Hi Users,
Today OPENSER's I st Anniversary ,
hi dedues please solve my issues
In Local network openser with radius (AAA) is working fine. by using xten
softphones
But when i'm deploy in some other places with different network .
Its successfully logined , but when making the calls its gives log as '
called failed : 408 request time out..."
here is my openser,cfg below
...........................
modparam("usrloc", "db_mode", 2)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "calculate_ha1", yes)
modparam("usrloc|acc|auth_db|group|msilo", "db_url", "
mysql://openser:openserrw@localhost/openser")
########333333
modparam("acc","log_level",1)
modparam("acc","log_flag",1)
modparam("acc","log_missed_flag",2)
modparam("acc", "log_fmt", "cdfimorstup")
#modparam("acc", "failed_transaction_flag",3)
modparam("acc", "report_cancels", 1)
modparam("acc","report_ack",0)
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 2)
modparam("acc","radius_flag",1)
#modparam("acc","radius_level",1)
#modparam("acc","radius_missed_flag",2)
modparam("acc","service_type",15)
#modparam("acc", "detect_direction", 1)
modparam("acc","radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf")
modparam("auth_radius","radius_config","/usr/local/etc/radiusclient-ng/radiusclient.conf")
##########33
modparam("nathelper","natping_interval",30)
modparam("nathelper","ping_nated_only",1)
modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock")
#################33
modparam("rr", "enable_full_lr", 1)
###############################################
route {
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("13")) {
sl_send_reply("483","Too Many Hops........................!");
exit;
};
if ( msg:len > max_len ) {
sl_send_reply("513", "Message too big");
exit;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if(!method== "REGISTER")
{
record_route();
};
# loose-route processing
if (loose_route()) {
# add now 0n 9 setflag(1);
acc_rad_request("200 ok ");
t_relay();
exit;
};
# account completed transactions via syslog
#setflag(1);
# setflag(2);
if(uri==myself)
{
if(method=="REGISTER")
{
if(!radius_www_authorize("192.168.2.55"))
{
www_challenge("192.168.2.55","0");
exit;
};
#consume_credentials();
save("location");
if (m_dump())
{
log("MSILO: offline messages dumped - if they
were\n");
}else{
log("MSILO: no offline messages dumped\n");
};
exit;
};
if(method=="INVITE" && method=="ACK")
{ record_route();
acc_rad_request("200");
force_rtp_proxy();
#setflag(1);
t_on_reply("1");
};
/* if (method=="BYE") {
record_route();
};
*/
if (method=="MESSAGE") {
log(1, "MESSAGE\n");
setflag(1); /* set for accounting (the same value as in
log_flag!) */
};
if (method=="BYE" || method=="CANCEL") {
#log (1, "BYE or CANCEL\n");
#setflag(1);
acc_rad_request("200 ");
#acc_rad_request("Stop");
#unforce_rtp_proxy();
#setflag(1);
};
if(!lookup("location"))
{
sl_send_reply("404","Woo......... NOt found");
};
};
if(!t_relay())
{
sl_reply_error();
};
lookup("aliases");
#setflag(2);
exit;
}
onreply_route[1]
{
if(status=~"[0-9][0-9][0-9]")
{
force_rtp_proxy();
};
}
-----------------------------------------------------------------------------------
please help me........
--
Thanks and Regards with cheers
Sunkara Ravi Prakash (Voip Developer)
Hyperion Technology
www.hyperion-tech.com
<b>
I'm trying to use Asterisk's B2BUA with Openser. However I read somewhere that Asterisk can't handle as much concurrent calls as Openser can with SIP. So that's why I am trying to use both of them together.
This is gonna be somewhat I'm trying to do:
UA ----->Openser ---> Asterisk ------> PSTN Gateway or SIP
All UA should be regisetered on Openser using mysql database.
Question is, if I use this implementation, wouldn't all the call load be on Asterisk instead of Openser?
Maybe I'm getting the wrong picture, as I'm fairly new to this. So I appreciate as much help as possible. I'm trying to do this for my senior design. Thank you.
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Hi Greg,
looks like you will have to start debugging :-)
Normally, SER doesn't care of payload (you could do H.323 tunnelling in it
if you wished ;-)) I can only imagine SER can complain about the message
if it uses some SDP-mangling module like nathelper but even then i would
not expect SER dropping silently. Let me know what you have found out
in any case -- it looks like a perfectly legitimate SIP request.
-jiri
At 07:44 14/06/2006, Greg Fausak wrote:
>Before I start debugging I thought I'd ask.
>I have a sip message from a gateway (Cisco IOS) that
>is being sliently dropped. Should ser be able to handle
>a message that looks like this? I haven't done a multipart
>message before.
>
>Thanks,
>---greg
>
>INVITE sip:+19194724170@sn-sip-in.ca-sn1.cisco.com:5060 SIP/2.0
>Via: SIP/2.0/UDP 172.18.109.91:5060;branch=z9hG4bK4F0109C
>Remote-Party-ID: <sip:
>+19199915651(a)172.18.109.91>;party=calling;screen=yes;privacy=off
>From: <sip:+19199915651@172.18.109.91>;tag=249E4980-FFC
>To: <sip:+19194724170@sn-sip-in.ca-sn1.cisco.com>
>Date: Tue, 13 Jun 2006 19:37:55 GMT
>Call-ID: F241878C-FA4A11DA-81EFC5C8-2F1D7951(a)172.18.109.91
>Supported: 100rel,timer,resource-priority,replaces
>Min-SE: 1800
>Cisco-Guid: 4064260884-4199158234-2157445126-1397050494
>User-Agent: Cisco-SIPGateway/IOS-12.x
>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, UPDATE, REFER,
>SUBSCRIBE, NOTIFY, INFO, REGISTER
>CSeq: 101 INVITE
>Max-Forwards: 10
>Timestamp: 1150227475
>Contact: <sip:+19199915651@172.18.109.91:5060>
>Expires: 180
>Allow-Events: telephone-event
>Content-Type: multipart/mixed;boundary=uniqueBoundary
>Mime-Version: 1.0
>Content-Length: 797
>
>--uniqueBoundary
>Content-Type: application/sdp
>Content-Disposition: session;handling=required
>
>v=0
>o=CiscoSystemsSIP-GW-UserAgent 8340 2382 IN IP4 172.18.109.91
>s=SIP Call
>c=IN IP4 172.18.109.91
>t=0 0
>m=audio 19122 RTP/AVP 18 3 0 4 100 101
>c=IN IP4 172.18.109.91
>a=rtpmap:18 G729/8000
>a=fmtp:18 annexb=yes
>a=rtpmap:3 GSM/8000
>a=rtpmap:0 PCMU/8000
>a=rtpmap:4 G723/8000
>a=fmtp:4 annexa=yes
>a=rtpmap:100 X-NSE/8000
>a=fmtp:100 192-194
>a=rtpmap:101 telephone-event/8000
>a=fmtp:101 0-16
>
>--uniqueBoundary
>Content-Type: application/gtd
>Content-Disposition: signal;handling=optional
>
>IAM,
>PRN,isdn*,,NI***,
>USI,rate,c,s,c,1
>USI,lay1,ulaw
>TMR,00
>CPN,02,,1,4724170
>CGN,04,,1,y,2,9199915651
>CPC,09
>FCI,,,,,,,y,
>GCI,f23fb314fa4a11da8098000653454c7e
>
>
>--uniqueBoundary--
>
>
>
>--
>Greg Fausak
>greg(a)thursday.com
>_______________________________________________
>Serusers mailing list
>Serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
Hi,
sorry, i could not understand what u meant by that.... please elaborate a bit,
Thanks,
Padmaja
----- Original Message -----
From: ram
To: Padmaja RV
Sent: Wednesday, June 14, 2006 4:30 PM
Subject: Re: [Users] openser forking
Ngrep should help you for the same
ram
On 6/14/06, Padmaja RV <padmaja.rv(a)vodcalabs.com> wrote:
Hi all,
i need to implement openser for forking an incoming sip call.
can anyone pls guide me how to do this?
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http://openser.org/cgi-bin/mailman/listinfo/users
Hello everyone,
I've finally got openser working with mysql. Now I'm trying to integrate Openser with Asterisk.
I read Asterisk at large. However I have a few questions:
1) It is said that openser as a front end to asterisk will enable more concurrent calls comparing to asterisk alone. How is that so?
2) If openser acts as a front end connecting all SIP calls, what is the role of asterisks?
3) Will the load of SIP calls be handled by openser alone? Or is it both asterisk and openser?
I'm confuse with this, please help a newbie out here. Thank you in advance.
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Greger,
Im already doing this for calls coming into my SER installation.
However I want to do it the other way around, i.e. from SER/UA to PSTN
Therefore I want to rewrite the username if the outgoing INVITE message to
be the alias, not the location, when sending to the PSTN.
Andy
_____
From: Greger V. Teigre [mailto:greger@teigre.com]
Sent: 12 June 2006 09:33
To: Andy Thomas
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] rewriting the outgoing URI to its alias
I see a lot of confusion on the different user related ids in SIP. Let me
try to explain (and answer your question, Andy, at the end of the post...)
Let's start at the user agent(UA) side:
--------------------
AOR: Address of record is the sip:myuser@domain.com address that you are
known as, just like your email address. Many user agents don't ask for AOR,
but builds it from other parameters you need to set
Username: The username is normally the user (before @) of AOR)
Realm: Often the domain portion of the AOR, thus AOR = sip:username@realm
Friendly name/displayname: "My name"
Out of this info, the From header is constructed:
From: "My name" <sip:username@realm>
---
Entirely separate (but sometimes confused in user agents):
Authentication username: The user used for authentication
Authentication realm: The realm to authenticate within
Very often the authentication realm is implicitly assumed to be the same as
the AOR realm
---
Then, the user agent will create the Contact header. The Contact header
should be the public contact address of your current location. Thus:
Contact: sip:username@mypublicip:5060
If the user agent is behind a NAT, the mypublicip will be a private address.
---
A sidenote: Unless registration server is explicitly specified, the realm in
the AOR will be used for looking up the SIP registration server using DNS
SRV or A lookups. You should avoid putting the FQDN of your SIP server in
the realm.
Also, you may in some user agents specify outbound proxy. This is the proxy
where the user agent will send INVITEs (and other outbound messages).
-------------------
And on the SER side:
-------------------
Authentication user/realm are used to do Digest authentication, but are then
forgotten (i.e. not stored).
The AOR is registered in the location table, together with the Contact
header, as well as the source ip and port (if different from Contact). The
fix_nated_register() function handles this setting of the so-called received
parameter.
So, to the routing:
- Messages that need to be routed (i.e. do not have Route headers) will have
a Request URI; the first line and the part after the message type: INVITE
sip:username@domain.com
- It is by changing this request URI, you do routing. The t_relay() command
uses the URI to forward the message correctly
- Before forwarding to a user agent, you want the request uri to be the same
as the stored Contact header for the AOR you are looking up. If not, the
user agent may reply with a 404 User not found
- The From header is NOT used for routing, and for backwards RFC
compatibility, you should not change the From header as some UAs will use
the content of the From header to match a dialog (however, if your UAs from
experience still work, there should not be a problem doing it, it's just not
RFC-compliant and may pop up and kick your butt later ;-)
- The AOR in location table is used for looking up incoming messages if you
can find a direct match between the Request-URI in the incoming message and
the stored AOR. If not, you can use the aliases table (and
lookup("aliases")) to match the Request-URI with something in the aliases
table, that again will map to the AOR in the location table
---
So keeping this (fairly simple) concept in focus: Routing (regardless of
LCR, avps or whatever) should focus on finding the correct Request-URI
before you call t_relay().
BUT, there are ways of "messing" up this...
---
- There are several commands in SER you can use to override the Request-URI
(forward_*). They should be avoided, unless you have a valid reason for
having a Request-URI in the message you are forwarding that is NOT
resolvable (either IP address or DNS name or DNS SRV/A resolvable) to the
party you are forwarding to. The reason can be if you want the R-URI to
contain the AOR and then forward the message to a server handling voicemail
- The dst_uri parameter (implicitly set by lookup) will tell t_relay() to
send the message to dst_uri instead of the Request-URI. lookup() will set
this when it finds that the Contact stored for the AOR also has an
associated received ip:port (because the user agent was NATed)
In general trying all sorts of tricks o fix-up things the way you need it
may not be so smart. Stick to the basics and question yourself: Is this
something I really want to do?
----------------------
To Andy, you want the aliases table to map to the AOR (as registered by the
UA in the REGISTER command). You do lookup("aliases") to resolve your DID
into an AOR. Then, later you can do lookup("location") to map the AOR to the
location of the UA (i.e. Contact/dst_uri). Then your Request-URI will be
correct.
g-)
Andy Thomas wrote:
Does anyone know how I would do this-
For all users who have PSTN access, they are assigned a number in the MySQL
alias table which matches a PSTN DDI number.
e.g. user 8000 has an alias of 2071231234, so on an incoming call the
lookup("aliases") function correctly matches the DDI to the user.
I want my ser.cfg to rewrite the user on an outgoing call, if a number
exists in the alias table for that user
Obviously, the rewriteuser function will be used, but what do I put in after
that?
I have tried rewriteuser (lookup ("aliases")) but that doesn't work
Can anyone help?
_____
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Is it possible to configure the OpenSER with nathelper/mediaproxy to act as a remote-side outbound proxy? I have tried to make it, but unfortunately still can not do that. It will be best if someone shares a working openser.cfg file. The best I got is that when the UA makes call it works OK, but the UA can not be dialed from other UA in the network. Maybe it is not even registered.
Hi,
I have installed the Presence Version of SER on a SUSE Linux Server.
If I want to add a user with the command "serctl add ...", I get following
message:
"error: 500 Command 'ul_show_contact' not found"
The same with serctl moni:
500 Command 'version' not found
500 Command 'uptime' not found
...
I hope, someone can help me.
Thanks,
Alois
