[Serusers] Cisco AS 5300 can't send BYE for SER... It's bug?
Sahria Hao
sahria.hao at gmail.com
Mon Feb 5 18:06:18 CET 2007
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
>
>
> ------------------------------
>
>
>
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>
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>
>
>
>
>
> --
> ----------
> shosuke
> msn : anseie at hotmail.co.jp
> email : sahria.hao at gmail.com
>
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