I don’t seem to be able to get a match when I send a call from my gw to ser registered user. Debug indicates that user Not found in usrloc even though the database shows the users registered. I am able to make out bound calls the gw just fine. If anyone can shed some light in it it would be great.

Thanks

=============================================================================================================

mysql> select * from location;

+----------+--------+------------------------+----------+---------------------+-------+----------------------------------+------+----------------+-----------+-------+-------+----------------------------+

| 9901 | | sip:9901@69.250.205.31 | NULL | 2005-02-26 03:31:09 | -1.00 | e5201b3819008718@192.168.199.102 | 107 | 20050226033016 | 0 | 0 | 0 | Grandstream BT100 1.0.5.11 |

1 row in set (0.01 sec)

====debugt output===============================================================================================

20(18672) parse_headers: flags=256

20(18672) find_first_route: No Route headers found

20(18672) loose_route: There is no Route HF

20(18672) lookup(): '9901' Not found in usrloc

=============================================================================================================

U 69.19.88.16:5060 -> 69.19.88.19:5060

INVITE sip:9901@69.19.88.19 SIP/2.0..Via: SIP/2.0/UDP 69.19.88.16:5060;branch=z9hG4bK2fb13d17..From: "Cell Phone MD" <sip:4103532264

@69.19.88.16>;tag=as149fbe93..To: <sip:9901@69.19.88.19>..Contact: <sip:4103532264@69.19.88.16>..Call-ID: 745bcf587074e77d4a8118dc212

ccba7@69.19.88.16..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Date: Sat, 26 Feb 2005 08:41:01 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS

, BYE, REFER..Content-Type: application/sdp..Content-Length: 240....v=0..o=root 11581 11581 IN IP4 69.19.88.16..s=session..c=IN IP4 66.

159.88.16..t=0 0..m=audio 14934 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:1

01 0-16..a=silenceSupp:off - - - -..

U 69.19.88.19:5060 -> 69.19.88.16:5060

SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 69.19.88.16:5060;branch=z9hG4bK2fb13d17..From: "Cell Phone MD" <

sip:4103532264@69.19.88.16>;tag=as149fbe93..To: <sip:9901@69.19.88.19>..Call-ID: 745bcf587074e77d4a8118dc212ccba7@69.19.88.16..CSeq:

102 INVITE..Server: Sip EXpress router (0.9.0 (i386/linux))..Content-Length: 0..Warning: 392 69.19.88.19:5060 "Noisy feedback tells: p

id=18682 req_src_ip=69.19.88.16 req_src_port=5060 in_uri=sip:9901@69.19.88.19 out_uri=sip:9901@69.19.88.16:5060 via_cnt==1"....

U 69.19.88.19:5060 -> 69.19.88.16:5060

INVITE sip:9901@69.19.88.16:5060 SIP/2.0..Max-Forwards: 10..Record-Route: <sip:69.19.88.19;ftag=as149fbe93;lr=on>..Via: SIP/2.0/UDP 66

.159.88.19;branch=z9hG4bK96c9.80b8ddd7.0..Via: SIP/2.0/UDP 69.19.88.16:5060;branch=z9hG4bK2fb13d17..From: "Cell Phone MD" <sip:410353

2264@69.19.88.16>;tag=as149fbe93..To: <sip:9901@69.19.88.19>..Contact: <sip:4103532264@69.19.88.16>..Call-ID: 745bcf587074e77d4a8118d

c212ccba7@69.19.88.16..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Date: Sat, 26 Feb 2005 08:41:01 GMT..Allow: INVITE, ACK, CANCEL, OPT

IONS, BYE, REFER..Content-Type: application/sdp..Content-Length: 240..P-hint: GATEWAY....v=0..o=root 11581 11581 IN IP4 69.19.88.16..s=

session..c=IN IP4 69.19.88.16..t=0 0..m=audio 14934 RTP/AVP 0 8 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:101 telephone

-event/8000..a=fmtp:101 0-16..a=silenceSupp:off - - - -..

U 69.19.88.16:5060 -> 69.19.88.19:5060

SIP/2.0 482 Loop Detected..Via: SIP/2.0/UDP 69.19.88.19;branch=z9hG4bK96c9.80b8ddd7.0..Via: SIP/2.0/UDP 69.19.88.16:5060;branch=z9hG4b

K2fb13d17..From: "Cell Phone MD" <sip:4103532264@69.19.88.16>;tag=as149fbe93..To: <sip:9901@69.19.88.19>;tag=as149fbe93..Call-ID: 74

5bcf587074e77d4a8118dc212ccba7@69.19.88.16..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER

..Contact: <sip:4103532264@69.19.88.16>..Content-Length: 0....

==ser.cfg===========================================================================================================

# $Id: serconf.sh,v 1.3 2003/11/03 13:03:58 janakj Exp $

# autogenerated SER configuration

# user: uid=0(root) gid=0(root) groups=0(root),1(bin),2(daemon),3(sys),4(adm),6(disk),10(wheel)

# system: Linux baltodb.popdial.com 2.6.10-1.760_FC3 #1 Wed Feb 2 00:14:23 EST 2005 i686 i686 i386 GNU/Linux

# date: Sat Feb 26 02:57:50 EST 2005

# ----------- global configuration parameters ------------------------

debug=4

fork=yes

port=5060

log_stderror=no

memlog=5

mhomed=yes

fifo="/tmp/ser_fifo"

alias=baltodb.popdial.com

# uncomment to override config values for test

# /*

debug=4 # debug level (cmd line: -ddd)

fork=yes

port=5060

log_stderror=yes # (cmd line: -E)

fifo="/tmp/ser_fifox"

# */

check_via=no # (cmd. line: -v)

dns=no # (cmd. line: -r)

rev_dns=no # (cmd. line: -R)

children=16

# if changing fifo mode to a more restrictive value, put

# decimal value in there, e.g. dec(rw|rw|rw)=dec(666)=438

#fifo_mode=438

# ------------------ module loading ----------------------------------

loadmodule "/usr/local/lib/ser/modules/tm.so"

loadmodule "/usr/local/lib/ser/modules/sl.so"

loadmodule "/usr/local/lib/ser/modules/acc.so"

loadmodule "/usr/local/lib/ser/modules/rr.so"

loadmodule "/usr/local/lib/ser/modules/maxfwd.so"

loadmodule "/usr/local/lib/ser/modules/mysql.so"

loadmodule "/usr/local/lib/ser/modules/usrloc.so"

loadmodule "/usr/local/lib/ser/modules/registrar.so"

loadmodule "/usr/local/lib/ser/modules/auth.so"

loadmodule "/usr/local/lib/ser/modules/auth_db.so"

loadmodule "/usr/local/lib/ser/modules/textops.so"

loadmodule "/usr/local/lib/ser/modules/uri.so"

loadmodule "/usr/local/lib/ser/modules/group.so"

loadmodule "/usr/local/lib/ser/modules/msilo.so"

loadmodule "/usr/local/lib/ser/modules/enum.so"

loadmodule "/usr/local/lib/ser/modules/uri_db.so"

# ----------------- setting module-specific parameters ---------------

# all DB urls here

# -- usrloc params --

/* 0 -- dont use mysql, 1 -- write_through, 2--write_back */

modparam("usrloc", "db_mode", 2)

modparam("usrloc", "timer_interval", 10)

# -- auth params --

modparam("auth_db", "calculate_ha1", yes)

#modparam("auth_db", "user_column", "user_id")

modparam("auth_db", "password_column", "password")

modparam("auth", "nonce_expire", 300)

# -- rr params --

# add value to ;lr param to make some broken UAs happy

modparam("rr", "enable_full_lr", 1)

# -- acc params --

# that is the flag for which we will account -- don't forget to

modparam("acc", "db_flag", 1 )

modparam("acc", "db_missed_flag", 3 )

# -- tm params --

modparam("tm", "fr_timer", 20 )

modparam("tm", "fr_inv_timer", 90 )

modparam("tm", "wt_timer", 20 )

# -- msilo params

modparam("msilo", "registrar", "sip:registrar@baltodb.popdial.com")

# -- enum params --

modparam("enum", "domain_suffix", "e164.arpa.")

# ------------------------- request routing logic -------------------

# main routing logic

route{

/* ********* ROUTINE CHECKS ********************************** */

# filter too old messages

if (!mf_process_maxfwd_header("10")) {

log("LOG: Too many hops\n");

sl_send_reply("483","Alas Too Many Hops");

break;

};

#if (len_gt( max_len )) {

if ( msg:len > max_len ) {

sl_send_reply("513", "Message too large sorry");

break;

};

# Make sure that requests dont advertise addresses

# from private IP space (RFC1918) in Contact HF

# (note: does not match with folded lines)

if (search("^(Contact|m): .*@(192\.168\.|10\.|172\.16)")) {

# 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:"))

&& !( src_ip==192.168.0.0/16 ||

src_ip==10.0.0.0/8 || src_ip==172.16.0.0/12 )) {

log("LOG: Someone trying to register from private IP again\n");

sl_send_reply("479", "We dont accept private IP contacts" );

break;

};

# anti-spam -- if somene claims to belong to our domain in From,

# challenge him (skip REGISTERs -- we will chalenge them later)

if (search("(From|F):.*(baltodb\.popdial\.com|66\.159\.88\.19)")) {

# invites forwarded to other domains, like FWD may cause subsequent

# request to come from there but have iptel in From -> verify

# only INVITEs (ignore FIFO/UAC's requests, i.e. src_ip==myself)

if (method=="INVITE" & !(src_ip==69.19.88.19)) {

if (!(proxy_authorize( "baltodb.popdial.com" /* realm */,

"subscriber" /* table name */ ))) {

proxy_challenge("baltodb.popdial.com" /* realm */, "0" /* no-qop */);

break;

};

# to maintain outside credibility of our proxy, we enforce

# username in From to equal digest username; user with

# "john.doe" id could advertise "bill.gates" in From otherwise;

if (!check_from()) {

log("LOG: From Cheating attempt in INVITE\n");

sl_send_reply("403", "That is ugly -- use From=id next time (OB)");

break;

};

# we better don't consume credentials -- some requests may be

# spiraled through our server (sfo@iptel->7141@iptel) and the

# subsequent iteration may challenge too, for example because of

# iptel claim in From; UACs then give up because they

# already submitted credentials for the given realm

#consume_credentials();

}; # INVITEs claiming to come from our domain

} else if (method=="INVITE" && !(uri=~"[@:\.](baltodb\.popdial\.com|66\.159\.88\.19)([;:].*)*"

# ... and we serve our gateway too if present

| uri=~"@66\.159\.88\.16([;:].*)*" )) {

#the INVITE neither claims to come from our domain nor is it targeted to it

# -> junk it

sl_send_reply("403", "No relaying");

break;

};

/* ********* RR ********************************** */

# to be safe, record route everything; UAs may use different

# transport protocols and need to have SER in path

record_route();

# if route forces us to forward to some explicit destination,

# do so; check however first that a cheater didn't preload

# a gateway destination to bypass PSTN ACLs

if (loose_route()) {

if (uri=~"sip:[+0-9]+@66\.159\.88\.16") {

# it is gateway -- proceed to ACLs

route(3);

break;

};

# route HF determined next hop; forward there

append_hf("P-hint: rr-enforced\r\n");

t_relay();

break;

};

/* ********* check for requests targeted out of our domain... ******* */

# sign of our domain: there is '@' (username) or : (nothing) in

# front of our domain name ; ('.' is not there -- we handle all

# xxx.iptel.org as outbound hosts);if none of these cases matches,

# proceed with processing of outbound requests in route[2]

if (!(uri=~"[@:](sip[\.)?(baltodb\.popdial\.com|66\.159\.88\.19)([;:].*)*" | uri=~"@66\.159\.88\.16([;:].*)*")) {

route(2);

break;

};

/* ************ requests for our domain ********** */

/* now, the request is for sure for our domain */

# registers always MUST be authenticated to

# avoid stealing incoming calls

if (method=="REGISTER") {

# Make sure that user's dont register infinite loops

# (note: does not match with folded lines)

if (search("^(Contact|m): .*@(baltodb\.popdial\.com|66\.159\.88\.19)")) {

log(1, "LOG: alert: someone trying to set aor==contact\n");

sl_send_reply("476", "No Server Address in Contacts Allowed" );

break;

};

if (search("^(Contact|m): .*66\.159\.88\.16")) {

log(1, "LOG: alert: protected contacts\n");

sl_send_reply("476", "No Server Address in Contacts Allowed" );

break;

};

if (!www_authorize( "baltodb.popdial.com" /* realm */,

"subscriber" /* table name */ )) {

# challenge if none or invalid credentials

www_challenge( "baltodb.popdial.com" /* realm */,

"0" /* no qop -- some phones can't deal with it */);

break;

};

# prohibit attempts to grab someone else's To address

# using valid credentials;

if (!check_to()) {

log("LOG: To Cheating attempt\n");

sl_send_reply("403", "That is ugly -- use To=id in REGISTERs");

break;

};

# it is an authenticated request, update Contact database now

if (!save("location")) {

sl_reply_error();

};

m_dump();

break;

};

# some UACs might be fooled by Contacts our UACs generate to make MSN

# happy (web-im, e.g.) -- tell its urneachable

if (uri=~"sip:daemon@" ) {

sl_send_reply("410", "daemon is gone");

break;

};

# is this an ENUM destination (leading +?)? give it a try, if the lookup

# doesn't change URI, just continue

if (uri=~"sip:\+[0-9]+@") {

if (!enum_query("voice")) { # if parameter empty, it defaults to "e2u+sip"

enum_query(""); # E2U+sip

};

} else {

# aliases (take precedences over PSTN number; provisioning interface

# is set up to assinge aliases beginning with 8)

lookup("aliases");

};

# check again, if it is still for our domain after aliases are resolved

if (!(uri=~"[@:](sip[\.)?(baltodb\.popdial\.com|66\.159\.88\.19)([;:].*)*" | uri=~"@66\.159\.88\.16([;:].*)*")) {

route(5);

break;

};

# now check if it's about PSTN destinations through our gateway;

# note that 8.... is exempted for numerical non-gw destinations

if (uri=~"sip:\+?[0-79][0-9]*@.*") {

route(3);

break;

};

# native SIP destinations are handled using our USRLOC DB

if (!lookup("location")) {

# handle user which was not found ...

route(4);

break;

};

# check whether some inventive user has uploaded gateway

# contacts to UsrLoc to bypass our authorization logic

if (uri=~"sip:[+0-9]+@66\.159\.88\.16") {

# it is gateway -- proceed to ACLs

route(3);

break;

};

/* ... and also report on missed calls ... */

setflag(3);

# we now know we may, we know where, let it go out now!

append_hf("P-hint: USRLOC\r\n");

if (!t_relay()) {

sl_reply_error();

break;

};

}

#------------------- OUTBOUND ----------------------------------------

# routing logic for outbound requests targeted out of our domain

# (keep in mind messages to our users can end up here too: for example,

# an INVITE may be UsrLoc-ed, then the other party uses outbound

# proxy with r-uri=the usr_loced addredd (typically IP))

route[2] {

append_hf("P-hint: OUTBOUND\r\n");

t_relay();

}

#------- ALIASED OUTBOUND --------------------------------------------

# routing logic for inbound requests aliased outbound; unlike

# with real outbound requests we do not force authentication

# as these calls are server by our server and we do not want

# to disqualify unathenticated request originatiors from other

# domains

route[5] {

append_hf("P-hint: ALIASED-OUTBOUND\r\n");

t_relay();

}

#----------------- PSTN ----------------------------------------------

# logic for calls to the PSTN

route[3] {

# turn accounting on

setflag(1);

/* require all who call PSTN to be members of the "int" group;

apply ACLs only to INVITEs -- we don't need to protect other requests, as they

don't imply charges; also it could cause troubles when a call comes in via PSTN

and goes to a party that can't authenticate (voicemail, other domain) -- BYEs would

fail then; exempt Cisco gateway from authentication by IP address -- it does not

support digest

if (method=="INVITE" && (!src_ip==69.19.88.16)) {

if (!proxy_authorize( "baltodb.popdial.com" /* realm */,

"subscriber" /* table name */)) {

proxy_challenge( "baltodb.popdial.com" /* realm */, "0" /* no qop */ );

break;

};

# let's check from=id ... avoids accounting confusion

if (method=="INVITE" & !check_from()) {

log("LOG: From Cheating attempt\n");

sl_send_reply("403", "That is ugly -- use From=id next time (gw)");

break;

};

if(!is_user_in("credentials", "int")) {

sl_send_reply("403", "NO PSTN Privileges...");

break;

};

consume_credentials();

}; # INVITE to authorized PSTN

# if you have passed through all the checks, let your call go to GW!

rewritehostport("69.19.88.16:5060");

# snom conditioner

if (method=="INVITE" && search("User-Agent: snom")) {

replace("100rel, ", "");

};

append_hf("P-hint: GATEWAY\r\n");

# use UDP to guarantee well-known sender port (TCP ephemeral)

t_relay_to_udp("69.19.88.16","5060");

}

/* *********** handling of unavailable user ******************* */

route[4] {

/**/

# message store

if (method=="MESSAGE") {

t_newtran();

if (m_store("0")) {

t_reply("202", "Accepted for Later Delivery");

} else {

t_reply("503", "Service Unavailable");

};

break;

};

/**/

# non-Voip -- just send "off-line"

if (!(method=="INVITE" || method=="ACK" || method=="CANCEL")) {

sl_send_reply("404", "Not Found");

break;

};

# voicemail subscribers ...

t_newtran();

t_reply("404", "Not Found");

# we account missed incoming calls; previous statteful processing

# guarantees that retransmissions are not accounted

if (method=="INVITE") {

acc_db_request("404 missed call", "missed_calls");

};

}