2007/2/6, Edson <4lists@gmail.com>:

I have this same behaviour, but never give it great importance, since we didn't bill incomming calls…

But it would be great to know if it's because of a misconfiguration or a bug… but we notice that many ports become unavaliable (blocked) over time. To release we programmed a reboot every day on 3AM… J

Even with 'ngrep' the BYE, when PSTN side disconnects, didn't show up…

Edson.
From: serusers-bounces@lists.iptel.org [mailto: serusers-bounces@lists.iptel.org] On Behalf Of Sahria Hao
Sent: segunda-feira, 5 de fevereiro de 2007 08:36
To: serusers@lists.iptel.org
Subject: Re: [Serusers] Cisco AS 5300 can't send BYE for SER... It's bug?
Hi Greger,

And I'm very sorry for my poor exposition.

>>Do you get an error on the 5300?

No, my 5300 works well and there's no error.

>> Is it sent, but never reaches SER?

No, when I finished call by PSTN side, 5300 didn't send BYE for SER.
>>Does SER receive, but does not recognize it?

SER didn't receive a message from 5300 entirely.

I think that when I finished this call, 5300 must send a BYE message for SER... but didn't send it.
2007/2/5, Greger V. Teigre <greger@teigre.com>:

09. [Cisco] can't send BYE for SER *****why??*****

What does that mean?! Do you get an error on the 5300? Is it sent, but never reaches SER?
Does SER receive, but does not recognize it?
g-)

Sho Aihara wrote:

Hi all.

I have a problem for the following scenario.
When I make a call for PSTN and on hook by PSTN side,
Cisco As can't send BYE for SER.

01. [UA via Asterisk] dialing "08022223333" -> [SER]
02. [SER] prefix("0333") and rewritehostport("my.cisco.ip.address :5060") -> [Cisco]
03. [Cisco] dial-peer voice 6000 pots, translate-outgoing called from "033308022223333" to "008022223333"
04. [Cisco] process an outbound call to "008022223333" -> [ e.g. Mobile]
05. [e.g. Mobile] Catch call
06. [SER] log CDR start
07. [Cisco] talking
08. [e.g. Mobile] On hook and call disconnect
09. [Cisco] can't send BYE for SER *****why??*****
10. [UA via Asterisk] On hook
11. [UA via Asterisk] Send BYE for SER
12. [SER] log CDR End [Cisco] Call finished

But another scenario, if make a call from PSTN to Asterisk and
on hook by PSTN side, Cisco As send BYE to SER.

01. [e.g. Mobile] dialing "0377771111(Asterisk user number)"
02. [Cisco] receive "77771111" call number
03. [Cisco] dial-peer voice 5000 voip, session target ipv4: my.ser.ip.address -> [SER]
04. [SER] process an incoming call to "0377771111" -> [UA via Asterisk]
05. [UA via Asterisk] Catch call
06. [SER] log CDR start
07. [UA via Asterisk] talking
08. [e.g. Mobile] On hook and call disconnect
09. [Cisco] Send BYE to SER
10. [SER] log CDR End [Cisco] Call finished
11. [UA via Asterisk] receive BYE from SER

And sorry for my diffucult example.

Why Cisco AS 5300 can't send BYE to SER
When PSTN call is disconnected by PSTN side?

My ser.cfg as follows:

# --------------------------------------------------------------------------
# global configuration parameters
# --------------------------------------------------------------------------
fork=no
log_stderror=yes
check_via=no
dns=no
rev_dns=no
listen=my.ser.ip.address
port=5060
fifo="/tmp/ser_fifo"
fifo_db_url="mysql://ser:heslo@localhost/ser"

# --------------------------------------------------------------------------
# module loading
# --------------------------------------------------------------------------
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/textops.so"
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/avpops.so"
loadmodule "/usr/local/lib/ser/modules/permissions.so"
loadmodule "/usr/local/lib/ser/modules/acc.so"
loadmodule "/usr/local/lib/ser/modules/exec.so"

# --------------------------------------------------------------------------
# setting module-specific parameters
# --------------------------------------------------------------------------
modparam("usrloc", "db_mode", 2)
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("rr", "enable_full_lr", 1)
modparam("usrloc", "db_url", " mysql://ser:heslo@localhost/ser")
modparam("auth_db", "db_url", "mysql://ser:heslo@localhost/ser")
modparam("permissions", "db_url", " mysql://ser:heslo@localhost /ser")
modparam("tm", "fr_inv_timer", 27)
modparam("tm", "fr_inv_timer_avp", "inv_timeout")
modparam("permissions", "db_mode", 1)
modparam("permissions", "trusted_table", "trusted")
modparam("acc", "db_url", "mysql://ser:heslo@localhost/ser")
modparam("acc", "db_flag", 2)
modparam("acc", "db_missed_flag", 3)

# --------------------------------------------------------------------------
# route pattern
# --------------------------------------------------------------------------
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()) {
    if (method=="ACK") {
      acc_db_request("01:CallStart\n", "acc");
    };
    if (method=="BYE" || method=="CANCEL") {
      acc_db_request("02:CallEnd\n", "acc");
    };
    t_relay();
    break;
};

if (uri==myself) {
    if (method=="REGISTER") {
      if (!www_authorize("", "subscriber")) {
        www_challenge("", "0");
        break;
      };
      save("location");
      break;
    };

    if (search("^(f|From): .*@(my\.cisco\.ip\.address)")) {
      #PSTN Incoming call from Cisco AS 5300 e.g. 0377771111
      rewritehost("my.asterisk.ip.address ");
    };

    lookup("aliases");

    if (!lookup("location")) {
      if (method=="INVITE" && !search("^(f|From): .*@(my\.cisco\.ip\.address)")) {
        if (!proxy_authorize("", "subscriber")) {
          proxy_challenge("", "0");
          break;
        };
        if (uri=~"^sip:0[0-9]{10}@") {
          # PSTN Outgoing call to Cisco AS 5300 e.g. 08022223333
          prefix("0333");
          rewritehostport("my.cisco.ip.address:5060");
          avp_write("i:45", "inv_timeout");
        } else {
          sl_send_reply("404", "Not Found");
          break;
        };
        consume_credentials();
      };
    };

};

if (!t_relay()) {
    sl_reply_error();
};

}

And my Cisco AS 5300 config as follows:

voice call send-alert
voice rtp send-recv

voice service pots
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco

voice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
sip
min-se 60

translation-rule 50
Rule 0 0333 0
Rule 1 ^7777 037777

voice class codec 2
codec preference 1 g711ulaw
codec preference 2 g711alaw

dial-peer voice 5000 voip
tone ringback alert-no-PI
description ser-asterisk-cisco-test
huntstop
destination-pattern 77771111$
translate-outgoing called 50
voice-class codec 2
session protocol sipv2
session target ipv4:my.ser.ip.address
dtmf-relay rtp-nte
max-conn 1

dial-peer voice 6000 pots
application session
max-conn 2
destination-pattern 0333T
progress_ind alert enable 8
translate-outgoing called 50
port 0:D

Thanks,
Sahria

_______________________________________________
Serusers mailing list
Serusers@lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers

--
----------
shosuke
msn : anseie@hotmail.co.jp
email : sahria.hao@gmail.com