Bogdan-Andrei Iancu wrote:
> if you provide the SER2SEMS (with vm) script, maybe will be possible
> to adapt it to openser.
>
> regards,
> bogdan
>
> Taymour A. El Erian wrote:
>
>> Bogdan-Andrei Iancu wrote:
>>
>>
>>
>>> Hi Taymour,
>>>
>>> there is no such example/tutorial for openser. But if you have a
>>> working SER script, I can help you to adapt it - will be a good
>>> reference for later.
>>>
>>> regards,
>>> bogdan
>>>
>>> Taymour A. El Erian wrote:
>>>
>>>
>>>
>>>> Hi.
>>>>
>>>> Is there any how to on how to configure OpenSER, SEMS for voicemail
>>>> ?, the one with SEMS does not work
>>>>
>>>>
>>>>
>>>>
>>>
>>
>> I had a SER setup using 2 instances one of them for voicemail and voice
>> responses (using vm and SEMS). I upgraded things to OpenSER but am
>> unable to have the voicemail working.
>>
>> P.S. I inherited the script for SER and it looks so ugly, what I need is
>> a simple script which should forward to voicemail on user unavailable
>> and play messages for user not found and other things.
>
Attached the configuration I used, one (openser.cfg) used for the main
SIP server and the second was used as a second instance which serve for
voicemail.
--
Taymour A El Erian
System Division Manager
RHCE, LPIC, CCNA, MCSE, CNA
TE Data
E-mail: [EMAIL PROTECTED]
Web: www.tedata.net
Tel: +(202)-4166600
Fax: +(202)-4166700
Ext: 1101
#
# $Id: rules.m4,v 1.1 2004/01/13 19:42:56 janakj Exp $
#
# (c) 2003 iptel.org
#
# Rules to process ser.cfg templates
#
### m4 macros to make the configuration easier
### End of m4 macro section
#
# $Id: ser.cfg,v 1.2 2003/11/01 18:18:16 jiri Exp $
#
#
#
# Set the following in your CISCO PSTN gateway:
# sip-ua
# nat symmetric role passive
# nat symmetric check-media-src
#
fork=yes
port=5060
#log_stderror=yes
fifo="/tmp/openser_fifo"
debug=3
memlog=4 # memlog set high (>debug) -- no final time-consuming memory reports
on exit
mhomed=yes
listen=ip-addr-1
alias="domain.tld"
check_via=yes
dns=yes
rev_dns=no
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=0666
loadmodule "/usr/lib/openser/modules/tm.so"
loadmodule "/usr/lib/openser/modules/sl.so"
loadmodule "/usr/lib/openser/modules/acc.so"
loadmodule "/usr/lib/openser/modules/rr.so"
loadmodule "/usr/lib/openser/modules/maxfwd.so"
loadmodule "/usr/lib/openser/modules/mysql.so"
loadmodule "/usr/lib/openser/modules/usrloc.so"
loadmodule "/usr/lib/openser/modules/registrar.so"
loadmodule "/usr/lib/openser/modules/auth.so"
loadmodule "/usr/lib/openser/modules/auth_db.so"
loadmodule "/usr/lib/openser/modules/textops.so"
loadmodule "/usr/lib/openser/modules/uri.so"
loadmodule "/usr/lib/openser/modules/uri_db.so"
loadmodule "/usr/lib/openser/modules/group.so"
loadmodule "/usr/lib/openser/modules/msilo.so"
loadmodule "/usr/lib/openser/modules/nathelper.so"
loadmodule "/usr/lib/openser/modules/enum.so"
loadmodule "/usr/lib/openser/modules/domain.so"
#loadmodule "/usr/lib/openser/modules/permissions.so"
# --------------------- database fifo settings -----------------------
fifo_db_url="mysql://openser:[EMAIL PROTECTED]/openser"
modparam("usrloc|acc|auth_db|group|msilo", "db_url", "mysql://openser:[EMAIL
PROTECTED]/openser")
# -- usrloc params --
/* 0 -- dont use mysql, 1 -- write_through, 2--write_back */
modparam("usrloc", "db_mode", 2)
modparam("usrloc", "timer_interval", 10)
modparam("usrloc", "use_domain", 1)
modparam("registrar", "use_domain", 1)
# -- auth params --
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
#modparam("auth_db", "use_rpid", 1)
modparam("auth", "nonce_expire", 300)
modparam("auth", "rpid_prefix", "<sip:")
modparam("auth", "rpid_suffix",
"@ip-addr-3>;party=calling;id-type=subscriber;screen=yes;privacy=off")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# -- acc params --
# report ACKs too for sake of completeness -- as we account PSTN
# destinations which are RR, ACKs should show up
modparam("acc", "report_ack", 1)
modparam("acc", "log_level", 1)
# if BYE fails (telephone is dead, record-routing broken, etc.), generate
# a report nevertheless -- otherwise we would have no STOP event; => 1
modparam("acc", "failed_transactions", 1)
# that is the flag for which we will account -- don't forget to
# set the same one :-)
# Usage of flags is as follows:
# 1 == should account(all to gateway),
# 3 == should report on missed calls (transactions to iptel.org's users),
# 4 == destination user wishes to use voicemail
# 6 == nathelper
#
modparam("acc", "log_flag", 1)
modparam("acc", "db_flag", 1)
modparam("acc", "log_missed_flag", 3)
modparam("acc", "db_missed_flag", 3)
# report to syslog: From, i-uri, status, digest id, method
modparam("acc", "log_fmt", "fisumdpr")
# -- tm params --
modparam("tm", "fr_timer", 15)
modparam("tm", "fr_inv_timer", 25)
modparam("tm", "wt_timer", 30)
# -- msilo params
modparam("msilo", "registrar", "sip:registrar at domain.tld")
# -- enum params --
modparam("enum", "domain_suffix", "e164.arpa.")
# -- multi-domain
modparam("domain", "db_mode", 1)
# NAT features turned off -- smartnat available only in nat-capable release
# We will you flag 6 to mark NATed contacts
modparam("registrar", "nat_flag", 6)
# Enable NAT pinging
modparam("nathelper", "natping_interval", 15)
# Ping only contacts that are known to be behind NAT
modparam("nathelper", "ping_nated_only", 1)
# --------------------- request routing logic -------------------
route {
if (!mf_process_maxfwd_header("10")) {
log("LOG: Too many hops\n");
sl_send_reply("483", "Alas Too Many Hops");
break;
};
if (msg:len >= max_len) {
sl_send_reply("513", "Message too large");
break;
};
#emad setflag(3);
# special handling for natted clients; first, nat test is
# executed: it looks for via!=received and RFC1918 addresses
# in Contact (may fail if line-folding used); also,
# the received test should, if complete, should check all
# vias for presence 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(1,"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 smart enough to be symmetric. In some phones,
like
# it takes a configuration option. With Cisco 7960, it
is
# called NAT_Enable=Yes, with kphone it is called
# "symmetric media" and "symmetric signaling". (The
latter
# not part of public released yet.)
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
append_to_reply("P-NATed-Caller: Yes\r\n");
};
};
# 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):.*@((ip-addr-1)|(domain.tld))")) {
# 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==fox)
if ((method == "INVITE" || method == "SUBSCRIBE") && !((src_ip
== ip-addr-1) || ((src_ip == ip-addr-2) || (src_ip == ip-addr-3)))) {
if (!(proxy_authorize("domain.tld", "subscriber"))) {
proxy_challenge("domain.tld", "0");
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 at iptel->7141 at
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();
}; # non-REGISTER from other domain
} else if ((method == "INVITE" || method == "SUBSCRIBE" ||
method=="REGISTER" ) &&
!(uri == myself || uri =~
"(@((ip-addr-2)|(ip-addr-3))([;:].*)*)")) {
# and we serve our gateway too (we RR requests to it, so that
# its address may show up in subsequent requests after
loose_route
sl_send_reply("403", "No relaying");
break;
};
# By default we record route everything except REGISTERs
if (!(method=="REGISTER")) record_route();
# if route forces us to forward to some explicit destination, do so
#
# loose_route returns true in case that a request included
# route header fields instructing SER where to relay a request;
# if that is the case, stop script processing and just forward there;
# one could alternatively ignore the return value and treat the
# request as if it was an outbound one; that would not work however
# with broken UAs which strip RR parameters from Route. (What happens
# is that with two RR /tcp2udp, spirals, etc./ and stripped parameters,
# SER a) rewrites r-uri with RR1 b) matches uri==myself against RR1
# c) applies mistakenly user-lookup to RR1 in r-uri
if (loose_route()) {
# check if someone has not introduced a pre-loaded INVITE -- if
so,
# verify caller's privileges before accepting rr-ing
if ((method=="INVITE" || method=="ACK" || method=="CANCEL") &&
uri =~ "(@((ip-addr-2)|(ip-addr-3))([;:].*)*)") {
route(3); # Forward to PSTN gateway
} else {
append_hf("P-hint: rr-enforced\r\n");
# account all BYEs
if (method=="BYE") {log(1,"loose setflag(1)\n");
setflag(1);
}
route(1); # Generic forward
};
break;
};
# ------- check for requests targeted out of our domain... -------
if (!(uri == myself || uri =~ "(@((ip-addr-2)|(ip-addr-3))([;:].*)*)"))
{
# ... and we serve our gateway too (we RR requests to it, so
that
# its address may show up in subsequent requests after
# rewriteFromRoute
append_hf("P-hint: OUTBOUND\r\n");
route(1);
break;
};
# ------- now, the request is for sure for our domain -----------
# registers always MUST be authenticated to
# avoid stealing incoming calls
if (method == "REGISTER") {
/*
if (!allow_register("register.allow", "register.deny")) {
log(1, "LOG: alert: Forbidden IP in Contact\n");
sl_send_reply("403", "Forbidden");
break;
};
*/
# prohibit attempts to grab someone else's To address
# using valid credentials;
if (!www_authorize("domain.tld", "subscriber")) {
# challenge if none or invalid credentials
www_challenge("domain.tld", "0");
break;
};
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;
};
# aliases
# note: through a temporary error in provisioning interface, there
# are now aliases 905xx ... they take precedence overy any PSTN numbers
# as they are resolved first
lookup("aliases");
# check again, if it is still for our domain after aliases
if (!(uri == myself || uri =~ "(@((ip-addr-2)|(ip-addr-3))([;:].*)*)"))
{
append_hf("P-hint: ALIASED-OUTBOUND\r\n");
route(1);
break;
};
# Remove leading + if it is a number begining with +
if (uri =~ "^[a-zA-Z]+:\+[0-9]+@") {
strip(1);
prefix("00");
};
if (!does_uri_exist()) {
# Try numeric destinations through the gateway
if (uri =~ "^[a-zA-Z]+:[0-9]+@") {
route(3);
} else {
sl_send_reply("604", "Does Not Exist Anywhere");
};
break;
};
# does the user wish redirection on no availability? (i.e., is he
# in the voicemail group?) -- determine it now and store it in
# flag 4, before we rewrite the flag using UsrLoc
if (is_user_in("Request-URI", "voicemail")) {
setflag(4);
};
# 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 =~ "(@((ip-addr-2)|(ip-addr-3))([;:].*)*)") {
log(1, "LOG: Weird! Gateway address in UsrLoc!\n");
route(3);
break;
};
# if user is on-line and is in voicemail group, enable redirection
# no voicemail currently activated
if (method == "INVITE" && isflagset(4)) {
t_on_failure("1");
};
# ... and also report on missed calls ... note that reporting
# on missed calls is mutually exclusive with silent C timer
setflag(3);
log(1,"Set flag to 3\n");
# we now know we may, we know where, let it go out now!
append_hf("P-hint: USRLOC\r\n");
route(1);
}
#
# Forcing media relay if necesarry
#
route[1] {
# 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(6)) {
if (!is_present_hf("P-RTP-Proxy")) {
force_rtp_proxy();
append_hf("P-RTP-Proxy: YES\r\n");
};
append_hf("P-NATed-Calee: Yes\r\n");
};
# nat processing of replies; apply to all transactions (for example,
# re-INVITEs from public to private UA are hard to identify as
# natted at the moment of request processing); look at replies
t_on_reply("1");
if (method=="BYE"||method=="INVITE") {
log(1,"route[1] setflag(1)\n");
setflag(1);
};
if (!t_relay()) {
sl_reply_error();
break;
};
}
onreply_route[1] {
# natted 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();
};
# keep Cisco gatweay sending keep-alives
if (isflagset(7) && status=~"2[0-9][0-9]") {
remove_hf("Session-Expires");
append_hf("Session-Expires: 60;refresher=UAC\r\n");
fix_nated_sdp("1");
};
# if (method=="BYE") {
if (method=="BYE"||method=="INVITE") {
log(1,"On Reply[1] setflag\n");
setflag(1);
};
}
#
# logic for calls to the PSTN
#
route[3] {
#log(1,"In route [3]\n");
# discard non-PSTN methods
if (!(method == "INVITE" || method == "ACK" || method == "CANCEL" ||
method == "OPTIONS" || method == "BYE")) {
sl_send_reply("500", "only VoIP methods accepted for GW");
break;
};
# turn accounting on
# if (method=="BYE") {
setflag(1);
# };
# continue with requests to PSTN gateway ...
# no authentication needed if the destination is on our free-pstn
# list or if the caller is the digest-less gateway
#
# 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
if (method == "INVITE") {
if (!(is_user_in("Request-URI", "free-pstn") || ((src_ip ==
ip-addr-2) || (src_ip == ip-addr-3)))) {
if (!proxy_authorize("domain.tld", "subscriber")) {
proxy_challenge("domain.tld", "0");
break;
};
# let's check from=id ... avoids accounting confusion
if (!check_from()) {
log("LOG: From Cheating attempt\n");
sl_send_reply("403", "That is ugly -- use
From=id next time (gw)");
break;
};
};
if (uri=~"sip:[EMAIL PROTECTED]") {
if (!is_user_in("credentials", "local")) {
rewriteuri("sip:[EMAIL
PROTECTED]:5090");
t_relay_to_udp("ip-addr-1",
"5090");
# sl_send_reply("403", "No permission for
local calls");
break;
};
rewritehostport("ip-addr-3:5060");
#t_on_failure("7");
# the same for long-distance (destinations begin with
two zeros")
} else if (uri=~"sip:[EMAIL PROTECTED]") {
if (!is_user_in("credentials", "ld")) {
rewriteuri("sip:[EMAIL PROTECTED]:5090");
t_relay_to_udp("ip-addr-1", "5090");
# sl_send_reply("403", " no permission
for LD ");
break;
};
rewritehostport("ip-addr-3:5060");
# t_on_failure("7");
# the same for international calls (three zeros)
} else if (uri=~"sip:[EMAIL PROTECTED]") {
if (!is_user_in("credentials", "int")) {
rewriteuri("sip:[EMAIL PROTECTED]:5090");
t_relay_to_udp("ip-addr-1", "5090");
# sl_send_reply("403", "International
permissions needed");
break;
};
strip(2);
rewritehostport("ip-addr-5:5060");
# everything else (e.g., interplanetary calls) is denied
} else if (uri=~"sip:[EMAIL PROTECTED]") {
if (!is_user_in("credentials", "int")) {
rewriteuri("sip:[EMAIL PROTECTED]:5090");
t_relay_to_udp("ip-addr-1", "5090");
# sl_send_reply("403", "International
permissions needed");
break;
};
strip(2);
rewritehostport("ip-addr-6:5060");
# everything else (e.g., interplanetary calls) is denied
}else {
rewriteuri("sip:[EMAIL PROTECTED]:5090");
t_relay_to_udp("ip-addr-1", "5090");
# sl_send_reply("403", "Forbidden");
break;
};
consume_credentials();
append_hf("P-Hint: GATEWAY\r\n");
log(1,"TRELAY IN ROUTE[3]\n");
t_relay();
}; # authorized PSTN
break;
}
failure_route[7] {
rewritehostport("ip-addr-4:5060");
append_branch();
t_relay();
}
# ------------- handling of unavailable user ------------------
route[4] {
# message store
if (method == "MESSAGE") {
if (!t_newtran()) {
sl_reply_error();
break;
};
if (m_store("0")) {
t_reply("202", "Accepted for Later Delivery");
break;
};
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;
};
if (method == "INVITE") {
acc_log_request("404 missed call\n");
acc_db_request("404 missed call", "missed_calls");
};
# if (t_newtran()) {
# if (method == "ACK") {
# log(1, "CAUTION: strange thing: ACK passed
t_newtran\n");
# break;
# };
if (!isflagset(4)) {
# route(6);
rewriteuri("sip:[EMAIL PROTECTED]:5090");
t_relay_to_udp("ip-addr-1", "5090");
# sl_send_reply("404", "Not Found and no voicemail turned
on");
break;
};
# forward to voicemail now
rewritehostport("ip-addr-1:5090");
t_relay_to_udp("ip-addr-1", "5090");
# t_reply("404", "Not Found");
# };
# we account missed incoming calls; previous statteful processing
# guarantees that retransmissions are not accounted
}
#route[5]{
# log(1,"route 5555555555555555555555555555\n");
# if (status=~"408") {
# rewritehostport("ip-addr-1:5090");
# append_branch();
# t_relay_to_udp("ip-addr-1", "5090");
# };
#}
# if forwarding downstream did not succeed, try voicemail running
# at bat.iptel.org:5090
failure_route[1] {
# if (method == "INVITE") {
# acc_log_request("404 missed call\n");
# acc_db_request("404 missed call", "missed_calls");
# };
log(1,"FAILURE ROUTE 1\n");
if (t_check_status("408|486")) {
log(1,"408 or 486\n");
revert_uri();
rewritehostport("ip-addr-1:5090");
append_branch();
t_relay_to_udp("ip-addr-1", "5090");
}
}
#
# $Id: voicemail.cfg,v 1.5 2004/01/14 18:23:50 rco Exp $
#
# this script is configured for use as voicemail UAS; it processes
# INVITEs and BYEs and asks SEMS to record media via "vm"; in this
# script, all record-routing and other constructs known from proxy
# scripts are not present -- it is a simple UAS
#
# ----------- global configuration parameters ------------------------
debug=7 # debug level (cmd line: -dddddddddd)
#fork=no
#log_stderror=yes # (cmd line: -E)
#listen=212.103.160.198
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5090
children=4
fifo="/tmp/vm_ser_fifo"
# ------------------ module loading ----------------------------------
loadmodule "/usr/lib/ser/modules/sl.so"
loadmodule "/usr/lib/ser/modules/tm.so"
loadmodule "/usr/lib/ser/modules/maxfwd.so"
loadmodule "/usr/lib/ser/modules/mysql.so"
loadmodule "/usr/lib/ser/modules/vm.so"
# ----------------- setting module-specific parameters ---------------
#modparam("vm", "db_url","mysql://ser:[EMAIL PROTECTED]/ser")
# ------------------------- request routing logic -------------------
# main routing logic
route{
# initial sanity checks -- messages with
# max_forwars==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;
};
if (!uri==myself) {
sl_send_reply("404", "not reponsible for host in r-uri");
break;
};
# Voicemail specific configuration - begin
if(method=="ACK" || method=="INVITE" || method=="BYE"){
if (!t_newtran()) {
log("could not create new transaction\n");
sl_send_reply("500","could not create new transaction");
break;
};
t_reply("100","Trying -- just wait a minute !");
if(method=="INVITE"){
if (uri=~"sip:[EMAIL PROTECTED]")
{
if (!vm("/tmp/am_fifo", "announcement"))
{
log(1,"couldn't contact announcement
server\n");
t_reply("500", "couldn not contact
announcement server");
};
log("!!!!!!!!!!!!!!!!!!!!!ANNOUNCEMENT!!!!!!!!!!!!!!!!!!!!!!\n");
}
else if(uri=~"sip:[EMAIL PROTECTED]")
{
if (!vm("/tmp/am_fifo", "number_reader"))
{
log("could not contact the number
reader server\n");
t_reply("500","could not contact ivr
server");
};
log
("!!!!!!!!!!!!!!!!!!!!!!!!!IVR!!!!!!!!!!!!!!!!!!!!!!!!!!!!!!\n");
}
else if (uri=~"sip:[EMAIL PROTECTED]")
{
if (!vm("/tmp/am_fifo", "conference"))
{
log("Could not contact conference
server\n");
t_reply("500","Could not contact
conference server");
};
log
("!!!!!!!!!!!!!!!!!!!!!CONFERENCE!!!!!!!!!!!!!!!!!!!!!!!!!\n");
}
else
{
if(!vm("/tmp/am_fifo","voicemail"))
#if(!vm("/tmp/am_fifo","ivr"))
{
log(1,"could not contact the answer
machine\n");
t_reply("500","could not contact the
answer machine");
};
log("!!!!!!!!!!!!!!!!!!!!!!VOICEMAIL!!!!!!!!!!!!!!!!!!!!!!!!!\n");
};
#
# log("**************** vm start - begin
******************\n");
# if(!vm("/tmp/am_fifo","voicemail")){
# log("could not contact the answer machine\n");
# t_reply("500","could not contact the answer
machine");
# };
# log("**************** vm start - end
******************\n");
} else if(method=="BYE"){
log("**************** vm end - begin
******************\n");
if(!vm("/tmp/am_fifo","bye")){
log("could not contact the answer machine\n");
t_reply("500","could not contact the answer
machine");
};
log("**************** vm end - end
******************\n");
};
break;
};
if (method=="CANCEL") {
sl_send_reply("200", "cancels are junked here");
break;
};
sl_send_reply("501", "method not understood here");
}
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