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@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@srv5.agile.ne.jp;tag=as2deea38f To: sip:08077771111@MY.SER.IP.ADDRESS Contact: sip:0355558888@MY.ASTERISK.IP.ADDRESS Call-ID: 089003277f7ea22d113bd56b186b6bc1@MY.SER.IP.ADDRESS CSeq: 103 INVITE User-Agent: Asterisk Max-Forwards: 16 Remote-Party-ID: "12" <sip:0355558888@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@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@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:
- [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.
- [UA via Asterisk] dialing "08022223333" -> [SER]
- [SER] prefix("0333") and rewritehostport("my.cisco.ip.address:5060") -> [Cisco]
- [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.
- [e.g. Mobile] dialing "0377771111(Asterisk user number)"
- [Cisco] receive "77771111" call number
- [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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shosuke msn : anseie@hotmail.co.jp email : sahria.hao@gmail.com
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