[Serusers] PSTN Fowarding Problem
jht2
hackglacier at 163.com
Mon Dec 20 20:45:15 CET 2004
Hi:
I am using SER to fowrad calls to a Cisco 2600 Gateway to PSTN call.But strange problem occured.
1, When I use X-lite to dial out,everything is OK.But when I use SJphone and some other hardware phone to dial out,the call cann't be fowarded and get "400" message from Cisco,which I found different from X-lite's call is there is an error message during the invite:"ERROR: extract_mediaip: no `c=' in SDP"(no apper during the X-lite's calling) . And If I get rid of the "fix_nated_sdp("1")" in the ser.cfg,no more 400 message feedback,the Sjphone and Hardware phone can connect with the PSTN number but only single way audio,it seems the RTP stream is abnormal.while get rid of "fix_nated_sdp("1")",the internal call between 2 X-lites is OK.
2. And another problem is while I use Radius for accounting,even I use X-lite make a successful call to Cisco,it hasn't start message in Radius log detail file,only "408" is logged as "Failed" logged on Acct-Status-Type.While internal calls between X-lites,the Radius log is properly correct with Invite 200 starts and 200 stops.The detail file is also pasted below.
Any Advice? Thanks.
A. Debug Log of Hardwarephone(Invite part,then get 400 bad requrest) :
11(23018) Sending:
INVITE sip:008613381786981 at 84.233.140.73:5060 SIP/2.0
Record-Route: <sip:008613381786981 at 62.164.130.1;ftag=qy9DjT5ubwqB6Ttp;lr=on>
Via: SIP/2.0/UDP 62.164.130.1;branch=0
Via: SIP/2.0/UDP 192.168.1.88:5060;rport=60487;received=218.82.26.6;branch=z9hG4bKSaoOR5zbirM6Xbvg
Max-Forwards: 69
User-Agent: PA168S
From: "8888" <sip:8888 at 62.164.130.1 >;tag=qy9DjT5ubwqB6Ttp
To: "008613381786981" <sip:008613381786981 at 62.164.130.1 >
Call-ID: 2GRBd0SARGev9jNA at 192.168.1.88
Contact: <sip:8888 at 218.82.26.6:60487>
CSeq: 1 INVITE
Supported: 100rel, replaces
Content-Type: application/sdp
Content-Length: 312
CC-Diversion:sip:008613381786981 at 62.164.130.1
v=0
o=8888 08882186 49218023 IN IP4 192.168.1.88
s=SIP CALL
c=IN IP4 192.168.1.88
t=0 0
m=audio 8000 RTP/AVP 4 18 0 8 3 101
a=rtpmap:4 G723/8000
a=rtpmap:18 G729/8000
a=rtpmap:0 PCMU/8000
a=rtpmap:8 PCMA/8000
a=rtpmap:3 GSM/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-15
a=direction:active
.
11(23018) orig. len=744, new_len=968, proto=1
11(23018) lookup(): '008613381786981' Not found in usrloc
11(23018) check_self - checking if host==us: 13==9 && [84.233.140.73] == [127.0.0.1]
11(23018) check_self - checking if port 5060 matches port 5060
11(23018) check_self - checking if host==us: 13==12 && [84.233.140.73] == [192.168.1.17]
11(23018) check_self - checking if port 5060 matches port 5060
11(23018) check_self - checking if host==us: 13==12 && [84.233.140.73] == [62.164.130.1]
11(23018) check_self - checking if port 5060 matches port 5060
11(23018) check_self: host != me
11(23018) parse_headers: flags=-1
11(23018) parse_headers: flags=-1
11(23018) DEBUG:check_content_type: type <application/sdp> found valid
11(23018) ERROR: extract_mediaip: no `c=' in SDP
11(23018) DEBUG: t_addifnew: msg id=26 , global msg id=24 , T on entrance=(nil)
11(23018) parse_headers: flags=-1
11(23018) parse_headers: flags=60
B. ser.cfg
# ----------- global configuration parameters
------------------------
debug=9 # debug level (cmd line:-d)
fork=yes
log_stderror=yes # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
fifo="/tmp/ser_fifo"
# ------------------ module loading
----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/exec.so"
loadmodule "/usr/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/rr.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
loadmodule "/usr/local/lib/ser/modules/domain.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_radius.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/xlog.so"
loadmodule "/usr/local/lib/ser/modules/uri.so"
loadmodule "/usr/local/lib/ser/modules/uri_radius.so"
# !! Nathelper
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
# ----------------- setting module-specific parameters
---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
#modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
#modparam("auth_db", "password_column", "password")
modparam("auth_radius","radius_config","/usr/local/etc/radiusclient/radiusclient.conf")
modparam("uri_radius","radius_config","/usr/local/etc/radiusclient/radiusclient.conf")
modparam("auth_radius","service_type",15)
modparam("acc","radius_config","/usr/local/etc/radiusclient/radiusclient.conf")
modparam("acc", "log_level", 1)
modparam("acc", "log_flag", 1)
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 2)
modparam("acc", "log_fmt", "miocfst")
modparam("acc", "failed_transactions" ,1)
modparam("acc", "radius_flag", 1)
modparam("acc", "service_type", 15)
modparam("acc", "radius_missed_flag", 3)
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# !! Nathelper
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
#xlog
#modparam("xlog", "buf_size", 8192)
#tm
modparam("tm", "fr_inv_timer", 400)
# ------------------------- request routing logic
-------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
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;
};
# !! Nathelper
# Special handling for NATed clients; first,NAT test is
# executed: it looks for via!=received and RFC1918 addresses
# in Contact (may fail if line-folding is used); also,
# the received test should, if completed,should check all
# vias for rpesence of received
if (nat_uac_test("3")) {
# Allow RR-ed requests, as these may indicate that
# a NAT-enabled proxy takes care of it; unless it is
# a REGISTER
if (method == "REGISTER" || !search("^Record-Route:")) {
log("LOG: Someone trying to register from private IP, rewriting\n");
# This will work only for user agents that support symmetric
# communication. We tested quite many of them and majority is
# smart enough to be symmetric. In some phones it takes a configuration
# option. With Cisco 7960, it is called NAT_Enable=Yes, with kphone it is
# called "symmetric media" and "symmetric signalling".
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
};
};
setflag(1);
setflag(2);
# 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();
# subsequent messages withing a dialog should take the
# path determined by record-routing
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;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
# Uncomment this if you want to use digest authentication
if (uri==myself) {
if (method=="REGISTER"){
if (!radius_www_authorize("")) {
www_challenge("", "0");
break;
};
if (!check_to()) {
log("LOG: To Cheating attempt\n");
sl_send_reply("403", "That is ugly -- use To=id in REGISTERs");
break;
};
save("location");
break;
};
if (method=="INVITE") {
log(1, "INVITE\n");
setflag(1); /* set for accounting(the same value as in log_flag!) */
};
if (method=="ACK") {
if (uri=~"sip:0[1-9][0-9]+ at .*") {
log(1, "ACK\n");
setflag(1); /* set for accounting(the same value as in log_flag!) */
};
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);
};
record_route();
if (uri=~"sip:00[1-9][0-9]+ at .*") {
rewritehostport("84.xx.xx.xxx:5060");
append_urihf("CC-Diversion:","\r\n");
forward(84.xx.xx.xxx, 5060);
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
# !! Nathelper
if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){
sl_send_reply("479", "We don't forward to private IP addresses");
break;
};
# if client or server know to be behind a NAT,enable relay
if (isflagset(6)) {
force_rtp_proxy();
};
# NAT processing of replies; apply to alltransactions (for example,
# re-INVITEs from public to private UA are hard to identify as
# NATed at the moment of request processing);look at replies
t_on_reply("1");
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
}
# !! Nathelper
onreply_route[1] {
# NATed transaction ?
if (isflagset(6) && status =~ "(183)|2[0-9][0-9]")
{
fix_nated_contact();
force_rtp_proxy();
# otherwise, is it a transaction behind a NAT and we did not
# know at time of request processing ? (RFC1918 contacts)
} else if (nat_uac_test("1")) {
fix_nated_contact();
};
}
C,Radius Detail File:
Sun Dec 19 19:52:19 2004
Acct-Status-Type = Failed
User-Service-Type = Sip-Session
Sip-Response-Code = 408
Sip-Method = Invite
User-Name = "491001 at 62.xx.xx.xxx"
Caller-ID = "sip:491001 at 62.xx.xx.xxx"
Client-Port-DNIS = "sip:003238777697 at 62.xx.xx.xxx"
Sip-Translated-Req-ID = "sip:003238777697 at 84.xx.xx.xxx:5060"
Acct-Session-Id = "FE2AF475-2066-475F-B960-D51E8FE7D051 at 212.202.103.93"
Sip-To-Tag = "n/a"
Sip-From-Tag = "1913233880"
Sip-Cseq = "15841"
Client-Id = 127.0.0.1
NAS-Port = 5060
Acct-Delay-Time = 0
Client-IP-Address = 127.0.0.1
Acct-Unique-Session-Id = "2e94ece290cdd8ce"
Timestamp = 1103457139
Sun Dec 19 19:59:42 2004
Acct-Status-Type = Stop
User-Service-Type = Sip-Session
Sip-Response-Code = 200
Sip-Method = 8
User-Name = "491001 at 62.xx.xx.xxx"
Caller-ID = "sip:491001 at 62.xx.xx.xxx"
Client-Port-DNIS = "sip:003238777697 at 62.xx.xx.xxx"
Sip-Translated-Req-ID = "sip:003238777697 at 84.xx.xx.xxx:5060"
Acct-Session-Id = "FE2AF475-2066-475F-B960-D51E8FE7D051 at 212.202.103.93"
Sip-To-Tag = "3312445751-681955"
Sip-From-Tag = "1913233880"
Sip-Cseq = "15842"
Client-Id = 127.0.0.1
NAS-Port = 5060
Acct-Delay-Time = 0
Client-IP-Address = 127.0.0.1
Acct-Unique-Session-Id = "2e94ece290cdd8ce"
Timestamp = 1103457582
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