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
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, orplease 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>: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 finishedBut 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 SERAnd 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-recvvoice service pots
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback ciscovoice service voip
fax protocol t38 ls-redundancy 0 hs-redundancy 0 fallback cisco
sip
min-se 60translation-rule 50
Rule 0 0333 0
Rule 1 ^7777 037777voice class codec 2
codec preference 1 g711ulaw
codec preference 2 g711alawdial-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 1dial-peer voice 6000 pots
application session
max-conn 2
destination-pattern 0333T
progress_ind alert enable 8
translate-outgoing called 50
port 0:DThanks,
Sahria
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--
----------
shosuke
msn : anseie@hotmail.co.jp
email : sahria.hao@gmail.com
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