serusers Hi everyone !
I have installed ser-0.8.10 and ser-mysql-0.8.10 on Red Hat Linux 7.2. After I configured the script . The SIP server can worked well in LAN . Two telephones both in the LAN can get thought . but when I put ser on Internet, they failed . I have no idea what I can do .Here is a document describe more detail.
I hope anyone can help me .
Thank you!
gg
gdut200104024532(a)163.com
2004-08-19
Hi , I am trying to add Voicemail services... the problem is that when the call is missed (busy, not answered or not connected) I can see some debug in ser´s second instance for users in "voicemail group" but the call is closed and a busy tone return to the caller...
bellow you can see my 3 config files and details from both ser debugs, please someone send me some advice.
thank you
Rafael
- ser.cfg for Main SER running on 5060
- voicemail.cfg for second instance on port 5090
- sems.conf for Media Server (sems)
SEMS.CFG ------->
----------- global configuration parameters ------------------------
#debug=3 # debug level (cmd line: -dddddddddd)
#fork=yes
#log_stderror=no # (cmd line: -E)
#/* Uncomment these lines to enter debugging mode
debug=9
fork=yes
log_stderror=yes
#*/
listen=200.110.2.131
listen=127.0.0.1
port=5060
# hostname matching an alias will satisfy the condition uri==myself".
alias=call.millicom.com.pe
alias=200.110.2.131
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
children=4
fifo="/tmp/ser_fifo"
# ------------------ 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/group.so"
loadmodule "/usr/local/lib/ser/modules/uri.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
# digest authentication
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters ---------------
modparam("usrloc", "db_mode", 2)
# storing passwords in our database in plain text:
# modparam("auth_db", "calculate_ha1", yes)
# modparam("auth_db", "password_column", "password")
# For Rad Accounting
modparam("acc","radius_config","/usr/local/etc/radiusclient/radiusclient.conf")
modparam("acc", "service_type", 15)
modparam("acc", "radius_flag", 1)
modparam("acc", "radius_missed_flag", 3)
modparam("acc", "report_ack", 0)
modparam("tm", "fr_timer", 20 )
modparam("tm", "fr_inv_timer", 30 )
modparam("tm", "wt_timer", 20 )
modparam("tm", "uac_from", "sip:avisos@millicom.net.pe" )
modparam("rr", "enable_full_lr", 1)
modparam("group", "db_url", "sql://ser:heslo@localhost/ser")
modparam("uri", "db_url", "sql://ser:heslo@localhost/ser")
# --------------------- request routing logic -------------------
route {
# initial sanity checks -- messages with
# max_forwards==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;
};
# 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
if (loose_route()) {
t_relay();
break;
};
# set for accounting:
if (method=="INVITE") {
log(1, "INVITE\n");
setflag(1); /* set for accounting (the same value as in log_flag!) */
};
if (method=="BYE" || method=="CANCEL") {
log (1, "BYE or CANCEL\n");
setflag(1);
};
if (!uri==myself) {
t_relay();
break;
};
if (method == "REGISTER") {
# to use digest authentication
if (!www_authorize("call.millicom.com.pe", "subscriber")) {
www_challenge("call.millicom.com.pe", "0");
break;
};
if (!save("location")) {
sl_reply_error();
};
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);
};
setflag(3);
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
# handle user which was not found
route(4);
break;
};
# if user is on-line and is in voicemail group, enable redirection
if (method == "INVITE" && isflagset(4)) {
t_on_failure("1");
};
t_relay();
}
# ------------- handling of unavailable user ------------------
route[4] {
# non-Voip -- just send "off-line"
if (!(method=="INVITE" || method=="ACK" || method=="CANCEL")) {
sl_send_reply("404", "Not Found");
acc_rad_request("404 Not Found");
break;
};
# not voicemail subscriber
if (!isflagset(4)) {
sl_send_reply("404", "Not Found and no voicemail turned on");
acc_rad_request("404 Not Found");
break;
};
# forward to voicemail now
rewritehostport("call.millicom.com.pe:5090");
t_relay_to_udp("call.millicom.com.pe", "5090");
}
# if forwarding downstream did not succeed, try voicemail running
# at bat.iptel.org:5090
failure_route[1] {
revert_uri();
rewritehostport("call.millicom.com.pe:5090");
append_branch();
t_relay_to_udp("call.millicom.com.pe", "5090");
}
################ VOICEMAIL.CFG -------->
# ----------- global configuration parameters ------------------------
#debug= # debug level (cmd line: -dddddddddd)
#fork=no
#log_stderror=yes # (cmd line: -E)
#/* Uncomment these lines to enter debugging mode
debug=20
fork=yes
log_stderror=yes
#*/
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/local/lib/ser/modules/sl.so"
loadmodule "/usr/local/lib/ser/modules/tm.so"
loadmodule "/usr/local/lib/ser/modules/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/mysql.so"
loadmodule "/usr/local/lib/ser/modules/vm.so"
# ----------------- setting module-specific parameters ---------------
modparam("voicemail", "db_url","sql://ser:heslo@localhost/ser")
modparam("voicemail", "subscriber_table", "subscriber")
modparam("voicemail", "email_column", "email_address")
# ------------------------- 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"){
log("**************** vm start - begin ******************\n");
if (uri=~"sip:as_welcome@.*" || uri=~"sip:as_nomoney@.*") {
if (!vm("/tmp/am_fifo", "announcement")) {
log("couldn't contact announcement server\n");
t_reply("500", "couldn not contact announcement server");
};
} else {
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");
}
###################
SEMS.CFG ---------->
[root@gkproxy01 answer_machine]# more /usr/local/etc/sems/sems.conf
# $Id: sems.conf.sample,v 1.11.2.5 2004/05/24 14:33:07 rco 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=no is the same as -E)
fork=yes
# optional parameter: stderr={yes|no}
#
# - debug mode: do not fork and log to stderr
# (stderr=yes is the same as -E)
stderr=no
# optional parameter: loglevel={0|1|2|3}
#
# - sets log level (error=0, warning=1, info=2, debug=3)
# (same as -D)
loglevel=1
# optional parameter: fifo_name=<filename>
#
# - path and file name of our fifo file (same as -i)
fifo_name=/tmp/am_fifo
# optional parameter: ser_fifo_name=<filename>
#
# - path and file name of Ser's fifo file (same as -o)
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/lib/sems/plug-in/
# optional parameter: smtp_server=<hostname>
#
# - sets address of smtp server
smtp_server=200.110.2.44
# optional parameter: smtp_port=<port>
#
# - sets port of smtp server
smtp_port=25
# optional parameter: rtp_low_port=<port>
#
# - sets port of rtp lowest server
#rtp_low_port=1024
# optional parameter: rtp_high_port=<port>
#
# - sets port of rtp highest server
#rtp_high_port=65535
##################################
# module specific parameters #
##################################
# sample voicemail configuration (inline)
config.voicemail=inline
# optional parameter: announce_path=<path>
#
# - sets the path where announce files are searched for
# - the file to be played is determined the following way:
# <announce_path>/<domainname>/<username>.wav
# if this file is not available <announce_path>/<default_anounce> is used
announce_path=/usr/local/lib/sems/audio/
# parameter: default_announce=<filename>
#
# - sets the name of the default announce WAV file
default_announce=default_en.wav
# parameter: max_record_time=<seconds>
#
# - maximum record time
max_record_time=30
# parameter: email_template=<filename>
#
# - email template file
#
# See the README file in <sems-src>/plug-in/voicemail
# for more information on the syntax used.
#
email_template=/usr/local/lib/sems/plug-in/mail.template
# end of configuration section for voicemail module
config.voicemail=end
# sample announcement configuration (inline)
config.announcement=inline
# optional parameter: announce_path=<path>
#
# - sets the path where announce files are searched for
announce_path=/usr/local/lib/sems/audio/
# parameter: default_announce=<filename>
#
# - sets the name of the default announce WAV file
default_announce=default_en.wav
# end of configuration section for announcement module
config.announcement=end
# sample isdngw module configuration (external file)
# config.isdngw=/etc/isdngw.conf
# sample ivr module configuration (inline)
config.ivr=inline
#parameter: python_script_path=<full path>
python_script_path=/etc/ivr
#parameter: python_script_file=<filename>
python_script_file=example.py
# end of configuration section for ivr module
config.ivr=end
# sample conference configuration (inline)
config.conference=inline
# parameter: default_announce=<filename>
#
# - sets the full pathed name of the default announce WAV file.
# Will be played to lonely users.
default_announce=/usr/local/lib/sems/audio/first_participant.wav
# parameter: join_sound=<filename>
#
# - sets the full pathed name of a WAV file
# which will be played to conference users
# when a new user joins the conference.
join_sound=/usr/local/lib/sems/audio/beep.wav
# parameter: drop_sound=<filename>
#
# - sets the full pathed name of a WAV file
# which will be played to conference users
# when a user drops the conference.
drop_sound=/usr/local/lib/sems/audio/beep.wav
# end of configuration section for conference module
config.conference=end
# example configuration for number reader
config.number_reader=inline
number_path=/usr/local/lib/sems/audio/
prolog_file=welcome_to_number_reader.wav
epilog_file=thanks_calling_number_reader.wav
# end of number_reader configuration
config.number_reader=end
# add more module configurations here (inline or external):
#
# config.mymodule=<filename>
# or
# config.mymodule=inline
# ...
# config.mymodule=end
[root@gkproxy01 answer_machine]#
DEBUG from 1st instamce (port 5060)
---------------------------------------
4(31737) DEBUG: relay_reply: branch=1, save=0, relay=1
4(31737) old size: 669, new size: 607
4(31737) build_res_from_sip_res: copied size: orig:108, new: 46, rest: 561 msg=
SIP/2.0 404 not reponsible for host in r-uri
Via: SIP/2.0/UDP 10.0.0.236:5060;branch=z9hG4bKa200bcf5a4114
From: <sip:6603000@call.millicom.com.pe>;tag=a200bcf5a4
To: <sip:6604000@call.millicom.com.pe>;tag=3749ec7003921b5c92fe06c5dc660395.5093
Call-ID: a2609600-48ac-bcf1-81f5-0002a40055b2(a)10.0.0.236
CSeq: 114 INVITE
Server: Sip EXpress router (0.8.14 (i386/linux))
Content-Length: 0
Warning: 392 200.110.2.131:5090 "Noisy feedback tells: pid=31707 req_src_ip=200.110.2.131 req_src_port=5060 in_uri=sip:6604000@call.millicom.com.pe:5090 out_uri=sip:6604000@call.millicom.com.pe:5090 via_cnt==2"
4(31737) DEBUG: reply relayed. buf=0x80dec48: SIP/2.0 4..., shmem=0x422be8a8: SIP/2.0 4
4(31737) DBG: callback type 7, id 1 entered
4(31737) DEBUG: cleanup_uacs: RETR/FR timers reset
4(31737) DEBUG: add_to_tail_of_timer[4]: 0x422c287c
4(31737) DEBUG: add_to_tail_of_timer[0]: 0x422c2890
4(31737) DEBUG:destroy_avp_list: destroing list (nil)
4(31737) receive_msg: cleaning up
[root@gkproxy01 admin]#
[root@gkproxy01 admin]# 2(31735) SIP Request:
2(31735) method: <ACK>
2(31735) uri: <sip:6604000@call.millicom.com.pe>
2(31735) version: <SIP/2.0>
2(31735) parse_headers: flags=1
2(31735) Found param type 232, <branch> = <z9hG4bKa200bcf5a4114>; state=16
2(31735) end of header reached, state=5
2(31735) parse_headers: Via found, flags=1
2(31735) parse_headers: this is the first via
2(31735) After parse_msg...
2(31735) preparing to run routing scripts...
2(31735) DEBUG : sl_filter_ACK: to late to be a local ACK!
2(31735) DEBUG : is_maxfwd_present: searching for max_forwards header
2(31735) parse_headers: flags=128
2(31735) DEBUG: add_param: tag=3749ec7003921b5c92fe06c5dc660395.5093
2(31735) end of header reached, state=29
2(31735) DEBUG: get_hdr_field: <To> [78]; uri=[sip:6604000@call.millicom.com.pe]
2(31735) DEBUG: to body [<sip:6604000@call.millicom.com.pe>]
2(31735) get_hdr_field: cseq <CSeq>: <114> <ACK>
2(31735) DEBUG: get_hdr_body : content_length=0
2(31735) DEBUG: is_maxfwd_present: value = 70
2(31735) DEBUG: add_param: tag=a200bcf5a4
2(31735) end of header reached, state=29
2(31735) parse_headers: flags=256
2(31735) found end of header
2(31735) find_first_route(): No Route headers found
2(31735) loose_route(): There is no Route HF
2(31735) check_self - checking if host==us: 20==13 && [call.millicom.com.pe] == [200.110.2.131]
2(31735) check_self - checking if port 5060 matches port 5060
2(31735) check_self - checking if host==us: 20==9 && [call.millicom.com.pe] == [127.0.0.1]
2(31735) check_self - checking if port 5060 matches port 5060
2(31735) query="select grp from grp where username='6604000' AND grp='voicemail'"
2(31735) is_user_in(): User is in group 'voicemail'
2(31735) rwrite(): Rewriting Request-URI with 'sip:6604000@200.110.6.58'
2(31735) DEBUG: t_addifnew: msg id=11467 , global msg id=11434 , T on entrance=0xffffffff
2(31735) parse_headers: flags=-1
2(31735) parse_headers: flags=60
2(31735) t_lookup_request: start searching: hash=31284, isACK=1
2(31735) DEBUG: RFC3261 transaction matched, tid=a200bcf5a4114
2(31735) DEBUG: t_lookup_request: transaction found (T=0x422c27c8)
2(31735) DEBUG: cleanup_uacs: RETR/FR timers reset
2(31735) DEBUG: add_to_tail_of_timer[2]: 0x422c2810
2(31735) DEBUG:destroy_avp_list: destroing list (nil)
2(31735) receive_msg: cleaning up
DEBUG FROM 2nd INSTANCE (PORT 5090)
--------------------------------------------------
]
7(31709) get_hdr_field: cseq <CSeq>: <113> <INVITE>
7(31709) DEBUG: get_hdr_body : content_length=180
7(31709) DEBUG: is_maxfwd_present: value = 69
7(31709) check_self - checking if host==us: 20==9 && [call.millicom.com.pe] == [127.0.0.1]
7(31709) check_self - checking if port 5090 matches port 5090
7(31709) check_self - checking if host==us: 20==13 && [call.millicom.com.pe] == [200.110.2.131]
7(31709) check_self - checking if port 5090 matches port 5090
7(31709) check_self: host != me
7(31709) parse_headers: flags=-1
7(31709) found end of header
7(31709) check_via_address(200.110.2.131, 200.110.2.131, 0)
7(31709) DEBUG:destroy_avp_list: destroing list (nil)
7(31709) receive_msg: cleaning up
5(31705) SIP Request:
5(31705) method: <ACK>
5(31705) uri: <sip:6605454@call.millicom.com.pe:5090>
5(31705) version: <SIP/2.0>
5(31705) parse_headers: flags=1
5(31705) Found param type 232, <branch> = <z9hG4bKd517.2ea1e793.0>; state=16
5(31705) end of header reached, state=5
5(31705) parse_headers: Via found, flags=1
5(31705) parse_headers: this is the first via
5(31705) After parse_msg...
5(31705) preparing to run routing scripts...
5(31705) parse_headers: flags=4
5(31705) DEBUG: add_param: tag=3749ec7003921b5c92fe06c5dc660395.01b8
5(31705) end of header reached, state=29
5(31705) DEBUG: get_hdr_field: <To> [78]; uri=[sip:6605454@call.millicom.com.pe]
5(31705) DEBUG: to body [<sip:6605454@call.millicom.com.pe>]
5(31705) DEBUG: sl_filter_ACK : local ACK found -> dropping it!
5(31705) DEBUG:destroy_avp_list: destroing list (nil)
5(31705) receive_msg: cleaning up
6(31707) SIP Request:
6(31707) method: <INVITE>
6(31707) uri: <sip:6604000@call.millicom.com.pe:5090>
6(31707) version: <SIP/2.0>
6(31707) parse_headers: flags=1
6(31707) Found param type 232, <branch> = <z9hG4bK43a7.d99b0dc7.1>; state=16
6(31707) end of header reached, state=5
6(31707) parse_headers: Via found, flags=1
6(31707) parse_headers: this is the first via
6(31707) After parse_msg...
6(31707) preparing to run routing scripts...
6(31707) DEBUG : is_maxfwd_present: searching for max_forwards header
6(31707) parse_headers: flags=128
6(31707) Found param type 232, <branch> = <z9hG4bKa200bcf5a4114>; state=16
6(31707) end of header reached, state=5
6(31707) parse_headers: Via found, flags=128
6(31707) parse_headers: this is the second via
6(31707) end of header reached, state=9
6(31707) DEBUG: get_hdr_field: <To> [36]; uri=[sip:6604000@call.millicom.com.pe]
6(31707) DEBUG: to body [<sip:6604000@call.millicom.com.pe>
]
6(31707) get_hdr_field: cseq <CSeq>: <114> <INVITE>
6(31707) DEBUG: get_hdr_body : content_length=180
6(31707) DEBUG: is_maxfwd_present: value = 69
6(31707) check_self - checking if host==us: 20==9 && [call.millicom.com.pe] == [127.0.0.1]
6(31707) check_self - checking if port 5090 matches port 5090
6(31707) check_self - checking if host==us: 20==13 && [call.millicom.com.pe] == [200.110.2.131]
6(31707) check_self - checking if port 5090 matches port 5090
6(31707) check_self: host != me
6(31707) parse_headers: flags=-1
6(31707) found end of header
6(31707) check_via_address(200.110.2.131, 200.110.2.131, 0)
6(31707) DEBUG:destroy_avp_list: destroing list (nil)
6(31707) receive_msg: cleaning up
8(31711) SIP Request:
8(31711) method: <ACK>
8(31711) uri: <sip:6604000@call.millicom.com.pe:5090>
8(31711) version: <SIP/2.0>
8(31711) parse_headers: flags=1
8(31711) Found param type 232, <branch> = <z9hG4bK43a7.d99b0dc7.1>; state=16
8(31711) end of header reached, state=5
8(31711) parse_headers: Via found, flags=1
8(31711) parse_headers: this is the first via
8(31711) After parse_msg...
8(31711) preparing to run routing scripts...
8(31711) parse_headers: flags=4
8(31711) DEBUG: add_param: tag=3749ec7003921b5c92fe06c5dc660395.5093
8(31711) end of header reached, state=29
8(31711) DEBUG: get_hdr_field: <To> [78]; uri=[sip:6604000@call.millicom.com.pe]
8(31711) DEBUG: to body [<sip:6604000@call.millicom.com.pe>]
8(31711) DEBUG: sl_filter_ACK : local ACK found -> dropping it!
8(31711) DEBUG:destroy_avp_list: destroing list (nil)
8(31711) receive_msg: cleaning up
---------------------------------
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Hi there,
It sounds like radius doesn't know about the SIP specific attributes.
Did you make sure you added the contents of the dictionary.ser to the
main radius client/radius server dictionary files? Or at least include
them?
To add simply, then place at the end of each respective dictionary file.
Normally for radius server at /etc/raddb/dictionary or
/usr/local/etc/raddb/dictionary
Normally for radius client at /etc/radiusclient/dictionary or
/usr/local/etc/radiusclient/dictionary
Or make sure you have it in the correct include directory.
Best regards,
Dave
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of Masoud Safi
Sent: 18 August 2004 06:55
To: serusers(a)lists.iptel.org
Subject: [Serusers] SER - ERROR: acc: can't get code for the
Sip-Methodattribute
In my SER installation, I have RADIUS accounting enabled, but when I
start
ser I get the following:
Maxfwd module- initializing
acc - initializing
0(27928) ERROR: acc: can't get code for the Sip-Method attribute
0(27928) init_mod(): Error while initializing module acc
ERROR: error while initializing modules
I found the advise from a ser user on removing the comments from the
dictionary.ser file, which I did but it did not help.
I downloaded the latest (supposedly) from
http://download.berlios.de/radiusclient-ng/radiusclient-0.4.3.tar.gz and
installed, that did not help either.
Now I am wondering if any one has solved such an issue?
Thanks
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I have just been added to a project to investigate SIP. One of my UN*X
admins installed SER on a RedHat box. I have attempted to use the command:
$/usr/sbin/serctl add username password e-mail (with the appropriate
values), but am asked for a MySQL password. MySQL is not installed on the
machine, and /usr/lib/ser/modules/mysql.so is not loaded. Do I need MySQL
in order to run SER? The documentation doesn't state that I need it.
-Todd
Hi all members,
Is there any way to authorize a call via radius? I mean, to check the
account's credit availability in order to let the user make the call and
then setup a timer for the longest time user can call? Just like a prepaid
service.
My question actually is, Exists that module or I should develop
something to do this?
Thank you very much in advance for your answer...
JIM
Hi, simple question :
- Main SER running on 5060
- Second SER instance running on 5090 to work with SEMS for Voicemail.
When a call arrive on the main SER:5060 and that is redirected to Voicemail, does my customer USER AGENT need to be able to connect to the 5090 port ?
Do i need to open this port ? Or it is the main ser instance that negociate with the SER:5090 ??
thanks
Hi all, i am testing rel 0.8.14 and see that when enable natping from nathelper module, for each registered user the module print an "ERROR: nathelper::timer: can't resolve_hos" message after N seconds (N=modparam natping interval).
I see in the 1503 nathelper.c line code the DBG line after funcion sip_resolvehost()
1501 he = sip_resolvehost(&curi.host, &curi.port_no, PROTO_UDP);
1502 if (he == NULL){
1503 LOG(L_ERR, "ERROR: nathelper::timer: can't resolve_hos\n");
1504 continue;
1505 }
* Note: in the case of ERROR the current contact is in the form:
<sip:someuser@172.16.22.33:5060>;q=0.00;expires=565
Thanks for comments
Ezequiel Colombo
Can SER at least authenticate its request sent to an uplink-proxy?
(Using different creditentials then used from the user-client? Ie.
can concatenated SERs authenticate against each other?
Dietmar
Soren (Home) said:
>
> Hi Dietmar,
>
> SER does not contain any UserAgent. It is just a proxy. Probably what you
> want, is to use Asterisk instead (http://www.asterisk.org/). It can
> register to foreign proxies.
I have tried Asterisk before but was not very happy as Asterisk seems to have
its own idea on how to forward SIP messages, there is no 1:1 forwarding
of messages, eg. a BUSY from uplink proxy is not forwarded as BUSY to
the UA.
>
>
> Cheers, Soren
>
>>
>> Hello,
>>
>> after reading some documents on iptel, googling and going
>> through the mailing list I still do not know how to set up
>> the following configuration - or if such a configuration is
>> possible with SER at all:
>>
>> +---------+ +-----+ +-------+
>> |sip-phone|----| SER |----|sipgate|--PSTN...
>> +---------+ +-----+ +-------+
>> (...)
>> I have some sip-phones which should register at "my" SER
>> (this is the easy part).
>> I also have an account at sipgate (and other GW-providers)
>> which I would like to use for incoming and outgoing calls.
>>
>> How can I configure my SER so that it REGISTERs at sipgate,
>> so that sipgate will send INVITEs to my SER when there is an
>> incoming call?
>>
>> How can I configure SER so that is authenticates
>> against sipgate when setting up an ougoing call?
>>
>>
>> The final setup should allow me to make "internal" calls
>> between the sip-phones, receive incoming calls from some
>> "public" GW-providers and use "least-cost-routing" for
>> external calls (use cheapest GW depending on destination).
>> I believe that such a configuration is not that exotic,
>> but I could not find a usable sample...
>>
>> Thank you
>>
>> Dietmar Wessel
>>
>>
>> _______________________________________________
>> Serusers mailing list
>> serusers(a)lists.iptel.org
>> http://lists.iptel.org/mailman/listinfo/serusers
>>
>
>
> --
> Soren Davidsen,
> Univ: AUE department of computer science, room H302, +45-20676081
> Home: http://tanesha.net/, +45-36947805 or +1-3602266324
> Ring billigt med IP-telefoni: http://musimi.dk/, http://GratisSip.dk/
>
--
Hello,
after reading some documents on iptel, googling and going
through the mailing list I still do not know how to set up
the following configuration - or if such a configuration is
possible with SER at all:
+---------+ +-----+ +-------+
|sip-phone|----| SER |----|sipgate|--PSTN...
+---------+ +-----+ +-------+
(...)
I have some sip-phones which should register at "my" SER
(this is the easy part).
I also have an account at sipgate (and other GW-providers)
which I would like to use for incoming and outgoing calls.
How can I configure my SER so that it REGISTERs at sipgate,
so that sipgate will send INVITEs to my SER when there is an
incoming call?
How can I configure SER so that is authenticates
against sipgate when setting up an ougoing call?
The final setup should allow me to make "internal" calls
between the sip-phones, receive incoming calls from some
"public" GW-providers and use "least-cost-routing" for
external calls (use cheapest GW depending on destination).
I believe that such a configuration is not that exotic,
but I could not find a usable sample...
Thank you
Dietmar Wessel
Hello,
I'm using ser 0.8.14 CVS and mediaproxy 1.2.0
(i have the same error in SER 0.8.12 lates CVS and mediaproxy 1.1.0)
I have seen that other users have a similar error like this one:
----------------------------------------------------------
[9882]: DBG: tcp_main_loop: dead child 20 (shutting down?)
[9764]: child process 9784 exited by a signal 11
[9764]: core was not generated
[9764]: INFO: terminating due to SIGCHLD
----------------------------------------------------------
method: <INVITE>
[9784]: uri: <sip:ê@sipy.callmyway.net>
[9784]: version: <SIP/2.0>
----------------------------------------------------------
please note that there is a strange user in the URI
This error always ocurs when i call to a ALIAS that have many users, for
example, 1111
ALIAS USER
1111 sip:123@domain.com
1111 sip:456@domain.com
1111 sip:789@domain.com
1111 sip:012@domain.com
Thanks
Ri Liang
