5 Feb
2007
5 Feb
'07
7:06 p.m.
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(a)gmail.com>om>:
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(a)lists.iptel.org [mailto:
serusers-bounces(a)lists.iptel.org] *On Behalf Of *Sahria Hao
*Sent:* segunda-feira, 5 de fevereiro de 2007 08:36
*To:* serusers(a)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(a)teigre.com>om>:
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):
.*(a)(my\.cisco\.ip\.address <.*(a)%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(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
--
----------
shosuke
msn : anseie(a)hotmail.co.jp
email : sahria.hao(a)gmail.com
6 Feb
6 Feb
9:27 a.m.
New subject: [Serusers] Cisco AS 5300 can't send BYE for SER... It's bug?
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:
HiEdson.
I want to knowthat why my Cisco AS 5300 didn't send BYE for SER...?
Maybe... Idoubt that maybe my 5300 haveonlydial-peer 6000voice"POTS"
configurefor outgoing PSTN call.
In case of PSTN incoming call have no problem about sendingBYE 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
pleasegive mea hintIf anyone have some way to solve this problem.
Thanks,
Sahria
2007/2/6, Edson <4lists(a)gmail.com <mailto: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(a)lists.iptel.org
<mailto:serusers-bounces@lists.iptel.org> [mailto:
serusers-bounces(a)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(a)lists.iptel.org <mailto:serusers@lists.iptel.org>
*Subject:* Re: [Serusers] Cisco AS 5300 can't send BYE for SER...
It's bug?
Hi Greger,
AndI'm verysorry 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 callby 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,5300must send aBYE message
for SER... but didn't send it.
2007/2/5, Greger V. Teigre <greger(a)teigre.com
<mailto:greger@teigre.com>>:
09. [Cisco] can't send BYEfor 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 BYEfor 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 BYEfor SER *****why??*****
10. [UA via Asterisk] On hook
11. [UA via Asterisk] Send BYEfor 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] Callfinished
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): .*(a)(my\.cisco\.ip\.address
<mailto:.*@%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):
.*(a)(my\.cisco\.ip\.address
<mailto:.*@%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(a)lists.iptel.org <mailto:Serusers@lists.iptel.org>
http://lists.iptel.org/mailman/listinfo/serusers
--
----------
shosuke
msn : anseie(a)hotmail.co.jp <mailto:anseie@hotmail.co.jp>
email : sahria.hao(a)gmail.com <mailto:sahria.hao@gmail.com>
------------------------------------------------------------------------
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
3:20 p.m.
New subject: [Serusers] Cisco AS 5300 can't send BYE for SER... It's bug?
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(a)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(a)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(a)gmail.com>om>:
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(a)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(a)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.
------------------------------
_______________________________________________
Serusers mailing
listSerusers@lists.iptel.orghttp://lists.iptel.org/mailman/listinfo/serusers
>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(a)teigre.com>om>:
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):
.*(a)(my\.cisco\.ip\.address <.*(a)%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(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
--
----------
shosuke
msn : anseie(a)hotmail.co.jp
email : sahria.hao(a)gmail.com
7 Feb
7 Feb
6:02 a.m.
New subject: [Serusers] Cisco AS 5300 can't send BYE for SER... It's bug?
I got an additional SIP message log from my 5300.
Scenario:
[UA] => [Asterisk] => [SER] => [CISCO AS5300] => [PSTN]
[UA] "12"
[Asterisk] 0355558888(a)MY.ASTERISK.IP.ADDRESS
[SER] MY.SER.IP.ADDRESS
[CISCO AS5300] MY.AS5300.IP.ADDRESS
[PSTN] T1 Line
And maybe I found little bad Via header in
ACK message from Asterisk (via SER) to AS5300.
Feb 7 03:22:34.401: Received:
ACK sip:056708077771111@MY.AS5300.IP.ADDRESS:5060 SIP/2.0
Record-Route: <sip:MY.SER.IP.ADDRESS;ftag=as2ad98fe4;lr>
Via: SIP/2.0/UDP MY.SER.IP.ADDRESS;branch=0
Via: SIP/2.0/UDP MY.ASTERISK.IP.ADDRESS
:5060;branch=z9hG4bK0a0daaf9;rport=5060
...
Why Via header looks like no branch value and port value...?
And all SIP messages have no Route header.
I attached AS5300 SIP message log (see below).
Any ideas?
Sahria
----------------------------------------------------------------------
Feb 7 03:22:27.284: Received:
INVITE sip:056708077771111@MY.AS5300.IP.ADDRESS:5060 SIP/2.0
Record-Route: <sip:MY.SER.IP.ADDRESS;ftag=as2ad98fe4;lr>
Via: SIP/2.0/UDP MY.SER.IP.ADDRESS;branch=z9hG4bK4e7f.1aea06e5.0
Via: SIP/2.0/UDP MY.ASTERISK.IP.ADDRESS
:5060;branch=z9hG4bK7a1f165b;rport=5060
From: "12" <sip:0355558888@MY.SER.IP.ADDRESS>;tag=as2ad98fe4
To: < sip:08077771111@MY.SER.IP.ADDRESS>
Contact: <sip:0355558888@MY.ASTERISK.IP.ADDRESS>
Call-ID: 351d8bb44819a06a4ad0ee7730518ea6(a)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
Proxy-Authorization:
Digest username="0355558888", realm="MY.SER.IP.ADDRESS",
algorithm=MD5, uri=" sip:08077771111@MY.SER.IP.ADDRESS",
nonce="45c9470e4c66d95af025624109fe107c5c292604",
response="f94ac0f87471cc79c86cd2253fecf71f", opaque=""
Date: Wed, 07 Feb 2007 03:22:27 GMT
Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY
Content-Type: application/sdp
Content-Length: 222
Min-SE: 60
v=0
o=root 29956 29957 IN IP4 MY.ASTERISK.IP.ADDRESS
s=session
c=IN IP4 MY.ASTERISK.IP.ADDRESS
t=0 0
m=audio 13250 RTP/AVP 0 101
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
a=silenceSupp:off - - - -
Feb 7 03:22:27.300: Sent:
SIP/2.0 100 Trying
Via: SIP/2.0/UDP MY.SER.IP.ADDRESS;branch=z9hG4bK4e7f.1aea06e5.0,SIP/2.0/UDP
MY.ASTERISK.IP.ADDRESS:5060;branch=z9hG4bK7a1f165b;rport=5060
From: "12" < sip:0355558888@MY.SER.IP.ADDRESS>;tag=as2ad98fe4
To: <sip:08077771111@MY.SER.IP.ADDRESS>;tag=28713870-2583
Date: Wed, 07 Feb 2007 03:22:27 GMT
Call-ID: 351d8bb44819a06a4ad0ee7730518ea6(a)MY.SER.IP.ADDRESS
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow-Events: telephone-event
Content-Length: 0
Feb 7 03:22:28.552: Sent:
SIP/2.0 183 Session Progress
Via: SIP/2.0/UDP MY.SER.IP.ADDRESS;branch=z9hG4bK4e7f.1aea06e5.0,SIP/2.0/UDP
MY.ASTERISK.IP.ADDRESS:5060;branch=z9hG4bK7a1f165b;rport=5060
From: "12" < sip:0355558888@MY.SER.IP.ADDRESS>;tag=as2ad98fe4
To: <sip:08077771111@MY.SER.IP.ADDRESS>;tag=28713870-2583
Date: Wed, 07 Feb 2007 03:22:27 GMT
Call-ID: 351d8bb44819a06a4ad0ee7730518ea6(a)MY.SER.IP.ADDRESS
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow-Events: telephone-event
Contact: <sip:056708077771111@MY.AS5300.IP.ADDRESS:5060>
Record-Route: <sip:MY.SER.IP.ADDRESS;ftag=as2ad98fe4;lr>
Content-Disposition: session;handling=required
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsSIP-GW-UserAgent 792 3780 IN IP4 MY.AS5300.IP.ADDRESS
s=SIP Call
c=IN IP4 MY.AS5300.IP.ADDRESS
t=0 0
m=audio 16672 RTP/AVP 0 101
c=IN IP4 MY.AS5300.IP.ADDRESS
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Feb 7 03:22:34.377: %ISDN-6-CONNECT: Interface Serial0:0 is now connected
to 08077771111 N/A
Feb 7 03:22:34.381: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP MY.SER.IP.ADDRESS;branch=z9hG4bK4e7f.1aea06e5.0,SIP/2.0/UDP
MY.ASTERISK.IP.ADDRESS:5060;branch=z9hG4bK7a1f165b;rport=5060
From: "12" <sip:0355558888@MY.SER.IP.ADDRESS>;tag=as2ad98fe4
To: < sip:08077771111@MY.SER.IP.ADDRESS>;tag=28713870-2583
Date: Wed, 07 Feb 2007 03:22:27 GMT
Call-ID: 351d8bb44819a06a4ad0ee7730518ea6(a)MY.SER.IP.ADDRESS
Server: Cisco-SIPGateway/IOS-12.x
CSeq: 103 INVITE
Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE,
NOTIFY, INFO
Allow-Events: telephone-event
Contact: <sip:056708077771111@MY.AS5300.IP.ADDRESS :5060>
Record-Route: <sip:MY.SER.IP.ADDRESS;ftag=as2ad98fe4;lr>
Content-Type: application/sdp
Content-Length: 237
v=0
o=CiscoSystemsSIP-GW-UserAgent 792 3780 IN IP4 MY.AS5300.IP.ADDRESS
s=SIP Call
c=IN IP4 MY.AS5300.IP.ADDRESS
t=0 0
m=audio 16672 RTP/AVP 0 101
c=IN IP4 MY.AS5300.IP.ADDRESS
a=rtpmap:0 PCMU/8000
a=rtpmap:101 telephone-event/8000
a=fmtp:101 0-16
Feb 7 03:22:34.401: Received:
ACK sip:056708077771111@MY.AS5300.IP.ADDRESS:5060 SIP/2.0
Record-Route: <sip:MY.SER.IP.ADDRESS;ftag=as2ad98fe4;lr>
Via: SIP/2.0/UDP MY.SER.IP.ADDRESS;branch=0
Via: SIP/2.0/UDP MY.ASTERISK.IP.ADDRESS
:5060;branch=z9hG4bK0a0daaf9;rport=5060
From: "12" <sip:0355558888@MY.SER.IP.ADDRESS>;tag=as2ad98fe4
To: < sip:08077771111@MY.SER.IP.ADDRESS>;tag=28713870-2583
Contact: <sip:0355558888@MY.ASTERISK.IP.ADDRESS>
Call-ID: 351d8bb44819a06a4ad0ee7730518ea6(a)MY.SER.IP.ADDRESS
CSeq: 103 ACK
User-Agent: Asterisk
Max-Forwards: 16
Remote-Party-ID: "12" <sip:0355558888@MY.SER.IP.ADDRESS
;privacy=off;screen=no
Content-Length: 0
###############################################
hook on PSTN side, but 5300 doesn't send BYE.
###############################################
Feb 7 03:22:41.577: %ISDN-6-DISCONNECT: Interface Serial0:0 disconnected
from 08077771111 , call lasted 7 seconds
###############################################
hook on Asterisk side, Asterisk send BYE.
###############################################
Feb 7 03:22:47.629: Received:
BYE sip:056708077771111@MY.AS5300.IP.ADDRESS:5060 SIP/2.0
Record-Route: <sip:MY.SER.IP.ADDRESS;ftag=as2ad98fe4;lr>
Via: SIP/2.0/UDP MY.SER.IP.ADDRESS;branch=z9hG4bK1e7f.d4c7e994.0
Via: SIP/2.0/UDP MY.ASTERISK.IP.ADDRESS
:5060;branch=z9hG4bK02d74e09;rport=5060
From: "12" <sip:0355558888@MY.SER.IP.ADDRESS>;tag=as2ad98fe4
To: < sip:08077771111@MY.SER.IP.ADDRESS>;tag=28713870-2583
Call-ID: 351d8bb44819a06a4ad0ee7730518ea6(a)MY.SER.IP.ADDRESS
CSeq: 104 BYE
User-Agent: Asterisk
Max-Forwards: 16
Remote-Party-ID: "12" <sip:0355558888@MY.SER.IP.ADDRESS
;privacy=off;screen=no
Proxy-Authorization:
Digest username="0355558888", realm="MY.SER.IP.ADDRESS",
algorithm=MD5, uri="sip:056708077771111@MY.AS5300.IP.ADDRESS:5060",
nonce="45c9470e4c66d95af025624109fe107c5c292604",
response="f615ca4b679b290671315e85cbba0388", opaque=""
Content-Length: 0
Feb 7 03:22:47.637: %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 2,
ConnectionId 452AF2D8 B59111DB A7A28749 8A4F7B28, SetupTime 12:22:27.293 JST
Wed Feb 7 2007, PeerAddress 0355558888, PeerSubAddress , DisconnectCause 10
, DisconnectText normal call clearing (16), ConnectTime 12:22: 34.383 JST
Wed Feb 7 2007, DisconnectTime 12:22:47.627 JST Wed Feb 7 2007, CallOrigin
2, ChargedUnits 0, InfoType 2, TransmitPackets 626, TransmitBytes 100160,
ReceivePackets 949, ReceiveBytes 151840
Feb 7 03:22:47.641 : %VOIPAAA-5-VOIP_CALL_HISTORY: CallLegType 1,
ConnectionId 452AF2D8 B59111DB A7A28749 8A4F7B28, SetupTime 12:22:27.371 JST
Wed Feb 7 2007, PeerAddress 056708077771111, PeerSubAddress ,
DisconnectCause 10 , DisconnectText normal call clearing (16), ConnectTime
12:22: 34.381 JST Wed Feb 7 2007, DisconnectTime 12:22:47.641 JST Wed Feb 7
2007, CallOrigin 1, ChargedUnits 0, InfoType 2, TransmitPackets 949,
TransmitBytes 151840, ReceivePackets 626, ReceiveBytes 100160
Feb 7 03:22:47.645: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP MY.SER.IP.ADDRESS;branch=z9hG4bK1e7f.d4c7e994.0,SIP/2.0/UDP
MY.ASTERISK.IP.ADDRESS:5060;branch=z9hG4bK02d74e09;rport=5060
From: "12" < sip:0355558888@MY.SER.IP.ADDRESS>;tag=as2ad98fe4
To: <sip:08077771111@MY.SER.IP.ADDRESS>;tag=28713870-2583
Date: Wed, 07 Feb 2007 03:22:47 GMT
Call-ID: 351d8bb44819a06a4ad0ee7730518ea6(a)MY.SER.IP.ADDRESS
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 104 BYE
Feb 7 03:22:47.645: Received:
BYE sip:056708077771111@MY.AS5300.IP.ADDRESS:5060 SIP/2.0
Record-Route: <sip:MY.SER.IP.ADDRESS;ftag=as2ad98fe4;lr>
Via: SIP/2.0/UDP MY.SER.IP.ADDRESS;branch=z9hG4bK1e7f.d4c7e994.0
Via: SIP/2.0/UDP MY.ASTERISK.IP.ADDRESS
:5060;branch=z9hG4bK02d74e09;rport=5060
From: "12" <sip:0355558888@MY.SER.IP.ADDRESS>;tag=as2ad98fe4
To: < sip:08077771111@MY.SER.IP.ADDRESS>;tag=28713870-2583
Call-ID: 351d8bb44819a06a4ad0ee7730518ea6(a)MY.SER.IP.ADDRESS
CSeq: 104 BYE
User-Agent: Asterisk
Max-Forwards: 16
Remote-Party-ID: "12" <sip:0355558888@MY.SER.IP.ADDRESS
;privacy=off;screen=no
Proxy-Authorization:
Digest username="0355558888", realm="MY.SER.IP.ADDRESS",
algorithm=MD5, uri="sip:056708077771111@MY.AS5300.IP.ADDRESS:5060",
nonce="45c9470e4c66d95af025624109fe107c5c292604",
response="f615ca4b679b290671315e85cbba0388", opaque=""
Content-Length: 0
Feb 7 03:22:47.649: Sent:
SIP/2.0 200 OK
Via: SIP/2.0/UDP MY.SER.IP.ADDRESS;branch=z9hG4bK1e7f.d4c7e994.0,SIP/2.0/UDP
MY.ASTERISK.IP.ADDRESS:5060;branch=z9hG4bK02d74e09;rport=5060
From: "12" < sip:0355558888@MY.SER.IP.ADDRESS>;tag=as2ad98fe4
To: <sip:08077771111@MY.SER.IP.ADDRESS>;tag=28713870-2583
Date: Wed, 07 Feb 2007 03:22:47 GMT
Call-ID: 351d8bb44819a06a4ad0ee7730518ea6(a)MY.SER.IP.ADDRESS
Server: Cisco-SIPGateway/IOS-12.x
Content-Length: 0
CSeq: 104 BYE
----------------------------------------------------------------------
****************************************************
old message was deleted. (size too big)
****************************************************
10:51 a.m.
New subject: [Serusers] Cisco AS 5300 can't send BYE for SER... It's bug?
Hi,
rport missing is no problem at all, default 5060 is then used.
The branch=0 in the ACK might be the problem (if Cisco does not match it
with the INVITE (but then it should retransmit the 200 OK reply)).
You can try add syn_branch=0 to the ser.cfg to have the branch id
calculated for ACKs too.
Michal
On Wed, 2007-02-07 at 13:02 +0900, Sahria Hao wrote:
I got an additional SIP message log from my 5300.
Scenario:
[UA] => [Asterisk] => [SER] => [CISCO AS5300] => [PSTN]
[UA] "12"
[Asterisk] 0355558888(a)MY.ASTERISK.IP.ADDRESS
[SER] MY.SER.IP.ADDRESS
[CISCO AS5300] MY.AS5300.IP.ADDRESS
[PSTN] T1 Line
And maybe I found little bad Via header in
ACK message from Asterisk (via SER) to AS5300.
Feb 7 03:22:34.401: Received:
ACK sip:056708077771111@MY.AS5300.IP.ADDRESS:5060 SIP/2.0
Record-Route: <sip:MY.SER.IP.ADDRESS;ftag=as2ad98fe4;lr>
Via: SIP/2.0/UDP MY.SER.IP.ADDRESS;branch=0
Via: SIP/2.0/UDP
MY.ASTERISK.IP.ADDRESS:5060;branch=z9hG4bK0a0daaf9;rport=5060
...
Why Via header looks like no branch value and port value...?
And all SIP messages have no Route header.
I attached AS5300 SIP message log (see below).
Any ideas?
Sahria
1:44 p.m.
New subject: [Serusers] Cisco AS 5300 can't send BYE for SER... It's bug?
You may see some hint in the log of Cisco AS if you enable more detailed
debug, for example:
debug sip all
debug voip ccapi inout
Just be prepared for the huge about of log and system slowdown!
Miklos
Michal Matyska wrote:
Hi,
rport missing is no problem at all, default 5060 is then used.
The branch=0 in the ACK might be the problem (if Cisco does not match it
with the INVITE (but then it should retransmit the 200 OK reply)).
You can try add syn_branch=0 to the ser.cfg to have the branch id
calculated for ACKs too.
Michal
On Wed, 2007-02-07 at 13:02 +0900, Sahria Hao wrote:
I got an additional SIP message log from my
5300.
Scenario:
[UA] => [Asterisk] => [SER] => [CISCO AS5300] => [PSTN]
[UA] "12"
[Asterisk] 0355558888(a)MY.ASTERISK.IP.ADDRESS
[SER] MY.SER.IP.ADDRESS
[CISCO AS5300] MY.AS5300.IP.ADDRESS
[PSTN] T1 Line
And maybe I found little bad Via header in
ACK message from Asterisk (via SER) to AS5300.
Feb 7 03:22:34.401: Received:
ACK sip:056708077771111@MY.AS5300.IP.ADDRESS:5060 SIP/2.0
Record-Route: <sip:MY.SER.IP.ADDRESS;ftag=as2ad98fe4;lr>
Via: SIP/2.0/UDP MY.SER.IP.ADDRESS;branch=0
Via: SIP/2.0/UDP
MY.ASTERISK.IP.ADDRESS:5060;branch=z9hG4bK0a0daaf9;rport=5060
...
Why Via header looks like no branch value and port value...?
And all SIP messages have no Route header.
I attached AS5300 SIP message log (see below).
Any ideas?
Sahria
_______________________________________________
Serusers mailing list
Serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
2:08 p.m.
New subject: [Serusers] Cisco AS 5300 can't send BYE for SER... It's bug?
Hi Michel,
Thank you for your advice.
And finally I successed to get BYE message from Cisco AS 5300! :-)
Cisco setting as follows...
MYAS5300#show run
Building configuration...
(omitted)
!
voice call disc-pi-off
voice call carrier capacity active
voice rtp send-recv
!
(omitted)
Here is an grobal configuration for 5300 voice service.
And I have no change for my ser.cfg.
(I appreciate for your advice, thanks Michel)
So my SER and AS 5300 call scenario makes BYE each other
when call is done (hooked on).
Very thanks.
Sahria
2007/2/7, Miklos Tirpak <miklos(a)iptel.org>rg>:
You may see some hint in the log of Cisco AS if you
enable more detailed
debug, for example:
debug sip all
debug voip ccapi inout
Just be prepared for the huge about of log and system slowdown!
Miklos
Michal Matyska wrote:
--
----------
shosuke
msn : anseie(a)hotmail.co.jp
email : sahria.hao(a)gmail.com
Hi,
rport missing is no problem at all, default 5060 is then used.
The branch=0 in the ACK might be the problem (if Cisco does not match it
with the INVITE (but then it should retransmit the 200 OK reply)).
You can try add syn_branch=0 to the ser.cfg to have the branch id
calculated for ACKs too.
Michal
On Wed, 2007-02-07 at 13:02 +0900, Sahria Hao wrote:
I got an additional SIP message log from my
5300.
Scenario:
[UA] => [Asterisk] => [SER] => [CISCO AS5300] => [PSTN]
[UA] "12"
[Asterisk] 0355558888(a)MY.ASTERISK.IP.ADDRESS
[SER] MY.SER.IP.ADDRESS
[CISCO AS5300] MY.AS5300.IP.ADDRESS
[PSTN] T1 Line
And maybe I found little bad Via header in
ACK message from Asterisk (via SER) to AS5300.
Feb 7 03:22:34.401: Received:
ACK sip:056708077771111@MY.AS5300.IP.ADDRESS:5060 SIP/2.0
Record-Route: <sip:MY.SER.IP.ADDRESS;ftag=as2ad98fe4;lr>
Via: SIP/2.0/UDP MY.SER.IP.ADDRESS;branch=0
Via: SIP/2.0/UDP
MY.ASTERISK.IP.ADDRESS:5060;branch=z9hG4bK0a0daaf9;rport=5060
...
Why Via header looks like no branch value and port value...?
And all SIP messages have no Route header.
I attached AS5300 SIP message log (see below).
Any ideas?
Sahria
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6 comments
4 participants
participants (4)
-
Greger V. Teigre -
Michal Matyska -
Miklos Tirpak -
Sahria Hao