Hi,
Has any one done using Asterisk/Digium T1 card as PSTN gateway, and SER as
the backend soft switch?
I am looking to install Asterisk boxes in a few cities to act as PSTN
gateways and using SER to accept these calls. SER will have to communicate
with other long distance carriers to allow world wide calling.
Will this combination work? I like to know your comments or suggestions on
this combination.
thanks
Tony Lum
You have several choices (like always on Unix):
1. Do like this:
if (method=="INVITE") {
# save messages in mysql for CDRTool
exec_msg("/etc/ser/sertrace.py; exit 1");
};
Make your own script to replace /etc/ser/sertrace.py
You can log the whole SIP packet from STDIN to mysql and index it by
environment variable set by SER (SIP_HF_CALLID).
To scale it up a bit you demonize the logger to reuse only one mysql
connection.
2. Log all SIP packets to a log file with a script like this one:
<<<<<<<<<<<<<<<<<<<<<<<<<<<<
#!/bin/sh
#
# chkconfig: 2345 20 80
# description: controls execution of SIP sniffer
# processname: ng
NAME=ngrep
OPTIONS="-L -d bond0 -t port 5060"
BIN=/usr/bin/ngrep
ROTATE=/usr/sbin/rotatelogs
RETVAL=0
# See how we were called.
case "$1" in
start)
killall $NAME
echo -n "Starting SIP sniffer: "
$BIN $OPTIONS 2>&1 | $ROTATE /var/log/ser/trace/sip 86400 &
RETVAL=$?
echo "ngrep."
;;
stop)
echo -n "Stopping SIP ngrep: "
killall $NAME
RETVAL=$?
echo "done."
;;
restart)
#$0 stop
$0 start
RETVAL=$?
;;
*)
echo "Usage: $0 {start|stop|restart}"
exit 1
esac
exit $RETVAL
<<<<<<<<<<<<<<<<<<<<<<<<
--
Adrian
How to get the call log and registration request in a
different file other that /var/log/messages.
Also, can i get some more details on autherisation and
accountingpackets. Rt now, iam getting only details
given below.
ACC: transaction answered: method=BY
E, i-uri=sip:12334 at 10.10.0.1,
o-uri=sip:12334 at 10.10.0.1, call_id
=4F6F069B-4BF4-4CC7-B99C-4E24F0E2C667 at 192.168.10.154,
from=Manu Sip <sip:manu at s
ip.test.net>;tag=2654486816, code=200
thank you.
MANU.
Hi all,
I am configuring SER in order to use the SEMS's voicemail
when one of my subscribed users is not avaiable or
when all the INVITEs to their contacts fail (fork).
I'am using the following configuration:
route {
....
if (isflagset(4)) { /*voicemail enabled*/
t_on_failure("1");
}
if (!t_relay()) {
sl_reply_error();
break;
}
}
failure_route[1] {
t_reply("100","Trying -- going to voicemail");
if (method=="INVITE" || method=="REFER") {
log("**************** vm start - begin ******************\n");
log("**************** voicemail ******************\n");
if(!vm("/tmp/am_fifo","voicemail")){
log("could not contact the answer machine\n");
t_reply("500","could not contact the answer machine");
};
log("**************** vm start - end ******************\n");
}
else if (method=="BYE") {
log("**************** vm end - begin ******************\n");
log("**************** voicemail ******************\n");
if(!vm("/tmp/am_fifo","bye")){
log("could not contact the answer machine\n");
t_reply("500","could not contact the answer machine");
};
log("**************** vm end - end ******************\n");
};
}
...
This works fine when the system calls a user's contact whose username part
matches the subscribed user name.
Unfortunately, if the system tries to reach such a contact at a phone
number (using a gateway) the result is that SEMS reports:
"Error: 404 voicemail: no email address for user 034823213213."
Can you help me ?
Best regards,
Andrea
Hi !
I have a problem with correct accounting for
calls to the PSTN
SIP UA implemented inside gateway (Mediatrix 2402 ) sends ACK to SER
even if the connection wasn't estabilished (no answer
or reject by second site).
in this way I have a few second in my stats for not estabilished
connections.
Where is problem ? Inside SER config, gateway config or
PBX config ??
How do you resolve this problem ?
Thanks
Andrzej
Hi guys,
I was create a script to route some kinds of routes calls to PSTN ( like
that ). So after 59 sec. it is droped.
Someone can help me ?
Thank you all for attention and kindness in advance.
Jadylson Bomfim
if (uri=~"sip:0[0-9]*@.*") {
route(6);
break;
};
route[6] {
setflag(1);
if (uri=~"^sip:0[1-9]*@") {
strip(1);
prefix("01155");
rewritehostport("ip address:5060");
acc_rad_request("Saindo Para PSTN no Brasil");
if (method=="INVITE" && search("User-Agent: snom")) {
replace("100rel, ", "");
};
append_hf("P-hint: GATEWAY\r\n");
# use UDP to guarantee well-known sender port (TCP ephemeral)
t_relay_to_udp("ip address","5060");
} else {
if (uri=~"^sip:00[0-9][0-9]*@") { # ... forward to gateways then;
#Cheque se inicia com 00
if (uri=~"^sip:00") {
# strip the leading "1"
strip(2);
prefix("011");
# if you have passed through all the checks, let your call go to GW!
rewritehostport("ip address:5060");
# snom conditioner
if (method=="INVITE" && search("User-Agent: snom")) {
replace("100rel, ", "");
};
append_hf("P-hint: GATEWAY\r\n");
# use UDP to guarantee well-known sender port (TCP ephemeral)
t_relay_to_udp("ip address","5060");
};
} else {
if (uri=~"^sip:001*@") { # ... forward to gateways then;
#Cheque se inicia com 00
if (uri=~"^sip:00") {
# strip the leading "1"
strip(2);
# if you have passed through all the checks, let your call go to GW!
rewritehostport("ip address:5060");
# snom conditioner
if (method=="INVITE" && search("User-Agent: snom")) {
replace("100rel, ", "");
};
append_hf("P-hint: GATEWAY\r\n");
# use UDP to guarantee well-known sender port (TCP ephemeral)
t_relay_to_udp("ip address","5060");
};
};
};
};
}
Jadylson da Rocha Passos Bomfim
Redevox Telecom
Uberlandia +55 34 3234-7813
São Paulo +55 11 5055-6888
Móvel +55 34 9103-6854
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Hello,
I am Benoit and I am working on the SER and the server and the database is
working well.
The problem is when I try to registrer SIP agent, the error message is "Error
483:" Too Many Hops". In the file ser.cfg there is a function
mf_process_maxforward_header("10"), and the problem is that I don't know where
is this function and how does it testing the max forward header because the SIP
agent that I use set this field to 70.
Kind regards
Benoit
Hi,
i'm trying to use t_relay_to_tcp() on a machine with several IP
addresses on just one ethernet interface.
SER seems not to set the source address of the TCP connection, so the OS
(more or less) randomly chooses one of the virtual addresses on the
interface, which makes in turn filtering based on source ip address
a B<- (pain in the a** ;).
Is there any way to configure SER 0.8.12 to use a specific source
address in outbound tcp connections (it works for udp, though)?
thanks
axelm
In the main.c file, there is a comment saying that 'we need to open fifo
before do_suid() and start the fifo server after all the socket are
initialized, to inherit them'. I'm wondering why SER has to do it like that.
Can't SER lower its privileges first (do_suid), then open fifo and start the
fifo server? The reason being that I want to run SER with an unprivileged
user. With the current implementation, the fifo file will be created by root
which is not accessible to anyone else. Unless I change the fifo_mode to
0666, which I don't want to, I can't make SER more secure and yet
accessible.
I've tried to move the do_suid() before init_fifo_server() and compile
successfully. The server can run as ser:apache with fifo_mode set to 0660.
That way, my Apache server can talk to SER yet limiting other user directly
talking to SER via serctl. However, I would like to know the reason why it
is implemented the way it is now. Maybe I'm not insightful to see the
obviously reason.
Also, will unix_sock get something like unix_sock_mode in the future?
Zeus Ng
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I have also had problems with getting the ACK back.
I don't completely understand your configuration, you
must allow for packets going both directions, right?
Here is my config :
route
{
# check to see if the message has been around too long
# probably means that it is looping
#
if (!mf_process_maxfwd_header("10"))
{
log("LOG: Too many hops\n");
sl_send_reply("483","Too Many Hops");
break;
};
#
# make sure the length of the message isn't too long!
#
if (len_gt( max_len ))
{
sl_send_reply("513", "Wow -- Message too large");
break;
};
#
# do the loose-routing thing, this is important!
#
if(loose_route())
{
log(1,"doing top loose route");
t_relay();
break;
};
# this is where I was dropping the ACKS.
# I was simply dropping these, but they must be relayed
# because they can be ACKs
if(!(uri==myself))
{
if(!t_relay())
{
sl_reply_error();
break;
};
break;
};
This gets the ACKs through for me.
By the way, I have this configured with Cisco ATAs, version 2.16.
---greg
>
>I have the same problem and posed it to the group yesterday ([Serusers]
>Ignored 200 OK message.) So far the only workaround that I have found is to
>use the rules in my gateway to rewrite the dialed digits before sending them
>to the PSTN PRI, thus leaving the origianl URI intact for SIP
>communications.
>
>One person told me that this is a bug in the Cisco ATA, but it happens on my
>IPDialog phones also. It seems to me that the INVITE is being processed by
>the SER dial rules and is rewritten, but the ACK is not.
>
>Sean
>_______________________________________________
>
>Sean Robertson
>
>NETXUSA
>p. 800-289-6389
>f. 864-233-4344 "Ask me about Voice over IP."
>http://www.netxusa.com/
>
>----- Original Message -----
>From: "Alexander Mayrhofer" <axelm(a)nic.at>
>To: <serusers(a)lists.iptel.org>
>Sent: Friday, June 27, 2003 12:15 PM
>Subject: [Serusers] rewrite & ACK forwarding problem
>
>
>>
>> Hi,
>>
>> we're running SER together with a PSTN Gateway. Before a call get's
>> forwarded to the gateway, we are rewriting the request URI to make
>> rewriting on the GW as simple as possible:
>>
>> route {
>> ...
>> strip(3); # +43xxx -> xxx
>> prefix("0"); # xxx -> 0xxx
>> rewritehostport(xxx.xxx.xxx.xxx, 5060); # request to gateway
>> route(1);
>> break;
>> ...
>>
>> SIP call flow looks like (record route enabled):
>>
>> (1) phone -> SER
>> INVITE sip:*43699xxxxxxxx@nic.at43.at SIP/2.0
>>
>> (2) SER -> phone
>> SIP/2.0 100 trying -- your call is important to us
>>
>> (3) SER -> GW
>> INVITE sip:0699xxxxxxxx@xx.xx.xx.xx:5060 SIP/2.0
>>
>> (4) GW -> SER
>> SIP/2.0 100 Trying
>>
>> (5) GW -> SER
>> SIP/2.0 183 Session Progress
>>
>> (6) SER -> phone
>> SIP/2.0 183 Session Progress
>>
>> (7) GW -> SER
>> SIP/2.0 180 Ringing
>>
>> (8) SER -> phone
>> SIP/2.0 180 Ringing
>>
>> (9) GW -> SER
>> SIP/2.0 200 OK
>> Contact: <sip:0699xxxxxxxx@xx.xx.xx.xx:5060>
>>
>> (10) SER -> phone
>> SIP/2.0 200 OK
>> Contact: <sip:0699xxxxxxx@xx.xx.xx.xx:5060>
>>
>> [ call established, we can talk, but ... ]
>>
>> (11) phone -> SER
>> ACK sip:0699xxxxxxxx@xx.xx.xx.xx:5060 SIP/2.0
>>
>> --> Here starts the problem. That ACK (11) never gets forwarded to the
>> Gateway, so after a few seconds, the GW starts over at (9). Those three
>> packets (9-11) repeat a few times until GW runs into a timeout and drops
>> the call.
>>
>> I have the impression that SER can't match the packet to the previous
>> requests because of the rewritten URI. Is that correct?
>>
>> The only output at debug level 3 is:
>>
>> Warning: sl_send_reply: I won't send a reply for ACK!!
>>
>> Is that a routing goof somewhere in our scripts or is that a more
>> generic problem? Is the problem that the warning indicates somehow
>> related to the fact that the ACK is not being forwarded?
>>
>> Help appreciated.
>>
>> cheers
>>
>> axelm
>>
>> _______________________________________________
>> Serusers mailing list
>> serusers(a)lists.iptel.org
>> http://lists.iptel.org/mailman/listinfo/serusers
>>
>
>
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>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
>
