Hello,
There is a new release of OpenXCAP server available. It contains many
bug fixes and additions for important features like XML partial
manipulation, xcap-caps, xcap-diff and RLS services that were not
available at the initial release last year. With this features
OpenXCAP fulfills now most if not all relevant requirements for an
XCAP server implementation. A generic xcap client library and command
line client is available in the clients directory, you can use it to
manipulate documents on OpenXCAP or other XCAP server implementation.
The server software is also operational at http://sip2sip.info so you
can test against it by registering a SIP account.
Changes in version 1.0.0
* Added RLS services (RFC 4662 and RFC 4826)
* Added support for xcap-diff based on draft-ietf-simple-xcap-diff-09
* Added partial get/put/delete of elements in the XCAP documents
* Added test suite for rls-services, resource-lists and partial
updates
* Many bug fixes from field experiences
* Development status from beta to production
* Switched to Python 2.5
* Improved documentation and testing suite
* Changed license to GPL
* Fixed MySQL operational error (2006)
The software can be downloaded from:
http://openxcap.org
Many thanks to Denis Bilenko who did most of the works for this new
release.
Kind regards,
Adrian Georgescu
Hello
Thanks for the feedback from everyone.
You can try cdrtool version 6.6.0, I have added parallel prepaid call
support. The algorithm is described in PREPAID.txt
cdrtool (6.6.0) unstable; urgency=low
* Allow simultaneous calls for prepaid accounts, see the Simultaneous
prepaid calls section in doc/PREPAID.txt
* Added pseudo code examples for call control engine
* Apply the changes from setup/mysql/alter_tables.mysql
* Added interface to kill ongoing sessions in Prepaid page
Testing and feedback is useful and welcome.
Regards,
Adrian
hey all
after making call forward feature using avpops module and its exported
functions avp_db_load and avp_pushto
i have a problem :
assuming all calls to extension 12001 will be forwarded to extension 12008,
in opensers's database i have added the following :
username=12001
domain =mydomain
attribute = callfwd
value = sip:12008@mydomain
type = 0
untill that every thing was fine and all calls comes to 12001 forward to
12008 without any problem , the problem comes from when i am using 12008 to
call 12001 it calls itself ( mean 12008)
so anyone can tell me what i have missed ?
i have configured my openser.cfg like this
in loaded modules section added
loadmodules "avpops.so"
>
in module parameter section added :
modparam("avpops", "avp_url", "mysql://openser:openserrw@localhost/openser")
modparam("avpops", "avp_table", "usr_preferences")
in Invite message handler added :
if(avp_db_load("$ru/username","$avp(s:callfwd)")) {
avp_pushto("$ru", "$avp(s:callfwd)");
route(1);
exit;
}
any suggestions ?
Greetings,
I have 2 proxies having the same database on backend. One is the primary,
second is standby - OpenSER 1.2.1.
That's how I implement DNS based failover. What I see, when all UA's
registered on primary and I start standby, I's starting to cleanup location
table according to it's own timer. So record is gone, when primary trying to
updated the record on re-register it of course fails, as a result location
table is wrong.
What will be a solution?
Thanks,
Toly
--
View this message in context: http://www.nabble.com/two-proxies-sharing-the-same-database-tp19335185p1933…
Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.
Hi, AFAIK there is an issue in typical SIP/PSTN enviroments in the case of
failover using different gateways serially.
The main issue is that SIP doesn't define how a UAC should handle the case in
which it receives various "183 Session Progress" from different
early-dialogs, each of them containing early-media.
It occurs to me that a SIP/PSTN gateway sometimes replies 183 with early media
for 1 or 2 seconds, and inmediatelly it replies a 502. Well, this 502 could
be caused by other node in the PSTN network, it's not necessarialy a problem
fo the gateway [1].
In this case I do serial forking forwarding the request to other gateway.
But the UAC already received early media from the first gateway, and most of
the SIP phones just render the first early media they receive in case of
multiple 183, and drop others. So basically the problem is that the UAC user
doesn't listen the ringing coming from PSTN after the request has been
forwarded to the second gateway.
Is there something I could do at proxy level to avoid this issue? Sincerely I
don't think so, since it makes no sense to disallow the first 183 (I cannot
know that the gateway will later reply a 502).
Thanks.
[1] Anyway, *who* is creating the early media? Why the PSTN generates it if
the request doesn't arrive to the called? "imagine there is no PSTN..."
--
Iñaki Baz Castillo
Greetings,
Suppose I need to do custom routing for outbound calls from UA - softphone.
Each user account may have the following info attached to it:
1:provider_a:2:provider_b:3:provider_c
which means:
forward call to provider_a,
if failed forward call to provider_b
if failed forward call to provider_c
if failed send response back to UA
I do database lookup based on from, and get above string.
Suppose I wrote my own dispatcher module which initially gets set to above
string:
1:provider_a:2:provider_b:3:provider_c
and forwards to first provider, then on failure_route I call next_provider
and so on.
Will it work across multiple messages? Will it work correctly for each
child - I mean joe@domain and
paul@domain should be handled according their own providers?
Any other suggestions?
Thanks,
Toly
--
View this message in context: http://www.nabble.com/custom-routing-tp19322478p19322478.html
Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.
I am using Kamailio 1.2 , in t openser.cfg I defined the the url for db,
which is not local
("auth_db|permissions|uri_db|usr_loc", "db_url",
"mysql://openser:db_password@db_ip_address/openser")
but still getting the error message below when start openser;
/sbin/openser[13064]: permissions - initializing
/sbin/openser[13064]: WARNING: File not found:
//etc/openser/permissions.allow
/sbin/openser[13064]: Default allow file (//etc/openser/permissions.allow)
not found => empty rule set
/sbin/openser[13064]: WARNING: File not found:
//etc/openser/permissions.deny
/sbin/openser[13064]: Default deny file (//etc/openser/permissions.deny) not
found => empty rule set
/sbin/openser[13064]: new_connection: Can't connect to local MySQL server
through socket '/var/run/mysqld/mysqld.sock' (2)
/sbin/openser[13064]: ERROR: group_db_bind: unable to connect to the
database
/sbin/openser[13064]: ERROR:group:mod_init: unable to open database
connection
/sbin/openser[13064]: init_mod(): Error while initializing module group
/sbin/openser[13064]: INFO:mi_fifo:mi_destroy: process hasn't been created
-> nothing to kill
any idea ?
thanks in advance
Asim
Hello,
I have some problem with, openser & centile!
When do a call from centile to openser, the call don't do "ringing", just
"trying" and after, centile say "BYE"
Have you an idea?
Thank you
I could have a look, but
Asterisk-Openseràno problems
Centile-openseràdont work !
Cordialement
BERGANZ François
<http://www.acropolistelecom.net> http://www.acropolistelecom.net
TEL/FAX : 33 (0) 1 70 72 50 15
De : Darren Sessions [mailto:dmsessions@gmail.com]
Envoyé : vendredi 5 septembre 2008 15:26
À : BERGANZ François
Objet : Re: [Kamailio-Users] Openser & Centile
Would be helpful to see some of the SIP signaling.
_____________________________
Darren Sessions
dmsessions(a)gmail.com
http://www.darrensessions.com
_____________________________
On Sep 5, 2008, at 7:24 AM, BERGANZ François wrote:
Hello,
I have some problem with, openser & centile!
When do a call from centile to openser, the call don't do "ringing", just
"trying" and after, centile say "BYE"
Have you an idea?
Thank you
_______________________________________________
Users mailing list
Users(a)lists.kamailio.org
http://lists.kamailio.org/cgi-bin/mailman/listinfo/users
The problem with concurrent prepaid calls and single balance is that
you have to correlate between the call control and rating angine
somehow so that all calls terminate when balnce becomes zero. The
problem is a bit complex:
Example:
Balance = 10.
A call starts to destination XXX, for the sake of example max session
time = 2 minutes
After one minute, you start second call to destination YYY which has a
different price and your balance is not anymore 10 but depends on the
duration of the first call which is in progress.
What is the maximum session time for it given that the first call is
already in progress?
What should happen with the first call?
I am looking for suggestions on implementing a proper algorithm to
deal with this situation in the rating engine. If you have any I would
be glad to hear it.
Adrian