I'm having a problem with the route sections below. When my proxy
receives an Invite to proxy to a PSTN gateway, the proxy is sending 2
invites very fast to the gateway in the first route block which ends up
creating a mess with the gateway.
Either I have something wrong in the routing code or I have an
SER/OS/hardware timing problem.
Any suggestions ?
route[1] {
xlog("L_INFO", "Sending to route 1\n%mb\n");
rewritehostport ("1.1.1.1:5060");
append_branch();
t_on_failure("1");
t_relay();
}
failure_route[1] {
if(t_check_status("487")) {
break;
}
xlog( "L_INFO", "failure on route 1\n%mb\n");
append_branch();
route(2);
break;
}
route[2] {
xlog("L_INFO", "Sending to route 2\n%mb\n");
rewritehostport ("2.2.2.2:5060");
append_branch();
t_on_failure("2");
t_relay();
}
failure_route[2] {
if(t_check_status("487")) {
break;
}
xlog( "L_INFO", "failure on route 2\n%mb\n");
append_branch();
route(3);
break;
}
route[3] {
xlog("L_INFO", "Sending to rout 3\n%mb\n");
rewritehostport ("3.3.3.3:5060");
t_relay();
}
Thanks,
Matt
Dear All,
I need your help for establishing SER as a SIP server for my organization.
I have downloaded it at my fedora 8.0 machine, but as i am new with linux
and SER it is hard to find me how to install/configure.
kindly help me as a new SER user.
regards
--
=========================
Noman
=========================
hi friends:
i need some help!
frist,my asterisk receive a sip message from sip proxy via SER.when my asterisk send 200 to SER and then receive ACK from proxy directly!
when asterisk send BYE to the proxy and can hangup the call.
but now i redirect the BYE to SER, i found the SER can not send the message to proxy! what can i do in the ser.cfg?
thanks in advance
2008-09-10
邱磊
I need to add 1 to an avp like :
$avp(i:681)=$avp(i:681)+1;
but don't work!
how to?
thank you
Cordialement
BERGANZ François
http://www.acropolistelecom.net
TEL/FAX : 33 (0) 1 70 72 50 15
Hi all,
I've a doubt regarding a statement at the end of page 5
"... Some important design decisions are associated with carrying
assertions in a SIP request. If an assertion is carried by value, or
uses a MIME-based content indirection system, then proxy servers will be
unable to inspect the assertion themselves. If the assertion were
referenced in a header, however, it might be possible for the proxy
to acquire and inspect the assertion itself. There ..."
Why a proxy server will be unable to inspect the assertion themself ?
SER has functions by which it could be read message body, does this
behavior not respect any rfc ?
I'm missing something :-S
Thanks in advance,
Francesco la Torre
Hi Gustavo,
thanks for your reply.
I've changed the *avp type* from "2" to "0" but I get always the same
error:
ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57>
Now I'm going to reboot the server... If I'll have news I'll let you know.
Regards,
Stefano
-------- Messaggio Originale --------
Oggetto: Re: [Kamailio-Users] Unconditional call forward
Data: Wed, 3 Sep 2008 10:38:42 -0300
Da: Gustavo Mistrinelli <gmistrinelli(a)gmail.com>
A: Stefano Palleschi <stefano.palleschi(a)okcom.it>
CC: users(a)lists.kamailio.org
Referenze: <48BE6299.5000805(a)okcom.it>
Hi Stefano,
Please check *avp type* that you are using, use type 0 instead of 2
See below the different types for avps:
0 - *AVP* with string name and string value
1 - *AVP* with string name and integer value
2 - *AVP* with integer name and string value
3 - *AVP* with integer name and integer value
Regards,
Gustavo
On Wed, Sep 3, 2008 at 7:10 AM, Stefano Palleschi
<stefano.palleschi(a)okcom.it <mailto:stefano.palleschi@okcom.it>> wrote:
Hello all,
i'm trying to do an unconditional call forward with openser. I've added
the instructions below in the script:
if (avp_db_load("$ru/username", "$avp(s:callfwd)")) {
avp_pushto("$ru", "$avp(s:callfwd)");
xlog("$avp(s:callfwd)");
route(1);
exit;
}
and i've created a record on the usr_preferences table of openser
database:
mysql> select * from usr_preferences;
+------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+
| uuid | username | domain | attribute |
value | type | modified |
+------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+
| 1 | 0666620996 | 81.29.176.57 <http://81.29.176.57> | callfwd |
sip:0666653218@81.29.176.57 <mailto:sip%3A0666653218@81.29.176.57> |
2 | 2008-09-03 09:51:33 |
+------+------------+-----------------------------+-----------+-----------------------------+------+---------------------+
1 row in set (0.00 sec)
like showed, i would like forward the call that arrive at the sip client
0666620996 to the pstn number 0666653218.
At openser restarting I get always this error:
ERROR:avpops:load_avps: incomplet uri <sip:81.29.176.57
<http://81.29.176.57>>
without make any call.
Why in the uri there isn't the first part (username@)?
Thanks in advance.
Stefano Palleschi
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--
Gustavo Mistrinelli
Ok,
I am trying to get a redirect server working correctly, only cache is I need to hit another redirect server first, and grab info from the contact in the that 302 reply before processing and sending back a 300 to the original switch.
call flow would be something like this:
call comes into kamailio,
1) kamailio sends the invite to another proxy(1),
2) that proxy returns a 302,(if it times out it moves on to step 5)
3) kamailio sends an ACK back to that proxy(1),
4) kamailio looks at the contact header of the 302 reply,
5) depending on that info, I decide where to tell the originating switch where to go,
5) kamailio sends originating switch a 300 redirect with new contact header.
6) originating switch sends back ACK.
I have tried doing this with a first step going to route(1) , which does a "t_on_reply" and "t_relay" to the second proxy.
The onreply route, graps the contact saves it into variable and then sends to route(2),
Route(2) looks at the variable, does some other lookups and sends t_reply("300","redirect").
The switch then sends the ACK, but kamailio says its not a local ACK and ignores it..
Then it keeps sending 300's back to the switch...
Please help.. Thanks in Advance.
kent
Folks,
My problem for forward call, that I need to change username/password
and realm to do my
forward.
So, I try this:
modparam("uac","from_restore_mode", "auto")
modparam("uac","auth_realm_avp","$avp(s:uac_realm)")
modparam("uac","auth_username_avp","$avp(s:uac_username)")
modparam("uac","auth_password_avp","$avp(s:uac_password)"
route {
...
if(uri=~"^sip:0[0-9]{10}@") {
xlog("L_INFO","uac_atual: $avp(s:uac_username) !!");
$avp(s:uac_username)="username";
$avp(s:uac_realm)="1.1.1.1";
$avp(s:uac_password)="password";
xlog("L_INFO","uac_trocado $avp(s:uac_username) !!");
route(3);
exit;
}
..
}
...
route[3] {
t_on_failure("3");
# reset flag to mark no authentication yet performed
resetflag(7);
# forward to PSTN
uac_replace_from("username","sip:username@1.1.1.1");
rewritehostport("1.1.1.1:5060");
xlog("L_INFO","rewritehost !!!!!!!!");
t_relay();
}
..
So, when I do xlog, in the first entry, uac_username don't display, why ?
And its strange, because in one supplier its ok, and in another its not run.
But I don't undestand why uac_username is empty in the first xlog!
Thanks,
-Thiago Rondon
Anyone lnow what this means?
dialog:populate_leg_info: bad sip message or missing Contact hdr
ERROR:dialog:dlg_onreq: could not add further info to the dialog
openser 1.3.2
thnks