Hello,
Has anyone tried the voicemail module along with
SEMS (Sip Express Media Server)?
I have tried to use it with an ATA 186 phone but I
get the following erros:
(13328) ERROR: parse_sdp_line_ex
(AmSdp.cpp:317): parse_sdp_line : parameter 'v=' was not
found
(process:13354): oRTP-WARNING **: Error receiving udp packet: Socket
operation on non-socket.
(process:13354): oRTP-WARNING **: Error receiving
udp packet: Socket operation on non-socket.
(process:13354): oRTP-WARNING **: Error receiving
udp packet: Socket operation on non-socket.
(process:13354):
oRTP-WARNING **: Error receiving udp packet: Socket operation on
non-socket.
...... it keeps logging this message
.....
Any ideas why is this happening?
I also would like to know what codecs
does this voicemail supports?
Is there a way to retrieve the messages by phone or
just by email?
Attached are ethereal captures and config
files.
Regards,
Claudio Thorell
##########################
SER.CFG
##########################
# ----------- global configuration parameters
------------------------
debug=3
# debug level (cmd line:
-dddddddddd)
fork=yes
log_stderror=no
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
----------------------------------
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/acc.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/exec.so"
loadmodule
"/usr/local/lib/ser/modules/mysql.so"
loadmodule
"/usr/local/lib/ser/modules/auth.so"
loadmodule
"/usr/local/lib/ser/modules/textops.so"
loadmodule
"/usr/local/lib/ser/modules/auth_db.so"
loadmodule
"/usr/local/lib/ser/modules/uri.so"
loadmodule
"/usr/local/lib/ser/modules/vm.so"
# ----------------- setting module-specific
parameters ---------------
# -- usrloc params --
modparam("usrloc",
"db_mode", 2)
modparam("usrloc", "timer_interval", 10)
# -- auth params --
modparam("auth", "secret",
"alsdkhglaksdhfkloiwr")
modparam("auth_db", "calculate_ha1",
yes)
modparam("auth_db", "password_column", "password")
# -- tm params
--
modparam("tm","ruri_matching",0)
# ------------------------- 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 (len_gt( max_len ))
{
sl_send_reply("513", "Message too
big");
break;
};
# 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
record_route();
# loose-route
processing
loose_route();
# Make
MSN Messenger happy...
if
(method=="REGISTER")
{
log(1,"Register
message\n");
save("tln_location");
sl_send_reply("200","ok");
break;
};
# Voicemail specific configuration - begin
if(method=="ACK" || method=="INVITE" || method=="BYE"){
if(t_newtran()){
t_reply("100","Trying -- just wait a minute !");
if(method=="INVITE"){
log(1,"**************** 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(1,"**************** vm start - end
******************\n");
break;
};
if(method=="BYE"){
log(1,"**************** 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(1,"**************** vm end - end
******************\n");
break;
};
}
else
{
log("could not create new
transaction\n");
sl_send_reply("500","could not create new
transaction");
};
};
# Voicemail specific configuration - end
}
##########################
SEMS.CFG
##########################
# $Id: sems.conf.sample,v 1.1 2003/06/17 16:05:01 ullstar Exp $
#
#
sems.conf.sample
#
# Sip Express Media Server (sems)
#
# sample
configuration file
#
#
# whitespaces (spaces and tabs) are ignored
#
comments start with a "#" and may be used inline
#
# example:
option=value1, value2 # i like this option
#
##################################
# global
parameters
#
##################################
# optional parameter: fork={yes|no}
#
# - specifies if sems should
run in daemon mode (background)
fork=yes
# optional parameter: stderr={yes|no}
#
# - debug mode: do not fork
and log to stderr
stderr=no
# optional parameter: loglevel={0|1|2|3}
#
# - sets log level
(error=0, warning=1, info=2, debug=3)
loglevel=1
# optional parameter: fifo_name=<filename>
#
# - path and file
name of our fifo file
fifo_name=/tmp/am_fifo
# optional parameter: ser_fifo_name=<filename>
#
# - path and
file name of Ser's fifo file
ser_fifo_name=/tmp/ser_fifo
# optional parameter: plugin_path=<path>
#
# - sets the path to
the plug-ins
# - may be absolute or relative to
CWD
plugin_path=/usr/local/src/answer_machine/lib
##################################
# voicemail specific
parameters #
##################################
# optional parameter: announce_path=<path>
#
# - sets the path
where announce files are searched
for
announce_path=/usr/local/src/answer_machine/wav/
# optional parameter: default_announce=<filename>
#
# - sets
the name of the default announce WAV
file
#default_announce=/usr/local/src/answer_machine/wav/default.wav
default_announce=default.wav
# optional parameter: max_record=<seconds>
#
# - maximum record
time
max_record=30
# optional parameter: smtp_server=<hostname>
#
# - sets address
of smtp server
smtp_server=localhost
# optional parameter: smtp_port=<port>
#
# - sets port of smtp
server
smtp_port=25
##################################
# module specific
parameters #
##################################
# sample isdngw module configuration (external file)
#
config.isdngw=/etc/isdngw.conf
# sample isdngw module configuration (inline)
config.isdngw=inline
# parameters for outgoing service (SIP -> PSTN)
# required parameter: outdevices=<dev1>, <dev2>, <dev3>,
...
#
# - specifies which ttyI* devices to use for outgoing
telephony calls
# - devices must be fully accessible by the vm process'
user
# - the number of devices listed is the maximum of
simultaneous
# outgoing phone calls (if not otherwise
restricted)
outdevices=/dev/ttyI10, /dev/ttyI11, /dev/ttyI12
# required parameter: outmsn=<msn>
#
# - specifies the
default msn for outgoing calls
outmsn=
# optional parameter: lockdir=/where/to/store/logfiles
#
# -
specifies the directory where to put the lockfiles
# - default:
lockdir=/var/lock
lockdir=/var/lock
# optional parameter: outmaxconn=<number>
#
# - specifies
the maximum number of outgoing connections
# - parameter is max-limited
by:
# * number of devices specified in
outdevices
# * number of available ISDN
b-channels
# - setting to 0 or omitting the parameter allows any number
of calls
outmaxcon=0
# optional parameter: outlogfile=<file>
#
# - specifies a
log file, where all outgoing calls are listed
outlogfile=
# optional parameter: forcenumber=<start of number>
#
# -
specifies allowed numbers
# - e.g. forcenumber=030 means allow only
numbers starting with 030
forcenumber=
# end of configuration section for isdngw module
config.isdngw=end
# add more module configurations here (inline or external):
#
#
config.mymodule=<filename>
# or
# config.mymodule=inline
#
...
# config.mymodule=end