On Wed, Mar 31, 2010 at 10:28 AM, Anca Vamanu <a...@opensips.org> wrote:
> James Lamanna wrote:
>> Anca Vamanu Wrote:
>>
>>
>>> Andrew, this patch is already in 1.6.2 and trunk.
>>> James, the first thing that you need to check is that you receive
>>> Subscribes from the phones with event 'dialog'. And indeed as Andrew
>>> said, you need to load pua and pua_dialoginfo modules.
>>>
>>
>> Ok thanks. I'll upgrade to 1.6.2.
>> Do I still need to explicitly call dialoginfo_set()?
>>
>>
> Yes, you have to call it.
Hi Anca,
I'm still having problems getting this to work at all.
I've now upgraded to 1.6.2.
Here is my entire config:
debug=3 # debug level (cmd line: -dddddddddd)
fork=yes
log_stderror=no # (cmd line: -E)
log_facility=LOG_LOCAL0
tos=0x60
# Uncomment these lines to enter debugging mode
#fork=no
log_stderror=yes
debug=6
check_via=no # (cmd. line: -v)
dns=no # (cmd. line: -r)
rev_dns=no # (cmd. line: -R)
port=5060
children=4
listen=udp:my.ip.address:5060
listen=udp:my.ip.address:5061
# ------------------ module loading ----------------------------------
#set module path
#mpath="/usr/local/lib/opensips/modules/"
mpath="/usr/local/lib64/opensips/modules/"
# Uncomment this if you want to use SQL database
loadmodule "db_mysql.so"
loadmodule "sl.so"
loadmodule "maxfwd.so"
loadmodule "textops.so"
loadmodule "avpops.so"
loadmodule "tm.so"
loadmodule "rr.so"
loadmodule "dialog.so"
loadmodule "signaling.so"
loadmodule "options.so"
loadmodule "localcache.so"
loadmodule "usrloc.so"
loadmodule "presence.so"
loadmodule "presence_xml.so"
loadmodule "presence_dialoginfo.so"
loadmodule "pua.so"
loadmodule "pua_dialoginfo.so"
#loadmodule "pua_bla.so"
loadmodule "pua_usrloc.so"
loadmodule "registrar.so"
loadmodule "mi_fifo.so"
loadmodule "xlog.so"
# Uncomment this if you want digest authentication
# db_mysql.so must be loaded !
loadmodule "auth.so"
loadmodule "auth_db.so"
# !! Nathelper
loadmodule "nathelper.so"
# ----------------- setting module-specific parameters ---------------
# -- mi_fifo params --
modparam("mi_fifo", "fifo_name", "/tmp/opensips_fifo")
modparam("usrloc", "db_mode", 2)
modparam("usrloc|dialog|dispatcher|presence|presence_xml|pua|avpops",
"db_url", "mysql://opensips:myp...@localhost/opensips")
modparam("avpops","avp_table","usr_preferences")
#modparam("dispatcher", "force_dst", 1)
# Only use username
#modparam("dispatcher", "flags", 1)
# Store passwords for 1 hour in cache
modparam("auth","username_spec","$avp(i:54)")
modparam("auth","password_spec","$avp(i:55)")
modparam("auth","calculate_ha1",1)
modparam("auth_db", "db_url",
"mysql://opensipsro:mypas...@localhost/opensips")
modparam("auth_db", "calculate_ha1", yes)
modparam("auth_db", "password_column", "password")
modparam("auth_db", "load_credentials", "$avp(i:55)=password")
modparam("rr", "enable_full_lr", 1)
modparam("dialog", "dlg_flag", 4)
modparam("dialog", "profiles_with_value", "caller")
modparam("usrloc","nat_bflag",6)
modparam("nathelper","sipping_bflag",8)
#modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT
#modparam("nathelper", "natping_interval", 30)
modparam("nathelper", "sipping_from", "sip:pin...@my.ip.address")
modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy/rtpproxy.sock")
modparam("presence", "server_address", "sip:s...@my.ip.address:5060")
modparam("presence", "expires_offset", 10)
modparam("presence_xml", "force_active", 1)
modparam("presence_dialoginfo", "force_single_dialog", 1)
modparam("pua_dialoginfo", "presence_server", "sip:s...@my.ip.address:5060")
modparam("pua_dialoginfo", "include_callid", 1)
modparam("pua_dialoginfo", "include_tags", 1)
modparam("pua_dialoginfo", "caller_confirmed", 1)
modparam("pua_usrloc", "default_domain", "my.ip.address")
modparam("pua_usrloc", "presence_server", "sip:s...@my.ip.address:5060")
# ------------------------- request routing logic -------------------
# main routing logic
route{
if (!is_method("NOTIFY"))
xlog("L_INFO", "New request - Request/failure/branch routes:
M=$rm
RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n");
# max_forwards==0, or excessively long requests
if (!mf_process_maxfwd_header("10")) {
sl_send_reply("483","Too Many Hops");
exit;
};
if (msg:len >= 2048 ) {
sl_send_reply("513", "Message too big");
exit;
};
# !! Nathelper
# Special handling for NATed clients; first, NAT test is
# executed: it looks for via!=received and RFC1918 addresses
# in Contact (may fail if line-folding is used); also,
# the received test should, if completed, should check all
# vias for rpesence of received
if (nat_uac_test("3")) {
# Allow RR-ed requests, as these may indicate that
# a NAT-enabled proxy takes care of it; unless it is
# a REGISTER
if (is_method("REGISTER") || !is_present_hf("Record-Route")) {
#xlog("L_INFO", "LOG:Someone trying to register from
private IP,
rewriting\n");
#xlog("L_INFO", "$rb\n");
# This will work only for user agents that support
symmetric
# communication. We tested quite many of them and
majority is
# smart enough to be symmetric. In some phones it takes
a
# configuration option. With Cisco 7960, it is called
# NAT_Enable=Yes, with kphone it is called "symmetric
media" and
# "symmetric signalling".
# Rewrite contact with source IP of signalling
fix_nated_contact();
if ( is_method("INVITE") ) {
#xlog("L_INFO", "NAT: FIXING SDP");
fix_nated_sdp("1"); # Add direction=active to
SDP
};
force_rport(); # Add rport parameter to topmost Via
setbflag(6); # Mark as NATed
# if you want sip nat pinging
# setbflag(8);
};
};
# 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);
exit;
};
# 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 (!is_method("REGISTER"))
record_route();
if (method == "INVITE") {
setflag(4);
}
if (!uri==myself) {
# mark routing logic in request
append_hf("P-hint: outbound\r\n");
route(1);
exit;
};
# 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 (is_method("OPTIONS") && (! uri=~"sip:....@]+.*")) {
options_reply();
exit;
}
if (is_method("INVITE|ACK")) {
unforce_rtp_proxy();
}
if (is_method("REGISTER")) {
#xlog("L_INFO", "trying to register $au $ad\n");
if(cache_fetch("local","passwd_$tu",$avp(i:55))) {
$avp(i:54) = $tU;
xlog("SCRIPT: stored password is $avp(i:55)\n");
# perform auth from variables
# $avp(i:54) contains the username
# $avp(i:55) contains the password
if (!pv_www_authorize("asterisk")) {
# authentication failed -> do challenge
www_challenge("asterisk", "0");
exit;
};
} else {
# perform DB authentication ->
# password will be loaded from DB automatically
if (!www_authorize("asterisk", "subscriber")) {
# authentication failed -> do challenge
www_challenge("asterisk", "0");
exit;
};
# after DB authentication, the password is
available
# in $avp(i:55) because of the
"load_credentials"
# module parameter.
xlog("SCRIPT: storing password <$avp(i:55)>\n");
# use a 20 minutes lifetime for the password;
# after that, it will erased from cache and we
do
# db authentication again (refresh the passwd
from DB)
cache_store("local","passwd_$tu","$avp(i:55)",3600);
}
# Uncomment this if you want to use digest
authentication
#if (!www_authorize("asterisk", "subscriber")) {
# www_challenge("asterisk", "0");
# return;
#};
#bla_set_flag();
save("location");
pua_set_publish();
exit;
};
lookup("aliases");
if (!uri==myself) {
append_hf("P-hint: outbound alias\r\n");
route(1);
exit;
};
#xlog("L_INFO", "TESTING FOR $hdr(Event)\n");
if (is_method("NOTIFY") && $hdr(Event) == "message-summary") {
#xlog("L_INFO", "MWI Notification $rb\n");
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
exit;
}
} else if (is_method("SUBSCRIBE") && uri =~
"sip:[7-9][0...@my.ip.address") {
xlog("L_INFO", "SUBSCRIBE FOR PAGE\n");
if(!cache_fetch("local","ast_$fU",$avp(i:200)))
avp_db_load("$fu/username","$avp(i:200)");
if ($avp(i:200) == NULL || $avp(i:200) == '') {
xlog("INVALID DIALPLAN SERVER URL\n");
sl_send_reply("404", "Not Found");
exit;
} else {
cache_store("local","ast_$fU","$avp(i:200)",3600);
}
#rewritehostport("$avp(i:200)");
$rd = $(avp(i:200){s.select,0,:});
$rp = $(avp(i:200){s.select,1,:});
}
else if (is_method("PUBLISH|SUBSCRIBE|NOTIFY")) {
route(2);
} else if (dst_port==5061) {
if (!lookup("location")) {
sl_send_reply("404", "Not Found");
exit;
}
xlog("L_INFO", "request from asterisk $ru $tu\n");
if (to_uri =~ ".*intercom=true") {
xlog("INTERCOM REQUEST\n");
$var(checkuser) = $tU;
get_profile_size("caller","$var(checkuser)","$var(rcalls)");
if ($var(rcalls) > 0) {
xlog("DENY INTERCOM\n");
sl_send_reply("486", "Busy Here");
exit;
}
}
if (!isflagset(31)) {
#get_profile_size("caller","$avp(s:caller_uuid)","$var(calls)");
create_dialog();
#set_dlg_profile("caller","$avp(s:caller_uuid)");
set_dlg_profile("caller","$tU");
setflag(31);
get_profile_size("caller","$tU","$var(calls)");
xlog("NUM CALLS $tU $ru $mf $var(calls) \n");
}
} else if (is_method("INVITE")) {
#if (!proxy_authorize("asterisk", "subscriber")) {
# proxy_challenge("asterisk", "1"); # Realm will
be autogenerated
# exit;
#};
if(!cache_fetch("local","ast_$fU",$avp(i:200)))
avp_db_load("$fu/username","$avp(i:200)");
if ($avp(i:200) == NULL || $avp(i:200) == '') {
xlog("INVALID DIALPLAN SERVER URL\n");
sl_send_reply("404", "Not Found");
exit;
} else {
cache_store("local","ast_$fU","$avp(i:200)",3600);
}
#rewritehostport("$avp(i:200)");
$rd = $(avp(i:200){s.select,0,:});
$rp = $(avp(i:200){s.select,1,:});
if (!isflagset(31)) {
#get_profile_size("caller","$avp(s:caller_uuid)","$var(calls)");
create_dialog();
#set_dlg_profile("caller","$avp(s:caller_uuid)");
set_dlg_profile("caller","$fU");
setflag(31);
get_profile_size("caller","$fU","$var(calls)");
dialoginfo_set();
xlog("NUM CALLS $fU $ru $mf $var(calls) \n");
}
}
};
append_hf("P-hint: usrloc applied\r\n");
route(1);
}
route[1]
{
# !! Nathelper
if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" &&
!search("^Route:")){
sl_send_reply("479", "We don't forward to private IP
addresses");
exit;
};
# if client or server know to be behind a NAT, enable relay
if (isbflagset(6)) {
force_rtp_proxy();
};
# NAT processing of replies; apply to all transactions (for example,
# re-INVITEs from public to private UA are hard to identify as
# NATed at the moment of request processing); look at replies
t_on_reply("1");
# send it out now; use stateful forwarding as it works reliably
# even for UDP2TCP
if (!t_relay()) {
sl_reply_error();
};
}
# !! Nathelper
onreply_route[1] {
# NATed transaction ?
if (isbflagset(6) && status =~ "(183)|2[0-9][0-9]") {
fix_nated_contact();
force_rtp_proxy();
# otherwise, is it a transaction behind a NAT and we did not
# know at time of request processing ? (RFC1918 contacts)
} else if (nat_uac_test("1")) {
fix_nated_contact();
} else if (status =~ "407") {
xlog("AUTH unset profile $tu $fu\n");
unset_dlg_profile("caller","$fU");
}
}
route[2] {
xlog("L_INFO", "NOTIFY SUBSCRIBE PUBLISH route\n");
if (!t_newtran()) {
sl_reply_error();
exit;
};
if(is_method("PUBLISH")) {
if ($hdr(Sender) != NULL)
handle_publish("$hdr(Sender)");
else
handle_publish();
}
else if( is_method("SUBSCRIBE")) {
xlog("L_INFO", "Handle Subscribe\n");
handle_subscribe();
}
else if (is_method("NOTIFY")) {
# bla_handle_notify();
t_reply("200", "OK");
}
exit;
}
>
> --
> Anca Vamanu
> www.voice-system.ro
>
>
>> Thanks.
>>
>> -- James
>>
>>
>>> Regards,
>>>
>>> --
>>> Anca Vamanu
>>> www.voice-system.ro
>>>
>>
>>
>>>> Andrew Pogrebennyk wrote:
>>>> James,
>>>> Are you using pua_dialoginfo to get device state? If so are you telling
>>>> the dialog module to monitor the interesting dialogs and calling
>>>> dialoginfo_set()?
>>>> Note that once you get this working you will likely need this fix:
>>>> http://sourceforge.net/tracker/?func=detail&atid=1086412&aid=2847397&group_id=232389
>>>>
>>> On 31.03.2010 08:31, James Lamanna wrote:
>>>
>>>
>>>>> Sorry, I realized I had a configuration error on my phone, but the
>>>>> presence still does not work.
>>>>> The phone now subscribes to the event: dialog.
>>>>> Here are relevant parts of my opensips config:
>>>>>
>>>>> modparam("presence", "server_address", "sip:s...@xxx.xxx.xxx.xxx:5060")
>>>>> modparam("presence", "expires_offset", 10)
>>>>> modparam("presence_xml", "force_active", 1)
>>>>> modparam("presence_dialoginfo", "force_single_dialog", 1)
>>>>>
>>>>> I have also verified that handle_subscribe() is being called when a
>>>>> SUBSCRIBE message comes in.
>>>>> Calling the phone doesn't seem to produce any PUBLISH messages or
>>>>> anything pertaining to presence.
>>>>>
>>
>> _______________________________________________
>> Users mailing list
>> Users@lists.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>>
>
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