Hello,
the dispatcher module (kind of fast load balancing support) has now
support for failover. Along with selected destination, the rest of
addresses in the destination set are stored in AVP list. If the selected
destination fails to manage the request, then the next address can be
used from failure_route.
Some destinations can be dynamically marked as inactive (e.g., based on
reply code) so the are not going to be used until the admin enables them
again, via fifo command.
For more info, please see the readme file:
http://openser.org/docs/modules/1.1.x/dispatcher.html
Cheers,
Daniel
Hello,
I was just wondering if there is a way to get music on hold to work with OpenSER. Hold currently works however it would be nice to transfer the RTP stream to a media server like SEMS or even Asterisk, instead of giving the end user deathly silence. I have tried to make my own script but I think it might need an OpenSER module? Just wanted to know how/if other users are dealing with this.
Thank you.
Justin
Thanks Tim. Klaus responded today with an email that gave me enough information to finally manipulate an AVP, and print it with xlog(). Wooo!
-----Original Message-----
From: Tim Klein [mailto:tkpublic@timklein.fastmail.fm]
Sent: Fri 12/16/2005 7:55 PM
To: users(a)openser.org
Cc:
Subject: RE: [Users] Variables
>I found something somewhere that said something like [si]$avp(name)
>or whatever. What the heck does [si] mean???
Was there a colon after the "[si]"? If so, then I'll bet [si] was
just a regular expression meaning "either the letter 's' or the
letter 'i'". My limited understanding is that every AVP reference
starts with either "i:" or "s:", depending on whether you're
referencing it by its integer name or its string-format name. (The
latter is more human-friendly, but slower.)
Tim
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Tim,
I've read those docs many many times. They still don't make any sense. I can't even find an example of how to print an avp with xlog.
I found something somewhere that said something like [si]$avp(name) or whatever. What the heck does [si] mean???
Doug.
-----Original Message-----
From: Tim Klein [mailto:tkpublic@timklein.fastmail.fm]
Sent: Thu 12/15/2005 4:01 PM
To: users(a)openser.org
Cc:
Subject: RE: [Users] Variables
>Is there any useful information ANYWHERE about avp's? I mean, don't
>take this the wrong way, but I've never had so much trouble
>interpreting/finding documentation on a subject.
At http://www.openser.org there's a link to this set of tutorials:
http://www.voice-system.ro/docs/
The one about the AVPops module is excellent. I hope this set of
tutorials will grow over time!
Tim
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Users(a)openser.org
http://openser.org/cgi-bin/mailman/listinfo/users
I was attempting to muck with the Jabber module from ser-0.9.4 stable
release... but, while the module compiles without a hitch, installing it kills
the server.
I get pages and pages of:
Dec 16 15:12:02 death /usr/local/sbin/ser[3993]: get_connection(): Inherited
open database connections, this is not a good idea
Messages in the log with the occasional
Dec 16 15:12:02 death /usr/local/sbin/ser[4000]: db_init(): Could not create a
connection
I didn't make any config changes. The database exists and is accessible. All
I did was load the module and set the modparams.
Is the Jabber module no longer really usable? If not, are there alternatives?
I know the new presence stuff uses an Xcap server, but I was kind of hoping to
hold off on all that until it became a little less constantly tinkered with. :)
N.
Klaus,
That's it!
Doug.
-----Original Message-----
From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
Sent: Friday, December 16, 2005 12:25 AM
To: Douglas Garstang
Cc: users(a)openser.org
Subject: Re: [Users] Transaction already in Process
So how does it look like when Client A calls Client B? Is the signaling
Like in the figure?
Asterisk
/ \
/ \
Openser \
/ \
/ \
Client A Client B
regards
klaus
Douglas Garstang wrote:
> Klaus,
>
> All calls are going through OpenSER from the phones. However, they don't go BACK to OpenSER. Asterisk terminates the call. Putting all calls BACK through OpenSER would be a nightmare. It would make a lot of Asterisk features such as ACD Queues, MeetMe Conferencing etc very hard to manage. I'm not even sure if they would work.
>
> Doug.
>
> -----Original Message-----
> From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
> Sent: Thursday, December 15, 2005 8:23 AM
> To: Douglas Garstang
> Cc: users(a)openser.org
> Subject: Re: [Users] Transaction already in Process
>
>
> I can't explain it, but I do not like the idea replicating with failure
> routes ...
>
> IMO it would be better to modify t_replicate to allow to replicate to
> moultiple instances.
>
> Would i be possible to route all calls via openser? Then you do not need
> the location info in the Asterisks.
>
>
> regards
> klaus
>
> Douglas Garstang wrote:
>
>>Klaus,
>>
>>Asterisk can't use a database for location/contact information. It also has no native means to transfer registrations between itself and another Asterisk system. It can't even perform a user defined action when a phone registers... doesn't leave us with many options. Right now, using OpenSER to replicate (forward,t_relay,t_replicate whatever) seems to be working ok. Do you have any reason to believe it wouldn't scale up well?
>>
>>Doug.
>>
>>-----Original Message-----
>>From: Klaus Darilion [mailto:klaus.mailinglists@pernau.at]
>>Sent: Thursday, December 15, 2005 3:23 AM
>>To: Douglas Garstang
>>Cc: users(a)openser.org
>>Subject: Re: [Users] Transaction already in Process
>>
>>
>>Hi Douglas!
>>
>>I still think it is a bad idea to replicate REGISTER to Asterisk 4
>>times. Where does Asterisk store the location table? In a database? Why
>>not replicate the location data on DB level?
>>
>>regards
>>klaus
>>
>>Douglas Garstang wrote:
>>
>>
>>>Arrgh!
>>>
>>>Now I'm getting "t_newtran: transaction already in process" messages whenever I try to call t_relay() or t_replicate twice, even when the first attempt has FAILED. In the case of t_relay() I have new addresses in the destination set and I still get this error. Why? When you attempt to deliver, and it fails, how can to attempt again without getting those messages? What fundamental piece am I missing?
>>>
>>>How could I do this? Forget the fancy stuff. I don't care about checking for failure anymore.
>>>
>>>t_replicate("192.168.10.7","5060");
>>>t_replicate("192.168.10.8","5060");
>>>t_replicate("192.168.10.200","5060");
>>>
>>>Doug.
>>>
>>>_______________________________________________
>>>Users mailing list
>>>Users(a)openser.org
>>>http://openser.org/cgi-bin/mailman/listinfo/users
>>>
>>>
>>
>>
>>
>
>
Hello all,
Can anybody provide me working ser.cfg with rtpproxy + nathelper.
I have already installed rtpproxy. I have gone through Onsip.org download but config didnt work. Can any body provide me the config.
Regards,
joel
I I got it right know, you need the following:
avp_printf("s:$hdr(call-id)", "$hdr(call-id)-$Ts");
The AVP ist string based, and the name of the AVP is the Call-id. Guess
AVPOPs need to be extended for this.
What is the main purpose of this new AVP? Adding as header or storing in
the database, so that another proxy can access it?
avp_pushto("$Myheader","$hdr(call-id)");
But this would just duplicate the Callid header.
regards
klaus
Joachim Fabini wrote:
> Hi,
>
> Is there a way or workaround to generate AVPs with header
> values as AVP name? It seems to me that the current AVP
> concept defines the AVPs at OpenSER startup, so
> header-value-based names are not feasible.
>
> What I need is the following: Store/retrieve a (key,value)
> pair where the key is the current message's call-id and
> the value some combination of header values and pseudo-
> variables.
>
> e.g. use the ops
> avp_printf("$hdr(call-id)", "$hdr(call-id)-$Ts");
> avp_pushto("$Myheader","$hdr(call-id)");
>
> in order to append the following header to the SIP message:
> Myheader: 23459(a)10.0.0.1-4235627623
>
> Any idea how to do this in OpenSER?
>
> thanks in advance
> --Joachim
>
>
> _______________________________________________
> Users mailing list
> Users(a)openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users
>
>
Hi,
after some hours of fighting trying to use sipsak to register a user on a
server i lost and decided to test older version 0.9.2 and it is working.
The command:
sipsak -U -x 120 -a password -u test -p 12.34.56.78 -C sip:s@11.22.33.44:5060
-s sip:test@domain
Didn't worked in 0.9.5 but work perfectly on 0.9.2 .
In 0.9.5 nothing was exiting from the server and seems like the sipsak was
sending the request to itself.
What can be the problem??
Bye,
Marcello
Hi,
We are trying to call from SER to Vocal , but call is failing saying
403 forbidden.
We are using kphone as user agent . UA1 is registered with SER .
UA2 is registered with vocal. When we call to UA2 from UA1 , call is
failing saying 403 forbidden. But if we call UA1 from UA2 then it is
working properly
Has anybody faced this type of problem ?
Does vocal requires some config changes ?
Any pointers on this will be helpful
Regards,
Rajesh
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