Hi All
I am using OpenSIP 1.6.2.
I would like to register 20 Million users on OpenSip..
Question ...
1] How many registration is possible ? Is it possible to register 20 Million
users on 1 OpenSip Server.
2] How many simultaneous call will be processed at a time.
3] And what about Voice
Hi,
I've configured OpenSIPs with Radius and now working to configure NAT on
OpenSIPs using module mod_nathelper. After configuring, I'm getting
following errors as listed down below;
Apr 1 11:53:31 rose /usr/local/sbin/opensips[11386]:
ERROR:nathelper:select_rtpp_node: script error -no valid
James Lamanna wrote:
On Wed, Mar 31, 2010 at 9:16 PM, James Lamanna jlama...@gmail.com wrote:
On Wed, Mar 31, 2010 at 10:28 AM, Anca Vamanu a...@opensips.org wrote:
James Lamanna wrote:
Anca Vamanu Wrote:
Andrew, this patch is already in 1.6.2 and trunk.
James,
While testing with drouting function is_from_gw(), it seems that in addition
to be present in the dr_gateways table, there must also exist at least one
entry in dr_rules referencing the GW otherwise the function returns false.
Is this intentional?
In my case, I have incoming-only GWs so there are
James Lamanna wrote:
On Wed, Mar 31, 2010 at 9:28 PM, James Lamanna jlama...@gmail.com wrote:
On Wed, Mar 31, 2010 at 9:16 PM, James Lamanna jlama...@gmail.com wrote:
On Wed, Mar 31, 2010 at 10:28 AM, Anca Vamanu a...@opensips.org wrote:
James Lamanna wrote:
Anca
On Thu, Apr 1, 2010 at 4:28 PM, mayamatakeshi mayamatake...@gmail.comwrote:
While testing with drouting function is_from_gw(), it seems that in
addition to be present in the dr_gateways table, there must also exist at
least one entry in dr_rules referencing the GW otherwise the function
Hi all,
This mail is a request for your comments about presence aggregation.
What is your opinion about this? Do you consider it to be a real problem
and which could be the solutions?
Although it is possible to have more clients registering for the same
account and publishing presence
Hello Ag-projects Team,
I would like to know if the next version of CDRTool will be fully-
compatible with opensips 1.6.
In a test environment I have CDRTool 7.0.2 and the following error
(because opensips 1.6 doesn't use anymore the trusted table):
Database DB_opensips error: Table
Hi,
On 1/4/10 11:48 AM, Carlo Dimaggio wrote:
Hello Ag-projects Team,
I would like to know if the next version of CDRTool will be fully-
compatible with opensips 1.6.
In a test environment I have CDRTool 7.0.2 and the following error
(because opensips 1.6 doesn't use anymore the trusted
Hi Takeshi,
To confirm :
1) is_from_gw() check only the dr_gateway table, without checking if the
GW are used by any rules
2) if the dr_rules table is empty, the DR engine will not load any info
(like gws) at all.
Regards,
Bogdan
mayamatakeshi wrote:
On Thu, Apr 1, 2010 at 4:28 PM,
Hello Ahmed,
you script does not configure any rtpproxy to be used - the
rtpproxy_sock parameter is empty:
modparam(nathelper,rtpproxy_sock,)
You need to set a valid link to a running rtpproxy :
http://www.opensips.org/html/docs/modules/1.6.x/nathelper.html#id228332
Regards,
This problem is hard to solve in a deterministic way, practically
there is conflicting information and in general conflicts require a
manual decision to solve. Asking users to set priorities in their UA
is not something you can rely upon, people simply forget or machines
go to sleep or
Hi,
Chandrakant Solanki wrote:
Hi All
I am using OpenSIP 1.6.2.
I would like to register 20 Million users on OpenSip..
Question ...
1] How many registration is possible ? Is it possible to register 20
Million users on 1 OpenSip Server.
theoretically you can put 20 M of users if you
On Thu, Apr 1, 2010 at 4:03 PM, Bogdan-Andrei Iancu
bog...@voice-system.rowrote:
Hi,
Chandrakant Solanki wrote:
Hi All
I am using OpenSIP 1.6.2.
I would like to register 20 Million users on OpenSip..
Question ...
1] How many registration is possible ? Is it possible to
Use MediaProxy unless you have a specific need that cannot be
addressed by its default configuration. You need to add a single line
in your routing configuration to enable it.
Adrian
On Apr 1, 2010, at 12:39 PM, Chandrakant Solanki wrote:
For voice quality, should I go for RTP Proxy /
Hi Franck,
I guess the xmlrpc library changed and this is why is not compiling
anymore
I personally tried up to version 1.06.27 and still works.
So, you have 2 options:
1) use an rpm, but not newer than 1.06.27
2) compile from sources 0.9.10
Regards,
Bogdan
Madovsky wrote:
well
Hi Anca,
On 1/4/10 10:02 AM, Anca Vamanu wrote:
Hi all,
This mail is a request for your comments about presence aggregation.
What is your opinion about this? Do you consider it to be a real problem
and which could be the solutions?
Although it is possible to have more clients registering
Hi Brett,
Based on the RFC quotes, I would say the serialize function is
considering 0.0 as higest priority and RFC suggest the other way around.
Could you try the attached patch to see if it fixes the problem ?
Regards,
Bogdan
Brett Nemeroff wrote:
Hi All,
I'm trying to figure out an
Hi Bogdan,
Are you able to reproduce the issue?
Regards,
Daniel
On Tue, Mar 30, 2010 at 1:03 PM, Daniel Ribeiro
ribeiro.dani...@gmail.comwrote:
Bogdan,
Attached there is :
opensips.cfg
usr_preferences
core dump file for version 1.6
wireshark with call scenario
Thanks,
Daniel
--
Hi
I think the topic touches client-side implementations more. On the server-side,
it is the best to distribute all information and leave it up to the client to
interpret it. This is also how XMPP does it.
-Original Message-
From: users-boun...@lists.opensips.org [mailto:users-
Il giorno 01/apr/10, alle ore 12:02, Saúl Ibarra Corretgé ha scritto:
You need to create a MySQL view to 'emulate' the trusted table:
[...]
PS: The view was suggested by Jeff Pyle IIRC :)
Ok very good! Thank you Saúl and Jeff :)
Best regards,
Carlo Dimaggio
Hi Rajib,
Maybe you can run a parallel network tracer to dump all the SIP traffic
(on proxy). At the end of the your test, you can list the dummy calls
and correlate them with the signalling (to see if the signalling for the
that calls was correct or not).
Regards,
Bogdan
rajib deka wrote:
Hi Sebastian,
Ideally, that is correct - I agree that this is something to be done by
the UAC.
But the problem I see is that the UACs are not able to handle multiple
presence information (per contact) and because of this, the watcher may
get some really confusing view over the other party.
Hi Leon,
It should work, but I haven't tried. It is built using Java6. I was checking
in the Netbeans IDE and I can generate it using Java5 (never tested) if it
matters. In the development environment it actually runs using Jetty, Grails
1.05 and Netbeans 1.6.8. Let me know if you have any
Hi Leon,
It should work, but I haven't tried. It is built using Java6. I was checking
in the Netbeans IDE and I can generate it using Java5 (never tested) if it
matters. In the development environment it actually runs using Jetty, Grails
1.05 and Netbeans 1.6.8. Let me know if you have any
Hi Bogdan
Yes, in an ideal world, this re-publishing with BUSY and ONLINE is wanted,
but the toggling not. Priorities would help (you'd receive that but not trigger
anything on the user interface as the highest priority will keep its state),
but we don't have them yet, so in that case the
Bogdan,
Hey, I was working on this problem some more and this is what I found...
Serialize branches does in fact order the branches, but lowest to
highest.. if you just do a:
1. serialize_branches
2. t_relay
3. next_branches
4. t_relay
5. next_branches
6. t_relay
7. next_branches
8. t_relay
and
Hello,
During my first adventure into topology hiding it seemed the b2b modules
weren't bringing all my custom_headers from one side to the other. During my
testing I encountered this problem:
ERROR:b2b_entities:client_new: Buffer too small
ERROR:b2b_logic:create_top_hiding_entities: failed
Hi Jeff,
This error is generated when when constructing extra headers to be added
to the message that will be sent outside. Indeed the size might be too
little - 256. I will fix this.
Regards,
--
Anca Vamanu
www.voice-system.ro
Jeff Pyle wrote:
Hello,
During my first adventure into
Thanks I think ive got it right but ill have a check, I'm using the OSP
module so used the OSP sample .cnf file so Im not sure if this differes
anyway to the normal config file in the way it registers users?
Thanks
--
View this message in context:
On Apr 1, 2010, at 3:10 PM, Schumann Sebastian wrote:
Hi Bogdan
Yes, in an ideal world, this re-publishing with BUSY and ONLINE is
wanted, but the toggling not. Priorities would help (you'd receive
that but not trigger anything on the user interface as the highest
priority will keep
Hi Jeff,
Jeff Pyle wrote:
I found the BUF_LEN value in the b2b_entities/client.c and b2b_entities/dlg.c
value, and increased it from 256 to 512. This seems to have taken care of
the errors. I wonder what else I've broken by changing this.
That was indeed the right place for the
Hello,
it is my pleasure to announce that SEMS 1.2.0 has been released. This
release again brings many improvements to this fine, free SIP Media
and Application server.
Get the source at
http://ftp.iptel.org/pub/sems/sems-1.2.0.tar.gz
Some debian packages at
Hi all,
I am running Opensips 1.6.2 and am running a strange problem of Opensips not
responding to any SIP messages after a day or two. It needs to be restarted
for it to get working again.
The overview of my setup is as follows:
1) I am using Localcache module to authorize REGISTER and INVITE
Ok Bogdan,
I will try it
Regards
Franck
- Original Message -
From: Bogdan-Andrei Iancu bog...@voice-system.ro
To: OpenSIPS users mailling list users@lists.opensips.org
Sent: Thursday, April 01, 2010 7:21 AM
Subject: Re: [OpenSIPS-Users] installing opensips on Fedora 10 64bits
Hi
Hi,
just a quick note to tell you that mi_xmlrpc works fine on FC11 64b.
svn rev: 6752 ( branch 1.6 ).
stock xmlrpc rpms:
xmlrpc-c-c++-1.16.6-2.1582.fc11.x86_64
xmlrpc-c-1.16.6-2.1582.fc11.x86_64
xmlrpc-c-devel-1.16.6-2.1582.fc11.x86_64
xmlrpc-c-client-1.16.6-2.1582.fc11.x86_64
Greetings all;
I'm attempting to set up a fail-over only scenario using dispatcher and am
encountering some problems. I'm using dispatcher since we're already
utilising it for load balancing, so it makes sense to reuse the tool, and
according to the OpenSIPS 1.6 dispatcher module documentation it
Where is your failure route? :)
-Brett
On Thu, Apr 1, 2010 at 11:20 AM, Jock McKechnie
jock.mckech...@gmail.com wrote:
Greetings all;
I'm attempting to set up a fail-over only scenario using dispatcher and am
encountering some problems. I'm using dispatcher since we're already
utilising it
On Thu, Apr 1, 2010 at 10:26 AM, Brett Nemeroff br...@nemeroff.com wrote:
Where is your failure route? :)
-Brett
I intentionally chose to not include it, along with the other 200 lines of
config, for simplicity, but you're right, given this is a failure, I clearly
should've, duh :)
Hello All,
I'm working with a Broadsoft which is setup to send outbound calls to
OpenSIPs. The Broadsoft extension is set to ring simultaneously
multiple extensions. One which hits the proxy and the other is
internal on the broadsoft.
What I'm seeing is an outbound call from broadsoft to the
This goes way back. Bogdan addressed it last year sometime, citing a reason
why ACK processing is slower than other processing. And, since the two
messages hit different children of Opensips, the ACK hits the exit gate after
the reINVITE, even though the ACK arrived first. I've seen it with
WOW
Ok, well the real question is.. did the 100ms sleep fix your problem?
And what was the result of the ordering issue for you? Did you get a
400 from the provider like I'm seeing?
I'll give it a shot..
-Brett
On Thu, Apr 1, 2010 at 2:56 PM, Jeff Pyle jp...@fidelityvoice.com wrote:
This
We're moving our mediaproxies to 2.0 and have run into the following: in
the old setup we used the priority value in the mediaproxy SRV records
to prefer local (same datacenter) relays but to failover to a different
datacenter if all local relays were unavailable. We then used the SRV
weight
On Thu, Apr 1, 2010 at 12:26 AM, Anca Vamanu a...@opensips.org wrote:
[snip]
Ok I think I got this somewhat working.
I was missing a dialoginfo_set() in another INVITE path.
However, does anyone know how, if you add a new phone, to make the
presence initialize to idle?
The BLF light blinks
Also, I've found a case where the BLF light stays red, even when a
call is hung up.
This seems to happen in the intercom case, where the SIP URI is
sip:u...@ip;intercom=true.
It doesn't happen on every intercom call, but once it does happen, it
is impossible to clear without clearing the
First, the nuts and bolts.
In the loose_route section:
if (!is_method(ACK)) {
perl_exec(nonack_delay);
}
And, in the Perl file:
sub nonack_delay {
select(undef,undef,undef,0.060);
return 1;
}
I lied. Not 100ms,
Hi,
I use Opensips as an SMS gateway. Now I need to check all message's
username firstly. If the username is in my mysql database,thus it belong to
the white name-list,then I realy it normally.If not ,I will drop it
directlly.
So I think I should exec some some mysql commands to check the
Yeah, that's OK
2010/3/30 Andrew Pogrebennyk andrew.pogreben...@portaone.com
On 30.03.2010 15:49, CheeWii wrote:
Now I used Opensips as a sip sms gateway. I used save(location)
to store the register information. However, when my client is behind
NAT,opensips will relay MESSAGE to an
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