7 Jun
2004
7 Jun
'04
9:15 a.m.
I am having problem using PSTN(AS5300). I have a VOCAL setup on the same machine and PSTN
works fine with VOCAL. Now I want to test SER on the same machine. I shut down VOCAL and
start SER. I can make SIP to SIP calls but can not make SIP to PSTN call. In my ser.cfg, I
redirect all the calls starting with 1 to the PSTN like this:
if (uri=~"^1.*"){
log(1,"Forwarding to PSTN\n");
record_reoute();
rewritehostport("PSTN IP:PORT");
};
Do I need any specific module for PSTN to work? How can I debug this?
Gary
8 Jun
8 Jun
5:42 a.m.
The reqular expression is not correct, it should be something like this:
if (uri =~ "^sip:1.*")
Jan.
On 07-06 09:15, gc wrote:
I am having problem using PSTN(AS5300). I have a VOCAL
setup on the same machine and PSTN works fine with VOCAL. Now I want to test SER on the
same machine. I shut down VOCAL and start SER. I can make SIP to SIP calls but can not
make SIP to PSTN call. In my ser.cfg, I redirect all the calls starting with 1 to the PSTN
like this:
if (uri=~"^1.*"){
log(1,"Forwarding to PSTN\n");
record_reoute();
rewritehostport("PSTN IP:PORT");
};
Do I need any specific module for PSTN to work? How can I debug this?
Gary
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
10:48 a.m.
Sorry for the misspell. I did have "^SIP:" in there. I tried forward(),
t_relay_to_udp() , t_relay_to(). None of them work. I also used ngrep to
display the sip messages. it seems the message never get forward to PSTN
gateway. Although I can directly connect to PSTN gateway with my sip phone
and make phone call. How can I debug this thing?
gary
----- Original Message -----
From: "Jan Janak" <jan(a)iptel.org>
To: "gc" <garych(a)unidial.com>
Cc: <serusers(a)lists.iptel.org>
Sent: Tuesday, June 08, 2004 5:42 AM
Subject: Re: [Serusers] Problem using PSTN.
The reqular expression is not correct, it should be
something like this:
if (uri =~ "^sip:1.*")
Jan.
On 07-06 09:15, gc wrote:
> I am having problem using PSTN(AS5300). I have a VOCAL setup on the same
machine and PSTN works fine with VOCAL. Now I want to test SER on the same
machine. I shut down VOCAL and start SER. I can make SIP to SIP calls but
can not make SIP to PSTN call. In my ser.cfg, I redirect all the calls
starting with 1 to the PSTN like this:
if
(uri=~"^1.*"){
log(1,"Forwarding to PSTN\n");
record_reoute();
rewritehostport("PSTN IP:PORT");
};
Do I need any specific module for PSTN to work? How can I debug this?
Gary
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
12:02 p.m.
OK, to clarify it a bit, it should be something like this:
if (uri =~ "^sip:1[0-9]+@.*") {
record_route();
rewritehostport("GW:PORT");
if (!t_relay()) {
sl_reply_error();
break;
};
};
Jan.
On 08-06 10:48, gc wrote:
Sorry for the misspell. I did have "^SIP:"
in there. I tried forward(),
t_relay_to_udp() , t_relay_to(). None of them work. I also used ngrep to
display the sip messages. it seems the message never get forward to PSTN
gateway. Although I can directly connect to PSTN gateway with my sip phone
and make phone call. How can I debug this thing?
gary
----- Original Message -----
From: "Jan Janak" <jan(a)iptel.org>
To: "gc" <garych(a)unidial.com>
Cc: <serusers(a)lists.iptel.org>
Sent: Tuesday, June 08, 2004 5:42 AM
Subject: Re: [Serusers] Problem using PSTN.
The reqular expression is not correct, it should
be something like this:
if (uri =~ "^sip:1.*")
Jan.
On 07-06 09:15, gc wrote:
> I am having problem using PSTN(AS5300). I have a VOCAL setup on the same
machine and PSTN works fine with VOCAL. Now I want to test SER on the same
machine. I shut down VOCAL and start SER. I can make SIP to SIP calls but
can not make SIP to PSTN call. In my ser.cfg, I redirect all the calls
starting with 1 to the PSTN like this:
if
(uri=~"^1.*"){
log(1,"Forwarding to PSTN\n");
record_reoute();
rewritehostport("PSTN IP:PORT");
};
Do I need any specific module for PSTN to work? How can I debug this?
Gary
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
3:41 p.m.
Yes, I do have this. It is still not working. Anything else you can think of
that may cause PSTN problem? How about hosts file on this machine? How can I
debug this? I looked at /var/log/messes, it did not help me.
Gary
----- Original Message -----
From: "Jan Janak" <jan(a)iptel.org>
To: "gc" <garych(a)unidial.com>
Cc: <serusers(a)lists.iptel.org>
Sent: Tuesday, June 08, 2004 12:02 PM
Subject: Re: [Serusers] Problem using PSTN.
OK, to clarify it a bit, it should be something like
this:
if (uri =~ "^sip:1[0-9]+@.*") {
record_route();
rewritehostport("GW:PORT");
if (!t_relay()) {
sl_reply_error();
break;
};
};
Jan.
On 08-06 10:48, gc wrote:
> Sorry for the misspell. I did have "^SIP:" in there. I tried forward(),
> t_relay_to_udp() , t_relay_to(). None of them work. I also used ngrep to
> display the sip messages. it seems the message never get forward to PSTN
> gateway. Although I can directly connect to PSTN gateway with my sip
phone
> and make phone call. How can I debug this thing?
>
> gary
> ----- Original Message -----
> From: "Jan Janak" <jan(a)iptel.org>
> To: "gc" <garych(a)unidial.com>
> Cc: <serusers(a)lists.iptel.org>
> Sent: Tuesday, June 08, 2004 5:42 AM
> Subject: Re: [Serusers] Problem using PSTN.
>
>
> > The reqular expression is not correct, it should be something like
this:
> >
> > if (uri =~ "^sip:1.*")
> >
> > Jan.
> >
> > On 07-06 09:15, gc wrote:
> > > I am having problem using PSTN(AS5300). I have a VOCAL setup on the
same
> machine and PSTN works fine with VOCAL. Now I
want to test SER on the
same
> machine. I shut down VOCAL and start SER. I can
make SIP to SIP calls
but
> can not make SIP to PSTN call. In my ser.cfg, I
redirect all the calls
> starting with 1 to the PSTN like this:
> > > if (uri=~"^1.*"){
> > > log(1,"Forwarding to PSTN\n");
> > > record_reoute();
> > > rewritehostport("PSTN IP:PORT");
> > > };
> > >
> > > Do I need any specific module for PSTN to work? How can I debug
this?
> > >
> > > Gary
> > >
> >
> > > _______________________________________________
> > > Serusers mailing list
> > > serusers(a)lists.iptel.org
> > > http://lists.iptel.org/mailman/listinfo/serusers
> >
> >
>
3:57 p.m.
On Tue, 8 Jun 2004, gc wrote:
Sorry for the misspell. I did have "^SIP:"
in there. I tried forward(),
t_relay_to_udp() , t_relay_to(). None of them work. I also used ngrep to
display the sip messages. it seems the message never get forward to PSTN
gateway. Although I can directly connect to PSTN gateway with my sip phone
and make phone call. How can I debug this thing?
Where exactly in the route() function did you add the above into the
config? The placement matters. If for example you're using the stock
ser.cfg and placed the code at the very end of route() then it wont work
because the stock portion of the config that looks like:
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
will trap the call before it even gets to your addition at the end of the
config.
Try changing the above to (or something similar):
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
# attempt handoff to PSTN.
log( "Forwarding to PSTN\n" );
rewritehost( "your pstn gateway router" );
forward( "your pstn gateway router", 5060 );
};
Antonio Querubin
Pacific LightNet Communications / Hawaii OnLine
System Administrator
tony(a)aloha.net (808) 791-2898
4:42 p.m.
Thanks a lot. It worked this time.
Gary
----- Original Message -----
From: "Antonio Querubin" <tony(a)aloha.net>
To: "gc" <garych(a)unidial.com>
Cc: "Jan Janak" <jan(a)iptel.org>rg>; <serusers(a)lists.iptel.org>
Sent: Tuesday, June 08, 2004 3:57 PM
Subject: Re: [Serusers] Problem using PSTN.
On Tue, 8 Jun 2004, gc wrote:
> Sorry for the misspell. I did have "^SIP:" in there. I tried forward(),
> t_relay_to_udp() , t_relay_to(). None of them work. I also used ngrep to
> display the sip messages. it seems the message never get forward to PSTN
> gateway. Although I can directly connect to PSTN gateway with my sip
phone
and make phone
call. How can I debug this thing?
Where exactly in the route() function did you add the above into the
config? The placement matters. If for example you're using the stock
ser.cfg and placed the code at the very end of route() then it wont work
because the stock portion of the config that looks like:
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
break;
};
will trap the call before it even gets to your addition at the end of the
config.
Try changing the above to (or something similar):
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
# attempt handoff to PSTN.
log( "Forwarding to PSTN\n" );
rewritehost( "your pstn gateway router" );
forward( "your pstn gateway router", 5060 );
};
Antonio Querubin
Pacific LightNet Communications / Hawaii OnLine
System Administrator
tony(a)aloha.net (808) 791-2898
7335
days inactive
7336
days old
6 comments
3 participants
participants (3)
-
Antonio Querubin -
gc -
Jan Janak