I met a problem in register new user using SERWEB. When I click the register button after i fill out the register item, the page return back itself, not the finish page.
Anyone know why?
Hi,
In traditional PBX, there is a feature to indicate if
another user (more precisely his phone) is being used
or not. This would help before transfering a call.
Is there a way to support this in ser? If record-route
is used, ser knows that users are on the phone bewteen
ACK and BYE methods. Is possible somehow tell other
phones about the status?
Thanks,
Richard
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hello friends ,
i want to establish a session as described
192.*.*.* ---> dhcpserver (202.*.*.*) -->
ser(202.*.*.*) --> 202.*.*.*
my ser.cfg file is
please guide me
with regards
rama kanth
*******************************************
#
# $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=yes # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
fork=no
log_stderror=yes
*/
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
----------------------------------
# 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/textops.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"
# !! Nathelper
loadmodule "/usr/local/lib/ser/modules/nathelper.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_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)
# !! 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(1,"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();
};
}
*******************************************
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Hello guys
Could anyone register an ATA 186 with sip version v.3.1.0 to ser using
authentication?
I am running ser v0.8.12 on a RH9 machine. All equipments has public IPs.
I configured the following parameters in the ATA:
UID0:1000
PWD0:(the pass configured in ser)
GkOrProxy:(ip address of ser)
UseLoginID:1
LoginID0:1000@myrealm.com (where myrealm.com is the realm configured in ser)
SIPRegOn:1
When ATA sends a REGISTER ser responds with 483 to many hops. So ATA does
not register to ser.
I tried the same user with Windows Messenger 5.0 and ser responds with 401
unauthorized user and the nuance, so MSN sends the new REGISTER with all the
information. Finally MSN does register to ser.
Any help on this matter is appreciated.
Pablo.
hi,
When I load X-Lite behind a Single Mapped Port Symmetric NAT Firewall and
make a call to it, the X-Lite rings and I can hear voice in both directions.
The problem is when I leave the X-Lite sitting idle for over 1 minute, the
inbound calls to client start to time out. Outbound calls from client work
no problem. Im not receiving anything on the X-Lite debugs, and when i debug
the SER, I see repeatedly sending Invites with no response. Below is the
debug...64.72.107.2 is my Gateway, .4 is my SER, and .1 is my firewall NAT
address where my client X-Lite sits behind. If i close the X-Lite and
open it back up, I am able to make calls to it again until I leave it sit
idle...Any ideas? Could something be wrong with my nathelper?
Failed Attempt...
U 64.72.107.2:55580 -> 64.72.107.4:5060
INVITE sip:7133154693@64.72.107.4;user=phone;phone-context=n ational
SIP/2.0..Via: SIP/2.0/UDP 64.72.107.2:5060..From: " 2814497314"
<sip:2814497314@64.72.107.2>..To: <sip:713315469
3(a)64.72.107.4;user=phone;phone-context=national>..Date: Mon, 24 May 2004
17:00:37 GMT..Call-ID: B67CA151-ACDA11D8-A5E08B
34-F8DB44FC@64.72.107.2..Cisco-Guid: 3061621073-2899972568-2
782825268-4175119612..User-Agent: Cisco VoIP Gateway/ IOS 12 .x/ SIP
enabled..CSeq: 101 INVITE..Max-Forwards: 6..Timestam p:
1085418037..Contact: <sip:2814497314@64.72.107.2:5060;use
r=phone>..Expires: 180..Content-Type: application/sdp..Conte nt-Length:
132....v=0..o=CiscoSystemsSIP-GW-UserAgent 6230 2 341 IN IP4
64.72.107.2..s=SIP Call..c=IN IP4 64.72.107.2..t= 0 0..m=audio 20478
RTP/AVP 8..
#
U 64.72.107.4:5060 -> 64.72.107.2:5060
SIP/2.0 100 trying -- your call is important to us..Via: SIP /2.0/UDP
64.72.107.2:5060..From: "2814497314" <sip:28144973 14(a)64.72.107.2>..To:
<sip:7133154693@64.72.107.4;user=phone; phone-context=national>..Call-ID:
B67CA151-ACDA11D8-A5E08B34 -F8DB44FC@64.72.107.2..CSeq: 101
INVITE..Server: Sip EXpress router (0.8.12-tcp_nonb
(i386/linux))..Content-Length: 0..W arning: 392 64.72.107.4:5060 "Noisy
feedback tells: pid=159 6 req_src_ip=64.72.107.2 req_src_port=55580
in_uri=sip:71331 54693(a)64.72.107.4;user=phone;phone-context=national
out_uri= sip:7133154693@64.72.107.1:55246 via_cnt==1"....
#
U 64.72.107.4:5060 -> 64.72.107.1:55246
INVITE sip:7133154693@64.72.107.1:55246 SIP/2.0..Via: SIP/2. 0/UDP
64.72.107.4;branch=z9hG4bKeb02.9f521926.0..Via: SIP/2. 0/UDP
64.72.107.2:5060..From: "2814497314" <sip:2814497314@ 64.72.107.2>..To:
<sip:7133154693@64.72.107.4;user=phone;pho ne-context=national>..Date:
Mon, 24 May 2004 17:00:37 GMT..C all-ID:
B67CA151-ACDA11D8-A5E08B34-F8DB44FC@64.72.107.2..Cis co-Guid:
3061621073-2899972568-2782825268-4175119612..User-A gent: Cisco VoIP
Gateway/ IOS 12.x/ SIP enabled..CSeq: 101 I NVITE..Max-Forwards:
5..Timestamp: 1085418037..Contact: <sip
:2814497314@64.72.107.2:5060;user=phone>..Expires: 180..Cont ent-Type:
application/sdp..Content-Length: 132..P-hint: USRL
OC....v=0..o=CiscoSystemsSIP-GW-UserAgent 6230 2341 IN IP4 6
4.72.107.2..s=SIP Call..c=IN IP4 64.72.107.2..t=0 0..m=audio 20478
RTP/AVP 8..
#
U 64.72.107.4:5060 -> 64.72.107.1:55246
INVITE sip:7133154693@64.72.107.1:55246 SIP/2.0..Via: SIP/2. 0/UDP
64.72.107.4;branch=z9hG4bKeb02.9f521926.0..Via: SIP/2. 0/UDP
64.72.107.2:5060..From: "2814497314" <sip:2814497314@ 64.72.107.2>..To:
<sip:7133154693@64.72.107.4;user=phone;pho ne-context=national>..Date:
Mon, 24 May 2004 17:00:37 GMT..C all-ID:
B67CA151-ACDA11D8-A5E08B34-F8DB44FC@64.72.107.2..Cis co-Guid:
3061621073-2899972568-2782825268-4175119612..User-A gent: Cisco VoIP
Gateway/ IOS 12.x/ SIP enabled..CSeq: 101 I NVITE..Max-Forwards:
5..Timestamp: 1085418037..Contact: <sip
:2814497314@64.72.107.2:5060;user=phone>..Expires: 180..Cont ent-Type:
application/sdp..Content-Length: 132..P-hint: USRL
OC....v=0..o=CiscoSystemsSIP-GW-UserAgent 6230 2341 IN IP4 6
4.72.107.2..s=SIP Call..c=IN IP4 64.72.107.2..t=0 0..m=audio 20478
RTP/AVP 8..
#
U 64.72.107.4:5060 -> 64.72.107.1:55246
INVITE sip:7133154693@64.72.107.1:55246 SIP/2.0..Via: SIP/2. 0/UDP
64.72.107.4;branch=z9hG4bKeb02.9f521926.0..Via: SIP/2. 0/UDP
64.72.107.2:5060..From: "2814497314" <sip:2814497314@ 64.72.107.2>..To:
<sip:7133154693@64.72.107.4;user=phone;pho ne-context=national>..Date:
Mon, 24 May 2004 17:00:37 GMT..C all-ID:
B67CA151-ACDA11D8-A5E08B34-F8DB44FC@64.72.107.2..Cis co-Guid:
3061621073-2899972568-2782825268-4175119612..User-A gent: Cisco VoIP
Gateway/ IOS 12.x/ SIP enabled..CSeq: 101 I NVITE..Max-Forwards:
5..Timestamp: 1085418037..Contact: <sip
:2814497314@64.72.107.2:5060;user=phone>..Expires: 180..Cont ent-Type:
application/sdp..Content-Length: 132..P-hint: USRL
OC....v=0..o=CiscoSystemsSIP-GW-UserAgent 6230 2341 IN IP4 6
4.72.107.2..s=SIP Call..c=IN IP4 64.72.107.2..t=0 0..m=audio 20478
RTP/AVP 8..
-----------------------
Harold Workman
CCNA, CCNP
Cytel Communications
hworkman(a)cytelcom.com
Ph. 281-449-4000 x3098
Hello list,
I am new to SER and have a question about NAT penetration in general.
All parts of SER and mediaproxy are running fine but I am still am
struggling a bit with NAT penetration.
According the readme for Mediaproxy ser module any endpoint that is behind
NAT needs to configured with outbound proxy enabled. Are there any examples
available how to do this when using mediaproxy.
This way the nat penetration is not likely to be really transparent but
needs a configuration issue on the EP.
Can anybody confirm me this please or hint me to reading material on this
subject.
Kindest Regards,
Tjapko Smits
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Hi all,
sorry for bothering you with my rtpproxy problem again.
I now looked for a potential error output of rtpproxy and found:
"rtpproxy: command syntax error" whenever force_rtp_proxy is called.
In addition I found a comment from Jan Janak in the mail-archive on a
similar problem (ser together with rtpproxy): "My guess is that you are
using incompatible versions of rtpproxy and ser, as far as I know the
rtpproxy from cvs can be used with unstable only."
That's obviously my case. I am using stable ser 0.8.12.
Therefore I have to switch to unstable ser and try again.
Greetings
Franz
Hi all,
I am trying to read the import MySQLdb into the ivr.py script to read values
from the sql database but I got error in missing modules. Please help. What
should I do to be able to read values from the database in the ivr.py script
for development purpose. Please help......Thanks in advance.
Regards,
Shirley
I'm using SER and SERWEB both from cvs, because i thought they were
compatible!
anyway with SER from cvs do i have to use stable SERWEB rel_0_8_12?
or i must change SER too!?
Stefano
>You are using incompatible versions of SER and serweb, use -r rel_0_8_12
>when checking out serweb for stable ser.
>
> Jan.
>
>On 20-05 11:51, Stefano wrote:
>> Hi,
>>
>> I have a problem about new user subscription from SERWEB; SERWEB correctly
>> send the confirmation email to users, but when the user check the link to
>> finalize the subscription it fails and the error "400 ul_add: flags
>>expected"
>> appears.
>>
>> What does it means, is there some "flag" to set to allow user's
>>subscribing?
>>
>> Thanks for your help.
>>
>> Stefano
>>