- sr-users - kamailio.org

Kamailio 3.1.5 crashing during TLS connection setup
by Kamal Palei 22 Oct '12

22 Oct '12

Dear All I have setup Kamailio 3.1.5 with TLS enabled. Whenever I receive the REGISTER request from endpoint, trying to forward to next proxy. For that I have added below code in kamailio.cfg * if(is_method("REGISTER")) { t_relay_to("tls:115.114.48.75:5061"); exit(); } *I see once REGISTER received at Kamailio proxy, it is sending TCP synch packet to next proxy, next proxy sends back a synch+reset packet. Then kamailio proxy is crashing with log as below. *root@B2BUA:/usr/local/src/scripts# 9(2347) ERROR: <core> [tcp_main.c:4139]: connect 115.114.48.75:5061 failed 9(2347) : <core> [mem/q_malloc.c:431]: BUG: qm_free: bad pointer (nil) (out of memory block!) - aborting * The complete log is given as below. Please let me know how to get out of this issue. I used -m 64 option while starting proxy, but same result. Thanks, Kamal, NECS, Bangalore * 0(2336) INFO: <core> [tcp_main.c:4730]: init_tcp: using epoll_et io watch method (config) 0(2338) INFO: usrloc [hslot.c:53]: locks array size 512 0(2338) INFO: tls [tls_init.c:511]: tls: _init_tls_h: compiled with openssl version "OpenSSL 1.0.1c 10 May 2012" (0x1000103f), kerberos support: off, compression: on 0(2338) INFO: tls [tls_init.c:519]: tls: init_tls_h: installed openssl library version "OpenSSL 1.0.1c 10 May 2012" (0x1000103f), kerberos support: off, zlib compression: off compiler: gcc -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -Wa,--noexecstack -DL_ENDIAN -DTERMIO -O3 -fomit-frame-pointer -Wall -DOPENSSL_BN_ASM_PART_WORDS -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_GF2m -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DRMD160_ASM -DAES_ASM -DVPAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM 0(2338) INFO: tls [tls_init.c:373]: tls: init_tls: disabling compression... 0(2338) WARNING: tls [tls_init.c:587]: tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 5242880 and 2621440 bytes 0(2338) INFO: <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 5242880 0(2338) INFO: <core> [cfg/cfg_ctx.c:411]: INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 2621440 0(2338) INFO: <core> [udp_server.c:184]: INFO: udp_init: SO_RCVBUF is initially 112640 0(2338) INFO: <core> [udp_server.c:235]: INFO: udp_init: SO_RCVBUF is finally 262142 0(2338) INFO: <core> [udp_server.c:184]: INFO: udp_init: SO_RCVBUF is initially 112640 0(2338) INFO: <core> [udp_server.c:235]: INFO: udp_init: SO_RCVBUF is finally 262142 0(2338) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 0(2338) INFO: tls [tls_domain.c:176]: TLSs<default>: tls_method=9 0(2338) INFO: tls [tls_domain.c:188]: TLSs<default>: certificate='/etc/certs/pocserver.com/cert.pem' 0(2338) INFO: tls [tls_domain.c:195]: TLSs<default>: ca_list='(null)' 0(2338) INFO: tls [tls_domain.c:202]: TLSs<default>: crl='(null)' 0(2338) INFO: tls [tls_domain.c:206]: TLSs<default>: require_certificate=0 0(2338) INFO: tls [tls_domain.c:213]: TLSs<default>: cipher_list='(null)' 0(2338) INFO: tls [tls_domain.c:220]: TLSs<default>: private_key='/etc/certs/pocserver.com/key.pem' 0(2338) INFO: tls [tls_domain.c:224]: TLSs<default>: verify_certificate=0 0(2338) INFO: tls [tls_domain.c:227]: TLSs<default>: verify_depth=9 0(2338) INFO: tls [tls_domain.c:544]: TLSs<default>: No client certificate required and no checks performed 0(2338) INFO: tls [tls_domain.c:176]: TLSc<default>: tls_method=9 0(2338) INFO: tls [tls_domain.c:188]: TLSc<default>: certificate='(null)' 0(2338) INFO: tls [tls_domain.c:195]: TLSc<default>: ca_list='(null)' 0(2338) INFO: tls [tls_domain.c:202]: TLSc<default>: crl='(null)' 0(2338) INFO: tls [tls_domain.c:206]: TLSc<default>: require_certificate=1 0(2338) INFO: tls [tls_domain.c:213]: TLSc<default>: cipher_list='(null)' 0(2338) INFO: tls [tls_domain.c:220]: TLSc<default>: private_key='(null)' 0(2338) INFO: tls [tls_domain.c:224]: TLSc<default>: verify_certificate=1 0(2338) INFO: tls [tls_domain.c:227]: TLSc<default>: verify_depth=9 0(2338) INFO: tls [tls_domain.c:529]: TLSc<default>: Server MUST present valid certificate 2(2340) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 1(2339) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 4(2342) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 0(2338) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 6(2344) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 3(2341) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 7(2345) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 6(2344) INFO: ctl [io_listener.c:224]: io_listen_loop: using epoll_et io watch method (config) root@B2BUA:/usr/local/src/scripts# 9(2347) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled 8(2346) INFO: rtpproxy [rtpproxy.c:1403]: rtp proxy <udp:127.0.0.1:7729> found, support for it enabled root@B2BUA:/usr/local/src/scripts# 9(2347) ERROR: <core> [tcp_main.c:4139]: connect 115.114.48.75:5061 failed 9(2347) : <core> [mem/q_malloc.c:431]: BUG: qm_free: bad pointer (nil) (out of memory block!) - aborting 0(2338) ALERT: <core> [main.c:742]: child process 2347 exited by a signal 6 *

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Listen on all IPv6 addresses
by Anton Kvashenkin 22 Oct '12

22 Oct '12

Hello guys. In the Kamailio v.3.3.0 release notes http://www.kamailio.org/w/kamailio-v3-3-0-release-notes/ it says: all IPv6 network interfaces are auto-detected and Kamailio start listening on them (if no strict listening rules are set) But it doesn't happen. version: kamailio 3.3.1 (x86_64/linux) d75cb5 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, DEFAULT PKG_SIZE 4MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: d75cb5 listen is commented out @ cfg

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Re: [SR-Users] rtpproxy delete fails
by Juha Heinanen 22 Oct '12

22 Oct '12

the patch below adds 't' flag to rtpproxy_destroy flags. if 't' flag is present when rtpproxy_destroy is called, to tag is not included in D command sent to rtpproxy thus causing destroy of the full call. 't' flag can be used to avoid piling up of unused rtpproxy calls by making it possible to call rtpproxy_destroy when 200 ok is received from a branch where rtpproxy is not used. is it ok for me to commit this patch or do long term rtpproxy users have a better idea? it would still make sense to include a timer in rtpproxy that cleans up un-used calls at an interval that corresponds to sip proxy's invite timeout. -- juha *** /usr/src/orig/sip-router/modules/rtpproxy/rtpproxy.c 2012-10-19 14:31:27.658485534 +0300 --- modules/rtpproxy/rtpproxy.c 2012-10-20 09:52:10.308389743 +0300 *************** *** 1678,1683 **** --- 1678,1684 ---- str callid, from_tag, to_tag, viabranch; char *cp; int via = 0; + int to = 1; int ret; struct rtpp_node *node; struct iovec v[1 + 4 + 3 + 2] = {{NULL, 0}, {"D", 1}, {" ", 1}, {NULL, 0}, {NULL, 0}, {NULL, 0}, {" ", 1}, {NULL, 0}, {" ", 1}, {NULL, 0}}; *************** *** 1700,1705 **** --- 1701,1710 ---- via = 1; break; + case 't': + case 'T': + to = 0; + break; case 'a': case 'A': case 'i': *************** *** 1736,1742 **** return -1; } to_tag.s = 0; ! if (get_to_tag(msg, &to_tag) == -1) { LM_ERR("can't get To tag\n"); return -1; } --- 1741,1748 ---- return -1; } to_tag.s = 0; ! to_tag.len = 0; ! if ((to == 1) && get_to_tag(msg, &to_tag) == -1) { LM_ERR("can't get To tag\n"); return -1; }

2 1

mediaproxy-ng fails to parse start_recording() command
by Juha Heinanen 22 Oct '12

22 Oct '12

i tried what happens when i call start_recording() after calling rtpproxy_manage() and got this kind of error to syslog: Oct 20 19:57:43 siika mediaproxy-ng[8423]: Failed to properly parse UDP command line '8554_13 R ngsgznnwgzpuovt(a)siika.tutpro.com mgpwe' from 127.0.0.1:51497, using fallback RE why the error? it is so that mediaproxy-ng does not support recording? -- juha

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Re: [SR-Users] dangers of excluding 100rel?
by Alex Balashov 19 Oct '12

19 Oct '12

You can strip 100rel as long as it's Supported. However, not so good things happen if you strip it from Required. :-) -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Andrew Mortensen <admorten(a)isc.upenn.edu> wrote:Can I safely strip 100rel support from INVITEs sent by callers registered to my proxy? We're using proxy-generated 302 redirects for call forwarding in our deployment. In general this works better for us than appending branches, as we rapidly ran into branch limits when dealing with large multiple appearance lines. However, we've run into a case where the caller, a Polycom 550, seems to get confused when a parallel forked call is redirected, 100rel is in use and the redirection target is also a multiple appearance line. The caller is correctly responding with PRACKs to all 180s, before and after the redirection. However, hanging up the call on the caller after the redirection, but before any remote party has answered, does *not* cause the caller to send a CANCEL to tear down the call. In fact, the caller doesn't send anything at all. The remote parties continue to ring until the ring timer expires, and the proxy sends CANCELs to all callee endpoints. That's clearly a Polycom bug, and we'll need to work with them to get it resolved. In the meantime, I'm looking for workarounds. One quick, crude option is to strip 100rel from the Supported header in the caller's INVITE. I've tested this, and it does solve the problem, but I'm concerned about side effects. I'm aware of the value 100rel has for calls to the PSTN, but in this case the calls never leave the local network. Are there situations where 100rel is critical for a parallel forked call? andrew _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

dangers of excluding 100rel?
by Andrew Mortensen 19 Oct '12

19 Oct '12

Can I safely strip 100rel support from INVITEs sent by callers registered to my proxy? We're using proxy-generated 302 redirects for call forwarding in our deployment. In general this works better for us than appending branches, as we rapidly ran into branch limits when dealing with large multiple appearance lines. However, we've run into a case where the caller, a Polycom 550, seems to get confused when a parallel forked call is redirected, 100rel is in use and the redirection target is also a multiple appearance line. The caller is correctly responding with PRACKs to all 180s, before and after the redirection. However, hanging up the call on the caller after the redirection, but before any remote party has answered, does *not* cause the caller to send a CANCEL to tear down the call. In fact, the caller doesn't send anything at all. The remote parties continue to ring until the ring timer expires, and the proxy sends CANCELs to all callee endpoints. That's clearly a Polycom bug, and we'll need to work with them to get it resolved. In the meantime, I'm looking for workarounds. One quick, crude option is to strip 100rel from the Supported header in the caller's INVITE. I've tested this, and it does solve the problem, but I'm concerned about side effects. I'm aware of the value 100rel has for calls to the PSTN, but in this case the calls never leave the local network. Are there situations where 100rel is critical for a parallel forked call? andrew

1 0

rtpproxy_destroy([flags]) does not accept flags
by Juha Heinanen 19 Oct '12

19 Oct '12

according to rtpproxy module readme, rtpproxy_destroy function takes optional flags param: 5.4. rtpproxy_destroy([flags]) however, the code does not support it: {"unforce_rtp_proxy", (cmd_function)unforce_rtp_proxy_f, 0, 0, 0, ANY_ROUTE}, {"rtpproxy_destroy", (cmd_function)unforce_rtp_proxy_f, 0, 0, 0, ANY_ROUTE}, what has happened? -- juha

2 3

MSILO Configuration
by Ed Brady 19 Oct '12

19 Oct '12

Hi, I am having trouble configuring MSILO. I copied the example config into my kamailio.cfg file and edited the areas that apply to my domain, but when restarting Kamailio I am faced with an error. Below is the error followed by my complete config file. Please advise what I am doing wrong. Is there some sort of resource for HowTos on Kamailio? The module documentation seems very thorough but the examples at the end seem to only be useful if that module is the only one you need to configure. I am trying to do a fairly straight forward configuration I think (SIP messaging with offline message storing and a single incoming/outgoing gateway that is used for all accounts that do not locally exist), but I am unable to find straight forward documentation on how to configure something like this. Please let me know what I am missing. Thanks, Ed Not starting Kamailio: invalid configuration file! > -e > 0(30863) INFO: tls [tls_init.c:377]: tls: init_tls: disabling > compression... > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 817, column 5: syntax error > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 817, column 5: bad command > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 817, column 5: bad command > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 878, column 12: syntax error > 0(30863) : <core> [cfg.y:3594]: parse error in config file > /etc/kamailio/kamailio.cfg, line 878, column 12: > ERROR: bad config file (5 errors) > *Config* #!KAMAILIO # # Kamailio (OpenSER) SIP Server v3.3 - default configuration script # - web: http://www.kamailio.org # - git: http://sip-router.org # # Direct your questions about this file to: <sr-users(a)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 # # #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_TLS #!define WITH_NAT #!define WITH_ACCDB # # *** 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 # # *** 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 ####### 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://openser:R4U8oO8Er872sNHr@localhost/openser" #!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=yes #!else debug=2 log_stderror=no #!endif memdbg=5 memlog=5 log_facility=LOG_LOCAL0 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="sip.mydomain.com" /* uncomment and configure the following line if you want Kamailio to bind on a specific interface/port/proto (default bind on all available) */ #listen=udp:10.0.0.10:5060 /* 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 = "" desc "VoiceMail IP Address" voicemail.srv_port = "5060" desc "VoiceMail 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/lib64/kamailio/modules_k/:/usr/lib64/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" loadmodule "cfgutils.so" loadmodule "msrp.so" loadmodule "htable.so" loadmodule "msilo.so" #!ifdef WITH_AUTH loadmodule "auth.so" loadmodule "auth_db.so" #!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 # ----------------- setting module-specific parameters --------------- # ----- mi_fifo params ----- modparam("mi_fifo", "fifo_name", "/tmp/kamailio_fifo") # ----- 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) # -- msilo params -- modparam("msilo","db_url",DBURL) modparam("msilo","from_address","sip:registrar@MYDOMAIN.com") modparam("msilo","contact_hdr","Contact: registrar(a)MYDOMAIN.com :5060;msilo=yes\r\n") modparam("msilo","content_type_hdr","Content-Type: text/plain\r\n") modparam("msilo","offline_message","*** User $rU is offline!") # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- registrar params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # max value for expires of registrations modparam("registrar", "max_expires", 3600) # set it to 1 to enable GRUU modparam("registrar", "gruu_enabled", 0) # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) /* by default ww do not adjust the direct of the sequential requests. if you enable this parameter, be sure the enable "append_fromtag" in "rr" module */ modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$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=$rd") #!endif # ----- usrloc params ----- /* enable DB persistency for location entries */ #!ifdef WITH_USRLOCDB modparam("usrloc", "db_url", DBURL) modparam("usrloc", "db_mode", 2) modparam("usrloc", "use_domain", MULTIDOMAIN) #!endif # ----- auth_db params ----- #!ifdef WITH_AUTH modparam("auth_db", "db_url", DBURL) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") modparam("auth_db", "use_domain", MULTIDOMAIN) # ----- permissions params ----- #!ifdef WITH_IPAUTH modparam("permissions", "db_url", DBURL) modparam("permissions", "db_mode", 1) #!endif #!endif # ----- alias_db params ----- #!ifdef WITH_ALIASDB modparam("alias_db", "db_url", DBURL) modparam("alias_db", "use_domain", MULTIDOMAIN) #!endif # ----- speedial params ----- #!ifdef WITH_SPEEDDIAL modparam("speeddial", "db_url", DBURL) modparam("speeddial", "use_domain", MULTIDOMAIN) #!endif # ----- domain params ----- #!ifdef WITH_MULTIDOMAIN modparam("domain", "db_url", DBURL) # use caching modparam("domain", "db_mode", 1) # register callback to match myself condition with domains list modparam("domain", "register_myself", 1) #!endif #!ifdef WITH_PRESENCE # ----- presence params ----- modparam("presence", "db_url", DBURL) # ----- presence_xml params ----- modparam("presence_xml", "db_url", DBURL) modparam("presence_xml", "force_active", 1) #!endif #!ifdef WITH_NAT # ----- rtpproxy params ----- modparam("rtpproxy", "rtpproxy_sock", "udp:127.0.0.1:7722") # ----- nathelper params ----- modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", FLB_NATSIPPING) modparam("nathelper", "sipping_from", "sip:pinger@kamailio.org") # params needed for NAT traversal in other modules modparam("nathelper|registrar", "received_avp", "$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=1800;") #!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); # NAT detection route(NATDETECT); # handle requests within SIP dialogs route(WITHINDLG); ### only initial requests (no To tag) # CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) t_relay(); exit; } t_check_trans(); # authentication route(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(); # account only INVITEs if (is_method("INVITE")) { setflag(FLT_ACC); # do accounting } # dispatch requests to foreign domains route(SIPOUT); ### requests for my local domains # handle presence related requests route(PRESENCE); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # dispatch destinations to PSTN route(PSTN); # user location service route(LOCATION); route(RELAY); } route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } if (!t_relay()) { 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] { if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails } if ( is_method("ACK") ) { # ACK is forwarded statelessy route(NATMANAGE); } route(RELAY); } else { if (is_method("SUBSCRIBE") && uri == myself) { # in-dialog subscribe requests route(PRESENCE); exit; } if ( is_method("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 t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard 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(); 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 $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 #!ifdef WITH_IPAUTH if((!is_method("REGISTER")) && allow_source_address()) { # source IP allowed return; } #!endif if (is_method("REGISTER") || from_uri==myself) { # authenticate requests if (!auth_check("$fd", "subscriber", "1")) { auth_challenge("$fd", "0"); exit; } # user authenticated - remove auth header if(!is_method("REGISTER|PUBLISH")) consume_credentials(); } # if caller is not local subscriber, then check if it calls # a local destination, otherwise deny, not an open relay here if (from_uri!=myself && 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(); 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); } } # Route to MSILO 10-18-12 route{ if ( !mf_process_maxfwd_header("10") ) { sl_send_reply("483","To Many Hops"); exit; }; if (uri==myself) { { # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; exit; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); exit; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; exit; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("$ru")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; exit; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); exit; }; # forward anything else t_relay(); } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { exit; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("$ou")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } # 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 = "sip:" + $avp(oexten) + "@" + $sel(cfg_get.voicemail.srv_ip) + ":" + $sel(cfg_get.voicemail.srv_port); route(RELAY); exit; #!endif return; } # manage outgoing branches branch_route[MANAGE_BRANCH] { xdbg("new branch [$T_branch_idx] to $ru\n"); route(NATMANAGE); } # manage incoming replies onreply_route[MANAGE_REPLY] { xdbg("incoming reply\n"); if(status=~"[12][0-9][0-9]") route(NATMANAGE); } # manage failure routing cases failure_route[MANAGE_FAILURE] { route(NATMANAGE); if (t_is_canceled()) { exit; } #!ifdef WITH_BLOCK3XX # block call redirect based on 3xx replies. if (t_check_status("3[0-9][0-9]")) { t_reply("404","Not found"); exit; } #!endif #!ifdef WITH_VOICEMAIL # serial forking # - route to voicemail on busy or no answer (timeout) if (t_check_status("486|408")) { route(TOVOICEMAIL); exit; } #!endif } event_route[msrp:frame-in] { xdbg("============#[[$msrp(method)]]===========\n"); xdbg("============*[[$si:$sp]]\n"); xdbg("============ crthop: [$msrp(crthop)]\n"); xdbg("============ prevhop: [$msrp(prevhop)]\n"); xdbg("============ nexthop: [$msrp(nexthop)]\n"); xdbg("============ firsthop: [$msrp(firsthop)]\n"); xdbg("============ lasthop: [$msrp(lasthop)]\n"); xdbg("============ prevhops: [$msrp(prevhops)]\n"); xdbg("============ nexthops: [$msrp(nexthops)]\n"); xdbg("============ srcaddr: [$msrp(srcaddr)]\n"); xdbg("============ srcsock: [$msrp(srcsock)]\n"); xdbg("============ sessid: [$msrp(sessid)]\n"); msrp_reply_flags("1"); if(msrp_is_reply()) { msrp_relay(); exit; } # handle AUTH MSRP requests if($msrp(method)=="AUTH") { if($msrp(nexthops)>0) { msrp_relay(); exit; } # frame for local server - send Use-Path # -- passwd can be loaded from DB based on $au $var(passwd) = "xyz123"; if(!pv_www_authenticate("myrealm", "$var(passwd)", "0")) { if(auth_get_www_authenticate("myrealm", "0", "$var(wauth)")) { msrp_reply("401", "Authorization Required", "$var(wauth)"); } else { msrp_reply("500", "Server Error"); } exit; } $var(cnt) = $var(cnt) + 1; pv_printf("$var(sessid)", "s.$(pp).$(var(cnt)).$(RANDOM)"); $sht(msrp=>$var(sessid)::srcaddr) = $msrp(srcaddr); $sht(msrp=>$var(sessid)::srcsock) = $msrp(srcsock); # - Use-Path: the MSRP address for server + session id $var(UsePath) = "Use-Path: msrp://127.0.0.1:5060/" + $var(sessid) + ";tcp\r\n"; msrp_reply("200", "OK", "$var(UsePath)"); exit; } if($msrp(method)=="SEND") { if($msrp(nexthops)>1) { msrp_reply("200", "Received"); msrp_relay(); exit; } $var(sessid) = $msrp(sessid); if($sht(msrp=>$var(sessid)::srcaddr) == $null) { # one more hop, but we don't have address in htable msrp_reply("481", "No Such Session"); exit; } msrp_relay_flags("1"); msrp_set_dst("$sht(msrp=>$var(sessid)::srcaddr)", "$sht(msrp=>$var(sessid)::srcsock)"); msrp_relay(); exit; } msrp_relay(); }

3 2

Re: [SR-Users] Kamailio v3.3.2 Released
by Paul Cupis 18 Oct '12

18 Oct '12

On 16/10/12 22:29, Jon Bonilla (Manwe) wrote: > El Tue, 16 Oct 2012 17:00:03 +0200 > Daniel-Constantin Mierla<miconda(a)gmail.com> escribió: >> >> Kamailio SIP Server v3.3.2 stable release is out. > > Debian and Ubuntu packages are ready. Please find the instructions > below: > > http://www.kamailio.org/dokuwiki/doku.php/packages:debs I'm seeing a problem with the 3.3.2 packages on Debian at the moment. There is a file conflict between the kamailio package and kamailio-mysql-modules preventing the latter from installing. $ dpkg-deb -c /var/cache/apt/archives/kamailio_3.3.2+squeeze1_amd64.deb | grep srdb -rw-r--r-- root/root 74296 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so.1.0 -rw-r--r-- root/root 28360 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so.1 -> libsrdb1.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so -> libsrdb1.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so.1 -> libsrdb2.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so -> libsrdb2.so.1.0 $ dpkg-deb -c /var/cache/apt/archives/kamailio-mysql-modules_3.3.2+squeeze1_amd64.deb | grep srdb -rw-r--r-- root/root 28360 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so.1.0 -rw-r--r-- root/root 74296 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so.1 -> libsrdb2.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb2.so -> libsrdb2.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so.1 -> libsrdb1.so.1.0 lrwxrwxrwx root/root 0 2012-10-16 21:04 ./usr/lib64/kamailio/libsrdb1.so -> libsrdb1.so.1.0 Regards,

2 3

Kamailio- Redirect message 300 with LCR reroute
by Fatima Chahrour~Vanrise Support 18 Oct '12

18 Oct '12

Dear List, I'd like to get help regarding my case. I have the following script where many thanks to Daniel has helped me in if (is_method("INVITE")) { if (!load_gws(1, $rU, $fu)) { sl_send_reply("502", "Unable To lOad GatEwAyS"); exit;} if(!next_gw()){ sl_send_reply("503", "Unable To fInD a gateWaY"); exit;} while(next_gw()){ km_append_branch(); } sl_send_reply("302","Moved Temporary"); exit; } The problem am facing is that the call is rerouting from the first gateway to the next gateway successfully when I have two gateways. BUT when I have three gateways, and the first two gateways are off, the call is not rerouted to the third gateway. It keeps hitting the seconds gateway and gives request time out at the end Why is that? Thanks in advance, F Chahrour

1 0

Kamailio v3.3.2 RPM Packages
by Daniel-Constantin Mierla 18 Oct '12

18 Oct '12

Hello, for the record, RPM packages of Kamailio v3.3.2 were built for Centos 5.5, Centos 6, RedHat 5, RedHat 6, Fedora 15, Fedora 16, Fedora 17, OpenSuse 11.4 and OpenSuse 12.1, on i586 and x86_64 architectures. They are available on opensuse build service repos, see the links at: * http://www.kamailio.org/wiki/packages/rpms Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Question about compliance with RCS/OMA Simple Presence
by Sangeeta Shah 17 Oct '12

17 Oct '12

Hello All, I need some guidance on whether the Kamailio presence/presence_xml and integrated xcap module implementations are RCS/OMA Simple presence compliant. Based on a few things I've seen in the code I don't see any handling for the oma presence rule xml extensions. Is there any plan to be RCS/OMA compliant in the new future? I can get SIMPLE PRESENCE with XCAP to work but not with OMA presence rules and resource lists. Thanks, Sangeeta -- Sangeeta Shah

1 0

has_sdp()??
by Juha Heinanen 17 Oct '12

17 Oct '12

while reading rtpproxy module readme, i found this kind of example: if (is_method("INVITE")) { if (has_sdp()) { if (rtpproxy_offer()) t_on_reply("1"); } else { t_on_reply("2"); } } if (is_method("ACK") && has_sdp()) rtpproxy_answer(); does has_sdp() function really exist? it is not listed on wiki page: http://www.kamailhttp://www.kamailio.org/wiki/alphaindexes/3.4.x/modfunctio… -- juha

4 13

make option to clean the installed binaries
by Ajay Sabat 17 Oct '12

17 Oct '12

Hi, I am a newbie here and recently started using kamailio/ser. I am wondering is there a makefile option to cleanup the installed binaries and library files. Thanks, Ajay

2 3

how to search previous messages in the forum
by Ajay Sabat 17 Oct '12

17 Oct '12

Hi, Many times I get doubts and questions about Kamailio that I want to search in the SR-Users forum before posting a new message, as it might have already been answered in the past. Is there a easier way to do that than looking up the subject lines in the monthly archive? Thanks, Ajay

2 1

Errors issued by Kamailio
by Sangeeta Shah 17 Oct '12

17 Oct '12

Hi, I am seeing the following errors in syslog: Oct 8 16:21:54 RCS-Presence /usr/local/sbin/kamailio[6723]: ERROR: presence_xml [notify_body.c:635]: while extracting tuple node Oct 8 16:21:54 RCS-Presence /usr/local/sbin/kamailio[6723]: ERROR: presence_xml [notify_body.c:75]: while constructing offline body Oct 8 16:21:54 RCS-Presence /usr/local/sbin/kamailio[6723]: ERROR: presence [presentity.c:1220]: preparing body Oct 8 16:21:54 RCS-Presence /usr/local/sbin/kamailio[6723]: ERROR: presence [publish.c:171]: Marking presentity Oct 8 16:21:55 RCS-Presence /usr/local/sbin/kamailio[6734]: INFO: presence [notify.c:1581]: NOTIFY sip:8475551004@10.50.251.12 via sip:8475551004@10.50.0.93:60562;transport=udp on behalf of sip:8475551004@10.50.251.12 for event message-summary Oct 8 16:21:55 RCS-Presence /usr/local/sbin/kamailio[6734]: INFO: presence [notify.c:1581]: NOTIFY sip:8475551004@10.50.251.12 via sip:8475551004@10.50.0.93:60562;transport=udp on behalf of sip:8475551004@10.50.251.12 for event presence.winfo Oct 8 16:21:57 RCS-Presence /usr/local/sbin/kamailio[6734]: INFO: presence [notify.c:1581]: NOTIFY sip:8475551004@10.50.251.12 via sip:8475551004@10.50.0.93:60562;transport=udp on behalf of sip:8475551004@10.50.251.12 for event presence I am using Kamailio 3.3 with Presence and RLS with integrated XCAP. The only significant change i made today is update the presence and RLS config to DB ONLY mode. Thanks, Sangeeta -- Sangeeta Shah

2 9

$fs syntax
by Juha Heinanen 17 Oct '12

17 Oct '12

wiki tells: $fs - reference to the forced socket for message sending (if any) in the form proto:ip:port should it be [proto:]ip:port instead? -- juha

3 7

Kamailio 3.2.4 presence
by Vitaliy Aleksandrov 17 Oct '12

17 Oct '12

Hi all, I'm facing problems with active subscription restoration after sipproxy restart. db_mode is 0 (DB_MEMORY_ONLY) and according to comments in the presence.h kamailio holds subscription in memory and periodically updates to db, but retrieves from db only at startup. And it's true, I see that kamailio tries to update the active_watchers table, but it never inserts new subscriptions there. There are no problems with access to the database since when db_mode is 2 (DB_ONLY) kamailio inserts new records into the active_watchers without any problems. Here are presence module settings: modparam("presence", "db_url", "sqlite:////var/db/presence.db") modparam("presence", "db_mode", 0) modparam("presence", "timeout_rm_subs", 1) modparam("presence", "pres_htable_size", 8) modparam("presence", "subs_htable_size", 8) modparam("presence", "max_expires", 3600) modparam("presence", "db_update_period", 60) modparam("presence", "clean_period", 60) Does anybody have success with db_mode = DB_MEMORY_ONLY and active_watchers db table ?

2 2

is kamilio support RFC5626 now ?
by mike 16 Oct '12

16 Oct '12

dear kamilio developers : i read the kamilio 3.3.0' s new features , it say it support outbound , and i have place to handle reg-id and sip.instance . so , i wonder if the kamilio3.3.0 support RFC5626 now，so that the kamilio can be used as a edge proxy . i ask why because i can't find where kamilio add flow-token and path for the register request. thanks

5 8

IPv6 addresses in dispatcher on IPv4 only proxy
by Alex Hermann 16 Oct '12

16 Oct '12

Hello, I am using the dispatcher module with a database table shared among multiple proxies. Some proxies do both IPv4 and IPv6, others only do IPv4. The problem is when i use an IPv6 address in the dispatcher table. Then the IPv4-only proxies fail to load the table. ERROR: dispatcher [dispatch.c:325]: could not resolve [IPv6 address] WARNING: <core> [mem/f_malloc.c:474]: WARNING:fm_free: free(0) called ERROR: dispatcher [dispatcher.c:321]: could not initiate a connect to the database ERROR: <core> [sr_module.c:889]: init_mod(): Error while initializing module dispatcher (/usr/lib/kamailio/modules_k/dispatcher.so) Is there any setting i can use on the IPv4-only proxies to make it skip the IPv6 records? (apart from listening on an IPv6 address) -- Greetings, Alex Hermann

2 9

Error while running TLS enabled Kamailio proxy
by Kamal Palei 16 Oct '12

16 Oct '12

Dear All While executing TLS enabled Kamailio proxy (3.1.5), getting below error. * 0(15895) INFO: tls [tls_domain.c:227]: TLSs<default>: verify_depth=9 0(15895) ERROR: tls [tls_domain.c:393]: TLSs<default>: Unable to load certificate file './modules/tls/ser-selfsigned.pem' 0(15895) ERROR: tls [tls_domain.c:394]: load_cert:error:02001002:system library:fopen:No such file or directory 0(15895) ERROR: tls [tls_domain.c:394]: load_cert:error:20074002:BIO routines:FILE_CTRL:system lib 0(15895) ERROR: tls [tls_domain.c:394]: load_cert:error:140DC002:SSL routines:SSL_CTX_use_certificate_chain_file:system lib 0(15895) ERROR: <core> [sr_module.c:832]: init_mod_child(): Error while initializing module tls (/usr/local/lib/kamailio/modules/tls.so) 0(15895) ERROR: <core> [main.c:1532]: ERROR: main: error in init_child(PROC_INT) -- exiting 0(15895) : <core> [mem/q_malloc.c:431]: BUG: qm_free: bad pointer (nil) (out of memory block!) - aborting 0(15893) ERROR: <core> [daemonize.c:307]: Main process exited before writing to pipe * I performed below steps to compile and install. 1. I had a working Kamailio-3.1.5 code base without TLS 2. Downloaded latest openssl source code 3. Compiled and generated libssl.a and libcrypto.a 4. Next, I recompiled kamailio proxy, compilation did not go through. I had to modify Makefile in modules/tls module, added path for openssl/bio.h and path for libssl.a 5. Then executed make all clean and make all include_modules=tls , this time compilation succeeded 6. Next I did make install 7. Generated certificates using openssl library that comes by default with ubuntu 10.04 lte 8. Added below entries in kamailio.cfg #!define WITH_TLS #!ifdef WITH_TLS loadmodule "tls.so" modparam("tls", "config", "/usr/local/etc/kamailio/tls.cfg") modparam("tls", "private_key", "/etc/certs/localb2bua.com/key.pem") modparam("tls", "certificate", "/etc/certs/localb2bua.com/cert.pem") modparam("tls", "ca_list", "/etc/certs/demoCA/cert.pem") enable_tls=yes #!endif 9. Then I tried running kamailio proxy It complained not getting tls.so and tls.cfg Copied these files from modules/tls to /usr/local/etc/kamailio/ path 10. Then I was able to run Kamailio proxy and got the errors pasted at top of email (* Unable to load certificate file './modules/tls/ser-selfsigned.pem') * Please guide me what should I do to avoid this error. Thanks a lot. -Kamal * *

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unsubscribe
by Pablo R. Digonzelli 16 Oct '12

16 Oct '12

Ing. Pablo Digonzelli Sofware Solutions IP Soluciones SRL 25 de Mayo 521. Entrepiso A email: pdigonzelli(a)softsargentina.com email: pdigonzelli(a)gmail.com twitter: @pdigonzelli Tel: 0381 4227378 Cel: 0381 155982714

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Re: [SR-Users] Kamailio 3.3.0 CDR logging call "duration"
by Daniel-Constantin Mierla 16 Oct '12

16 Oct '12

Hello, do not reply to private email only when it is about a discussion on the mailing list - such messages will be ignored. Try to store the value in an avp when processing the BYE, after loose_route(), and store that avp in acc. Daniel On 10/15/12 1:13 PM, Nord7 wrote: > Hello, > no I'm not getting error messages. > Problem: > I can not get kamailio record duration dialogue created field "duration". > I make a test call, answer it, wait 5-6 seconds and hang up. > Now I have to get 5-6 seconds duration in the "duration", but the > field is blank. The rest of the field information about the call is > entered correctly. > So the variable $ DLG_lifetime information about the duration of the > call does not get. > But most of all I am wrong somewhere. > > > 2012/10/15 Daniel-Constantin Mierla <miconda(a)gmail.com > <mailto:miconda@gmail.com>> > > Hello, > > do you get any error message or what is the problem you encounter? > > Cheers, > Daniel > > > On 10/15/12 10:14 AM, Nord7 wrote: >> Hello >> >> Kamailio 3.3.0 >> All default "extra fields" created in database. I create in >> Database new field "duration" and trying write in this field >> variable $DLG_lifetime (from dialog module) , but can't do this. >> What i'm doing wrong? >> >> installed modules: ACC, DIALOG >> # ----- acc params ----- >> /* what special events should be accounted ? */ >> modparam("acc", "early_media", 0) >> modparam("acc", "report_ack", 0) >> modparam("acc", "report_cancels", 0) >> modparam("acc", "detect_direction", 0) >> /* account triggers (flags) */ >> modparam("acc", "log_level", 1) >> modparam("acc", "log_flag", FLT_ACC) >> modparam("acc", "log_missed_flag", FLT_ACCMISSED) >> modparam("acc", "log_extra", >> "src_user=$fU;src_domain=$fd;src_ip=$si;" >> "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" >> "duration=$DLG_lifetime") >> modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) >> /* enhanced DB accounting */ >> 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=$si;" /* extra fields >> created in DataBase */ >> "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" >> "duration=$DLG_lifetime") >> modparam("acc", "cdr_enable", 1) >> modparam("acc", "cdr_start_on_confirmed", 0) >> modparam("acc", "cdr_facility", "LOG_DAEMON") >> >> # ----- dialog params ----- >> modparam("dialog", "enable_stats", 1) >> modparam("dialog", "hash_size", 4096) >> modparam("dialog", "rr_param", "did") >> modparam("dialog", "dlg_flag", DLG_FLAG) /* DLG_FLAG=4 */ >> modparam("dialog", "default_timeout", 21600) >> modparam("dialog", "timeout_avp", "$avp(i:10)") >> modparam("dialog", "dlg_match_mode", 0) >> modparam("dialog", "detect_spirals", 1) >> modparam("dialog", "db_url", DBURL) >> modparam("dialog", "db_mode", 1) >> modparam("dialog", "db_fetch_rows", 500) >> >> >> ########Routing Logic######## >> >> route[RELAY] { >> >> # enable additional event routes for forwarded requests >> # - serial forking, RTP relaying handling, a.s.o. >> if (is_method("INVITE|SUBSCRIBE")) { >> t_on_branch("MANAGE_BRANCH"); >> t_on_reply("MANAGE_REPLY"); >> } >> if (is_method("INVITE")) { >> t_on_failure("MANAGE_FAILURE"); >> } >> setflag(DLG_FLAG); # flag 4 duration >> setflag(FLT_ACC); # do accounting ... >> setflag(FLT_ACCFAILED); # ... even if the transaction fails >> if (!t_relay()) { >> sl_reply_error(); >> } >> exit; >> } >> >> >> >> >> >> >> >> _______________________________________________ >> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list >> sr-users(a)lists.sip-router.org <mailto:sr-users@lists.sip-router.org> >> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > -- > Daniel-Constantin Mierla -http://www.asipto.com > http://twitter.com/#!/miconda <http://twitter.com/#%21/miconda> -http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 -http://asipto.com/u/kat > Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 -http://asipto.com/u/katu > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

2 1

segfault in siptrace module using setflag?
by samuel 16 Oct '12

16 Oct '12

Hi folks, I was making some tests with kamailio (both 3.3.0 and 3.3.1 tgz) with siptrace module sending "homer" packets to a remote central agent where the data is stored and presented to the user. The related config is: # ----- siptrace params from HOMER ----- modparam("siptrace", "duplicate_uri", "sip:A.B.C.D:9060") modparam("siptrace", "trace_to_database", 0) modparam("siptrace", "trace_flag", 22) modparam("siptrace", "trace_on", 1) #HEPv2 == timestamp will be included to HEP header modparam("siptrace", "hep_version", 2) modparam("siptrace", "hep_mode_on", 1) modparam("siptrace", "hep_capture_id", 1) basically for all incoming messages, a route is called with the following actions: route[HOMER] { setflag(22); sip_trace(); } I've been making some tests and if I disable the setflag action, there is no problem (well, no transaction messages are sent to the central agent), but as soon as I activated setflag, kamailio crashes with the following error: <core> [pass_fd.c:293]: ERROR: receive_fd: EOF on 28 There's no concrete point in the config file where it fails so I think the problem is more tm-internals related. I can provide further information in case you need it. Just for adding information, and without any intention to start a flame-war, with opensips 1.8.0-tls the server does not crash... Thanks in advance! Samuel.

3 14

Supporting TLS and DTLS in RTP Proxy
by Kamal Palei 16 Oct '12

16 Oct '12

Hi All I am planning to enhance RTP proxy to support TLS and DTLS. We have some requirements where we need to send RTP packets either over TLS or over DTLS. I really do not know where to start :-) Few things that I have thought over is 1. I will take opnssl code base and integrate with rtpproxy code. 2. Will refer kamailio proxy code how tls connection framework is there and try to copy the code from kamailio code base to rtp proxy code base, OR is there a better alternative here. If somebody has a better idea, kindly share. Thanks a lot. -kamal

5 6

Guidance on using Kamailio as Presence Server and Scalability issues for millions of subsribers
by Thomas Hoffman 16 Oct '12

16 Oct '12

We're trying to determine how best to design/architect a solution where we use Kamailio as a Presence/XCAP server and be scalable to support millions of subscribers. I see the Dispatcher module, but it appears that requires having dedicated instances of the server running just in Dispatcher mode - then you can load balance across the actual instances of the presence servers. It's not clear if a server can be configured to work as a dispatcher and a presence server at the same time. Does anyone have any high-level guidance how to approach this problem? Thank you, Tom Hoffman ______________________________________________________________________________________________________________________________ This e-mail contains PRIVILEGED AND CONFIDENTIAL INFORMATION intended solely for the use of the addressee(s). If you are not the intended recipient, please notify so to the sender by e-mail and delete the original message. In such cases, please notify us immediately at sanchar-sadhan(a)infinite.com . Further, you are not to copy, disclose, or distribute this e-mail or its contents to any unauthorized person(s) .Any such actions are considered unlawful. This e-mail may contain viruses. Infinite has taken every reasonable precaution to minimize this risk, but is not liable for any damage you may sustain as a result of any virus in this e-mail. You should carry out your own virus checks before opening the e-mail or attachments. Infinite reserves the right to monitor and review the content of all messages sent to or from this e-mail address. Messages sent to or from this e-mail address may be stored on the Infinite e-mail system. ***INFINITE******** End of Disclaimer********INFINITE********

2 1

ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1)
by MARINA SERRANO MONTES 16 Oct '12

16 Oct '12

Hi, My kamailio installation usually works right, but sometimes, it crashes and it shows the following error: "ERROR: ctl [ctl.c:379]: ERROR: ctl: could not delete unix socket /tmp/kamailio_ctl: Operation not permitted (1) " Do you know why it happens randomly or what's the cause ? Many thanks. Br, Marina ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n de correo electr?nico en el enlace situado m?s abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at: http://www.tid.es/ES/PAGINAS/disclaimer.aspx

2 1

preparing kamailio 3.3.2
by Daniel-Constantin Mierla 16 Oct '12

16 Oct '12

Hello, I am considering packaging Kamailio v3.3.2 at the beginning of next week (Monday or Tuesday). If any of the devs have backports to be done, do them before the end of this week. Also, if someone spotted some fixes in the master that were not backported, write to the lists so we can care of them. Thanks, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 2

Delay message
by pars3c 15 Oct '12

15 Oct '12

Hi, how can i delay sending of a message ? A ----> proxy (after x seconds) ---> B If I use the sleep function of the cfgutils module,I block all the calls for the sleep time or the single one? Thanks

3 4

Error on compiling Kamailio with TLS support in Ubuntu 10.04LTE
by Kamal Palei 15 Oct '12

15 Oct '12

Hi All I am trying to compile Kamailio - 3.1.5 in Ubuntu 10.04LTE . Tried below comands one by one make -C modules/tls make modules modules=modules/tls make all include_modules=tls But always getting the error *In file included from tls_bio.c:31: tls_bio.h:32:25: warning: openssl/bio.h: No such file or directory * It looks that openssl is not installed in machine. I tried installing openssl as belowB sudo apt-get install libssl-dev But it fails always. The error it gets is *root@B2BUA:/usr/local/src/kamailio-3.1.5# sudo apt-get install libssl-dev sudo: unable to resolve host B2BUA Reading package lists... Done Building dependency tree Reading state information... Done Some packages could not be installed. This may mean that you have requested an impossible situation or if you are using the unstable distribution that some required packages have not yet been created or been moved out of Incoming. The following information may help to resolve the situation: The following packages have unmet dependencies: libssl-dev: Depends: libssl0.9.8 (= 0.9.8k-7ubuntu8) but 0.9.8k-7ubuntu8.8 is to be installed E: Broken packages* * root@B2BUA:/usr/local/src/kamailio-3.1.5# * Has anybody faced this issue previously. Whats the workaround here, if somebody knows please share. Thanks Kamal

2 1

Re: [SR-Users] R: Delay message
by Alex Balashov 15 Oct '12

15 Oct '12

You can't, exactly. But you can t_continue() inside an rtimer route that fires at a given interval. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/pars3c <pars3c(a)gmail.com> wrote:With t_suspend of tmx module, how can I set seconds to sleep? -----Messaggio originale----- Da: sr-users-bounces(a)lists.sip-router.org [mailto:sr-users-bounces@lists.sip-router.org] Per conto di Alex Balashov Inviato: lunedì 15 ottobre 2012 15:52 A: sr-users(a)lists.sip-router.org Oggetto: Re: [SR-Users] Delay message It sounds like you're after the recent asynchronous TM innovations: http://www.kamailio.org/docs/modules/3.3.x/modules_k/tmx.html#id2535968 http://www.kamailio.org/docs/modules/3.3.x/modules_k/tmx.html#id2525882 On 10/15/2012 09:49 AM, pars3c wrote: > Hi, > > how can i delay sending of a message ? > > A ----> proxy (after x seconds) ---> B > > If I use the sleep function of the cfgutils module,I block all the > calls for the sleep time or the single one? > > Thanks > > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing > list sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users ----- Nessun virus nel messaggio. Controllato da AVG - www.avg.com Versione: 2012.0.2221 / Database dei virus: 2441/5332 - Data di rilascio: 14/10/2012 _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

1 0

Kamailio 3.3.0 CDR logging call "duration"
by Nord7 15 Oct '12

15 Oct '12

Hello Kamailio 3.3.0 All default "extra fields" created in database. I create in Database new field "duration" and trying write in this field variable $DLG_lifetime (from dialog module) , but can't do this. What i'm doing wrong? installed modules: ACC, DIALOG # ----- acc params ----- /* what special events should be accounted ? */ modparam("acc", "early_media", 0) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "detect_direction", 0) /* account triggers (flags) */ modparam("acc", "log_level", 1) modparam("acc", "log_flag", FLT_ACC) modparam("acc", "log_missed_flag", FLT_ACCMISSED) modparam("acc", "log_extra", "src_user=$fU;src_domain=$fd;src_ip=$si;" "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" "duration=$DLG_lifetime") modparam("acc", "failed_transaction_flag", FLT_ACCFAILED) /* enhanced DB accounting */ 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=$si;" /* extra fields created in DataBase */ "dst_ouser=$tU;dst_user=$rU;dst_domain=$rd;" "duration=$DLG_lifetime") modparam("acc", "cdr_enable", 1) modparam("acc", "cdr_start_on_confirmed", 0) modparam("acc", "cdr_facility", "LOG_DAEMON") # ----- dialog params ----- modparam("dialog", "enable_stats", 1) modparam("dialog", "hash_size", 4096) modparam("dialog", "rr_param", "did") modparam("dialog", "dlg_flag", DLG_FLAG) /* DLG_FLAG=4 */ modparam("dialog", "default_timeout", 21600) modparam("dialog", "timeout_avp", "$avp(i:10)") modparam("dialog", "dlg_match_mode", 0) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_url", DBURL) modparam("dialog", "db_mode", 1) modparam("dialog", "db_fetch_rows", 500) ########Routing Logic######## route[RELAY] { # enable additional event routes for forwarded requests # - serial forking, RTP relaying handling, a.s.o. if (is_method("INVITE|SUBSCRIBE")) { t_on_branch("MANAGE_BRANCH"); t_on_reply("MANAGE_REPLY"); } if (is_method("INVITE")) { t_on_failure("MANAGE_FAILURE"); } setflag(DLG_FLAG); # flag 4 duration setflag(FLT_ACC); # do accounting ... setflag(FLT_ACCFAILED); # ... even if the transaction fails if (!t_relay()) { sl_reply_error(); } exit; }

2 1

DTLS support in Kamailio
by Kamal Palei 15 Oct '12

15 Oct '12

Hi List Is DTLS supported in Kamailio? If not, can somebody give me initial idea/pointer how can we support it with minimal code change. Best Regards Kamal

2 1

Re: [SR-Users] Get sip clients to work with sip:user@domain and ldap
by Christophe ROY 15 Oct '12

15 Oct '12

Hello I have created a SRV record for sip.domain.tld pointing to our kamailio server, the domain by itself resolves to our DNS IP. Regarding the sockets: root@kamailio1:~# netstat -lnp | grep kamailio tcp 0 0 192.168.14.25:5060 0.0.0.0:* LISTEN 4752/kamailio unix 2 [ ACC ] STREAM LISTENING 17100 4745/kamailio /tmp/kamailio_ctl Thanks 2012/10/11 Daniel-Constantin Mierla <miconda(a)gmail.com> > Hello, > > I recommend using 3.3.x if you start a new deployment. > > Is the domain resolvable by DNS? What are the sockets kamailio is > listening on? > > Cheers, > Daniel > > On 10/10/12 5:21 PM, Christophe ROY wrote: > > Hello > > I'm quite new to SIP and TOIP so I may not understand the concepts > correctly. > I'm supposed to make work SIP clients (for now, linphone on android and > windows) with a Kamailio 3.2 server and ldap authentication. > > Ldap authentication seems to be working fine, as the login/pass is > validated against the Active Directory before registering the SIP client. > However, Linphone seems to register the SIP client as > sip:user@true_ip_address instead of sip:user@domain. > > If I try to make a call to sip:anotheruser@domain it doesn't work, and I > can find in /var/log/syslog the following lines: > > Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm [ut.h:333]: > no corresponding socket for af 2 > Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm > [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding > listening socket) > Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm > [t_fwd.c:1534]: ERROR: t_forward_nonack: failure to add branches > Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: sl > [sl_funcs.c:282]: ERROR: sl_reply_error used: I'm terribly sorry, server > error occurred (7/SL) > > However, if I try to make a direct call to sip:anotheruser@his_true_IP, > it works > > How can I make the registered users to be reachable with user@domain with > Kamailio? > And by the way, is there a way to get a list of registered users? > > Thank you for the help > > Christophe > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing listsr-users@lists.sip-router.orghttp://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > > > -- > Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda > Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat > Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu > > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >

3 4

Wesip Netbeans
by pars3c 15 Oct '12

15 Oct '12

Hi , anyone know if it is possible to use Wesip with Glassfish instead of Tomcat? I'm new in using Glassfish and sip servlet, and I don't know how to start the sip servlet. In tomcat there is the file server.xml, and the connector . How "say" to glassfish to start a listener on sip packet ? If it's not possible to use Glassfish, how i can open a sar project with netbeans e debug it? Thanks to all

1 0

update_subs_rlsdb: NO MATCH
by Sangeeta Shah 12 Oct '12

12 Oct '12

I have Kamailio 3.3 with Presence, RLS and integrated XCAP. I see this information message in the log file: Oct 8 14:34:05 RCS-Presence /usr/local/sbin/kamailio[4530]: INFO: rls [rls_db.c:779]: update_subs_rlsdb: NO MATCH Anyone know why the application is calling that method if it doesn't exist in rls_db.c. I still haven't been able to get presence with rls and integrated xcap working so not sure if this is an issue or no. Currently for both presence and RLS i am using "DB-Only" mode. Any help is appreciated. It seems like when client A adds client B as a contact, the kamalio server is not sending out an auth request to client B. Even though it's registered for presence.winfo events. Thanks, Sangeeta -- Sangeeta Shah

2 6

TLS support in Kamailio
by Kamal Palei 12 Oct '12

12 Oct '12

Hi List I have a specific requiremnt where I want Kamailio proxy need to work as a intermideiate proxy . The endpoints will talk to Kamailio using UDP as transport and Kamailio should interact with main proxy using TLS. Is this scenario possible with Kamailio ? If so, can you give me some pointer how can I acheive it. Thanks Kamal Net clooud systems Bangalore

3 5

Kamailio Group voice Call
by Vijay Thakur 12 Oct '12

12 Oct '12

All Experts, I have setup kamailio 3.1.5 server and it is working fine in my production environment. I want to implement group voice calls (audio conferencing). How can i do this. Kindly guide me. With Regards, Vijay Thakur

3 2

Skype-like service in less than one hour
by శ్రీధర్ 12 Oct '12

12 Oct '12

I set up my server following the tutorial at http://kb.asipto.com/kamailio:skype-like-service-in-less-than-one-hour, with the latest kamailio 3.3, and updated the config file accordingly. After the reboot the sip server is not running here are the logs. When I do 'sudo /etc/init.d/kamailio start' the server starts, but has problems communicating with the rtpproxy, the log for the same is also below. Please advice if any changes needed to make this working with the latest version Thanks S $ cat /var/log/syslog | grep kamailio Oct 12 01:07:48 sips kamailio: INFO: tls [tls_init.c:377]: tls: init_tls: disabling compression... Oct 12 01:07:48 sips kamailio: INFO: <core> [tcp_main.c:4829]: init_tcp: using epoll_lt as the io watch method (auto detected) Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: usrloc [hslot.c:53]: locks array size 512 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: auth [auth_mod.c:329]: auth: qop set, but nonce-count (nc_enabled) support disabled Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: tls [tls_init.c:541]: tls: _init_tls_h: compiled with openssl version "OpenSSL 1.0.1 14 Mar 2012" (0x1000100f), kerberos support: off, compression: on Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: tls [tls_init.c:549]: tls: init_tls_h: installed openssl library version "OpenSSL 1.0.1 14 Mar 2012" (0x1000100f), kerberos support: off, zlib compression: on#012 compiler: cc -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DL_ENDIAN -DTERMIO -g -O2 -fstack-protector --param=ssp-buffer-size=4 -Wformat -Wformat-security -Werror=format-security -D_FORTIFY_SOURCE=2 -Wl,-Bsymbolic-functions -Wl,-z,relro -Wa,--noexecstack -Wall -DOPENSSL_NO_TLS1_2_CLIENT -DOPENSSL_MAX_TLS1_2_CIPHER_LENGTH=50 -DOPENSSL_BN_ASM_PART_WORDS -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_GF2m -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DRMD160_ASM -DAES_ASM -DVPAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: WARNING: tls [tls_init.c:603]: tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 9961472 and 4980736 bytes Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: <core> [cfg/cfg_ctx.c:613]: INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 9961472 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: INFO: <core> [cfg/cfg_ctx.c:613]: INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 4980736 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: db_mysql [km_my_con.c:109]: driver error: Can't connect to local MySQL server through socket '/var/run/mysqld/mysqld.sock' (2) Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: <core> [db.c:320]: could not add connection to the pool Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: usrloc [dlist.c:513]: failed to open database connection Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: registrar [reg_mod.c:472]: failed to register domain Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:623 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:626 Oct 12 01:07:48 sips /usr/sbin/kamailio[794]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/etc/kamailio/kamailio.cfg:628 Oct 12 01:54:55 sips kamailio: INFO: tls [tls_init.c:377]: tls: init_tls: disabling compression... Oct 12 01:54:55 sips kamailio: INFO: <core> [tcp_main.c:4829]: init_tcp: using epoll_lt as the io watch method (auto detected) Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: usrloc [hslot.c:53]: locks array size 512 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: auth [auth_mod.c:329]: auth: qop set, but nonce-count (nc_enabled) support disabled Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_init.c:541]: tls: _init_tls_h: compiled with openssl version "OpenSSL 1.0.1 14 Mar 2012" (0x1000100f), kerberos support: off, compression: on Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_init.c:549]: tls: init_tls_h: installed openssl library version "OpenSSL 1.0.1 14 Mar 2012" (0x1000100f), kerberos support: off, zlib compression: on#012 compiler: cc -fPIC -DOPENSSL_PIC -DZLIB -DOPENSSL_THREADS -D_REENTRANT -DDSO_DLFCN -DHAVE_DLFCN_H -DL_ENDIAN -DTERMIO -g -O2 -fstack-protector --param=ssp-buffer-size=4 -Wformat -Wformat-security -Werror=format-security -D_FORTIFY_SOURCE=2 -Wl,-Bsymbolic-functions -Wl,-z,relro -Wa,--noexecstack -Wall -DOPENSSL_NO_TLS1_2_CLIENT -DOPENSSL_MAX_TLS1_2_CIPHER_LENGTH=50 -DOPENSSL_BN_ASM_PART_WORDS -DOPENSSL_IA32_SSE2 -DOPENSSL_BN_ASM_MONT -DOPENSSL_BN_ASM_GF2m -DSHA1_ASM -DSHA256_ASM -DSHA512_ASM -DMD5_ASM -DRMD160_ASM -DAES_ASM -DVPAES_ASM -DWHIRLPOOL_ASM -DGHASH_ASM Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: tls [tls_init.c:603]: tls: openssl bug #1491 (crash/mem leaks on low memory) workaround enabled (on low memory tls operations will fail preemptively) with free memory thresholds 9961472 and 4980736 bytes Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [cfg/cfg_ctx.c:613]: INFO: cfg_set_now(): tls.low_mem_threshold1 has been changed to 9961472 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [cfg/cfg_ctx.c:613]: INFO: cfg_set_now(): tls.low_mem_threshold2 has been changed to 4980736 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 163840 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [udp_server.c:179]: INFO: udp_init: SO_RCVBUF is initially 163840 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: <core> [udp_server.c:230]: INFO: udp_init: SO_RCVBUF is finally 262142 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:187]: TLSs<default>: tls_method=12 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:199]: TLSs<default>: certificate='/etc/kamailio/kamailio-selfsigned.pem' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:206]: TLSs<default>: ca_list='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:213]: TLSs<default>: crl='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:217]: TLSs<default>: require_certificate=0 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:224]: TLSs<default>: cipher_list='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:231]: TLSs<default>: private_key='/etc/kamailio/kamailio-selfsigned.key' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:235]: TLSs<default>: verify_certificate=0 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:238]: TLSs<default>: verify_depth=9 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:576]: TLSs<default>: No client certificate required and no checks performed Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:187]: TLSc<default>: tls_method=9 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:199]: TLSc<default>: certificate='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:206]: TLSc<default>: ca_list='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:213]: TLSc<default>: crl='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:217]: TLSc<default>: require_certificate=1 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:224]: TLSc<default>: cipher_list='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:231]: TLSc<default>: private_key='(null)' Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:235]: TLSc<default>: verify_certificate=1 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:238]: TLSc<default>: verify_depth=9 Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: INFO: tls [tls_domain.c:561]: TLSc<default>: Server MUST present valid certificate Oct 12 01:54:55 sips /usr/sbin/kamailio[1135]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1134]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1136]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1135]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1134]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1135]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1137]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1136]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1135]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1134]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1136]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1134]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1136]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1133]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1137]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1137]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1138]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1137]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1133]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1133]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1133]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1138]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1138]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1138]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1140]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1141]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1139]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1139]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1139]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1139]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1141]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1140]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1140]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1141]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1140]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1141]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1132]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1132]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1132]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1132]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1130]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1153]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1153]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1153]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1153]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1152]: INFO: ctl [io_listener.c:225]: io_listen_loop: using epoll_lt io watch method (config) Oct 12 01:54:55 sips /usr/sbin/kamailio[1159]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1159]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1159]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1159]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1158]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1158]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1158]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1157]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1158]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1157]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1157]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1157]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1156]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1156]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1156]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1156]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily Oct 12 01:54:55 sips /usr/sbin/kamailio[1155]: ERROR: rtpproxy [rtpproxy.c:1464]: can't connect to RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1155]: ERROR: rtpproxy [rtpproxy.c:1536]: proxy <unix:/var/run/rtpproxy/rtpproxy.sock> does not respond, disable it Oct 12 01:54:55 sips /usr/sbin/kamailio[1155]: WARNING: rtpproxy [rtpproxy.c:1393]: can't get version of the RTP proxy Oct 12 01:54:55 sips /usr/sbin/kamailio[1155]: WARNING: rtpproxy [rtpproxy.c:1430]: support for RTP proxy <unix:/var/run/rtpproxy/rtpproxy.sock> has been disabled temporarily

4 6

log alerting time, progress time connect time
by pars3c 12 Oct '12

12 Oct '12

Hi , i need to register the time when arrive the following messages: invite, alerting, progress , connected and hangup. How can I make it ? it's is possible to store these value in the extra field in the acc module ? Thanks

2 1

TCP error
by Ed Brady 12 Oct '12

12 Oct '12

I am testing connecting an MSRP client to my Kamailio instance, and I keep getting the following error in my syslog: /usr/sbin/kamailio[20055]: ERROR: <core> [tcp_read.c:293]: error reading: Connection reset by peer (104) I believe this must be a misconfiguration on my part, I just don't know where to start. Any ideas what may cause a TCP read error? If this is an uncommon error, I can supply configs to help determine the cause. Thanks, Ed

3 2

Wrong onreply_route is called after serial forking
by Alex Hermann 11 Oct '12

11 Oct '12

Hello, i noticed thet in serial forking, replies to earlier branches arriving after sending out a new branch use the onreply_route of the later branch instead of the onreply_route set before sending the earlier branch.... How to reproduce: 1) set onreply_route to A 2) relay 1st branch 3) 1st branch times out, internal 408 is created 4) tm send CANCEL to 1st branch 5) in failure route, onreply_route is set to B 6) relay 2nd branch 7) 1st branch responds with 487, and goes into reply_route B instead of A I think each branch should take the reply_route which was set before it got relayed and not pick up later changes meant for other branches. How would i fix this? -- Greetings, Alex Hermann

5 7

Question about how the RLS module handle subscribe
by Sangeeta Shah 11 Oct '12

11 Oct '12

Hello All, I am trying to understand how the RLS module handles a subscribe request. My client is sending the following: Subscribe, event = reg, no Allowed header - hence it is getting handled by the presence module (entry make its to active_watchers table) Subscribe, event - presence.winfo, no Allowed header - hence it's getting handled by the presence module (entry makes it to active_watchers table) Publish - handled by presence module (makes it to presentity table) Subscribe, event = presence, Allowed = eventlist - hence it's handled by the rls module. RLS module is having trouble sending a notify per this data in the syslog. Also I don't see any entries in the watcher table or rls_watchers or rls_presentity table. Errors: Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: presence [hash.c:342]: record not found in hash table Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: ERROR: rls [notify.c:387]: updating in hash table Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: ERROR: rls [notify.c:273]: in function agg_body_sendn_update Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:81]: freeing 5 columns Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x7fb2a8daa220 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x7fb2a8daa598 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x7fb2a8da9f90 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x7fb2a8da9b48 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[4] at 0x7fb2a8da9b28 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x7fb2a8daa608 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x7fb2a8daa570 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x7fb2a8da9410 Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: ERROR: rls [subscribe.c:732]: failed sending full state notify Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: ERROR: rls [subscribe.c:781]: occured in rls_handle_subscribe Oct 11 11:29:26 RCS-Presence /usr/local/sbin/kamailio[23085]: DEBUG: <script>: ========= RLS Handle subscriber returned with ret code: -1 === What's it trying to do here? Any help is appreciated. Thanks, Sangeeta -- Sangeeta Shah

1 0

Kamailio presence with RLS
by Sangeeta Shah 11 Oct '12

11 Oct '12

Hello Everyone, Has anyone had any luck getting RLS working with presence. I am seeing the following errors in my log file: ct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: presence [hash.c:342]: record not found in hash table Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: tm [t_reply.c:1543]: DEBUG: cleanup_uac_timers: RETR/FR timers reset Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [usr_avp.c:644]: DEBUG:destroy_avp_list: destroying list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [xavp.c:365]: destroying xavp list (nil) Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2020]: DEBUG: <core> [receive.c:293]: receive_msg: cleaning up Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: ERROR: rls [notify.c:385]: updating in hash table Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: ERROR: rls [notify.c:271]: in function agg_body_sendn_update Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:81]: freeing 5 columns Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[0] at 0x7fbf415c4178 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[1] at 0x7fbf415c48a0 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[2] at 0x7fbf415c38a8 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[3] at 0x7fbf415c3bc8 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:85]: freeing RES_NAMES[4] at 0x7fbf415c4880 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:94]: freeing result names at 0x7fbf415c3c28 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:99]: freeing result types at 0x7fbf415c3648 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <core> [db_res.c:136]: freeing result set at 0x7fbf415c3940 Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: ERROR: rls [subscribe.c:732]: failed sending full state notify Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: ERROR: rls [subscribe.c:781]: occured in rls_handle_subscribe Oct 10 17:20:22 RCS-Presence /usr/local/sbin/kamailio[2017]: DEBUG: <script>: ========= RLS Handle subscriber returned with ret code: -1 === I think this is my first problem. There is an error later in the log file as well, but it seems Apparently from what i understand we are trying to update the entry in the presence_shtable and we cannot find it and hence we are failing to send the full state notify. Any help is appreciated. If further information is required please let me know. I can try to print out more data in the update_presence_shtable function. I have another htread with a lot of log info but for some reason the thread went cold and I just need to keep moving on getting this to work. Anyone familiar with this please let me know what the likely cause of this issue might be. Thanks, Sangeeta -- Sangeeta Shah

1 0

How can I directly access the domain of additional branches??
by Stephen Dodge (Bistech) 11 Oct '12

11 Oct '12

Hello, Could someone advise if it is possible to directly access the domain of additional branches as $rd references the domain of the main URI only. If it's not possible does anyone have any tips on how I can achieve this. Many Thanks. Steve. ________________________________ Information in this message, including any attachments, is confidential to the person to whom it is addressed and may be legally privileged. If you are not the intended recipient please notify the sender and delete the message from your system. Please note that Bistech Group plc, Bistech plc, Bisnet Limited and the sender do not accept any responsibility for viruses. It is your responsibility to check the e-mail and any attachments for viruses. Calls may be monitored and recorded.

4 5

Kamailio direct interconnectivity with PRI
by SamyGo 11 Oct '12

11 Oct '12

Hello, I've a scenario in which I've to deploy a couple Sangoma PRI cards with kamailio. What I wish is that I've some drivers for this purpose and so I don't ned to install FreeSWITCH or Asterisk in between the PRIs and Kamailio. Kindly give any feedback on what are the possibilities and options. Thanks, Sammy

4 7

Get sip clients to work with sip:user@domain and ldap
by Christophe ROY 11 Oct '12

11 Oct '12

Hello I'm quite new to SIP and TOIP so I may not understand the concepts correctly. I'm supposed to make work SIP clients (for now, linphone on android and windows) with a Kamailio 3.2 server and ldap authentication. Ldap authentication seems to be working fine, as the login/pass is validated against the Active Directory before registering the SIP client. However, Linphone seems to register the SIP client as sip:user@true_ip_address instead of sip:user@domain. If I try to make a call to sip:anotheruser@domain it doesn't work, and I can find in /var/log/syslog the following lines: Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm [ut.h:333]: no corresponding socket for af 2 Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm [t_fwd.c:424]: ERROR: can't fwd to af 2, proto 1 (no corresponding listening socket) Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: tm [t_fwd.c:1534]: ERROR: t_forward_nonack: failure to add branches Oct 10 17:19:21 kamailio1 /usr/sbin/kamailio[1790]: ERROR: sl [sl_funcs.c:282]: ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) However, if I try to make a direct call to sip:anotheruser@his_true_IP, it works How can I make the registered users to be reachable with user@domain with Kamailio? And by the way, is there a way to get a list of registered users? Thank you for the help Christophe

2 1

kamailio 3.3.1 - uac - remote register - init delay possible?
by René Bormann 11 Oct '12

11 Oct '12

Hello @all, want to use the remote register function of the UAC module. We need to register 42 accounts on one openSER server. At kamailio startup those 42 remote register requests will be startet at the same time. This leads to a kind of FlodAttact on the openSER server of our provider, which will block us :-( We would like to delay each initial remote register by 1 second. We searched the UAC module and taught that at some place the uac.so module has to load the uacreg table and do some for loop to start the remote register process. We found function child_init() in uac.c where the function uac_reg_load_db() is called followed uac_reg_timer(). Our change to uac_reg_timer() is the sleep(1). After compiling it seems to us - that this function is not called. Is there anybody how has a tip for us? Thanks, René /** * */ void uac_reg_timer(unsigned int ticks, unsigned int initial_reg_call) { LM_ERR("----------------------- uac_reg_timer() -- enter \n"); <------>int i; <------>reg_item_t *it = NULL; <------>time_t tn; <------>tn = time(NULL); <------>for(i=0; i<_reg_htable->htsize; i++) <------>{ <------><------>/* free entries */ <------><------>it = _reg_htable->entries[i].byuuid; <------><------>while(it) <------><------>{ LM_ERR("----------------------- uac_reg_timer() -- 1 \n"); <------><------><------>if (initial_reg_call == 1) <------><------><------>{ LM_ERR("----------------------- uac_reg_timer() -- 2 \n"); <------><------><------><------>sleep(1); <------><------><------>} <------><------><------>uac_reg_update(it->r, tn); <------><------><------>it = it->next; <------><------>} <------>} }

3 8

MSRP Config
by Ed Brady 10 Oct '12

10 Oct '12

Hi, I am working on an MSRP project with Kamailio. I have Kamailio installed and configured for SIP with several test accounts created, but am unable to send MSRP messages so far. I am using Blink as a test client, though the final intentions of the project will be to use iDoubs and IMSDroid for client connections. We are having these developed to our needs right now though, so I must use another client to begin testing. I have been unable to find good documentation on the initial configuration of MSRP in Kamailio. I have seen the module documentation, but am unclear how to utilize the information there to configure Kamailio properly. Is there any documentation on getting started with a basic MSRP relay setup so that two clients can chat and send files? I am not new to SIP, but this is my first project with MSRP. Thanks, Ed

4 10

OpenSER/Kamailio Support? - Forwarding SIP INVITE Packets
by Eddie Elric 10 Oct '12

10 Oct '12

Hello fellow OpenSER/Kamailio users, I hope this is the right method of contact - I was unable to find a forum on which to post any queries. In essence, I've set up a Kamailio server of the latest version - logging on to clients works fine, etc. The idea of my setting up was for forwarding SIP messages to our test server, that's where I got stuck. The following link explains everything on Kamailio it seems - http://www.kamailio.org/wiki/cookbooks/3.3.x/core#forward So, I've added to the kamailio.cfg the following - forward("10.0.0.10","5060"); But you see, I'm not sure where I should have added it. I've tried adding to a variety of places, such as - route[SIPOUT] and onreply_route[MANAGE_REPLY] - but none are working. Cheers guys/girls. Best, Eddie

1 1

Kamailio/RTP Proxy issue
by Kamal Palei 10 Oct '12

10 Oct '12

Hi All Facing a strange issue in Kamailio / RTP proxy setup. Using 1. Kamailio - 3.1.5 2. RTP proxy 1.1 I have pne sip client, lets say it is 'A'. 'A' sent INVITE to Kamailio proxy. Kamailio proxy modified the IP address present in SDP to RTP proxy IP address, and sent to main proxy. Main proxy did respond back with 200 OK to INVITE. In 200 OK SDP, Main proxy put the IP address of Media server (media server ip address is different than main proxy server ip address). After the call establishment, it is expected that RTP proxy should send media packets to media server, instead it is sending to proxy server. Pleae do let know, how to put a work around to avoid this problem. Thanks Kamal NECS, Bangalore

2 2

TCP replace alias can cause core
by Jijo 10 Oct '12

10 Oct '12

We found a problem regarding TCP connection alias in the following code at tcp_main.c: int tcpconn_finish_connect( struct tcp_connection* c, union sockaddr_union* from) { : : /* remove all the aliases except the first one and re-add them * (there shouldn't be more then the 3 default aliases at this * stage) */ for (r=1; r<c->aliases; r++){ a=&c->con_aliases[r]; tcpconn_listrm(tcpconn_aliases_hash[a->hash], a, next, prev); } c->aliases=1; As TCP_ALIAS_REPLACE flag is set for the default TCP options value, in the function _tcpconn_add_alias_unsafe() a TCP connection alias can be moved from connection A to connection B based on the TCP alias hash. In this case, the number of aliases is incremented in the connection A, and decremented from connection B. However, in the connection B the number of aliases can reach zero (no alias). And the code above can be executed for connection B setting the number of aliases to 1 unconditionally. When this case happens, the connection B keeps an invalid alias (already excluded from connection B by tcpconn_add_alias_unsafe() function called from connection A). When the connection A is released, the aliases are also released, and this memory area can be filled with different data. As connection B has references to an invalid alias it can try to access invalid areas, and can crash Kamailio. This access happens, for example, when another alias is added to connection B. To fix it we include a check before the code: if (c->aliases>0) { for (r=1; r<c->aliases; r++){ a=&c->con_aliases[r]; tcpconn_listrm(tcpconn_aliases_hash[a->hash], a, next, prev); memset(a,0xbb,sizeof(struct tcp_conn_alias)); } c->aliases=1; } Please let us know if any comments. Thanks Jijo

2 1

[snmpstats] openserSIPRegRejectedRegistrations not being updated
by Anatoly Ivanov 10 Oct '12

10 Oct '12

Hi, I am observing weird behavior of snmpstats.so - I see openserSIPRegAcceptedRegistrations being updated, but openserSIPRegRejectedRegistrations stays at zero. Could somebody please explain under what conditions this counter is being incremented? I am using auth_radius, so I suspect I simply do not reject failed registartions the 'proper' way, thus causing the counter to stay at 0. Thank you, Anatoly.

2 3

how to use fix_nated_sdp("2") with multiple media types
by MARINA SERRANO MONTES 10 Oct '12

10 Oct '12

Hi, I have a doubt about how to use "fix_nated_sdp("2")" function when we have an scenario with multiple media types (audio and video) and the param "c" in sdp for each media type has a different IP. We are using a SCOPIA 400 MCU to multi-party in a call, and it will set up different IP to send RTP depend on media type. I have tested this function, but it rewrites always all the "c" parameter with the same value. Is there any option to use the function or to rewrite "c" value for each media type independently? Many thanks. Br, Marina ________________________________ Este mensaje se dirige exclusivamente a su destinatario. Puede consultar nuestra pol?tica de env?o y recepci?n de correo electr?nico en el enlace situado m?s abajo. This message is intended exclusively for its addressee. We only send and receive email on the basis of the terms set out at: http://www.tid.es/ES/PAGINAS/disclaimer.aspx

3 11

Nginx dispatching of websocket requests?
by Juha Heinanen 09 Oct '12

09 Oct '12

just to complete the thread, nginx does not currently support proxying of requests that use websocket transport. it is in the roadmap, but no date is set. when nginx receives request over websocket it converts 'Upgrade: Connection: Upgrade' header to 'Connection: close' header and removes Upgrade header. -- juha

1 0

moving all to kamailio namespace
by Daniel-Constantin Mierla 09 Oct '12

09 Oct '12

Hello, some bits were left with the old project name OpenSER, not to disturb too much at that time and see where everything goes. Other voices expressed same idea in the past, now everything is stable and development goes smooth, so it is good to make a decision because openser is too way back and new comers could be confused. In my list is renaming from openser to kamailio next attributes: - default database name (exiting configs should not be impacted that much as most of them have database provided by module parameter) - default database access usernames and passwords, respectively openser, openserro and openserrw should become kamailio, kamailioro and kamailiorw - perl module API is packed as OpenSER - this could affect some people using it, but a search and replace should do it - snmpstats mibs - probably it is going to have some impact as well Eventually we can start doing it gradually, one or two per release, but is probably better to do all at once, so people will do one upgrade with next major release. Another thing is sercmd, it is in conflict for packaging as both ser and kamailio install it. It means that one cannot install both kamailio and ser at same time from packages. One option is to install it as kamcmd. Because these changes affect users and devels, I sent the message to both mailing lists. Opinions? Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

6 7

storing and using route-set in dialog module
by Klaus Darilion 09 Oct '12

09 Oct '12

Hi! Are there any means to store the route set of a dialog (on dialog creation) and use the stored route set when forwarding in-dialog requests instead of using the requests' route set? thanks Klaus

1 0

Force to use mediaproxy/rtpproxy
by Mino Haluz 09 Oct '12

09 Oct '12

Hi, I am doing some performance tests and I cannot force SIP INVITEs to use mediaproxy within LAN, without any NAT. I'm using engage_media_proxy or rtpproxy_manage but phones are still exchanging voice data between them, not through the proxy. Is there some trick how to forcibly tell kamailio to use media/rtp proxy ? Maybe kamailio finds out that there is no nat and therefore will not use any nat traversal mechanism. Mino

1 1

get $sht array of values
by Uri Shacked 09 Oct '12

09 Oct '12

Hi, If i have two keys on a HTABLE that are the same. i undestand i need to set the key type to 1 (array). i use the $shtcn and get 3 (size is an entry as well). so far, so good. But, now i am trying to get all the values from the table to an AVP oto say it in another way - i need all values because i parse it afterwards. So, what is the syntax? using $sht() didn't work..... Uri

2 1

kamailio cores on corrupted route header
by Jijo 09 Oct '12

09 Oct '12

Hello, kamailio cores when receives a corrupted route header. For example, this was causing the core. Route: sip:10.236.236.100;transport=tcp;r2=on;lr;ftag=1348218287134-Test-553188;osb-tag=NM;nat=yes;twan=yes?[=& [=<sip:10.236.236.100;transport=tcp;r2=on;lr;ftag=1348218287134-Test-553188;osb-tag=NM;nat=yes;twan=yes?[=&%20[=> I found the problem, the pointer was not initializing to null after freeing it. Please apply this fix in the next version. Here is the diff with the original(3.2.2) and changed version. PGA:/mnt/o/kamailio-3.2.2/parser # diff -u parse_param.c.orig parse_param.c --- parse_param.c.orig 2012-10-09 09:42:58.372003500 -0300 +++ parse_param.c 2012-10-09 21:34:14.556367900 -0300 @@ -545,6 +545,7 @@ error: if (t) pkg_free(t); free_params(*_p); + *_p = 0; return -2; ok: Thanks Jijo

1 0

Kamailio XMPP support
by Vijay Thakur 09 Oct '12

09 Oct '12

Dear All, I have set up my kamailio server successfully and it is working very fine. i have followed the below link for setup: http://kb.asipto.com/kamailio:presence:k31-made-simple. Now i want to enable xmpp module. At Kamailio wiki page, i have found that there can be two way to implement xmpp module in kamilio. And they are (1) Inside the server (2) Remoter jabber server. But there is no specific "how to" to enable this module. How can i compile and enable this module in kamailio. And after this what will be the setting of my kamailio.cfg. Currently jabberd2 is not available as directed in wiki. Can i use openfire with kamailio for xmpp? How ? Kindly point me any direction. With Warm Wishes, Vijay Thakur information intended for a specific individual and purpose, and is protected by law. If you are not the intended recipient, you should delete this message and are hereby notified that any disclosure, copying, or distribution of this message, or the taking of any action based on it, is strictly prohibited. -----------------------------------------------------------------------------------------------------------------

1 0

Kamailio XMPP Module Setup
by Vijay Thakur 09 Oct '12

09 Oct '12

Dear All, I have set up my kamailio server successfully and it is working very fine. i have followed the below link for setup: http://kb.asipto.com/kamailio:presence:k31-made-simple. Now i want to enable xmpp module. At Kamailio wiki page, i have found that there can be two way to implement xmpp module in kamilio. And they are (1) Inside the server (2) Remoter jabber server. But there is no specific "how to" to enable this module. How can i compile and enable this module in kamailio. And after this what will be the setting of my kamailio.cfg. Kindly point me any direction. With Warm Wishes, Vijay Thakur

1 0

Kamailio FAQ Week
by Daniel-Constantin Mierla 09 Oct '12

09 Oct '12

Hello, I'm looking at getting richer content for the FAQ wiki page: * http://www.kamailio.org/wiki/tutorials/faq/main Probably we can get it good enough with a bit of mobilization for one week. If anyone is having some questions fitting the purpose of the FAQ, just add them to the page or send via email. If you don't have the right answer, just leave it as TBA in the wiki, via email is fine to send just the question. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Nginx dispatching of websocket requests?
by Juha Heinanen 09 Oct '12

09 Oct '12

peter suggested using nginx for dispatching xcap requests and that works fine. has someone tried using nginx to dispatch also sip over websocket requests? is there some drawbacks? -- juha

2 1

Kamailio issues errors
by Sangeeta Shah 08 Oct '12

08 Oct '12

I have Kamailio 3.3 with Presence, RLS and integrated XCAP. Now I am noticing the errors in the attached syslog capture. The only thing I have changes is the DB mode to DB Only. Thanks, Sangeeta -- Sangeeta Shah

1 0

implementing call-quota with dialog module
by Aft nix 08 Oct '12

08 Oct '12

Hi, I'm trying to implement a "callquota" based on destination domains. The idea is following: every destation domain has a limit of, say 50 calls. As the sip clients are registered with the "destination domains", i've extracted the domain using "$fd" because it results as same value as destination domain. Problem is, i wanted to have separate quota for each domains. But it seems, that the quota is reached "as total" not as per domain basis. Another thing is, the number of calls reported by it is also unusually large than actual calls. This is implemenated as bellow: #!ifdef WITH_CALL_LIMIT modparam("dialog","enable_stats",1) modparam("dialog","dlg_flag", DLG_FLAG) modparam("dialog","hash_size", 4096) modparam("dialog", "profiles_with_value","callquota") modparam("dialog", "default_timeout", 300) modparam("dialog", "dlg_match_mode", 2) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_mode", 0) #!endif #!ifdef WITH_CALL_LIMIT if (is_method("INVITE")) { dlg_manage(); $var(100) = 0; get_profile_size("callquota", "$fd", "$var(100)"); if ($var(100) >= 50 ) { xdbg("DEBUG: Simultaneous calls limit reached"); sl_send_reply("503","Simultaneous calls limit reached"); exit; } set_dlg_profile("callquota","$fd"); if (get_profile_size("callquota","$fd", "$var(100)")) { xdbg("DEBUG: there are $var(100) total calls for $fd"); } } #!endif Thanks in advance. -- -aft

2 2

Newbie question about RLS/XCAP/batch'd NOTIFYs
by Dmytro Bogovych 06 Oct '12

06 Oct '12

Greetings. I try to make iOS voip application with presence information. iOS does not let too frequent network updates from server side when runs in background mode. So I cannot rely on plain PUBLISH / SUBSCRIBE scheme - it leads to restart of application by iOS in many cases. Please see this post for details: https://devforums.apple.com/message/565667#565667 The question. Can Kamailio provide way to reduce the number of NOTIFYs? I see it as 1) iOS application creates & uploads RLS via XCAP 2) iOS application asks server about changes in presence information once per 10 minutes or when application goes to foreground state. The best way is to receive single batch'd NOTIFY answer for all watched peers. Is it possible to implement with kamailio? Thank you :)

2 2

compilation warnings
by Vieri 05 Oct '12

05 Oct '12

Hi, On a Gentoo Linux system I'm getting the following compile-time warnings (kamailio-3.3.1 ebuild): * python_msgobj.c:529:5: warning: dereferencing type-punned pointer will break strict-aliasing rules * ld_con.c:241:4: warning: implicit declaration of function 'ldap_simple_bind_s' * perl.c:340:2: warning: null argument where non-null required (argument 2) * perl.c:340:2: warning: null argument where non-null required (argument 1) Hopefully I can just ignore these messages. Thanks, Vieri

3 3

kamailio 3.3 sipcapture and Homer Capture Agent 4
by Juha Heinanen 05 Oct '12

05 Oct '12

is Homer Capture Agent 4 hep protocol incompatible with Kamalio 3.3 sipcapture module? in captagent.xml i set core_hep.conf version to 2, but looks like kamailio is not able to parse the messages: Oct 5 12:40:18 siika sip-proxy[6779]: INFO: <core> [parser/parse_fline.c:145]: ERROR:parse_first_line: method not followed by SP Oct 5 12:40:18 siika sip-proxy[6779]: INFO: <core> [parser/parse_fline.c:242]: ERROR:parse_first_line: bad message Oct 5 12:40:18 siika sip-proxy[6779]: ERROR: <core> [parser/msg_parser.c:705]: ERROR: parse_msg: message=<#002#020#002#006#023ĳ��bg#012�bg#012#002�nP#037#035#010> Oct 5 12:40:18 siika sip-proxy[6779]: ERROR: <core> [receive.c:146]: ERROR: receive_msg: parse_msg failed in kamailio config i have modparam("sipcapture", "capture_on", 1) modparam("sipcapture", "hep_capture_on", 1) -- juha

2 2

Does Kamailio support ECDHE ciphers for TLS?
by AJ 05 Oct '12

05 Oct '12

Hi, I have Kamailio 3.3.1 working fine with TLS 1.0, running on CentOS 6.3 However, now I want to use some ECDHE ciphersuites, for instance: ECDHE-ECDSA-AES128-SHA I have built an appropriate OpenSSL (1.0.0j) in CentOS to support this, and have generated the appropriate certs: openssl ecparam -out ca-privkey.pem -name secp256r1 -genkey -outform pem openssl req -x509 -new -key ca-privkey.pem -out ca_cert.pem -outform PEM -days 3650 openssl ecparam -out privkey.pem -name secp256r1 -genkey -outform pem openssl req -new -nodes -key privkey.pem -outform pem -out cert_req.pem openssl ca -keyfile ca-privkey.pem -cert ca_cert.pem -in cert_req.pem -out cert.pem However, when I try to connect from my phone I am always getting: Oct 3 20:27:54 /usr/sbin/kamailio[2905]: ERROR: tls [tls_server.c:1190]: TLS accept:error:1408A0C1:SSL routines:SSL3_GET_CLIENT_HELLO:no shared cipher Oct 3 20:27:54 /usr/sbin/kamailio[2905]: ERROR: <core> [tcp_read.c:1127]: ERROR: tcp_read_req: error reading If I use the OpenSSL s_server test-tool, I can connect from my phone just fine with these certs: openssl s_server -accept 8888 -cert cert.pem -key key.pem -pass stdin -CAfile calist.pem -cipher ECDHE-ECDSA-AES128-SHA Does Kamailio support ECDHE ciphersuites? If so, what am I doing wrong? Thanks! AJ

3 2

Kamailio question
by Jostein Aardal 05 Oct '12

05 Oct '12

Hi! I am a student at Oslo and Akershus University college of applied sciences. I'm working on a project where I am supposed to setup a kamailio server. Up until now I have been able to install kamailio and add a user. When I use a sip-client I am able to register and authenticate this user. And I am able to make calls between different users, as long as they both are registered on the same server. But when I try to make a call between users that are registered on different servers/proxies I fail to establish the call. When I look at the sip-messages in wireshark, I see that there is a message "407 proxy authentication required". I thought that enabling "multidomain" in kamailio.cfg would fix this problem, but it doesn't. What I have written in the kamailio.cfg-file is: #!define WITH_MYSQL #!define WITH_AUTH #!define WITH_USRLOCDB #!define WITH_MULTIDOMAIN #!define WITH_ALIASDB #!define WITH_DEBUG yours Jostein Aardal Oslo and Akershus University College of Applied Sciences

3 2

hello i want to active international call
by mark shaw 05 Oct '12

05 Oct '12

hello i want to active international call in my account for important . radioyork2 thedept

2 1

Client info
by Sangeeta Shah 04 Oct '12

04 Oct '12

Hello All, I am trying to use Kamailio 3.3 configured as a presence server and rls server with integrated XCAP. I am trying to test the presence and rls functionality with clients such as Jitsi (doesn't seem to support rls and doesn't let me specify presence rules per contact) and Boghe (doesn't seem to be behaving as expected!). Anyone recommend other clients that work for the functionality I described above. Thanks, Sangeeta -- Sangeeta Shah

1 0

Kamailio LCR - From Uri
by Fatima Chahrour~Vanrise Support 04 Oct '12

04 Oct '12

Dears, Am able to route call using LCR module from anywhere. I need to use certain option in the lcr tables in order to specify the ip of which calls is generated. I suppose that the From_uri column in LCR module meets my requirement, yet am not able to make it work and confused after reading several docs. My question is in what form should the from_uri column be? Is the ip enough? Are the Load params used below missing? modparam("lcr", "db_url","mysql://openser:openserrw@localhost/openser") modparam("lcr","lcr_gw_table","lcr_gw") modparam("lcr","from_uri_column","from_uri") modparam("lcr","priority_column","priority") modparam("lcr","gw_uri_avp", "$avp(i:709)") modparam("lcr","ruri_user_avp", "$avp(i:500)") modparam("lcr", "flags_avp", "$avp(i:712)") modparam("lcr","lcr_id_avp","$avp(s:lcr_id_avp)") What should I change in the routing script to consider the from_uri? if (!load_gws("1")) { sl_send_reply("503", "Unable To lOad GatEwAyS"); exit;} if(!next_gw()){ sl_send_reply("503", "Unable To fInD a gateWaY"); exit;} Thanks in advance, F Chahrour

3 3

sipcapture unsupported family
by Juha Heinanen 04 Oct '12

04 Oct '12

after adding a few module params modparam("sipcapture", "db_url", "MYSQL_SIP_PROXY_USAGE_URL") modparam("sipcapture", "table_name", "sip_capture") modparam("sipcapture", "hep_capture_on", 1) modparam("sipcapture", "capture_node", "EXTERNAL_HOSTNAME") i restarted my sip proxy and started to get these kind of errors to syslog: Oct 4 18:09:02 siika /usr/sbin/sip-proxy[22914]: ERROR: sipcapture [sipcapture.c:733]: ERROR: sipcapture:hep_msg_received: unsupported family [71] why they appear and how to turn them off? -- juha

3 4

Presence with RLS and integrated XCAP on Kamailio 3.3
by Sangeeta Shah 04 Oct '12

04 Oct '12

Hello All, Does anyone have presence working on Kamailio 3.3 with integrated XCAP. In theory my setup should work. I have two clients: Client A: Registers with the presence server Subscribe 1 - event = reg Subscribe 2 - event = presence.winfo Subscribe 3 - event = presence Publishes its status I see entries in the foll databases for A: Active Watchers DB: presence, presence.winfo, and reg event entries Watchers DB: No entries Presentity: presence, reg, and presence???? entries PUA list: 1 entry RLS watchers: No entries RLS presentity list: No entries XCAP List: 3 entries - rls services, resource list, and pres rules Client B: Registers with the presence server Subscribe 1 - event = presence.winfo Subscribe 2 - event = presence Subscribe 3 - event = message-summary Publishes its status I see entries in the foll databases for B: Active Watchers DB: message-summary, presence.winfo Watchers DB: 1 entry - with B as the presentity and A as the watcher Presentity: reg, presence PUA list: 1 entry RLS watchers: No entries RLS presentity list: No entries XCAP List: 2 entries resource-lists, pres rules My first question is - Is this all correct or is something going missing? Now the real problem: When Client A adds Client B to it's contact list, on Client A I see the following message: Client B Package (presence), Pending (Subscribe). This information from what I gather is coming back from the periodic notifies for the server because of the presence.winfo event. However, the notify going out to Client B does not have any indication that Client A is a watcher and needs to be authorized. This should be the case per RFC 3857. Excerpts from notifies to client A and client B: Client A NOTIFY sip:8475551001@10.51.2.181:58150;transport=udp SIP/2.0 Via: SIP/2.0/UDP 10.50.251.12;branch=z9hG4bKdcef.e5df11e.0 To: sip:8475551001@10.50.251.12;tag=1988526 From: sip:8475551001@10.50.251.12;tag=a6a1c5f60faecf035a1ae5b6e96e979a-c52c CSeq: 3 NOTIFY Call-ID: 19607474-addd-0ffb-a9da-031235052e9d Content-Length: 347 User-Agent: kamailio (3.3.1 (x86_64/linux)) Max-Forwards: 70 Event: presence.winfo Contact: <sip:10.50.251.12:5060;transport=udp> Subscription-State: active;expires=3180 Content-Type: application/watcherinfo+xml <?xml version="1.0"?> <watcherinfo xmlns="urn:ietf:params:xml:ns:watcherinfo" version="2" state="partial"> <watcher-list resource="sip:8475551001@10.50.251.12" package="presence"> <watcher id="9e65d804dbc4fc269c5f1d2f6eae0650(a)0.0.0.0" event="subscribe" status="pending">sip:8475551002@10.50.251.12</watcher> </watcher-list> </watcherinfo> Client B: NOTIFY sip:8475551002@10.50.0.113:5060;transport=udp;registering_acc=10_50_251_12 SIP/2.0 Via: SIP/2.0/UDP 10.50.251.12;branch=z9hG4bK7f71.6fca9e81.0 To: sip:8475551002@10.50.251.12;tag=a30db42f From: sip:8475551002@10.50.251.12;tag=a6a1c5f60faecf035a1ae5b6e96e979a-6692 CSeq: 3 NOTIFY Call-ID: f71666432f236f38c7c4dc467b4483e0(a)0.0.0.0 Content-Length: 195 User-Agent: kamailio (3.3.1 (x86_64/linux)) Max-Forwards: 70 Event: presence.winfo Contact: <sip:10.50.251.12:5060;transport=udp> Subscription-State: active;expires=3600 Content-Type: application/watcherinfo+xml <?xml version="1.0"?> <watcherinfo xmlns="urn:ietf:params:xml:ns:watcherinfo" version="2" state="full"> <watcher-list resource="sip:8475551002@10.50.251.12" package="presence"/> </watcherinfo> Also the frequency of notifies to client B is a lot less for event: presence.winfo. Any help is greatly appreciated since I've been trying to get presence to work for a few weeks now and keep running into new problems. Thanks, Sangeeta -- Sangeeta Shah

1 0

sip_capture schema?
by Juha Heinanen 04 Oct '12

04 Oct '12

i couldn't find in the source directory database schema for sip_capture table. why it is not included? also, looks like sipcapture module README is missing many of module's params. is this module still supported or has it become obsolete? -- juha

3 9

Kamailio - can't write any logs to file
by Nord7 04 Oct '12

04 Oct '12

Hello everyone. I have installed: CentOS6, Kamailio 3.3, rsyslogd Part of my Kamailio config: debug=9 log_stderror=yes memdbg=9 memlog=9 log_facility=LOG_LOCAL0 loaded modules: debugger.so modparam("debugger", "cfgtrace", 1) modparam("debugger", "log_facility", "LOG_LOCAL0") modparam("debugger", "log_prefix", "from-debugger-with-love:") xlog.so modparam("xlog", "buf_size", 8192) modparam("xlog", "log_facility", "LOG_LOCAL0") rsyslog.conf default config with two added rows: # Save log messages with LOG_LOCAL0 in /var/log/kamailio local0.* /var/log/kamailio rsyslogd and kamailio restarted. When i run command "kamailio -ddddddd" i see debug all my xlog("L_INFO","blabla\n") or xlog("LOG_LOCAL0","L_INFO","blabla\n" ); messages. When i run "kamctl start" i don't have any logs from kamailio in my system. I'm stuck, any suggestions, will be helpful -- Best regards, Kirill Ushakov

2 2

Re: [SR-Users] Error when client tries to put XCAP resource list
by Daniel-Constantin Mierla 03 Oct '12

03 Oct '12

Hello, do not reply privately to discussions on the mailing list. It is important that the thread has a conclusion for the people searching in archive for the same topic. Cheers, Daniel On 10/3/12 10:13 PM, Sangeeta Shah wrote: > Thanks for the response. I was able to figure it out and now it works. > > Appreciate your response. > > On Wed, Oct 3, 2012 at 3:02 PM, Daniel-Constantin Mierla > <miconda(a)gmail.com> wrote: >> Hello, >> >> the http uri is available in available $hu. You can print it with xlog and >> be sure you give it as parameter to xcap server function. >> >> A basic example in the readme shows it: >> * >> http://kamailio.org/docs/modules/stable/modules_k/xcap_server.html#id2532050 >> >> Cheers, >> Daniel >> >> >> >> On 10/3/12 8:34 PM, Sangeeta Shah wrote: >>> Hello All, >>> I am using Kamailio 3.3 with Presence/RLS and integrated XCAP. When >>> my client tries to "PUT" the resource list for the user, the server is >>> returning a 500 Server Error response. Attached is the TCP trace for >>> the put and the corresponding response from the server. I also see the >>> following error in the server logs: >>> Oct 3 12:13:04 RCS-Presence /usr/local/sbin/kamailio[17593]: ERROR: >>> <script>: ===== xhttp: put received sip:8475551001@10.50.251.12 >>> Oct 3 12:13:04 RCS-Presence /usr/local/sbin/kamailio[17593]: ERROR: >>> xcap_server [xcap_misc.c:109]: invalid http uri - shorter than >>> xcap-root >>> Oct 3 12:13:04 RCS-Presence /usr/local/sbin/kamailio[17593]: ERROR: >>> xcap_server [xcap_server.c:560]: cannot parse xcap uri [0] >>> >>> Maybe it's a client/server configuration issue. What is the expected >>> http uri. How do I print out the received HTTP URI And the XCAP root. >>> >>> Any debugging help is appreciated. >>> >>> TCP trace: >>> >>> PUT /xcap-root/resource-lists/users/8475551001/index HTTP/1.1 >>> >>> Host: 10.50.251.12:5060 >>> >>> Content-Length: 2037 >>> >>> Connection: Keep-Alive >>> >>> User-Agent: XDM-client/OMA1.1 >>> >>> X-3GPP-Intended-Identity: 8475551001 >>> >>> Content-Type: application/resource-lists+xml >>> >>> >>> >>> <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> >>> >>> <list name="rcs"> >>> >>> <display-name>All Contacts</display-name> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> <entry uri="8475551001" /> >>> >>> </list> >>> >>> <list name="rcs_blockedcontacts"> >>> >>> <display-name>Blocked Contacts</display-name> >>> >>> </list> >>> >>> <list name="rcs_revokedcontacts"> >>> >>> <display-name>Revoked Contacts</display-name> >>> >>> </list> >>> >>> <list name="oma_allcontacts"> >>> >>> <display-name>OMA All Contacts</display-name> >>> >>> </list> >>> >>> <list name="oma_blockedcontacts"> >>> >>> <display-name>OMA Blocked Contacts</display-name> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> </list> >>> >>> <list name="oma_buddylist"> >>> >>> <display-name>OMA BuddyList</display-name> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> </list> >>> >>> <list name="oma_grantedcontacts"> >>> >>> <display-name>OMA Granted Contacts</display-name> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> <external >>> anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" >>> /> >>> >>> </list> >>> >>> <list name="oma_pocbuddylist"> >>> >>> <display-name>OMA POC BuddyList</display-name> >>> >>> </list> >>> >>> <list name="doubango"> >>> >>> <display-name>My Default Contacts</display-name> >>> >>> </list> >>> >>> </resource-lists> >>> >>> HTTP/1.1 500 Server error >>> >>> Via: SIP/2.0/TCP 10.51.2.181:49889 >>> >>> Server: kamailio (3.3.1 (x86_64/linux)) >>> >>> Content-Length: 0 >>> >>> Thanks, >>> Sangeeta >>> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.com >> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat >> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - >> http://asipto.com/u/katu >> > > -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Error when client tries to put XCAP resource list
by Sangeeta Shah 03 Oct '12

03 Oct '12

Hello All, I am using Kamailio 3.3 with Presence/RLS and integrated XCAP. When my client tries to "PUT" the resource list for the user, the server is returning a 500 Server Error response. Attached is the TCP trace for the put and the corresponding response from the server. I also see the following error in the server logs: Oct 3 12:13:04 RCS-Presence /usr/local/sbin/kamailio[17593]: ERROR: <script>: ===== xhttp: put received sip:8475551001@10.50.251.12 Oct 3 12:13:04 RCS-Presence /usr/local/sbin/kamailio[17593]: ERROR: xcap_server [xcap_misc.c:109]: invalid http uri - shorter than xcap-root Oct 3 12:13:04 RCS-Presence /usr/local/sbin/kamailio[17593]: ERROR: xcap_server [xcap_server.c:560]: cannot parse xcap uri [0] Maybe it's a client/server configuration issue. What is the expected http uri. How do I print out the received HTTP URI And the XCAP root. Any debugging help is appreciated. TCP trace: PUT /xcap-root/resource-lists/users/8475551001/index HTTP/1.1 Host: 10.50.251.12:5060 Content-Length: 2037 Connection: Keep-Alive User-Agent: XDM-client/OMA1.1 X-3GPP-Intended-Identity: 8475551001 Content-Type: application/resource-lists+xml <resource-lists xmlns="urn:ietf:params:xml:ns:resource-lists"> <list name="rcs"> <display-name>All Contacts</display-name> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> <entry uri="8475551001" /> </list> <list name="rcs_blockedcontacts"> <display-name>Blocked Contacts</display-name> </list> <list name="rcs_revokedcontacts"> <display-name>Revoked Contacts</display-name> </list> <list name="oma_allcontacts"> <display-name>OMA All Contacts</display-name> </list> <list name="oma_blockedcontacts"> <display-name>OMA Blocked Contacts</display-name> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> </list> <list name="oma_buddylist"> <display-name>OMA BuddyList</display-name> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> </list> <list name="oma_grantedcontacts"> <display-name>OMA Granted Contacts</display-name> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> <external anchor="http://10.50.251.12:5060/xcap-root/resource-lists/users/8475551001/index/~~…" /> </list> <list name="oma_pocbuddylist"> <display-name>OMA POC BuddyList</display-name> </list> <list name="doubango"> <display-name>My Default Contacts</display-name> </list> </resource-lists> HTTP/1.1 500 Server error Via: SIP/2.0/TCP 10.51.2.181:49889 Server: kamailio (3.3.1 (x86_64/linux)) Content-Length: 0 Thanks, Sangeeta -- Sangeeta Shah

2 1

Re: [SR-Users] Handling Tel URIs
by Kamal Palei 03 Oct '12

03 Oct '12

Hi Daniel It worked fine. I added the below code. if(is_method("PUBLISH")) { loose_route(); t_relay(); exit; } Thanks Kamal On Wed, Oct 3, 2012 at 5:58 PM, Kamal Palei <palei.kamal(a)gmail.com> wrote: > Thanks Daniel , let me try this out. > > Regards > Kamal > > On Wed, Oct 3, 2012 at 5:33 PM, Daniel-Constantin Mierla < > miconda(a)gmail.com> wrote: > >> Hello, >> >> do you have a Route header also for kamailio proxy? If yes, try to do >> loose_route() and then t_relay(). >> >> If not, get the address from Route in $du and relay: >> $du = $(hdr(Route){nameaddr.uri}); >> remove_hf("Route"); >> t_relay(); >> exit; >> >> Cheers, >> Daniel >> >> >> On 10/3/12 1:46 PM, Kamal Palei wrote: >> >> Hi Daniel >> Thanks for your response. >> >> I have a message as below. >> >> PUBLISH tel:+915346586495469 <+915346586495469> SIP/2.0 >> Via: >> To: >> From: >> Route: <sip:main_proxy_ip> >> .. >> .. >> This message comes from phone to Kamailio proxy. >> I just need Kamailio proxy just forward this message to main proxy >> without modifying tel uri present in request uri. >> >> Can this be acheived by tel2sip() api. >> >> Thanks >> Kamal >> >> >> On Wed, Oct 3, 2012 at 4:37 PM, Daniel-Constantin Mierla < >> miconda(a)gmail.com> wrote: >> >>> Hello, >>> >>> >>> On 10/3/12 12:36 PM, Kamal Palei wrote: >>> >>> Changing the subject line >>> >>> On Wed, Oct 3, 2012 at 3:54 PM, Kamal Palei <palei.kamal(a)gmail.com>wrote: >>> >>>> Hi All >>>> I am using Kamailio - 3.1.5 version. >>>> >>>> When a message of tel URI comes to Kamailio, Kamailio is not forwarding >>>> the message to next proxy. >>>> If is returning "478 Unresolved response" . >>>> >>>> Can somebody give me some pointer how do I handle this situation? >>>> >>> kamailio cannot forward when there is tel uri -- it requires a domain >>> part (hostname/ip) to know where to send the requests. >>> >>> See this function that should help you convert the tel uri to sip uri: >>> http://kamailio.org/docs/modules/stable/modules_k/siputils.html#id2494464 >>> >>> Cheers, >>> Daniel >>> >>> -- >>> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >>> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat >>> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu >>> >>> >> >> -- >> Daniel-Constantin Mierla - http://www.asipto.comhttp://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda >> Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat >> Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu >> >> >

1 0

Re: [SR-Users] Handling Tel URIs
by Kamal Palei 03 Oct '12

03 Oct '12

Changing the subject line On Wed, Oct 3, 2012 at 3:54 PM, Kamal Palei <palei.kamal(a)gmail.com> wrote: > Hi All > I am using Kamailio - 3.1.5 version. > > When a message of tel URI comes to Kamailio, Kamailio is not forwarding > the message to next proxy. > If is returning "478 Unresolved response" . > > Can somebody give me some pointer how do I handle this situation? > > Thanks > Kamal > > _______________________________________________ > SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list > sr-users(a)lists.sip-router.org > http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users > >

2 3

Kamailio CDR logging to DB
by Tayeb Meftah 03 Oct '12

03 Oct '12

hi everyone, can someone show me how to save cdr's into db ? i enabled acc in the default config but still no cdr in the acc table thank you -- Tayeb Meftah General director Voice of the blind ™ broadcast freedom http://www.vobradio.org/ Phones: Algeria: +213-660-347746 UK, London: +442032983627 USA, NewYork: +1(347)759-5883 iNum: +883510001288000 we're blind, but not deaf!

2 1

Locate UA in usrloc table
by Vitaliy Aleksandrov 03 Oct '12

03 Oct '12

Hi all. Is there any easy method to find if somebody with $si:$sp is registered (usrloc->received.ip == $si, usrloc->received.port == $sp) ? reg_fetch_contacts uses uri as search key, but in my case I don't have it. The only way I know is to write a module that will use "get_all_ucontacts" function exported by usrloc module in its API.

2 1

FreeSWITCH | Kamailio MWI forwarding
by Tayeb Meftah 03 Oct '12

03 Oct '12

hello we're using Kamailio / freeswitch freeswitch is the Voicemail server now. please how to forward MWI bethwan kamailio and Freeswitch? if user 1000 have a vm i want kamailio to fetch it from FS and advertise it to the subscriber thank you -- Tayeb Meftah General director Voice of the blind ™ broadcast freedom http://www.vobradio.org/ Phones: Algeria: +213-660-347746 UK, London: +442032983627 USA, NewYork: +1(347)759-5883 iNum: +883510001288000 we're blind, but not deaf!

2 1

WARNING:fm_free at stop
by Juha Heinanen 03 Oct '12

03 Oct '12

at some point i have started to get these kind of warning when i stop my sip proxy: --text follows this line-- Oct 1 10:23:01 siika /usr/sbin/sip-proxy[16970]: WARNING: <core> [mem/f_malloc.c:512]: WARNING:fm_free: free(0) called any idea where they come from or how i could found out where they come from? -- juha

2 20

(no subject)
by Kamal Palei 03 Oct '12

03 Oct '12

Hi All I am using Kamailio - 3.1.5 version. When a message of tel URI comes to Kamailio, Kamailio is not forwarding the message to next proxy. If is returning "478 Unresolved response" . Can somebody give me some pointer how do I handle this situation? Thanks Kamal

1 0

Re: [SR-Users] uac From restore not working in 3.3?
by Alex Balashov 02 Oct '12

02 Oct '12

I will check. The problem is that this installation is exotic and deviant from my usual order of operations, which creates further confusion. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: On 10/2/12 9:45 PM, Alex Balashov wrote: Perhaps the source of my misunderstanding is something to do with branching. The actual case I am troubleshooting is a 503 generated via t_reply out of a failure route. The complaint is that this 503 contains a modified From header, exposed to the originator of the call. If you created the transaction after calling msg_apply_changes(), then I am quite sure it is how it worked all the time. Try with an older version and see. Cheers, Daniel -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Those originated locally were never using the original value if you called msg apply changes -- you can try with an older version, if it does that, then it is a bug somewhere :-). Btw, if you called msg apply changes after creating the transaction, then SL is out of scope and tm local reply is using the initial values, but that means the changes are only for that branch, a failure route will get the initial request content. Cheers, Daniel On 10/2/12 9:30 PM, Alex Balashov wrote: The expected behaviour is that all replies, both relayed and endogenously originated (via SL or TM), as well as sequential requests directed back to the caller, will preserve the original headers. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Not sure what is 'work as you expect', provided the details I gave in the previous mail related to local replies. Otherwise, if you don't track all dialogs, could be an issue if some of the replies are ok and some are not. I will check the uac module and add a parameter to not use dialog module even if loaded, if such option is not already there. Cheers, Daniel On 10/2/12 9:11 PM, Alex Balashov wrote: So, you're saying that if all the dialogs were tracked, everything would work as I expect it to? -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: I think there was an addition to use dialog module to store initial and new values. I don't know if it can be turned off or is automatically when the dialog module is loaded - just to see if that is the cause. However, if you do msg_apply_changes(), local replies will use the new headers. I'm pretty sure it happened the same before -- e.g., it was no way to revert back so that sl can use older values of the headers. Perhaps you can revert from config, changing the header to initial value that you store in an avp, apply changes and send the reply. Received and relayed replies should be converted. Cheers, Daniel On 10/2/12 8:57 PM, Alex Balashov wrote: I should be clearer: when I say "internally generated", I don't mean it in the sense of 408 or something else opaquely generated by the proxy. I mean internally generated as in originating from the proxy, e.g. something sent with sl_send_reply() or t_reply(). -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Alex Balashov <abalashov(a)evaristesys.com> wrote: Hello, I am using the latest cloned 3.3 branch and performing a uac_replace_from() operation. After the operation, I do msg_apply_changes(). My 'uac' modparams are: modparam("uac", "rr_from_store_param", "vsfc") modparam("uac", "restore_mode", "auto") modparam("uac", "restore_passwd", "xxxxxxxxx") However, the modified From URI is getting back to the originator, both in internally generated errors, and in relayed replies. This didn't used to happen before 3.3, but has been outstanding for quite some time (I didn't look at it much until now). Is there some possibility that there is a >= 3.3 bug? Or, have I simply misconfigured something? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

2 1

Re: [SR-Users] uac From restore not working in 3.3?
by Alex Balashov 02 Oct '12

02 Oct '12

Perhaps the source of my misunderstanding is something to do with branching. The actual case I am troubleshooting is a 503 generated via t_reply out of a failure route. The complaint is that this 503 contains a modified From header, exposed to the originator of the call. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Daniel-Constantin Mierla <miconda(a)gmail.com> wrote:Those originated locally were never using the original value if you called msg apply changes -- you can try with an older version, if it does that, then it is a bug somewhere :-). Btw, if you called msg apply changes after creating the transaction, then SL is out of scope and tm local reply is using the initial values, but that means the changes are only for that branch, a failure route will get the initial request content. Cheers, Daniel On 10/2/12 9:30 PM, Alex Balashov wrote: The expected behaviour is that all replies, both relayed and endogenously originated (via SL or TM), as well as sequential requests directed back to the caller, will preserve the original headers. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: Not sure what is 'work as you expect', provided the details I gave in the previous mail related to local replies. Otherwise, if you don't track all dialogs, could be an issue if some of the replies are ok and some are not. I will check the uac module and add a parameter to not use dialog module even if loaded, if such option is not already there. Cheers, Daniel On 10/2/12 9:11 PM, Alex Balashov wrote: So, you're saying that if all the dialogs were tracked, everything would work as I expect it to? -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: I think there was an addition to use dialog module to store initial and new values. I don't know if it can be turned off or is automatically when the dialog module is loaded - just to see if that is the cause. However, if you do msg_apply_changes(), local replies will use the new headers. I'm pretty sure it happened the same before -- e.g., it was no way to revert back so that sl can use older values of the headers. Perhaps you can revert from config, changing the header to initial value that you store in an avp, apply changes and send the reply. Received and relayed replies should be converted. Cheers, Daniel On 10/2/12 8:57 PM, Alex Balashov wrote: I should be clearer: when I say "internally generated", I don't mean it in the sense of 408 or something else opaquely generated by the proxy. I mean internally generated as in originating from the proxy, e.g. something sent with sl_send_reply() or t_reply(). -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Alex Balashov <abalashov(a)evaristesys.com> wrote: Hello, I am using the latest cloned 3.3 branch and performing a uac_replace_from() operation. After the operation, I do msg_apply_changes(). My 'uac' modparams are: modparam("uac", "rr_from_store_param", "vsfc") modparam("uac", "restore_mode", "auto") modparam("uac", "restore_passwd", "xxxxxxxxx") However, the modified From URI is getting back to the originator, both in internally generated errors, and in relayed replies. This didn't used to happen before 3.3, but has been outstanding for quite some time (I didn't look at it much until now). Is there some possibility that there is a >= 3.3 bug? Or, have I simply misconfigured something? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

2 1

Re: [SR-Users] uac From restore not working in 3.3?
by Alex Balashov 02 Oct '12

02 Oct '12

The expected behaviour is that all replies, both relayed and endogenously originated (via SL or TM), as well as sequential requests directed back to the caller, will preserve the original headers. -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Daniel-Constantin Mierla <miconda(a)gmail.com> wrote:Not sure what is 'work as you expect', provided the details I gave in the previous mail related to local replies. Otherwise, if you don't track all dialogs, could be an issue if some of the replies are ok and some are not. I will check the uac module and add a parameter to not use dialog module even if loaded, if such option is not already there. Cheers, Daniel On 10/2/12 9:11 PM, Alex Balashov wrote: So, you're saying that if all the dialogs were tracked, everything would work as I expect it to? -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Daniel-Constantin Mierla <miconda(a)gmail.com> wrote: I think there was an addition to use dialog module to store initial and new values. I don't know if it can be turned off or is automatically when the dialog module is loaded - just to see if that is the cause. However, if you do msg_apply_changes(), local replies will use the new headers. I'm pretty sure it happened the same before -- e.g., it was no way to revert back so that sl can use older values of the headers. Perhaps you can revert from config, changing the header to initial value that you store in an avp, apply changes and send the reply. Received and relayed replies should be converted. Cheers, Daniel On 10/2/12 8:57 PM, Alex Balashov wrote: I should be clearer: when I say "internally generated", I don't mean it in the sense of 408 or something else opaquely generated by the proxy. I mean internally generated as in originating from the proxy, e.g. something sent with sl_send_reply() or t_reply(). -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Alex Balashov <abalashov(a)evaristesys.com> wrote: Hello, I am using the latest cloned 3.3 branch and performing a uac_replace_from() operation. After the operation, I do msg_apply_changes(). My 'uac' modparams are: modparam("uac", "rr_from_store_param", "vsfc") modparam("uac", "restore_mode", "auto") modparam("uac", "restore_passwd", "xxxxxxxxx") However, the modified From URI is getting back to the originator, both in internally generated errors, and in relayed replies. This didn't used to happen before 3.3, but has been outstanding for quite some time (I didn't look at it much until now). Is there some possibility that there is a >= 3.3 bug? Or, have I simply misconfigured something? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

2 1

Re: [SR-Users] uac From restore not working in 3.3?
by Alex Balashov 02 Oct '12

02 Oct '12

So, you're saying that if all the dialogs were tracked, everything would work as I expect it to? -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Daniel-Constantin Mierla <miconda(a)gmail.com> wrote:I think there was an addition to use dialog module to store initial and new values. I don't know if it can be turned off or is automatically when the dialog module is loaded - just to see if that is the cause. However, if you do msg_apply_changes(), local replies will use the new headers. I'm pretty sure it happened the same before -- e.g., it was no way to revert back so that sl can use older values of the headers. Perhaps you can revert from config, changing the header to initial value that you store in an avp, apply changes and send the reply. Received and relayed replies should be converted. Cheers, Daniel On 10/2/12 8:57 PM, Alex Balashov wrote: I should be clearer: when I say "internally generated", I don't mean it in the sense of 408 or something else opaquely generated by the proxy. I mean internally generated as in originating from the proxy, e.g. something sent with sl_send_reply() or t_reply(). -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/ Alex Balashov <abalashov(a)evaristesys.com> wrote: Hello, I am using the latest cloned 3.3 branch and performing a uac_replace_from() operation. After the operation, I do msg_apply_changes(). My 'uac' modparams are: modparam("uac", "rr_from_store_param", "vsfc") modparam("uac", "restore_mode", "auto") modparam("uac", "restore_passwd", "xxxxxxxxx") However, the modified From URI is getting back to the originator, both in internally generated errors, and in relayed replies. This didn't used to happen before 3.3, but has been outstanding for quite some time (I didn't look at it much until now). Is there some possibility that there is a >= 3.3 bug? Or, have I simply misconfigured something? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

2 1

Re: [SR-Users] uac From restore not working in 3.3?
by Alex Balashov 02 Oct '12

02 Oct '12

I should be clearer: when I say "internally generated", I don't mean it in the sense of 408 or something else opaquely generated by the proxy. I mean internally generated as in originating from the proxy, e.g. something sent with sl_send_reply() or t_reply(). -- Alex -- Sent from my Samsung mobile, and thus lacking in the refinement one might expect from a proper keyboard. Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Web: http://www.evaristesys.com/Alex Balashov <abalashov(a)evaristesys.com> wrote:Hello, I am using the latest cloned 3.3 branch and performing a uac_replace_from() operation. After the operation, I do msg_apply_changes(). My 'uac' modparams are: modparam("uac", "rr_from_store_param", "vsfc") modparam("uac", "restore_mode", "auto") modparam("uac", "restore_passwd", "xxxxxxxxx") However, the modified From URI is getting back to the originator, both in internally generated errors, and in relayed replies. This didn't used to happen before 3.3, but has been outstanding for quite some time (I didn't look at it much until now). Is there some possibility that there is a >= 3.3 bug? Or, have I simply misconfigured something? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/ _______________________________________________ SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list sr-users(a)lists.sip-router.org http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users

2 1

uac From restore not working in 3.3?
by Alex Balashov 02 Oct '12

02 Oct '12

Hello, I am using the latest cloned 3.3 branch and performing a uac_replace_from() operation. After the operation, I do msg_apply_changes(). My 'uac' modparams are: modparam("uac", "rr_from_store_param", "vsfc") modparam("uac", "restore_mode", "auto") modparam("uac", "restore_passwd", "xxxxxxxxx") However, the modified From URI is getting back to the originator, both in internally generated errors, and in relayed replies. This didn't used to happen before 3.3, but has been outstanding for quite some time (I didn't look at it much until now). Is there some possibility that there is a >= 3.3 bug? Or, have I simply misconfigured something? Thanks! -- Alex -- Alex Balashov - Principal Evariste Systems LLC 235 E Ponce de Leon Ave Suite 106 Decatur, GA 30030 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/, http://www.alexbalashov.com/

1 0

Kamailio project booth at Astricon 2012
by Daniel-Constantin Mierla 02 Oct '12

02 Oct '12

Hello, Kamailio project will participate at Astricon 2012 with its own booth in the open source area, courtesy of our Asterisk friends organizing the show. The event takes place in Atlanta, GA, USA, between Oct 23-25, 2012: * http://www.astricon.net/2012/default.aspx Special thanks to community members JR Richardson, Nir Simionovich and Eric Klein, which coordinate the activity at the booth. I made a news about it at: * http://www.kamailio.org/w/2012/10/kamailio-booth-at-astricon12/ If you are interested in joining the team at the booth and present how you use Kamailio, reply to me in order discuss further. If you participate at Astricon, stop by booth #20 to chat about use cases of Kamailio and what is new around the project. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

Kamailio - VoIP dinner in Berlin, next Monday, Oct 8, 2012
by Daniel-Constantin Mierla 02 Oct '12

02 Oct '12

Hello, we're planning a dinner in Berlin, next Monday, Oct 8, 2012 - exact time and place to be decided soon. Besides me, among people interested to participate are Alex Balashov, Andrei Pelinescu-Onciul, Jiri Kuthan and Raphael Coeffic. If anyone here wants to participate, reply to count the heads for reservation. It's the usual german rule, everyone pays for him/herself. Cheers, Daniel -- Daniel-Constantin Mierla - http://www.asipto.com http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda Kamailio Advanced Training, Berlin, Nov 5-8, 2012 - http://asipto.com/u/kat Kamailio Advanced Training, Miami, USA, Nov 12-14, 2012 - http://asipto.com/u/katu

1 0

XCAP configuration with Presence Server and Resource lists not working on Kamailio 3.3.1
by Sangeeta Shah 01 Oct '12

01 Oct '12

Hello All, I am trying to get the presence module on Kamailio 3.3.1 working with integrated XCAP and resource lists. The clients I am using to test are: Jitsi and Boghe. I am following the configuration as specified by the respective module documentation: http://kamailio.org/docs/modules/devel/modules_k/xcap_server.html http://kb.asipto.com/kamailio:presence:k31-made-simple http://www.kamailio.org/dokuwiki/doku.php/presence:presence-module and http://kamailio.org/docs/modules/devel/modules_k/rls.html The server is up and running but it doesn't seem like it's working for either one of the clients. I checked the archives and someone had a similar issue in Aug and switched back to releases 3.2/3.1. Has anything been resolved? Are there any changes that need to be made to the documentation for 3.3 to support this functionality. Any help is appreciated. If tcpdumps for the respective clients are desired I can provide them. Thanks, Sangeeta -- Sangeeta Shah

2 2

Error starting kamailio with dialog module usage
by Aft nix 01 Oct '12

01 Oct '12

Hi, After implementing a "Call limit" application with the following config: #!ifdef WITH_CALL_LIMIT modparam("dialog","enable_stats",1) modparam("dialog","dlg_flag", 4) modparam("dialog","hash_size", 4096) modparam("dialog", "profiles_with_value","callquota") modparam("dialog", "default_timeout", 300) modparam("dialog", "dlg_match_mode", 2) modparam("dialog", "detect_spirals", 1) modparam("dialog", "db_mode", 0) #!endif request_route { ................ #!ifdef WITH_CALL_LIMIT if (is_method("INVITE")) { dlg_manage(); $var(100) = 0; get_profile_size("callquota", "$rU", "var(100)"); if ($var(100) >= 5 ) { xlog("DEBUG: Simultaneous calls limit reached"); sl_send_reply("503","Simultaneous calls limit reached"); exit; } set_dlg_profile("callquota","$ru"); if (get_profile_size("callquota", "$var(100)")) { xlog("DEBUG: there are $var(100) total calls for $rU\n"); } } #!endif ............... } Now it passes cleanly when i issue : $kamailio -c but it fails to start with following info: Sep 29 19:37:49 108 /usr/local/sbin/kamailio[1865]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:507 Sep 29 19:37:49 108 /usr/local/sbin/kamailio[1865]: ERROR: <core> [route.c:1216]: fixing failed (code=-1) at cfg:/usr/local/etc/kamailio/kamailio.cfg:523 line 507 : get_profile_size("callquota", "$rU", "var(100)"); Thanks in advance -- -aft

2 2

kamailio core at qm_status
by Jijo 01 Oct '12

01 Oct '12

Hi All, I'm observing a core intermittently at "qm_status (qm=0x786cd000) at mem/q_malloc.c:763" for kamailio version 3.1.0 looking at the backtrace this is occurring while doing a tcp buffer overrun. Please have a look and let me know if anyone observed this issue or how can we debug it. Please find the backtrace and kamailio version osbprod-V2R0:~ # /usr/sbin/kamailio -V version: kamailio 3.1.0 (i386/linux) 21a375 flags: STATS: Off, USE_IPV6, USE_TCP, USE_TLS, TLS_HOOKS, USE_RAW_SOCKS, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, USE_FUTEX, FAST_LOCK-ADAPTIVE_WAIT, USE_DNS_CACHE, USE_DNS_FAILOVER, USE_NAPTR, USE_DST_BLACKLIST, HAVE_RESOLV_RES ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535, PKG_SIZE 15MB poll method support: poll, epoll_lt, epoll_et, sigio_rt, select. id: 21a375 compiled on 12:38:44 Apr 26 2012 with gcc 4.5.0 #0 qm_status (qm=0x786cd000) at mem/q_malloc.c:763 #1 0x0819c0f3 in qm_debug_frag (qm=0x786cd000, f=<value optimized out>) at mem/q_malloc.c:167 #2 0x0819c352 in qm_free (qm=0x786cd000, p=0x7a6cb6ac, file=0x821cce2 "<core>: tcp_main.c", func=0x822285f "wbufq_run", line=880) at mem/q_malloc.c:457 #3 0x0815f686 in wbufq_run (tcpconn=0x7a1a7548, ev=4, fd_i=-1) at tcp_main.c:880 #4 handle_tcpconn_ev (tcpconn=0x7a1a7548, ev=4, fd_i=-1) at tcp_main.c:4141 #5 0x08169bf2 in io_wait_loop_epoll () at io_wait.h:1092 #6 tcp_main_loop () at tcp_main.c:4606 #7 0x080b0404 in main_loop () at main.c:1655 #8 0x080b1f84 in main (argc=9, argv=0xbfb4c9f4) at main.c:2446 (gdb) l 758 LOG_(DEFAULT_FACILITY, memlog, "qm_status: ", 759 "dumping free list stats :\n"); 760 for(h=0,i=0;h<QM_HASH_SIZE;h++){ 761 unused=0; 762 for (f=qm->free_hash[h].head.u.nxt_free,j=0; 763 f!=&(qm->free_hash[h].head); f=f->u.nxt_free, i++, j++){ 764 if (!FRAG_WAS_USED(f)){ 765 unused++; 766 #ifdef DBG_QM_MALLOC 767 LOG_(DEFAULT_FACILITY, memlog, "qm_status: ",

4 7

How to force tcp connection reuse for NOTIFY
by Owen Lynch 01 Oct '12

01 Oct '12

Hi, I want to configure Kamailio so that it sends a NOTIFY using the tcp connection already established by the client SUBSCRIBE request. The aim is for clients to establish secure TLS connections to Kamailio when they first register or subscribe, then for kamailio to reuse these connections for all further NOTIFY/INVITE requests. I thought that adding force_tcp_alias() in the route script would accomplish this, but it doesn't. The NOTIFY always uses the ip:port tuple specified in the SUBSCRIBE Contact header. Can anyone shed any light on this? Thanks, Owen Lynch

2 2

Registration Authentication Error
by Nathaniel L Keeling III 30 Sep '12

30 Sep '12

Hello, we are testing the new version of Kamailio, 3.3 and are getting authentication error when trying to register a sip account. We have a db backend in which we used kamctl to add the user to the db. We are using the kamailio.cfg script created during the install with changes for our environment. Is there some other changes required besides the normal. I can see in the logs where everything looks ok except for when it compares the calculated values. I have doubled check the subscriber table and password but to no avail. 11(28548) DEBUG: db_postgres [km_dbase.c:393]: PQclear(1008b8560) result set 11(28548) DEBUG: auth_db [authorize.c:124]: HA1 string calculated: e13d569284e76a33ca63e4d6203f844a 11(28548) DEBUG: auth [api.c:211]: check_response: Our result = '18210b5da2b3d27e6ba50977072599ea' 11(28548) DEBUG: auth [api.c:221]: check_response: Authorization failed Thanks Nathaniel

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