I need some newbie help to configure ser so that it forwards all
requests to another server. I don't need mysql authentication, or any
authentication, just a plain proxy that runs on port 5080. Can someone
help me modify the config? All the examples I have seen spend a lot of
time on mysql authentication, but little on the proxy routng.
Chris
Hi Serusers,
I have 2 ATAs, one is V3.1 and one is V3.2, they are both behind NAT and are
behaving differently when registering. With 3.2, the nat_uac_test does not
apparently detect it is behind NAT and leaves the flag at 0 in the location
table. V3.1 works fine.
I can see why nat_uac_test() may be failing, because the ATA seems to have
auto-magically detected its public IP and is using it in the via: header,
thus receive!=via test fails! Also the contact has the public IP, so the
RFC1813 test fails. I have not configured the NAT IP or server in the ATA as
the NAT public IP is dynamic.
Although the v3.2 ATA seems to have figured out its public IP, it still uses
its private IP in the INVITE sdp c= headers, so we get no media. And because
the nat flag is 0, the rtpproxy doesn't kick in and fix things. Notice the
call-id of the register message includes the private ip.
Anyone familiar with ATA firmware versions and features know the magic
incarnation to shut off the "auto-detect public IP", ahem, "feature" in
v3.2? Or any other suggestions for how to make nat_uac_test() get a positive
when it gets one of these smart REGISTERs from behind NAT?
Im running the old version of SER 0.8.12 still where nat_uac_test only has 1
2 or 3 options, not the new one that has "19", would that make a difference,
upgrading to v0.9?
Here are the ngreps of REGISTERs from the 2 versions: (public Ips masked)
V3.1 works fine, nat_uac_test results in flag 1 in location table, rtpproxy
works fine
>From ata v3.1 behind nat1 to ser server:
U <nat1 pub ip>:49121 -> <pub ip of ser>:5060
REGISTER sip:sip.mycompany.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.11:
5060..From: <sip:20210@sip.mycompany.com;user=phone>;tag=933299358..To
: <sip:20210@sip.mycomapny.com;user=phone>..Call-ID: 4033019825(a)192.16
8.0.11..CSeq: 1 REGISTER..Contact:
<sip:20210@192.168.0.11:5060;user=phone;
transport=udp>;expires=3600..User-Agent: Cisco ATA 186 v3.1.0 atasip
(0402
11A)..Content-Length: 0....
And from ata with v3.2
U <pub ip of nat2>:5060 -> <pub ip of ser>:5060
REGISTER sip:sip.mycompany.com SIP/2.0..Via: SIP/2.0/UDP <pub ip of nat2>
:5060;branch=z9hG4bK1c6d78542c385f5e..From: <sip:202111@sip.mycompany.com
;user=phone>;tag=2453464613..To: <sip:202111@sip.mycomapny.com
;user=phone>..Call-ID: 1193202309@172.16.1.100..CSeq: 1 REGISTER..Con
tact: <sip:202111@<pub ip of nat2>:5060;user=phone;transport=udp>;expire
s=3600..User-Agent: Cisco ATA 186 v3.2.0 atasip
(041111A)..Content-Length:
0....
Thx in advance,
Jon
Hi all,
I downloaded serweb-0.9.3.tgz for ser-0.9.3
I just (re)install serweb however in
config/config_data_layer.php i changed.
it's ok for me !!
Regards
Harry
$config->data_sql->table_user_preferences_types="preferences_types";
to
$config->data_sql->table_user_preferences_types="usr_preferences_types";
I set in sql file : scripts/sql/usr_preferences.sql
# $Id: usr_preferences.sql,v 1.2.2.1 2005/06/28
08:50:08 kozlik Exp $
#
# Dumping data for table 'usr_preferences_types'
#
INSERT INTO usr_preferences_types (att_name,
att_rich_type, att_raw_type, att_type_spec,
default_value)
VALUES("fw_voicemail", "boolean", "1", NULL, "0");
INSERT INTO usr_preferences_types (att_name,
att_rich_type, att_raw_type, att_type_spec,
default_value)
VALUES("sw_user_status_visible", "boolean", "1", NULL,
"1");
INSERT INTO usr_preferences_types (att_name,
att_rich_type, att_raw_type, att_type_spec,
default_value)
VALUES("send_daily_missed_calls", "boolean", "1",
NULL, "0");
___________________________________________________________________________
Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger
Téléchargez cette version sur http://fr.messenger.yahoo.com
Hi everybody!
recently I have been testing calls to/from ISDN with a AS5350. My SER
version is 0.8.14 and when I make a call from ISDN towards a SIP UA
registered in the SER the problem is that the AS5350 is no aware of Call
Release.
I explain it in more detail...
1. AS5350 sends an INVITE to the SER with destination number 93222222
2. SER forwards the INVITE to the corresponding SIP UA.
3. UA reponds with a 180 Ringing and it is forwarded to the AS5350 by the
SER
4. When I pick up the phone, the SIP UA sends a 200 OK message, which is
frowarded by the SER as well.
5. AS5350 replies with a ACK to the 200 OK but beware!! the AS5350 has
changed UDP source port!! it is different from the one set in INVITE.
6. The SIP UA hangs up the call by sending a BYE message to the SER (it is
Record Routing)
7. The SER forwards the BYE to the AS5350 but with the old UDP port!!
8. AS5350 sends ICMP Destination Unreachable (Cause code: port unreachable)
every time that the SER retries the BYE.
My guess is that the AS5350 expects the BYE packet through the new UDP
port...
Has anybody found such a problem?
Thanks
Victor
I'm currently running Fedora Core 4, SER 0.9.2, and have followed the
instruction of running:
# pear install -f classkit
and when I run:
# pear list
I receive the following output:
Installed packages:
===================
Package Version State
Archive_Tar 1.1 stable
Console_Getopt 1.2 stable
DB 1.7.6 stable
HTML_Template_IT 1.1 stable
HTTP 1.3.5 stable
Mail 1.1.4 stable
Net_SMTP 1.2.6 stable
Net_Socket 1.0.6 stable
Net_UserAgent_Detect 2.0.1 stable
PEAR 1.3.5 stable
XML_Parser 1.2.6 stable
XML_RPC 1.2.2 stable
classkit 0.4 beta
runkit 0.3.0 beta
but it's still giving me the "Function aggregate_methods() doesn't exists.
Try install Classkit extension. http://pecl.php.net/package/classkit" error.
I even installed the runkit which is supposed to be the next version up.
Can anyone recommend anything to resolve this error?? I'm more of a perl
man, so I'm lost with PHP.
Thanks.
Gareth
Hello Serusers,
I have 2 ATAs, one is V3.1 and one is V3.2, they are both behind NAT and are
behaving differently when registering. With 3.2, the nat_uac_test does not
apparently detect it is behind NAT and leaves the flag at 0 in the location
table. V3.1 works fine.
I can see why nat_uac_test() may be failing, because the ATA seems to have
auto-magically detected its public IP and is using it in the via: header,
thus receive!=via test fails! Also the contact has the public IP, so the
RFC1813 test fails. I have not configured the NAT IP or server in the ATA as
the NAT public IP is dynamic.
Although the v3.2 ATA seems to have figured out its public IP, it still uses
its private IP in the INVITE sdp c= headers, so we get no media. And because
the nat flag is 0, the rtpproxy doesn't kick in and fix things. Notice the
call-id of the register message includes the private ip.
Anyone familiar with ATA firmware versions and features know the magic
incarnation to shut off the "auto-detect public IP", ahem, "feature" in
v3.2? Or any other suggestions for how to make nat_uac_test() get a positive
when it gets one of these smart REGISTERs from behind NAT?
Im running the old version of SER 0.8.12 still where nat_uac_test only has 1
2 or 3 options, not the new one that has "19", would that make a difference,
upgrading to v0.9?
Here are the ngreps of REGISTERs from the 2 versions: (public Ips masked)
V3.1 works fine, nat_uac_test results in flag 1 in location table, rtpproxy
works fine
>From ata v3.1 behind nat1 to ser server:
U <nat1 pub ip>:49121 -> <pub ip of ser>:5060
REGISTER sip:sip.mycompany.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.11:
5060..From: <sip:20210@sip.mycompany.com;user=phone>;tag=933299358..To
: <sip:20210@sip.mycomapny.com;user=phone>..Call-ID: 4033019825(a)192.16
8.0.11..CSeq: 1 REGISTER..Contact:
<sip:20210@192.168.0.11:5060;user=phone;
transport=udp>;expires=3600..User-Agent: Cisco ATA 186 v3.1.0 atasip
(0402
11A)..Content-Length: 0....
And from ata with v3.2
U <pub ip of nat2>:5060 -> <pub ip of ser>:5060
REGISTER sip:sip.mycompany.com SIP/2.0..Via: SIP/2.0/UDP <pub ip of nat2>
:5060;branch=z9hG4bK1c6d78542c385f5e..From: <sip:202111@sip.mycompany.com
;user=phone>;tag=2453464613..To: <sip:202111@sip.mycomapny.com
;user=phone>..Call-ID: 1193202309@172.16.1.100..CSeq: 1 REGISTER..Con
tact: <sip:202111@<pub ip of nat2>:5060;user=phone;transport=udp>;expire
s=3600..User-Agent: Cisco ATA 186 v3.2.0 atasip
(041111A)..Content-Length:
0....
Thx in advance,
Jon
Hi I'm trying to compile the lcr backported module from Onsip.org but
I'm getting the following error:
make[1]: Entering directory `/root/sources/sip_router/modules/lcr'
gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall
-minline-all-stringops -malign-double -falign-loops -mcpu=athlon
-DNAME='"ser"' -DVERSION='"0.9.3"' -DARCH='"i386"' -DOS='"linux"'
-DCOMPILER='"gcc 3.3"' -D__CPU_i386 -D__OS_linux
-DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP
-DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE
-DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024
-DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD
-DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c
lcr_mod.c -o lcr_mod.o
lcr_mod.c: In function `load_gws':
lcr_mod.c:591: error: `HDR_FROM_F' undeclared (first use in this function)
lcr_mod.c:591: error: (Each undeclared identifier is reported only once
lcr_mod.c:591: error: for each function it appears in.)
lcr_mod.c: In function `load_contacts':
lcr_mod.c:890: error: too many arguments to function `next_branch'
lcr_mod.c:912: error: too many arguments to function `next_branch'
make[1]: *** [lcr_mod.o] Error 1
make[1]: Leaving directory `/root/sources/sip_router/modules/lcr'
Does anybody know if I'm missing something?
Regards
Alberto Cruz
hi,
i installed ser with mysql support and authentication on solaris9, i have 2 ordinary phones, connected to a linksys 2phone port pap2 device. what i need a simple ser routing configuration where i can call from one of the phones to the other phone on the same pap2 device using ser.
pls. help.. thnks in advance.
chris.
guys,
just want to ask if i can make ser to account session durations
for billing purposes?
Thanks,
--
Ryan Pagquil
Infodyne Inc. - PhilOnline.com
3603 Antel Global Corporate Center
Doña Julia Vargas Ave.
Ortigas Center Pasig City
Tel: 687-0715
Web: www.philonline.com
i'm Installaing sipsak on a solaris 9, when i run ./configure i got this:
checking ctype.h presence... yes
checking for ctype.h... yes
checking arpa/inet.h usability... yes
checking arpa/inet.h presence... yes
checking for arpa/inet.h... yes
checking getopt.h usability... no
checking getopt.h presence... no
checking for getopt.h... no
configure: error: missing required header (see above)
where can i get the getopt.h? what package will i install?
thnks