Hi everyone
I'm trying to integrate Asterisk with Kamailio for voicemail. I tried to follow this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb BUT:
- I had to adapt it because I use LDAP authentication with Kamailio - I had problems with Asterisk 10.7 (problems with chan_sip module crashing) so I've installed Asterisk 11 on another VM - we have high-availability with 2 Kamailio servers, with Kamailio listening on TCP (constraint from our SSL gateway in front of Kamailio) on a "virtual IP" (created by keep-alive): this VIP is not visible with ifconfig, but you can see it with the command "ip addr sh eth0"
For now, we use Linphone on Windows as SIP clients to test. If I don't define WITH_ASTERISK, calls work, I can call someone@domain.tld However, if I define WITH_ASTERISK, calls fail (even with destination registered and available) and I have these errors in the logfile:
Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [ut.h:333]: no corresponding socket for af 2 Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket) Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: INFO: <script>: reply error Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL)
It seems to happen on the if (!t_relay()) line in ROUTE[RELAY]
192.168.14.25 is the real IP of the Kamailio server, 192.168.14.24 is the VIP of the Kamailio "cluster" 192.168.14.28 is the IP of the Mysql server 192.168.14.32 is the IP of the Asterisk server
I can't find why the relay doesn't work. I've tried to bypass the VIP and have Kamailio listen on the real IP, but it still doesn't work: I don't seem to have the same errors as above, but I don't see any traffic between Kamailio and Asterisk.
What could be the problem? Thanks for your help
Christophe
Below is my kamailio.cfg:
#!WITH_DEBUG
#!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.2 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: sr-users@lists.sip-router.org # # Refer to the Core CookBook at http://www.kamailio.org/dokuwiki/doku.php # for an explanation of possible statements, functions and parameters. # # Several features can be enabled using '#!define WITH_FEATURE' directives: # # *** To run in debug mode: # - define WITH_DEBUG # # *** To enable mysql: # - define WITH_MYSQL # # *** To enable authentication execute: # - enable mysql # - define WITH_AUTH # - add users using 'kamctl' # # *** To enable IP authentication execute: # - enable mysql # - enable authentication # - define WITH_IPAUTH # - add IP addresses with group id '1' to 'address' table # # *** To enable persistent user location execute: # - enable mysql # - define WITH_USRLOCDB # # *** To enable presence server execute: # - enable mysql # - define WITH_PRESENCE # # *** To enable nat traversal execute: # - define WITH_NAT # - install RTPProxy: http://www.rtpproxy.org # - start RTPProxy: # rtpproxy -l _your_public_ip_ -s udp:localhost:7722 # ###!define WITH_NAT
# *** To enable PSTN gateway routing execute: # - define WITH_PSTN # - set the value of pstn.gw_ip # - check route[PSTN] for regexp routing condition # # *** To enable database aliases lookup execute: # - enable mysql # - define WITH_ALIASDB # # *** To enable speed dial lookup execute: # - enable mysql # - define WITH_SPEEDDIAL # # *** To enable multi-domain support execute: # - enable mysql # - define WITH_MULTIDOMAIN # # *** To enable TLS support execute: # - adjust CFGDIR/tls.cfg as needed # - define WITH_TLS # # *** To enable XMLRPC support execute: # - define WITH_XMLRPC # - adjust route[XMLRPC] for access policy # # *** To enable anti-flood detection execute: # - adjust pike and htable=>ipban settings as needed (default is # block if more than 16 requests in 2 seconds and ban for 300 seconds) # - define WITH_ANTIFLOOD # # *** To block 3XX redirect replies execute: # - define WITH_BLOCK3XX # # *** To enable VoiceMail routing execute: # - define WITH_VOICEMAIL # - set the value of voicemail.srv_ip # - adjust the value of voicemail.srv_port # # *** To enhance accounting execute: # - enable mysql # - define WITH_ACCDB # - add following columns to database #!ifdef ACCDB_COMMENT ALTER TABLE acc ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE acc ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE acc ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_domain VARCHAR(128) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN src_ip varchar(64) NOT NULL default ''; ALTER TABLE missed_calls ADD COLUMN dst_ouser VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_user VARCHAR(64) NOT NULL DEFAULT ''; ALTER TABLE missed_calls ADD COLUMN dst_domain VARCHAR(128) NOT NULL DEFAULT ''; #!endif
###!define WITH_ASTERISK ###!define WITH_VOICEMAIL #!define WITH_LDAP #!define WITH_AUTH #!define WITH_MYSQL ####### Defined Values #########
# *** Value defines - IDs used later in config #!ifdef WITH_MYSQL # - database URL - used to connect to database server by modules such # as: auth_db, acc, usrloc, a.s.o. #!define DBURL "mysql://openserrw:openserrw@192.168.14.28/openser"
#!ifdef WITH_ASTERISK #!define DBASTURL "mysql://asterisk:asteriskpwd@192.168.14.28/asterisk" #!endif
#!endif #!ifdef WITH_MULTIDOMAIN # - the value for 'use_domain' parameters #!define MULTIDOMAIN 1 #!else #!define MULTIDOMAIN 0 #!endif
# - flags # FLT_ - per transaction (message) flags # FLB_ - per branch flags #!define FLT_ACC 1 #!define FLT_ACCMISSED 2 #!define FLT_ACCFAILED 3 #!define FLT_NATS 5
#!define FLB_NATB 6 #!define FLB_NATSIPPING 7
####### Global Parameters #########
#!ifdef WITH_DEBUG debug=4 log_stderror=no #!else debug=2 log_stderror=no #!endif
memdbg=5 memlog=5
log_facility=LOG_LOCAL6
fork=yes children=4
/* uncomment the next line to disable TCP (default on) */ #disable_tcp=yes
/* uncomment the next line to disable the auto discovery of local aliases based on reverse DNS on IPs (default on) */ auto_aliases=no
/* add local domain aliases */ alias="mydomain.corp"
/* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ listen=tcp:192.168.14.24:5060 #listen=tcp:192.168.14.25:5060
/* port to listen to * - can be specified more than once if needed to listen on many ports */ #port=5060
#!ifdef WITH_TLS enable_tls=yes #!endif
# life time of TCP connection when there is no traffic # - a bit higher than registration expires to cope with UA behind NAT tcp_connection_lifetime=3605
####### Custom Parameters #########
# These parameters can be modified runtime via RPC interface # - see the documentation of 'cfg_rpc' module. # # Format: group.id = value 'desc' description # Access: $sel(cfg_get.group.id) or @cfg_get.group.id #
#!ifdef WITH_PSTN # PSTN GW Routing # # - pstn.gw_ip: valid IP or hostname as string value, example: # pstn.gw_ip = "10.0.0.101" desc "My PSTN GW Address" # # - by default is empty to avoid misrouting pstn.gw_ip = "" desc "PSTN GW Address" #!endif
#!ifdef WITH_VOICEMAIL # VoiceMail Routing on offline, busy or no answer # # - by default Voicemail server IP is empty to avoid misrouting voicemail.srv_ip = "192.168.14.32" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail Port" #!endif
#!ifdef WITH_ASTERISK asterisk.bindip = "192.168.14.32" desc "Asterisk IP Address" asterisk.bindport = "5060" desc "Asterisk Port" kamailio.bindip = "192.168.14.24" desc "Kamailio IP Address" kamailio.bindport = "5060" desc "Kamailio Port" #!endif
####### Modules Section ########
# set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/lib/kamailio/modules_k/:/usr/lib/kamailio/modules/" #!endif
#!ifdef WITH_MYSQL loadmodule "db_mysql.so" #!endif
loadmodule "mi_fifo.so" loadmodule "kex.so" loadmodule "tm.so" loadmodule "tmx.so" loadmodule "sl.so" loadmodule "rr.so" loadmodule "pv.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "siputils.so" loadmodule "xlog.so" # loadmodule "sanity.so" loadmodule "ctl.so" loadmodule "cfg_rpc.so" loadmodule "mi_rpc.so" loadmodule "acc.so"
#!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #chris loadmodule "ldap.so" modparam ("ldap", "config_file", "/etc/kamailio/ldap.cfg") #!ifdef WITH_IPAUTH loadmodule "permissions.so" #!endif #!endif
#!ifdef WITH_ALIASDB loadmodule "alias_db.so" #!endif
#!ifdef WITH_SPEEDDIAL loadmodule "speeddial.so" #!endif
#!ifdef WITH_MULTIDOMAIN loadmodule "domain.so" #!endif
#!ifdef WITH_PRESENCE loadmodule "presence.so" loadmodule "presence_xml.so" #!endif
#!ifdef WITH_NAT loadmodule "nathelper.so" loadmodule "rtpproxy.so" #!endif
#!ifdef WITH_TLS loadmodule "tls.so" #!endif
#!ifdef WITH_ANTIFLOOD loadmodule "htable.so" loadmodule "pike.so" #!endif
#!ifdef WITH_XMLRPC loadmodule "xmlrpc.so" #!endif
#!ifdef WITH_DEBUG loadmodule "debugger.so" #!endif
#!ifdef WITH_ASTERISK loadmodule "uac.so" #!endif
# ----------------- setting module-specific parameters ---------------
# ----- mi_fifo params ----- #modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") modparam("mi_fifo", "fifo_name", "/tmp/kamailio_tmp")
# ----- tm params ----- # auto-discard branches from previous serial forking leg modparam("tm", "failure_reply_mode", 3) # default retransmission timeout: 30sec modparam("tm", "fr_timer", 30000) # default invite retransmission timeout after 1xx: 120sec modparam("tm", "fr_inv_timer", 120000)
# suppress the check for the CSEQ method # modparam("sanity", "default_checks", 967)
# ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) #edit asterisk #!ifdef WITH_ASTERISK modparam("rr", "append_fromtag", 1) #!else modparam("rr", "append_fromtag", 0) #!endif
# ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ modparam("registrar", "max_contacts", 256) # max value for expires of registrations modparam("registrar", "max_expires", 3600)
# ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ #!ifdef WITH_ACCDB modparam("acc", "db_flag", FLT_ACC) modparam("acc", "db_missed_flag", FLT_ACCMISSED) modparam("acc", "db_url", DBURL) modparam("acc", "db_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd") #!endif
# ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif
#chris commented out this part # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "load_credentials", "") #!ifdef WITH_ASTERISK modparam("auth_db", "user_column", "username") modparam("auth_db", "password_column", "sippasswd") modparam("auth_db", "db_url", DBASTURL) modparam("auth_db", "version_table", 0) #!else modparam("auth_db", "db_url", DBURL) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", MULTIDOMAIN) #!endif
# ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif
#!endif
# ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif
# ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif
# ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif
#!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL)
# ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif
#!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:192.168.14.25:22222")
# ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@teopad-toip.corp")
# params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") modparam("usrloc", "nat_bflag", FLB_NATB) #!endif
#!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/etc/kamailio/tls.cfg") #!endif
#!ifdef WITH_ANTIFLOOD # ----- pike params ----- modparam("pike", "sampling_time_unit", 2) modparam("pike", "reqs_density_per_unit", 16) modparam("pike", "remove_latency", 4)
# ----- htable params ----- # ip ban htable with autoexpire after 5 minutes modparam("htable", "htable", "ipban=>size=8;autoexpire=300;") #!endif
#!ifdef WITH_XMLRPC # ----- xmlrpc params ----- modparam("xmlrpc", "route", "XMLRPC"); modparam("xmlrpc", "url_match", "^/RPC") #!endif
#!ifdef WITH_DEBUG # ----- debugger params ----- modparam("debugger", "cfgtrace", 1) #!endif
####### Routing Logic ########
# Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route {
# per request initial checks route(REQINIT); xlog("L_INFO","apres REQINIT");
# NAT detection route(NATDETECT); xlog("L_INFO","apres NATDETECT");
# handle requests within SIP dialogs route(WITHINDLG); xlog("L_INFO","apres WITHINDLG");
### only initial requests (no To tag)
# CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; }
t_check_trans(); xlog("L_INFO","apres t_check_trans");
# authentication route(AUTH); xlog("L_INFO","apres AUTH");
# record routing for dialog forming requests (in case they are routed) # - remove preloaded route headers remove_hf("Route"); if (is_method("INVITE|SUBSCRIBE")) record_route(); xlog("L_INFO","apres RECORD ROUTE");
# account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } xlog("L_INFO","apres INVITE");
# dispatch requests to foreign domains route(SIPOUT); xlog("L_INFO","apres SIPOUT");
### requests for my local domains
# handle presence related requests route(PRESENCE); xlog("L_INFO","apres PRESENCE");
# handle registrations route(REGISTRAR); xlog("L_INFO","apres REGISTRAR");
if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; }
# dispatch destinations to PSTN route(PSTN); xlog("L_INFO","apres PSTN");
# user location service route(LOCATION); xlog("L_INFO","apres LOCATION");
route(RELAY); xlog("L_INFO","apres RELAY"); }
route[RELAY] {
# enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. xlog("L_INFO","Dans route relay"); if (is_method("INVITE|SUBSCRIBE")) { xlog("L_INFO","avant manage branch"); t_on_branch("MANAGE_BRANCH"); xlog("L_INFO","avant manage reply"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { xlog("L_INFO","avant manage failure"); t_on_failure("MANAGE_FAILURE"); }
if (!t_relay()) { xlog("L_INFO","reply error"); sl_reply_error(); } exit; }
# Per SIP request initial checks route[REQINIT] { #!ifdef WITH_ANTIFLOOD # flood dection from same IP and traffic ban for a while # be sure you exclude checking trusted peers, such as pstn gateways # - local host excluded (e.g., loop to self) if(src_ip!=myself) { if($sht(ipban=>$si)!=$null) { # ip is already blocked xdbg("request from blocked IP - $rm from $fu (IP:$si:$sp)\n"); exit; } if (!pike_check_req()) { xlog("L_ALERT","ALERT: pike blocking $rm from $fu (IP:$si:$sp)\n"); $sht(ipban=>$si) = 1; exit; } } #!endif
if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }
# if(!sanity_check("1511", "7")) # { # xlog("Malformed SIP message from $si:$sp\n"); # exit; # } }
# Handle requests within SIP dialogs route[WITHINDLG] { xlog("L_INFO","Dans WITHINDLG"); if (has_totag()) { xlog("L_INFO","dans has totag"); # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { xlog("L_INFO","looseroute"); if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { xlog("L_INFO","ack"); # ACK is forwarded statelessy route(NATMANAGE); } xlog("L_INFO","relay"); route(RELAY); } else { xlog("L_INFO","else"); if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests xlog("L_INFO","subscribe avant presence"); route(PRESENCE); xlog ("L_INFO","apres presence"); exit; } if ( is_method("ACK") ) { xlog("L_INFO","else ack"); if ( t_check_trans() ) { # no loose-route, but stateful ACK; # must be an ACK after a 487 # or e.g. 404 from upstream server xlog("L_INFO","else ack avant relay"); t_relay(); xlog("L_INFO","else ack apres relay"); exit; } else { # ACK without matching transaction ... ignore and discard xlog("L_INFO","else final"); exit; } } sl_send_reply("404","Not here"); } exit; } }
# Handle SIP registrations route[REGISTRAR] { if (is_method("REGISTER")) { if(isflagset(FLT_NATS)) { setbflag(FLB_NATB); # uncomment next line to do SIP NAT pinging setbflag(FLB_NATSIPPING); } if (!save("location")) sl_reply_error(); #edit asterisk #!ifdef WITH_ASTERISK xlog ("L_INFO","avant regfwd dans registrar"); route(REGFWD); xlog ("L_INFO","apres regfwd dans registrar");
#!endif
exit; } }
# USER location service route[LOCATION] {
#!ifdef WITH_SPEEDIAL # search for short dialing - 2-digit extension if($rU=~"^[0-9][0-9]$") if(sd_lookup("speed_dial")) route(SIPOUT); #!endif
#!ifdef WITH_ALIASDB # search in DB-based aliases if(alias_db_lookup("dbaliases")) route(SIPOUT); #!endif
#edit asterisk #!ifdef WITH_ASTERISK if(is_method("INVITE") && (!route(FROMASTERISK))) { #if new call from out there - send to Asterisk # - non-INVITE requests are routed directly by Kamailio # - traffic from Asterisk is router also directly by Kamailio xlog ("L_INFO", "avant toasterisk dans location"); route(TOASTERISK); xlog ("L_INFO", "apres toasterisk dans location"); exit; } #!endif $avp(oexten) = $rU; if (!lookup("location")) { $var(rc) = $rc; route(TOVOICEMAIL); t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } }
# when routing via usrloc, log the missed calls also if (is_method("INVITE")) { setflag(FLT_ACCMISSED); } }
# Presence server route route[PRESENCE] { if(!is_method("PUBLISH|SUBSCRIBE")) return;
#!ifdef WITH_PRESENCE if (!t_newtran()) { sl_reply_error(); exit; };
if(is_method("PUBLISH")) { handle_publish(); t_release(); } else if( is_method("SUBSCRIBE")) { handle_subscribe(); t_release(); } exit; #!endif
# if presence enabled, this part will not be executed if (is_method("PUBLISH") || $rU==$null) { sl_send_reply("404", "Not here"); exit; } return; }
# Authentication route route[AUTH] { #!ifdef WITH_AUTH
#edit asterisk #!ifdef WITH_ASTERISK #do not auth traffic from Asterisk: trusted! xlog ("L_INFO", "avant if route fromasterisk"); if(route(FROMASTERISK)) return; #!endif
if (is_method("REGISTER")) # { # # authenticate the REGISTER requests (uncomment to enable auth) # if (!www_authorize("$td", "subscriber")) # { # www_challenge("$td", "0"); # exit; # } # # if ($au!=$tU) # { # sl_send_reply("403","Forbidden auth ID"); # exit; # }
{
#edit asterisk ##!ifdef WITH_ASTERISK # xlog ("L_INFO", "dans auth / authcheck sipusers"); # if (!auth_check("$fd","sipusers","1")) ##!else if(is_present_hf("Authorization")) ##!endif
{ # ldap search if (!ldap_search("ldap://sipaccounts/OU=SIP,OU=Utilisateurs,DC=teopad-toip,DC=corp?teopad-Sip-Username,teopadSipPassword?one?(teopad-Sip-Username=$fU)")) # if (!ldap_search("ldap://sipaccounts/OU=SIP,OU=Utilisateurs,DC=teopad-toip,DC=corp?sAMAccountName,?one?(sAMAccountName=$fU)")) { switch ($retcode) { case -1: # no LDAP entry found sl_send_reply("404", "User Not Found"); xlog("L_INFO", "ldap_search: NO found [$retcode] entries for (uid=$fU)"); exit;
case -2: # internal error sl_send_reply("500", "Internal server error"); exit;
default: exit; } } ldap_result("teopad-Sip-Username/$avp(s:username)"); ldap_result("teopadSipPassword/$avp(s:password)"); xlog("L_INFO", "ldap_search: found [$retcode] entries for (uid=$fU)"); if(!pv_www_authenticate("$td", "$avp(s:password)", "0")) { xlog ("L_INFO", "ldap pv_authenticate failed") ; www_challenge("$td","1"); exit; } save("location"); sl_send_reply("200", "ok"); xlog ("L_INFO", "ldap pv_authenticate ok") ; exit; } else { www_challenge("$td","1"); exit; }
} else {
#!ifdef WITH_IPAUTH if(allow_source_address()) { # source IP allowed return; } #!endif
# # authenticate if from local subscriber if (from_uri==myself) { # if (!proxy_authorize("$fd", "subscriber")) { # proxy_challenge("$fd", "0"); # exit; # } if (is_method("PUBLISH")) { xlog ("L_INFO", "au = $au") ; xlog ("L_INFO", "fU = $fU") ; xlog ("L_INFO", "tU = $tU") ; xlog ("L_INFO", "fd = $fd") ; xlog ("L_INFO", "rd = $rd") ;
if ($au!=$fU || $au!=$tU) { sl_send_reply("403","Forbidden auth ID au!=fu ou au!=tu"); exit; } if ($au!=$rU) { sl_send_reply("403","Forbidden R-URI"); exit; } #!ifdef WITH_MULTIDOMAIN if ($fd!=$rd) { sl_send_reply("403","Forbidden R-URI domain"); exit; } #!endif } else { xlog ("L_INFO", "au = $au") ; xlog ("L_INFO", "fU = $fU") ; # if ($au!=$fU) { # sl_send_reply("403","Forbidden auth ID au!=fu"); # exit; # } }
#consume_credentials(); # caller authenticated } else { # caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (!uri==myself) { sl_send_reply("403","Not relaying"); exit; } } } #!endif return; }
# Caller NAT detection route route[NATDETECT] { #!ifdef WITH_NAT force_rport(); if (nat_uac_test("19")) { if (is_method("REGISTER")) { fix_nated_register(); } else { fix_nated_contact(); } setflag(FLT_NATS); } #!endif return; }
# RTPProxy control route[NATMANAGE] { #!ifdef WITH_NAT if (is_request()) { if(has_totag()) { if(check_route_param("nat=yes")) { setbflag(FLB_NATB); } } } if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB))) return;
rtpproxy_manage(); #rtpproxy_manage("co","82.127.95.167");
if (is_request()) { if (!has_totag()) { add_rr_param(";nat=yes"); } } if (is_reply()) { if(isbflagset(FLB_NATB)) { fix_nated_contact(); } } #!endif return; }
# Routing to foreign domains route[SIPOUT] { if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(RELAY); } }
# PSTN GW routing route[PSTN] { #!ifdef WITH_PSTN # check if PSTN GW IP is defined if (strempty($sel(cfg_get.pstn.gw_ip))) { xlog("SCRIPT: PSTN rotuing enabled but pstn.gw_ip not defined\n"); return; }
# route to PSTN dialed numbers starting with '+' or '00' # (international format) # - update the condition to match your dialing rules for PSTN routing if(!($rU=~"^(+|00)[1-9][0-9]{3,20}$")) return;
# only local users allowed to call if(from_uri!=myself) { sl_send_reply("403", "Not Allowed"); exit; }
$ru = "sip:" + $rU + "@" + $sel(cfg_get.pstn.gw_ip);
route(RELAY); exit; #!endif
return; }
# XMLRPC routing #!ifdef WITH_XMLRPC route[XMLRPC] { # allow XMLRPC from localhost if ((method=="POST" || method=="GET") && (src_ip==127.0.0.1)) { # close connection only for xmlrpclib user agents (there is a bug in # xmlrpclib: it waits for EOF before interpreting the response). if ($hdr(User-Agent) =~ "xmlrpclib") set_reply_close(); set_reply_no_connect(); dispatch_rpc(); exit; } send_reply("403", "Forbidden"); exit; } #!endif
# route to voicemail server route[TOVOICEMAIL] { #!ifdef WITH_VOICEMAIL if(!is_method("INVITE")) return;
# check if VoiceMail server IP is defined if (strempty($sel(cfg_get.voicemail.srv_ip))) { xlog("SCRIPT: VoiceMail rotuing enabled but IP not defined\n"); return; } if($avp(oexten)==$null) return;
$ru = "tcp:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); xlog("L_INFO","tovoicemail ru: $ru"); route(RELAY); exit; #!endif
return; }
# manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); }
# manage incoming replies onreply_route[MANAGE_REPLY] { xlog("L_INFO","dans managereply"); xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]"){ xlog("L_INFO","avant route natmanage"); route(NATMANAGE); } }
# manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE);
if (t_is_canceled()) { exit; }
#!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif
#!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif }
#edit asterisk #!ifdef WITH_ASTERISK # Test if coming from Asterisk route[FROMASTERISK] { xlog ("L_INFO", "Dans FROMASTERISK? $si / $sp"); if($si==$sel(cfg_get.asterisk.bindip) && $sp==$sel(cfg_get.asterisk.bindport)) return 1; return -1; }
# Send to Asterisk route[TOASTERISK] { $du = "sip:" + $sel(cfg_get.asterisk.bindip) + ":" + $sel(cfg_get.asterisk.bindport); xlog ("L_INFO", "Dans TOASTERISK $du") ; xlog ("L_INFO", "Juste avant route relay"); route(RELAY); exit; }
# Forward REGISTER to Asterisk route[REGFWD] { xlog("L_INFO", "Dans REGFWD"); if(!is_method("REGISTER")) { return; } $var(rip) = $sel(cfg_get.asterisk.bindip); $uac_req(method)="REGISTER"; $uac_req(ruri)="sip:" + $var(rip) + ":" + $sel(cfg_get.asterisk.bindport); $uac_req(furi)="sip:" + $au + "@" + $var(rip); $uac_req(turi)="sip:" + $au + "@" + $var(rip); $uac_req(hdrs)="Contact: <sip:" + $au + "@" + $sel(cfg_get.kamailio.bindip) + ":" + $sel(cfg_get.kamailio.bindport) + ">\r\n"; xlog("L_INFO","avant if dans regfwd"); if($sel(contact.expires) != $null) $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $sel(contact.expires) + "\r\n"; else $uac_req(hdrs)= $uac_req(hdrs) + "Expires: " + $hdr(Expires) + "\r\n"; uac_req_send(); } #!endif
15 nov 2012 kl. 11:58 skrev Christophe ROY christophe.roy.thales@gmail.com:
Hi everyone
I'm trying to integrate Asterisk with Kamailio for voicemail. I tried to follow this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb BUT:
- I had to adapt it because I use LDAP authentication with Kamailio
- I had problems with Asterisk 10.7 (problems with chan_sip module crashing) so I've installed Asterisk 11 on another VM
- we have high-availability with 2 Kamailio servers, with Kamailio listening on TCP (constraint from our SSL gateway in front of Kamailio) on a "virtual IP" (created by keep-alive): this VIP is not visible with ifconfig, but you can see it with the command "ip addr sh eth0"
For now, we use Linphone on Windows as SIP clients to test. If I don't define WITH_ASTERISK, calls work, I can call someone@domain.tld However, if I define WITH_ASTERISK, calls fail (even with destination registered and available) and I have these errors in the logfile:
Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [ut.h:333]: no corresponding socket for af 2 Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket)
Seems like Kamailio and ASterisk is not using the same transports. Check sip.conf in Asterisk so that you enable the proper transports that you are using for forwarding. If Asterisk is ONLY listening to udp, add ";transport=udp" to the forwarding URI. To force TCP, use "transport=tcp".
Now since the error message indicates proto 1, which in Kamailio-speak is UDP, it seems like you have an issue with that.
Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: INFO: <script>: reply error Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL)
It seems to happen on the if (!t_relay()) line in ROUTE[RELAY]
192.168.14.25 is the real IP of the Kamailio server, 192.168.14.24 is the VIP of the Kamailio "cluster" 192.168.14.28 is the IP of the Mysql server 192.168.14.32 is the IP of the Asterisk server
I can't find why the relay doesn't work. I've tried to bypass the VIP and have Kamailio listen on the real IP, but it still doesn't work: I don't seem to have the same errors as above, but I don't see any traffic between Kamailio and Asterisk.
What could be the problem? Thanks for your help
If you forward register to Asterisk, you have to configure outboundproxy in sip.conf in asterisk so that you get messages back from Asterisk. Or use one of my branchces with support for the SIP Path header in Asterisk (using the PATH module in Kamailio).
Using the onsend route you can check IP, port and transport used to deliver a message from Kamailio. CHeck the Kamailio cookbook on the wiki for more information about that.
/O
-- * Olle E. Johansson - oej@edvina.net * Kamailio & SIP Masterclass Miami FL December 2012 * http://edvina.net/training/
Thanks Olle, it helped a lot Now, calls come through Asterisk and voicemail is working.... but it's "working too well" ;)
When I try to call someone, Asterisk tells me that the subscriber is absent and I'm sent directly to voicemail:
app_dial.c:2433 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
if I take a look in the asterisk CLI, I have that:
rtpproxy1*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status Description Realtime kamailio (Unspecified) a A 0 Unmonitored siptest2/siptest2 (Unspecified) D 0 Unmonitored Cached RT testteopad2/testteopad2 (Unspecified) D 0 Unmonitored Cached RT 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 3 offline]
rtpproxy1*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time 0 SIP registrations.
And I'm not sure I understand correctly the line "Be sure you configure Asterisk to not authenticate SIP requests coming from Kamailio." in the tutorial: I've tried to add in sip.conf these lines:
[kamailio] type=friend permit=192.168.14.0/24 secret= transport=tcp outboundproxy=192.168.14.25
(Kamailio is 192.168.14.25)
Thanks for your help
Christophe
2012/11/15 Olle E. Johansson oej@edvina.net
15 nov 2012 kl. 11:58 skrev Christophe ROY christophe.roy.thales@gmail.com:
Hi everyone
I'm trying to integrate Asterisk with Kamailio for voicemail. I tried to follow this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb BUT:
- I had to adapt it because I use LDAP authentication with Kamailio
- I had problems with Asterisk 10.7 (problems with chan_sip module crashing) so I've installed Asterisk 11 on another VM
- we have high-availability with 2 Kamailio servers, with Kamailio listening on TCP (constraint from our SSL gateway in front of Kamailio) on a "virtual IP" (created by keep-alive): this VIP is not visible with ifconfig, but you can see it with the command "ip addr sh eth0"
For now, we use Linphone on Windows as SIP clients to test. If I don't define WITH_ASTERISK, calls work, I can call someone@domain.tld However, if I define WITH_ASTERISK, calls fail (even with destination registered and available) and I have these errors in the logfile:
Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [ut.h:333]: no corresponding socket for af 2 Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket)
Seems like Kamailio and ASterisk is not using the same transports. Check sip.conf in Asterisk so that you enable the proper transports that you are using for forwarding. If Asterisk is ONLY listening to udp, add ";transport=udp" to the forwarding URI. To force TCP, use "transport=tcp".
Now since the error message indicates proto 1, which in Kamailio-speak is UDP, it seems like you have an issue with that.
Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: INFO: <script>: reply error Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL)
It seems to happen on the if (!t_relay()) line in ROUTE[RELAY]
192.168.14.25 is the real IP of the Kamailio server, 192.168.14.24 is the VIP of the Kamailio "cluster" 192.168.14.28 is the IP of the Mysql server 192.168.14.32 is the IP of the Asterisk server
I can't find why the relay doesn't work. I've tried to bypass the VIP and have Kamailio listen on the real IP, but it still doesn't work: I don't seem to have the same errors as above, but I don't see any traffic between Kamailio and Asterisk.
What could be the problem? Thanks for your help
If you forward register to Asterisk, you have to configure outboundproxy in sip.conf in asterisk so that you get messages back from Asterisk. Or use one of my branchces with support for the SIP Path header in Asterisk (using the PATH module in Kamailio).
Using the onsend route you can check IP, port and transport used to deliver a message from Kamailio. CHeck the Kamailio cookbook on the wiki for more information about that.
/O
--
- Olle E. Johansson - oej@edvina.net
- Kamailio & SIP Masterclass Miami FL December 2012
- http://edvina.net/training/
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
If that helps, here's the debug seen from asterisk when I try to call from testteopad2@domain.corp to siptest2@domain.corp: (.188 is the IP of testteopad2 and .181 the ip of siptest2)
<--- SIP read from TCP:192.168.14.25:44622 ---> INVITE sip:siptest2@192.168.14.25 SIP/2.0 Record-Route: sip:192.168.14.25;transport=tcp;lr=on;ftag=19669 Via: SIP/2.0/TCP 192.168.14.25;branch=z9hG4bK4c88.74c368d3.0;i=c Via: SIP/2.0/TCP 192.168.14.188:5060;rport=3075;branch=z9hG4bK5884 From: sip:testteopad2@domain.corp;tag=19669 To: "Alexis" sip:siptest2@192.168.14.25 Call-ID: 8250 CSeq: 20 INVITE Contact: sip:testteopad2@192.168.14.188:3075;transport=tcp Content-Type: application/sdp Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Max-Forwards: 69 User-Agent: Linphone/3.5.0 (eXosip2/3.6.0) Subject: Phone call Content-Length: 320
v=0 o=testteopad2 752 752 IN IP4 192.168.14.188 s=Talk c=IN IP4 192.168.14.188 t=0 0 m=audio 7078 RTP/AVP 112 111 110 3 0 8 101 a=rtpmap:112 speex/32000 a=fmtp:112 vbr=on a=rtpmap:111 speex/16000 a=fmtp:111 vbr=on a=rtpmap:110 speex/8000 a=fmtp:110 vbr=on a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 <-------------> --- (15 headers 14 lines) --- Sending to 192.168.14.25:44622 (NAT) Using INVITE request as basis request - 8250 Found peer 'testteopad2' for 'testteopad2' from 192.168.14.25:44622 Found RTP audio format 112 Found RTP audio format 111 Found RTP audio format 110 Found RTP audio format 3 Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Found audio description format speex for ID 112 Found audio description format speex for ID 111 Found audio description format speex for ID 110 Found audio description format telephone-event for ID 101 Capabilities: us - (gsm), peer - audio=(gsm|ulaw|alaw|speex|speex16|speex32)/video=(nothing)/text=(nothing), combined - (gsm) Non-codec capabilities (dtmf): us - 0x1 (telephone-event|), peer - 0x1 (telephone-event|), combined - 0x1 (telephone-event|) Peer audio RTP is at port 192.168.14.188:7078 Looking for siptest2 in teopad (domain 192.168.14.25) list_route: hop: sip:192.168.14.25;transport=tcp;lr=on;ftag=19669
<--- Transmitting (no NAT) to 192.168.14.25:5060 ---> SIP/2.0 100 Trying Via: SIP/2.0/TCP 192.168.14.25;branch=z9hG4bK4c88.74c368d3.0;i=c;received=192.168.14.25 Via: SIP/2.0/TCP 192.168.14.188:5060;rport=3075;branch=z9hG4bK5884 Record-Route: sip:192.168.14.25;transport=tcp;lr=on;ftag=19669 From: sip:testteopad2@domain.corp;tag=19669 To: "Alexis" sip:siptest2@192.168.14.25 Call-ID: 8250 CSeq: 20 INVITE Server: Asterisk PBX 11.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:siptest2@192.168.14.32:5060;transport=TCP Content-Length: 0
<------------> Really destroying SIP dialog '35ccffe644eb697214d4d7e34f968a2b@192.168.14.32:5060' Method: INVITE [Nov 19 16:00:59] WARNING[25058][C-00000002]: app_dial.c:2433 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent) Audio is at 19994 Adding codec 100002 (gsm) to SDP Adding non-codec 0x1 (telephone-event) to SDP
<--- Reliably Transmitting (no NAT) to 192.168.14.25:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.14.25;branch=z9hG4bK4c88.74c368d3.0;i=c;received=192.168.14.25 Via: SIP/2.0/TCP 192.168.14.188:5060;rport=3075;branch=z9hG4bK5884 Record-Route: sip:192.168.14.25;transport=tcp;lr=on;ftag=19669 From: sip:testteopad2@domain.corp;tag=19669 To: "Alexis" sip:siptest2@192.168.14.25;tag=as73c46699 Call-ID: 8250 CSeq: 20 INVITE Server: Asterisk PBX 11.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Contact: sip:siptest2@192.168.14.32:5060;transport=TCP Content-Type: application/sdp Content-Length: 261
v=0 o=root 500542260 500542260 IN IP4 192.168.14.32 s=Asterisk PBX 11.0.1 c=IN IP4 192.168.14.32 t=0 0 m=audio 19994 RTP/AVP 3 101 a=rtpmap:3 GSM/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - a=ptime:20 a=sendrecv
<------------>
<--- SIP read from TCP:192.168.14.25:44622 ---> ACK sip:siptest2@192.168.14.32:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 192.168.14.25;branch=z9hG4bKcydzigwkX;i=c Via: SIP/2.0/TCP 192.168.14.188:5060;rport=3075;branch=z9hG4bK22228 From: sip:testteopad2@domain.corp;tag=19669 To: "Alexis" sip:siptest2@192.168.14.25;tag=as73c46699 Call-ID: 8250 CSeq: 20 ACK Contact: sip:testteopad2@192.168.14.188:3075;transport=tcp Max-Forwards: 69 User-Agent: Linphone/3.5.0 (eXosip2/3.6.0) Content-Length: 0
<-------------> --- (11 headers 0 lines) ---
<--- SIP read from TCP:192.168.14.25:44622 ---> BYE sip:siptest2@192.168.14.32:5060;transport=TCP SIP/2.0 Via: SIP/2.0/TCP 192.168.14.25;branch=z9hG4bK5c88.bc649332.0;i=c Via: SIP/2.0/TCP 192.168.14.188:5060;rport=3075;branch=z9hG4bK1571 From: sip:testteopad2@domain.corp;tag=19669 To: "Alexis" sip:siptest2@192.168.14.25;tag=as73c46699 Call-ID: 8250 CSeq: 21 BYE Contact: sip:testteopad2@192.168.14.188:5060;transport=TCP Max-Forwards: 69 User-Agent: Linphone/3.5.0 (eXosip2/3.6.0) Content-Length: 0
<-------------> --- (11 headers 0 lines) --- Sending to 192.168.14.25:5060 (no NAT) Scheduling destruction of SIP dialog '8250' in 32000 ms (Method: BYE)
<--- Transmitting (no NAT) to 192.168.14.25:5060 ---> SIP/2.0 200 OK Via: SIP/2.0/TCP 192.168.14.25;branch=z9hG4bK5c88.bc649332.0;i=c;received=192.168.14.25 Via: SIP/2.0/TCP 192.168.14.188:5060;rport=3075;branch=z9hG4bK1571 From: sip:testteopad2@domain.corp;tag=19669 To: "Alexis" sip:siptest2@192.168.14.25;tag=as73c46699 Call-ID: 8250 CSeq: 21 BYE Server: Asterisk PBX 11.0.1 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH Supported: replaces, timer Content-Length: 0
<------------>
2012/11/19 Christophe ROY christophe.roy.thales@gmail.com:
Thanks Olle, it helped a lot Now, calls come through Asterisk and voicemail is working.... but it's "working too well" ;)
When I try to call someone, Asterisk tells me that the subscriber is absent and I'm sent directly to voicemail:
app_dial.c:2433 dial_exec_full: Unable to create channel of type 'SIP' (cause 20 - Subscriber absent)
if I take a look in the asterisk CLI, I have that:
rtpproxy1*CLI> sip show peers Name/username Host Dyn Forcerport ACL Port Status Description Realtime kamailio (Unspecified) a A 0 Unmonitored siptest2/siptest2 (Unspecified) D 0 Unmonitored Cached RT testteopad2/testteopad2 (Unspecified) D 0 Unmonitored Cached RT 3 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 3 offline]
rtpproxy1*CLI> sip show registry Host dnsmgr Username Refresh State Reg.Time 0 SIP registrations.
And I'm not sure I understand correctly the line "Be sure you configure Asterisk to not authenticate SIP requests coming from Kamailio." in the tutorial: I've tried to add in sip.conf these lines:
[kamailio] type=friend permit=192.168.14.0/24 secret= transport=tcp outboundproxy=192.168.14.25
(Kamailio is 192.168.14.25)
Thanks for your help
Christophe
2012/11/15 Olle E. Johansson oej@edvina.net
15 nov 2012 kl. 11:58 skrev Christophe ROY christophe.roy.thales@gmail.com:
Hi everyone
I'm trying to integrate Asterisk with Kamailio for voicemail. I tried to follow this tutorial: http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb BUT:
- I had to adapt it because I use LDAP authentication with Kamailio
- I had problems with Asterisk 10.7 (problems with chan_sip module crashing) so I've installed Asterisk 11 on another VM
- we have high-availability with 2 Kamailio servers, with Kamailio listening on TCP (constraint from our SSL gateway in front of Kamailio) on a "virtual IP" (created by keep-alive): this VIP is not visible with ifconfig, but you can see it with the command "ip addr sh eth0"
For now, we use Linphone on Windows as SIP clients to test. If I don't define WITH_ASTERISK, calls work, I can call someone@domain.tld However, if I define WITH_ASTERISK, calls fail (even with destination registered and available) and I have these errors in the logfile:
Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [ut.h:333]: no corresponding socket for af 2 Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket)
Seems like Kamailio and ASterisk is not using the same transports. Check sip.conf in Asterisk so that you enable the proper transports that you are using for forwarding. If Asterisk is ONLY listening to udp, add ";transport=udp" to the forwarding URI. To force TCP, use "transport=tcp".
Now since the error message indicates proto 1, which in Kamailio-speak is UDP, it seems like you have an issue with that.
Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: tm [t_fwd.c:1530]: ERROR: t_forward_nonack: failure to add branches Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: INFO: <script>: reply error Nov 15 11:45:08 kamailio1 /usr/sbin/kamailio[25308]: ERROR: sl [sl_funcs.c:371]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL)
It seems to happen on the if (!t_relay()) line in ROUTE[RELAY]
192.168.14.25 is the real IP of the Kamailio server, 192.168.14.24 is the VIP of the Kamailio "cluster" 192.168.14.28 is the IP of the Mysql server 192.168.14.32 is the IP of the Asterisk server
I can't find why the relay doesn't work. I've tried to bypass the VIP and have Kamailio listen on the real IP, but it still doesn't work: I don't seem to have the same errors as above, but I don't see any traffic between Kamailio and Asterisk.
What could be the problem? Thanks for your help
If you forward register to Asterisk, you have to configure outboundproxy in sip.conf in asterisk so that you get messages back from Asterisk. Or use one of my branchces with support for the SIP Path header in Asterisk (using the PATH module in Kamailio).
Using the onsend route you can check IP, port and transport used to deliver a message from Kamailio. CHeck the Kamailio cookbook on the wiki for more information about that.
/O
--
- Olle E. Johansson - oej@edvina.net
- Kamailio & SIP Masterclass Miami FL December 2012
- http://edvina.net/training/
SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users@lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users