Hi,
On 03/28/2012 06:37 PM, Iñaki Baz Castillo wrote:
> 2012/3/28 Min Wang :
>> In order to properly proxy the msg to GW1, Kamailio seems need to change the
>> to tag from B to A.
>
> Totally wrong. Multiple (early-)dialogs are 100% valid according to
> RFC 3261. If you find some SIP device faili
Hi Kluas and Iñaki:
Thank you a lot for the information!
min
On 03/28/2012 12:37 PM, Iñaki Baz Castillo wrote:
2012/3/28 Min Wang:
In order to properly proxy the msg to GW1, Kamailio seems need to change the
to tag from B to A.
Totally wrong. Multiple (early-)dialogs are 100% vali
Hello Krishna,
do you use git version or release 3.2 ?
I don't really understand what's a problem do you have ?
Wbr,
Alexandr
3/28/2012 12:53 PM, Krishna Kurapati wrote:
I tried to pull null terminator to see if that fixes the issue. That
is why there is shift in the lines. It is same as lin
On 3/26/12 10:54 AM, Reda Aouad wrote:
Congratulations Daniel and all the Kamailio team and developers !
Thank you all for this great community :D
Thank you, it is a result of both developers and users, the project
would haven't been here without the two working together. Of course, all
as a
On 3/28/12 12:55 PM, Krishna Kurapati wrote:
Thanks Daniel, What is a typical configuration for 1000 and 1
connections?
It depends of what is associated with a connection, if there is a
registration, then also the usrloc record has to be taken in account.
The best is that you start is an
have you changed the log line to print the port? can you print the
entire string, with proto and ip to see if there are spaces/invalid
characters (print it between special chars, like [])?
The line is inside the mirroring traffic via hep protocol, is what you
want to get, right?
Cheers,
Dani
hello,
Is there any billing application or modules to use with kamailio , for
example if i want to check the credits of the user on every passing
minute , and when he's out of credits i stop the call .
Please reply ASAP .
Thank you
___
SIP Express Rout
OK...
It worked great... and, no append_branch() was needed... only t_on_branch
with the main branch route.
When will i need to use the append_branch?
BR,
Uri
On Wed, Mar 28, 2012 at 4:00 PM, Uri Shacked wrote:
> Hi,
>
> Thanks, this was very helpful for understanding.
>
> Still let’s see if i
2012/3/28 Min Wang :
> In order to properly proxy the msg to GW1, Kamailio seems need to change the
> to tag from B to A.
Totally wrong. Multiple (early-)dialogs are 100% valid according to
RFC 3261. If you find some SIP device failing when it receives
multiple 180/183/200 responses with different
Hi,
Thanks, this was very helpful for understanding.
Still let’s see if i got it right:
I get the INVITE.
Do whatever I do on the main route.
Then do:
“append_branch();”
“changes….changes…..;”
Now before the t_relay do t_on_branch(name of branch).
Get the reply (301 in my case)
And from
Hi Karsten,
On 03/28/2012 12:52 PM, Karsten Horsmann wrote:
> Now some ask if we can act as UAC to get calls from them, for example
> authenticate
> to sipgate and feed the landline calls to our ivr system via kamailio.
>
> AFAIK the module UAC provides only one pair of user/password credentials.
We are using kamailio for user registration and authentication of users.
SIP clients get registered to kamailio with username and password, and all
their calls are forwarded to Asterisk.
Authentication in Asterisk servers is by IP of incoming calls.
Then calls are forwarded or served in place.
Mess
Thanks Daniel, What is a typical configuration for 1000 and 1
connections?
On Wed, Mar 28, 2012 at 4:04 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
> looks like you haven't reserved enough shared memory for the load you want
> to handle. Increase it via -m command line parameter, by defa
I tried to pull null terminator to see if that fixes the issue. That is why
there is shift in the lines. It is same as line 1701.
Krish Kura
On Wed, Mar 28, 2012 at 3:57 AM, Daniel-Constantin Mierla wrote:
> Hello,
>
> are you working with a custom siptrace module? the line in siptrace.c does
Hello,
my kamailio proxy handles calls between public sip-clients and
internal ivr systems.
Some customer give us calls via sip-trunks to our proxy and this goes
to the ivr too.
Now some ask if we can act as UAC to get calls from them, for example
authenticate
to sipgate and feed the landline c
Hi Daniel,
Thanks for your reply. We are going to upgrade to 3.2 so that we can make
use of the variables for now and start playing around with the devel version
ready for the next release :-)
Cheers,
Charles
_
From: Daniel-Constantin Mierla [mailto:mico...@gmail.com]
Sent
Hello,
looks like you haven't reserved enough shared memory for the load you
want to handle. Increase it via -m command line parameter, by default is
32MB which is quite low for stress testing.
Cheers,
Daniel
On 3/26/12 3:57 PM, Krishna Kurapati wrote:
I keep getting these... I am using mas
On 3/7/12 12:13 PM, Mino Haluz wrote:
Hm, the same settings works on testing 3.2.1. But this is not the
solution - I have to get it working on 3.2.0...
3.2.1 is same as 3.2.0 but with bug fixes. They work on the same
database structure and configuration file.
You can look at the changes be
Hello,
are you working with a custom siptrace module? the line in siptrace.c
does not match. If you changed the sources, paste here the lines in
siptrace.c around 1705, 10 before and 10 after.
Cheers,
Daniel
On 3/26/12 3:23 PM, Krishna Kurapati wrote:
Hi,
I keep getting this error in the s
Hello,
some time ago there were some discussions about missing symbols (not
sure if this one), the cause being a different version of perl than the
one used to generate modules_k/perl/openserxs.xs. Delete that file and
compile again.
Cheers,
Daniel
On 3/28/12 9:49 AM, Karsten Horsmann wrote
Hi,
i am using Centos 6 (32bit) and kamailio 3.2.2 build from the tar.gz.
I installed all dependencies for the modules
and the make all, make install seems to be fine.
But something is wrong with my kamailio - how can i fix this?
If i try to load the perl module i get this error messages:
ERR
There is no need to change the tag. GW1 must accept the 200 OK with
different tag, otherwise the gateway is buggy.
If the gateway is really buggy, the usual workaround is to strip the
to-tag from the provisional responses, i.e. remove the to-tag from
Asterisk's 183 response, e.g. using functio
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