Hi,
I have got Serweb going but when I access admin/index.php or
user_interface/index.php I get the login prompt but when I try to login it
returns to the login prompt.
Looking back at the archives I see references to the domain or realm being
incorrect, what realm / domain does this refer to? What needs to match this
domain.
When I set up the MySQL users I had used the command export
SIP_DOMAIN="SIP-PROXY.LAB" to allow me to use serctl, I see this in the
output from select * from subscriber; run in mysql.
I have changed the line in the html/config.php to match how I had configured
my ser.cfg: (in the ser howto these commands were on separate lines, in my
config.php they were on one line, does this matter?)
/* your domain name */
$this->realm="boxb.lab"$this->domainname="boxb.lab"$this->default_domain=
ereg_replace( "(www\.|sip\.)?(.*)", "\\2",
$_SERVER['SERVER_NAME']);
/* initial nummerical alias for new subscriber -- don't
forget to
align your SER routing script to it !
*/
$this->first_alias_number=82000;
I'm sure this is a mismatch between the domain / realm somewhere but I am
not sure what I need to change, the SIP_DOMAIN? The server hostname? The
MYSQL users? or the aliases in ser.cfg?
Many thanks in advance...
Regards,
Steven Godfrey
Hi Ronald!
Please send your emails always Cc: to the mailing list.
Your ser.cfg is very strange, because your loading and configuring the
modules several times - this sould be done only once!
regarding the loose_route block - this is here:
>> # loose-route processing
>> if (loose_route()) {
--> setflag(9);
>> t_relay();
>> break;
>> };
try adding setflag(9) before t_relay!
klaus
Hi all,
I am using ser 0.8.12 with nathelper and rtpproxy. When I tried to make
a call between two clients which are behind the same NAT, everything
work fine. However, when I try to make a call between clients which are
behind different NATs, niether client can hear each other's audio. my
configuration file is shown in the end of this message. Can someone help
me, please?
Thanks,
Dan.
#
# $Id: nathelper.cfg,v 1.1.2.1 2003/11/24 14:47:18 janakj Exp $
#
# simple quick-start config script including nathelper support
# This default script includes nathelper support. To make it work
# you will also have to install Maxim's RTP proxy. The proxy is enforced
# if one of the parties is behind a NAT.
#
# If you have an endpoing in the public internet which is known to
# support symmetric RTP (Cisco PSTN gateway or voicemail, for example),
# then you don't have to force RTP proxy. If you don't want to enforce
# RTP proxy for some destinations than simply use t_relay() instead of
# route(1)
#
# Sections marked with !! Nathelper contain modifications for nathelper
#
# NOTE !! This config is EXPERIMENTAL !
#
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
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 ----------------------------------
",#
# $Id: nathelper.cfg,v 1.2 2003/04/15 20:35:29 jiri Exp $
#
# example script showing use of nathelper module
# (incomplete for sake of brevity)
#
# ----------- global configuration parameters ------------------------
# ------------------ 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/maxfwd.so"
loadmodule "/usr/local/lib/ser/modules/usrloc.so"
loadmodule "/usr/local/lib/ser/modules/registrar.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
# !! Nathelper
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 0)
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# !! Nathelper
modparam("registrar", "nat_flag", 6)
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{
# 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;
};
# !! Nathelper
# Special handling for NATed clients; first, NAT test is
# executed: it looks for via!=received and RFC1918 addresses
# in Contact (may fail if line-folding is used); also,
# the received test should, if completed, should check all
# vias for rpesence 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("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 enough to be symmetric. In some phones it takes a
configuration
# option. With Cisco 7960, it is called NAT_Enable=Yes, with
kphone it is
# called "symmetric media" and "symmetric signalling".
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
};
};
# 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
if (!method=="REGISTER") record_route();
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
break;
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
break;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
# if (!www_authorize("iptel.org", "subscriber")) {
# www_challenge("iptel.org", "0");
# break;
# };
save("location");
break;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
# !! Nathelper
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 client or server know to be behind a NAT, enable relay
if (isflagset(6)) {
force_rtp_proxy();
};
# NAT processing of replies; apply to all transactions (for example,
# re-INVITEs from public to private UA are hard to identify as
# NATed at the moment of request processing); look at replies
t_on_reply("1");
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
}
# !! Nathelper
onreply_route[1] {
# NATed 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();
};
}
Hi,
I have a problem. I am trying to redirect calls when user is not online to
PSTN gateway. I have problem changing the FROM field to the Original Request
user for accounting purpose. I am using the append_rpid_hf() but still
unable to go through. Please help or is there any other way to resolve this
problem. Thank in advance....:) Please help help.....
Regards,
Shirley
Hi all,
A domain has their own 3 or 4 digit internal dial plan.
When users dial out SIP/PSTN, callers should add a prefix plus their
internal number.
When PSTN calls come in, should I use prefix2domain
in pdt module to convert the PSTN number to a local
domain number.
Is it possible with ser/sems isdngw to do that?
Anybody could send me back an example file in order to help me to
understand
Regards
Harry
Hi all,
I have ser operating with a combination of X-lite and grandstream
phones. I am trying to get it it play nicely with Asterisk and have run
into a strange situation. Calling from ser to asterisk works fine but
when I try to call back from asterisk to ser I run into problems, the
sip device (behind NAT) rings but when I answer the call ser tries to
send the ACK back to the private IP and not the public IP (found using
STUN) that it was using up until that point.
I have the following debug from ngrep to illustrate the problem.
Offending SIP packet marked with ########s
Any ideas?
#
U 222.152.11.236:36595 -> xx.xx.xx.52:5060
SIP/2.0 100 trying.
Via: SIP/2.0/UDP xx.xx.xx.52;branch=z9hG4bKbe2d.2a520d95.0.
Via: SIP/2.0/UDP xx.xx.xx.51:5080;branch=z9hG4bK6484622b.
From: "Hamish Archer" <sip:100@xx.xx.xx.51:5080>;tag=as12db357a.
To: <sip:17778134000@proxy02.blahblah.xxx>.
Call-ID: 1738a1ee3d2310af4082784e4d0a6d89(a)xx.xx.xx.51.
CSeq: 102 INVITE.
User-Agent: Grandstream BT100 1.0.4.54.
Content-Length: 0.
.
#
U 222.152.11.236:36595 -> xx.xx.xx.52:5060
SIP/2.0 180 ringing.
Via: SIP/2.0/UDP xx.xx.xx.52;branch=z9hG4bKbe2d.2a520d95.0.
Via: SIP/2.0/UDP xx.xx.xx.51:5080;branch=z9hG4bK6484622b.
Record-Route: <sip:xx.xx.xx.52;ftag=as12db357a;lr=on>.
From: "Hamish Archer" <sip:100@xx.xx.xx.51:5080>;tag=as12db357a.
To: <sip:17778134000@proxy02.blahblah.xxx>;tag=760cdf6859149410.
Call-ID: 1738a1ee3d2310af4082784e4d0a6d89(a)xx.xx.xx.51.
CSeq: 102 INVITE.
User-Agent: Grandstream BT100 1.0.4.54.
Content-Length: 0.
.
#
U 222.152.11.236:36595 -> xx.xx.xx.52:5060
SIP/2.0 180 ringing.
Via: SIP/2.0/UDP xx.xx.xx.52;branch=z9hG4bKbe2d.2a520d95.0.
Via: SIP/2.0/UDP xx.xx.xx.51:5080;branch=z9hG4bK6484622b.
Record-Route: <sip:xx.xx.xx.52;ftag=as12db357a;lr=on>.
From: "Hamish Archer" <sip:100@xx.xx.xx.51:5080>;tag=as12db357a.
To: <sip:17778134000@proxy02.blahblah.xxx>;tag=efb3a3fe9b832ccc.
Call-ID: 1738a1ee3d2310af4082784e4d0a6d89(a)xx.xx.xx.51.
CSeq: 102 INVITE.
User-Agent: Grandstream BT100 1.0.4.54.
Content-Length: 0.
.
#
U 222.152.11.236:36595 -> xx.xx.xx.52:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP xx.xx.xx.52;branch=z9hG4bKbe2d.2a520d95.0.
Via: SIP/2.0/UDP xx.xx.xx.51:5080;branch=z9hG4bK6484622b.
Record-Route: <sip:xx.xx.xx.52;ftag=as12db357a;lr=on>.
From: "Hamish Archer" <sip:100@xx.xx.xx.51:5080>;tag=as12db357a.
To: <sip:17778134000@proxy02.blahblah.xxx>;tag=760cdf6859149410.
Call-ID: 1738a1ee3d2310af4082784e4d0a6d89(a)xx.xx.xx.51.
CSeq: 102 INVITE.
User-Agent: Grandstream BT100 1.0.4.54.
Contact: <sip:17778134000@192.168.1.158>.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Type: application/sdp.
Content-Length: 209.
.
v=0.
o=17778134000 8000 8000 IN IP4 192.168.1.158.
s=SIP Call.
c=IN IP4 222.152.11.236.
t=0 0.
m=audio 5004 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
#################################################
U xx.xx.xx.52:5060 -> 192.168.1.158:5060
ACK sip:17778134000@192.168.1.158 SIP/2.0.
Max-Forwards: 10.
Record-Route: <sip:xx.xx.xx.52;ftag=as12db357a;lr=on>.
Via: SIP/2.0/UDP xx.xx.xx.52;branch=0.
Via: SIP/2.0/UDP xx.xx.xx.51:5080;branch=z9hG4bK6484622b.
Route: <sip:17778134000@192.168.1.158>.
From: "Hamish Archer" <sip:100@xx.xx.xx.51:5080>;tag=as12db357a.
To: <sip:17778134000@proxy02.blahblah.xxx>;tag=760cdf6859149410.
Contact: <sip:100@xx.xx.xx.51:5080>.
Call-ID: 1738a1ee3d2310af4082784e4d0a6d89(a)xx.xx.xx.51.
CSeq: 102 ACK.
User-Agent: Asterisk PBX.
Content-Length: 0.
###################################################
#
U 222.152.11.236:36595 -> xx.xx.xx.52:5060
SIP/2.0 200 OK.
Via: SIP/2.0/UDP xx.xx.xx.52;branch=z9hG4bKbe2d.2a520d95.0.
Via: SIP/2.0/UDP xx.xx.xx.51:5080;branch=z9hG4bK6484622b.
Record-Route: <sip:xx.xx.xx.52;ftag=as12db357a;lr=on>.
From: "Hamish Archer" <sip:100@xx.xx.xx.51:5080>;tag=as12db357a.
To: <sip:17778134000@proxy02.blahblah.xxx>;tag=760cdf6859149410.
Call-ID: 1738a1ee3d2310af4082784e4d0a6d89(a)xx.xx.xx.51.
CSeq: 102 INVITE.
User-Agent: Grandstream BT100 1.0.4.54.
Contact: <sip:17778134000@192.168.1.158>.
Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE.
Content-Type: application/sdp.
Content-Length: 210.
.
v=0.
o=17778134000 8000 8000 IN IP4 192.168.1.158.
s=SIP Call.
c=IN IP4 222.152.11.236.
t=0 0.
m=audio 35337 RTP/AVP 0 101.
a=rtpmap:0 PCMU/8000.
a=ptime:20.
a=rtpmap:101 telephone-event/8000.
a=fmtp:101 0-11.
Greetings,
I am wondering if there is any writing on the use of wildcards and what they
do? Basically, I would like to know how I can wildcard something like
12125551212 to 1*, for routing purposes. Any help will be highly
appreciated.
Masoud
_________________________________________________________________
Check out the coupons and bargains on MSN Offers! http://youroffers.msn.com
Hi,
This message is original asked in sems list, but I
haven't got an answer yet. Does anyone here know
answer?
Thanks,
Richard
--- Richard <mypop3mail(a)yahoo.com> wrote:
> Date: Wed, 5 May 2004 20:05:20 -0700 (PDT)
> From: Richard <mypop3mail(a)yahoo.com>
> To: sems(a)lists.iptel.org
> Subject: [Sems] ivr.record and DTMF tone
>
> Hi,
>
> In sems, is there a way to record the message and
also detect
> DTMF tones? For example, after the recording, user
> can
> press # to stop or * to canel. Then he can play and
re-record it
> if
> necessary. This would be very useful to record
> personalize greetings.
>
> Thanks,
> Richard
>
__________________________________
Do you Yahoo!?
Win a $20,000 Career Makeover at Yahoo! HotJobs
http://hotjobs.sweepstakes.yahoo.com/careermakeover
