Hi,
since yesterday we have a problem with the resourcelistserver of a sipx
4.2.1 installation. Our system monitoring threw an alert CPU 100% Load > 70.
/var/log/messages.1 contains the following information several times:
Sep 7 20:23:47 master kernel: INFO: task sipxrls:3896 blocked for more
It seam that the cause are NOTIFY messages send from ua due to
[CID=22077233] SIP/2.0 404 Subscription Does Not Exist
On Wed, Sep 7, 2011 at 9:15 PM, Joegen Baclor wrote:
> **
> I agree. I'll have that disabled by default in next release. current
> default rate limit is 50 messages per secon
I agree. I'll have that disabled by default in next release. current
default rate limit is 50 messages per second per UA. If that is being
violated by a single UA, then i suggest you investigate why it is
blocked in the first place.
On 09/08/2011 09:02 AM, Roman Gelfand wrote:
I don't know
I don't know why but rate limiting on karoo bridge caused messages from
remote worker to be blocked after a while.
Anyway, I do rate limiting from firewall. So, I added the entire internet
to while list.
My guess the issue is there should be fine-grained rate limitting based on
specific sip mess
Then by all means open a jira and ask the dev-list their opinions on this. i
would maintain you would want to try to maintain a native dialplan (no "99"
stuff). That works fine for sending a call to a gateway to get processed and
simply place a call. Transfers are more complicated. You should dial
Gee, thanks :) (About what you did with the grandstreams).
I know there is some sentiment against them, but other than this issue (and
one other one, whereby when a CALLER (not the called-party) puts a call on
hold, they're unable to retrieve it, but only if the phone is configured for
VL
sipx is a proxy. it is not a proxy for another sip system such as an
unmanaged gateway acting as its own domain. the gateway os not part of its
framework, so yeah, we're not agreeing but systemically it (the gateway) is
its own entity.
aks patton about a b2bua function on it, that might be what yo
see http://track.sipfoundry.org/browse/XX-9776 and comment accordingly.
what we've found is one or two libraries are all that are needed to do the
actual conversion, and sipxivr.java is where all the functionality and
editing need to be done. There are recents threads in the sipx-dev list that
ela
I have just hired a developer to create the ability to convert TIFF files to
PDF within sipXecs. I expect it to be ready in the next 14 to 21 days. In
its current development, it will be an all or nothing - the system all gets
TIFF, or the system all gets PDF. It will be donated to the community
Thanks, look forward to seeing what you come up with. Any plans for SipX
having the faxes in PDF as an option?
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Tony Graziano
Sent: Wednesday, September 07, 2011 5:07 PM
To: Discussion list
I have some polishing to do with my configurations. I have some small
details and custom needs to works out. When I am done I will publish my
findings, and sample config.
On Wed, Sep 7, 2011 at 5:46 PM, Ly Tran wrote:
> T.38 faxing is now working for us incoming through Voxitas. They had to
>
Counterpath seems to be a company struggling in many ways. Their reseller
program was sunk over a year ago. So, resellers such as myself has no
motivation to promote their product. Of course the support issues are now
famous.
It may well be time to start collaborating towards integration wi
Thanks Tony.
You're correct, I would not like to hear 'use Polycom handsets', as the same
issue exists for me using Grandstream, Snom, and Counterpath softphone UAs, but
specifically because I have already deployed over 100 Grandstream handsets in
production, and have another 370 on hand, re
T.38 faxing is now working for us incoming through Voxitas. They had to move
us to another server. The current one was not T.38 capable. The only thing we
have to work on now is getting reliable outgoing. Two physical fax machines
that goes through an ATA works fine, but not T.38. We have a
you don't. you do if you don't have something parsing or rewriting it
correctly., or is incapable of doing it.
sipx is a proxy. if you are connecting as an unmanaged gateway, this should
work fine. at the same tine I ha e seen this dine with older necessary pbx
systems and have not heard of call t
You realize that in the case of a dialed call (if a SipXecs user picks up a
grandstream phone and dials) to 4495, the SipXecs server DOES rewrite the
headers, and the rules exist on the patton to route the call correctly?
I'm afraid that I don't understand why I should need an SBC simply to hand
I think its important to say that sipx and sipxbridge as an sbc is designed
to handle pstn sip trunks.
I think you expect sipx and an unmanaged gateway to do more than they
should.
I do think with the patton handles a refer properly. I don't think id expect
sipx or the patton to rewrite the header
Content-Type: text/plain;
charset="utf-8"
Content-Transfer-Encoding: 8bit
Organization: SipXecs Forum
In-Reply-To:
X-FUDforum: 08063afcdd00a6e76393c5b9527381e8 <63116>
Message-ID:
Hi Tony,
So, in reading, re-reading, and re-evaluating things
based on what you and others here have said,
I have used Audio Codes when I setup SipX at Houghton College. I was just
checking to see if they still hold up well. I will look at Patton as well.
SipX and any of these should be a breeze compared to managing several Cisco
Gateways and a Unified Communications cluster. :)
Really pushing the coll
I've got slightly better experience with Counterpath.
I just send mails to supp...@counterpath.com and then wait, remind them,
wait (reat n times) and eventually, maybe, the problem gets sorted.
But for example the XMPP part of Bria is still far from stable although
it's known for about half a ye
HA! When I have complained, I've always gotten the response; "We can
fix this for you if you want to pay us for a custom build."
On Tue, Sep 6, 2011 at 9:35 AM, Worley, Dale R (Dale) wrote:
>> From: pscheep...@epo.org [pscheep...@epo.org]
>>
>> If the sip domain has more then 1 sip SRV
>> record
its all pfsense stuff.
On Wed, Sep 7, 2011 at 9:58 AM, Max DiOrio wrote:
>
>
> What are people using for their rate limit settings?
>
> ** **
>
> Can anyone provide a link to the country blocker package?
>
> ** **
>
> Thanks.
>
>
> Max
>
> ** **
>
What are people using for their rate limit settings?
Can anyone provide a link to the country blocker package?
Thanks.
Max
From: sipx-users-boun...@list.sipfoundry.org
[mailto:sipx-users-boun...@list.sipfoundry.org] On Behalf Of Nathaniel Watkins
Sent: Wednesda
I'm been doing both (well, I'm blocking the countries that are questionable) :)
- they were only able to get 3 attempts in total - so I'm actually quite
pleased - the rate limiter in conjunction with the country blocker seem to be
doing a wonderful job.
Nathaniel Watkins
IT Director
Garrett Cou
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