Is the c-timer of RFC3261 the same as the fr_inv_timer? Then you're
right:
RC3261
" Update timer C for provisional responses
For an INVITE transaction, if the response is a provisional
response with status codes 101 to 199 inclusive (i.e., anything
but 100), the proxy MUST reset timer C for that client
transaction. The timer MAY be reset to a different value, but
this value MUST be greater than 3 minutes
"
But setting the timer to 3 minutes is much to long for redirection to
voicemail.
Shouldn't timer-c be activated for all transactions (as stated in the
RFC) and not only for calls which will be redirected to the voicemail?
Therefore, a separate timer for redirection to voicemail would be
necessary. The fr_inv_timer would be set to 3 minutes to cancel
neverending transactions, and a separate fr_inv_voicemail_timer can be
set to e.g. 10 seconds and this timer will not be reset after
provisional responses.
regards,
Klaus
> -----Original Message-----
> From: Jiri Kuthan [mailto:jiri@iptel.org]
> Sent: Tuesday, November 18, 2003 12:47 PM
> To: Klaus Darilion; serusers(a)lists.iptel.org
> Subject: Re: [Serusers] fr_inv_timer problem when using SIPPS
>
>
> Thanks, it is very nice to get some real questions :)
>
> I'm now running in my bed, so I may be confused but I think that
> the spec tells 180s to extend life of transaction context (i.e.,
> extend C-timer). I don't necessarily like it, and I think it was
> just a too quick result of a discussion which the sip mailing list
> wanted to complete (roughly during bis-07 time).
>
> I need to review what SER actually does. Not extending the transaction
> lifetime through 18x should be at least a config option.
>
> -jiri
>
> At 12:00 PM 11/18/2003, Klaus Darilion wrote:
> >Hi!
> >
> >I tried voicemail with redirect to the vm-proxy if the user
> doesn't pick up
> >the phone after 10 seconds. It works fine if I call a kphone
> user, but not
> >with SIPPS users. SIPPS sends a 180 Ringing every five seconds which
> >probably resets the timer - if I choose timer intervalls
> shorter than 5
> >seconds it also works with SIPPS.
> >
> >Is this a bug?
> >
> >tried with ser:
> >0.8.12dev-t14
> >0.8.11rc1
> >0.8.11-r1
> >
> >regards,
> >Klaus
> >
> >
> >----------
> >#
> ># $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $
> >#
> ># simple quick-start config script
> >#
> >
> ># ----------- global configuration parameters
> ------------------------
> >
> >debug=3 # debug level (cmd line: -dddddddddd)
> >fork=yes
> >log_stderror=no # (cmd line: -E)
> >
> >/* Uncomment these lines to enter debugging mode
> >debug=7
> >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"
> >alias=obelix.ict.tuwien.ac.at
> ># ------------------ 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"
> >
> ># Uncomment this if you want digest authentication
> ># mysql.so must be loaded !
> >loadmodule "/usr/local/lib/ser/modules/auth.so"
> >loadmodule "/usr/local/lib/ser/modules/auth_db.so"
> >
> ># load the voicemail module
> >#loadmodule "/usr/local/lib/ser/modules/vm.so"
> >
> ># load the enum module
> >loadmodule "/usr/local/lib/ser/modules/enum.so"
> >
> ># load the group module, to verify if a user forwards to voicemail
> >loadmodule "/usr/local/lib/ser/modules/group.so"
> >
> ># load the nathelper module
> >#loadmodule "/usr/local/lib/ser/modules/nathelper.so"
> >
> ># ----------------- setting module-specific parameters
> ---------------
> >
> ># -- 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)
> >modparam("usrloc", "db_url", "sql://ser:ser2000@localhost/ser")
> >
> ># -- 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")
> >modparam("auth_db", "db_url", "sql://ser:ser2000@localhost/ser")
> >
> ># -- rr params --
> ># add value to ;lr param to make some broken UAs happy
> >modparam("rr", "enable_full_lr", 1)
> >
> ># -- voicemail params --
> >#modparam("voicemail", "db_url","sql://ser:ser2000@localhost/ser")
> >
> ># -- voicemail params --
> >modparam("group", "db_url","sql://serro:serro2000@localhost/ser")
> >
> ># -- nathelper params --
> >#modparam("nathelper", "natping_interval", 10)
> >
> >modparam("tm", "fr_inv_timer", 8 )
> >modparam("tm", "fr_inv_timer", 8 )
> >
> ># ------------------------- request routing logic
> -------------------
> >
> ># main routing logic
> >
> >route{
> >
> > lookup("aliases");
> > # 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 > max_len) {
> > if (len_gt( max_len )) {
> > 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
> >
> >#remove record route for dissipate
> >#if (!(method=="REGISTER")) record_route();
> >
> > # loose-route processing
> > if (loose_route()) {
> > t_relay();
> > 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 (method=="REGISTER") {
> >
> ># Uncomment this if you want to use digest authentication
> ># if (!www_authorize("obelix.ict.tuwien.ac.at",
> >"subscriber")) {
> >#
> www_challenge("obelix.ict.tuwien.ac.at",
> >"0");
> ># break;
> ># };
> >
> > if (!save("location")) {
> > sl_reply_error();
> > };
> > break;
> > };
> >
> > # check if number beginns with 00, then
> convert it into a +
> > if (uri=~"^sip:00[0-9]*@") {
> > # strip booth leading "0"
> > strip(2);
> > prefix("+");
> > };
> > # check if request uri begins with an
> internation phone
> >number, if yes, try enum to resolve
> > if (uri=~"sip:\+?[0-9]+@.*") {
> > enum_query("voice");
> > };
> >
> > #mark transaction for voicemail
> > if (is_user_in("Request-URI", "voicemail")) {
> > setflag(4);
> > };
> >
> > # native SIP destinations are handled using
> our USRLOC DB
> > if (!lookup("location")) {
> > # handle user which was not found
> > route(4);
> > break;
> > };
> > };
> >
> > #add failure route which should be performed if
> response code >=300
> > if (method=="INVITE" && isflagset(4)) {
> > t_on_failure("1");
> > };
> >
> > # forward to current uri now; use stateful forwarding; that
> > # works reliably even if we forward from TCP to UDP
> > if (!t_relay()) {
> > sl_reply_error();
> > };
> >
> >}
> >
> >route[4]{
> >
> > # non-Voip -- just send "off-line"
> > if (!(method == "INVITE" || method == "ACK" ||
> method == "CANCEL" ||
> >method == "REFER" || method == "BYE")) {
> > sl_send_reply("404", "Not Found");
> > break;
> > };
> >
> > # not voicemail subscriber
> > if (!isflagset(4)) {
> > sl_send_reply("404", "Not Found and no
> voicemail turned
> >on");
> > break;
> > };
> >
> > # forward to voicemail now
> > #rewritehostport("machtnix.ict.tuwien.ac.at:5060");
> > t_relay_to_udp("machtnix.ict.tuwien.ac.at", "5060");
> >
> > #klaus
> > #forward(machtnix.ict.tuwien.ac.at, 5060)
> >}
> >
> >
> >
> >failure_route[1] {
> > /* XX: note: unsafe if preloaded routes without username used */
> > revert_uri();
> > revert_uri();
> > #rewritehostport("machtnix.ict.tuwien.ac.at:5060");
> > append_branch();
> > t_relay_to_udp("machtnix.ict.tuwien.ac.at", "5060");
> >}
> >
> >
> >_______________________________________________
> >Serusers mailing list
> >serusers(a)lists.iptel.org
> >http://lists.iptel.org/mailman/listinfo/serusers
>
> --
> Jiri Kuthan http://iptel.org/~jiri/
>
>
You have to enable accounting!
http://www.iptel.org/ser/doc/seruser/seruser.html#AEN892
and
http://www.iptel.org/ser/doc/seruser/seruser.html#MISSEDCALLS
regards,
Klaus
> -----Original Message-----
> From: Kannaiyan Natesan [mailto:nkans@lycos.co.uk]
> Sent: Tuesday, November 18, 2003 12:17 PM
> To: serusers(a)lists.iptel.org
> Subject: [Serusers] Missed Calls Record --
>
>
> I was checking my mysql db table 'missed_calls' by making a
> missed called to
> one of my sip clients. But I could not see the missed call
> record in the
> database.
>
> Can anyone help me where I'm going wrong with this ?
>
> Kannaiyan
>
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
>
>
Thanks, it is very nice to get some real questions :)
I'm now running in my bed, so I may be confused but I think that
the spec tells 180s to extend life of transaction context (i.e.,
extend C-timer). I don't necessarily like it, and I think it was
just a too quick result of a discussion which the sip mailing list
wanted to complete (roughly during bis-07 time).
I need to review what SER actually does. Not extending the transaction
lifetime through 18x should be at least a config option.
-jiri
At 12:00 PM 11/18/2003, Klaus Darilion wrote:
>Hi!
>
>I tried voicemail with redirect to the vm-proxy if the user doesn't pick up
>the phone after 10 seconds. It works fine if I call a kphone user, but not
>with SIPPS users. SIPPS sends a 180 Ringing every five seconds which
>probably resets the timer - if I choose timer intervalls shorter than 5
>seconds it also works with SIPPS.
>
>Is this a bug?
>
>tried with ser:
>0.8.12dev-t14
>0.8.11rc1
>0.8.11-r1
>
>regards,
>Klaus
>
>
>----------
>#
># $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $
>#
># simple quick-start config script
>#
>
># ----------- global configuration parameters ------------------------
>
>debug=3 # debug level (cmd line: -dddddddddd)
>fork=yes
>log_stderror=no # (cmd line: -E)
>
>/* Uncomment these lines to enter debugging mode
>debug=7
>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"
>alias=obelix.ict.tuwien.ac.at
># ------------------ 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"
>
># Uncomment this if you want digest authentication
># mysql.so must be loaded !
>loadmodule "/usr/local/lib/ser/modules/auth.so"
>loadmodule "/usr/local/lib/ser/modules/auth_db.so"
>
># load the voicemail module
>#loadmodule "/usr/local/lib/ser/modules/vm.so"
>
># load the enum module
>loadmodule "/usr/local/lib/ser/modules/enum.so"
>
># load the group module, to verify if a user forwards to voicemail
>loadmodule "/usr/local/lib/ser/modules/group.so"
>
># load the nathelper module
>#loadmodule "/usr/local/lib/ser/modules/nathelper.so"
>
># ----------------- setting module-specific parameters ---------------
>
># -- 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)
>modparam("usrloc", "db_url", "sql://ser:ser2000@localhost/ser")
>
># -- 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")
>modparam("auth_db", "db_url", "sql://ser:ser2000@localhost/ser")
>
># -- rr params --
># add value to ;lr param to make some broken UAs happy
>modparam("rr", "enable_full_lr", 1)
>
># -- voicemail params --
>#modparam("voicemail", "db_url","sql://ser:ser2000@localhost/ser")
>
># -- voicemail params --
>modparam("group", "db_url","sql://serro:serro2000@localhost/ser")
>
># -- nathelper params --
>#modparam("nathelper", "natping_interval", 10)
>
>modparam("tm", "fr_inv_timer", 8 )
>modparam("tm", "fr_inv_timer", 8 )
>
># ------------------------- request routing logic -------------------
>
># main routing logic
>
>route{
>
> lookup("aliases");
> # 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 > max_len) {
> if (len_gt( max_len )) {
> 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
>
>#remove record route for dissipate
>#if (!(method=="REGISTER")) record_route();
>
> # loose-route processing
> if (loose_route()) {
> t_relay();
> 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 (method=="REGISTER") {
>
># Uncomment this if you want to use digest authentication
># if (!www_authorize("obelix.ict.tuwien.ac.at",
>"subscriber")) {
># www_challenge("obelix.ict.tuwien.ac.at",
>"0");
># break;
># };
>
> if (!save("location")) {
> sl_reply_error();
> };
> break;
> };
>
> # check if number beginns with 00, then convert it into a +
> if (uri=~"^sip:00[0-9]*@") {
> # strip booth leading "0"
> strip(2);
> prefix("+");
> };
> # check if request uri begins with an internation phone
>number, if yes, try enum to resolve
> if (uri=~"sip:\+?[0-9]+@.*") {
> enum_query("voice");
> };
>
> #mark transaction for voicemail
> if (is_user_in("Request-URI", "voicemail")) {
> setflag(4);
> };
>
> # native SIP destinations are handled using our USRLOC DB
> if (!lookup("location")) {
> # handle user which was not found
> route(4);
> break;
> };
> };
>
> #add failure route which should be performed if response code >=300
> if (method=="INVITE" && isflagset(4)) {
> t_on_failure("1");
> };
>
> # forward to current uri now; use stateful forwarding; that
> # works reliably even if we forward from TCP to UDP
> if (!t_relay()) {
> sl_reply_error();
> };
>
>}
>
>route[4]{
>
> # non-Voip -- just send "off-line"
> if (!(method == "INVITE" || method == "ACK" || method == "CANCEL" ||
>method == "REFER" || method == "BYE")) {
> sl_send_reply("404", "Not Found");
> break;
> };
>
> # not voicemail subscriber
> if (!isflagset(4)) {
> sl_send_reply("404", "Not Found and no voicemail turned
>on");
> break;
> };
>
> # forward to voicemail now
> #rewritehostport("machtnix.ict.tuwien.ac.at:5060");
> t_relay_to_udp("machtnix.ict.tuwien.ac.at", "5060");
>
> #klaus
> #forward(machtnix.ict.tuwien.ac.at, 5060)
>}
>
>
>
>failure_route[1] {
> /* XX: note: unsafe if preloaded routes without username used */
> revert_uri();
> revert_uri();
> #rewritehostport("machtnix.ict.tuwien.ac.at:5060");
> append_branch();
> t_relay_to_udp("machtnix.ict.tuwien.ac.at", "5060");
>}
>
>
>_______________________________________________
>Serusers mailing list
>serusers(a)lists.iptel.org
>http://lists.iptel.org/mailman/listinfo/serusers
--
Jiri Kuthan http://iptel.org/~jiri/
Hi!
I tried voicemail with redirect to the vm-proxy if the user doesn't pick up
the phone after 10 seconds. It works fine if I call a kphone user, but not
with SIPPS users. SIPPS sends a 180 Ringing every five seconds which
probably resets the timer - if I choose timer intervalls shorter than 5
seconds it also works with SIPPS.
Is this a bug?
tried with ser:
0.8.12dev-t14
0.8.11rc1
0.8.11-r1
regards,
Klaus
----------
#
# $Id: ser.cfg,v 1.21 2003/06/04 13:47:36 jiri Exp $
#
# simple quick-start config script
#
# ----------- global configuration parameters ------------------------
debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
/* Uncomment these lines to enter debugging mode
debug=7
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"
alias=obelix.ict.tuwien.ac.at
# ------------------ 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"
# Uncomment this if you want digest authentication
# mysql.so must be loaded !
loadmodule "/usr/local/lib/ser/modules/auth.so"
loadmodule "/usr/local/lib/ser/modules/auth_db.so"
# load the voicemail module
#loadmodule "/usr/local/lib/ser/modules/vm.so"
# load the enum module
loadmodule "/usr/local/lib/ser/modules/enum.so"
# load the group module, to verify if a user forwards to voicemail
loadmodule "/usr/local/lib/ser/modules/group.so"
# load the nathelper module
#loadmodule "/usr/local/lib/ser/modules/nathelper.so"
# ----------------- setting module-specific parameters ---------------
# -- 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)
modparam("usrloc", "db_url", "sql://ser:ser2000@localhost/ser")
# -- 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")
modparam("auth_db", "db_url", "sql://ser:ser2000@localhost/ser")
# -- rr params --
# add value to ;lr param to make some broken UAs happy
modparam("rr", "enable_full_lr", 1)
# -- voicemail params --
#modparam("voicemail", "db_url","sql://ser:ser2000@localhost/ser")
# -- voicemail params --
modparam("group", "db_url","sql://serro:serro2000@localhost/ser")
# -- nathelper params --
#modparam("nathelper", "natping_interval", 10)
modparam("tm", "fr_inv_timer", 8 )
modparam("tm", "fr_inv_timer", 8 )
# ------------------------- request routing logic -------------------
# main routing logic
route{
lookup("aliases");
# 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 > max_len) {
if (len_gt( max_len )) {
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
#remove record route for dissipate
#if (!(method=="REGISTER")) record_route();
# loose-route processing
if (loose_route()) {
t_relay();
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 (method=="REGISTER") {
# Uncomment this if you want to use digest authentication
# if (!www_authorize("obelix.ict.tuwien.ac.at",
"subscriber")) {
# www_challenge("obelix.ict.tuwien.ac.at",
"0");
# break;
# };
if (!save("location")) {
sl_reply_error();
};
break;
};
# check if number beginns with 00, then convert it into a +
if (uri=~"^sip:00[0-9]*@") {
# strip booth leading "0"
strip(2);
prefix("+");
};
# check if request uri begins with an internation phone
number, if yes, try enum to resolve
if (uri=~"sip:\+?[0-9]+@.*") {
enum_query("voice");
};
#mark transaction for voicemail
if (is_user_in("Request-URI", "voicemail")) {
setflag(4);
};
# native SIP destinations are handled using our USRLOC DB
if (!lookup("location")) {
# handle user which was not found
route(4);
break;
};
};
#add failure route which should be performed if response code >=300
if (method=="INVITE" && isflagset(4)) {
t_on_failure("1");
};
# forward to current uri now; use stateful forwarding; that
# works reliably even if we forward from TCP to UDP
if (!t_relay()) {
sl_reply_error();
};
}
route[4]{
# non-Voip -- just send "off-line"
if (!(method == "INVITE" || method == "ACK" || method == "CANCEL" ||
method == "REFER" || method == "BYE")) {
sl_send_reply("404", "Not Found");
break;
};
# not voicemail subscriber
if (!isflagset(4)) {
sl_send_reply("404", "Not Found and no voicemail turned
on");
break;
};
# forward to voicemail now
#rewritehostport("machtnix.ict.tuwien.ac.at:5060");
t_relay_to_udp("machtnix.ict.tuwien.ac.at", "5060");
#klaus
#forward(machtnix.ict.tuwien.ac.at, 5060)
}
failure_route[1] {
/* XX: note: unsafe if preloaded routes without username used */
revert_uri();
revert_uri();
#rewritehostport("machtnix.ict.tuwien.ac.at:5060");
append_branch();
t_relay_to_udp("machtnix.ict.tuwien.ac.at", "5060");
}
Adrian Georgescu wrote:
> I just did a cvs update on the devel tree, I find nothing like this. Is
> this the right CVS?
Ah, OK, probably it is in my private version, I'll commit it into the
trunk today or tomorrow.
Sorry for confusion.
-Maxim
>
> Adrian
>
> On Tuesday, Nov 18, 2003, at 10:05 Europe/Amsterdam, Maxim Sobolev wrote:
>
>> Adrian Georgescu wrote:
>>
>>> How can I have SER instruct RTPProxy to stop relaying after a BYE is
>>> received?
>>> RTPproxy stops now only after the predefined timeout of 60 seconds.
>>
>>
>> You can use unforce_rtp_proxy() command available in the latest cvs
>> version of the module. This command asks rtp proxy to delete session
>> immediately, something along the lines:
>>
>> if (method == "BYE" || method == "CANCEL") {
>> unforce_rtp_proxy();
>> };
>>
>> -Maxim
>>
>>> Adrian Georgescu
>>> ag(a)ag-projects.com
>>> http://ag-projects.com
>>> Tel: +31-23-5458104
>>> IP phone: sip:ag@ag-projects.com
>>> ------------------------------------------------
>>> DNS, ENUM & IP telephony http://managed-dns.org/
>>> _______________________________________________
>>> Serusers mailing list
>>> serusers(a)lists.iptel.org
>>> http://lists.iptel.org/mailman/listinfo/serusers
>>
>>
>>
>
>
>
How can I have SER instruct RTPProxy to stop relaying after a BYE is
received?
RTPproxy stops now only after the predefined timeout of 60 seconds.
Adrian Georgescu
ag(a)ag-projects.com
http://ag-projects.com
Tel: +31-23-5458104
IP phone: sip:ag@ag-projects.com
------------------------------------------------
DNS, ENUM & IP telephony http://managed-dns.org/
Hello,
Is there a function in nathelper to detect whether the called party is
behind NAT or not? This is useful to decide if RTPProxy should be used
or not. Nathelper maintains somehow this information in memory in
order to UDP ping the phones behind NAT.
Regards,
Adrian Georgescu
ag(a)ag-projects.com
http://ag-projects.com
Tel: +31-23-5458104
IP phone: sip:ag@ag-projects.com
------------------------------------------------
DNS, ENUM & IP telephony http://managed-dns.org/
Does this mean that we can't call a SIP phone and redirect it to voicemail
after some time when nobody picks the phone? I have put these two statements
in ser.cfg but they don't seem to work!
modparam("tm", "fr_inv_timer", 15)
modparam("tm", "fr_timer", 10)
I am getting the voicemail greeting directly after I call the SIP phone?
Please let me know. Thank you.
-----Original Message-----
From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org]On
Behalf Of Jan Janak
Sent: Thursday, November 13, 2003 5:41 PM
To: Samy Touati
Cc: serusers(a)lists.iptel.org
Subject: Re: [Serusers] voicemail working finally
Hello,
dial voicemail allows you to call your own voicemail, the server will
first send an INVITE to your phone and then REFER to redirect it to the
voicemail. Because voicemail currently has no IVR interface, you can
only record a message.
Jan.
On 12-11 12:46, Samy Touati wrote:
> Hi,
>
> I finally got the vm working, thanks for the help.
>
> In serweb there's a dial voicemail, what doesn it do ? Doest it
> call the vm and let you retrieve your messages ?
>
>
> Thanks.
>
> Samy.
>
>
> _______________________________________________
> Serusers mailing list
> serusers(a)lists.iptel.org
> http://lists.iptel.org/mailman/listinfo/serusers
_______________________________________________
Serusers mailing list
serusers(a)lists.iptel.org
http://lists.iptel.org/mailman/listinfo/serusers
Hi!
Bitte
1. nicht meinen Namen (meine Adresse - da wird man dann mit 100%iger Wahrscheinlichkeit vollgespammed) im CC an irgendwelche Mailinglisten schicken
2. Solche banalen Sachen bitte nicht vom DAI aus in Mailinglisten posten, da das einen etwas lächerlichen Eindruck machen könnte
3. Einfach ein exsitierendes Init-Script nehmen, kopieren, anpassen - go
Frank
-----Ursprüngliche Nachricht-----
Von: Stephan Schmidt
Gesendet: Mo 17.11.2003 16:44
An: 'serusers(a)iptel.org'
Cc: Frank Steuer; Nicolas Braun; Alexander Nieswand; Harry Behrens
Betreff: Init.d script for Solaris
Hi,
I have installed SER from source distribution on Solaris 8 with MySQL properly installed and the ser.cfg edited according to the Installation Notes. At this point the manual says that the init.d script is not included for this system. Has anyone done such a script for Solaris yet, and what other than the full path to the ser-folder must be included? The manual says that it is possible to adapt one of the given ser.init files from the source distribution, but the rpm and the debian version are very much different, so I do not know which one to adapt.
Regards,
Stephan Schmidt,
Technical University of Berlin
