Hi there!
I would like to implement the following scenario using OpenSER:
- when a SIP client sends a request consisting of only a simple phone
number (on the form +CC-AAA-NNNNNN, with or without plus sign, hyphens,
possible spaces etc where C, A and N are simple numbers) OpenSER should
route the request to another, remote SIP-proxy acting as a PSTN gateway.
- either the same credentials should be used or preferebly credentials
dependent on the domain name of the callee for all requests to the
SIP-proxy
- after the session with the remote proxy has been set up OpenSER either
works as proxy between the client and the remote proxy, or better
transfers the session to the client
- all requests for a client not on the form of a phone number (i.e. with
a fully qualified domain name should be routed to the correct location
as decided by SRV-records etc.
Is this possible? Any hints on how to accomplish this?
Sincerely,
Kris
PS/Please note that I am new to OpenSER, SIP and SIP terminology...
Does anyone know if it is possible (and then how) to specify the RTP
port range on cisco TDM gateways (3640, 5300, 5400, 5800, etc)
It seems inconceivable that it can't be done on cisco hardware, but i
havn't been able to find any information on how to do it, i've talked to
several cisco engineers that don't know how to do it either.
Again, sorry for the OT question
tavis
Hi,
I'm wondering how to prevent bandwitdh lost with (open)SER+MediaProxy in
this type of network configuration:
UA0 --- [(open)SER+MediaProxy] --- [ Internet ] --- [ FW/NAT ] --- [UA1,
UA2]
UA1, UA2 are one the same network and they are registered in the (open)SER
server.
If UA1 call UA2 with default media proxy configuration, the RTP flow will
not be direct between this two UA. So I lost twice bandwith (from SER to UA1
and SER to UA2).
How can i prevent this ? On the REGISTRAR process, I think I could store the
public IP used by UA1 and UA2 and the two private IP of this UA. Since they
have the same public IP but a different private IP, on the INVITE procces I
suppose it's could be possible to indicate that RTP flow can be direct
between this two devices, no ? Now which tools (modules) could be used to
resolve this ?
Thanks,
Christophe
Dear serusers,
I am running 0.9.4 and am successfully logging calls to MySQL using the ACC
module.
I have added a statement to my ser.cfg to get the ACC module to NOT log
OPTIONS as follows:
if (method!="OPTIONS")
{
setflags..
}
The above works and I am not logging OPTIONS.
I tried the modification below, but it didn't work to get rid of ACK's. They
are still being logged.
if ((method!="OPTIONS") & (method!="ACK"))
{
setflags..
}
How can I get ACC NOT to log ACK's records into the database?
Leo P.
Hello All :
I have installed SER + SEMS to implement voicemail, but only the first call can connect to voicemail,
and can mail message.wav to callee's e-mail address .
but after the first call , i can not connect to voicemail until i restart SEMS
then i execute "sems -E " it shows that:
(29104) ERROR: run (AmServer.cpp:161): odd trailer
(29104) ERROR: run (AmServer.cpp:161): odd trailer
(29104) ERROR: run (AmServer.cpp:161): odd trailer
(29104) ERROR: run (AmServer.cpp:161): odd trailer
(29104) ERROR: run (AmServer.cpp:161): odd trailer
(29104) ERROR: run (AmServer.cpp:161): odd trailer
(29104) ERROR: run (AmServer.cpp:161): odd trailer
(29104) ERROR: run (AmServer.cpp:161): odd trailer
(29104) ERROR: run (AmServer.cpp:161): odd trailer
(29104) ERROR: run (AmServer.cpp:161): odd trailer
(29104) ERROR: run (AmServer.cpp:161): odd trailer
(29104) ERROR: run (AmServer.cpp:161): odd trailer
(29104) WARNING: sig_usr_fifo (sems.cpp:84): signal 2 received
(29103) WARNING: sig_usr_fifo (sems.cpp:84): signal 2 received
(29103) WARNING: sig_usr_fifo (sems.cpp:84): signal 17 received
How to solve this problem ??
*ser.cfg and sems.conf was attached.
本郵件附件清單如下:
(1). sems.conf.txt (1.4 K)
(2). ser.cfg.txt (7.3 K)
Hello All
In "ser.cfg" file how to configure MYSQL DB to use unix socket??
In my "sems.conf" have configure as fllow :
......
socket_name=/tmp/am_sock
ser_socket_name=/tmp/ser_socket
send_method=unix_socket
......
hi friends
there are no functions with name *init_hfname_parser* & *init_digest_parser
*in any file of the TAR package. can anybody plz tell me the function name
corresponding to the above functions in SER-0.9.4
Thanks
Nitin
I swear, getting OpenSER to do anything is worse than pulling teeth.
The avpops module documentation at:
http://www.openser.org/docs/modules/1.0.x/avpops.html#AEN165
has the following example under the db_scheme command:
...
modparam("avpops","db_scheme",
"scheme1:table=subscriber;uuid_column=uuid;value_column=first_name")
...
which causes the following error to be reported in syslog.conf:
Nov 27 14:24:52 ser1 openser: ERROR:avpops:parse_avp_sb_scheme: unknown attribute <uuid_column>
Nov 27 14:24:52 ser1 openser: ERROR:avpops:avp_add_db_scheme: falied to parse scheme
Nov 27 14:24:52 ser1 openser: parse error (102,3-4): Can't set module parameter
Why?
The example at the 'alleged' tutorial at:
http://www.voice-system.ro/docs/avpops/
which is just the docs at openser.org cut and pasted mostly has this example:
...
modparam("avpops","db_scheme",
"scheme1:uuid_col=uid;value_col=job;value_type=string;table=emp")
...
which does not generate any errors in syslog.
HOWEVER, when avp_db_load() is called, the following is logged to syslog.conf:
Nov 27 14:21:42 ser1 openser[2424]: ERROR:avpops_init: "AVP_DB" present but "AVP_TABLE" found empty
Nov 27 14:21:42 ser1 openser[2424]: init_mod(): Error while initializing module avpops
Why??? The docs say I can use dbscheme to use a non standard AVP table, which is what I am trying to do, namely the subscriber table. If I can use dbscheme to define the columns from another table, why is OpenSER complaining that I haven't called modparam("avpops","avp_table","avptable")?????
Doug.
That doesn't look like an RFC to me. It's not cataloged at rfc.net, and doesn't have a number. I didn't even know it existed. Thanks though. Like most things related to Openser, the knowledge is just assumed.
-----Original Message-----
From: Norman Brandinger [mailto:norm@goes.com]
Sent: Sun 11/27/2005 9:02 AM
To: Douglas Garstang
Cc: users(a)openser.org
Subject: Re: [Users] t_check_status
Before you can take advantage of what OpenSER has to offer, you should
become familiar with the the various RFC's associated with SIP
communication.
AVP stands for Attribute - Value Pair. Basically, it maps an attribute
for example, "NPA" to a value such as "212". The AVP's do have the
ability to use databases to store and retrieve values, however, this is
not a requirement. The configuration file can set, test and/or access
an AVP without any database access. In real life situations, you will
most probably want to use a database to store and retrieve AVP's.
RPID is defined here:
http://www.iptel.org/ietf/security/identity/draft-ietf-sip-privacy-04.txt
Regards,
Norm
Douglas Garstang wrote:
> I thought AVP stuff was database related.... this confuses me even more. I am reading the documentation you know.... constantly. Considering it looks like it was written by the people who developed OpenSER, there's a lot of implied/assumed knowledge there, and the scripting language is very high level, it's bloody difficult to work out what the documentation means.
>
> For example... where's the definitiaion of AVP? I still can't find what an 'rpid' is.
>
> -----Original Message-----
> From: Juha Heinanen [mailto:jh@tutpro.com]
> Sent: Sat 11/26/2005 9:29 PM
> To: Douglas Garstang
> Cc: users(a)openser.org
> Subject: RE: [Users] t_check_status
>
>
>
> Douglas Garstang said:
>
> > I really don't even get the point of what an AVP is. Something to do with
> > databases. I don't see why database interaction is needed to set a timer.
> > That's crazy.
> >
>
> i don't know what is cracy, since no database interaction is needed to set
> a timer. timers can be set in many different ways. read usrav
> documentation.
>
> --juha
>
>
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
> Users mailing list
> Users(a)openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users
>
I have found a strange bug in the ZyXEL 2602HW (firmware "V3.40(MV.3)")
that make it impossible to use it to make calls: the Request URI of the
ACK and BYE messages is addressed to the SIP server
("sip:195.120.250.50") instead of the other UA (i.e. something like
"sip:123@1.2.3.4").
I cannot understand how such an huge bug is not experienced by somebody
else.
Anybody had the same problem with the 2602HW?
Thanks.
--
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|ederico Giannici http://www.neomedia.it
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