Hi,
Would just like to ask, I have an SER SIP Proxy and I setup an Asterisk,
i used an SER local as a trunk for the Asterisk.
When the Asterisk box register to SER it will have this URI
sip:s@10.10.10.10, instead of sip:1001@10.10.10.10.
Anyone has encountered this problem? Because I'm checking the From part,
and "s" is not a valid extension number so it will deny it calling to
the gateway.
TIA
Regards
Nhadie
Hi fellows
Im trying to install Openser 1.0.1 in other server but it doenst running at all. It is the error:
0(24823) set_mod_param_regex: parameter <radius_config> not found in module <acc>
0(24823) parse error (130,19-20): Can't set module parameter
0(24823) set_mod_param_regex: parameter <service_type> not found in module <acc>
0(24823) parse error (131,20-21): Can't set module parameter
0(24823) set_mod_param_regex: parameter <radius_flag> not found in module <acc>
0(24823) parse error (132,19-20): Can't set module parameter
0(24823) set_mod_param_regex: parameter <radius_missed_flag> not found in module <acc>
0(24823) parse error (133,18-19): Can't set module parameter
0(24823) set_mod_param_regex: parameter <radius_extra> not found in module <acc>
0(24823) parse error (135,20-21): Can't set module parameter
I already installed MySQL, RadiusClient and Radius and configured the config files from all of them. When I compiled the Openser, I needed to compile the modules "AUTH_RADIUS", "URI_RADIUS" and "MYSQL" manually. After "make install", I copied the 3 so files to "/usr/local/lib/openser/modules".
The openser.cfg:
.
.
.
123: modparam("acc", "failed_transaction_flag", 1)
124: #modparam("acc", "report_ack", 1)
125: modparam("acc", "log_level", 1)
126: modparam("acc", "report_cancels", 1)
127: modparam("acc", "log_flag", 1)
128: modparam("acc", "db_flag", 1)
129: modparam("acc", "db_missed_flag", 1)
130: modparam("acc", "radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf")
131: modparam("acc", "service_type", 15)
132: modparam("acc", "radius_flag", 1)
133: modparam("acc", "radius_missed_flag",2)
134: modparam("acc", "log_fmt","cdfimorstup")
.
.
What im doing wrong?
Thanks for your help.
Bruno Machado
---------------------------------
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Hi All
I hope someone can help me figure this out. We've had a ser server running
for quite some time now on a Dell 2850 with dual cpu's. The load is very low
due to the use of multiple mediaproxy servers to handle the RTP streams. We
have about 2300 UID in the mysql database but at any given time only about
1300 show as registered in the location table and we have customers
complaining about not receiving calls. When tech support has a customer
power cycle their ATA, it registers just fine and they start receiving
calls. What could cause this? From the beginning, I noticed this was
happening but didn't give it much thought as the number of UIDs was low. But
now it is a growing problem. SER version is 0.9.0
Any hints or help would be appreciated.
Rick
Hello,
I use OpenSer with Cisco AS5300/5350/5400 with IOS 12.3.
When Cisco sends an OK 200 SIP message to OpenSer, there are two "c=IN
IPV4" tags in sdp body with the same IP address.
The nathelper module only replaces the last occurence of c= tag so some
SIP clients works wrong because those get the first IP from the first
"c=" tag...
How can i replace the all occurences of "c=" tag in sdp body with the
help of nathelper?
Thx,
Antal
Content-Type: application/sdp.
Content-Length: 277.
.
v=0.
o=CiscoSystemsSIP-GW-UserAgent 7144 773 IN IP4 213.253.220.242.
s=SIP Call.
c=IN IP4 213.253.xxx.xxx. -> this is the original address of Cisco GW
t=0 0.
m=audio 35944 RTP/AVP 18.
c=IN IP4 213.253.xxx.yyy. -> this is the nated address replaced by
nathelper
a=rtpmap:18 G729/8000.
a=fmtp:18 annexb=no.
m=video 35946 RTP/AVP.
c=IN IP4 213.253.220.253.
a=nortpproxy:yes.
Hi,
The docs of the siptrace module of v1.1.0,refer to the function
"siptrace()", where in fact it is defined as "sip_trace()" in the source.
Cheers,
Andy
hi all,
can a t_relay that got no answer be forwarded to a voicemail?
The scenario is the following:
i have a 'call group' sip-user (implemented in mysql database) that call
t_relay when an INVITE is rcvd. As consequence of the t_relay the mentioned
INVITE is forwarded to different destinations, and when one destination
answers the call a CANCEL is sent to other destinations. The problem exists
if no destination attends the call. How can I tell ser to forward the call
to voicemail if no user belonging to group 'call group' answers the call?
thanks!!!
victor
--
Victor Pascual Ávila
mail: victor.pascual.avila(a)gmail.com
sip: 17476374121(a)proxy01.sipphone.com
linux user: 224443
Hi,
I was wondering how I can route my SER users to other
SIP servers, without need of authentication to other
server.
I mean this:
My User -> Auth -> My SER
My SER -> Auth -> Other SIP Server
My User --------make call--------> My SER
--------route call-------> Other SIP Server
Thank you.
Kaveh
__________________________________
Yahoo! Mail - PC Magazine Editors' Choice 2005
http://mail.yahoo.com
Hi!
Does somebody know why 'fr_inv_timer_avp' does nothing when I use it as
follow:
...
modparam("tm", "fr_inv_timer_avp", "inv_timeout")
...
modparam("avpops","avp_table","avptimer")
# 'avptimer' is like 'usr_preferences'
...
avp_db_load("$ruri/username","i:inv_timeout/avptimer");
avp_write("i:inv_timeout","inv_timeout");
# In avptimer there is an attribute 'inv_timeout'
Actually I try to set a fr_inv_timer different from an user to another.
'avptimer' timer defines, for each users, their proper fr_inv_timer.
If somebody can help...
Thanks in advance,
Michel
Hi,
Is it posible to compile openser on Hp-Ux??
Has somebody try this??
Would I need something special??
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On Thursday 03 August 2006 00:46, Bill Zhang wrote:
> Anyone knows why I keep getting following error message?
>
>
>
> Aug 2 13:10:26 localhost openser[30695]: ERROR: extract_body: message body
> has length zero
>
> Aug 2 13:10:26 localhost openser[30695]: ERROR: force_rtp_proxy2: can't
> extract body from the message
The message says it, and your SIP message proves it: the SIP message has no
body.
nathelper operates on the SDP body, it mangles them (together with rtpproxy)
so that the SDP body sent to the receiver contains the address of your
machine running OpenSER instead of the real phone. This is so that the
machine can act as RTP proxy on network edges.
Since your SIP message has no SDP body there is nothing nathelper can do :-)
C'ya,
Marc
--
Marc Haisenko
Comdasys AG
Rüdesheimer Straße 7
D-80686 München
Tel: +49 (0)89 - 548 43 33 0
Fax: +49 (0)89 - 548 43 33 29
e-mail: haisenko(a)comdasys.com
http://www.comdasys.com
