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