Hello I Have a problem with my gatewas, because I want if one Gateway is
down sends to the other gateway i implemented something like this:
rewritehost("GATEWAY1");
t_on_failure("1");
t_relay();
And in failure 1:
rewritehost("GATEWAY2");
append_branch();
t_relay();
In the INVITE works fine because if GATEWAY1 is down or something is bad,
this sends a error and the INVITE is send to the GATEWAY2, but in the ACK is
the problem because the ACK allways is send to the GATEWAY1 and this dont
respond with an error or something like that.
Do you know how can I do to resolv this problem?
Regards
Atentamente:
Ing. Jesús Terán Blásquez
MCM Telecomunicaciones
53.50.00.57
Planeación
Helo everyone!
I've a problem with forwarding. My gateway return a "480" response when line is offline and I'm trying to forward to another gateway when line is offline (480 is return). I don't want to use t_on_failure. I write something like this:
if (uri="sip:0@.*"){
sethost("192.168.10.1")
t_relay();
if (t_check_status ("480"))
{
sethost("192.168.10.2");
t_relay();
break;
}
break;
}
But this script dont't work. I want to forward all calls only if return 480 sip response (not 486). Thanks for any help. Sorry for my bad english.
Swaper
Hi everybody.
As we know SER/RadiusClient sent the accounting information in the
following way:
04/05/2006 22:25:12 Acct-Status-Type = Start
04/05/2006 22:25:12 User-Service-Type = 15
04/05/2006 22:25:12 Sip-Response-Code = 200
04/05/2006 22:25:12 Sip-Method = 1
04/05/2006 22:25:12 User-Name = "55999999(a)192.168.75.4"
04/05/2006 22:25:12 Calling-Station-Id = "sip:55999999@192.168.75.4"
04/05/2006 22:25:12 Called-Station-Id = "sip:015527350900@192.168.75.4"
04/05/2006 22:25:12 Sip-Translated-Request-URI =
"sip:015527350900@201.37.23.105"
04/05/2006 22:25:12 Acct-Session-Id = "3565c759-7f2a08a6(a)172.16.1.50"
04/05/2006 22:25:12 Sip-To-Tag = "1FFFF924-24EF"
04/05/2006 22:25:12 Sip-From-Tag = "4c676fbdf6dbdb92o0"
04/05/2006 22:25:12 Sip-Cseq = "101"
04/05/2006 22:25:12 Client-Port-Id = 5060
04/05/2006 22:25:12 Acct-Delay-Time = 0
04/05/2006 22:25:12 Client-Id = 192.168.75.4
Is there a way at SER accounting parameters or Radius Client
Configuration that we can give a format to the
Calling-Station-Id/Called-Station-Id in order to strip the "sip:" string?
Regards
Alberto Cruz
Hi,
As I understood the SIP authentication process is based on RFC2617 i.e. the
same authentication process used in HTTP. So both Basic and Digest
authentication are possible.
Is-it possible to force always digest authentication ? The goal is to avoid
to have credential in clear if basic authentication is used. It's possible
on web server so I supposed it's possible also in SIP. Perhaps this feature
is by default on (open)SER since I never viewed in my tests passwords in
clear.
Thanks in advance,
Christophe
Hello to everybody,
I configured the openser.cfg as follow:
#-----------------------------------
# AUT_DB parameters
#----------------------------------
modparam("auth_db", "db_url", "mysql://openser:openser@localhost/openser")
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "domain_column", "domain")
modparam("auth_db", "user_column", "phone")
modparam("auth_db", "password_column", "password")
In this way openser checks for authentication the phone number instead of the username (it works fine!!). My problem is realted to domain. Entering the modparam(.....,"domain") I was expecting to see the domain column populated in the table location, but it doesn't happen. Do I have to configure something else? Moreover, anybody could explain me the meaning of the column location.received?
thanks in advance
Alessandro
Hello,I have the following setup:
UA<->NAT<-->SER/MediaProxy<->Prepaid(B2BUA)<-->PSTNGW
|
IVR
Nat'd user calls a number that is forwarded (t_relay)
to Prepaid and the user is prompted for
PIN#/Destination Phone# and all works fine. Mediaproxy
is invoked and a mediaproxy port# (35774) is assigned
during the whole process.
Once the user enters the destination number, the
Prepaid forwards the call to the PSTNGW and the PSTN
phone rings (Call signalling working fine). When the
PSTN phone is answered, no media can be heard!!!
I did some investigation and here is what is
happening:
Once the user enters the destination# to dial, The
Prepaid/B2BUA does two things:
1) Sends a re-invite to put the UA on hold.
2) Sends an INVITE to the PSTN GW and retrieves the
SDP from the response from PSTN GW
3) Sends a second-reinvite to the UA via SER with the
SDP info of the PSTNGW
4) SER invokes the mediaproxy (since it is reinvite)
and assigns the SAME MEDIAPROXY PORT# as earlier on
when the media was flowing fine (SDP has audio port#
35774)
5) SER forwards the re-INVITE TO UA
6) UA responds with a 200 OK and sends 200 OK to SER.
SER agains invokes mediaproxy and assigns the SAME
MEDIAPROXY PORT# (audio port 35774) and forwards the
response to Prepaid (which sends it to PSTNGW).
I checked on Cisco PSTNGW that it is creating a
session with the mediaproxy (port:35774). BUT NO AUDIO
CAN BE HEARD IN ANY DIRECTION.
Can anyone please help? Is this a mediaproxy issue
that when reinvites are sent and mediaproxy is invoked
multiple times, issues arise?
I am running the latest mediaproxy
version.
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Dear all,
I have installed rtpproxy from
http://www.openser.org/downloads/snapshots/rtpproxy/rtpproxy-cvs-latest.tgz
on the same computer as Openser.
I started it without options. /var/run/rtpproxy.pid and .sock are created.
I suppose a communication is established between OpenSer and Rtpproxy as SDP
messages are correctly replaced, it means "Connection Information c" shows
rtpproxy IP address and ports.
Also both UAs sent RTP media to rtpproxyIP:port described in SDP body, but
rtpproxy doesn't forward any RTP packets from one UA to the other.
Tcpdump -n udp and host 213.48.xxx.yyy result:
11:48:49.191130 IP 213.48.xxx.yyy.2236 > 207.245.aaa.bbb.35000: UDP, length
60
11:48:49.250183 IP 213.48.xxx.yyy.2236 > 207.245.aaa.bbb.35000: UDP, length
60
11:48:49.255407 IP 213.48.xxx.yyy.2237 > 207.245.aaa.bbb.35004: UDP, length
60
11:48:49.297050 IP 213.48.xxx.yyy.2236 > 207.245.aaa.bbb.35000: UDP, length
60
11:48:49.297275 IP 213.48.xxx.yyy.2237 > 207.245.aaa.bbb.35004: UDP, length
60
11:48:49.335890 IP 213.48.xxx.yyy.2237 > 207.245.aaa.bbb.35004: UDP, length
60
11:48:49.339723 IP 213.48.xxx.yyy.2236 > 207.245.aaa.bbb.35000: UDP, length
60
11:48:49.393997 IP 213.48.xxx.yyy.2237 > 207.245.aaa.bbb.35004: UDP, length
60
Have you ever come across that issue? How can I get 207.245.aaa.bbb.35000 >
213.48.xxx.yyy.2237? Does this problem occur because both UAs are in the
same local network? (I don't thing so)
I would be very thankful if you could give me an idea about it.
Thanks a lot
Paul
Dear All,
i have many users to add to my mysql. But if i use e.g.
serctl add user1 123456 user1@mydomain
ich need to enter the password per hand.
So, howto can i do it quickly?
Thanks
Lei
Dear all,
we are trying to replace the from_uri with uac_replace_from function as
following, but the actual from_uri is not replaced after
uac_replace_from is called as debug log. so when we call load_gw() and
next_gw() later in the configuration, the gateway came up is not for the
replaced from_uri, it queried with original from_uri. Does anyone know
how to completely replace the from_uri from original one. or a avp id
that can force the replacement?
uac_replace_from("$avp(s:newcallerid)","sip:$avp(s:newcallerid)@xxx.xxx.xxx.xxx:8080");
avp_print();
0(20433) DEBUG:avpops:print_avp: p=0xf4f18568, flags=3
0(20433) DEBUG: name=<currentcallerid>
0(20433) DEBUG: val_str=<orignalcallerid>
thanks
Ray
