sr-users May 2007

sr-users@lists.kamailio.org

164 participants
296 discussions

[Users] Question about forking
by Tim Madorma 18 May '07

18 May '07

Hi, I want to start openser with runsv which requires that the starting process run in the foreground. My problem is that I also want to listen on a couple of different ports. When I set forking = yes, it will listen on multiple ports, but runsv won't work. When I set forking=no, then openser will only listen on the first specified port. Is there any way around this? Can I have the starting process run in the foreground and fork other processes that listen to the ports in the background? Here is the error message: WARNING: no fork mode and more than one listen address found(will use only the the first one) Here are the associated configuration lines: fork=no children=32 # Local IP address/port pairs to listen to listen=65.185.233.55:5061 listen=65.185.233.55:5062 # Alias IP address/port pairs alias=65.185.233.104:5061 alias=65.185.233.104:5062 thanks, Tim

3 14

[Users] How to log the username of caller and callee to acc table?
by Charles Wang 18 May '07

18 May '07

Hi all, I ever used SER and I know it will save the From information such as 101(a)192.168.11.200 to sip_from or from_uri in its acc table. But in my openser, I also setup the acc module. Where is my caller info? I can find a lot of number such as "2806655523" in my from_tag. I wanna store the caller and callee info in my acc table when I make an internal call or an external call. Can anybody kind to help me to solve it? or give me some tips please. PS: I got this openser.cfg from http://www.sipwise.com/wizard. Best regards, Charles ######################################################################## # Parts of my openser.cfg Configuration ######################################################################## loadmodule "acc.so" modparam("acc", "early_media", 0) modparam("acc", "failed_transaction_flag", 24) modparam("acc", "report_ack", 0) modparam("acc", "report_cancels", 0) modparam("acc", "log_flag", 0) modparam("acc", "log_missed_flag", 0) modparam("acc", "log_level", 2) modparam("acc", "db_flag", 25) modparam("acc", "db_missed_flag", 0) modparam("acc", "db_table_acc", "acc") modparam("acc", "db_table_missed_calls", "missed_calls") modparam("acc", "db_url", "mysql://openser:openserrw@localhost /openser") modparam("acc", "acc_method_column", "method") modparam("acc", "acc_callid_column", "callid") modparam("acc", "acc_time_column", "time") modparam("acc", "acc_totag_column", "to_tag") modparam("acc", "acc_from_tag_column", "from_tag") modparam("acc", "detect_direction", 1) modparam("acc", "acc_sip_code_column", "sip_code") modparam("acc", "acc_sip_reason_column", "sip_reason") modparam("acc", "multi_leg_info", "src_leg=$avp(i:901);dst_leg=$avp(i:902)") ######################################################################## # Request route 'main' ######################################################################## route[0] { xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if(msg:len > max_len) { xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("513", "Message Too Big"); exit; } if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("483", "Too Many Hops"); exit; } if(!is_method("REGISTER")) { if(nat_uac_test("19")) { record_route(";nat=yes"); } else { record_route(); } } if(is_method("CANCEL") || is_method("BYE")) { end_media_session(); } if(loose_route()) { if(!has_totag()) { xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if(nat_uac_test("19") || search("^Route:.*;nat=yes")) { fix_nated_contact(); if(!search("^Content-Length:[ ]*0")) { setbflag(6); } } if(is_method("BYE")) { setflag(24); # account failed transactions setflag(25); # account successful transactions } # mark as loose-routed for acc setflag(26); route(11); } if(is_method("REGISTER")) { route(10); } setflag(24); # account failed transactions setflag(25); # account successful transactions if(is_method("INVITE")) { route(12); } if(is_method("CANCEL") || is_method("ACK")) { route(16); } route(17); } ######################################################################## # Request route 'clear-usr-preferences-caller' ######################################################################## route[1] { xlog("L_INFO", "Clear caller preferences - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:caller_cli)/g"); avp_delete("$avp(s:clir)/g"); } ######################################################################## # Request route 'clear-usr-preferences-callee' ######################################################################## route[2] { xlog("L_INFO", "Clear callee preferences - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_delete("$avp(s:callee_fr_inv_timer)/g"); avp_delete("$avp(s:cfu)/g"); } ######################################################################## # Request route 'usr-preferences-caller' ############################################################ ############ route[3] { route(1); xlog("L_INFO", "Load caller preferences for uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # load caller avps avp_db_load("$avp(s:caller_uuid)", "*"); avp_copy("$avp(s:cli)", "$avp(s:caller_cli)/d"); if(is_avp_set("$avp(s:clir)/n") && avp_check("$avp(s:clir)", "eq/i:1")) { # mark for anonymization setflag(28); } } ######################################################################## # Request route 'usr-preferences-callee' ######################################################################## route[4] { xlog("L_INFO", "Load callee preferences for uuid '$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # load callee avps avp_db_load("$avp(s:callee_uuid)", "*"); if(is_avp_set("$avp(s:cfu)/s")) { xlog("L_INFO", "Call-forward-unconditional to '$avp(s:cfu)' found - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(5); route(6); avp_delete("$avp(s:caller_uuid)/g"); avp_copy("$avp(s:callee_uuid)", "$avp(s:caller_uuid)/d"); avp_pushto("$ru", "$avp(s:cfu)"); route(3); route(13); exit; } if(is_avp_set("$avp(s:ringtimeout)/n")) { xlog("L_INFO", "Setting ring timeout to $avp(s:ringtimeout) secs - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_copy("$avp(s:ringtimeout)", "$avp(s:callee_fr_inv_timer)/d"); } } ######################################################################## # Request route 'acc-caller' ############################################################ ############ route[5] { xlog("L_INFO", "Setting acc source-leg for uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_printf("$avp(i:901)", "$avp(s:caller_uuid)|$fU|$fd"); } ######################################################################## # Request route 'acc-callee' ######################################################################## route[6] { xlog("L_INFO", "Setting acc destination-leg for uuid '$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); avp_printf("$avp(i:902)", "$avp(s:callee_uuid)|$rU|$rd"); } ######################################################################## # Request route 'acc-failure' ######################################################################## route[7] { xlog("L_INFO", "Accounting failed request for uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(5); route(6); resetflag(24); acc_db_request("404", "acc"); } ######################################################################## # Request route 'clir' ######################################################################## route[8] { if(isflagset(28) && !isflagset(27)) { setflag(27); xlog("L_INFO", "Anonymize caller - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); uac_replace_from("Anonymous"," sip:anonymous@anonymous.invalid"); if(is_present_hf("Privacy")) { remove_hf("Privacy"); } append_hf("Privacy: id\r\n"); } } ######################################################################## # Request route 'stop-media-proxy' ######################################################################## route[9] { if(isflagset(22)) { end_media_session(); } } ######################################################################## # Request route 'base-route-register' ######################################################################## route[10] { sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("", "0"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Request route 'base-outbound' ######################################################################## route[11] { if(is_present_hf("P-Asserted-Identity")) { remove_hf("P-Asserted-Identity"); } if(is_present_hf("Remote-Party-ID")) { remove_hf("Remote-Party-ID"); } if(is_avp_set("$avp(s:caller_cli)/s")) { if(!isflagset(28)) { xlog("L_INFO", "Set caller CLI '$avp(s:caller_cli)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); append_hf("P-Asserted-Identity: <$avp(s:caller_cli)>\r\n"); } } route(8); if(isbflagset(6) && !isflagset(22)) { setflag(22); fix_nated_contact(); use_media_proxy(); t_on_reply("2"); } else { t_on_reply("1"); } xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!isflagset(21)) { t_on_failure("2"); if(!isflagset(26)) { route(5); route(6); } } if(!t_relay()) { sl_reply_error(); if(is_method("INVITE") && isbflagset(6)) { end_media_session(); } } exit; } ######################################################################## # Request route 'base-route-invite' ######################################################################## route[12] { sl_send_reply("100", "Trying"); if(from_gw()) { $avp(s:caller_uuid) = "0"; xlog("L_INFO", "Call from PSTN' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); setflag(23); } else { if(allow_trusted()) { if(is_avp_set("$avp(s:peer_uuid)/s")) { # use tag-column from trusted-table as uuid for this caller avp_copy("$avp(s:peer_uuid)", "$avp(s:caller_uuid)/d"); } else { # if no uuid is set, use "0" as default uuid $avp(s:caller_uuid) = "0"; } xlog("L_INFO", "Call from trusted peer with uuid '$avp(s:caller_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!is_domain_local("$rd")) { xlog("L_INFO", "Rejecting peering attempt with non-local request domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Relaying Denied"); exit; } setflag(23); } else { if(!proxy_authorize("", "subscriber")) { xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); proxy_challenge("", "0"); exit; } if(!check_from()) { xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed From-URI Detected"); exit; } consume_credentials(); } } route(3); if(nat_uac_test("19")) { setbflag(6); } route(13); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[13] { if(lookup("aliases")) { xlog("L_INFO", "Callee was aliased - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } route(2); if(!is_domain_local("$rd")) { setflag(20); $avp(s:callee_uuid) = "0"; route(15); } avp_delete("$avp(s:callee_uuid)"); avp_db_query("select uuid from subscriber where username = '$rU'", "$avp(s:callee_uuid)"); if(is_avp_set("$avp(s:callee_uuid)/s")) { xlog("L_INFO", "Callee is local, uuid='$avp(s:callee_uuid)' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(14); } else { $avp(s:callee_uuid) = "0"; xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(15); } exit; } ######################################################################## # Request route 'invite-to-internal' ######################################################################## route[14] { route(4); if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); sl_send_reply("404", "User Offline"); } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(11); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[15] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(11); exit; } if(!isflagset(23)) { # don't allow calls relaying from PSTN to PSTN, if not explicitely forwarded if(uri =~ "^sip:0[0-9]+@") { # only route numeric users to PSTN xlog("L_INFO", "Loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!load_gws()) { xlog("L_ERR", "Error loading PSTN gateways - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } if(!next_gw()) { xlog("L_ERR", "No PSTN gateways available - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("503", "PSTN Termination Currently Unavailable"); exit; } setflag(21); t_on_failure("1"); route(11); } } xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); sl_send_reply("404", "User Not Found"); exit; } ######################################################################## # Request route 'base-route-local' ######################################################################## route[16] { t_on_reply("1"); if(t_check_trans()) { xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!t_relay()) { sl_reply_error(); } } else { xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } exit; } ######################################################################## # Request route 'base-route-generic' ######################################################################## route[17] { xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); exit; } # Request route 'base-filter-failover' route[18] { if(!t_check_status("408|500|503")) { xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(9); exit; } } # Reply route 'base-standard-reply' onreply_route[1] { xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); exit; } # Reply route 'base-nat-reply' onreply_route[2] { xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); if(nat_uac_test("1")) { fix_nated_contact(); } if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if(!search("^Content-Length:[ ]*0")) { use_media_proxy(); } } exit; } -- Best Regards Charles

2 2

[Serusers] No Audio
by Arun Kumar 18 May '07

18 May '07

Hi I've configured my ser server with dedicated Media proxy server. When I call from IP to IP it works fine without any problem but when I try to dialout using ser there is no audio. But if I'll enable STUN in my phone I'm able to hear audio. my phones are behind Nat( Port rest con nat). Please guide me how to setup my ser or mediaproxy so I don't have to use STUN server. thanks arun

2 5

[Users] ACK loop after receiving 404 from remote GW
by Taylor Carpenter 18 May '07

18 May '07

I am getting a ACK loop after receiving a 404 from a remote GW. I am starting the call using sipp for testing. sipp -r 1 -l 1 -d 5000 -s 8989 -p 5061 -sn uac 10.14.18.22 The ngrep trace is showing Via and From set to 127.0.0.1 for the communication between OpenSER and the sipp system. The ngrep output and the routes from the openser.cfg are at http://pastebin.ca/493807 I assume the Via/From being set to 127.0.0.1 may be the issue, which I assume I could change from sipp, but I would also like OpenSER to catch the loop. Taylor

2 1

[Users] Openser 1.2 Presence Module
by Curt Moore 18 May '07

18 May '07

Hello all. I've been experimenting with the new presence module in openser-1.2 and have a few questions which hopefully someone can help answer. I'm using an SVN checkout of the 1.2 branch, r2237, and am testing with Polycom SoundPoint IP phones, firmware 2.1.1.0052. >From what I can tell, openser 1.2 does not fully support the xpidf content type, it only supoorts the winfo content type. I've looked through the code for the presence module extensively and it appears that whenever a SUBSCRIBE comes in where the Event header is simply "Event: presence" and the Accept header is "Accept: application/xpidf+xml", the message is acceped and a 202 is sent in reply but in any NOTIFY sent from openser as a result, there is no message body and the message does not contain the XML information for pidf. Just as a test, I changed the event field in active_watchers to "presence.winfo" and noticed that with this change, openser generates a NOTIFY with a message body which contains watcherinfo. This, of course, generates a 481 from the phone since it does not know about the "Event: presence.winfo" event type. Also as a test, I tried to use the pa module as I'd had success with this module in the past since it _does_ support xpidf. In 1.2 it appears that the database structure specified in the databse setup scripts does not agree with the table structure being used in the module. As a result, I could not get the pa module to work either. Am I overlooking some small detail? Please let me know if further information is needed in order to help diagnose and resolve this issue. Cheers, -Curt

2 3

[Users] Failure_route help
by Yago Bargalló 18 May '07

18 May '07

Hello, I'm new in Openser and now I'm trying to use a failure_route but it doesn't work and I don't know why. The version that I'm using is Openser 1.2. What I'm trying is to forward a call after fr_inv_timer to Asterisk but it doesn't work. This is my openser.cfg : (...) modparam("tm", "fr_inv_timer" , 8) # ------------------------- request routing logic ------------------- # main routing logic route{ (...) if (uri==myself) { if (method=="REGISTER") { if (!www_authorize("172.16.51.150", "subscriber")) { www_challenge("172.16.51.150", "0"); exit; }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; if(is_method("INVITE") && (uri=~"sip:2[0-9]+@")) { t_on_failure("1"); } # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; append_hf("P-hint: usrloc applied\r\n"); }; route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; exit; } failure_route[1] { #redirect to Asterisk rewritehostport("172.16.51.160:5060"); rewriteuser("2002"); }

3 2

[Users] Compiling Openser 1.2.0 on Solaris
by Lesley Liu 18 May '07

18 May '07

Hi Users, I tried to compile openser-1.2.0-tls on Solaris 10. The gcc, bison, flex, make, libiconv packages are added. However, when command "make all", the message shows the following without further information at beginning. When command "make install", it only shows the first 4 lines as following. Please advise how to solve it. Thanks a lot. ; # pwd /opt/openser-1.2.0-tls # make all yacc -d -b cfg cfg.y conflicts: 1 shift/reduce yacc -d -b cfg cfg.y conflicts: 1 shift/reduce yacc -d -b cfg cfg.y conflicts: 1 shift/reduce yacc -d -b cfg cfg.y conflicts: 1 shift/reduce Compiling action.c gcc -g -O9 -funroll-loops -Wall -mcpu=ultrasparc -mtune=ultrasparc -DNAME='"openser"' -DVERSION='"1.2.0-notls"' -DARCH='"sparc64"' -DOS='"solaris"' -DCOMPILER='"gcc 3.4.6"' -D__CPU_sparc64 -D__OS_solaris -D__SMP_no -DCFG_DIR='"/usr/local/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -DHAVE_DEVPOLL -DHAVE_SELECT -c action.c -o action.o Compiling blacklists.c .... ; Regards, Lesley

2 1

[Users] Compiling Openser 1.2.0 on Solaris
by Lesley Liu 18 May '07

18 May '07

1 0

[Users] Call forward when no answer
by Howard Tang 18 May '07

18 May '07

Hi All, I have followed a tutorial and set up Asterisk as a voice mail server. http://www.voip-info.org/wiki/view/Realtime+Integration+Of+Asterisk+With+Op… It works fine when the UA is offline. Now, I want a call forwarded to the Voice mail server when there is no answer from the UA after 60 seconds(UA is registered on the openser). What should I do? Below is my config (copy from the above link). # requests for Media server if(is_method("INVITE") && !has_totag() && uri=~"sip:\*9") { route(3); exit; } # mark transaction if user is in voicemail group if(is_method("INVITE") && !has_totag() && is_user_in("Request-URI","voicemail")) { xdbg("user [$ru] has voicemail redirection enabled\n"); # backup R-URI avp_write("$ruri", "i:10"); setflag(2); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { if(isflagset(2)) { # route to Asterisk Media Server prefix("1"); rewritehostport("10.10.10.11:5060"); route(1); } else { sl_send_reply("404", "Not Found"); exit; } }; # voicemail access # - *98 - listen caller's voice messages, being prompted for pin # - *981 - listen voice messages, being promted for mailbox and pin # - *98XXXX - leave voice message to XXXX # route[3] { # direct voicemail if (uri =~ "sip:\*98@" ) { rewriteuser("1"); xdbg("voicemail access\n"); } else if (uri =~ "sip:\*981@" ) { strip(4); rewriteuser("11"); } else if (uri =~ "sip:\*98.+@" ) { strip(3); prefix("1"); } else { xlog("unknown media extension $rU\n"); sl_send_reply("404", "Unknown media service"); exit; } # route to Asterisk Media Server rewritehostport("10.10.10.11:5060"); route(1); } failure_route[1] { if (t_was_cancelled()) { xdbg("transaction was cancelled by UAC\n"); return; } # restore initial uri avp_pushto("$ruri", "i:10"); prefix("1"); # route to Asterisk Media Server rewritehostport("10.10.10.11:5060"); resetflag(2); route(1); }

3 6

RE: AW: [Users] OpenSER crashed with Segmentation Fault using PresenceModule
by Schumann Sebastian 17 May '07

17 May '07

Thank you both very much for this hints, I will change the versions I currently use immediately. Regards Sebastian -----Original Message----- From: Anca-Maria Vamanu [mailto:anca@voice-system.ro] Sent: Thursday, May 17, 2007 5:55 PM To: Ovidiu Sas Cc: Schumann Sebastian; users(a)openser.org Subject: Re: AW: [Users] OpenSER crashed with Segmentation Fault using PresenceModule I was just about to tell you this myself. Please take the svn version, either 1.2 or the trunk. Ovidiu Sas wrote: > Hi Sebastian, > > You should use the 1.2 svn branch (or maybe the trunk). There are > several fixes already present in the tree. There's no point testing > against the original 1.2 tarball. > > > Regards, > Ovidiu Sas

1 0

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sr-users May 2007