On 2/22/11 11:10 PM, Ovidiu Sas wrote:
> If a virtual PRI is set up (23 channels for NA or 30 channels for
> Europe), again the cps doesn't really count. As soon as the virtual
> PRI is maxed out (in terms of channels) all subsequent calls will be
> rejected (and the cps will be 0).
I'd suggest n
I tested setting up acc module to use the cdr_flag with dialog module.
Works nicely until opensips is restarted while there is an open call.
After an opensips restart, calls that were started before the restart do not
get an entry in the DB. Not even an old style BYE record.
I verified that the dia
On Tue, Feb 22, 2011 at 3:32 PM, Jeff Pyle wrote:
> Indeed. We've had to resort to combing the accounting database after the
> fact, and when certain minute volumes or calls-per-minute thresholds have
> been exceeded, disable the trunk. This has saved us on more than a few
> occasions.
>
> Outsi
On Tue, Feb 22, 2011 at 3:28 PM, Adrian Georgescu wrote:
> Ovidiu,
>
> With stolen account credentials one can cause major frauds during a single
> weekend without looking like a DOS attack.
That is correct, but I don't really see how ratelimitation would help
here for regular accounts.
A regula
typo "start our own"
On Tue, Feb 22, 2011 at 3:54 PM, Duane Larson wrote:
> With so many attendees we could start or own global federated Telecom
> company and beat AT&T into the ground.
>
>
>
>
> On Tue, Feb 22, 2011 at 3:51 PM, David J. wrote:
>
>> 11:01:07 AM bogdan_vs: Hi all
>> SylkServer [
With so many attendees we could start or own global federated Telecom
company and beat AT&T into the ground.
On Tue, Feb 22, 2011 at 3:51 PM, David J. wrote:
> 11:01:07 AM bogdan_vs: Hi all
> SylkServer [~sylkse...@node10.dns-hosting.info] entered the room.
> (11:01:15 AM)
> bogdan_vs has cha
11:01:07 AM bogdan_vs: Hi all
SylkServer [~sylkse...@node10.dns-hosting.info] entered the room.
(11:01:15 AM)
bogdan_vs has changed the topic to: - OpenSIPS community meeting in
progress (11:01:28 AM)
11:01:34 AM saghul: hi bogdan_vs
11:01:38 AM aidanna: howdy bogdan_vs
11:01:45 AM SylkServer:
I'd prefer to realize in the database my own routing logic and looking the
way to pass all parameters needed for accepting and directing a call (SRC
IP, SRC NUM, DST NUM, TIME and maybe SRC media format) to the database with
a SELECT statement and return the gateway or gateway list and processed ca
Indeed. We've had to resort to combing the accounting database after the
fact, and when certain minute volumes or calls-per-minute thresholds have
been exceeded, disable the trunk. This has saved us on more than a few
occasions.
Outside of fraudulant or DoS activities, it would be very useful to
Ovidiu,
With stolen account credentials one can cause major frauds during a single
weekend without looking like a DOS attack. Rate limiting of normal SIP accounts
to a few simultaneous calls or whatever is normal usage is the best defensive
strategy. Pike is not useful for non-DOS situations li
Yes you can use the permissions module to check the incoming IP. Use the
function called check_source_address() before calling do_routing.
Sven
On 2/22/11 2:05 PM, "Georgy Goshin" wrote:
> Hello!
>
> I need to replace Mera MVTS in VoIP wholesale setup, need help with it
> because I did not f
The ratelimit module was designed to deal with SIP trunks and not with
subscriber traffic.
Under normal circumstances, the subscriber traffic does not need to be
ratelimit-ed.
The pike module can be used to identify DoS traffic from a particular
subscriber and then take appropriate action.
Regards
Hello!
I need to replace Mera MVTS in VoIP wholesale setup, need help with it
because I did not found any instructions that will fit my needs. My OpenSIPS
instance should do the following:
It should route and proxy calls between SIP servers with T.38 support and
transcoding, do accounting.
Decis
Ronald,
I got into this very same question:
http://www.openser.org/pipermail/users/2009-November/009405.html
Despite my apparent enthusiasm at the time I never did implement anything
useful.
- Jeff
On 2/22/11 12:27 PM, "Ovidiu Sas" wrote:
>Yes, you can do this with opensips. Just assig
Yes, you can do this with opensips. Just assign a pipe number to an
account and use that pipe to limit traffic for that account:
http://www.opensips.org/html/docs/modules/devel/ratelimit.html#id250282
For now, there is a limit of 16 pipes:
http://www.opensips.org/html/docs/modules/devel/ratelimit
Hello all,
OpenSIPS trunk was enhanced with some new functionality for load
monitoring and load debugging. The idea is to make easier for the user
to figure out what is bottlenecking the system, whether it's the DNS,
the database connection or the actual processing of the SIP message
while go
Hi,
I have been trying to understand the workings of the MediaProxy with respect
to forwarding packets in the context of OpenSips. The accompanying
documentation and mailing list articles have been clear enough to get me to
the point where I can use MediaProxy to relay traffic between two NATed
te
Hi Dave,
2011/2/21 Dave Singer
> 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
18 matches
Mail list logo
