Thanks, how could I have missed that!
Is what it does it loads usrloc table piece-wise?
And if so, why does it do only for mysql?
And also, it is saying that old version would have
increased memory consumption with TCP/TLS -- does it
mean that it does not increase with the new version?
-jiri
At 14:32 30/11/2006, Ovidiu Sas wrote:
>Maybe you are looking for this one:
>http://www.openser.org/index.php?option=com_content&task=view&id=48&Itemid=9
>
>
>Regards,
>Ovidiu Sas
>
>On 11/30/06, Jiri Kuthan <jiri(a)iptel.org> wrote:
>>At 12:53 22/11/2006, Weiter Leiter wrote:
>>>I know that OpenSER loads (only?) faster.
>>
>>Can folks share with me what the fast-usrloc-loading feature is about?
>>I was not successful finding it out.
>>
>>Thanks!
>>
>>-jiri
>>
>>
>>--
>>Jiri Kuthan http://iptel.org/~jiri/
>>
>>
>>_______________________________________________
>>Users mailing list
>>Users(a)openser.org
>>http://openser.org/cgi-bin/mailman/listinfo/users
>
>--
>Jiri Kuthan http://iptel.org/~jiri/
Hi!
I can nowhere found the setting for the fifo use inside openserctl. The
default is /tmp/openser_fifo - where can I change this?
thanks
klaus
--
Klaus Darilion
nic.at
Hello everyone,
I found many references in serdev, openser users, etc. MLs to the issue
at hand, but only came up with diverging opinions and no solution
whatsoever.
The issue appears in case of a UA sending a re-INVITE inside of an
established dialog. Depending on your configuration file, version,
network conditions, there is NO guarantee that this re-INVITE will be
sent to (I'm not even talking about reaching) the next hop (typically a
PSTN gateway) AFTER the ACK to the first INVITE of the dialog.
I perfectly understand why (openser being transaction and not dialog
statefull being the only reason that really matters, given the fact that
INVITE and ACK are separate transactions, often handled by separate
openser processes) it is this way.
However, it seems that several gateways are not handling this well at
all, and just drop the transaction typically sending a "500 Server
internal error". I've read reports about cisco AS5300, Asterisk, and I
am myself experiencing this with an Andiocodes Mediant 2000 gateway.
In my case, the issue is there 99% of the time. I know I could profile a
bit my config file to ensure ACKs are processed faster than INVITEs
(which propably is already the case), but this is hardly a workaround. I
got tempted to even call an external "timer" via exec_msg when detecting
a re-INVITE (if it is an INVITE, has a to_tag and is loose routed), but
come on, I'm sure we can do better than this !
So no matter what the RFCs say or mean to say, no matter how long we
argue about this, the facts are :
- gateways often don't stand this ordering issue
- many iPBXs are using reINVITEs for several call features
- the only way to solve this is to be dialog statefull, at least for
ACKs and INVITES
So my conclusion is :
I'm going to code an ugly little hack - using an external database and
avpops - so that ACKs are logged per Cseq+callid, and when reinvites are
detected, relay will be delayed until the last ACK is logged as sent.
What do you think ?
Is there a way to use the dialog module to optimize this (flagging the
SIP headers themselves instead of an external DB) ?
Best Regards,
Jerome Martin
Hi,
thanks to Lavinia Andrei, we added today a new transport implementation
for the Management Interface (MI). This is XMRPC and it the second after
the initial FIFO transport, but not the last :).
The module is build by using the xmlrpc library, following all XMLRPC
standards. It is in alpha stage, so more testing is needed.
NOTE that only commands ported to the new MI interface (and not all
former FIFO cmds) are available via XMLRPC, but this is a temporary
problem - there are only 2 module left to update (TM and USRLOC).
For more info, please see the online documentation:
http://www.openser.org/docs/modules/1.2.x/mi_xmlrpc.html
or ask on the mailing list.
Any feedback - bugs or improvements - are welcomed.
Regards,
Bogdan
Hi all!
Does someone know where I can find documentation about the database
structure used with OpenSER. I mean the signification of the tables and
their fields.
Thanks in advance
Greg
I am trying to forward requests so that the request is
completely gone and the UA is forwarded to the final
destination with no signaling from the forwarding
system.
For example: "5551212@" would be forwarded to 1.1.1.2
but 1.1.1.2 would not get a redirect request from
1.1.1.1 rather, directly from the UA.
Any thoughts?
Thanks! FR
____________________________________________________________________________________
Access over 1 million songs - Yahoo! Music Unlimited
(http://music.yahoo.com/unlimited)
Hi,
I'm running an OpenSER server and I've got a couple of usage questions I'd
like some pointers on how I can implement them.
- I want to link multiple incoming telephone numbers to one single account
so 123456 and 45678 and 999933 can all point to the account customer1
Basically I want to do SIP trunking, but with requiring the CPE to
authenticate itself. How do I do this?
- I'm now using an Asterisk server to do some incoming call routing (mainly
some call forwarding scenarios) and I'd like to remove the Asterisk
dependency. How do I retrieve values based on the called destination from a
database? I've tried looking at AVPops, but haven't yet found any clear
documentation on how to use it.
Thanks!
--
Andreas Sikkema
Hi All,
We are currently using radius server for authentication. Is it possible to use oracle database with SER without using radius server. We have no problem if need to develop our own module for it, but the concern is whether we still be able to do digest authentication if we take out the radius server.
Best Regards,
Abdul Qadir
---------------------------------
Want to start your own business? Learn how on Yahoo! Small Business.
Hi,
Is it true that MSILO module only can handle text message?
So what is the usage of voice_silo table that written on message_store.php on serweb then?
Thanx
Regards,
Meidiana
---------------------------------
Access over 1 million songs - Yahoo! Music Unlimited.
Not sure why that's happening. Probably setting canreinvite=no on the
asterisk side will eliminate the re-INVITEs as a temporary solution, but
still would like to know what is happening...
wrote:
> Sometimes, a calls b and b hears a, and a hears b for a second but a
second
> INVITE comes to phone B that causes it to redirect rtp to be point to
point.
> Sometimes there is no audio.
> Sometimes, everything works fine.
> At one point, rtp from a was going to asterisk, but asterisk was not
sending
> the rtp on to b, and b was trying to send traffic point to point.
