Hi Eneref,
Thanks for your reply. So, do u mean that I'm gonna have the follwing
applications in the SAME box?
- 2 SER servers ( 1 existing SER + 1 new SER for SEMS ) with different port
- 1 SEMS
Thanks.
Roa Yu
-----Original Message-----
From: Eneref [mailto:eneref@arcdiv.com]
Sent: Friday, November 09, 2007 4:14 PM
To: roayu
Subject: Re: [Serusers] Voicemail setup on SER v0.9.6
Yes. You need either SEMS or Asterisk or some such. Meshing SEMS with
SER, though it LOOKS complex, is really not that bad.
The easiest thing to do is to run another SER server to handle voicemail
(on a different port from the first SER if it's on the same box) and
forward messages to it if voicemail is required (on time outs, or
whatever conditions you set). There are some good instructions on how
to do that in the SEMS docs.
This also makes it quite easy to swap out with Asterisk or any other
sort of voicemail-capable server should the need arise in future.
N.
roayu wrote:
>
> Hi there!
>
>
>
> Could anyone let me know how to setup Voicemail on SER v.0.9.6 ? Do I
> need to install SEMS as well?
>
>
>
> Thanks
>
>
>
>
>
> Regards,
>
> Roa Yu
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Serusers mailing list
> Serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
Hi Guys
I really think there is something wrong with MediaProxy, I can not get
the radius or the mysql accounting to report any details, as best as I
can see there is never any attempt to even try.
There is nothing in the logs about there being anything wrong at all.
I had added print statement thought out the code and I can see that the
code for both mysql update and the code for sending radius updates never
gets call.
I have not been able to find the connection between the radius or mysql
thread and the rtphandler but the print statement a in the actual calls
to the mysql or radius and the print messages are never seen in the logs.
Other prints added by me do get logged.
Has any one actual tried the current code and got do the stop updates ?
Mike
Hi Folks,
I'm struggling with bridging openser between a public and a
private network.
The private 10.10.12.0/24 network connects openser and an asterisk
server that plays a file and hangs up. I've got 2 sip clients connected
to the public and private sides of the openser server.
When I make calls via openser to the asterisk box, I only get audio on
the private networks sip client, and neither client gets hung up
correctly at the end of the call.
I've put the SIP debug from the external client here:
http://pastebin.com/m388a707d
My openser and rtpproxy configs are here:
http://pastebin.com/m3ed309d3
Can anyone give me any pointers? I've been staring at this for days :(
-Cheers Max
I'm using 1.2.X and the LCR module... following syntax fails at start up...
$avp(i:20) = 1;
if(!load_gw( $avp(i:20) ) ) {
with an invalid arg to load_gw...
anyone have a suggestion?
Stanisław Pitucha wrote:
> ----- "Anca Vamanu" <anca(a)voice-system.ro> wrote:
>
>> I am not sure if snom 190 supports server presence. I've searched a
>> bit
>> but did not find sufficient info.
>> Does the phone send other Subscribe messages apart from the ones with
>>
>> event: dialog? Does it send any Publish message?
>>
>
> It has "Publish Presence" option turned on. It has also line monitoring with lights. I've found this site with example notify packets:
> http://www.abptech.com/support/faqs/faq_snomphone_LED_explained.html
> I have both support broken registrar on and filtering off, whatever that means for snom phone. That means it supports something.
> Still - it subscribes for dialog properly (as far as I can tell from rfc4235), so I don't understand the bad event response... Or maybe openser just doesn't support 'dialog' events?
>
Yes, as I have mentioned before, OpenSer does not support 'dialog'
event. It has support only for dialog with sla parameter (Event:
dialog;sla) , used for BLA implementation (draft
draft-anil-sipping-bla-03.txt).
Please send a ngrep capture with a Subscribe message with Event:
presence , its reply and the Notify that follows.
Anca
Hi,
When I invoke proxy_authorize and radius_proxy_authorize for an ACK
message the is no uid avp present.
All other messages which are authorized work fine.
Why is that so??
Is it the right behaviour??
Bests regards
Tomasz
At 17:49 08/11/2007, Tomasz Zieleniewski wrote:
>Hi Piotr:)
>
>RFC only implies that proxy should not challenge the ACK message and
>moreover it says that client sending ACK should duplicate
>Authorization and Proxy-Authorization headers. This it what my client
>does. But according to the description of the proxy_authorize function
>it only verifies that credentials are valid. This function doesn't
>cause the challenge response to be sent to the client. This is
>performed with the usage of proxy_challenge function.
>"The function verifies credentials according to RFC2617. If the
>credentials are verified successfully then the function will succeed
>and mark the credentials as authorized (marked credentials can be
>later used by some other functions). If the function was unable to
>verify the credentials for some reason then it will fail and the
>script should call proxy_challenge which will challenge the user
>again."
>In my opinion, that is why proxy_authorize function should generate
>the uid avp for ACK request if verification of credential gives
>positive result.
Well -- it should be stored somewhere in transaction state and someway
I guess you should be able to get access to it (not sure if with or
without some hack) ... but do you relaly need that? ACK has a transport
function and as such, I don't see the use value for doing more processing
with it than abosrbing/forwarding...
-jiri
--
Jiri Kuthan http://iptel.org/~jiri/
At 15:53 08/11/2007, Tomasz Zieleniewski wrote:
>Hi,
>
>When I invoke proxy_authorize and radius_proxy_authorize for an ACK
>message the is no uid avp present.
>All other messages which are authorized work fine.
>Why is that so??
>Is it the right behaviour??
yes.
ACk has only a transport function and does not really impact SIP processing.
-jiri
>Bests regards
>Tomasz
>_______________________________________________
>Serusers mailing list
>Serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
Hi,
I have the following problem after I insert a partical hf value woth
the usage of the textops module (insert_attr_hf)
the is_present_hf function returns negative result.
is this correct behaviour??
Regards
Tomasz