hi,
i'm installing appache 2.0.54 on solaris 9, i already installed pkges from sunfreeware.com gcc automake, binutils, make, autoconf, etc. i laready did fixed my path /usr/cc/bin/ /usr/ucb ......, i tried ln -s gcc cc, however, i 'm still getting the error.
but i dont have all this packages on my system
SUNWbtool, SUNWsprot, SUNWtoo
for libraries & headers:
SUNWhea, SUNWarc, SUNWlibm, SUNWlibms
SUNWdfbh, SUNWcg6h, SUNWxwinc, SUNWolinc,
SUNWxglh
for 64 bit development:
SUNWarcx, SUNWbtoox, SUNWdplx, SUNWscpux, SUNWsprox,
SUNWtoox, SUNWlmsx, SUNWlmx, SUNWlibCx
for ucb compat:
SUNWsra, SUNWsrh
i do not have the /opt/SUNWsprot directory either.
how can i install this packages, where can i get it? is it downloadable?
thnks.
checking for chosen layout... apr
checking for gcc... gcc
checking for C compiler default output file name... configure: error: C compiler cannot create executables
See `config.log' for more details.
configure failed for srclib/apr
--- gunawan <gun_cool182(a)yahoo.com> wrote:
Hi, All..
I used ser.cfg from getting started document 4a from
www.onsip.org
but I get error 404 user not found when I tried to
call to PSTN number ( I tried 0811xxxxxx)
is something I missed in my ser.cfg????/
the following is my ser.cfg :
debug=3
fork=yes
log_stderror=yes
listen=202.xx.xxx.xxx # put your server IP
address here
port=5060
children=4
dns=no
rev_dns=no
fifo="/tmp/ser_fifo"
fifo_db_url="mysql://ser:heslo@localhost/ser"
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/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/uri.so"
loadmodule "/usr/local/lib/ser/modules/uri_db.so"
loadmodule "/usr/local/lib/ser/modules/mediaproxy.so"
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
loadmodule "/usr/local/lib/ser/modules/domain.so"
loadmodule "/usr/local/lib/ser/modules/avpops.so"
loadmodule "/usr/local/lib/ser/modules/domain.so"
loadmodule "/usr/local/lib/ser/modules/permissions.so"
modparam("auth_db|permissions|uri_db|usrloc","db_url",
"mysql://ser:heslo@localhost/ser")
modparam("auth_db|uri_db|usrloc", "db_url",
"mysql://ser:heslo@localhost/ser")
modparam("auth_db", "calculate_ha1", 1)
modparam("auth_db", "password_column", "password")
modparam("nathelper", "rtpproxy_disable", 1)
modparam("nathelper", "natping_interval", 0)
modparam("mediaproxy","natping_interval", 30)
modparam("mediaproxy","mediaproxy_socket",
"/var/run/mediaproxy.sock")
modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-
clients")
modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-
clients")
modparam("usrloc", "db_mode", 2)
modparam("registrar", "nat_flag", 6)
modparam("rr", "enable_full_lr", 1)
modparam("tm", "fr_inv_timer", 27)
modparam("tm", "fr_inv_timer_avp", "inv_timeout")
modparam("permissions", "db_mode", 1)
modparam("permissions", "trusted_table", "trusted")
alias='pcr.ac.id'
route {
#
------------------------------------------------------------
-----
# Sanity Check Section
#
------------------------------------------------------------
-----
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483", "Too Many Hops");
break;
};
if (msg:len > max_len) {
sl_send_reply("513", "Message
Overflow");
break;
};
#
------------------------------------------------------------
-----
# Record Route Section
#
------------------------------------------------------------
-----
if (method=="INVITE" && client_nat_test("3"))
{
# INSERT YOUR IP ADDRESS HERE
record_route_preset("202.xx.xxx.xxx:5060;nat=yes");
} else if (method!="REGISTER") {
record_route();
};
#
------------------------------------------------------------
-----
# Call Tear Down Section
#
------------------------------------------------------------
-----
if (method=="BYE" || method=="CANCEL") {
end_media_session();
};
#
------------------------------------------------------------
-----
# Loose Route Section
#
------------------------------------------------------------
-----
if (loose_route()) {
if (has_totag() && (method=="INVITE"
|| method=="ACK"))
{
if (client_nat_test("3") ||
search
("^Route:.*;nat=yes")) {
setflag(6);
use_media_proxy();
};
};
route(1);
break;
};
#
------------------------------------------------------------
-----
# Call Type Processing Section
#
------------------------------------------------------------
-----
if (uri!=myself) {
route(5);
route(1);
break;
};
if (uri==myself) {
if (method=="ACK") {
route(6);
break;}
else if (method=="CANCEL") {
route(3);
break;
} else if (method=="INVITE") {
route(3);
break;
} else if (method=="REGISTER") {
route(2);
break;
};
lookup("aliases");
if (uri!=myself) {
route(5);
route(1);
break;
};
if (!lookup("location")) {
sl_send_reply("404", "User Not
Found");
break;
};
};
route(1);
}
route[1] {
#
------------------------------------------------------------
-----
# Default Message Handler
#
------------------------------------------------------------
-----
t_on_reply("1");
if (!t_relay()) {
if (method=="INVITE" || method=="ACK")
{
end_media_session();
};
sl_reply_error();
};
}
route[2] {
#
------------------------------------------------------------
-----
# REGISTER Message Handler
#
------------------------------------------------------------
----
sl_send_reply("100", "Trying");
if (!search("^Contact:\ +\*") &&
client_nat_test("7")) {
setflag(6);
fix_nated_register();
force_rport();
};
if (!www_authorize("pcr.ac.id","subscriber"))
{
www_challenge("pcr.ac.id","0");
break;
};
if (!check_to()) {
sl_send_reply("401", "Unauthorized");
break;
};
consume_credentials();
if (!save("location")) {
sl_reply_error();
};
}
route[3] {
#
------------------------------------------------------------
-----
# CANCEL and INVITE Message Handler
#
------------------------------------------------------------
-----
if (client_nat_test("3")) {
setflag(7);
force_rport();
fix_nated_contact();
};
if (method=="INVITE" && !allow_trusted()) {
if (!proxy_authorize("pcr.ac.id","subscriber")) {
proxy_challenge("pcr.ac.id","0");
break;
} else if (!check_from()) {
sl_send_reply("403", "Use From=ID");
break;
};
consume_credentials();
};
if (uri=~"^sip:081[0-9]*@") {
route(4);
break;
};
lookup("aliases");
if (uri!=myself) {
route(5);
route(1);
break;
};
if (!lookup("location")) {
if (uri=~"^sip:[0-9]{10}@") {
route(4);
break;
};
sl_send_reply("404", "User Not
Found");
break;
};
if (method=="CANCEL") {
route(1);
break;
};
route(5);
route(1);
}
route[4] {
#
-----------------------------------------------------------------
# PSTN Handler
#
-----------------------------------------------------------------
rewritehostport("202.xx.xxx.xxx:5060"); # INSERT YOUR
PSTN GATEWAY IP
ADDRESS
avp_write("i:45", "inv_timeout");
route(5);
route(1);
}
route[5] {
#
-----------------------------------------------------------------
# RTP Proxy Enabler
#
-----------------------------------------------------------------
if (isflagset(6) || isflagset(7)) {
use_media_proxy();
};
}
route[6] {
#
---------------------------------------------------------------------
# ACK Handler
#
---------------------------------------------------------------------
#
---------------------------------------------------------------------
# Aliases Section
#
---------------------------------------------------------------------
lookup("aliases");
if (uri!=myself) {
route(5);
route(1);
break;
};
lookup("location");
route(1);
}
onreply_route[1] {
if ((isflagset(6) || isflagset(7)) &&
(status=~"(180)|(183)|2
[0-9][0-9]")) {
if (!search("^Content-Length:\ +0")) {
use_media_proxy();
};
};
if (client_nat_test("1")) {
fix_nated_contact();
};
}
and I already configured The Cisco Router... when I
tried dial to Cisco gateway from PSTN.... for the
first
few seconds I heard sound which like the router
answered the calls, after that I received busy
tone....
Why did I receive busy tone after the router answered
my call??? Is my configuration in Router wrong????
I already use debug ccsip messages, but nothing
appeared in my console... ????? :?
could somebody help me... thanks all..... v(^_^)v
> --- Steve Blair <blairs(a)isc.upenn.edu> wrote:
>
> >
> >
> > gunawan wrote:
> >
> > >Hi, Steve.....
> > >
> > >I want to ask about dial-peers u provide in ur
> >
> >website(http://mit.edu/sip/sip.edu/ciscoGW.html)...
> > >
> > >
> > >
> > Just for clarification this isn't my site. This is
> > where some of the
> > documents associated
> > with the Internet2 working group of which I am a
> > member.
> >
> > >1. dial-peer voice 680010 voip
> > > description Only peer for inbound
> > to
> > >SIP Proxy 215-746-8001:8009 extensions
> > > huntstop
> > > preference 2
> > > destination-pattern 6800[1-9]
> > > progress_ind setup enable 3
> > > voice-class codec 1
> > > session protocol sipv2
> > > session target sip-server
> > > dtmf-relay rtp-nte
> > > no vad
> > >
> > >what is 680010?????? is that ur phone number????
> > >something related to phone number???
> > >
> > >
> > No it is not my phon enumber. This is an example
> > only. The value following
> > dial-peer voice is just a label to uniquely
> identify
> > that dial-peer. The
> > value
> > specified in the destination-pattern parameter is
> > what is matched
> > against the
> > dialed digits. In this example the 6800[1-9]
> > specifies a range of numbers
> > from 68001 through 68009. These are real Centrex
> > extensions within our
> > numbering plan bu tthey are test numbers reserved
> > for use by our VoIP
> > project.
> > We use 5 digits because that is how many digits
> our
> > Centrex provider hands
> > us in the call setup message.
> >
> > >2. dial-peer voice 61 pots
> > > description Only peer for outbound 5-digit 746
> > >campus calls
> > > translation-profile outgoing Prefix
> > > preference 3
> > > destination-pattern 6....
> > > direct-inward-dial
> > > port 1/0:23
> > > prefix 215746
> > >
> > >why do u use 61???? something related to ur pone
> > >number????
> > >
> > >
> > >
> > Same as above. This is an illustration. The 61 is
> > just a label . In this
> > example
> > 61 is the first dial-peer defined to handle
> outbound
> > calls to Centrex
> > extensions
> > 6xxxx.
> >
> > >3. could u give me some other example
> > configuration,
> > >bcoz I dun use PABX or analog router here.. I
> plug
> > in
> > >my Telephone line direct to VIC2FXO card...
> > >so I Wish that my SIP client can call to PSTN
> > >client...
> > >my Telephone number that I plug to cisco router
> is
> > >62(761) 53808 , 62 is country code, 761 is area
> > kode,
> > >53808 is my phone number...
> > >
> > >
> > >
> > The FXS and FXO peers are similar to those
> mentioned
> > in the document. You
> > may need to fiddle with parameters for your
> specific
> > type of connection.
> > You may need to know if you are using wink-start
> or
> > loop-start signaling,
> > what port to use, etc. A very basic setup would
> > include something like:
> >
> > voice-port 1/0/0
> > input gain 10
> > connection plar opx 32766
> > description FXO 1/0/0 5732767 for VoIP
> >
> > dial-peer voice 91 pots
> > destination-pattern 9
> > port 1/0/0
> > prefix 9
> > !
> >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >__________________________________
> > >Yahoo! Mail Mobile
> > >Take Yahoo! Mail with you! Check email on your
> > mobile phone.
> > >http://mobile.yahoo.com/learn/mail
> > >
> > >
> >
> > --
> >
> > ISC Network Engineering
> > The University of Pennsylvania
> > 3401 Walnut Street, Suite 221A
> > Philadelphia, PA 19104
> >
> >
> > voice: 215-573-8396
> >
> > 215-746-8001
> >
> > fax: 215-898-9348
> >
> > sip:blairs@upenn.edu
> >
> >
>
>
>
>
> ____________________________________________________
>
> Yahoo! Sports
> Rekindle the Rivalries. Sign up for Fantasy Football
>
> http://football.fantasysports.yahoo.com
>
____________________________________________________
Sell on Yahoo! Auctions no fees. Bid on great items.
http://auctions.yahoo.com/
--- gunawan <gun_cool182(a)yahoo.com> wrote:
Hi, All..
I used ser.cfg from getting started document 4a from
www.onsip.org
but I get error 404 user not found when I tried to
call to PSTN number ( I tried 0811xxxxxx)
is something I missed in my ser.cfg????/
the following is my ser.cfg :
debug=3
fork=yes
log_stderror=yes
listen=202.xx.xxx.xxx # put your server IP
address here
port=5060
children=4
dns=no
rev_dns=no
fifo="/tmp/ser_fifo"
fifo_db_url="mysql://ser:heslo@localhost/ser"
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/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
loadmodule "/usr/local/lib/ser/modules/uri.so"
loadmodule "/usr/local/lib/ser/modules/uri_db.so"
loadmodule "/usr/local/lib/ser/modules/mediaproxy.so"
loadmodule "/usr/local/lib/ser/modules/nathelper.so"
loadmodule "/usr/local/lib/ser/modules/textops.so"
loadmodule "/usr/local/lib/ser/modules/domain.so"
loadmodule "/usr/local/lib/ser/modules/avpops.so"
loadmodule "/usr/local/lib/ser/modules/domain.so"
loadmodule "/usr/local/lib/ser/modules/permissions.so"
modparam("auth_db|permissions|uri_db|usrloc","db_url",
"mysql://ser:heslo@localhost/ser")
modparam("auth_db|uri_db|usrloc", "db_url",
"mysql://ser:heslo@localhost/ser")
modparam("auth_db", "calculate_ha1", 1)
modparam("auth_db", "password_column", "password")
modparam("nathelper", "rtpproxy_disable", 1)
modparam("nathelper", "natping_interval", 0)
modparam("mediaproxy","natping_interval", 30)
modparam("mediaproxy","mediaproxy_socket",
"/var/run/mediaproxy.sock")
modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-
clients")
modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-
clients")
modparam("usrloc", "db_mode", 2)
modparam("registrar", "nat_flag", 6)
modparam("rr", "enable_full_lr", 1)
modparam("tm", "fr_inv_timer", 27)
modparam("tm", "fr_inv_timer_avp", "inv_timeout")
modparam("permissions", "db_mode", 1)
modparam("permissions", "trusted_table", "trusted")
alias='pcr.ac.id'
route {
#
------------------------------------------------------------
-----
# Sanity Check Section
#
------------------------------------------------------------
-----
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483", "Too Many Hops");
break;
};
if (msg:len > max_len) {
sl_send_reply("513", "Message
Overflow");
break;
};
#
------------------------------------------------------------
-----
# Record Route Section
#
------------------------------------------------------------
-----
if (method=="INVITE" && client_nat_test("3"))
{
# INSERT YOUR IP ADDRESS HERE
record_route_preset("202.xx.xxx.xxx:5060;nat=yes");
} else if (method!="REGISTER") {
record_route();
};
#
------------------------------------------------------------
-----
# Call Tear Down Section
#
------------------------------------------------------------
-----
if (method=="BYE" || method=="CANCEL") {
end_media_session();
};
#
------------------------------------------------------------
-----
# Loose Route Section
#
------------------------------------------------------------
-----
if (loose_route()) {
if (has_totag() && (method=="INVITE"
|| method=="ACK"))
{
if (client_nat_test("3") ||
search
("^Route:.*;nat=yes")) {
setflag(6);
use_media_proxy();
};
};
route(1);
break;
};
#
------------------------------------------------------------
-----
# Call Type Processing Section
#
------------------------------------------------------------
-----
if (uri!=myself) {
route(5);
route(1);
break;
};
if (uri==myself) {
if (method=="ACK") {
route(6);
break;}
else if (method=="CANCEL") {
route(3);
break;
} else if (method=="INVITE") {
route(3);
break;
} else if (method=="REGISTER") {
route(2);
break;
};
lookup("aliases");
if (uri!=myself) {
route(5);
route(1);
break;
};
if (!lookup("location")) {
sl_send_reply("404", "User Not
Found");
break;
};
};
route(1);
}
route[1] {
#
------------------------------------------------------------
-----
# Default Message Handler
#
------------------------------------------------------------
-----
t_on_reply("1");
if (!t_relay()) {
if (method=="INVITE" || method=="ACK")
{
end_media_session();
};
sl_reply_error();
};
}
route[2] {
#
------------------------------------------------------------
-----
# REGISTER Message Handler
#
------------------------------------------------------------
----
sl_send_reply("100", "Trying");
if (!search("^Contact:\ +\*") &&
client_nat_test("7")) {
setflag(6);
fix_nated_register();
force_rport();
};
if (!www_authorize("pcr.ac.id","subscriber"))
{
www_challenge("pcr.ac.id","0");
break;
};
if (!check_to()) {
sl_send_reply("401", "Unauthorized");
break;
};
consume_credentials();
if (!save("location")) {
sl_reply_error();
};
}
route[3] {
#
------------------------------------------------------------
-----
# CANCEL and INVITE Message Handler
#
------------------------------------------------------------
-----
if (client_nat_test("3")) {
setflag(7);
force_rport();
fix_nated_contact();
};
if (method=="INVITE" && !allow_trusted()) {
if (!proxy_authorize("pcr.ac.id","subscriber")) {
proxy_challenge("pcr.ac.id","0");
break;
} else if (!check_from()) {
sl_send_reply("403", "Use From=ID");
break;
};
consume_credentials();
};
if (uri=~"^sip:081[0-9]*@") {
route(4);
break;
};
lookup("aliases");
if (uri!=myself) {
route(5);
route(1);
break;
};
if (!lookup("location")) {
if (uri=~"^sip:[0-9]{10}@") {
route(4);
break;
};
sl_send_reply("404", "User Not
Found");
break;
};
if (method=="CANCEL") {
route(1);
break;
};
route(5);
route(1);
}
route[4] {
#
-----------------------------------------------------------------
# PSTN Handler
#
-----------------------------------------------------------------
rewritehostport("202.xx.xxx.xxx:5060"); # INSERT YOUR
PSTN GATEWAY IP
ADDRESS
avp_write("i:45", "inv_timeout");
route(5);
route(1);
}
route[5] {
#
-----------------------------------------------------------------
# RTP Proxy Enabler
#
-----------------------------------------------------------------
if (isflagset(6) || isflagset(7)) {
use_media_proxy();
};
}
route[6] {
#
---------------------------------------------------------------------
# ACK Handler
#
---------------------------------------------------------------------
#
---------------------------------------------------------------------
# Aliases Section
#
---------------------------------------------------------------------
lookup("aliases");
if (uri!=myself) {
route(5);
route(1);
break;
};
lookup("location");
route(1);
}
onreply_route[1] {
if ((isflagset(6) || isflagset(7)) &&
(status=~"(180)|(183)|2
[0-9][0-9]")) {
if (!search("^Content-Length:\ +0")) {
use_media_proxy();
};
};
if (client_nat_test("1")) {
fix_nated_contact();
};
}
and I already configured The Cisco Router... when I
tried dial to Cisco gateway from PSTN.... for the
first
few seconds I heard sound which like the router
answered the calls, after that I received busy
tone....
Why did I receive busy tone after the router answered
my call??? Is my configuration in Router wrong????
I already use debug ccsip messages, but nothing
appeared in my console... ????? :?
could somebody help me... thanks all..... v(^_^)v
> --- Steve Blair <blairs(a)isc.upenn.edu> wrote:
>
> >
> >
> > gunawan wrote:
> >
> > >Hi, Steve.....
> > >
> > >I want to ask about dial-peers u provide in ur
> >
> >website(http://mit.edu/sip/sip.edu/ciscoGW.html)...
> > >
> > >
> > >
> > Just for clarification this isn't my site. This is
> > where some of the
> > documents associated
> > with the Internet2 working group of which I am a
> > member.
> >
> > >1. dial-peer voice 680010 voip
> > > description Only peer for inbound
> > to
> > >SIP Proxy 215-746-8001:8009 extensions
> > > huntstop
> > > preference 2
> > > destination-pattern 6800[1-9]
> > > progress_ind setup enable 3
> > > voice-class codec 1
> > > session protocol sipv2
> > > session target sip-server
> > > dtmf-relay rtp-nte
> > > no vad
> > >
> > >what is 680010?????? is that ur phone number????
> > >something related to phone number???
> > >
> > >
> > No it is not my phon enumber. This is an example
> > only. The value following
> > dial-peer voice is just a label to uniquely
> identify
> > that dial-peer. The
> > value
> > specified in the destination-pattern parameter is
> > what is matched
> > against the
> > dialed digits. In this example the 6800[1-9]
> > specifies a range of numbers
> > from 68001 through 68009. These are real Centrex
> > extensions within our
> > numbering plan bu tthey are test numbers reserved
> > for use by our VoIP
> > project.
> > We use 5 digits because that is how many digits
> our
> > Centrex provider hands
> > us in the call setup message.
> >
> > >2. dial-peer voice 61 pots
> > > description Only peer for outbound 5-digit 746
> > >campus calls
> > > translation-profile outgoing Prefix
> > > preference 3
> > > destination-pattern 6....
> > > direct-inward-dial
> > > port 1/0:23
> > > prefix 215746
> > >
> > >why do u use 61???? something related to ur pone
> > >number????
> > >
> > >
> > >
> > Same as above. This is an illustration. The 61 is
> > just a label . In this
> > example
> > 61 is the first dial-peer defined to handle
> outbound
> > calls to Centrex
> > extensions
> > 6xxxx.
> >
> > >3. could u give me some other example
> > configuration,
> > >bcoz I dun use PABX or analog router here.. I
> plug
> > in
> > >my Telephone line direct to VIC2FXO card...
> > >so I Wish that my SIP client can call to PSTN
> > >client...
> > >my Telephone number that I plug to cisco router
> is
> > >62(761) 53808 , 62 is country code, 761 is area
> > kode,
> > >53808 is my phone number...
> > >
> > >
> > >
> > The FXS and FXO peers are similar to those
> mentioned
> > in the document. You
> > may need to fiddle with parameters for your
> specific
> > type of connection.
> > You may need to know if you are using wink-start
> or
> > loop-start signaling,
> > what port to use, etc. A very basic setup would
> > include something like:
> >
> > voice-port 1/0/0
> > input gain 10
> > connection plar opx 32766
> > description FXO 1/0/0 5732767 for VoIP
> >
> > dial-peer voice 91 pots
> > destination-pattern 9
> > port 1/0/0
> > prefix 9
> > !
> >
> > >
> > >
> > >
> > >
> > >
> > >
> > >
> > >__________________________________
> > >Yahoo! Mail Mobile
> > >Take Yahoo! Mail with you! Check email on your
> > mobile phone.
> > >http://mobile.yahoo.com/learn/mail
> > >
> > >
> >
> > --
> >
> > ISC Network Engineering
> > The University of Pennsylvania
> > 3401 Walnut Street, Suite 221A
> > Philadelphia, PA 19104
> >
> >
> > voice: 215-573-8396
> >
> > 215-746-8001
> >
> > fax: 215-898-9348
> >
> > sip:blairs@upenn.edu
> >
> >
>
>
>
>
> ____________________________________________________
>
> Yahoo! Sports
> Rekindle the Rivalries. Sign up for Fantasy Football
>
> http://football.fantasysports.yahoo.com
>
__________________________________________________
Do You Yahoo!?
Tired of spam? Yahoo! Mail has the best spam protection around
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hello!
> Thank you ,Karel! I resolve one question that show "DB Error: no such table" by your guidance.but I have a new question now it is when I want to show accounting and missed calls .it's show me "DB Error: syntax error" why that's for?
> ZhaoMin
> ----- Original Message -----
> From: "Karel Kozlik" <karel(a)iptel.org>
> To: "zhaomin" <zhaomin(a)kingdream.com>
> Cc: <serusers(a)iptel.org>
> Sent: Monday, July 04, 2005 7:39 PM
> Subject: Re: [Serusers] SerWeb0.9.3 show me the error :DB Error: no such table Why?
>
>
> > Hello,
> > ser-0.9.3 and ser cvs useing a little diferent DB schema. There is a
> > confusion in names of tables for user preference. Check the names of
> > tables in config/config_data_layer.php and match them with you DB schema.
> >
> > Karel
> >
> > zhaomin napsal(a):
> > > I used serweb-0.9.3 and Ser CVS version, I can register and login but
> > > when I login and open the my account show me error: DB Error: no such
> > > table.but it's can open the data in the mysql .I have this error prompt
> > > at many place include admin,anyone have this prompt? How can I resolve it?
> > > Thanks advance !
> > > ZhaoMin
> > >
> > >
> > > ------------------------------------------------------------------------
> > >
> > > _______________________________________________
> > > Serusers mailing list
> > > Serusers(a)iptel.org
> > > http://mail.iptel.org/mailman/listinfo/serusers
Valued Colleagues,
I am trying to figure out how ANI and CallerID are handled/processed by
SER
and by SIP in general.
I have understood append_rpid_hf can be used to add a Remote-Party-ID
SIP header field.
Does Remote-Party-ID represent the CallerID or the ANI?!
How does SER include CallerID and ANI in the SIP message?
Can anyone kindly point me in the right direction as to where to
look/read?!
Best Regards,
Ramin
Valued Colleagues,
I am trying to figure out how ANI and CallerID are handled/processed by
SER
and by SIP in general.
I have understood append_rpid_hf can be used to add a Remote-Party-ID
SIP header field.
which I think
Can anyone kindly point me in the right direction as to where to
look/read?!
Best Regards,
Ramin
Valued Colleagues,
I am trying to figure out how ANI and CallerID are handled/processed by
SER
and by SIP in general.
Can anyone kindly point me in the right direction as to where to
look/read?!
Best Regards,
Ramin
Hi all!
I'm trying to set-up user authentification, but it seems like I have some
problem with my ser.cfg This is the output I get after runnin "ser -c":
-----------------------
[root@Linux tmp]# ser -c
0(16552) parse error (158,6-7): syntax error
0(16552) parse error (158,6-7): missing '(' or ')' ?
0(16552) parse error (158,6-7): bad command: missing ';'?
0(16552) parse error (158,7-8): bad command
0(16552) parse error (158,8-9): bad command
0(16552) parse error (159,1-2): bad command
0(16552) parse error (175,1-2): syntax error
0(16552) parse error (175,1-2): bad command
0(16552) parse error (175,1-2): bad command
0(16552) parse error (175,2-3): bad command
0(16552) parse error (175,23-24): syntax error
0(16552) parse error (175,23-24): bad command: missing ';'?
0(16552) parse error (175,23-24): bad command
0(16552) parse error (175,25-26): bad command
0(16552) parse error (178,2-3): syntax error
0(16552) parse error (178,2-3):
ERROR: bad config file (16 errors)
-----------------------
This is my config file:
------------------------------------------------------------
#
# $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
#debug=3 # debug level (cmd line: -dddddddddd)
#fork=yes
#log_stderror=no # (cmd line: -E)
#listen=10.29.1.2
/* Uncomment these lines to enter debugging mode
fork=no
log_stderror=yes
*/
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
#port=5060
children=4
fifo="/tmp/ser_fifo"
fifo_db_url="mysql://ser:heslo@localhost/ser"
# ------------------ module loading ----------------------------------
# Uncomment this if you want to use SQL database
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"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/uri_db.so"
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
# ----------------- setting module-specific parameters ---------------
modparam("auth_db|uri_db|usrloc", "db_url",
"mysql://ser:heslo@localhost/ser")
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("usrloc", "db_mode", 2)
modparam("rr", "enable_full_lr", 1)
# -- usrloc params --
#modparam("usrloc", "db_mode", 0)
# Uncomment this if you want to use SQL database
# for persistent storage and comment the previous line
#modparam("usrloc", "db_mode", 2)
# -- auth params --
# Uncomment if you are using auth module
#
#modparam("auth_db", "calculate_ha1", yes)
#
# If you set "calculate_ha1" parameter to yes (which true in this config),
# uncomment also the following parameter)
#
#modparam("auth_db", "password_column", "password")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
#modparam("rr", "enable_full_lr", 1)
# ------------------------- request routing logic -------------------
##############################################################
route{
# initial sanity checks -- messages with
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
break;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
break;
};
# we record-route all messages -- to make sure that
# subsequent messages will go through our proxy; that's
# particularly good if upstream and downstream entities
# use different transport protocol
if (!method=="REGISTER") record_route();
# subsequent messages withing a dialog should take the
# path determined by record-routing
if (loose_route()) {
# mark routing logic in request
append_hf("P-hint: rr-enforced\r\n");
route(1);
break;
};
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
break;
};
# if the request is for other domain use UsrLoc
# (in case, it does not work, use the following command
# with proper names and addresses in it)
if (uri==myself) {
if (uri==myself) {
if (method=="INVITE") {
route(3);
break;
} else if (method=="REGISTER") {
route(2);
break;
};
# Uncomment this if you want to use digest authentication
# if (!www_authorize("voip.savica.net", "subscriber"))
{
# www_challenge("voip.savica.net", "0");
# break;
# };
# save("location");
# break;
# };
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
break;
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
sl_send_reply("404", "User Not Found");
break;
};
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
}
route[2] {
# -----------------------------------------------------------------
# REGISTER Message Handler
# ----------------------------------------------------------------
sl_send_reply("100", "Trying");
(!www_authorize("","subscriber")) {
www_challenge("","0");
break;
};
if (!check_to()) {
sl_send_reply("401", "Unauthorized");
break;
};
consume_credentials();
if (!save("location")) {
sl_reply_error();
};
}
route[3] {
# -----------------------------------------------------------------
# INVITE Message Handler
# -----------------------------------------------------------------
if (!proxy_authorize("","subscriber")) {
proxy_challenge("","0");
break;
} else if (!check_from()) {
sl_send_reply("403", "Use From=ID");
break;
};
consume_credentials();
lookup("aliases");
if (uri!=myself) {
route(1);
break;
};
if (!lookup("location")) {
sl_send_reply("404", "User Not Found");
break;
};
route(1);
}
-------------------------------------------------
You were right, it was the firewall. Once I added the NAT traversal
rules it worked right away.
What I don't understand is why I need this from the SER installation I
setup while I didn't need to do this to register with my iptel.org account?
If they're both outside the firewall wouldn't this have happened to both?
-mark
Iqbal wrote:
> can ser send a message back to you, run ngrep on ur machine and see
> whathappens after ser has received ur INVITE, because I am guessing it
> might be a FW in between somewhere,
>
> Iqbal
>
> Mark Jeftovic wrote:
>
>>
>> Hi folks, I was experimenting with SER awhile back and am now
>> revisiting it.
>>
>> I've just setup a SER installation using the latest debian packages on
>> a server outside of our office firewall on a public IP address.
>>
>> From home, my laptop on a private IP behind a linksys router/firewall
>> can login/register with my SER server fine. It can also login/register
>> into my iptel.org account as well.
>>
>> Then when I get to the office, behind our firewall, I have problems.
>> I can still login/register my iptel.org account from my laptop on a
>> private IP behind the office firewall, but when I try to log into the
>> SER server I setup it seems to go into a loop (see logs below).
>>
>> I'm using stunserver.org in all cases.
>>
>> I wonder about the line pre_auth(): Credentials with given realm not
>> found but then it works fine using the same login info from home.
>>
>> Also, when all this gives up my softphone says:
>>
>> Registration rejected.
>> Response:Server could not be reached, or it did not respond.
>>
>> Any ideas appreciated.
>>
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: SIP Request:
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: method: <REGISTER>
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: uri:
>> <sip:sip.example.org>
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: version: <SIP/2.0>
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: parse_headers: flags=1
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: end of header reached, state=5
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: parse_headers: Via found,
>> flags=1
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: parse_headers: this is the
>> first vi
>> a
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: After parse_msg...
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: preparing to run routing
>> scripts...
>>
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: DEBUG : is_maxfwd_present:
>> searchin
>> g for max_forwards header
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: parse_headers: flags=128
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: end of header reached, state=8
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: DEBUG: get_hdr_field: <To>
>> [28]; ur
>> i=[sip:markjr@sip.example.org]
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: DEBUG: to body
>> [sip:markjr@sip.easy
>> dns.org^M ]
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: get_hdr_field: cseq <CSeq>:
>> <142240
>> 54> <REGISTER>
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: DEBUG: get_hdr_body :
>> content_lengt
>> h=0
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: found end of header
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: DEBUG: is_maxfwd_present:
>> max_forwa
>> rds header not found!
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: end of header reached, state=8
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: DEBUG: get_hdr_field: <To>
>> [28]; ur
>> i=[sip:markjr@sip.example.org]
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: DEBUG: to body
>> [sip:markjr@sip.easy
>> dns.org^M ]
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: get_hdr_field: cseq <CSeq>:
>> <142240
>> 54> <REGISTER>
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: DEBUG: get_hdr_body :
>> content_lengt
>> h=0
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: found end of header
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: DEBUG: is_maxfwd_present:
>> max_forwa
>> rds header not found!
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: end of header reached, state=8
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: parse_headers: flags=256
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: find_first_route(): No
>> Route header
>> s found
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: loose_route(): There is no
>> Route HF
>>
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: check_self - checking if
>> host==us:
>> 15==9 && [sip.example.org] == [127.0.0.1]
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: check_self - checking if
>> port 5060
>> matches port 5060
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: check_self - checking if
>> host==us:
>> 15==13 && [sip.example.org] == [66.207.199.42]
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: check_self - checking if
>> port 5060
>> matches port 5060
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: DBUG: entering REGISTER branch
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: parse_headers: flags=4096
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: pre_auth(): Credentials
>> with given realm not found
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: build_auth_hf():
>> 'WWW-Authenticate:
>> Digest realm="sip.example.org",
>> nonce="42cacdeb905244dcc9e7ebc71f60fb5aaa658faf"^M '
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: parse_headers: flags=-1
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]:
>> check_via_address(66.207.199.34, 19
>> 2.168.1.18, 0)
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: DEBUG:destroy_avp_list:
>> destroing l
>> ist (nil)
>> Jul 5 14:09:03 tmda /usr/sbin/ser[21028]: receive_msg: cleaning up
>>
>> _______________________________________________
>> Serusers mailing list
>> serusers(a)lists.iptel.org
>> http://lists.iptel.org/mailman/listinfo/serusers
>>
>> .
>>
>
>
Hello,
I've ser running on 192.168.1.222:5061 and asterisk on 192.168.1.222:5060
and I want to forward international calls to asterisk. I'm writing a small
accounting application and I need the methods "ACK" and "BYE".
Here a part of my ser.cfg:
if (method=="INVITE") {
# authenticated and authorized, now accounting is set
setflag(1);
record_route();
};
if (loose_route())
{
# mark the BYEs -- they are requests within a dialog
if(method=="BYE")
setflag(1);
t_relay();
break;
};
if (uri=~"sip:00") {
rewritehostport("192.168.1.222:5060"); //to asterisk...
}
else {
sl_send_reply("404", "Not Found");
break;
};
append_hf("P-hint: GATEWAY\r\n");
if (!t_relay()) {
sl_reply_error();
break;
};
Why don't I get the BYE messages?? I think I made it like it's written in
the documentation
Thanks!
Sebastian
