I am running a version of SER compiled from ser-0.8.12_src. I tried to load ser-0.8.14 but I could not get it to recognize the RTP proxy. Here is a copy of my ser.cfg. Any help would be greatly appreciated:
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
listen=64.77.16.75
listen=127.0.0.1
port=5060
children=16
check_via=yes # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
fifo="/tmp/ser_fifo"
# syn_branch - Shall the server use stateful synonym branches? It is
# faster but not reboot-safe. Default is yes.
syn_branch=yes
memlog=3
# hostname matching an alias will satisfy the condition uri==myself".
alias=ari.com
alias=64.77.16.75
# sip_warning - Should replies include extensive warnings? By default
# yes, it is good for trouble-shooting.
sip_warning=yes
# server_signature - Should locally-generated messages include server's
# signature? By default yes, it is good for trouble-shooting.
server_signature=yes
# 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
# mhomed -- enable calculation of outbound interface; useful on
# multihomed servers.
mhomed=0
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
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/acc.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
loadmodule "/usr/local/lib/ser/modules/uri.so"
loadmodule "/usr/local/lib/ser/modules/group.so"
loadmodule "/usr/local/lib/ser/modules/msilo.so"
loadmodule "/usr/local/lib/ser/modules/enum.so"
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters ---------------
modparam("usrloc|group|msilo|uri", "db_url","mysql://ser:heslo@localhost/ser")
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# -- tm params --
modparam("tm", "fr_timer", 20 )
modparam("tm", "fr_inv_timer", 90 )
modparam("tm", "wt_timer", 20 )
# -- msilo params
modparam("msilo", "registrar", "sip:registrar@ari.com")
# -- enum params --
#
modparam("enum", "domain_suffix", "e164.arpa.")
# -- nathelper params --
#
modparam("registrar", "nat_flag", 1)
modparam("nathelper", "natping_interval", 30) # Ping interval 30 s
#modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT
# -- acc params --
# that is the flag for which we will account -- don't forget to
#modparam("acc", "db_flag", 1 )
#modparam("acc", "db_missed_flag", 3 )
# ------------------------- request routing logic -------------------
# main routing logic
route {
/* ********* ROUTINE CHECKS ********************************** */
# filter too old messages
if (!mf_process_maxfwd_header("10")) {
log(1, "LOG: Too many hops\n");
sl_send_reply("483","Alas Too Many Hops");
break;
};
if ( msg:len >= max_len ) {
sl_send_reply("513", "Wow -- Message too large");
break;
};
if (search("^(Contact|m): .*(a)(192\.168\.|10\.|172\.16)")) {
log(1, "src address different than via header->NAT detected\n");
log(1, "force_rport and fix_nated_contact and setflag(5)\n");
#try NAT traversal, works only if the client is symmetrical
force_rport();
fix_nated_contact();
append_hf("P-hint: fixed NAT contact for request\r\n");
# flag 5 indicates that incoming request is from NATed client
setflag(5);
};
/* ********* RR ********************************** */
# to be safe, record route everything; UAs may use different
# transport protocols and need to have SER in path
record_route();
setflag(1);
# if route forces us to forward to some explicit destination,
# do so; check however first that a cheater didn't preload
# a gateway destination to bypass PSTN ACLs
if (loose_route()) {
log(1, "Loose Route\n");
if ((uri=~"sip:[+0-9]+@64\.77\.16\.126")) {
# it is gateway -- proceed to ACLs
route(3);
break;
};
# route HF determined next hop; forward there
append_hf("P-hint: rr-enforced\r\n");
t_relay();
break;
};
/* ********* check for requests targeted out of our domain... *******
*/
# sign of our domain: there is '@' (username) or : (nothing) in
# front of our domain name ; ('.' is not there -- we handle all
# xxx.iptel.org as outbound hosts);if none of these cases matches,
# proceed with processing of outbound requests in route[2]
if (!(uri==myself)) {
log(1, "Not myself\n");
route(2);
break;
};
/* ************ requests for our domain ********** */
/* now, the request is for sure for our domain */
# registers always MUST be authenticated to
# avoid stealing incoming calls
if (method=="REGISTER") {
# Make sure that user's dont register infinite loops
if (search("^(Contact|m): .*(a)(ari\.com|64\.77\.16\.75)")) {
log(1, "LOG: alert: someone trying to set aor==contact\n");
sl_send_reply("476", "No Server Address in Contacts Allowed" );
break;
};
if (search("^(Contact|m): .*64\.77\.16\.126")) {
log(1, "LOG: alert: protected contacts\n");
sl_send_reply("476", "No Server Address in Contacts Allowed" );
break;
};
if (!www_authorize("ari.com" /* realm */,"subscriber" /* table name */ )) {
# challenge if none or invalid credentials
www_challenge( "ari.com" /* realm */,"0" /* no qop -- some phones can't deal with it */);
break;
};
# prohibit attempts to grab someone else's To address
# using valid credentials;
if (!check_to()) {
log(1, "LOG: To Cheating attempt\n");
sl_send_reply("403", "That is ugly -- use To=id in REGISTERs");
break;
};
if (isflagset(5)) {
#register from nated client, save nat_flag=6
#in location table
setflag(6);
};
# it is an authenticated request, update Contact database now
if (!(save("location"))) {
sl_reply_error();
};
m_dump();
break;
};
# NAT Traversal Code
if ((isflagset(5)) || (isflagset(6))) {
log(1, "at least one of the participants is NATed->record_route\n");
record_route();
log(1, " -->setting up reply processing ->onreply_route[1]");
t_on_reply("1");
if (method=="INVITE") {
log(1, " INVITE request-->force_rtp_proxy, set NATED-INVITE flag(7)");
force_rtp_proxy();
append_hf("P-hint: request forced to rtp proxy\r\n");
setflag(7);
};
};
# native SIP destinations are handled using our USRLOC DB
if (lookup("location")) {
log(1, "Native SIP destination detected\n");
append_hf("P-hint: USRLOC\r\n");
if (!t_relay()) {
sl_reply_error();
break;
};
};
# now check if it's about PSTN destinations through our gateway;
if (uri=~"sip:\+?[0-9][0-9]*@.*") {
log(1, "PSTN destination to be used\n");
route(3);
break;
};
} # route
#----------------- PSTN ----------------------------------------------
# logic for calls to the PSTN
route[3] {
# turn accounting on
setflag(1);
/* require all who call PSTN to be members of the "int" group;
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; exempt Cisco gateway from authentication by IP address --it does not
support digest
*/
if (src_ip==64.77.16.126) {
log(1, "From ARI Gateway");
}
else {
if ((method=="INVITE") && (!(src_ip==64.77.16.126))) {
log(1, "LOG: need to auth PSTN invite");
if (!proxy_authorize( "ari.com" /* realm */,"subscriber" /* table name */)) {
proxy_challenge( "ari.com" /* realm */, "0" /* no qop */ );
break;
};
# let's check from=id ... avoids accounting confusion
if (method=="INVITE" & !check_from()) {
log(1, "LOG: From Cheating attempt\n");
sl_send_reply("403", "That is ugly -- use From=id next time (gw)");
break;
};
if(!is_user_in("credentials", "int")) {
sl_send_reply("403", "NO PSTN Privileges...");
break;
};
consume_credentials();
}; # INVITE to authorized PSTN
}
# if you have passed through all the checks, let your call go to GW!
log(1, "Rewriting Host and port");
rewritehostport("64.77.16.126:5060");
# snom conditioner
if (method=="INVITE" && search("User-Agent: snom")) {
replace("100rel, ", "");
};
t_relay_to_udp ("64.77.16.126","5060");
t_relay_to_tcp ("64.77.16.126","5060");
break;
}
# all incoming replies for t_onrepli-ed transactions enter here
onreply_route[1] {
log(1, "-------------------------------------------\n");
log(1, "onreply_route[1] entered\n");
if (isflagset(6)) {
log(1, "transaction was sent to a NATED client -> fix nated contact\n");
fix_nated_contact();
append_hf("P-hint: fixed NAT contact for response\r\n");
}
if ( (status=~"100") ) {
log(1, "status 100 received\n");
};
if ( (status=~"180") ) {
log(1, "status 180 received\n");
};
if ( (status=~"202") ) {
log(1, "status 202 received\n");
};
if ( (status=~"200" || status=~"183") ) {
log(1, "status 2xx or 183");
if ( isflagset(7) ) {
log(1, "marked(7) as NATED-INVITE -> force_rtp_proxy \n");
force_rtp_proxy();
append_hf("P-hint: response forced to rtp proxy\r\n");
};
};
}
#------------------- OUTBOUND ----------------------------------------
# routing logic for outbound requests targeted out of our domain
# (keep in mind messages to our users can end up here too: for example,
# an INVITE may be UsrLoc-ed, then the other party uses outbound
# proxy with r-uri=the usr_loced addredd (typically IP))
route[2] {
append_hf("P-hint: OUTBOUND\r\n");
t_relay();
}
#------- ALIASED OUTBOUND --------------------------------------------
# routing logic for inbound requests aliased outbound; unlike
# with real outbound requests we do not force authentication
# as these calls are server by our server and we do not want
# to disqualify unathenticated request originatiors from other
# domains
route[5] {
append_hf("P-hint: ALIASED-OUTBOUND\r\n");
t_relay();
}
> -----Original Message-----
> From: Brian J. Rathman
> Sent: Sunday, September 26, 2004 10:31 AM
> To: 'serusers(a)lists.iptel.org'
> Subject: One Way audio cut through problem
>
> I am noticing that when I try and make calls through SER using a Cisco ATA186 and terminating through a T1 card on a Cisco Router that I get one way audio until I speak into the phone connected to the ATA. The called party can hear me, but I do not hear anything from them until I say something, then the audio cuts through. The issue with this is that I am trying to setup these devices to call into an IVR on my end and if the calling party must speak first, then they are not going to hear the IVR. Has anyone else seen something like this before? I am running a version of SER compiled from the
I am noticing that when I try and make calls through SER using a Cisco ATA186 and terminating through a T1 card on a Cisco Router that I get one way audio until I speak into the phone connected to the ATA. The called party can hear me, but I do not hear anything from them until I say something, then the audio cuts through. The issue with this is that I am trying to setup these devices to call into an IVR on my end and if the calling party must speak first, then they are not going to hear the IVR. Has anyone else seen something like this before? I am running a version of SER compiled from the
I am trying to get SER and rtpproxy to run together on my OpenBSD 3.5
firewall. My intention is to have SER act as a SIP proxy for incoming SIP
calls that reach me by IP address or domain name. I also have an Asterisk
PBX behind my firewall. My problem is that SER can't find a running RTP
proxy and I am not sure why. I run rtpproxy without any switches. SER is
run as a non-root user and in an chroot directory. My ser.cfg is below:
#
# $Id: ser.cfg,v 1.12 2002/10/21 02:40:06 jiri Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
fork=no
log_stderror=yes # (cmd line: -E)
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=2
fifo="/tmp/ser_fifo"
alias="my.domainname.com"
listen=204.118.177.213
#listen=192.168.3.1
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
#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/nathelper.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
#loadmodule "/usr/local/lib/ser/modules/auth.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
#modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
#modparam("auth", "secret", "alsdkhglaksdhfkloiwr")
#modparam("auth", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
#modparam("auth", "password_column", "password")
#modparam("rr", "enable_full_lr", 1)
modparam("registrar", "nat_flag", 6)
#nathelper
modparam("nathelper", "natping_interval", 30) # Ping interval 30 s
modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT
# ------------------------- request routing logic -------------------
# main routing logic
route{
# compulsory processing of Route header fields and adding RR
loose_route();
# ATA's are symmetric but don't advertise it -- force use of rport
if (search("User-Agent: Cisco ATA.*")) {
setflag(1); # remember this is ATA
force_rport();
fix_nated_contact();
};
/* registration (uses rewritten contacts) */
if (method=="REGISTER") {
save("my.domainname.com");
break;
};
if (method=="INVITE") {
record_route();
if (isflagset(1)) { # ATA ?
force_rtp_proxy();
};
/* set up reply processing */
t_on_reply("1");
};
if (method == "INVITE" || method == "CANCEL") {
if (!lookup("my.domainname.com")) {
sl_send_reply("404", "Not Found");
break;
};
};
/* set up reply processing and forward statefuly */
t_relay();
}
# all incoming replies for t_onrepli-ed transactions enter here
onreply_route[1] {
if (status=~"2[0-9][0-9]" && search("Server: Cisco ATA.*"))
fix_nated_contact();
force_rtp_proxy();
}
Greetings,
One of my user names keep failing on registration and I am getting these
messages in my syslog. Any ideas here to look for the problem?
Thanks
MS
Sep 21 22:52:13 sipsvr2 /usr/local/sbin/ser[32122]: ERROR: parse_uri: bad
port in uri (error at char in state 3) parsed: <sip:1:xxx.xxx.xxx.xxx>(18)
/<sip:1:xxx.xxx.xxx.xxx> (18)
Sep 21 22:52:13 sipsvr2 /usr/local/sbin/ser[32122]: get_username(): Error
while parsing R-URI
Sep 21 22:52:13 sipsvr2 /usr/local/sbin/ser[32122]: insert_RR(): Error while
extracting username
Sep 21 22:52:13 sipsvr2 /usr/local/sbin/ser[32122]: record_route(): Error
while inserting Record-Route line
Sep 21 22:52:13 sipsvr2 /usr/local/sbin/ser[32122]: ERROR: parse_uri: bad
port in uri (error at char in state 3) parsed: <sip:1:xxx.xxx.xxx.xxx>(18)
/<sip:1:xxx.xxx.xxx.xxx> (18)
_________________________________________________________________
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Hello All.
I finally have my lookup("aliases") working thanks to Zeus Ng. Now that users
can have aliases on my ser proxy I have a question regarding voicemail. I'm
hoping someone can give me an idea of how best to address this issue.
I use ser for all SIP stuff and Asterisk for voicemail only. I have ser and
asterisk working nicely together.
A typical scenerio would be like this. I have a ser user named
1000(a)mycompany.com with a PSTN alias 4075551234. In addition this user has an
Asterisk mailbox configured as 1000(a)mycompany.com
When someone dials sip:1000@mycompany.com and there is no answer they will get
sent to voicemail, which then Asterisk will say "The user at extension
one-zero-zero-zero is unavailable. Please leave your message after the tone..."
But what happens when a caller dials 4075551234(a)mycompany.com and gets routed
to voicemail? 4075551234 doesn't exist in asterisk. If I use lookup("aliases")
in my ser.cfg routing plan can I revert back to the original
sip:1000@mycompany.com before sending the caller off to the asterisk voicemail?
If so, how?
Regards,
Paul
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I am trying to get SER and rtpproxy to run
together on my OpenBSD 3.5 firewall. My intention is
to have SER 0.8.14 act as a SIP proxy for incoming SIP
calls that reach me by IP address or domain name. I
also have an Asterisk PBX behind my firewall. My
problem is that SER can't find a running RTP proxy and
I am not sure why. I run rtpproxy without any
switches. SER is run as a non-root user and my
ser.cfg is below:
#
# $Id: ser.cfg,v 1.12 2002/10/21 02:40:06 jiri Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters
------------------------
debug=3 # debug level (cmd line: -dddddddddd)
fork=no
log_stderror=yes # (cmd line: -E)
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=2
fifo="/tmp/ser_fifo"
alias="my.domainname.com"
listen=204.118.177.213
#listen=192.168.3.1
# ------------------ module loading
----------------------------------
# Uncomment this if you want to use SQL database
#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/nathelper.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
#loadmodule "/usr/local/lib/ser/modules/auth.so"
# ----------------- setting module-specific parameters
---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
#modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
#modparam("auth", "secret", "alsdkhglaksdhfkloiwr")
#modparam("auth", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which
true in this config),
# uncomment also the following parameter)
#
#modparam("auth", "password_column", "password")
#modparam("rr", "enable_full_lr", 1)
modparam("registrar", "nat_flag", 6)
#nathelper
modparam("nathelper", "natping_interval", 30) # Ping
interval 30 s
modparam("nathelper", "ping_nated_only", 1) # Ping
only clients behind NAT
# ------------------------- request routing logic
-------------------
# main routing logic
route{
# compulsory processing of Route header fields and
adding RR
loose_route();
# ATA's are symmetric but don't advertise it -- force
use of rport
if (search("User-Agent: Cisco ATA.*")) {
setflag(1); # remember this is ATA
force_rport();
fix_nated_contact();
};
/* registration (uses rewritten contacts) */
if (method=="REGISTER") {
save("my.domainname.com");
break;
};
if (method=="INVITE") {
record_route();
if (isflagset(1)) { # ATA ?
force_rtp_proxy();
};
/* set up reply processing */
t_on_reply("1");
};
if (method == "INVITE" || method == "CANCEL") {
if (!lookup("my.domainname.com")) {
sl_send_reply("404", "Not Found");
break;
};
};
/* set up reply processing and forward statefuly */
t_relay();
}
# all incoming replies for t_onrepli-ed transactions
enter here
onreply_route[1] {
if (status=~"2[0-9][0-9]" && search("Server: Cisco
ATA.*"))
fix_nated_contact();
force_rtp_proxy();
}
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HI ALL;
Can anybody explain what exactly is "statefull proxy"???????
I read the ser related documents about that topic but I have not understood very well
Warmest Regards
mohammad
Don't know much about the asterisk voicemail capability. However, does the voicemail has "aliases" capability in terms it could take multiple phone number for the same mailbox?
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Java Rockx
Sent: September 25, 2004 9:53 PM
To: ser users
Subject: [Serusers] SER + Asterisk Voicemail -- Seeking Advise
Hello All.
I finally have my lookup("aliases") working thanks to Zeus Ng. Now that users
can have aliases on my ser proxy I have a question regarding voicemail. I'm
hoping someone can give me an idea of how best to address this issue.
I use ser for all SIP stuff and Asterisk for voicemail only. I have ser and
asterisk working nicely together.
A typical scenerio would be like this. I have a ser user named
1000(a)mycompany.com with a PSTN alias 4075551234. In addition this user has an
Asterisk mailbox configured as 1000(a)mycompany.com
When someone dials sip:1000@mycompany.com and there is no answer they will get
sent to voicemail, which then Asterisk will say "The user at extension
one-zero-zero-zero is unavailable. Please leave your message after the tone..."
But what happens when a caller dials 4075551234(a)mycompany.com and gets routed
to voicemail? 4075551234 doesn't exist in asterisk. If I use lookup("aliases")
in my ser.cfg routing plan can I revert back to the original
sip:1000@mycompany.com before sending the caller off to the asterisk voicemail?
If so, how?
Regards,
Paul
__________________________________
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_______________________________________________
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serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
HI ALL;
I use ser as my sip proxy/registrar and asterisk as ip-pbx server like voicemail
In the case on unavalibility the call is redirected to asterisk sucessfully and prompts to record the message
I allocate number 1000 for accessing recorded messages but when I try to listen to new messages nothing can be heard while when I directly register to
astersik everything is ok.
appreciate any suggestion
Warmest Regards
mohammad
