Hi all
I have downloaded from openser
and iam trying to integrate voice mail with asterisk
I have read all the docs in the document site
after config, and people recomendation iam able to run the openser
successfully
and able to fix the problem calling out side
but when the local user not available, iam sending to asterisk voice mail
and i get error
SIP/2.0 484 Address Incomplete in my x-lite client
any suggestions
ram
Hi,
Check library (.so files) paths in ser.cfg file and if not correct, give the correct path for each library. Then run ser again, it will definitely work.
Regards,
Benjamin.
-----Original Message-----
From: Alexandr Dubovikov [mailto:shurik@start4.info]
Sent: Thursday, June 08, 2006 5:14 PM
To: Shyamsundar, Purkayastha (Purkayastha)** CTR **
Cc: 'serusers(a)lists.iptel.org'
Subject: Re: [Serusers] novice issue
On Thu, Jun 08, 2006 at 04:48:24PM +0530, Shyamsundar, Purkayastha (Purkayastha)** CTR ** wrote:
> Hi all
>
> Just started using ser but i am stuck at the first step itself
>
> i downloaded ser-0.9.6_linux_i386.tar
>
> untared it , set the cfg file and modules in correct path as per cfg file
> and ran ./ser from the sbin directory
>
> and it says
>
> ERROR in cfg file ( 24 errors)
at first, check you configuration with "ser -c"
and set the debug level to 4 in your ser.cfg , after it you can see where is a
mistake.
Wbr,
--
Alexandr Dubovikov * baron@iRC RusNet * mailto:shurik@start4.info
AD1-UANIC * ICQ: 122351182 * http://www.start4.info
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by Shyamsundar, Purkayastha (Purkayastha)** CTR **
Hi all
Just started using ser but i am stuck at the first step itself
i downloaded ser-0.9.6_linux_i386.tar
untared it , set the cfg file and modules in correct path as per cfg file
and ran ./ser from the sbin directory
and it says
ERROR in cfg file ( 24 errors)
if i comment out the loadmodule statements on the cfg files then the errors
reduce but inspite of verifying the correct path several times its not
getting through.
That means its reading the cfg file correctly but not loading the modules or
not able to interpret loadmodule command maybe
Please help me
thanks in advance
Shyam
hi
I have been using OpenSER from CVS for some months now, as a
SIP proxy for a few domains. the OpenSER server has been very stable ;)
my goal is to have an experimental service using latest version of
all components.
we have a small PHP script to display the SER status, reading from
the MySQL database. you can see how it works here:
http://ser.symbianos.org/
(if you want to download the PHP script, go to the bottom of the page)
Many thanks for supporting the OpenSER server, I find
both the code and the docs to be excellent ;)
/alfred
--
Alfred E. Heggestad <aeh(a)db.org>
Web: http://aeh.db.org/
VoIP: <sip:alfredh@symbianos.org>
Phone: +47 21 98 71 20
Mobile: +47 98 23 67 05
Skype: alfredheggestad
I thought I was experiencing the same issue. But after looking at it
closely, I discovered the ACK is not being sent. I am not an expert with
OpenSER and may not be using the correct terminology, but I will try to
explain what I see happening.
I am using Openser as a gateway to our provider. We send invites from
asterisk to openser and we forward the invite to the provider. We are using
rewritehost followed by t_relay to route the requests. When we receive the
200 ok back from the provider we relay the message back to asterisk.
Asterisk responds with an ACK followed by the reinvites. The issue we are
seeing is the ack is not relayed at all, only the reinvites get relayed.
We then added the following to the top of openser.cfg:
# ---------------------------------------------
# Asterisk reinvite issue relay ACK immediately
# ---------------------------------------------
if (method=="ACK") {
forward(uri:host, uri:port);
route(1);
};
Now the ACK is being handled correctly. We are clearly missing something.
I suspect the same thing is happening on inbound calls with the relaying of
200 ok messages.
Gene
------------------------------
Message: 6
Date: Wed, 10 May 2006 10:54:52 +0200
From: Klaus Darilion <klaus.mailinglists(a)pernau.at>
Subject: Re: [Users] ACK is sent before the re-INVITE to OpenSER, but
the proxy relays the re-INVITE first (call flow in the body).
To: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>
Cc: users(a)openser.org
Message-ID: <4461AA5C.7090403(a)pernau.at>
Content-Type: text/plain; charset=ISO-8859-1; format=flowed
IMO the receiving client should be tolerant and accept the reINVITe
although the ACK is missing/delayed, as the reINVITE is an implict ACK.
regards
klaus
Bogdan-Andrei Iancu wrote:
> Hi Alexandre,
>
> if I'm not wrong ,this topic was previously discussed on the mailing
> list. The idea is that openser cannot guarantee that messages will be
> sent in same order as received. And this because of parallel
> multi-process execution.
>
> the idea is that the CC1 shouldn;t send a re-INVITE so fast and also the
> GW should wait a little bit to see if ACK is delayed or not.
>
> if you are running in full debug, the logs will might help you to see
> how the ACK and INVITE gets swapped during processing.
>
> regards,
> bogdan
>
<<snip>>
Dear all,
My name is Ferianto. I am a college student. I am still new in voip system. A few months ago, i read from the internet about how to communicate using voip,
and the usage of TLS to secure the communication in voip. I am interested in it and i have taken it as the topic of my paper in my college. So, i try to build it
in my college. For that reason, i have read many tutorials about TLS system, especially from www.openser.org, because I plan to build the TLS
system by using openser packet. So, i try to understand tls.htm file from www.openser.org. But, when i try to Configure it, i get a problem. I am confused what the first thing to do for configuration. I have asked it to my lecturer, but no body can help me in my college because TLS is new in my college.I am ALONE. Can any
one guide me to build it and so i can finish my paper?
Please help me.....Please.....
Thanks
Ferianto
---------------------------------
New Yahoo! Messenger with Voice. Call regular phones from your PC and save big.
Hi!
as you said, i have nt initially included record_route. that should
be the reason y the server is processing only transaction and not the entire
call. and sorry, it is only till the 200 Ok i see the responses going thru
the server, after which the uas learn each other's address and directly talk
to each other. i have traced the uas and i see the bye. but now i have
included record_route function in both the INVITE and the BYE code block.
still the bye is not going thru the proxy. also when the invite is routed
from the proxy to the called UA, i dont see the record_route header in the
invite.
is any other module needed to be loaded other than mediaproxy for the bye to
work?
Please let me know.....
Thanks a lot for your help,
Padmaja
----- Original Message -----
From: <fredler(a)ycn.com>
To: "Padmaja RV" <padmaja.rv(a)vodcalabs.com>
Cc: <users(a)openser.org>
Sent: Tuesday, June 06, 2006 5:35 PM
Subject: Re: [Users] cant see Bye when the call is diconnected
>> i have openser running and supports authentication and calling,. i made
>> calls and verified that the media session is established both ways. now
>> if i
>> disconnect the call, i cant see the BYE request sent. in the ethereal i
>> dont
>> see any call related messages after ACK is sent for the INVITE. please
>> help
>> me.....
>
> I think you did not set "record-route" in your cfg.
> But than you also should not see ACK.
>
> Are you able to trace on you User Agent?
> Than you would see where BYE is sent.
>
> regards
> Franz
>
Hello, Everyone:
I am newbie to openser. I would like to build a softwitch system by
openser. It is work when client behind NAT, But it don't work when the
openser server behind NAT where I add a DNAT rule in the router.
Who can help me to address the issue ?
FYI:
CLEINT --------------------->Router
++++++++++++++++++++++++++++++ Openser
202.99.1.23 202.99.3.5 192.168.0.1
192.168.0.2
I use sipphone software to connect 202.99.3.5(5060)
It don't work ...
Jerry.
sir
when I fill up Admin or User login form
It prompts a message :
DB Error: Connect Failed
Bad username or password.
PLZ tell me how to remove this error
waiting for your reply.
regards.
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