab
Hi,
With these changes (and a client that supports it) should I now be able
to use WebSockets without the aliasing from nathelper?
Regards,
Peter
On Mon, 2012-07-30 at 17:35 +0200, José Luis Millán wrote:
> Hi Daniel
>
>
>
> It works as expected. Binding is updated correctly now!
>
>
> Th
Hi Daniel
It works as expected. Binding is updated correctly now!
Thank you very much.
Regards
2012/7/30 Iñaki Baz Castillo
> 2012/7/30 Daniel-Constantin Mierla :
> > After playing a bit more with it, I pushed some commits on master branch.
> > Can you try to see if they fixed the issue you r
2012/7/30 Daniel-Constantin Mierla :
> After playing a bit more with it, I pushed some commits on master branch.
> Can you try to see if they fixed the issue you reported?
Hi Daniel, we are testing it. It looks ok for now but we need further
testing, please let us giving a better feedback today or
Hello,
It looks like the r-uri in the trace is different from the one kamailio
considers. From the ACK message captured with ngrep:
ACK sip:200.87.137.150:5060;user=phone SIP/2.0
But form the logs:
Jul 30 09:15:00 theseus-test /usr/local/kamailio/sbin/kamailio[1577]:
DEBUG: [parser/msg_parser.c:
Thank's for feedback Daniel,
This core is add to the tracker with ID 247...
Best Regards
2012/7/30 Daniel-Constantin Mierla
> Hello,
>
> it is in my to-do list to investigate, so far I was not able to reproduce.
>
> Vacation time together with other travelings makes a bit more slower
> proce
After playing a bit more with it, I pushed some commits on master
branch. Can you try to see if they fixed the issue you reported?
Cheers,
Daniel
On 7/30/12 12:06 PM, Daniel-Constantin Mierla wrote:
Hello,
can you send the requests for registration and re-registration (ngrep
with -W byline o
Hello,
can you send the requests for registration and re-registration (ngrep
with -W byline or pcap) in order to test them here? I tried to reproduce
with an UA I have here and the registration update does the right thing.
Cheers,
Daniel
On 7/27/12 1:43 PM, José Luis Millán wrote:
Yes,
Ver
Enable debugger module with cfgtrace and see what actions of your config
file are executed. Maybe there is a mistake in your routing logic.
Cheers,
Daniel
On 7/30/12 11:27 AM, Tayeb wrote:
Domain added
But same
Envoyé de mon iPhone
Le 30 juil. 2012 à 09:06, Daniel-Constantin Mierla a écrit :
Hello,
the log message shows that the transaction is not found. Is the ACK
coming late after 200ok ? There is a tm module parameter that you can
adjust to prolong the time a transaction is kept after completion, if
that is the case.
Also, be sure the INVITE is sent using tm functions, not st
Domain added
But same
Envoyé de mon iPhone
Le 30 juil. 2012 à 09:06, Daniel-Constantin Mierla a écrit :
> Hello,
>
> if your config is based on the default one, the 403 not relaying is sent when
> both caller and callee are not local users, preventing the proxy to act as an
> open relay and b
This is the code that's being executed:
route[WITHINDLG] {
if (has_totag()) {
# sequential request withing a dialog should
# take the path determined by record-routing
xlog("ESTAMOS EN WITHIN\n");
if (loose_route()){
xlog("LOOSE ROUTE DETECTED\n");
Hi,
We are trying to follow
http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdbto
setup Kamailio as front end and multiple PBXs as back end.
We need to forward sip messages to the respective servers based on the
contact. For example, users 101, 102 go to PBX at 1.1.1.1 and
Hello,
if your config is based on the default one, there is a check for
associated INVITE transaction and if that does not exist, then the ACK
is droppend.
You can use debugger module with cfgtrace set on in order to see what
actions in the config file are executed. That will help to see if
Hi Daniel,
This is the ACK message:
U 2012/07/30 04:23:31.604721 79.170.68.151:5060 -> 79.170.68.157:5060
ACK sip:200.87.137.150:5060;user=phone SIP/2.0.
Via: SIP/2.0/UDP 79.170.68.151:5060
;branch=z9hG4bK334faa4497ll114a52eACK450932302031.
Max-Forwards: 70.
Route: .
To: ;tag=ldb0cbn6-CC-23.
From
Hello,
these are only the error messages, with debug=4, there should be lot of
DEBUG messages that will help to see what parts of code were loaded.
Cheers,
Daniel
On 7/24/12 9:30 AM, Uri Shacked wrote:
this is the log when debug is 4...
Jul 24 10:19:37 RNDSRV kamailio: : [cfg.y:353
Hello,
if your config is based on the default one, the 403 not relaying is sent
when both caller and callee are not local users, preventing the proxy to
act as an open relay and be abused to do attacks.
So, first, be sure you have the domains you serve as aliases to match
the myself conditio
Hello,
first, such race can happen always and it is ok from sip rfc point of
view. The carrier UA should have received the BYE from the other side
and close the dialog, then ignore the rest. So it is a broken UA
implementation imo.
Let's say you just drop the 481, then the BYE will time out
Hello,
what email client application do you use? By hitting reply all button,
the email client should preserve the thread id in a header (which is not
visible unless you show the headers -- usually an option to most of
email clients).
Anyhow, the right way to reply is to be sure the mailing
Hello,
can you add a log message to print the source ip, call id and r-uri?
It may happen that the ACK is looping back if r-uri is pointing to itself.
Also, try to get the ngrep on all devices, like:
ngrep -d any -qt -W byline port 5060
Pasting the ACK request here will help to see if somethi
Hello,
it is in my to-do list to investigate, so far I was not able to reproduce.
Vacation time together with other travelings makes a bit more slower
process. Please add it also to the tracker:
http://sip-router.org/tracker/
A minor release for 3.3 branch is going to be soon, being there ma
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