gt;
...
Seems that there's a problem with "if($var(calls)>$avp(maxcalls))"
because in logs i got:
User has 2 calls (max 10 calls): BUSY
...
User has 1 calls (max 10 calls): ALLOW
Any hints or suggestion?
Thanks for your help.
--
Michele Pinassi
Università degli
grep):
INVITE 333@172.20.1.4:506 5...@voip..it:5060 28
172.20.1.4:5060193.xxx:5060 REJECTED 28ad94e804782466
0:10
INVITE 5...@voip..it5...@voip.xxx.it20
193.205.4.182:5060 193.xxx:5060
?
>
> Best regards,
>
> Liviu Chircu
> OpenSIPS Developer
> http://www.opensips-solutions.com
> On 16.10.2018 10:13, Michele Pinassi wrote:
>> Hi all,
>>
>> maybe a trivial question but i'm planning to upgrade my production
>> OpenSIPS server from lates
ay ?
Thanks for any hint :-)
Michele
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Ufficio Esercizio e tecnologie - Università degli Studi di Siena
tel: 0577.(23)5000 - helpd...@unisi.it
Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di
Ateneo, htt
R");
break;
case -2:
xlog("L_ERR","Auth error for $fU@$fd from $si:
INVALID PASSWORD");
break;
case -3: # Stale nonce - This is not an error, so don't
print anything
br
; Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 - central...@unisi.it
Per trovare una soluzione r
setflag(CALL_AST);
>create_dialog("B");
>do_accounting("log", "cdr");
> }
>
> in cosole debug I see
> ACC: call ended:
> created=1496654777;call_start_time=1496654780;duration=7;ms_duration=7393;setuptime=3;meth
t;
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
very well !
>
> But, I have to move back my OpenSIPS to the 'poor' hardware!
>
>
> So, what configurations in OpenSIPS files should I set, to get better
> performance?
>
>
> Any suggestion will be very helpful!
>
>
> Best regards.
>
>
--
Mich
le
On 16/03/2017 09:05, Michele Pinassi wrote:
> Opensips 1.11.9-notls on a Debian 8.7 x64, with more that 700 phones
> registered.
>
> Sometimes, almost twice a week, i got a lot of:
>
> /usr/sbin/opensips[1585]: ERROR:presence:update_presentity: No E_Tag
> match [a.1489409482.
memory available...
Thanks, Michele
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 - central...@unisi.it
Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le FAQ di
Ateneo, http
ute_to_carrier("toip")) {
t_on_failure("next_gw");
t_relay();
exit;
}
}
but i got an error. Maybe i'm doing somewthing wrong with number
manipulation.
Suggestions how i can do the same thing in a more elegant manner are
welcome !
Thanks for any
wrote:
>
> What is the preferred OS and version for OpenSIPS?
>
>
>
> I installed it on newer version Ubtunu, but and it seems to work, but
> the Control Panel does not. The Control Panel I am guessing expects
> an older version of Debian.
>
>
>
--
Michele Pi
On my opensips log i got:
/usr/sbin/opensips[17639]: CRITICAL:core:timer_ticker: timer handler
lasted (260 us) for more than timer tick (100
us) -> potential timer shifting
what's going on ?
Thanks, Michele
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, S
Hi all,
i'm going crazy with an issue with WiFi VoIP users. Our VoIP server,
Opensips 1.11.9, was inside DMZ with public IP address 193.x.x.110 and
WiFi network is 10.x.x.x. VoIP fixed phone are inside a separated
network 172.20.x.x and firewall allowed traffic in both directions
between those net
08 proxy-voip01 /usr/sbin/opensips[2978]: 403 - Forbidden -
Call blocked from 333xx to 5002
Any suggestions ?
Thanks, Michele
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 - central...@
t are you sure that $var(carrier) points to
> "toip" carrier ?
>
> Note that the module changes only the username in RURI, it does not
> change TO hdr .
>
> Best regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutio
all the routing scenarios (per prefix, per
> carrier, etc). Are you sure your call is routed via that GW ? try to
> print in cfg the GW ID to see it the right GW is used.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-soluti
rrier routing, i get
no number strip...
Maybe i'm missing something ?
Thanks, Michele
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 - central...@unisi.it
Per trovare una soluzione rapida a
0 phones.
Any hint/suggestions ?
Thanks, Michele
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 - central...@unisi.it
Per trovare una soluzione rapida ai tuoi problemi tecnici consulta le F
exit;//
//}/
but on establishing call, this is the tcpdump trace between VOIP01 and
VOIP02 i get this:
/IP VOIP01.5060 > VOIP02.5060: UDP, length 1170//
//
//INVITE sip:86472@VOIP02 SIP/2.0//
//Record-Route: //
//Via: SIP/2.0/UDP VOIP01:5060;branch=z9hG4bKe567.edebdee3.0//
//Via: SIP/2.0/UDP
172
er cannot call directly to Boss phone: firstly, the call will be
diverted to secretary and only in a second time, call should be
forwarded to boss phone (from secretary).
Any hint how to do this ?
Michele
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza In
ip.unisi.it") { #
> CUSTOMIZE ME
>
> Or set the contact in presence with the real IP:
> modparam("presence", "server_address",
> mailto:sip:presence@127.0.0.1:5060)
>
> Best regards,
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Develo
5030:
http://pastebin.com/h7qBTKbv
This is the full opensips.cfg -> http://pastebin.com/r7fNTxpy
Thanks for any help.
Michele
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 - fax: 0577.(23)2053
Pe
't figure out why
doesn't work...
Michele
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)5000 - fax: 0577.(23)2053
Per trovare una soluzione rapida ai tuoi problemi tecnici
consulta l
ndle_publish();
} else if (is_method("SUBSCRIBE")) {
handle_subscribe();
t_relay();
}
} else if(is_method("NOTIFY")) {
t_relay();
}
exit;
}
Is that correct ? There's another better way to do this ?
Michele
--
Michele
uot;)) {
t_relay();
exit;
}
sl_send_reply("404","Not here");
}
exit;
}
Hope this works.
Michele
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servi
binGAdVESTnG0.bin
Description: PGP/MIME version identification
encrypted.asc
Description: OpenPGP encrypted message
___
Users mailing list
Users@lists.opensips.org
http://lists.opensips.org/cgi-bin/mailman/listinfo/users
ror();
exit;
}
if (is_method("PUBLISH")) {
if($hdr(Sender)!= NULL)
handle_publish("$hdr(Sender)");
else
handle_publish();
} else if (is_method("SUBSCRIBE")) {
handle_subscribe();
}
exit;
}
username","$fU");
}
# fix some broken subscriptions
if(!search("^Accept: application/simple-message-summary")) {
append_hf("Accept: application/simple-message-summary\r\n");
}
setdsturi("si
modparam("presence", "server_address",
> "sip:presence@127.0.0.1:5060")
>
> Best regards,
> Bogdan-Andrei Iancu
> OpenSIPS Founder and Developer
> http://www.opensips-solutions.com
> On 24.02.2015 12:04, Michele Pinassi wrote:
>>
erver when i try to call (from another
phone) 5020.
The full opensips.cfg is available here: http://pastebin.com/e6SfbFfq
Thanks for any help.
Michele
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23
http://pastebin.com/rW3AKr22
My config: http://pastebin.com/9gP9xncd
Thanks for all your help.
Michele
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi di Siena
tel: 0577.(23)2169 - fax: 0577.(23)2053
Per trovar
ts.opensips.org
>> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
>
>
>
> ___
> Users mailing list
> Users@lists.opensips.org
> http://lists.opensips.org/cgi-bin/mailman/listinfo/users
--
Michele Pinassi
Responsabile Tel
Y+%60active_watchers%60.%60local_contact%60+ASC&token=02f2e9cb7d26d05e4d122aa4f77b558d>
sip:5...@voip.unisi.it 5002voip.unisi.it 5007voip.unisi.it
dialog f315b2d58ae8829149b784764c5a40e3-ef1f 1f2g3lai9y
5c5b6c540726-yq78t6gum1az 1 2 sip:5002@172.20.
presence);
}
[]
route(relay);
}
# RELAY
route[relay] {
# for INVITEs enable some additional helper routes
if (is_method("INVITE")) {
xlog("L_INFO","Route RELAY on INVITE [$fd/$fu/$rd/$ru/$si/]\n");
dialoginfo_set();
t_on
ndle_publish();
}
if (is_method("SUBSCRIBE")) {
handle_subscribe();
}
exit;
}
branch_route[per_branch_ops] {
xlog("L_INFO","Route NEW BRANCH [$fd/$fu/$rd/$ru/$si/]\n");
}
failure_route[missed_call] {
xlog("L_INFO","Route MI
update_subscription: notify
Oct 29 11:47:32 proxy-voip01 /usr/sbin/opensips[5496]:
INFO:presence:send_notify_request: NOTIFY sip:5...@voip.unisi.it via
sip:5022@172.20.2.12:32768 on behalf of sip:*9...@voip.unisi.it for event
message-summary, to_tag=f315b2d58ae8829149b784764c5a40e3-c2fb, cseq=1
M
MEDICINA [$fd/$fu/$rd/$ru/$si/]\n");
if(route_to_carrier("trunkmed")) {
t_on_failure("next_gw");
t_relay();
exit;
}
}
"trunkmed" gw is outside VoIP network (200.200.200.2) and i'm not able
to forward correctly packets from VoIP devices to trunkm
Hi Frana,
Just install an Asterisk and live happy.
If you need help i've already di what you want
Michele
Il 20/apr/2014 09:57 "H Yavari" ha scritto:
> Hi Frank,
> I tried to do with rtpproxy_stream2uac() the RBT but this not work for me.
>
> --
> Regards,
> H.Yavari
> --
sbin/opensips[10210]:
INFO:core:sig_usr: signal 15 received
Oct 29 13:59:04 proxy-voip01 /usr/sbin/opensips[10193]:
INFO:core:cleanup: cleanup
Oct 29 13:59:04 proxy-voip01 /usr/sbin/opensips[10193]:
NOTICE:presence:destroy: destroy module ...
Any hint ?
Michele
--
Michele Pinassi
Responsabile T
quot;,"Internal Error");
};
exit;
}
# Presence route
route[handle_presence]
{
if (!t_newtran())
{
sl_reply_error();
exit;
}
if(is_method("PUBLISH"))
{
handle_publish();
}
else
if( is_method("SUBSCRIBE"))
{
the
> configuration seems ok for that ; Could you confirm that OpenSIPS is
> sending RADIUS packages to the RADIUS server ? The RARDIUS server is the
> one responsible for writing in whatever file or DB the received data.
>
> Regards,
>
> Bogdan-Andrei Iancu
> OpenSIPS Foun
> http://www.opensips-solutions.com
>
>
> On 05/16/2013 01:18 PM, Michele Pinassi wrote:
>> Thanks Bodgan for your kindly reply but now accounting don't work:
>> nothing will be added to acc table !
>>
>> Here's the full routing logic. Maybe there's
1 [$fd/$fu/$rd/$ru/$si/]\n");
}
onreply_route[2] {
xlog("L_INFO","OnReply Route2 [$fd/$fu/$rd/$ru/$si/]\n");
}
failure_route[1] {
xlog("L_INFO","Failure Route1 [$fd/$fu/$rd/$ru/$si/]\n");
if (t_was_cancelled()) {
exit;
/$rd/$ru/$si/]\n");
if (!t_newtran()) {
sl_reply_error();
exit;
};
if(is_method("PUBLISH")) {
handle_publish();
} else if( is_method("SUBSCRIBE")) {
handle_subscribe();
}
exit;
}
route[4] {
xlog("L_INFO"
quot;);
}
onreply_route[2] {
xlog("L_INFO","OnReply Route2 [$fd/$fu/$rd/$ru/$si/]\n");
}
failure_route[1] {
xlog("L_INFO","Failure Route1 [$fd/$fu/$rd/$ru/$si/]\n");
if (t_was_cancelled()) {
exit;
}
if (t_check_stat
$ | NULL |
+----+-++--+--+---+--+--+
root@proxy-voip01:/etc/opensips#
Any hint ? Thanks, Michele
--
Michele Pinassi
Responsabile Telefonia di Ateneo
Servizio Reti, Sistemi e Sicurezza Informatica - Università degli Studi
di Siena
tel: 0577.(23)
48 matches
Mail list logo
