sr-dev August 2012

sr-dev@lists.kamailio.org

30 participants
169 discussions

Using t_suspend()/t_continue() multiple times on the same transaction

by Peter Dunkley

Hi, I am trying to use t_suspend()/t_continue() multiple times on the same transaction. Calling t_suspend() more than once works, but the second time I call t_continue() the transaction is killed and a 500 response is sent. It is the "if (branch == t->nr_of_outgoings)" check from the code fragment below (from t_suspend.c:t_continue()) that results in the transaction being killed - you can see the debug/error line I added to determine this in the fragment. Is using t_suspend()/t_continue() multiple times something that should work? Thanks, Peter if (t->uas.status < 200) { /* No final reply has been sent yet. * Check whether or not there is any pending branch. */ for ( branch = 0; branch < t->nr_of_outgoings; branch++ ) { if ((t->uac[branch].request.buffer != NULL) && (t->uac[branch].last_received < 200) ) break; } if (branch == t->nr_of_outgoings) { /* There is not any open branch so there is * no chance that a final response will be received. */ ret = 0; LM_ERR("branch == t->nr_of_outgoings\n"); goto kill_trans; } } -- Peter Dunkley Technical Director Crocodile RCS Ltd

8 years, 8 months

default config: do loose_route() for in-dialog NOTIFY as per new RFC 6665

by Iñaki Baz Castillo

Hi, the new RFC 6665 mandates that the subscription dialog is made *after* the first NOTIFY from the UAS (rather than after the 200 to the initial SUBSCRIBE). This means that the UAC MUST take the route set from the *first* received NOTIFY (instead of taking it from the 200 to the initial SUBSCRIBE). Therefore a proxy compliant with RFC 6665 MUST always add RR to in-dialog NOTIFY requests and thus, I propose that the default SR script file implements it. Regards. -- Iñaki Baz Castillo <ibc(a)aliax.net>

9 years, 1 month

git:master: modules_k/uac: uac_replace_from/to AUTO mode with dialog module

by Anca Vamanu

Module: sip-router Branch: master Commit: 0f702f6e236eb0cbb238bf83a0c4ae94d7b3cad8 URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=0f702f6… Author: Anca Vamanu <anca.vamanu(a)1and1.ro> Committer: Anca Vamanu <anca.vamanu(a)1and1.ro> Date: Thu Jul 19 17:49:13 2012 +0300 modules_k/uac: uac_replace_from/to AUTO mode with dialog module Added a implementation for uac_replace_from/to() that uses the dialog module for AUTO mode. In this mode the URIs are stored as dialog variables. The change in tm module fixes a bug: if uac_replace_to() was called, the URI was not changed accordingly in Cancel. --- modules/tm/t_cancel.c | 5 +- modules_k/dialog/dialog.c | 2 + modules_k/dialog/dlg_load.h | 6 + modules_k/uac/README | 111 +++++++++++----- modules_k/uac/doc/uac_admin.xml | 68 +++++++++- modules_k/uac/replace.c | 292 ++++++++++++++++++++++++++++++--------- modules_k/uac/uac.c | 17 ++- 7 files changed, 393 insertions(+), 108 deletions(-) Diff: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commitdiff;h=0f7…

9 years, 9 months

™ delete_timer obsolete?

by Olle E. Johansson

According to the 3.3 docs, this function in the ™ module is no longer needed. Is this correct? If so, should someone remove it? Hmm. All mentions of ™ gets automatically converted to a trademark sign in my mailer. I mean the T M Module :-) /O -------- 1.4.6. delete_timer (integer) Time after which a to-be-deleted transaction currently ref-ed by a process will be tried to be deleted again (in milliseconds). Note: this parameter is obsolete for ser 2.1 (in 2.1 the transaction is deleted the moment it's not referenced anymore). Default value is 200 milliseconds. Example 6. Set delete_timer parameter ... modparam("tm", "delete_timer", 100) ...

10 years, 5 months

kamailio core at qm_status

by Jijo

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: ",

10 years, 5 months

Websocket testing

by Carlos Ruiz Díaz

Hello, I have a few questions about the websocket module which I have successfully installed but I wasn't able to make calls through it: 1. After setting up the proxy ip:port in the call.htm file (of sipml5) to * 127.0.0.1:5060* the client started to work but kamailio script refused to establish my connection because the following condition was not satisfied: *if ($Rp != MY_WS_PORT && $Rp != MY_WSS_PORT) {* * xlog("L_WARN", "HTTP request received on $Rp\n");* * xhttp_reply("403", "Forbidden", "", "");* * exit;* *}* *MY_WS_PORT* and *MY_WSS_PORT *are set to 80 and 443 respectively, as the default config example of websocket module says so. Then, I decided to change the ip:port to *127.0.0.1:80*, always in the call.htm file and afterwards the condition was satisfied but sipml5 dies with SIP stack start: proxy='127.0.0.1:80', realm='<sip:127.0.0.1>', impi='2000', impu='<sip:2000@127.0.0.1>' tsk_utils.js:97<http://127.0.0.1/sipml5/src/tinySAK/src/tsk_utils.js?svn=5> Connecting to 'ws://127.0.0.1:80' tsk_utils.js:97<http://127.0.0.1/sipml5/src/tinySAK/src/tsk_utils.js?svn=5> Stack starting tsk_utils.js:97<http://127.0.0.1/sipml5/src/tinySAK/src/tsk_utils.js?svn=5> Unexpected response code: 200 :1 <http://127.0.0.1/> __tsip_transport_ws_onerror tsk_utils.js:97<http://127.0.0.1/sipml5/src/tinySAK/src/tsk_utils.js?svn=5> __tsip_transport_ws_onclose tsk_utils.js:97<http://127.0.0.1/sipml5/src/tinySAK/src/tsk_utils.js?svn=5> Failed to connet to the server Finally, I ended up commenting the condition block and restored the original values of ip:port to *127.0.0.1:5060* . Having done that, I tried again and another error was thrown but this time, in the next condition block: * if ($hdr(Host) == $null || !is_myself($hdr(Host))) * <script>: WebSocket Aug 8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <script>: Host: 127.0.0.1:5060 Aug 8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <script>: Origin: http://127.0.0.1 Aug 8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <core> [socket_info.c:589]: grep_sock_info - checking if host==us: 14==9 && [ 127.0.0.1:5060] == [127.0.0.1] Aug 8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <core> [socket_info.c:589]: grep_sock_info - checking if host==us: 14==9 && [ 127.0.0.1:5060] == [127.0.0.2] Aug 8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <core> [socket_info.c:589]: grep_sock_info - checking if host==us: 14==13 && [ 127.0.0.1:5060] == [192.168.10.95] Aug 8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <core> [socket_info.c:589]: grep_sock_info - checking if host==us: 14==13 && [ 127.0.0.1:5060] == [192.168.10.55] Aug 8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <core> [socket_info.c:589]: grep_sock_info - checking if host==us: 14==9 && [ 127.0.0.1:5060] == [127.0.0.1] Aug 8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <core> [socket_info.c:589]: grep_sock_info - checking if host==us: 14==9 && [ 127.0.0.1:5060] == [127.0.0.2] Aug 8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <core> [socket_info.c:589]: grep_sock_info - checking if host==us: 14==13 && [ 127.0.0.1:5060] == [192.168.10.95] Aug 8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <core> [socket_info.c:589]: grep_sock_info - checking if host==us: 14==13 && [ 127.0.0.1:5060] == [192.168.10.55] Aug 8 11:30:53 carlosrdcnx-laptop kamailio[16238]: DEBUG: <core> [forward.c:462]: *check_self: host != me* Aug 8 11:30:53 carlosrdcnx-laptop kamailio[16238]: WARNING: <script>: Bad host 127.0.0.1:5060 I commented the block too and only then sipml5 was able to register itself. What am I doing wrong here? 2. I registered a legacy softphone (twinkle) to attempt to initiate a call in both ways, but the was something wrong with the signaling, probably some frame decoding garbage in the buffer of the SIP message. Perhaps these bytes are part of the frame control header but since I haven't read the RFC (yet) I am mentioning it anyway. tcp_send: buf=*#012�~#003�*INVITE sip:2000@df7jal23ls0d.invalid;transport=ws SIP/2.0#015#012Record-Route: <sip:127.0.0.1;transport=ws;r2=on;lr=on>#015#012Record-Route: <sip:127.0.0.1;r2=on;lr=on>#015#012Via: SIP/2.0/WS 127.0.0.1;branch=z9hG4bK90a8.b1a7035e13ed19880dd12a1f4c86adbb.0#015#012Via: SIP/2.0/UDP 127.0.0.1:5062;rport=5062;branch=z9hG4bKimixlbyp#015#012Max-Forwards: 69#015#012To: <sip:2000@127.0.0.1>#015#012From: "1000" <sip:1000@127.0.0.1>;tag=lrtfz#015#012Call-ID: gxsqobolphfchfq(a)carlosrdcnx-laptop.site#015#012CSeq: 654 INVITE#015#012Contact: <sip:1000@127.0.0.1:5062>#015#012Content-Type: application/sdp#015#012Allow: INVITE,ACK,BYE,CANCEL,OPTIONS,PRACK,REFER,NOTIFY,SUBSCRIBE,INFO,MESSAGE#015#012Supported: replaces,norefersub,100rel#015#012User-Agent: Twinkle/1.4.2#015#012Content-Length: 302#015#012#015#012v=0#015#012o=twinkle 391470222 1383232165 IN IP4 127.0.0.1#015#012s=-#015#012c=IN IP4 127.0.0.1#015#012t=0 0#015#012m=audio 8008 RTP/AVP 98 97 8 0 3 101#015#012a=rtpmap:98 speex/16000#015#012a=rtpmap:97 speex/8000#015#012a=rtpmap:8 PCMA/8000#015#012a=rtpmap:0 PCMU/8000#015#012a=rtpmap:3 GSM/8000#015#012a=rtpmap:101 telephone-event/8000#015#012a=fmtp:101 0-15#015#012a=ptime:20#015#012 3. Does Twinkle support the minimum media requirements for testing? If not, what (Linux) softphone is suitable for this purpose? Thanks and sorry for the long email. Carlos.

10 years, 6 months

Testing the Websocket module with sipml5.org

by Juan Castro

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

10 years, 6 months

mem leak in sdpops

by Vicente Hernando

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.

10 years, 6 months

git:master: auth: print return code in log when nonce is invalid

by Daniel-Constantin Mierla

Module: sip-router Branch: master Commit: 494b383edde7a2d193c220f3117506e4cc95932f URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=494b383… Author: Daniel-Constantin Mierla <miconda(a)gmail.com> Committer: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Fri Aug 31 10:29:45 2012 +0200 auth: print return code in log when nonce is invalid - it will give a clue about why nonce is considered invalid --- modules/auth/api.c | 2 +- 1 files changed, 1 insertions(+), 1 deletions(-) diff --git a/modules/auth/api.c b/modules/auth/api.c index 2ee39a7..c79de5c 100644 --- a/modules/auth/api.c +++ b/modules/auth/api.c @@ -145,7 +145,7 @@ static int auth_check_hdr_md5(struct sip_msg* msg, auth_body_t* auth, *auth_res = NONCE_REUSED; return 0; } else { - DBG("auth:pre_auth: Invalid nonce value received\n"); + DBG("auth:pre_auth: Invalid nonce value received (ret %d)\n", ret); *auth_res = NOT_AUTHENTICATED; return 0; }

10 years, 6 months

git:master: core: removed append_branch() from cfg language structure

by Daniel-Constantin Mierla

Module: sip-router Branch: master Commit: 41918bca6d6a99f6773a72ab132e75c940ecd79a URL: http://git.sip-router.org/cgi-bin/gitweb.cgi/sip-router/?a=commit;h=41918bc… Author: Daniel-Constantin Mierla <miconda(a)gmail.com> Committer: Daniel-Constantin Mierla <miconda(a)gmail.com> Date: Thu Aug 30 22:36:10 2012 +0200 core: removed append_branch() from cfg language structure - cfg append_branch() is no longer implemented by core, being moved to corex module in order to allow variables in the parameters --- cfg.lex | 3 --- cfg.y | 23 ----------------------- 2 files changed, 0 insertions(+), 26 deletions(-) diff --git a/cfg.lex b/cfg.lex index 76eb2f1..f986354 100644 --- a/cfg.lex +++ b/cfg.lex @@ -236,7 +236,6 @@ PREFIX "prefix" STRIP "strip" STRIP_TAIL "strip_tail" SET_USERPHONE "userphone" -APPEND_BRANCH "append_branch" REMOVE_BRANCH "remove_branch" CLEAR_BRANCHES "clear_branches" IF "if" @@ -641,8 +640,6 @@ IMPORTFILE "import_file" <INITIAL>{PREFIX} { count(); yylval.strval=yytext; return PREFIX; } <INITIAL>{STRIP} { count(); yylval.strval=yytext; return STRIP; } <INITIAL>{STRIP_TAIL} { count(); yylval.strval=yytext; return STRIP_TAIL; } -<INITIAL>{APPEND_BRANCH} { count(); yylval.strval=yytext; - return APPEND_BRANCH; } <INITIAL>{REMOVE_BRANCH} { count(); yylval.strval=yytext; return REMOVE_BRANCH; } <INITIAL>{CLEAR_BRANCHES} { count(); yylval.strval=yytext; diff --git a/cfg.y b/cfg.y index e094223..0ae0e40 100644 --- a/cfg.y +++ b/cfg.y @@ -3289,29 +3289,6 @@ cmd: | STRIP LPAREN error RPAREN { $$=0; yyerror("bad argument, number expected"); } | SET_USERPHONE LPAREN RPAREN { $$=mk_action(SET_USERPHONE_T, 0); set_cfg_pos($$); } | SET_USERPHONE error { $$=0; yyerror("missing '(' or ')' ?"); } - | APPEND_BRANCH LPAREN STRING COMMA STRING RPAREN { - qvalue_t q; - if (str2q(&q, $5, strlen($5)) < 0) { - yyerror("bad argument, q value expected"); - } - $$=mk_action(APPEND_BRANCH_T, 2, STRING_ST, $3, NUMBER_ST, (void *)(long)q); - set_cfg_pos($$); - } - | APPEND_BRANCH LPAREN STRING RPAREN { - $$=mk_action(APPEND_BRANCH_T, 2, STRING_ST, $3, - NUMBER_ST, (void *)Q_UNSPECIFIED); - set_cfg_pos($$); - } - | APPEND_BRANCH LPAREN RPAREN { - $$=mk_action(APPEND_BRANCH_T, 2, STRING_ST, 0, - NUMBER_ST, (void *)Q_UNSPECIFIED); - set_cfg_pos($$); - } - | APPEND_BRANCH { - $$=mk_action(APPEND_BRANCH_T, 2, STRING_ST, 0, - NUMBER_ST, (void *)Q_UNSPECIFIED); - set_cfg_pos($$); - } | REMOVE_BRANCH LPAREN intno RPAREN { $$=mk_action(REMOVE_BRANCH_T, 1, NUMBER_ST, (void*)$3); set_cfg_pos($$);

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sr-dev August 2012