I do not have the possibility to use Patton 4960 with 1xPRI. I wished an alternative method for Patton1 + patton2 + patton3: 8+8+8=32. thanks
This is the code for 2 Patton
Hi!
On the pattons you can created Error Messages when not circuit channel is available And on Ser you have to register to Failure Route to the route of the first patton and the failure route point to the second Patton.
PATTON1:
profile sip default
map causes to-sip no-circuit-channel-available to 486
map causes to-sip resources-unavailable to 486
service hunt-group HUNT-BRI
cyclic
timeout 1
drop-causes normal-unspecified
drop-causes no-circuit-channel-available
drop-causes network-out-of-order
drop-causes temporary-failure
drop-causes switching-equipment-congestion
drop-causes access-info-discarded
drop-causes circuit-channel-not-available
drop-causes resources-unavailable
route call 1 dest-interface ISDN_01
route call 2 dest-interface ISDN_02
route call 3 dest-interface ISDN_03
SER:
Ser.cfg:
if (uri=~ ". *sip: 012345 ") # Patton 1
{t_on_failure (1);
t_relay_to_udp (11.11.11.10, 5060);
break;
}
failure_route [1]
{
if (method== " INVITE ")
{
if (t_check_status (600|486)) #### ErrorCode from Patton 1
{
#####call from Patton 1 -- 486 (Busy) ret -- call switch to Patton 2 ######
if (uri=~ ". *sip: 012345 ") # Costumer xy
{
append_branch ();
t_relay_to_udp (11.11.11.11, 5060); ##Unterroute ist hier nicht notwendig,
break; ##da sich die Verrechnung hier nicht aendert
}
}
}
Which code to use for 3 patton?
------------------------------------------------------
Leggi GRATIS le tue mail con il telefonino i-mode di Wind
http://i-mode.wind.it/
Hi, ppl!
I have installed OpenSER and radius server. OpenSER authorized all
call through radius. How I can limit call duration by
"Session-Timeout" attribute from radius ?
--
tarantul
Hi Team,
Please someone guide me in how I could forward for instance a call to
1111111 made on the SER to an * server.
I have tried quite a lot of things but nothing seems to work.
Thanks
--
Kevin Kibuku,
Intersat Africa Limited
Internet Via Satellite
Cell: +254720789158
Hi All,
I am trying to implement serial forking and am using the
example on voice-systems website as a guide. The problem I am
experiencing is as follows.
Failure route rewrite ruri with the next uri to try and calls route[17]
route 17 performs a lookup("location"), finds the usrloc information
and rewrites ruri, however, when t_relay is called it uses the
original ruri (one before the lookup) to do t_relay.
Another issue I'm experiencing is that when processing call after
failure route, ruri_user_avp. Once i rewrite the uri in failure
route, I have to use the following to manually write a value for
ruri_user_avp. Can someone explain why this is happening?
Thanks in advance for your help!!
--
Zahid
]
Some relevant config and debug info:
modparam("lcr", "gw_uri_avp", "1400")
modparam("lcr", "ruri_user_avp", "1402")
route[18] { # used to process pstn/ off system calls
......
if (!is_avp_set("$avp(i:1402)")) {
xlog("L_INFO","$ci: route[22] ** setting ruri_user_avp **\n");
$avp(i:1402) = $rU;
xlog("L_INFO","$ci: route[22] ** setting ruri_user_avp = $avp
(i:1402) **\n");
}
........
} #end route 18
route[11] {
# load all numbers asscociated with the pilot number in $avp
(serial_fork) array
xlog("L_INFO","$ci: route[11] - M=$rm RURI=$ru F=$fu T=$tu IP=$si
\n");
$avp(serial_fork) = "sip:*6" + $rU + "@sipdev.oursiphost.com";
if (avp_db_query("select username from fork where pilot = $rU
order by pri" , "$avp(serial_fork)")) {
xlog("L_INFO", "route[11] $ci: method $rm r-uri <$ru>
\n");
xlog("L_INFO", "route[11] serial_fork = $avp
(serial_fork) remote-user = $rU \n");
t_on_failure("5");
route(17);
exit;
} else {
xlog("L_ERR", "route[11] failed to load serial_fork
info.\n");
}
avp_print(); # debugging only
setflag(8);
} # end of route[11]
route[17] {
xlog("L_INFO","$ci: route[17] - M=$rm RURI=$ru F=$fu T=$tu IP=$si
\n");
$avp(s:inv_timeout) = "10";
if (lookup("location")) { # currently online
registered UA.
xlog("L_INFO","$ci: route[17] - M=$rm RURI=$ru F=$fu T=$tu
IP=$si -- inside location lookup \n");
if (!t_relay()) {
sl_reply_error();
};
return;
} else { # not online
if (does_uri_exist()) { # but exists
xlog("L_INFO", "$ci: Subscriber is offline [$ru]\n");
sl_send_reply("480", "Temporarily Unavailable.");
return;
}
xlog("L_INFO","$ci: route[17] - M=$rm RURI=$ru F=$fu T=$tu
IP=$si -- outside location lookup \n");
};
if (uri=~"^sip:(([0-9()+-])|(%20))+@") {
subst_user('/[()+-]//g'); # strip out the punctuation.
subst_user('/%20//g'); # strip out the white space.
route(18); # all numerics just go to route(18) --
off system calls
return;
};
} # end route[17]
failure_route[5] {
xlog("L_INFO","$ci: failure_route[5] - M=$rm RURI=$ru F=$fu
T=$tu IP=$si\n");
avp_print();
if (avp_pushto("$ruri", "$avp(serial_fork)")) {
append_branch();
xlog("L_INFO","$ci: failure_route[5] $rm r-uri <$ru>
from <$fu> to <$tu>\n");
avp_delete("$avp(serial_fork)");
t_on_failure("5");
route(17);
} else {
xlog("L_INFO","$ci: failure_route[5] $rm r-uri <$ru>
from <$fu> to <$tu>\n");
}
}
Debug info
Aug 31 10:54:56 mousse openser[6815]: a410a3c1-
e4befc20-961fd447(a)10.1.12.174: failure_route[5] - M=INVITE RURI=sip:
10512@10.1.12.172:8891 F=sip:10510@sipdev.oursiphost.com T=sip:
10512(a)sipdev.oursiphost.com;user=phone IP=10.1.12.174
Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp:
p=0xb6039c28, flags=0x0003
Aug 31 10:54:56 mousse openser[6815]: INFO:
name=<inv_timeout>
Aug 31 10:54:56 mousse openser[6815]: INFO:
val_str=<10 / 2>
Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp:
p=0xb6039bf0, flags=0x0002
Aug 31 10:54:56 mousse openser[6815]: INFO: id=<655>
Aug 31 10:54:56 mousse openser[6815]: INFO:
val_str=<sip:10511@sipdev.oursiphost.com / 29>
Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp:
p=0xb6039bb8, flags=0x0002
Aug 31 10:54:56 mousse openser[6815]: INFO: id=<655>
Aug 31 10:54:56 mousse openser[6815]: INFO:
val_str=<sip:10210@sipdev.oursiphost.com / 29>
Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp:
p=0xb6039b80, flags=0x0002
Aug 31 10:54:56 mousse openser[6815]: INFO: id=<655>
Aug 31 10:54:56 mousse openser[6815]: INFO:
val_str=<sip:10510@sipdev.oursiphost.com / 29>
Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp:
p=0xb6039b48, flags=0x0002
Aug 31 10:54:56 mousse openser[6815]: INFO: id=<655>
Aug 31 10:54:56 mousse openser[6815]: INFO:
val_str=<sip:*610512@sipdev.oursiphost.com / 31>
Aug 31 10:54:56 mousse openser[6815]: INFO:avpops:print_avp:
p=0xb6039b18, flags=0x0083
Aug 31 10:54:56 mousse openser[6815]: INFO:
name=<is_pilot>
Aug 31 10:54:56 mousse openser[6815]: INFO:
val_str=<y / 1>
Aug 31 10:54:56 mousse openser[6815]: DEBUG:avpops:pushto_avps: 1
avps were processed
Aug 31 10:54:56 mousse openser[6815]: ******* setting for branch 0
flags 0
Aug 31 10:54:56 mousse openser[6815]: a410a3c1-
e4befc20-961fd447(a)10.1.12.174: failure_route[5] INVITE r-uri <sip:
10511(a)sipdev.oursiphost.com> from <sip:10510@sipdev.oursiphost.com>
to <sip:10512@sipdev.oursiphost.com;user=phone>
Aug 31 10:54:56 mousse openser[6815]: DEBUG:avpops:delete_avps: 1
avps were removed
Aug 31 10:54:56 mousse openser[6815]: a410a3c1-
e4befc20-961fd447(a)10.1.12.174: route[17] - M=INVITE RURI=sip:
10511(a)sipdev.oursiphost.com F=sip:10510@sipdev.oursiphost.com T=sip:
10512(a)sipdev.oursiphost.com;user=phone IP=10.1.12.174
Aug 31 10:54:56 mousse openser[6815]: rewrite_uri: Rewriting Request-
URI with 'sip:10511@10.1.12.172:8891'
Aug 31 10:54:56 mousse openser[6815]: a410a3c1-
e4befc20-961fd447(a)10.1.12.174: route[17] - M=INVITE RURI=sip:
10511@10.1.12.172:8891 F=sip:10510@sipdev.oursiphost.com T=sip:
10512(a)sipdev.oursiphost.com;user=phone IP=10.1.12.174 -- inside
location lookup
Aug 31 10:54:56 mousse openser[6815]: DEBUG: mk_proxy: doing DNS
lookup...
Aug 31 10:54:56 mousse openser[6815]: DEBUG:sip_resolvehost2: no
port, no proto -> do NAPTR lookup!
Aug 31 10:54:56 mousse openser[6815]: DEBUG:filter_and_sort_naptr:
found valid SIP+D2U -> _sip._udp.sipdev.oursiphost.com
Aug 31 10:54:56 mousse openser[6815]: DEBUG:do_srv_lookup: SRV
(_sip._udp.sipdev.oursiphost.com) = mousse.cc.oursiphost.com:5060
Aug 31 10:54:56 mousse openser[6815]: DEBUG:sip_resolvehost2: found!
Aug 31 10:54:56 mousse openser[6815]: check_via_address(10.1.12.174,
10.1.12.174, 0)
Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:set_timer: relative
timeout is 500000
Aug 31 10:54:56 mousse openser[6815]: DEBUG: add_to_tail_of_timer[4]:
0xb6039eb4 (61500000)
Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:set_timer: relative
timeout is 30
Aug 31 10:54:56 mousse openser[6815]: DEBUG: add_to_tail_of_timer[0]:
0xb6039ed0 (91)
Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:relay_reply: branch=0,
save=1, relay=-1
>, CSEQ_N=<CSeq: 1> 2(a)sipdev.oursiphost.com;user=phone>ocal: using
FROM=<From: "Zahid 10510" <sip:
10510(a)sipdev.oursiphost.com>;tag=BF787F6-F6DD44ED
Aug 31 10:54:56 mousse openser[6815]: DEBUG: cancel_branch: sending
cancel...
Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:set_timer: relative
timeout is 500000
Aug 31 10:54:56 mousse openser[6815]: DEBUG: add_to_tail_of_timer[4]:
0xb6039e20 (61500000)
Aug 31 10:54:56 mousse openser[6815]: DEBUG:tm:set_timer: relative
timeout is 30
Aug 31 10:54:56 mousse openser[6815]: DEBUG: add_to_tail_of_timer[0]:
0xb6039e3c (91)
Je serai absent(e) du 03/09/2007 au 17/09/2007.
Je répondrai à votre message dès mon retour.
Contacter le service Ingénierie :
Patrice Galleau au +33 1 44 55 69 36, patrice.galleau(a)orange-ftgroup.com ou
Tony Capochichi au +33 1 44 55 66 34, tony capochichi(a)orange-ftgroup.com
I will reply to your message when I'll be back.
Please contact the Engineer department:
Patrice Galleau at +33 1 44 55 69 36, patrice.galleau(a)orange-ftgroup.com or
Tony Capochichi at +33 1 44 55 66 34, tony capochichi(a)orange-ftgroup.com
*********************************
This message and any attachments (the "message") are confidential and intended solely for the addressees.
Any unauthorised use or dissemination is prohibited.
Messages are susceptible to alteration.
France Telecom Group shall not be liable for the message if altered, changed or falsified.
If you are not the intended addressee of this message, please cancel it immediately and inform the sender.
********************************
Does Presence Module support 'policy document'? That means the user can do some settings to forbid specified watcher or something like this.
Thank you!
Kevin
_________________________________________________________________
Windows Live Custom Domain，您的免费电子邮局。
https://domains.live.com/default.aspx
Hello,
I would like to announce the launch of OpenXCAP.org
OpenXCAP is an open source, easy extensible, fully featured XCAP
server with
TLS security and support for multiple realms. OpenXCAP server can be
used in
combination with a SIP server that supports PUBLISH/SUBSCRIBE/NOTIFY
methods to provide a complete SIP SIMPLE server solution
The server is written in the Python programming language and is based
on the
following RFCs:
- The Extensible Markup Language Configuration Access Protocol RFC 4825
- Formats for Representing Resource Lists RFC 4826
- Usage for Manipulating Presence Document Contents RFC 4827
- Presence Authorization Rules draft-ietf-simple-presence-rules-10
The server supports TLS security, multiple back-end storage systems and
works out of the box with OpenSER Proxy/Registrar/Presence server. The
software is provided under the BSD license.
I wish to thank here to my IETF friends who helped me with advice,
testing
and guidance during the development phase of this project. I would
also like
to thank to Mircea Amarascu, the author of the software, without his
passion
and dedication this project would not have been possible.
The project is hosted at:
http://openxcap.org
The project is supported via the wiki collaboration system setup by
AG Projects.
To open ticket please Register first.
We are looking forward your feedback from real-life field deployments.
Kind regards,
Adrian Georgescu
AG Projects
Hello all,
Actually i have openser box which i used to make
calls among my offices which spread different
geographical places.
i am using in the configuration file just to make
changes to my rtp packets so without RTPPROXY
and MEDIAPROXY modules i was able to make calls
even my end UAC are behind NAT.
but if i want to control my media between different
UAs like a prepaid system as we used to see in case of
mobile prepaid system which stops you from continuing your
call when there was no balance left on ur mobile .
can anybody give me a hint how can i do that ?
if so what architecture it needs like any mediaproxy?
will session border controller help me out?
just give me a hint so that i can go ahead
thanks once again
--
Srinivas Antarvedi
Peoplefone AG jest firmą oferującą usługi telekomunikacyjne oparte na
technologii Voice over IP (VoIP) w wyjątkowo niskich stawkach. Peoplefone
jest certyfikowanym partnerem firm
Siemens<http://www.siemens.ch/index.jsp?sdc_p=c175fi1012637lmno1012637psuz1&sdc_sid…>i
AVM/FRITZ!Box <http://www.fritz-shop.ch/>. W związku z naszym dynamicznym
rozwojem, do nowo tworzonej spółki w Polsce poszukujemy osoby na stanowisko:
SPECJALISTA D/S VOIP
Miejsce pracy: Warszawa
*Wymagania:*
- wykształcenie wyższe techniczne (lub w trakcie studiów), najlepiej
informatyczne
- praktyczna umiejętność programowania w środowisku w PHP i językach C
i C++
- znajomość Linux
- znajomość mySQL i PostgreSQL
- znajomość zagadnień związanych z sieciami IP, UDP, TCP
- znajomość narzędzi do analizy sieci np.. Ethereal
- wiedza z zakresu VoIP, serwerów i konfiguracji oprzyrządowania
- umiejętność rozwiązywania problemów informatycznych
- znajomość języka angielskiego w mowie i piśmie w stopniu
zaawansowanym
- umiejętność szybkiego uczenia się
- chęć podnoszenia swoich kwalifikacji zawodowych
- profesjonalne i rzetelne podejście do obowiązków służbowych,
umiejętność pracy w zespole, samodzielność, komunikatywność,
systematyczność, terminowość
- doświadczenie zawodowe mile widziane
*Dodatkowe wymagania (mile widziane):*
- umiejętność programowania w PERL i JAVA
- znajomość Asterisk / SER / OPENSER
- umiejętność konfigurowania bramek i ruterów Cisco, Patton, lub
innych
- znajomość protokołów SIP lub STUN
- znajomość problematyki NAT I funkcjonowania outband Proxy
- znajomość narzędzi monitorujących jak Cactus, Nagios, MRTG
- znajomość narzędzi jak DRBD, Hearthbeat
- dyspozycyjność
Wszystkich zainteresowanych prosimy o przesłanie życiorysu oraz listu
motywacyjnego (zawierającego zgodę na przetwarzanie danych osobowych)
*do 10 września
br. **
*na adres:
*ewa.ciesla(a)peoplefone.com*
W temacie wiadomości wysyłanej pocztą elektroniczną prosimy umieścić nazwę
stanowiska.
Skontaktujemy się tylko z wybranymi kandydatami.
Prosimy o dopisanie następującej klauzuli "Wyrażam zgodę na przetwarzanie
moich danych osobowych zawartych w mojej ofercie pracy dla potrzeb
niezbędnych do realizacji procesu rekrutacji (zgodnie z ustawą z dn.
29.08.97 roku o Ochronie Danych Osobowych Dz. Ust Nr 133 poz. 883)".
