Jason,
That is very strange behavior. If there is no matching transaction I
think there is no way to catch the message as reply and failure routes
are triggered by tm module.
I'm not sure since I haven't setup a stateless config.
I'm curious too if there is something that can be done here.
Dave
You know what I do?
I add append_hf("GW: BLAH") and change BLAH for each step in the if blocks.
Then you can grab a tcpdump and see your custom header. This has helped me
ENORMOUSLY during testing to track packet flow.
I prefer looking at the packets themselves and the custom headers rather
tha
Tyler,
Yes, I have already thought the same thing. However that 403 SIP Message
didn't land there in the main routing block, yet through pcap traces I could
see the SIP Message being routed to the end-user client. I had this basic
log of every new messages routed to the main route, for example:
=
I would do this by matching the method of the packet in the main routing
block, and then using textops to search for the 403 in a subsequent if () {}
block, and then just drop; or exit;
My methods are usually not the most elegant solution - I'll wait for Dave or
one of the Devs to chime in with a
Hello everyone,
This certain SIP UAS keeps on sending "SIP/2.0 403" messages to my OpenSIPS
machine out of blue, this failure/warning messages has nothing to do with
this certain client on the call. I wanted to figure out how to drop ("not
proxy") that annoying non-transactional 4XX SIP Messages b
Anca,
I'm trying to understand the use of the database in the B2B modules. So far my
application is only topology hiding (baby steps).
When it comes to shutdown / db operations, in my mind I think of it like the
dialog module. Is that a useful analogy? On my non-B2B-enabled proxies I run
db
Toyima,
Try:
yum whatprovides '*/md5' '*/md5sum' '*/db_dump'
the db_dump should be apparent from that.
I'm using Centos and it shows several packages that provide a md5
command but none put it in the path. You could try installing one and
linking it into the path to see if that helps.
eg.: ln -s
Hi,
It seems that the sipp receiver script is crashing, anyone have a decent
sipp send and receive script for a statefull opensips with record
routing?
Thanks
Chris
-Original Message-
From: users-boun...@lists.opensips.org
[mailto:users-boun...@lists.opensips.org] On Behalf Of Zahid Me
Hi everyone,
I would like to ask all of you if it is possible to use ratelimit module to
limit cps per account/group (i.e.: account A has limit of 10 cps, account
B's is 20 cps, etc.)? Is it even possible to implement this set-up on
opensips?
Thanks for any kind of help.
Regards,
Ronald
Hi,
During the call, is OPENSIP support the client codec request of switching
between audio to video or video to audio codec request to other client. I am
working on the application but i don't know about the above question on SIP
side.
Thanks for your time and cooperation.
Warm Regards
Amir Ra
What:
http://www.opensips.org/Resources/PublicMeetings
When:
22th of Feb 2011, 17:00 PM CET -
http://permatime.com/Europe/Bucharest/2011-02-22/18:00
Where:
IRC #opensips
Topic:
Meetings - how and where : is chat ok (audio conf) ? is IRC ok ?
what about SIP chat ? What is the best
oops, sorry, i sent the message without finishing :S here i go again:
I've found a "solution" for this problem, the problem was that i had the ip
addres of the Asterisk server on the table "domain";
I have this configuration on my script: (only the important section to be
checked)
if (!(method=
Hello,
I've found a "solution" for this problem, the problem was that i had the ip
addres of the Asterisk server on the table "domain";
I have this configuratíon on my script:
if (!(method=="REGISTER") && is_from_local()) /*multidomain version*/
{
if(!check_source_address("0")){
if (!pro
Chris,
Are you currently using xlog to write messages to log file? If yes, make
sure that you configure syslog to use asynchronous logging for the log facility
used by opensips.
--
Zahid
On Feb 21, 2011, at 9:06 AM, chris wrote:
Hi,
I assume you mean increase the log level above 3 wh
Hi,
I assume you mean increase the log level above 3 when you say increase the
verbosity?
I am using a completely internal test rig on a LAN.
Running the Sipp instances back to back I can create whatever call rates I like
but as soon as I introduce opensips it dies after 5cps.
Hav
Try to increase the verbosity in the log and see what happens.
You should be able to push more cps than 5, unless you have something wrong
in you cfg, or your vendor limited your gw, etc.
-Laszlo
2011/2/21 chris
> Hi,
>
>
>
> Can anyone help with a performance problem.
>
>
>
> I have a very sim
Hi,
Can anyone help with a performance problem.
I have a very simple scenario that I am testing.
Using permissions to allow fixed gateway calls and sending them to
another fixed gateway.
No registrations, no internal users just purely switching calls from one
gateway to another.
I have a
Hi all,
Some of you may know that latest Android ships with a built-in SIP
client. This client uses the deprecated maddr parameter in the R-URI
pointing to the specified outbound proxy (which you need to specify,
because it doesn't do DNS SRV resolution) for routing.
After gettting reports f
Hi Dave,
This is what i did:
root@OpenSIPS home# yum search md5
Loaded plugins: rhnplugin, security
Matched: md5
=
cyrus
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