[SR-Users] Routing problem

[Klaus Darilion]

Hi Jerome!

As your config and the log messages seems to be correct, you should 
debug on the network and verify which SIP messages are sent to which 
location. You can use tcpdump/wireshark or the simple (my preferred 
choice) ngrep tool.

1. install ngrep
apt-get install ngrep

2. capture the SIP traffic
ngrep -d any -W byline -t -q -P "" port 5060

-d any: sniff on every interface (sometimes messages get lopped via the 
loopback interface, using -d any shows you alos the looped messages).
-W byline: nice line-break
-t: timestamp
-q: quiet mode
-P "": replace non printable charcaters with "" (e.g. CR LF)
finally the capture filter: port 5060

Start ngrep on the server where openser is installed and then compare 
the signalling of a successful call 
(sip:number at pabx-alcatel.example.com) with an unsuccessful call.

Once that work, it should be possible to solve the other direction too, 
e.g. by defining a trunk on the PABX and route certain extensions, e.g. 
390abc to the SIP proxy. Then openser can do the mapping of the address 
and forwarding of the call.

btw: openser 1.3 is rather old. Using Kamailio 3.0 gives you much more 
flexibility when writing the configuration. Anyway, using 1.3 should 
work too, you setup is rather simple, and is for sure not the cause of 
your problem.

regards
klaus


Am 18.05.2010 05:22, schrieb alet at librelogiciel.com:
> Hi,
>
> First I'm a complete SIP newbie.
>
> We've purchased a new pabx which is a complete piece of crap (Alcatel
> OmniPCX OXO) : no LDAP support, no registrar functionnality, and so
> on... and it seems there's no way to upgrade.
>
> This pabx is on a subnet different from the one used by desktop
> computers.
>
> What I'd like to achieve is to be able to install SIP soft phones onto
> desktop computers (like ekiga for example), allow them to call all
> non-IP phones already configured in the pabx, and be called by any
> non-IP phone through the pabx.
>
> Since the pabx doesn't have a registrar mode, it can't associate IP
> addresses with phone numbers. So in ekiga, I can configure the pabx as a
> SIP proxy, and my ekiga can call any normal (non-IP) phone connected to
> the pabx. This works fine, but is only one half of what I want, since I
> want these SIP phones to be callable from non-IP phones.
>
> Enters openser, which I plan to use as a registrar and router.
>
> So I've installed openser 1.3.2 onto a Debian Squeeze box.  I can
> successfully register SIP soft phones to it, and they can call each
> other just fine calling (number at sip.example.com). This openser box is on
> the same subnet as the Alcatel pabx (so different from the one used by
> SIP soft phones).
>
> All our "normal" phone numbers are 6 digits long and begin with 290,
> so I want to route all call numbers beginning with "290" to the Alcatel
> pabx.
>
> So I've modified the default openser.cfg to contain :
>
> --- CUT ---
> port=5060
> listen=udp:10.125.0.1:5060
> ...
> route {
>      ...
>      if (is_method("INVITE")) {
>         setflag(1);
>         if (uri=~"sip:290[0-9]{3}@sip.example.com") {
>              xlog("#### CALL TO NON-IP PHONE ####\n");
>              route(2);
>         }
>      }
>      ...
> }
> ...
> # route[1] was in original openser.cfg
> route[1] {
>           # for INVITEs enable some additional helper routes
>           if (is_method("INVITE")) {
>               t_on_branch("2");
>               t_on_reply("2");
>               t_on_failure("1");
>           }
>
>           if (!t_relay()) {
>               sl_reply_error();
>           };
>           exit;
> }
> #
> # I've added the following
> route[2] {
>          rewritehostport("pabx-alcatel.example.com:5060");
>          xlog("#### Server and port rewritten ####\n");
>          route(1);
> }...
> --- CUT ---
>
> Now in ekiga if I dial sip:number at pabx-alcatel.example.com:5060 it works
> fine as it always did, because this call is direct.
>
> However if I dial sip:number at sip.example.com:5060 the host and port in
> URL are correctly rewritten in sip:number at pabx-alcatel.example.com:5060
> but nothing happens : the call doesn't ring and fails immediately.
>
> In the logs when dialing sip:290081 at sip.example.com:5060 I've got this,
> so I suppose it *SHOULD* work :
>
> --- CUT ---
> /usr/sbin/openser[759]: #### CALL TO NON-IP PHONE ####
> /usr/sbin/openser[759]: #### Server and port rewritten ####
> /usr/sbin/openser[759]: new branch at sip:290081 at pabx-alcatel.example.com:5060
> /usr/sbin/openser[760]: incoming reply
> /usr/sbin/openser[758]: incoming reply
> --- CUT ---
>
> but it doesn't work as expected.
>
> any idea why ?
>
> NB : I understand this would only solve half of my problem, the other
> part will be solved (I hope) with the help of the guy who installed the
> pabx.
>
> Thanks in advance
>
> Jerome Alet
>
>
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