Hi everybody, i'm still new to SER and i dont fully understand my config file :)
I can make outgoing calls, but incoming calls dont get routed through. They stay in a loop at my external interface.
perhaps someone could tell where the mistake is.
Here is my config:
debug=4 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=yes # (cmd line: -E)
listen=10.*.*.*
listen=193.*.*.*
mhomed=yes
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
# ------------------ module loading ----------------------------------
loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/rr.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/usrloc.so"
loadmodule "/usr/lib/ser/modules/registrar.so"
loadmodule "/usr/lib/ser/modules/textops.so"
loadmodule "/usr/lib/ser/modules/nathelper.so"
modparam("usrloc", "db_mode", 0)
modparam("rr", "enable_full_lr", 1)
modparam("registrar", "nat_flag", 6)
modparam("nathelper", "natping_interval", 30) # Ping interval 30 s
modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT
route{
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if (msg:len >= max_len ) {
sl_send_reply("513", "Message too big");
break;
};
if (nat_uac_test("3")) {
if (method == "REGISTER" || ! search("^Record-Route:")) {
log("LOG: Someone trying to register from private IP, rewriting\n");
fix_nated_contact(); # Rewrite contact with source IP of signalling
if (method == "INVITE") {
fix_nated_sdp("1"); # Add direction=active to SDP
};
force_rport(); # Add rport parameter to topmost Via
setflag(6); # Mark as NATed
};
};
if (!method=="REGISTER") record_route();
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
break;
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
break;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
force_rtp_proxy("FAEI");
t_on_reply("1");
if (!t_relay()) {
sl_reply_error();
};
}
onreply_route[1] {
fix_nated_contact();
force_rtp_proxy("FAII");
# otherwise, is it a transaction behind a NAT and we did not
# know at time of request processing ? (RFC1918 contacts)
if (nat_uac_test("1")) {
fix_nated_contact();
};
}
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Hi,
I want to know if it is possible to pass
psuedo-variables, or avp variables insdie t_relay()
function.
Here is what I tried:
t_relay("tcp:$rd:$rp");
t_relay("tcp:\$rd:"\$rp");
avp_write("$rd", "$avp(reqhost)");
avp_write("$rp", "$avp(reqport)");
t_relay("tcp:$avp(reqhost):$avp(reqport)");
Then, I got the following error:
ERROR: parse_phostport: bad port number in ....
ERROR:tm:fixup_phohstport2proxy: invalid parameter...
ERROR: fix_actions: fixing failed (code=-1) at cfg
line 513
Is there any way to solve this problem?
I want to do this in order to specify which protocol
to use for sending the SIP Request using the Req-URI
domain and port.
Thanks,
Vivek
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Hello everybody,
I'm new to SER and try to write my first config file. After reading the Admin guide & the getting started Tutorial on onsip I still cannot configure my rtpproxy.
In the test setup I have two systems --
1. The Linux server ( where the SER is running)
2. one win XP system ( SIP UA).
The linux server has two LAN cards one with a private IP and one with a public one.
The SIP - Client has an account at a VoIP - Provider (sipgate).
What I want the SER to do is to forward the call to the provider and to accept incoming calls.
No authorisation or mysql things nedeed.
Until now SIP - communication works, but the RTP - Stream is not forwarded correctly.
I already tried to start rtpproxy as follows:
1. rtpproxy
2. rtproxy -l 10.*.*.*
3. rtpproxy -l public-IP
4. rtpproxy -l 10.*.*.*/public-IP
5. rtpproxy -l public-IP/10.*.*.*
I use this config for SER: http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/sip_router/etc/nathelper.cfg?…
Thanks in advance.
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I am trying to use avpops and getting the following message in the log file:
ERROR:avpops:fixup_db_avp: source/flags "ru" unknown!
The relevant part from openser.cfg is:
avp_db_load("$ru/username", "s:Variable");
However when I look at the pseudo-variables documentation $ru is defined.
Curiously, elsewhere I am using avp_db_load("$from", "s:Variable"); and that
works even though $from is not defined as a pseudo-variable in the
documentation. Am I missing something?
OpenSer 1.0.1
Any advice appreciated.
Cameron
Sorry, i ran the older version of openser not the new. :-)
thx
-----Original Message-----
From: Pletli Antal
Sent: Friday, August 04, 2006 8:45 AM
To: users(a)openser.org
Subject: RE: [Users] Two c= tags in sdp body
Hi,
It doesn't work, with the 'c' flag all the c= tags remain unchanged.
-----Original Message-----
From: Ovidiu Sas [mailto:sip.nslu@gmail.com]
Sent: Thursday, August 03, 2006 5:00 PM
To: Pletli Antal
Cc: users(a)openser.org
Subject: Re: [Users] Two c= tag in sdp body
Check the new 'c' flag in 1.1 release:
http://www.openser.org/docs/modules/1.1.x/nathelper.html#AEN275
-ovi
On 8/3/06, Pletli Antal <a.pletli(a)euromacc.hu> wrote:
>
>
>
> 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.
> _______________________________________________
> Users mailing list
> Users(a)openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users
>
>
>
Hi,
It doesn't work, with the 'c' flag all the c= tags remain unchanged.
-----Original Message-----
From: Ovidiu Sas [mailto:sip.nslu@gmail.com]
Sent: Thursday, August 03, 2006 5:00 PM
To: Pletli Antal
Cc: users(a)openser.org
Subject: Re: [Users] Two c= tag in sdp body
Check the new 'c' flag in 1.1 release:
http://www.openser.org/docs/modules/1.1.x/nathelper.html#AEN275
-ovi
On 8/3/06, Pletli Antal <a.pletli(a)euromacc.hu> wrote:
>
>
>
> 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.
> _______________________________________________
> Users mailing list
> Users(a)openser.org
> http://openser.org/cgi-bin/mailman/listinfo/users
>
>
>
Hi USERS,
Happy
FriendShip
Day
on this Sunday ,,
in Advance...................
--
Thanks and Regards with cheers
Sunkara Ravi Prakash (Voip Developer)
Hyperion Technology
Kondapur, Hi-tech city,
Hyderabad.
www.hyperion-tech.com
+91-9985077535
Dear all,
Now, I try to test my openser configuration by using xlite. I tried to call from client to test the openser configuration that I have built, can work sucessfully.
So, I must create a new user by using "openserctl add" command.
But when i tried to do that, I got an error message. The error message said that " WARINIG: no database engine found - tried ''
This command requires a database engine - none was loaded".
I think this message said that I can not add a new user because no database can be loaded. What is wrong? However, I can run mysql goodly.
This is the task that I have done:
[root@localhost openser]# openserctl add aldi aldi aldi(a)sip.pcr.ac.id
WARINIG: no database engine found - tried ''
This command requires a database engine - none was loaded
What can i do to solve this problem? I do hope anybody can give me a sugesstion.Please help me...Please
Note: I use mysql for the database, and it can work goodly. ( I can show all the table that there is in openser database). They are aliases, dbaliases, etc.
Thanks with cheers,
Ferianto
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Yes you can. We use prefix and strip to add and subtract prefixes before
t_relay
Rick
-----Original Message-----
From: serusers-bounces(a)lists.iptel.org
[mailto:serusers-bounces@lists.iptel.org] On Behalf Of Nhadie
Sent: Wednesday, August 02, 2006 9:43 AM
To: serusers(a)iptel.org
Subject: [Serusers] accounting for prefixed number
Hi All,
I have a client using an asterisk server for them to use my sip proxy, what
I did was add a trunk on their asteriks with a local extension from my SER
sip proxy.
Can I also just do a prefix ID for them instead of trunk? Meaning when they
send me a call they should prefix it with e.g 1234# so I will identify that
it came from that client, and how can I account that?
TIA
Regards
Nhadie
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