Hi,
Maybe you have forgotten to create the tables for opensips?
opensipsdbctl create .(creates a new
database)
opensips is missing the version table
BR
Uwe
anurag schrieb:
>
>
> Hi,
>
>
>
> I’m new to OpenSIPs. I want to use OpenSIPs as Presence server.
>
>
>
> I’ve compiled Op
This I am unclear about at this time. Everything will be SIP/IAX based, no
PSTN lines will be used. We may have some PRI's here and there at some point
but not initially.
>>> Asterisk you can use as VOIP / PSTN termniation
> 2 Opensips ( for loadbalance registration) replicate
> 1 Mysql Server
> 3
>
>
> > I have got my SIP proxy up and running, but I am able to work only with
> > the username 'opensips' and the password 'opensipsrw'. When I change the
> > username and password in opensipsctlrc file I get an error message
> > saying access denied. I am not asure about which variables to comme
Hi,
I'm new to OpenSIPs. I want to use OpenSIPs as Presence server.
I've compiled OpenSIPs (1.5.1) for presence server and installed it
on Linux. Also, I've enabled presence parameters in config file with
mysql database.
My DB details are: User-> opensips
Hey All,I've got a stateful dispatcher I've build using dialog profiles.
Apparently I'm doing something stupid in my script generating these errors:
Jul 3 00:54:20 opensips-a /usr/local/sbin/opensips[26189]:
ERROR:dialog:set_dlg_profile: dialog was not yet created - script error
Jul 3 00:54:20 op
Hello,
You may try to use xml-rpc instead of opensips-mi-proxy?
I don't use opensips-mi-proxy, so, can't give much help.
Searching "opensips/socket" in google may give you some hints on the
problem.
Thanks & Regards
Cao, Charles
-Original Message-
From: users-boun...@lists.opensips.org
Hi, draft-ietf-sip-outbound defined a outbound mechanism, reference to, http://tools.ietf.org/html/draft-ietf-sip-outbound-20 .Does OpenSip support this feature?Thanks,Sean Fengcienet___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.or
Julien,
If you use media proxy you actually let the end-points negotiate the
codec, MediaProxy is transparent and does not get involved at code
level. A transcoder would be just one of the endpoints. So you can use
any transcoder you want (like an SBC with transcoding capabilities),
if an
ny one box?
>
> Thank you very much for this information as it will help to first
> understand what the project can do.
>
Hi
thats very big project., for that you need to study the system indepth both
Opensips and Asterisk
at this moment iam working on the same project but my concep
Hi,
When using Opensips with Mediaproxy, what transcoding solution is mostly used
by Opensips users, anyone as recommendations / suggestions ?
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
Hi Dimitrios,
On 02 Jul 2009, at 13:17, Dimitrios Giannakopoulos wrote:
> Hi,
>
> So, according to our scenario the asterisk has private ip. Any traffic
> from/to the asterisk can be routed to/from our network.
> The Network trace (between asterisk and opensips) shows that
> mediaproxy does not fo
Hello, I had 6.7.3 until this morning in our testing machine, I just
upgraded cdrtool to the 6.8.1 version, and after cleaning all the
database and recreating the tables, there's on remaining thing left.
Every time i run the normalize script (normalize.php) i get this:
Warning: Missing argument 8
As I would be using opensips for registration/proxy/lb, and not RTP, assuming
that connections are handed off to the asterisk boxes, would it be possible to
run it as an esx guest?
I ask this because it would simplify remote management of the server/s, nothing
more.
Thanks.
_
Thanks for the input folks!
> thats very big project., for that you need to study the system indepth both
> Opensips and Asterisk
It is more of a plan to be able to handle growth while not having so many of
the usual growth pains that ultimately mean losing customers. We want to build
a scalabl
I've come across this project a few times but have been having a bit of a
> time confirming just what the project does. I thought perhaps the best way
> would be to join the list and ask.
>
> My task is to put together a scalable asterisk based pbx system. Because
> the boxes will initially have mo
El Jueves, 2 de Julio de 2009, Bogdan-Andrei Iancu escribió:
> > Pure B2BUA may be a really nice module.
> >
>
> There is already finished a full signalling b2bua module. It will be
> uploaded soon on SVN.
Hi Bogdan, I really wonder how will behave this b2bua module when having
parallel forking
Hello,
i want to upload my presence-rules.xml to authorize or not authorize watchers.
when i upload the document to openxcap i get these error in my syslog:
can anybody help me to resolve the problems?
i think a big problem is that the server can not find the socket from opensips
in /var/run/ope
Hi,
Is your xlite taking the challenge it get from opensips to REGISTER again?
The (blah) is just an cut&paste error?
BR
Uwe
Gordon Ross schrieb:
> On 02/07/2009 09:49, "Uwe Kastens" wrote:
>> Strange so far. I cannot see any wrong configuration on a 1st view.
>> Could you see if auth is work
On 2 Jul 2009, at 12:05, Stuart Marsden wrote:
> Hi,
>
> It can be made to work - trust me
I'm sure it can if enough effort is put into it. The problem you face
is that Linksys made many false assumptions when they devised their
MOH scheme and now you have to work around all of them. First o
Hi Victor,
Victor Gamov wrote:
> On 01.07.2009 17:07, Bogdan-Andrei Iancu wrote:
>
>> Hi Iñaki,
>>
>> not exactly - for hiding the network information is enough to hide some
>> headers (like RR, VIA, Contact).
>>
>
> Call-Id have IP info too.
>
it may - there is no standard way of gener
On 02/07/2009 09:49, "Uwe Kastens" wrote:
> Strange so far. I cannot see any wrong configuration on a 1st view.
> Could you see if auth is working and only writing to the USRLOC is
> failing? (Maybe put some xlog statements around the register part).
Small progress. I discovered another missed un
Hi,
It can be made to work - trust me
Phone A puts Phone B on hold
The new Session A-> MoH Invite contains the relay port of the existing
A->B call (functionality fixed by the phone, it thinks it is passing
phone B)
the MoH server will now start to send audio to the mediaproxy session
for t
Gordon,
Strange so far. I cannot see any wrong configuration on a 1st view.
Could you see if auth is working and only writing to the USRLOC is
failing? (Maybe put some xlog statements around the register part).
The error ocurs by saving the contact into the DB. Have you tried with
another client?
Hi,
On 30 Jun 2009, at 19:42, Stuart Marsden wrote:
>
> The problem is the Sipura/Linksys/Cisco Phones do it the other way
> round - the phone going on hold (A) sends a normal sendonly to the
> caller(B) . Then (A) sends a brand new invite to the MoH server(C) ,
> passing what it thinks i
On 02/07/2009 09:13, "Uwe Kastens" wrote:
> Which version of opensips you are testing with?
1.5.1
> Have you enabled multi
> domain support for register, urloc etc.pp.?
Yes. However, in the process of posting the config (below) I noticed that I
hadn't un-commented the line:
modparam("alias_db|
Hi,
On 02 Jul 2009, at 08:58, Dimitrios Giannakopoulos wrote:
> Hi all,
> Thanks for the help.
>
> I have set nat=no but problem persists.
So is your Asterisk on a public IP? Could you at least confirm with a
network trace that mediaproxy is forwarding RTP packets from the
gatways (from bot
Depends is a select would be faster than an LDAP bind.
Probably OpenLDAP would be faster and you have much more to gain by
having it in centrally in OpenLDAP (replication, standards based
access etc.)
Gavin.
On 01/07/2009, Bogdan-Andrei Iancu wrote:
> Hi Alan,
>
> Got your point! Theoretically,
Gordon,
Which version of opensips you are testing with? Have you enabled multi
domain support for register, urloc etc.pp.? Maybe you can post the head
of your config.
BR
Uwe
Gordon Ross schrieb:
> Starting with an empty DB, I created a domain and I created a subscriber in
> OpenSIPS.
>
> # o
Starting with an empty DB, I created a domain and I created a subscriber in
OpenSIPS.
# opensipsctl domain add blah
# opensipsctl add 2...@blah 1234
Looking at the database, the user & domain are in the tables.
Firing up X-Lite, I put the following in as the SIP account details:
Display Name: G
29 matches
Mail list logo
