sr-dev September 2012

sr-dev@lists.kamailio.org

32 participants
174 discussions

git:master: dialog: use long_hf_len instead of hardcoded value in table defs, no actual schema change
by Henning Westerholt 07 Sep '12

07 Sep '12

Module: sip-router Branch: master Commit: 976f801ae2df9597be0fccff89496bde34b22e62 URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=976f801… Author: Henning Westerholt <hw(a)kamailio.org> Committer: Henning Westerholt <hw(a)kamailio.org> Date: Fri Sep 7 12:44:33 2012 +0200 dialog: use long_hf_len instead of hardcoded value in table defs, no actual schema change --- lib/srdb1/schema/dialog.xml | 6 +++--- lib/srdb1/schema/dialog_vars.xml | 2 +- lib/srdb1/schema/entities.xml | 1 + 3 files changed, 5 insertions(+), 4 deletions(-) diff --git a/lib/srdb1/schema/dialog.xml b/lib/srdb1/schema/dialog.xml index 19bf242..fadb5aa 100644 --- a/lib/srdb1/schema/dialog.xml +++ b/lib/srdb1/schema/dialog.xml @@ -101,7 +101,7 @@ <column> <name>caller_route_set</name> <type>string</type> - <size>512</size> + <size>&long_hf_len;</size> <null/> <description>Route set on the caller side.</description> </column> @@ -109,7 +109,7 @@ <column> <name>callee_route_set</name> <type>string</type> - <size>512</size> + <size>&long_hf_len;</size> <null/> <description>Route set on on the caller side.</description> </column> @@ -204,7 +204,7 @@ <column> <name>xdata</name> <type>string</type> - <size>512</size> + <size>&long_hf_len;</size> <null/> <description>Extra data associated to the dialog (e.g., serialized profiles).</description> </column> diff --git a/lib/srdb1/schema/dialog_vars.xml b/lib/srdb1/schema/dialog_vars.xml index 429b01c..7c7a757 100644 --- a/lib/srdb1/schema/dialog_vars.xml +++ b/lib/srdb1/schema/dialog_vars.xml @@ -54,7 +54,7 @@ <column> <name>dialog_value</name> <type>string</type> - <size>512</size> + <size>&long_hf_len;</size> <description>The value of the dialog variable</description> </column> diff --git a/lib/srdb1/schema/entities.xml b/lib/srdb1/schema/entities.xml index ff12ca0..8ff237d 100644 --- a/lib/srdb1/schema/entities.xml +++ b/lib/srdb1/schema/entities.xml @@ -6,6 +6,7 @@ <!ENTITY id_len "64"> <!ENTITY table_id_len "10"> <!ENTITY hf_len "255"> +<!ENTITY long_hf_len "512"> <!ENTITY rule_len "255"> <!ENTITY method_len "16"> <!ENTITY avp_name_len "32">

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Siremis v3.3.0 Released
by Elena-Ramona Modroiu 06 Sep '12

06 Sep '12

Hi, Siremis v3.3.0 was released - the web management interface compatible with Kamailio v3.3.x. This version is an update of v3.2.1 to be able to work with the database schema of Kamailio v3.3.x. The news of the release, including links to installation and screenshots, is available at: - http://siremis.asipto.com/2012/09/06/siremis-v3-3-0-released/ Regards, Ramona

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Best way to detect when E2EACK has been sent by proxy using tm hooks?
by Andrew Mortensen 05 Sep '12

05 Sep '12

I'm working on a custom module for 3.2.x, and I'd like to send an UPDATE to both legs of a call after the ACK from the caller is relayed by the proxy to the callee. I've hooked TMCB_E2EACK_IN, which allows me to detect when the caller sends the ACK to the proxy, but there doesn't seem to be an obvious way for me to detect when the proxy has actually relayed the ACK to the callee. If I register TMCB_REQUEST_SENT in the TMCB_E2EACK_IN callback, the TMCB_REQUEST_SENT callback is never invoked, so for the moment I'm registering a TMCB_DESTROY callback in the TMCB_E2EACK_IN callback. This does get invoked, and I believe I can get everything I need from the struct cell and the param, but the callback is triggered fairly late by tm. I have two questions, though: First, is registering the TMCB_DESTROY callback in the ACK callback safe? Second, is there a better way to do this? I'd prefer options available with the 3.2 release, since our deployment is based on it, but if 3.3+ offers a solution, I'm willing to take a look. Thanks. andrew

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Testing the Websocket module with sipml5.org
by Juan Castro 05 Sep '12

05 Sep '12

Well, not the simpl5.org site itself, since it uses its own Websocket-SIP gateway to make calls. I downloaded their code and changed the gateway address from "sipml5.org" to "192.168.0.148" which is the VM where I'm testing. Also, Apache is using ports 8000 and 4443 since Kamailio uses 80 and 443. But when I try to make a call, filling 192.168.0.48 as the realm, I get "Disconnected: Failed to connet to the server" from the call.htm page. I monitored packets with Wireshark and I see no traffic that would make sense between the machine with the browser and the one with Kamailio. I suspect I have to make more changes in the sipml5.org Javascript, am I correct? Kamailio itself (Dev version) is up and running and normal SIP from normal UAs works. Here's the config I'm using. It's the example in the websocket module directory, just changed to use my db and the default TLS cert that was installed: ---------------------------------------------------------------------- #!KAMAILIO # # Simple/sample kamailio.cfg for running a proxy/registrar with TLS and # WebSockets support. #!substdef "!DBURL!mysql://ser:heslo@localhost/ser!g" #!substdef "!MY_IP_ADDR!192.168.0.148!g" #!substdef "!MY_WS_PORT!80!g" #!substdef "!MY_WSS_PORT!443!g" #!substdef "!MY_WS_ADDR!tcp:MY_IP_ADDR:MY_WS_PORT!g" #!substdef "!MY_WSS_ADDR!tls:MY_IP_ADDR:MY_WSS_PORT!g" ##!define LOCAL_TEST_RUN #!define WITH_TLS #!define WITH_WEBSOCKETS ####### Global Parameters ######### fork=yes children=4 #alias="example.com" #!ifdef WITH_TLS enable_tls=1 #!endif listen=MY_IP_ADDR #!ifdef WITH_WEBSOCKETS listen=MY_WS_ADDR #!ifdef WITH_TLS listen=MY_WSS_ADDR #!endif #!endif tcp_connection_lifetime=3604 tcp_accept_no_cl=yes tcp_rd_buf_size=16384 syn_branch=0 debug=2 # set paths to location of modules (to sources or installation folders) #!ifdef WITH_SRCPATH mpath="modules_k:modules" #!else mpath="/usr/local/kamailio/lib/kamailio/modules_k/:/usr/local/kamailio/lib/kamailio/modules/" #!endif loadmodule "db_mysql.so" loadmodule "tm.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 "auth.so" loadmodule "auth_db.so" loadmodule "kex.so" loadmodule "mi_rpc.so" #!ifdef WITH_TLS loadmodule "tls.so" #!endif #!ifdef WITH_WEBSOCKETS loadmodule "xhttp.so" loadmodule "websocket.so" loadmodule "nathelper.so" #!endif # ----------------- setting module-specific parameters --------------- # ----- 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) # ----- 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) modparam("registrar", "max_expires", 3600) modparam("registrar", "gruu_enabled", 0) # ----- usrloc params ----- modparam("usrloc", "db_url", "DBURL") modparam("usrloc", "db_mode", 0) # ----- auth_db params ----- modparam("auth_db", "db_url", "DBURL") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "load_credentials", "") #!ifdef WITH_TLS # ----- tls params ----- modparam("tls", "config", "/usr/local/kamailio/etc/kamailio/tls.cfg") #!endif #!ifdef WITH_WEBSOCKETS # ----- nathelper params ----- modparam("nathelper|registrar", "received_avp", "$avp(RECEIVED)") # Note: leaving NAT pings turned off here as nathelper is _only_ being used for # WebSocket connections. NAT pings are not needed as WebSockets have # their own keep-alives. #!endif ####### Routing Logic ######## # Main SIP request routing logic # - processing of any incoming SIP request starts with this route # - note: this is the same as route { ... } request_route { # per request initial checks route(REQINIT); #!ifdef WITH_WEBSOCKETS if (nat_uac_test(64)) { # Do NAT traversal stuff for requests from a WebSocket # connection - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. force_rport(); if (is_method("REGISTER")) fix_nated_register(); else { if (!add_contact_alias()) { xlog("L_ERR", "Error aliasing contact <$ct>\n"); sl_send_reply("400", "Bad Request"); exit; } } } #!endif # 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")) record_route(); # handle registrations route(REGISTRAR); if ($rU==$null) { # request with no Username in RURI sl_send_reply("484","Address Incomplete"); exit; } # user location service route(LOCATION); route(RELAY); } route[RELAY] { if (!t_relay()) { sl_reply_error(); } exit; } # Per SIP request initial checks route[REQINIT] { 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()) { #!ifdef WITH_WEBSOCKETS if ($du == "") { if (!handle_ruri_alias()) { xlog("L_ERR", "Bad alias <$ru>\n"); sl_send_reply("400", "Bad Request"); exit; } } #!endif route(RELAY); } else { 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 (!save("location")) sl_reply_error(); exit; } } # USER location service route[LOCATION] { if (!lookup("location")) { $var(rc) = $rc; t_newtran(); switch ($var(rc)) { case -1: case -3: send_reply("404", "Not Found"); exit; case -2: send_reply("405", "Method Not Allowed"); exit; } } } # Authentication route route[AUTH] { 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")) 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; } } #!ifdef WITH_WEBSOCKETS onreply_route { if (nat_uac_test(64)) { # Do NAT traversal stuff for replies to a WebSocket connection # - even if it is not behind a NAT! # This won't be needed in the future if Kamailio and the # WebSocket client support Outbound and Path. add_contact_alias(); } } event_route[xhttp:request] { set_reply_close(); set_reply_no_connect(); if ($Rp != MY_WS_PORT && $Rp != MY_WSS_PORT) { xlog("L_WARN", "HTTP request received on $Rp\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } xlog("L_DBG", "HTTP Request Received\n"); if ($hdr(Upgrade)=~"websocket" && $hdr(Connection)=~"Upgrade" && $rm=~"GET") { xlog("L_DBG", "WebSocket\n"); xlog("L_DBG", " Host: $hdr(Host)\n"); xlog("L_DBG", " Origin: $hdr(Origin)\n"); if ($hdr(Host) == $null || !is_myself($hdr(Host))) { xlog("L_WARN", "Bad host $hdr(Host)\n"); xhttp_reply("403", "Forbidden", "", ""); exit; } # Optional... validate Origin # Optional... perform HTTP authentication # ws_handle_handshake() exits (no further configuration file # processing of the request) when complete. if (ws_handle_handshake()) { # Optional... cache some information abou the # successful connection exit; } } xhttp_reply("404", "Not found", "", ""); } event_route[websocket:closed] { xlog("L_INFO", "WebSocket connection from $si:$sp has closed\n"); } #!endif ---------------------------------------------------------------------- -- Juan Carlos Castro y Castro Instant Solutions - Telefonia Gerando Resultado http://www.instant.com.br Principais capitais: 4063-6100 Demais regiões: (11)4063-6100

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git:3.3: sdpops: Fix memory leakage in w_get_sdp function.
by Vicente Hernando 05 Sep '12

05 Sep '12

Module: sip-router Branch: 3.3 Commit: 4f34d95ebc689702376bfdeb63be65d1e774455a URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=4f34d95… Author: Vicente Hernando <vhernando(a)systemonenoc.com> Committer: Vicente Hernando <vhernando(a)systemonenoc.com> Date: Wed Sep 5 06:44:39 2012 -0400 sdpops: Fix memory leakage in w_get_sdp function. (cherry picked from commit d1d2494c966d65828d1920296056da840a11efd9) --- modules/sdpops/sdpops_mod.c | 8 +------- 1 files changed, 1 insertions(+), 7 deletions(-) diff --git a/modules/sdpops/sdpops_mod.c b/modules/sdpops/sdpops_mod.c index df7d6e6..b70b553 100644 --- a/modules/sdpops/sdpops_mod.c +++ b/modules/sdpops/sdpops_mod.c @@ -974,14 +974,8 @@ static int w_get_sdp(sip_msg_t* msg, char *avp) LM_DBG("No SDP\n"); return -2; } else { - avp_val.s.s = pkg_malloc(sdp->raw_sdp.len); + avp_val.s.s = sdp->raw_sdp.s; avp_val.s.len = sdp->raw_sdp.len; - if (avp_val.s.s == NULL) - { - LM_ERR("Failed to alloc memory for SDP"); - return -1; - } - memcpy(avp_val.s.s, sdp->raw_sdp.s, avp_val.s.len); LM_DBG("Found SDP %.*s\n", sdp->raw_sdp.len, sdp->raw_sdp.s); } if (add_avp(AVP_VAL_STR | avp_type, avp_name, avp_val) != 0)

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git:master: sdpops: Fix memory leakage in w_get_sdp function.
by Vicente Hernando 05 Sep '12

05 Sep '12

Module: sip-router Branch: master Commit: d1d2494c966d65828d1920296056da840a11efd9 URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=d1d2494… Author: Vicente Hernando <vhernando(a)systemonenoc.com> Committer: Vicente Hernando <vhernando(a)systemonenoc.com> Date: Wed Sep 5 06:44:39 2012 -0400 sdpops: Fix memory leakage in w_get_sdp function. --- modules/sdpops/sdpops_mod.c | 8 +------- 1 files changed, 1 insertions(+), 7 deletions(-) diff --git a/modules/sdpops/sdpops_mod.c b/modules/sdpops/sdpops_mod.c index df7d6e6..b70b553 100644 --- a/modules/sdpops/sdpops_mod.c +++ b/modules/sdpops/sdpops_mod.c @@ -974,14 +974,8 @@ static int w_get_sdp(sip_msg_t* msg, char *avp) LM_DBG("No SDP\n"); return -2; } else { - avp_val.s.s = pkg_malloc(sdp->raw_sdp.len); + avp_val.s.s = sdp->raw_sdp.s; avp_val.s.len = sdp->raw_sdp.len; - if (avp_val.s.s == NULL) - { - LM_ERR("Failed to alloc memory for SDP"); - return -1; - } - memcpy(avp_val.s.s, sdp->raw_sdp.s, avp_val.s.len); LM_DBG("Found SDP %.*s\n", sdp->raw_sdp.len, sdp->raw_sdp.s); } if (add_avp(AVP_VAL_STR | avp_type, avp_name, avp_val) != 0)

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git:3.3: dialog(k): fixed the name of event route
by Daniel-Constantin Mierla 05 Sep '12

05 Sep '12

Module: sip-router Branch: 3.3 Commit: e663ddd6e3ff55cd43f1a21561f7c60a94420d36 URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=e663ddd… Author: Daniel-Constantin Mierla <miconda(a)gmail.com> Committer: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Wed Sep 5 08:58:18 2012 +0200 dialog(k): fixed the name of event route - it is 'dialog:failed' instead of 'dialog:failure' - reported by Uri Shacked (cherry picked from commit 163f860ef7c0ed9a720207499d054e7f68b139df) --- modules_k/dialog/README | 8 ++++---- modules_k/dialog/doc/dialog_admin.xml | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/modules_k/dialog/README b/modules_k/dialog/README index 5aa0b76..46997f7 100644 --- a/modules_k/dialog/README +++ b/modules_k/dialog/README @@ -149,7 +149,7 @@ Alex Balashov 11.1. event_route[dialog:start] 11.2. event_route[dialog:end] - 11.3. event_route[dialog:failure] + 11.3. event_route[dialog:failed] 2. Developer Guide @@ -348,7 +348,7 @@ Chapter 1. Admin Guide 11.1. event_route[dialog:start] 11.2. event_route[dialog:end] - 11.3. event_route[dialog:failure] + 11.3. event_route[dialog:failed] 1. Overview @@ -1783,7 +1783,7 @@ dlg_set_property("ka-dst"); 11.1. event_route[dialog:start] 11.2. event_route[dialog:end] - 11.3. event_route[dialog:failure] + 11.3. event_route[dialog:failed] 11.1. event_route[dialog:start] @@ -1793,7 +1793,7 @@ dlg_set_property("ka-dst"); Executed when BYE is processed or dialog timed out. -11.3. event_route[dialog:failure] +11.3. event_route[dialog:failed] Executed when dialog is not completed (+300 reply to INVITE). diff --git a/modules_k/dialog/doc/dialog_admin.xml b/modules_k/dialog/doc/dialog_admin.xml index 760ba4c..a31d5b5 100644 --- a/modules_k/dialog/doc/dialog_admin.xml +++ b/modules_k/dialog/doc/dialog_admin.xml @@ -2316,7 +2316,7 @@ dlg_set_property("ka-dst"); </para> </section> <section> - <title><varname>event_route[dialog:failure]</varname></title> + <title><varname>event_route[dialog:failed]</varname></title> <para> Executed when dialog is not completed (+300 reply to INVITE). </para>

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git:master: dialog(k): fixed the name of event route
by Daniel-Constantin Mierla 05 Sep '12

05 Sep '12

Module: sip-router Branch: master Commit: 163f860ef7c0ed9a720207499d054e7f68b139df URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=163f860… Author: Daniel-Constantin Mierla <miconda(a)gmail.com> Committer: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Wed Sep 5 08:58:18 2012 +0200 dialog(k): fixed the name of event route - it is 'dialog:failed' instead of 'dialog:failure' - reported by Uri Shacked --- modules_k/dialog/README | 8 ++++---- modules_k/dialog/doc/dialog_admin.xml | 2 +- 2 files changed, 5 insertions(+), 5 deletions(-) diff --git a/modules_k/dialog/README b/modules_k/dialog/README index 5aa0b76..46997f7 100644 --- a/modules_k/dialog/README +++ b/modules_k/dialog/README @@ -149,7 +149,7 @@ Alex Balashov 11.1. event_route[dialog:start] 11.2. event_route[dialog:end] - 11.3. event_route[dialog:failure] + 11.3. event_route[dialog:failed] 2. Developer Guide @@ -348,7 +348,7 @@ Chapter 1. Admin Guide 11.1. event_route[dialog:start] 11.2. event_route[dialog:end] - 11.3. event_route[dialog:failure] + 11.3. event_route[dialog:failed] 1. Overview @@ -1783,7 +1783,7 @@ dlg_set_property("ka-dst"); 11.1. event_route[dialog:start] 11.2. event_route[dialog:end] - 11.3. event_route[dialog:failure] + 11.3. event_route[dialog:failed] 11.1. event_route[dialog:start] @@ -1793,7 +1793,7 @@ dlg_set_property("ka-dst"); Executed when BYE is processed or dialog timed out. -11.3. event_route[dialog:failure] +11.3. event_route[dialog:failed] Executed when dialog is not completed (+300 reply to INVITE). diff --git a/modules_k/dialog/doc/dialog_admin.xml b/modules_k/dialog/doc/dialog_admin.xml index 760ba4c..a31d5b5 100644 --- a/modules_k/dialog/doc/dialog_admin.xml +++ b/modules_k/dialog/doc/dialog_admin.xml @@ -2316,7 +2316,7 @@ dlg_set_property("ka-dst"); </para> </section> <section> - <title><varname>event_route[dialog:failure]</varname></title> + <title><varname>event_route[dialog:failed]</varname></title> <para> Executed when dialog is not completed (+300 reply to INVITE). </para>

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mem leak in sdpops
by Vicente Hernando 05 Sep '12

05 Sep '12

Hello, within w_get_sdp function in modules/sdpops/sdpops_mod.c } else { avp_val.s.s = pkg_malloc(sdp->raw_sdp.len); avp_val.s.len = sdp->raw_sdp.len; if (avp_val.s.s == NULL) I do not find anywhere where avp_val.s.s variable is freed. So I attach two possible patches to solve that in two different ways. Kind regards, Vicente.

2 2

git:3.3: parser/sdp: prevent manipulation with freed structure
by Richard Fuchs 04 Sep '12

04 Sep '12

Module: sip-router Branch: 3.3 Commit: 5771e1ed6c7b92ebab9e2da0251315498a5190e1 URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=5771e1e… Author: Michal Karas <largon(a)largon.net> Committer: Richard Fuchs <rfuchs(a)sipwise.com> Date: Thu Jul 12 17:38:15 2012 +0200 parser/sdp: prevent manipulation with freed structure - FS#244 --- parser/sdp/sdp.c | 1 + 1 files changed, 1 insertions(+), 0 deletions(-) diff --git a/parser/sdp/sdp.c b/parser/sdp/sdp.c index db9003a..5521aec 100644 --- a/parser/sdp/sdp.c +++ b/parser/sdp/sdp.c @@ -721,6 +721,7 @@ int parse_sdp(struct sip_msg* _m) if (res != 0) { LM_DBG("free_sdp\n"); free_sdp((sdp_info_t**)(void*)&_m->body); + return res; } /* The whole body is SDP */ ((sdp_info_t*)_m->body)->raw_sdp.s = body.s;

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sr-dev September 2012