[Serusers] Cisco AS 5300 can't send BYE for SER... It's bug?

Sahria Hao sahria.hao at gmail.com
Tue Feb 6 13:20:13 UTC 2007


Thanks for Harry and Greger.

Then, I test incoming and outgoing call for capture SIP messages by cisco as
5300.


Feb  6 11:54:45.028: Received:
INVITE sip:056708077771111 at MY.AS5300.IP.ADDRESS:5060 SIP/2.0
Record-Route: <sip:MY.SER.IP.ADDRESS;ftag=as2deea38f;lr=on>
Via: SIP/2.0/UDP MY.SER.IP.ADDRESS;branch=z9hG4bK2b5e.7cc1de11.0
Via: SIP/2.0/UDP MY.ASTERISK.IP.ADDRESS
:5060;branch=z9hG4bK687f6fc7;rport=5060
From: "12" <sip:0355558888 at srv5.agile.ne.jp>;tag=as2deea38f
To: <sip:08077771111 at MY.SER.IP.ADDRESS>
Contact: <sip:0355558888 at MY.ASTERISK.IP.ADDRESS>
Call-ID: 089003277f7ea22d113bd56b186b6bc1 at MY.SER.IP.ADDRESS
CSeq: 103 INVITE
User-Agent: Asterisk
Max-Forwards: 16
Remote-Party-ID: "12" <sip:0355558888 at MY.SER.IP.ADDRESS
>;privacy=off;screen=no
Date: Tue, 06 Feb 2007 11:54:45 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 222
(From: "12" is an UA registered on Asterisk)

And I found it.

Via: SIP/2.0/UDP MY.SER.IP.ADDRESS;branch=z9hG4bK2b5e.7cc1de11.0

Why SER Via haven't port (5060) number or rport ?
(Asterisk Via have port number and rport)

Thanks,

Sahria
07/02/06 に Greger V. Teigre <greger at teigre.com> さんは書きました:
>
> Hm. I've worked with 5300s without any problems... But I haven't done the
> cisco config, though...
> I would have tried to listen directly on the Cisco network port to see if
> any packet shows up. Of course, debugging turned on to see what happens.
> Upgrade IOS => still no change, and I would've filed a ticket with Cisco.
> g-)
>
> Sahria Hao wrote:
>
> Hi Edson.
>
> I want to know that why my Cisco AS 5300 didn't send BYE for SER...?
>
> Maybe... I doubt that maybe my 5300 have only dial-peer 6000 voice "POTS"
> configure for outgoing PSTN call.
> In case of PSTN incoming call have no problem about sending BYE for SER,
> Because it is apply dial-peer voice 5000 "VOIP" confiure as follows:
>
>   [SER] <- [5300 (VOIP dial-peer)] <- [PSTN]
>
> So I'll try to re-configure my 5300 dial-peer, or
> please give me a hint If anyone have some way to solve this problem.
>
> Thanks,
>
> Sahria
>
> 2007/2/6, Edson <4lists at 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 at lists.iptel.org [mailto:serusers-bounces at lists.iptel.org]
> > *On Behalf Of *Sahria Hao
> > *Sent:* segunda-feira, 5 de fevereiro de 2007 08:36
> > *To:* serusers at 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 at 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<.*@%28my%5C.cisco%5C.ip%5C.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 <.*@%28my%5C.cisco%5C.ip%5C.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 at lists.iptel.org
> >
> >
> > http://lists.iptel.org/mailman/listinfo/serusers
> >
> >
> >
> >
> >
> >
> > --
> > ----------
> > shosuke
> > msn : anseie at hotmail.co.jp
> > email : sahria.hao at gmail.com
> >
> ------------------------------
>
> _______________________________________________
> Serusers mailing listSerusers at lists.iptel.orghttp://lists.iptel.org/mailman/listinfo/serusers
>
>
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