27 Jan
2005
27 Jan
'05
4:05 p.m.
Does anyone have a working example of a config file for ser 0.9
that they can share. My 0.8.14 config keeps failing. The line
numbers in the debug output indicate the problem is on a line
that is blank. The NEWS file is no help.
Thanks
--
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104
voice: 215-573-8396
215-746-8001
fax: 215-898-9348
sip:blairs@upenn.edu
27 Jan
27 Jan
5:06 p.m.
Steve,
Here is my ser.cfg. If you find any potential problems or errors with it then please let
me know.
The general setup here is ser-0.9 is the sip router, asterisk 1.0.1 is the voicemail
server and we
use a third party for PSTN access.
Click-2-Dial is not handled with FIFO commands, but rather with sipp from sourceforge
because this
lets us seperate the web console from the sip proxy. We do not use serweb, so we have our
own web
frontend replacement for serweb.
This ser.cfg makes use of the avpops module for call forwarding features and some voice
mail
stuff. The call forwarding stuff got rather complex because we had to make sure the number
a user
keyed in to their call forwarding settings was "valid". For example, if they did
not have
international dialing permissions then they wouldn't be allowed to have an
international number in
their call forwarding settings. The other complex part for the call forwarding section is
the fact
that you have to know in the reply_route which call forwarding number to use under
different
conditions, (ie forward busy versus forward no answer).
The final note is that we have our own custom ser module called xdz_tools which exposes
the
is_caller_blocked() and is_did_unassigned() functions.
The is_did_unassigned() function is needed so we can determine if a DID is assigned to our
PRI but
not allocated to a subscriber. This prevents infinite loops between our SIP proxy and the
PSTN
gateway.
The is_caller_blocked() function is used to determine if the callee has a line block on
the
caller.
Hope this helps.
Regards,
Paul
# IP Legend
# 99.99.99.100 - Asterisk Voicemail Server
# 99.99.99.101 - Domestic US PSTN Gateway
# 99.99.99.102 - International PSTN Gateway
# ------------- Initial global variables
debug=3
fork=yes
log_stderror=no
listen=99.99.99.99
port=5060
children=4
server_signature=yes
# hostname matching an alias will satisfy the condition uri==myself"
alias=mycompany1.com
alias=mycompany2.com
# dns - Uses dns to check if it is necessary to add a "received=" field
# to a via. Default is no.
# rev_dns - Same as dns but use reverse DNS.
dns=no
rev_dns=no
# check_via - Turn on or off Via host checking when forwarding replies.
# Default is no. arcane. looks for discrepancy between name and
# ip address when forwarding replies.
check_via=no
# syn_branch - Shall the server use stateful synonym branches? It is
# faster but not reboot-safe. Default is yes.
syn_branch=yes
# memlog - Debugging level for final memory statistics report. Default
# is L_DBG -- memory statistics are dumped only if debug is set high.
memlog=1000
# sip_warning - Should replies include extensive warnings?
# By default yes, it is good for trouble-shooting.
sip_warning=no
# fifo - FIFO special file pathname
fifo_mode=0666
fifo="/tmp/ser_fifo"
fifo_db_url="mysql://ser:number4@localhost/ser"
# reply_to_via - A hint to reply modules whether they should send reply
# to IP advertised in Via. Turned off by default, which means that
# replies are sent to IP address from which requests came.
reply_to_via=no
# user | uid - uid to be used by the server. 99 = nobody.
#uid="nobody"
# group | gid - gid to be used by the server. 99 = nobody.
#gid="nobody"
# mhomed -- enable calculation of outbound interface.
# useful on multihomed servers.
mhomed=0
# ------------- external module loading
loadmodule "/usr/local/lib/ser/modules/mysql.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/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/exec.so"
loadmodule "/usr/local/lib/ser/modules/group.so"
loadmodule "/usr/local/lib/ser/modules/print.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
loadmodule "/usr/local/lib/ser/modules/mediaproxy.so"
loadmodule "/usr/local/lib/ser/modules/uri.so"
loadmodule "/usr/local/lib/ser/modules/uri_db.so"
loadmodule "/usr/local/lib/ser/modules/domain.so"
loadmodule "/usr/local/lib/ser/modules/xlog.so"
loadmodule "/usr/local/lib/ser/modules/speeddial.so"
loadmodule "/usr/local/lib/ser/modules/options.so"
loadmodule "/usr/local/lib/ser/modules/avpops.so"
loadmodule "/usr/local/lib/ser/modules/permissions.so"
loadmodule "/usr/local/lib/ser/modules/xdz_tools.so"
# ------------ media proxy settings
modparam("mediaproxy", "natping_interval", 30)
modparam("mediaproxy", "sip_asymmetrics",
"/usr/local/etc/ser/sip-asymmetric-clients")
modparam("mediaproxy", "rtp_asymmetrics",
"/usr/local/etc/ser/rtp-asymmetric-clients")
# ------------- db_url setting
modparam("acc|auth_db|domain|group|permissions|speeddial|uri_db|usrloc|xdz_tools",
"db_url",
"mysql://ser:heslo@localhost/ser")
# ------------- use_domain setting
modparam("auth_db|group|registrar|speeddial|uri_db|usrloc",
"use_domain", 0)
# ------------- permissions settings
modparam("permissions", "db_mode", 1)
modparam("permissions", "trusted_table", "trusted")
# ------------- accounting parameters
modparam("acc", "db_flag", 1)
modparam("acc", "db_missed_flag", 1)
modparam("acc", "log_fmt", "cdfimorstup")
modparam("acc", "log_level", 1)
modparam("acc", "failed_transactions", 1)
modparam("acc", "report_cancels", 1)
modparam("acc", "report_ack", 0)
# ------------- auth parameters
# allows clear text passwords in the mysql database
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "avps_column_int", "domn")
modparam("auth_db", "avps_column_str", "uuid|rpid")
modparam("auth", "rpid_prefix", "<sip:")
modparam("auth", "rpid_suffix",
";user=phone>;party=calling;screen=yes;id-type=subscriber;privacy=full")
# ------------- domain parameters
modparam("domain", "db_mode", 1)
# ------------- exec parameters
modparam("exec", "setvars", 1)
modparam("exec", "time_to_kill", 10)
# ------------- registration parameters
modparam("registrar", "nat_flag", 2)
modparam("registrar", "min_expires", 60)
modparam("registrar", "max_expires", 86400)
modparam("registrar", "default_expires", 3600)
modparam("registrar", "desc_time_order", 1)
modparam("registrar", "append_branches", 1)
# ------------- rr parameters
# set ";lr" tag to lr=true
modparam("rr", "enable_full_lr", 1)
# ------------- tm parameters
modparam("tm", "fr_timer", 15)
modparam("tm", "fr_inv_timer", 22)
modparam("tm", "wt_timer", 5)
modparam("tm", "fr_inv_timer_avp", "inv_timeout")
# ------------- usrloc parameters
# 0 = disable
# 1 = write-through
# 2 = write-back
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "timer_interval", 60)
modparam("usrloc", "desc_time_order", 1)
#-------------- AVPOPS parameters
modparam("avpops", "avp_url",
"mysql://ser:heslo@localhost/ser")
modparam("avpops", "avp_table", "usr_preferences")
modparam("avpops", "avp_aliases",
"voicemail=i:500;calltype=i:700;fwd_no_answer_type=i:701;fwd_busy_type=i:702")
# ------------- logging parameters
modparam("xlog", "buf_size", 8192)
route {
# ------------------------------------------------------------------------
# Sanity Check Section
# ------------------------------------------------------------------------
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483", "Too Many Hops");
break;
};
if (msg:len > max_len) {
sl_send_reply("513", "Message Overflow");
break;
};
# ------------------------------------------------------------------------
# NOTIFY Keep-Alive Section
# ------------------------------------------------------------------------
if ((method=="NOTIFY") && search("^Event: keep-alive")) {
sl_send_reply("200", "OK");
break;
};
if ((method=="NOTIFY") && (uri=~"^sip:700@")) {
sl_send_reply("200", "OK");
break;
};
# ------------------------------------------------------------------------
# NOTIFY Message-Summary Section
#
# Event: Message-Summary is used to control the message waiting indicator
# ------------------------------------------------------------------------
if ((method=="NOTIFY") && (src_ip=="99.99.99.100") &&
search("^Event: message-summary")) {
if (lookup("location")) {
t_relay();
};
break;
};
# ------------------------------------------------------------------------
# OPTIONS Section
# ------------------------------------------------------------------------
if (method=="OPTIONS") {
options_reply();
break;
};
# ------------------------------------------------------------------------
# NAT Test Section #1
# ------------------------------------------------------------------------
if (method=="REGISTER" && client_nat_test("3")) {
fix_contact();
force_rport();
setflag(2);
};
# ------------------------------------------------------------------------
# Click2Dial Section
# ------------------------------------------------------------------------
if (src_ip=="99.99.99.99" && (search("^From: <sip:700@")
|| search("^To: <sip:700@"))) {
setflag(14);
};
# ------------------------------------------------------------------------
# Registration Section
# ------------------------------------------------------------------------
if (method=="REGISTER") {
# allow all requests from user 700 - the Click2Dial controller
if (!isflagset(14)) {
if (!is_from_local()) {
sl_send_reply("403", "Unknown Domain");
break;
};
if (!www_authorize("", "subscriber")) {
www_challenge("", "0");
break;
};
if (!check_to()) {
sl_send_reply("401", "Unauthorized");
break;
};
if (is_user_in("To", "demo-disabled")) {
sl_send_reply("403", "Your evaluation period has expired");
break;
};
if (is_user_in("To", "disabled")) {
sl_send_reply("403", "Your account has been disabled");
break;
};
};
# snom sip phones use this header to start their
# keep-alive mechanism for NAT bindings
append_to_reply("P-NAT-Refresh: 15\r\n");
if (!save("location")) {
sl_reply_error();
};
break;
};
# ------------------------------------------------------------------------
# Accounting Section
#
# NOTE: We test for flag 14 because we do not want to record Click2Dial
# entries
# ------------------------------------------------------------------------
if ((method=="INVITE" || method=="BYE") && !isflagset(14)) {
setflag(1);
};
# ------------------------------------------------------------------------
# NAT Tear-Down Section
# ------------------------------------------------------------------------
if ((method == "BYE" || method == "CANCEL")) {
end_media_session();
};
# ------------------------------------------------------------------------
# Record Route Section
#
# 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();
};
# ------------------------------------------------------------------------
# Loose Route Section
#
# Grant route routing if route headers present
# ------------------------------------------------------------------------
if (loose_route()) {
route(2);
break;
};
# ------------------------------------------------------------------------
# Message Waiting Indicator SUBSCRIBE Section
#
# Make sure all SUBSCRIBE messages with header "Event: message-summary"
# are sent to the voicemail server
# ------------------------------------------------------------------------
if ((method=="SUBSCRIBE") && search("^Event:
message-summary")) {
rewritehostport("99.99.99.100:5060");
t_relay();
break;
};
# ------------------------------------------------------------------------
# NAT Test Section #1
# ------------------------------------------------------------------------
if (client_nat_test("3") && !search("^Record-Route:")) {
force_rport();
fix_contact();
};
# ------------------------------------------------------------------------
# Enhanced Voicemail Applications Section
#
# Access additional Asterisk applications at extension 701 and 702
# ------------------------------------------------------------------------
if (uri=~"^sip:70[1|2]@") {
route(5);
break;
};
# ------------------------------------------------------------------------
# PSTN Section
#
# ------------------------------------------------------------------------
if (method=="INVITE") {
if (uri=~"^sip:1[0-9]{10}@") {
strip(1);
};
if ((uri=~"^sip:911@.*") || (uri=~"^sip:900[0-9]{7}@") ||
(uri=~"^sip:976[0-9]{7}@")) {
sl_send_reply("503", "Service Unavailable");
break;
};
};
# ------------------------------------------------------------------------
# Alias Routing Section
# ------------------------------------------------------------------------
lookup("aliases");
if (!uri==myself) {
route(2);
break;
};
# ------------------------------------------------------------------------
# Load ACL Section
# ------------------------------------------------------------------------
if (method=="INVITE" && !isflagset(14)) {
if (is_user_in("Request-URI", "voicemail")) {
setflag(31);
};
if (is_from_local() || is_uri_host_local()) {
if (is_user_in("From", "411")) {
setflag(30);
};
if (is_user_in("From", "int")) {
setflag(29);
};
if (is_user_in("From", "free-pstn")) {
setflag(28);
};
if (avp_db_load("$from/username", "s:callidblock")) {
if (avp_check("s:callidblock", "eq/y/i")) {
setflag(25);
};
};
};
if (avp_db_load("$ruri/username", "s:anoncallrej")) {
if (avp_check("s:anoncallrej", "eq/y/i")) {
setflag(24);
};
};
};
# ------------------------------------------------------------------------
# Anonymous Call Rejection Section
# ------------------------------------------------------------------------
if (isflagset(24) && (method=="INVITE") &&
search("^(f|F)rom:.*(a|A)nonymous")) {
route(8);
break;
};
# ------------------------------------------------------------------------
# 411 Directory Assistance Section
# ------------------------------------------------------------------------
if (method=="INVITE" && uri=~"^sip:411@.*") {
if (isflagset(30)) {
avp_write("$from", "i:34");
avp_pushto("$ruri", "i:34");
strip_tail(7);
subst_uri('/^sip:([0-9]+)@(.*)$/sip:\15551212@\2/i');
avp_delete("i:34");
route(3);
break;
};
sl_send_reply("503", "Service Unavailable");
break;
};
# ------------------------------------------------------------------------
# Speed Dialing Section
# ------------------------------------------------------------------------
if ((method=="INVITE") && (uri=~"^sip:[0-9]{2}@.*"))
{
sd_lookup("speed_dial");
};
# ------------------------------------------------------------------------
# 011 International Call Section
# ------------------------------------------------------------------------
if (method=="INVITE" && uri=~"^sip:011[0-9]*@") {
if (isflagset(29)) {
route(6);
} else {
sl_send_reply("503", "Service Unavailable");
};
break;
};
# ------------------------------------------------------------------------
# URI Compare Section
#
# Here we compare the "from" and "to" to see if the caller is
dialing
# their own extension. If so then we route to voicemail if needed
# ------------------------------------------------------------------------
if (method=="INVITE") {
avp_write("$from", "i:34");
if (avp_check("i:34", "eq/$ruri/i")) {
if (isflagset(31)) {
route(5);
break;
} else {
sl_send_reply("486", "Busy");
break;
};
};
};
# ------------------------------------------------------------------------
# Call Block Section
# ------------------------------------------------------------------------
if (is_caller_blocked()) {
route(7);
break;
};
# ------------------------------------------------------------------------
# Do Not Disturb Section
# ------------------------------------------------------------------------
if (avp_db_load("$ruri/username", "s:donotdisturb")) {
if (avp_check("s:donotdisturb", "eq/y/i")) {
route(5);
break;
};
};
# ------------------------------------------------------------------------
# Blind Call Forwarding Section
# ------------------------------------------------------------------------
if (method=="INVITE") {
# here we must store the current (aka original) R-URI because if
# we set call forwarding and the forwarded number is busy then we
# need to use this original R-URI to determine which voicemail
# box we should go to
if (isflagset(31)) {
avp_write("$ruri", "$voicemail");
};
if (avp_db_load("$ruri/username", "s:callfwd")) {
avp_pushto("$ruri", "s:callfwd");
# lookup the call fowarding number to see if it is a served
# sip number or a PSTN number
# check forwarding number rules
route(1);
if (avp_check("$calltype", "eq/-/i")) {
sl_send_reply("503", "Service Unavailable");
break;
};
# test for domestic PSTN gateway
if (avp_check("$calltype", "eq/dom/i")) {
route(3);
break;
};
# test for international PSTN gateway
if (avp_check("$calltype", "eq/int/i")) {
route(6);
break;
};
};
};
# ------------------------------------------------------------------------
# Call Routing Section
# ------------------------------------------------------------------------
if (!lookup("location")) {
# if flag 31 (ie voicemail) is set and we made it here this means
# the user's phone is not registered anywhere. We'll forward to
# voicemail after this block because we need to check the call
# forward settings first
if (isflagset(31)) {
# flag 19 means the user has voicemail but is not online
# so we need to remember to send to voicemail if call
# forwarding is not enabled
setflag(19);
};
if (method=="INVITE") {
if (does_uri_exist()) {
# subscriber record found, but they're offline
sl_send_reply("480", "Temporarily Unavailable");
break;
};
if (is_did_unassigned()) {
# User Does Not Exist, but the DID is ours
sl_send_reply("404", "User Not Found");
break;
};
if (uri=~"^sip:[0-9]{10}@") {
# Send to PSTN Gateway
if (isflagset(28)) {
route(3);
} else {
sl_send_reply("503", "Service Unavailable");
};
break;
};
sl_send_reply("404", "User Not Found");
break;
};
};
# ------------------------------------------------------------------------
# Call Forwarding Section
# ------------------------------------------------------------------------
if (method=="INVITE") {
# save R-URI in a temp AVP for later use
avp_write("$ruri", "i:99");
# only load the forward no answer option if voice mail is not enabled
if (!isflagset(31)) {
if (avp_db_load("$ruri/username", "s:fwdnoanswer")) {
route(1);
if (!avp_check("$calltype", "eq/-/i")) {
if (avp_check("$calltype", "eq/dom/i")) {
avp_write("dom", "$fwd_no_answer_type");
} else if (avp_check("$calltype", "eq/int/i")) {
avp_write("int", "$fwd_no_answer_type");
} else {
avp_write("sip", "$fwd_no_answer_type");
}
setflag(27);
};
};
};
if (avp_db_load("$ruri/username", "s:fwdbusy")) {
route(1);
if (!avp_check("$calltype", "eq/-/i")) {
if (avp_check("$calltype", "eq/dom/i")) {
avp_write("dom", "$fwd_busy_type");
} else if (avp_check("$calltype", "eq/int/i")) {
avp_write("int", "$fwd_busy_type");
} else {
avp_write("sip", "$fwd_busy_type");
}
setflag(26);
};
};
avp_pushto("$ruri", "i:99");
};
if (isflagset(19)) {
# send to voicemail
route(5);
} else {
route(2);
};
}
route[1] {
# Here we have route checks for all the call forwarding stuff. The return
# values are passed as AVP $calltype as follows:
#
# "-" = R-URI is not allowed
# "dom" = R-RURI is a domestic call
# "int" = R-RURI is an international call
# "sip" = R-RURI is a sip call
avp_write("-", "$calltype");
if (uri=~"^sip:1[0-9]{10}@") {
strip(1);
};
if ((uri=~"^sip:70[1|2]@") || (uri=~"^sip:411@.*") ||
(uri=~"^sip:911@.*") || (uri=~"^sip:900[0-9]{7}@") ||
(uri=~"^sip:976[0-9]{7}@")) {
break;
};
lookup("aliases");
if (!lookup("location")) {
if (uri=~"^sip:[0-9]{10}@") {
# test for domestic PSTN number
if (isflagset(28)) {
avp_write("dom", "$calltype");
};
} else if (uri=~"^sip:011[0-9]*@") {
# test for international PSTN number
if (isflagset(29)) {
avp_write("int", "$calltype");
};
};
break;
};
avp_write("sip", "$calltype");
}
route[2] {
if ((method=="INVITE") && !allow_trusted()) {
if (!proxy_authorize("", "subscriber")) {
proxy_challenge("", "0");
break;
} else if (!check_from()) {
log(1, "Spoofed SIP call attempt");
sl_send_reply("403", "Use From=ID");
break;
} else if (!(is_from_local() || is_uri_host_local())) {
sl_send_reply("403", "Please register to use our service");
break;
};
};
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 (isflagset(25)) {
replace("^From:(.*)>" , "From: \"Anonymous\"
<sip:someone@anonymous.invalid>");
};
if (method=="INVITE" || method=="ACK") {
use_media_proxy();
};
t_on_failure("1");
t_on_reply("1");
if (!t_relay()) {
if (method=="INVITE" || method=="ACK") {
end_media_session();
};
sl_reply_error();
};
}
route[3] {
if (method=="INVITE") {
if (!proxy_authorize("", "subscriber")) {
proxy_challenge("", "0");
break;
} else if (!check_from()) {
log(1, "Spoofed SIP call attempt");
sl_send_reply("403", "Use From=ID");
break;
} else if (!(is_from_local() || is_uri_host_local())) {
sl_send_reply("403", "Please register to use our service");
break;
};
# enable caller id blocking for PSTN calls
if (isflagset(25)) {
append_rpid_hf();
};
};
# SIP->PSTN calls get 45 seconds to timeout
avp_write("i:45", "inv_timeout");
rewritehost("99.99.99.101");
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 (method=="INVITE" || method=="ACK") {
use_media_proxy();
};
if (isflagset(31)) {
t_on_failure("1");
};
t_on_reply("1");
if (!t_relay()) {
if (method=="INVITE" || method=="ACK") {
end_media_session();
};
sl_reply_error();
};
}
route[4] {
# voicemail route #1
#
# this path this executed during these conditions:
#
# cond 1) the called number is in the location table
# but the callee did not answer the phone
# (ie, failover to voicemail)
if (isflagset(25)) {
replace("^From:(.*)>" , "From: \"Anonymous\"
<sip:someone@anonymous.invalid>");
};
rewritehostport("99.99.99.100:5060");
append_branch();
t_on_reply("1");
if (!t_relay()) {
if (method=="INVITE" || method=="ACK") {
end_media_session();
};
sl_reply_error();
};
}
route[5] {
# voicemail route #2
#
# this path this executed during these conditions:
#
# cond 1) the called number is not in the location table
# cond 2) the from_uri == to_uri (ie, caller==callee)
if (method=="INVITE" || method=="ACK") {
use_media_proxy();
};
rewritehostport("99.99.99.100:5060");
t_on_reply("1");
if (!t_relay()) {
if (method=="INVITE" || method=="ACK") {
end_media_session();
};
sl_reply_error();
};
}
route[6] {
# All International Calls Go To BroadVox
if (method=="INVITE") {
if (!proxy_authorize("", "subscriber")) {
proxy_challenge("", "0");
break;
} else if (!check_from()) {
log(1, "Spoofed SIP call attempt");
sl_send_reply("403", "Use From=ID");
break;
} else if (!(is_from_local() || is_uri_host_local())) {
sl_send_reply("403", "Please register to use our service");
break;
};
# enable caller id blocking for PSTN calls
if (isflagset(25)) {
append_rpid_hf();
};
};
# SIP->PSTN calls get 45 seconds to timeout
avp_write("i:45", "inv_timeout");
rewritehost("99.99.99.102");
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 (method=="INVITE" || method=="ACK") {
use_media_proxy();
};
t_on_reply("1");
if (!t_relay()) {
if (method=="INVITE" || method=="ACK") {
end_media_session();
};
sl_reply_error();
};
}
route[7] {
# caller blocked announcment
#
# this path this executed if a caller has been blocked
if (method=="INVITE" || method=="ACK") {
use_media_proxy();
};
rewriteuri("sip:699@99.99.99.100:5060");
t_on_reply("1");
if (!t_relay()) {
if (method=="INVITE" || method=="ACK") {
end_media_session();
};
sl_reply_error();
};
}
route[8] {
# anonymous call rejection announcment
#
# this path this executed for anonymous callers
if (method=="INVITE" || method=="ACK") {
use_media_proxy();
};
rewriteuri("sip:698@99.99.99.100:5060");
t_on_reply("1");
if (!t_relay()) {
if (method=="INVITE" || method=="ACK") {
end_media_session();
};
sl_reply_error();
};
}
onreply_route[1] {
# Not all 2xx messages have a content body so here we
# make sure our Content-Length > 0 to avoid a parse error
if (status=~"(180)|(183)|2[0-9][0-9]") {
if (!search("^Content-Length:\ 0")) {
use_media_proxy();
};
};
if (client_nat_test("1")) {
fix_contact();
};
}
failure_route[1] {
# if caller hung up then don't sent to voicemail
if (t_check_status("487")) {
break;
};
if (isflagset(26) && t_check_status("486")) {
# forward busy is flag 26
if (avp_pushto("$ruri", "s:fwdbusy")) {
avp_delete("s:fwdbusy");
append_branch();
resetflag(26);
# test for domestic PSTN gateway
if (avp_check("$fwd_busy_type", "eq/dom/i")) {
# test for domestic PSTN gateway
route(3);
} else if (avp_check("$fwd_busy_type", "eq/int/i")) {
# test for international PSTN gateway
route(6);
} else {
# default to sip call
route(2);
};
break;
};
};
# here we can have either voicemail __OR__ forward no answer
if (isflagset(27) && t_check_status("408")) {
# forward no answer is flag 27
if (avp_pushto("$ruri", "s:fwdnoanswer")) {
avp_delete("s:fwdnoanswer");
append_branch();
resetflag(27);
if (avp_check("$fwd_no_answer_type", "eq/dom/i")) {
# test for domestic PSTN gateway
route(3);
} else if (avp_check("$fwd_no_answer_type", "eq/int/i")) {
# test for international PSTN gateway
route(6);
} else {
# default to sip call
route(2);
};
break;
};
} else if (isflagset(31) && avp_pushto("$ruri",
"$voicemail")) {
avp_delete("$voicemail");
route(4);
break;
};
}
--- Steve Blair <blairs(a)isc.upenn.edu> wrote:
Does anyone have a working example of a config file for ser 0.9
that they can share. My 0.8.14 config keeps failing. The line
numbers in the debug output indicate the problem is on a line
that is blank. The NEWS file is no help.
Thanks
--
ISC Network Engineering
The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104
voice: 215-573-8396
215-746-8001
fax: 215-898-9348
sip:blairs@upenn.edu
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11:59 p.m.
Wow this is excellent - could you also post your asterisk
configuration (sip, extension, voicemail) -- it would greatly help me
on my quest to use asterisk as VM for SER. Many thanks!
Cheers, hank
7242
days inactive
7242
days old
2 comments
3 participants
participants (3)
-
Henning Verbeek -
Java Rockx -
Steve Blair