AJ,
I have found it is essential to get the ability to
trace and debug calls working. This requires some
effort, but it will answer many of your questions when
you do!
ngrep is a good start as mentioned before. At least then
you can fumble through the packet dump to see if the
request is being sent to your 5350 at all.
I use ngrep for quick stuff, but I prefer sipscenario and
ethereal (tethereal) for sniffing. You will need a hub that
is plugged into all of your devices, or you will need to
turn on port monitoring (depending on your switch type...I
use cisco). You then set up a sniffer on the monitored port.
A trace will reveal all. You will then *know* if a request is
being sent to your 5350.
In any event, you seem to have a syntax error:
rewritehost("216.157.199.1:5060");
forward(uri:host, uri:port);
You probably want:
forward("216.157.199.1", "5060");
or whatever...I don't recall the syntax.
I also don't remember exactly how the uri==myself works.
Perhaps you need more aliases to make that work, like:
etc..
there is a bit more work to do if you want bidirectional calls.
because the cisco you have has to know which way the call is going.
so, people invent dialplans. you may want to have
ser prepend '9' to an outbound call, then your dialplan recognizes the
leading '9' on the cisco and strips it and makes the outbound call.
You've
got to catch an inbound call and forward it to the sip-uas (the ser
proxy server).
these are vague hints, i know. Good Luck!
-g
On Jun 15, 2004, at 12:39 PM, AJ Grinnell wrote:
Before i get too far ahead of my self here, will
someone just tell me
if
this config will work or not.
# ----------- global configuration parameters ------------------------
debug=4 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=yes # (cmd line: -E)
listen=216.157.192.117
listen=127.0.0.1
alias=216.157.192.117
alias=x.x.x.x
check_via=yes # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
fifo="/tmp/ser_fifo"
sip_warning=yes
syn_branch=no
server_signature=yes
reply_to_via=no
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
loadmodule "/usr/lib/ser/modules/mysql.so"
loadmodule "/etc/lib/ser/modules/sl.so"
loadmodule "/etc/lib/ser/modules/tm.so"
loadmodule "/etc/lib/ser/modules/rr.so"
loadmodule "/etc/lib/ser/modules/maxfwd.so"
loadmodule "/etc/lib/ser/modules/usrloc.so"
loadmodule "/etc/lib/ser/modules/registrar.so"
loadmodule "/etc/lib/ser/modules/auth.so"
loadmodule "/etc/lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters ---------------
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
modparam("usrloc", "db_mode", 2)
modparam("auth_db", "db_url",
"sql://ser:heslo@localhost/ser")
# -- auth params --
modparam("auth_db", "db_url",
"sql://ser:heslo@localhost/ser")
modparam("auth_db", "calculate_ha1", yes)
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 > max_len ) {
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
record_route();
# loose-route processing
if (loose_route()) {
t_relay();
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("asterisk.crt.net",
"subscriber"))
{
www_challenge("asterisk.crt.net",
"0");
break;
};
save("location");
break;
};
if (uri=~"^sip:9742146@.*") {
# attempt handoff to PSTN.
log( "Forwarding to PSTN\n" );
rewritehost("216.157.199.1:5060");
forward(uri:host, uri:port);
};
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
};
}
AJ Grinnell
Network Operations Technician
CRT/ Arialink Broadband
1223 Turner Street, Suite A
Lansing, MI 48906
517.346.5041
517.492.1321 direct
-----Original Message-----
From: Vitaly Nikolaev [mailto:vitaly@switchgate.com]
Sent: Tuesday, June 15, 2004 1:17 PM
To: AJ Grinnell
Cc: serusers(a)lists.iptel.org
Subject: RE: [Serusers] No outgoing calls
Yea...
Find program called ngrep (should go with SER... find it there)
Compile it and execute "ngrep -q SIP"
Then call and watch... :)
That is tool number one for me
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On
Behalf Of AJ Grinnell
Sent: Tuesday, June 15, 2004 1:12 PM
To: serusers(a)lists.iptel.org
Subject: RE: [Serusers] No outgoing calls
I am beginning to think that maybe Ser is working, but not getting the
calls
to the Cisco 5350. Is there anything else I need to do to get this
working?
AJ Grinnell
Network Operations Technician
CRT/ Arialink Broadband
1223 Turner Street, Suite A
Lansing, MI 48906
517.346.5041
517.492.1321 direct
-----Original Message-----
From: Steve Blair [mailto:blairs@isc.upenn.edu]
Sent: Tuesday, June 15, 2004 12:47 PM
To: AJ Grinnell
Subject: Re: [Serusers] No outgoing calls
Oh, ok. I assumed that was the case but it wasn't clear from
the configuration data.
_Steve
AJ Grinnell wrote:
I am listening on x.x.x.x, which is a public IP
AJ Grinnell
Network Operations Technician
CRT/ Arialink Broadband
1223 Turner Street, Suite A
Lansing, MI 48906
517.346.5041
517.492.1321 direct
-----Original Message-----
From: Steve Blair [mailto:blairs@isc.upenn.edu]
Sent: Tuesday, June 15, 2004 11:56 AM
To: AJ Grinnell
Subject: Re: [Serusers] No outgoing calls
Well you need to indicate what address you are listen on. From
the config you sent it appears as if you are only listening on the
loopback address.
AJ Grinnell wrote:
> This is where i am getting lost. I am listening on the loopback, but
also
on
> a public IP. I really dont know if my statements are correct, I am
piecing
> together various bits here and there. This is
all greek to me right
now.
>
> AJ Grinnell
> Network Operations Technician
> CRT/ Arialink Broadband
> 1223 Turner Street, Suite A
> Lansing, MI 48906
> 517.346.5041
> 517.492.1321 direct
>
>
> -----Original Message-----
> From: Steve Blair [mailto:blairs@isc.upenn.edu]
> Sent: Tuesday, June 15, 2004 10:55 AM
> To: AJ Grinnell
> Subject: Re: [Serusers] No outgoing calls
> So what does your logging and ngrep output show? I see your listening
> on the loopback address. This may be correct but I have never seen
> it before. Do you really want to listen on that address? Also is the
> regular expression in the "native SIP destinations..." if statement
being
> matched? If not I'd check what domain
your UAs are register with
> and rewrite both the alias statement and regular expression as
> needed.
>
> -Steve
>
> AJ Grinnell wrote:
>
>
>> Can someone please help with this. We are switching from using
Asterisk
by
>> itself to using Ser w/ Asterisk. As it
stands now, I can not make
>> any
>> outgoing calls, authentication is not working, and I am about to
>> give
up
>
>
on
>> Ser. I know that Ser is a great server, but my configuration is not
going
>> well at all. Below is my config. All I
want to do for now is just be
able
>>
>>
>>
>>
> to
>
>> make outgoing calls, and maybe even recieve a call or two by the end
of
>
>
the
>> day. What am I doing wrong here? Help?
>>
>> # ----------- global configuration parameters
------------------------
>>
>> #debug=3 # debug level (cmd line: -dddddddddd)
>> #fork=yes
>> #log_stderror=no # (cmd line: -E)
>>
>> listen=x.x.x.x
>> listen=127.0.0.1
>> alias=x.x.x.x
>> alias=x.x.x.x
>>
>> check_via=yes # (cmd. line: -v)
>> dns=no # (cmd. line: -r)
>> rev_dns=no # (cmd. line: -R)
>> port=5060
>> children=4
>> fifo="/tmp/ser_fifo"
>> sip_warning=yes
>> syn_branch=no
>> server_signature=yes
>> reply_to_via=no
>>
>> # ------------------ module loading
----------------------------------
>>
>> # Uncomment this if you want to use SQL database
>> loadmodule "/usr/lib/ser/modules/mysql.so"
>> loadmodule "/etc/lib/ser/modules/sl.so"
>> loadmodule "/etc/lib/ser/modules/tm.so"
>> loadmodule "/etc/lib/ser/modules/rr.so"
>> loadmodule "/etc/lib/ser/modules/maxfwd.so"
>> loadmodule "/etc/lib/ser/modules/usrloc.so"
>> loadmodule "/etc/lib/ser/modules/registrar.so"
>> loadmodule "/etc/lib/ser/modules/auth.so"
>> loadmodule "/etc/lib/ser/modules/auth_db.so"
>>
>> # ----------------- setting module-specific parameters
---------------
>>
>> # -- usrloc params --
>>
>> #modparam("usrloc", "db_mode", 0)
>> modparam("usrloc", "db_mode", 2)
>> modparam("auth_db", "db_url",
"sql://ser:heslo@localhost/ser")
>>
>> # -- auth params --
>> modparam("auth_db", "db_url",
"sql://ser:heslo@localhost/ser")
>> modparam("auth_db", "calculate_ha1", yes)
>> 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 > max_len ) {
>> 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
>> record_route();
>> # loose-route processing
>> if (loose_route()) {
>> t_relay();
>> 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("arialink.com",
"subscriber"))
>
>
{
>> # www_challenge("arialink.com",
"0");
>> # break;
>> # };
>>
>> save("location");
>> break;
>> };
>>
>> ## native SIP destinations are handled using our USRLOC
DB
if (uri=~"^sip:9742146@.*") {
# attempt handoff to PSTN.
log( "Forwarding to PSTN\n" );
rewritehostport( "x.x.x.x:5060" );
forward(uri:host, uri:port);
};
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
};
}
AJ Grinnell
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The University of Pennsylvania
3401 Walnut Street, Suite 221A
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voice: 215-573-8396
215-746-7903
fax: 215-898-9348
sip:blairs@upenn.edu
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The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104
voice: 215-573-8396
215-746-7903
fax: 215-898-9348
sip:blairs@upenn.edu
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The University of Pennsylvania
3401 Walnut Street, Suite 221A
Philadelphia, PA 19104
voice: 215-573-8396
215-746-7903
fax: 215-898-9348
sip:blairs@upenn.edu
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believed to be clean. For more information please email
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