3 Jun
2005
3 Jun
'05
11:14 p.m.
Dear Friends,
I know this is annoying to post nearly same things but still no answer :( Is
it really so hard or can anybody tell me why this is not working.
There are :
ATA 1 : 12345 (registered and can call 54321, no problem)
ATA 2 : 54321 (registered and can call 12345, no problem)
ANALOG FXO : 201 (registered as 201, for test purposes i only registered one
port of it)
With Brekeke's Ondo SIP Proxy, i created a simple dial-plan and it worked,
even i can add more prefixes no problem. But in SER i cannot :( I'm using
ver 0.9.2. Normally it's like when i want to call a GSM, proxy simply adds a
prefix 201 which is same with registered sip number for regarding analog
port and forwards it (looks like sip:20105353490056@192.168.1.35) to Gateway
and Gateway simply strips 201 from it and dials 05353490056, yes it
connects. But what this same does not work on SER, any ideas because it's
very boring to play all day and night ?
Thanks....
>>>>>>>>> minimal SER.CFG
>>>>>>>>>>
debug=3
fork=yes
log_stderror=no
check_via=no
dns=no
rev_dns=no
listen=192.168.1.10
port=5060
children=4
fifo="/tmp/ser_fifo"
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"
modparam("usrloc", "db_mode", 0)
modparam("rr", "enable_full_lr", 1)
route {
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;
};
record_route();
if (loose_route()) {
t_relay();
break;
};
if (uri==myself) {
if (method=="REGISTER") {
save("location");
break;
};
if (uri=~"^sip:053[0-9]*@.*") {
prefix("201");
rewritehostport ("192.168.1.35:5060");
# forward ("192.168.1.35:5060");
t_relay_to_udp("192.168.1.35","5060");
break;
};
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
if (!t_relay()) {
sl_reply_error();
};
}
>>>>>>>>> call from 12345 to
54321 (SIP 2 SIP Call) >>>>>>>>>>
U 192.168.1.10:5060 -> 192.168.1.201:5060
INVITE sip:54321@192.168.1.201:5060 SIP/2.0..Record-Route:
<sip:192.168.1.1
0;ftag=xITO2cDMxID;lr=on>..Via: SIP/2.0/UDP
192.168.1.10;branch=z9hG4bK233f
.8f8b9d41.0..Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bKba4b3e1d3eed
9c7c..Max-Forwards: 16..To: <sip:54321@192.168.1.10>..From:
<sip:12345@192.
168.1.10;user=phone>;tag=xITO2cDMxID..Call-ID:
911D1B912D91212(a)192.168.1.20
0..CSeq: 1 INVITE..Contact:
<sip:12345@192.168.1.200:5060>..Session-Expires
: 100;refresher=uac..Supported: timer..Content-Type:
application/sdp..Conte
nt-Length: 185....v=0..o=12345 0 0 IN IP4 192.168.1.200..s=-..c=IN IP4
192.
168.1.200..t=0 0..m=audio 2142 RTP/AVP 18 18..a=rtpmap:18
G729/8000..a=send
recv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20..
#
U 192.168.1.10:5060 -> 192.168.1.200:5060
SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bKba4b
3e1d3eed9c7c..To: <sip:54321@192.168.1.10>;tag=xETOxUDMxED..From:
<sip:1234
5@192.168.1.10;user=phone>;tag=xITO2cDMxID..Call-ID:
911D1B912D91212(a)192.16
8.1.200..CSeq: 1 INVITE..Record-Route:
<sip:192.168.1.10;ftag=xITO2cDMxID;l
r=on>..Content-Length: 0....
>>>>>>>>> call from 12345 to
05353490056 (GSM Call) >>>>>>>>>>
U 192.168.1.10:5060 -> 192.168.1.200:5060
SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP
192.16
8.1.200:5060;branch=z9hG4bK31bc676961f91e18..To:
<sip:05353490056@192.168.1
.10>..From: <sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID:
8
76D1B912D91212@192.168.1.200..CSeq: 1 INVITE..Server: Sip EXpress router
(0
.9.2 (i386/linux))..Content-Length: 0..Warning: 392 192.168.1.10:5060
"Nois
y feedback tells: pid=4587 req_src_ip=192.168.1.200 req_src_port=5060
in_u
ri=sip:05353490056@192.168.1.10
out_uri=sip:20105353490056@192.168.1.35:506
0 via_cnt==1"....
#
U 192.168.1.10:5060 -> 192.168.1.35:5060
INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route:
<sip:192
.168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP
192.168.1.10;branch=z9h
G4bK188d.03855495.0..Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bK31bc
676961f91e18..Max-Forwards: 16..To: <sip:05353490056@192.168.1.10>..From:
<
sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID:
876D1B912D9121
2@192.168.1.200..CSeq: 1 INVITE..Contact:
<sip:12345@192.168.1.200:5060>..S
ession-Expires: 100;refresher=uac..Supported: timer..Content-Type:
applicat
ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4
192.168.1.200..s=-.
.c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18 18..a=rtpmap:18
G72
9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20..
#
U 192.168.1.10:5060 -> 192.168.1.35:5060
INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route:
<sip:192
.168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP
192.168.1.10;branch=z9h
G4bK188d.03855495.0..Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bK31bc
676961f91e18..Max-Forwards: 16..To: <sip:05353490056@192.168.1.10>..From:
<
sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID:
876D1B912D9121
2@192.168.1.200..CSeq: 1 INVITE..Contact:
<sip:12345@192.168.1.200:5060>..S
ession-Expires: 100;refresher=uac..Supported: timer..Content-Type:
applicat
ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4
192.168.1.200..s=-.
.c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18 18..a=rtpmap:18
G72
9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20..
#
U 192.168.1.10:5060 -> 192.168.1.35:5060
INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route:
<sip:192
.168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP
192.168.1.10;branch=z9h
G4bK188d.03855495.0..Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bK31bc
676961f91e18..Max-Forwards: 16..To: <sip:05353490056@192.168.1.10>..From:
<
sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID:
876D1B912D9121
2@192.168.1.200..CSeq: 1 INVITE..Contact:
<sip:12345@192.168.1.200:5060>..S
ession-Expires: 100;refresher=uac..Supported: timer..Content-Type:
applicat
ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4
192.168.1.200..s=-.
.c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18 18..a=rtpmap:18
G72
9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20..
#
U 192.168.1.10:5060 -> 192.168.1.35:5060
INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route:
<sip:192
.168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP
192.168.1.10;branch=z9h
G4bK188d.03855495.0..Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bK31bc
676961f91e18..Max-Forwards: 16..To: <sip:05353490056@192.168.1.10>..From:
<
sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID:
876D1B912D9121
2@192.168.1.200..CSeq: 1 INVITE..Contact:
<sip:12345@192.168.1.200:5060>..S
ession-Expires: 100;refresher=uac..Supported: timer..Content-Type:
applicat
ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4
192.168.1.200..s=-.
.c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18 18..a=rtpmap:18
G72
9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20..
#
U 192.168.1.10:5060 -> 192.168.1.200:5060
SIP/2.0 200 ok -- no more pending branches..Via: SIP/2.0/UDP
192.168.1.200:
5060;branch=z9hG4bK31bc676961f91e18..To:
<sip:05353490056@192.168.1.10>;tag
=2f9bfc2acf470ceacf4efdebbaa289b4-026c..From:
<sip:12345@192.168.1.10;user=
phone>;tag=xIjNxkDMxID..Call-ID: 876D1B912D91212@192.168.1.200..CSeq: 1
CAN
CEL..Server: Sip EXpress router (0.9.2 (i386/linux))..Content-Length:
0..Wa
rning: 392 192.168.1.10:5060 "Noisy feedback tells: pid=4586
req_src_ip=19
2.168.1.200 req_src_port=5060 in_uri=sip:05353490056@192.168.1.10
out_uri=s
ip:20105353490056@192.168.1.35:5060 via_cnt==1"....
3 Jun
3 Jun
11:16 p.m.
Hi
To be honest I never really designed/wrote it to sell, hence its tightly
intergrated into the way we do our system, thats the reason I wrote the
pseudocode in the email, if I can get time, I will try to see if I can
make the next version standalone, and then just dump it as opensource or
something, but dont hold ur breath on that, cause these days I barely
have time to goto sleep, seems like I am building ser setups for half
the world :-) . I went through the email below and hers my thoughts for
the part I understand. If you want one written we may look at it, but we
would look at a minimum cost £3000
1. Account for everything, and in ur script which does the
billing/rating just ignore all numbers which start with 833, its better
to do it here, than in ser itself, although in ser you could simply say
if method !=REGISTER and uri!=^sip:833) setflag(1) or whatever acc flag
is...that lots is not syntatically correct I just typed it out...but you
get the idea.
2. I dont follow much of the rest :-) , but it seems as if when a user
dial a particvular number you wish it to goto ext1, then 2 etc etc, if
that is the case look at usr_preferences table, where you can set onbusy
etc paratemers, and tell them where to send the call . Note I have had
varying success with this cause it does depend on your client a little.
And in ser.cfg what you do is to use avpops to pull that value out,
check what to do, set a flag, and then forward the call.
Hope the above is of some help...
Iqbal
Ozan Blotter wrote:
.
--------------080700020303020806010007--
.
Ozan Blotter wrote:
Dear Iqbal,
Do you ever want to sell it , mean i want to buy :) all i need is so
simple , please check this and tell me how we can arrange it, i 'm
ready to pay quickly and also there's a small problem for me;
Dear Friends,
A friend of me is writing Basic Billing for SER :) in PHP. I'm gonna
place it somewhere so anyone can add remove features to it and repost
it. Will let all of you know and download it. But for now i need to
create a dial plan. I do not want to make accounting for numbers
beginning with 833 prefix, is there a way for it ? Another thing is ,
i've tried with so many things but always give errors in config. I
have 2 x 4 Port Analog GSM Gateways, first IP is 192.168.1.10 and
second IP is 192.168.1.20, SER Server has two ethernets, one is for
Public IP 212.154.XXX.YYY for general use and the second ethernet is
192.168.1.5 connected to HUB where Gateways's LAN is. All ports of
FXO's are registered with SER like 1111,2222,3333,4444 (Gateway A) and
5555,6666,7777,8888 (Gateway B). I want SER to send some calls to that
numbers depending on Prefix :
Before i worked with Ondo SIP Proxy a little and was successful with
dial plan. Ex: a user dials 05353490056, proxys adds 1111 prefix,
route to registered SIP client 1111 and send the call as
111105353490056(a)IP.of.Proxy to Gateway, prefix 1111 is stripped from
SIP and the gateway connects to GSM like 05353490056(a)IP.of.Gateway
Ex for Gateways:
* for 1111,2222,3333 and 4444 : i want to use 053 prefix, so when a
user dial a number like 05353490056, SER has to route it to 1111, if
it's busy then route to 2222 so on till 4444.
* for 5555 and 6666 : i want to use 050 and 055 prefix, again same as
above.
* for 7777 and 8888 : i want to use 054 prefix.
* for prefixes apart from above and 833, i want to route all calls to
85.96.XXX.YYY , important thing is the clients are registered on SER
but i want to route calls over from SER and by keeping their
username/password combination since the clients are created on
85.96.XXX.YYY and that Gateway does auth for users, but SER also has
to do accounting for this calls.
Do i need so many things ?
Thanks,
Ozan Blotter
*ps : I'm using SER 0.9.2 from CVS :)
current SER.CFG as follows :
[root@localhost ser]# cat ser.cfg
debug=3
fork=yes
log_stderror=no
listen=192.168.1.9 # put your server IP address here
port=5060
children=4
dns=no
rev_dns=no
fifo="/tmp/ser_fifo"
fifo_db_url="mysql://ser:heslo@localhost/ser"
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/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/uri_db.so"
modparam("auth_db|uri_db|usrloc", "db_url",
"mysql://ser:heslo@localhost/ser")
modparam("auth_db", "calculate_ha1", 1)
modparam("auth_db", "password_column", "password")
modparam("usrloc", "db_mode", 1)
modparam("rr", "enable_full_lr", 1)
modparam("acc", "log_level", 1)
modparam("acc", "db_flag", 1)
modparam("acc", "log_missed_flag", 3)
modparam("acc", "log_fmt", "fimos")
modparam("acc", "db_url",
"mysql://ser:heslo@localhost/ser")
modparam("acc", "db_missed_flag", 2)
modparam("acc", "early_media", 1)
modparam("acc", "failed_transactions", 1)
modparam("acc", "log_flag", 1)
route {
setflag(1);
#
-----------------------------------------------------------------
# Sanity Check Section
#
-----------------------------------------------------------------
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483", "Too Many Hops");
break;
};
if (msg:len > max_len) {
sl_send_reply("513", "Message Overflow");
break;
};
#
-----------------------------------------------------------------
# Record Route Section
#
-----------------------------------------------------------------
if (method!="REGISTER") {
record_route();
};
#
-----------------------------------------------------------------
# Loose Route Section
#
-----------------------------------------------------------------
if (loose_route()) {
route(1);
break;
};
#
-----------------------------------------------------------------
# Call Type Processing Section
#
-----------------------------------------------------------------
if (uri!=myself) {
route(1);
break;
};
if (uri==myself) {
if (method=="INVITE" || method=="BYE" ||
method=="CANCEL") {
setflag(1);
route(3);
break;
} else if (method=="REGISTER") {
route(2);
break;
};
lookup("aliases");
if (uri!=myself) {
route(1);
break;
};
if (!lookup("location")) {
if (uri=~"^sip:053[0-9]*@") {
prefix("1111");
rewritehost ("192.168.1.10");
break;
};
sl_send_reply("404", "User Not Found");
break;
};
route(1);
};
}
route[1] {
#
-----------------------------------------------------------------
# Default Message Handler
#
-----------------------------------------------------------------
if (!t_relay()) {
sl_reply_error();
};
}
route[2] {
#
-----------------------------------------------------------------
# REGISTER Message Handler
# ----------------------------------------------------------------
sl_send_reply("100", "Trying");
if (!www_authorize("","subscriber")) {
www_challenge("","0");
break;
};
if (!check_to()) {
sl_send_reply("401", "Unauthorized");
break;
};
consume_credentials();
if (!save("location")) {
sl_reply_error();
};
}
route[3] {
#
-----------------------------------------------------------------
# INVITE Message Handler
#
-----------------------------------------------------------------
if (!proxy_authorize("localhost","subscriber")) {
proxy_challenge("localhost","0");
break;
} else if (!check_from()) {
sl_send_reply("403", "Use From=ID");
break;
};
consume_credentials();
lookup("aliases");
if (uri!=myself) {
route(1);
break;
};
if (!lookup("location")) {
sl_send_reply("404", "User Not Found");
break;
};
route(1);
}
[root@localhost ser]#
----- Original Message ----- From: "Iqbal" <iqbal(a)gigo.co.uk>
To: <cosmocid(a)ispro.net.tr>
Sent: Thursday, June 02, 2005 8:00 PM
Subject: [Fwd: Re: [Serusers] VoIp Billing Solution SIP server
compatible!!!]
> Not
sure it helps, but was my thought process when I looked at
> building mine
>
> Iqbal
>
.
Dear Friends,
I know this is annoying to post nearly same things but still no answer
:( Is it really so hard or can anybody tell me why this is not working.
There are :
ATA 1 : 12345 (registered and can call 54321, no problem)
ATA 2 : 54321 (registered and can call 12345, no problem)
ANALOG FXO : 201 (registered as 201, for test purposes i only
registered one port of it)
With Brekeke's Ondo SIP Proxy, i created a simple dial-plan and it
worked, even i can add more prefixes no problem. But in SER i cannot
:( I'm using ver 0.9.2. Normally it's like when i want to call a GSM,
proxy simply adds a prefix 201 which is same with registered sip
number for regarding analog port and forwards it (looks like
sip:20105353490056@192.168.1.35) to Gateway and Gateway simply strips
201 from it and dials 05353490056, yes it connects. But what this same
does not work on SER, any ideas because it's very boring to play all
day and night ?
Thanks....
>>>>>>>>> minimal
SER.CFG >>>>>>>>>>
>>>>>>>>
debug=3
fork=yes
log_stderror=no
check_via=no
dns=no
rev_dns=no
listen=192.168.1.10
port=5060
children=4
fifo="/tmp/ser_fifo"
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"
modparam("usrloc", "db_mode", 0)
modparam("rr", "enable_full_lr", 1)
route {
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;
};
record_route();
if (loose_route()) {
t_relay();
break;
};
if (uri==myself) {
if (method=="REGISTER") {
save("location");
break;
};
if (uri=~"^sip:053[0-9]*@.*") {
prefix("201");
rewritehostport ("192.168.1.35:5060");
# forward ("192.168.1.35:5060");
t_relay_to_udp("192.168.1.35","5060");
break;
};
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
};
if (!t_relay()) {
sl_reply_error();
};
}
>>>>>>>>> call from
12345 to 54321 (SIP 2 SIP Call) >>>>>>>>>>
>>>>>>>>
U 192.168.1.10:5060 -> 192.168.1.201:5060
INVITE sip:54321@192.168.1.201:5060 SIP/2.0..Record-Route:
<sip:192.168.1.1
0;ftag=xITO2cDMxID;lr=on>..Via: SIP/2.0/UDP
192.168.1.10;branch=z9hG4bK233f
.8f8b9d41.0..Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bKba4b3e1d3eed
9c7c..Max-Forwards: 16..To: <sip:54321@192.168.1.10>..From:
<sip:12345@192.
168.1.10;user=phone>;tag=xITO2cDMxID..Call-ID:
911D1B912D91212(a)192.168.1.20
0..CSeq: 1 INVITE..Contact:
<sip:12345@192.168.1.200:5060>..Session-Expires
: 100;refresher=uac..Supported: timer..Content-Type:
application/sdp..Conte
nt-Length: 185....v=0..o=12345 0 0 IN IP4 192.168.1.200..s=-..c=IN
IP4 192.
168.1.200..t=0 0..m=audio 2142 RTP/AVP 18 18..a=rtpmap:18
G729/8000..a=send
recv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20..
#
U 192.168.1.10:5060 -> 192.168.1.200:5060
SIP/2.0 180 Ringing..Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bKba4b
3e1d3eed9c7c..To: <sip:54321@192.168.1.10>;tag=xETOxUDMxED..From:
<sip:1234
5@192.168.1.10;user=phone>;tag=xITO2cDMxID..Call-ID:
911D1B912D91212(a)192.16
8.1.200..CSeq: 1 INVITE..Record-Route:
<sip:192.168.1.10;ftag=xITO2cDMxID;l
r=on>..Content-Length: 0....
>>>>>>>>> call from
12345 to 05353490056 (GSM Call) >>>>>>>>>>
>>>>>>>>
U 192.168.1.10:5060 -> 192.168.1.200:5060
SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP
192.16
8.1.200:5060;branch=z9hG4bK31bc676961f91e18..To:
<sip:05353490056@192.168.1
.10>..From:
<sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID: 8
76D1B912D91212@192.168.1.200..CSeq: 1 INVITE..Server: Sip EXpress
router (0
.9.2 (i386/linux))..Content-Length: 0..Warning: 392 192.168.1.10:5060
"Nois
y feedback tells: pid=4587 req_src_ip=192.168.1.200
req_src_port=5060 in_u
ri=sip:05353490056@192.168.1.10
out_uri=sip:20105353490056@192.168.1.35:506
0 via_cnt==1"....
#
U 192.168.1.10:5060 -> 192.168.1.35:5060
INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route:
<sip:192
.168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP
192.168.1.10;branch=z9h
G4bK188d.03855495.0..Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bK31bc
676961f91e18..Max-Forwards: 16..To:
<sip:05353490056@192.168.1.10>..From: <
sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID:
876D1B912D9121
2@192.168.1.200..CSeq: 1 INVITE..Contact:
<sip:12345@192.168.1.200:5060>..S
ession-Expires: 100;refresher=uac..Supported: timer..Content-Type:
applicat
ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4
192.168.1.200..s=-.
.c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18
18..a=rtpmap:18 G72
9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20..
#
U 192.168.1.10:5060 -> 192.168.1.35:5060
INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route:
<sip:192
.168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP
192.168.1.10;branch=z9h
G4bK188d.03855495.0..Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bK31bc
676961f91e18..Max-Forwards: 16..To:
<sip:05353490056@192.168.1.10>..From: <
sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID:
876D1B912D9121
2@192.168.1.200..CSeq: 1 INVITE..Contact:
<sip:12345@192.168.1.200:5060>..S
ession-Expires: 100;refresher=uac..Supported: timer..Content-Type:
applicat
ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4
192.168.1.200..s=-.
.c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18
18..a=rtpmap:18 G72
9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20..
#
U 192.168.1.10:5060 -> 192.168.1.35:5060
INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route:
<sip:192
.168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP
192.168.1.10;branch=z9h
G4bK188d.03855495.0..Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bK31bc
676961f91e18..Max-Forwards: 16..To:
<sip:05353490056@192.168.1.10>..From: <
sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID:
876D1B912D9121
2@192.168.1.200..CSeq: 1 INVITE..Contact:
<sip:12345@192.168.1.200:5060>..S
ession-Expires: 100;refresher=uac..Supported: timer..Content-Type:
applicat
ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4
192.168.1.200..s=-.
.c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18
18..a=rtpmap:18 G72
9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20..
#
U 192.168.1.10:5060 -> 192.168.1.35:5060
INVITE sip:20105353490056@192.168.1.35:5060 SIP/2.0..Record-Route:
<sip:192
.168.1.10;ftag=xIjNxkDMxID;lr=on>..Via: SIP/2.0/UDP
192.168.1.10;branch=z9h
G4bK188d.03855495.0..Via: SIP/2.0/UDP
192.168.1.200:5060;branch=z9hG4bK31bc
676961f91e18..Max-Forwards: 16..To:
<sip:05353490056@192.168.1.10>..From: <
sip:12345@192.168.1.10;user=phone>;tag=xIjNxkDMxID..Call-ID:
876D1B912D9121
2@192.168.1.200..CSeq: 1 INVITE..Contact:
<sip:12345@192.168.1.200:5060>..S
ession-Expires: 100;refresher=uac..Supported: timer..Content-Type:
applicat
ion/sdp..Content-Length: 185....v=0..o=12345 0 0 IN IP4
192.168.1.200..s=-.
.c=IN IP4 192.168.1.200..t=0 0..m=audio 2146 RTP/AVP 18
18..a=rtpmap:18 G72
9/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrecv..a=ptime:20..
#
U 192.168.1.10:5060 -> 192.168.1.200:5060
SIP/2.0 200 ok -- no more pending branches..Via: SIP/2.0/UDP
192.168.1.200:
5060;branch=z9hG4bK31bc676961f91e18..To:
<sip:05353490056@192.168.1.10>;tag
=2f9bfc2acf470ceacf4efdebbaa289b4-026c..From:
<sip:12345@192.168.1.10;user=
phone>;tag=xIjNxkDMxID..Call-ID: 876D1B912D91212@192.168.1.200..CSeq:
1 CAN
CEL..Server: Sip EXpress router (0.9.2 (i386/linux))..Content-Length:
0..Wa
rning: 392 192.168.1.10:5060 "Noisy feedback tells: pid=4586
req_src_ip=19
2.168.1.200 req_src_port=5060 in_uri=sip:05353490056@192.168.1.10
out_uri=s
ip:20105353490056@192.168.1.35:5060 via_cnt==1"....
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7119
days inactive
7119
days old
1 comments
2 participants
participants (2)
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Iqbal -
Ozan Blotter