Okay... I think with Steve's help, I've been getting somewhere with my avpops
stuff, but I have a couple of questions.
1) What is the format of the $from and $to headers as far as avpops is
concerned? How can I find that out?
2) How does avp_print() work (i.e. to where does it print) ? Is there any way
I can get intermediate values of things so I can see just what it is my little
scripts are doing?
N.
confirm 83b7fb1612811c64f85f47a6046dd84dea286d15
Dear........
I want to networking among three SIP servers.
But now I am trying using two servers likes three servers.
That is,
- source : computer A
ip address ; 192.168.178.194
sipuri : sip:mj@192.168.178.194
- via : computer B
ip address : 192.168.178.193
sipuri : sip:hj@192.168.178.193
- destination : computer A
ip address :192.168.178.194
sipuri : sip:mj@192.168.178.194
And I am using sipp to test (I modified the "via" part based on original
uac module).
=====================================================
<< computer B's test.cfg >>
#
# $Id: welcome.cfg,v 1.3 2003/11/03 13:03:56 janakj Exp $
#
# welcome message for new subscribers; based on exec
#
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
# must be yes since REGISTER processing causes an INVITE to be sent,
# which needs to be processed by another process
fork=yes
children=4
alias="kims.myserver.sampro.cse"
alias="myserver.sampro.cse"
alias="192.168.178.193"
# debugging
log_stderror=yes # (cmd line: -E)
mhomed=yes
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/exec.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 2)
#modparam("usrloc", "db_url", "mysql://ser:heslo@192.168.2.16/ser" )
modparam("usrloc", "db_url", "mysql://root:heslo@localhost/ser")
# ------------------------- request routing logic -------------------
# main routing logic
route{
log(1, "\n\n");
log(1, "START!!!\n");
# 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>=204822) {
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
# if (method=="INVITE") record_route(); # 1=loose routing
if(method=="INVITE"){
log(1, "receive INVITE\n");
record_route();
sl_send_reply("200", "ok");
};
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
# if(method=="ACK"){
# log(1, "receive ACK\n");
# };
log(1, "record-route processing completed\n");
# 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(uri=="sip:mijin@sam.cse"){
# forward(192.168.178.194, 5060);
# };
if (uri=~"sip:192.168.178.193"){
if (method=="REGISTER") {
log(1, "receive REGISTER\n");
# save location before initiating welcome
save("location");
log(1, "save 성공\n");
record_route();
sl_send_reply("200", "ok");
# welcome message
log(1, "route3으로 ��
\n");
route(3);
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
log(1, "Usrloc DB�서 찾
� 수 없�
\n");
break;
};
};
if(method=="BYE"){
log(1, "receive BYE\n");
record_route();
sl_send_reply("200", "ok");
break;
};
t_relay();
}
# welcome message -- if a REGISTER succeeded, look if it is the first-time;
# if so, initiate click-to-dial transaction to set up call to an
announcement
# server; edit the config values first to make it work
route[3] {
if (!exec_msg('
# config:
# --announcement server URI
ANS="sip:hjkim@myserver.sampro.cse"
# --SIP domain
# DOMAIN="192.168.178.193"
DOMAIN="192.168.178.193"
# ctd
CTD=${HOME}/sip_router/examples/ctd.sh
# ------------------------------------
# check if first time ...
SIP_UID=`echo $SIP_HF_TO | sed -e
"s/^.*sip:\([a-zA-Z0-9_\.]*\)@.*$/\1/g"`
QUERY="select flag from subscriber
where username=\"$SIP_UID\";
update subscriber set flag=\"x\"
where username=\"$SIP_UID\" ";
mysql -Bsuser -pheslo -e "$QUERY" ser| grep "^x$" > /dev/null
# ... if so, c-t-d to announcement server
if [ "$?" -ne 0 ] ; then
# flag was not set to x yet -- first-time registration;
# initiate a call from telephone of the user to an announcement server
$CTD "sip:$SIP_UID@$DOMAIN" "$ANS" > /dev/null 2>&1
fi
')) {
log(1, "welcome exec failed\n");
}
log(1, "-------route3 � \n");
}
As you see, I added
modparam("rr", "enable_full_lr", no)
if(uri=="sip:mijin@192.168.178.194"){
forward(192.168.178.194, 5060);
append_branch("sip:hjkim@192.168.178.193:5060"); log(1, "next
hop...\n");
break;
};
but, I received error message in computer B(via):
8(12253) ERROR: forward_reply: no 2nd via found in reply
What does it mean?
What can I do for doing what I want ???
Is there any problem in my test.cfg file ?????
I would like to know in detail .
please tell me what to do...............................
sipp module can't support that????
And Please tell me about forward funtion.
I don't know whether I understand forward function usage well.......
sorry ............
Song
_________________________________________________________________
고.. 감.. 도.. 사.. 랑.. 만.. 들.. 기.. MSN 러브
http://www.msn.co.kr/love/
I'm having some trouble trying to integrate both Radius authentication
and Radius AVP storage.
I'm using freeradius and there doesn't seem to be much documentation
regarding configuration semantics and such.
Could anyone share some information on how they were able to use a
Radius server to authenticate requests and (with avp_radius) as an AVP
retrieval mechanism?
I'm uncertain as to how I can stack these two uses of data on the
freeradius server
tavis
confirm 83b7fb1612811c64f85f47a6046dd84dea286d15
Dear........
I want to networking among three SIP servers.
But now I am trying using two servers likes three servers.
That is,
- source : computer A
ip address ; 192.168.178.194
sipuri : sip:mj@192.168.178.194
- via : computer B
ip address : 192.168.178.193
sipuri : sip:hj@192.168.178.193
- destination : computer A
ip address :192.168.178.194
sipuri : sip:mj@192.168.178.194
And I am using sipp to test (I modified the "via" part based on original
uac module).
=====================================================
<< computer B's test.cfg >>
#
# $Id: welcome.cfg,v 1.3 2003/11/03 13:03:56 janakj Exp $
#
# welcome message for new subscribers; based on exec
#
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
# must be yes since REGISTER processing causes an INVITE to be sent,
# which needs to be processed by another process
fork=yes
children=4
alias="kims.myserver.sampro.cse"
alias="myserver.sampro.cse"
alias="192.168.178.193"
# debugging
log_stderror=yes # (cmd line: -E)
mhomed=yes
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/exec.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
modparam("usrloc", "db_mode", 2)
#modparam("usrloc", "db_url", "mysql://ser:heslo@192.168.2.16/ser" )
modparam("usrloc", "db_url", "mysql://root:heslo@localhost/ser")
# ------------------------- request routing logic -------------------
# main routing logic
route{
log(1, "\n\n");
log(1, "START!!!\n");
# 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>=204822) {
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
# if (method=="INVITE") record_route(); # 1=loose routing
if(method=="INVITE"){
log(1, "receive INVITE\n");
record_route();
sl_send_reply("200", "ok");
};
# loose-route processing
if (loose_route()) {
t_relay();
break;
};
# if(method=="ACK"){
# log(1, "receive ACK\n");
# };
log(1, "record-route processing completed\n");
# 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(uri=="sip:mijin@sam.cse"){
# forward(192.168.178.194, 5060);
# };
if (uri=~"sip:192.168.178.193"){
if (method=="REGISTER") {
log(1, "receive REGISTER\n");
# save location before initiating welcome
save("location");
log(1, "save 성공\n");
record_route();
sl_send_reply("200", "ok");
# welcome message
log(1, "route3으로 이동 \n");
route(3);
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
log(1, "Usrloc DB에서 찾
을 수 없음 \n");
break;
};
};
if(method=="BYE"){
log(1, "receive BYE\n");
record_route();
sl_send_reply("200", "ok");
break;
};
t_relay();
}
# welcome message -- if a REGISTER succeeded, look if it is the first-time;
# if so, initiate click-to-dial transaction to set up call to an
announcement
# server; edit the config values first to make it work
route[3] {
if (!exec_msg('
# config:
# --announcement server URI
ANS="sip:hjkim@myserver.sampro.cse"
# --SIP domain
# DOMAIN="192.168.178.193"
DOMAIN="192.168.178.193"
# ctd
CTD=${HOME}/sip_router/examples/ctd.sh
# ------------------------------------
# check if first time ...
SIP_UID=`echo $SIP_HF_TO | sed -e
"s/^.*sip:\([a-zA-Z0-9_\.]*\)@.*$/\1/g"`
QUERY="select flag from subscriber
where username=\"$SIP_UID\";
update subscriber set flag=\"x\"
where username=\"$SIP_UID\" ";
mysql -Bsuser -pheslo -e "$QUERY" ser| grep "^x$" > /dev/null
# ... if so, c-t-d to announcement server
if [ "$?" -ne 0 ] ; then
# flag was not set to x yet -- first-time registration;
# initiate a call from telephone of the user to an announcement server
$CTD "sip:$SIP_UID@$DOMAIN" "$ANS" > /dev/null 2>&1
fi
')) {
log(1, "welcome exec failed\n");
}
log(1, "-------route3 끝 \n");
}
As you see, I added
modparam("rr", "enable_full_lr", no)
if(uri=="sip:mijin@192.168.178.194"){
forward(192.168.178.194, 5060);
append_branch("sip:hjkim@192.168.178.193:5060"); log(1, "next
hop...\n");
break;
};
but, I received error message in computer B(via):
8(12253) ERROR: forward_reply: no 2nd via found in reply
What does it mean?
What can I do for doing what I want ???
Is there any problem in my test.cfg file ?????
I would like to know in detail .
please tell me what to do...............................
Song
_________________________________________________________________
싸게 싸게 MSN공동구매
http://www.waawaa.com/cobuy/cobuy_default.asp?siteid=10160
confirm 83b7fb1612811c64f85f47a6046dd84dea286d15
Dear. All
Hi~ I was adviced many times by "seruser" but my problems has not been
solved.
I,m afraid that I have many questions.
So I intend to specify my present situation.
Operating System: Linux Fedora 3
================================================================================
[root@cse3 ~]# serctl start
Starting SER : started pid(8370)
[root@cse3 ~]# serctl ps
0 8370 attendant
1 8375 fifo server
2 8377 receiver child=0 sock= 127.0.0.1:5060
3 8379 receiver child=1 sock= 127.0.0.1:5060
4 8384 receiver child=2 sock= 127.0.0.1:5060
5 8386 receiver child=3 sock= 127.0.0.1:5060
6 8387 receiver child=0 sock= 192.168.178.193:5060
7 8390 receiver child=1 sock= 192.168.178.193:5060
8 8393 receiver child=2 sock= 192.168.178.193:5060
9 8397 receiver child=3 sock= 192.168.178.193:5060
10 8401 timer
11 8403 tcp receiver
12 8406 tcp receiver
13 8410 tcp receiver
14 8412 tcp receiver
15 8415 tcp main process
==============================================================================
my_ip: 192.168.178.193
sip_uri: sip:hjkim@192.168.178.193
I added my hostname and my ip address in /etc/host file.
And I added
alias="hostname.myserver.sampro.cse"
alias="192.168.178.193"
in /usr/local/etc/ser/ser.cfg file.
==================================== 1
============================================
[root@cse3 ~]# serctl ping sip:hjkim@192.168.178.193
[1]+ Done cat <$path | filter_fl
483
===========================================================================================
==================================== 2
===========================================
I received "SIP/2.0 483 Too Many Hops" .
but I don't know the reason.
Well...My computer is connected other computers including computer has
192.168.178.194 as ip address. And my /etc/host file has
127.0.0.1 cse4 localhost.localdomain localhost
192.168.178.194 kims myserver.sampro.cse
Is there reason in here??? I don't know .
===========================================================================================
[root@cse3 ~]# sipsak -T -s sip:hjkim@192.168.178.193
warning: IP extract from warning activated to be more informational
0: 192.168.178.193 (0.355 ms) SIP/2.0 483 Too Many Hops
1: 192.168.178.193 (0.982 ms) SIP/2.0 483 Too Many Hops
2: 192.168.178.193 (0.459 ms) SIP/2.0 483 Too Many Hops
3: 192.168.178.193 (0.520 ms) SIP/2.0 483 Too Many Hops
....................
177: timeout after 500 ms
177: timeout after 1000 ms
177: timeout after 2000 ms
177: 192.168.178.193 (3306.067 ms) SIP/2.0 483 Too Many Hops
178: 192.168.178.193 (3.073 ms) SIP/2.0 483 Too Many Hops
179: 192.168.178.193 (3.104 ms) SIP/2.0 483 Too Many Hops
..................
313: 192.168.178.193 (3.345 ms) SIP/2.0 483 Too Many Hops
314: 192.168.178.193 (3.494 ms) SIP/2.0 483 Too Many Hops
315: 192.168.178.193 (3.038 ms) SIP/2.0 483 Too Many Hops
316: 192.168.178.193 (3.518 ms) SIP/2.0 483 Too Many Hops
317: 192.168.178.193 (5.048 ms) SIP/2.0 500 I'm terribly sorry, server
error occurred (1/SL)
without Contact header
============================================================================================
======================================== 3
======================================
In response part,
I found the message "SIP/2.0 401 Unauthorized" and "authorization failed
request already contains (Proxy-) Authorization, but received 40[1|7], see
above"
What happened??
============================================================================================
[root@cse3 ~]# sipsak -U -vv -s sip:hjkim@192.168.178.193
warning: ignoring -i option when in usrloc mode
registering user hjkim... authorizing
registering user hjkim...
request:
REGISTER sip:192.168.178.193 SIP/2.0
Authorization: Digest username="hjkim", uri="sip:192.168.178.193",
algorithm=MD5, realm="myserver.sampro.cse",
nonce="4333e4b8b0199c2d9c0f8bd249bedc3b8b20bf87",
response="f58e11aff01e5123bcf5010405e91278"
Via: SIP/2.0/UDP 127.0.0.1:32860;branch=z9hG4bK.4f28e7fb;rport
From: sip:hjkim@192.168.178.193;tag=4de59540
To: sip:hjkim@192.168.178.193
Call-ID: 1306891584(a)127.0.0.1
CSeq: 2 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: sipsak 0.9.1
Expires: 15
Contact: sip:hjkim@127.0.0.1:32860
response:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
127.0.0.1:32860;branch=z9hG4bK.4f28e7fb;rport=32860;received=192.168.178.193
From: sip:hjkim@192.168.178.193;tag=4de59540
To: sip:hjkim@192.168.178.193;tag=b27e1a1d33761e85846fc98f5f3a7e58.ad3a
Call-ID: 1306891584(a)127.0.0.1
CSeq: 2 REGISTER
WWW-Authenticate: Digest realm="myserver.sampro.cse",
nonce="4333e4b8b0199c2d9c0f8bd249bedc3b8b20bf87"
Server: Sip EXpress router (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 192.168.178.193:5060 "Noisy feedback tells: pid=8397
req_src_ip=192.168.178.193 req_src_port=32860 in_uri=sip:192.168.178.193
out_uri=sip:192.168.178.193 via_cnt==1"
error: authorization failed
request already contains (Proxy-) Authorization, but received
40[1|7], see above
============================================================================================
[root@cse3 ~]# serctl ul show
Dumping all contacts may take long: are you sure you want to proceed? [Y|N]
y
===Domain list===
---Domain---
name : 'aliases'
size : 512
table: 0xf50b44c8
d_ll {
n : 0
first: (nil)
last : (nil)
}
---/Domain---
---Domain---
name : 'location'
size : 512
table: 0xf50b2218
d_ll {
n : 2
first: 0xf50b4220
last : 0xf50b4348
}
...Record(0xf50b4220)...
domain: 'location'
aor : 'kim'
~~~Contact(0xf50b4260)~~~
domain : 'location'
aor : 'kim'
Contact : 'sip:hjkim@myserver.sampro.cse'
Expires : Permanent
q : 1
Call-ID :
'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything'
CSeq : 42
User-Agent: 'SIP Express Router FIFO'
received : ''
State : CS_SYNC
Flags : 128
next : (nil)
prev : (nil)
~~~/Contact~~~~
.../Record...
...Record(0xf50b4348)...
domain: 'location'
aor : 'mj'
~~~Contact(0xf50b4388)~~~
domain : 'location'
aor : 'mj'
Contact : 'sip:mj@myserver.sampro.cse'
Expires : Permanent
q : 1
Call-ID :
'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything'
CSeq : 42
User-Agent: 'SIP Express Router FIFO'
received : ''
State : CS_SYNC
Flags : 128
next : (nil)
prev : (nil)
~~~/Contact~~~~
.../Record...
---/Domain---
===/Domain list===
====================================================================================================================================
4 =====================================
I received the message "received ICMP packet (type: 3, code: 10): from
192.168.178.194"
What does means this message??
============================================================================================
[root@cse3 ~]# sipsak -T -vv -s sip:iptel@192.168.178.194
warning: IP extract from warning activated to be more informational
Max-Forwards set to 0
received ICMP packet (type: 3, code: 10): from 192.168.178.194
=======================================================
=======================================================
<< my ser.cfg >>
#
# $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
#debug=3 # debug level (cmd line: -dddddddddd)
#fork=yes
#log_stderror=no # (cmd line: -E)
alias="192.168.178.193"
alias="myserver.sampro.cse"
alias="kims.myserver.sampro.cse"
/* 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"
# ----------------- 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)
# ------------------------- 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 >= 2000048 ) {
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
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=="sip:192.168.178.193") {
# 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=="sip:192.168.178.193") {
if (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
if (!www_authorize("myserver.sampro.cse", "subscriber")) {
www_challenge("myserver.sampro.cse", "0");
break;
};
save("location");
log(1, "save \n");
break;
};
lookup("aliases");
if (!uri=="sip:192.168.178.193") {
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]
{
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
}
######################################################################################
I’d appreciate it if you would give me some solutions.
Thank you in advance.....
mi-jin Song
_________________________________________________________________
행운의 주인공이 이번엔 나일꺼야, 진짜루... 인터넷 복권
http://www.msn.co.kr/money/interlotto/
hi
I am using Sip communicator. I cant receive any voice from the other calling party but the calling party can hear whatever I talk. The ReceiveStreamEvent is not getting generated. Would anyone help me reg this please?
Thanks
Rajesh
---------------------------------
Yahoo! India Matrimony: Find your partner now.
Thank you guys for the clarification. If you do not mind I would ask one
more question. Let say I have aliases 1234 of the account user@server
and I'm getting call to the 1234 alias but there is no one registered at
the moment with URI name@server. I also have SER configured in a way
that if call will come to name@server URI and no user is availeble the
call will be forwarded to voice mail box. Will same scenario work in
case of call to the alias if I do lookup("aliases") before forwarding or
I should do something more sofisticated to make it work?
Thank you in advance,
Igor
-----Original Message-----
From: Atle Samuelsen [mailto:clona@cyberhouse.no]
Sent: Friday, September 23, 2005 1:14 PM
To: Igor Balk
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] Permanent aliases
Hi Igor,
I think your wrong there.
AS far as you register with your box to your sipserver, and have a alias
pointing to the username@yoursipserver then you should'nt have a
problem.
-Atle
* Igor Balk <IBalk(a)intermedia.net> [050923 19:01]:
>
> Sorry if I'm asking stupid question but I'm trying to set up alias for
> example 1234<->john.doe and make it permanent. As far as I understand
> if I use serctl add alias command and then contact for the person will
> change (dynamic ip, login from other coputer etc) my alias would not
> work anymore. Please tell me if I'm worng and if there is a way to fix
> this problem.
>
> Thank you,
> Igor
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
Hi all,
I'm new to SER and I started to play around with it as we need a solution to connect UPD UA's connected to Asterisk to a Microsoft LCS and we tough to use SER for this purpose. I've read so far through the Admin guide and it seems that the TCP support is natively embedded into SER. However I've added to the ser.cfg file the following lines (within the route[0] block just before branching to route [1]):
#Forward all traffic with the uri*(a)172.16.0.201 <mailto:uri*@172.16.0.201>
If (uri=~"sip:*@172.16.0.201) {
forward_tcp(172.16.0.201,5060);
Break;
};
However while sniffing the packets it still appears that the INVITE requests are sent to the LCS using UDP and I receive an ICMP port unreachable. The original request for the UA is sent in UDP. I assume that I configured something wrong and I've checked the mailing lists and I didn't find any related documents. I'm using the latest version of SER available. Needless to say that UDP SIP calls between ASTERISK and SER work fine whatever clients I use.
Any clue or help would be highly appreciated.
Thank you very much in forecast for your help.
Cheers, Frank
Hello all,
Not really easy ton understand how to setup... If somebody can help, I'll really appreciate.
My config :
redhat 7.3 (SME E/Smith edition)
Asterisk 1.0.9
ser
serweb
IP phone 1 IP phone 2
IP public 1 IP public 2
! !
! Internet !
---------------------------------------------
!
!
!
IP public 3
!
! Lan
Asterisk
!
-------------------------------------------------
! !
IP Phone 3 IP Phone 4
IP Local 3 IP Local 4
Every IP Phone are registered in the AsterisK
IP Phone 3 comunique with IP Phone 4 whithout PB
IP Phone 1 comunique with IP Phone 3 or 4 whithout PB
IP Phone 1 comunique with IP Phone 2 whith delay ! So audio stream doesn't go directly from IP Public 1 to IP Public 2.
As I understand S.E.R. should help me to redirect audio stream...
Well I setup S.E.R. and SERWEB under the Asterisk box (wich is the gateway & the firewall)
I add some user whith serweb
I add register in my sip.conf to register my asterisk to ser
I try to do an echo test and nothing append.
Well, how to setup SER to build this config... I'm sure this question certainly basic for most of you, but I've good Knowledge of AsterisK and I start with SIP proxies. I'm really not find F.A.Q. or userguide to build this...
Thank you for your help, Florent.
confirm 83b7fb1612811c64f85f47a6046dd84dea286d15
Dear. All
Hi~ I was adviced many times by "seruser" but my problems has not been
solved.
I,m afraid that I have many questions.
So I intend to specify my present situation.
Operating System: Linux Fedora 3
================================================================================
[root@cse3 ~]# serctl start
Starting SER : started pid(8370)
[root@cse3 ~]# serctl ps
0 8370 attendant
1 8375 fifo server
2 8377 receiver child=0 sock= 127.0.0.1:5060
3 8379 receiver child=1 sock= 127.0.0.1:5060
4 8384 receiver child=2 sock= 127.0.0.1:5060
5 8386 receiver child=3 sock= 127.0.0.1:5060
6 8387 receiver child=0 sock= 192.168.178.193:5060
7 8390 receiver child=1 sock= 192.168.178.193:5060
8 8393 receiver child=2 sock= 192.168.178.193:5060
9 8397 receiver child=3 sock= 192.168.178.193:5060
10 8401 timer
11 8403 tcp receiver
12 8406 tcp receiver
13 8410 tcp receiver
14 8412 tcp receiver
15 8415 tcp main process
==============================================================================
my_ip: 192.168.178.193
sip_uri: sip:hjkim@192.168.178.193
I added my hostname and my ip address in /etc/host file.
And I added
alias="hostname.myserver.sampro.cse"
alias="192.168.178.193"
in /usr/local/etc/ser/ser.cfg file.
==================================== 1
============================================
[root@cse3 ~]# serctl ping sip:hjkim@192.168.178.193
[1]+ Done cat <$path | filter_fl
483
===========================================================================================
==================================== 2
===========================================
I received "SIP/2.0 483 Too Many Hops" .
but I don't know the reason.
Well...My computer is connected other computers including computer has
192.168.178.194 as ip address. And my /etc/host file has
127.0.0.1 cse4 localhost.localdomain localhost
192.168.178.194 kims myserver.sampro.cse
Is there reason in here??? I don't know .
===========================================================================================
[root@cse3 ~]# sipsak -T -s sip:hjkim@192.168.178.193
warning: IP extract from warning activated to be more informational
0: 192.168.178.193 (0.355 ms) SIP/2.0 483 Too Many Hops
1: 192.168.178.193 (0.982 ms) SIP/2.0 483 Too Many Hops
2: 192.168.178.193 (0.459 ms) SIP/2.0 483 Too Many Hops
3: 192.168.178.193 (0.520 ms) SIP/2.0 483 Too Many Hops
....................
177: timeout after 500 ms
177: timeout after 1000 ms
177: timeout after 2000 ms
177: 192.168.178.193 (3306.067 ms) SIP/2.0 483 Too Many Hops
178: 192.168.178.193 (3.073 ms) SIP/2.0 483 Too Many Hops
179: 192.168.178.193 (3.104 ms) SIP/2.0 483 Too Many Hops
..................
313: 192.168.178.193 (3.345 ms) SIP/2.0 483 Too Many Hops
314: 192.168.178.193 (3.494 ms) SIP/2.0 483 Too Many Hops
315: 192.168.178.193 (3.038 ms) SIP/2.0 483 Too Many Hops
316: 192.168.178.193 (3.518 ms) SIP/2.0 483 Too Many Hops
317: 192.168.178.193 (5.048 ms) SIP/2.0 500 I'm terribly sorry, server
error occurred (1/SL)
without Contact header
============================================================================================
======================================== 3
======================================
In response part,
I found the message "SIP/2.0 401 Unauthorized" and "authorization failed
request already contains (Proxy-) Authorization, but received 40[1|7], see
above"
What happened??
============================================================================================
[root@cse3 ~]# sipsak -U -vv -s sip:hjkim@192.168.178.193
warning: ignoring -i option when in usrloc mode
registering user hjkim... authorizing
registering user hjkim...
request:
REGISTER sip:192.168.178.193 SIP/2.0
Authorization: Digest username="hjkim", uri="sip:192.168.178.193",
algorithm=MD5, realm="myserver.sampro.cse",
nonce="4333e4b8b0199c2d9c0f8bd249bedc3b8b20bf87",
response="f58e11aff01e5123bcf5010405e91278"
Via: SIP/2.0/UDP 127.0.0.1:32860;branch=z9hG4bK.4f28e7fb;rport
From: sip:hjkim@192.168.178.193;tag=4de59540
To: sip:hjkim@192.168.178.193
Call-ID: 1306891584(a)127.0.0.1
CSeq: 2 REGISTER
Content-Length: 0
Max-Forwards: 70
User-Agent: sipsak 0.9.1
Expires: 15
Contact: sip:hjkim@127.0.0.1:32860
response:
SIP/2.0 401 Unauthorized
Via: SIP/2.0/UDP
127.0.0.1:32860;branch=z9hG4bK.4f28e7fb;rport=32860;received=192.168.178.193
From: sip:hjkim@192.168.178.193;tag=4de59540
To: sip:hjkim@192.168.178.193;tag=b27e1a1d33761e85846fc98f5f3a7e58.ad3a
Call-ID: 1306891584(a)127.0.0.1
CSeq: 2 REGISTER
WWW-Authenticate: Digest realm="myserver.sampro.cse",
nonce="4333e4b8b0199c2d9c0f8bd249bedc3b8b20bf87"
Server: Sip EXpress router (0.9.4 (i386/linux))
Content-Length: 0
Warning: 392 192.168.178.193:5060 "Noisy feedback tells: pid=8397
req_src_ip=192.168.178.193 req_src_port=32860 in_uri=sip:192.168.178.193
out_uri=sip:192.168.178.193 via_cnt==1"
error: authorization failed
request already contains (Proxy-) Authorization, but received
40[1|7], see above
============================================================================================
[root@cse3 ~]# serctl ul show
Dumping all contacts may take long: are you sure you want to proceed? [Y|N]
y
===Domain list===
---Domain---
name : 'aliases'
size : 512
table: 0xf50b44c8
d_ll {
n : 0
first: (nil)
last : (nil)
}
---/Domain---
---Domain---
name : 'location'
size : 512
table: 0xf50b2218
d_ll {
n : 2
first: 0xf50b4220
last : 0xf50b4348
}
...Record(0xf50b4220)...
domain: 'location'
aor : 'kim'
~~~Contact(0xf50b4260)~~~
domain : 'location'
aor : 'kim'
Contact : 'sip:hjkim@myserver.sampro.cse'
Expires : Permanent
q : 1
Call-ID :
'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything'
CSeq : 42
User-Agent: 'SIP Express Router FIFO'
received : ''
State : CS_SYNC
Flags : 128
next : (nil)
prev : (nil)
~~~/Contact~~~~
.../Record...
...Record(0xf50b4348)...
domain: 'location'
aor : 'mj'
~~~Contact(0xf50b4388)~~~
domain : 'location'
aor : 'mj'
Contact : 'sip:mj@myserver.sampro.cse'
Expires : Permanent
q : 1
Call-ID :
'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything'
CSeq : 42
User-Agent: 'SIP Express Router FIFO'
received : ''
State : CS_SYNC
Flags : 128
next : (nil)
prev : (nil)
~~~/Contact~~~~
.../Record...
---/Domain---
===/Domain list===
====================================================================================================================================
4 =====================================
I received the message "received ICMP packet (type: 3, code: 10): from
192.168.178.194"
What does means this message??
============================================================================================
[root@cse3 ~]# sipsak -T -vv -s sip:iptel@192.168.178.194
warning: IP extract from warning activated to be more informational
Max-Forwards set to 0
received ICMP packet (type: 3, code: 10): from 192.168.178.194
######################################################################################
I’d appreciate it if you would give me some solutions.
Thank you in advance.....
mi-jin Song
_________________________________________________________________
행운의 주인공이 이번엔 나일꺼야, 진짜루... 인터넷 복권
http://www.msn.co.kr/money/interlotto/
