sr-users May 2007

sr-users@lists.kamailio.org

164 participants
295 discussions

AW: [Users] OpenSER crashed with Segmentation Fault using PresenceModule

by Schumann Sebastian

Hello, here is the stderr console output (just the last lines which should be important as I saw there is a message size limit), and the gdb backtrace. Do you see any hint for the crash? Best regards Sebastian ________________________________ Von: Anca-Maria Vamanu [mailto:anca@voice-system.ro] Gesendet: Do 5/17/2007 16:57 An: Schumann Sebastian Cc: users(a)openser.org Betreff: Re: [Users] OpenSER crashed with Segmentation Fault using PresenceModule Hello, Please post the log and the gdb backtrace. regards, Anca Vamanu Schumann Sebastian wrote: >Dear all > >I am currently writing my final thesis and want to set up a Presence Core therefore. I am using OpenSER as Presence Agent (with Registrar function or now with separat Registrar, both configuration has the problem). It crashes after a certain time and certain amount of watcher entries. I cannot replicate this or deliminate to a certain action but it always happends, no matter if I clean up the tables, restart the machine or whatever. > >My configuration is one XCAP Server (XCAP Lite 1.0 from ag-projects, working properly), one Registrar (OpenSER with MySQL DB, which forwards SUBSCRIBE/PUBLISH to PA) and Presence Agent (OpenSER with Presence Module, MySQL DB, XMLRPC Module) and Eyebeam 1.5 Clients. All runs on VMware virtual machines. > >I have just three clients that try to subscribe each other. Without any Subscriptions everything works properly, XML data gets stored, PUBLISH is processed etc. Subscribing one user by another is possible and works properly. > >After allowing a watcher to subscription (using XCAP) and if the watcher logs in next time, the state in the watcher table changes from pending to active. It happens now and then, when a user goes online and one of his subscribed presentities is not online, that the OpenSER PA crashes. > >Analyzing DEBUG info and backtrace was not successful, because I found no information or hints about it, just the message segmentation fault appears. > >Does anyone have this problem as well or has anybody an idea about it? I can provide you logs, the dump or traces if you need it. > >Thanks for your help. > >Sebastian > > >

17 years, 6 months

[Users] OpenSER crashed with Segmentation Fault using Presence Module

by Schumann Sebastian

Dear all I am currently writing my final thesis and want to set up a Presence Core therefore. I am using OpenSER as Presence Agent (with Registrar function or now with separat Registrar, both configuration has the problem). It crashes after a certain time and certain amount of watcher entries. I cannot replicate this or deliminate to a certain action but it always happends, no matter if I clean up the tables, restart the machine or whatever. My configuration is one XCAP Server (XCAP Lite 1.0 from ag-projects, working properly), one Registrar (OpenSER with MySQL DB, which forwards SUBSCRIBE/PUBLISH to PA) and Presence Agent (OpenSER with Presence Module, MySQL DB, XMLRPC Module) and Eyebeam 1.5 Clients. All runs on VMware virtual machines. I have just three clients that try to subscribe each other. Without any Subscriptions everything works properly, XML data gets stored, PUBLISH is processed etc. Subscribing one user by another is possible and works properly. After allowing a watcher to subscription (using XCAP) and if the watcher logs in next time, the state in the watcher table changes from pending to active. It happens now and then, when a user goes online and one of his subscribed presentities is not online, that the OpenSER PA crashes. Analyzing DEBUG info and backtrace was not successful, because I found no information or hints about it, just the message segmentation fault appears. Does anyone have this problem as well or has anybody an idea about it? I can provide you logs, the dump or traces if you need it. Thanks for your help. Sebastian -- Sebastian Schumann Diploma Student Architecture and Design Department Slovak Telekom, a. s. T-Com, Operations Unit Production and Service Division Innovations and Enterprise Solution Subunit Address: Namestie Slobody 6, 817 62 Bratislava, Slovakia Office: Room 232, Jarabinkova 1, 821 09 Bratislava, Slovakia +421 910 643010 (mobile) sebastian.schumann(a)t-com.sk http://www.t-com.sk

17 years, 6 months

[Users] failure_route

by Yago Bargalló

Hello all, I don't know why my openser.cfg doesn't work correctly, I want my openser to catch the failure_route after 10 seconds to reroute a call to a Asterisk when nobody responds the phone on an openser. Does anybody know what I'm doing wrong?? Thanks in advance!!!!!!!!! Openser.cfg: modparam("tm","fr_timer",4) modparam("tm", "fr_inv_timer" ,10) # ------------------------- request routing logic ------------------- # main routing logic (...) route{ (...) # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication 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(uri=~"sip:2[0-9]+@") { t_on_failure("1"); # rewritehostport("172.16.51.160:5060"); # t_relay(); # rewriteuser("2002"); # exit; } # 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] { #reroute to asterisk rewritehost("172.16.51.150"); rewriteuser("2002"); t_relay(); exit; }

17 years, 6 months

[Serusers] 407 Proxy Authentication Required to a subscribed user

by flavio

Hi all, I'm testing Blind Trasfer with my IP Phones, SER and GW, in this scenario: AnalogPhone (GW) is the Transferee IP Phone1, registred to SER, is the Transferor IP Phone 2, registred to SER, is the Transfer Target When the Transferor send INVITE to put on hold the Trasferee, SER reply with 407 Proxy Authentication Requrire, as follow reported: 2007/05/17 12:45:49.370808 10.28.19.124:5060 -> 10.28.19.202:5060 INVITE sip:06XXXXXX15@10.28.19.230 SIP/2.0. Via: SIP/2.0/UDP 10.28.19.124;branch=z9hG4bKc442076492fe9954. Route: <sip:10.28.19.202;ftag=1c1272914830;lr=on>. From: <sip:06XXXXXX14@10.28.19.202>;tag=9bb7a2279efa8707. To: <sip:06XXXXXX15@10.28.19.230>;tag=1c1272914830. Contact: <sip:06XXXXXX14@10.28.19.124;user=phone>. Supported: replaces. Call-ID: 127291448221200023114(a)10.28.19.230. CSeq: 21593 INVITE. User-Agent: Grandstream BT110 1.0.8.12. Max-Forwards: 70. Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE. Content-Type: application/sdp. Content-Length: 206. . v=0. o=0660522014 8000 8001 IN IP4 10.28.19.124. s=SIP Call. c=IN IP4 0.0.0.0. t=0 0. m=audio 5004 RTP/AVP 8 18 0. a=sendonly. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729/8000. a=rtpmap:0 PCMU/8000. a=ptime:20. # U 2007/05/17 12:45:49.371204 10.28.19.202:5060 -> 10.28.19.124:5060 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 10.28.19.124;branch=z9hG4bKc442076492fe9954. From: <sip:06XXXXXX14@10.28.19.202>;tag=9bb7a2279efa8707. To: <sip:06XXXXXX15@10.28.19.230>;tag=1c1272914830. Call-ID: 127291448221200023114(a)10.28.19.230. CSeq: 21593 INVITE. Proxy-Authenticate: Digest realm="10.28.19.202", nonce="464c4fa95f251088d01666190944e53313bab22e". Server: Sip EXpress router (0.9.6 (i386/linux)). Content-Length: 0. Warning: 392 10.28.19.202:5060 "Noisy feedback tells: pid=15940 req_src_ip=10.28.19.124 req_src_port=5060 in_uri=sip:0660522015@10.28.19.230 out_uri=sip:0660522015@10.28.19.230 via_cnt==1". My question is: why SER reply with an authentication request to INVITE, if the Transferor party is a registred user? Have you any suggestion about? I verify user subscribtion in my ser.cfg file as follow: if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); } Thanks for support, Flavio -- ******************************** * (o< ing. Patria Flavio * //\ phone 0823451358 * V_/_ mobile 3407873357 * ********************************

17 years, 6 months

[Serusers] looped requests, syn_branch==1 => branch value is repeated for 2 Ts

by JF

Hi all, I have observed when looping a request through SER (using t_relay_to("127.0.0.1", 5060)) that sometimes the via branch parameter value generated when the request goes out is kept equal to the value in the previous via (when this request was looped), and two equal vias go out. This causes problems with on_reply routes. On_reply routes are then called only once, instead of two. Since these are 2 independent transactions, this should happen twice. Browsing through tm code, it seems that this is caused by the usage of syn_branch == 1 in main.c (the default), leading to the following function calls when an initial INVITE loops: tm/t_fwd.c: t_forward_nonack add_uac print_uac_request tm/t_msgbuilder.c: t_calc_branch msg_translator.c: branch_builder <-- here, when syn_branch==1, the transaction 'label' value is used for the branch value, but according to tm/h_table.h, this value is "sequence number within hash collision slot". Is this the reason why some times the branch value is the same in two transactions looped through SER? The label value can perfectly be the same across 2 transactions not in the same hash slot... right? Maybe this has nothing to do with the request being looped, or it just makes this situation more probable? Anyone else can replicate this behaviour? Thanks, João Filipe Plácido -- PT Inovação

17 years, 6 months

[Users] No answer when my openser 1.2.0 make a call to asterisk

by Charles Wang

Hi all, I try to make a call from my Openser 1.2.0 (SIP Proxy) to the asterisk in different machine. I use my asterisk as a trunking gateway. I can make a call from my openser to some trunking gateway such as my cisco 5300 or welltech 5250. In the same method, I try to make a call to asterisk ( sip listen on udp 5060 ) I use ngrep on my asterisk machine and list as below. But I can't find any sip debug in my asterisk CLI. In the message on my openser, I can see as below: openser[16801]: route[11] Base Outbound - M=INVITE RURI= sip:02123456789@asterisk.ip.addr:5060;transport=udp F= sip:101@openser.domain.name T=sip:00862123456789@openser.domain.name IP= 61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 But the To URI is still 00862123456789(a)openser.domain.name not 02123456789(a)asterisk.ip.addr. Is it correct? Does anybody kind to help me to solve it or give me some tips please? Best regards, Charles ##################### message on my Openser ################################# openser[16802]: New request - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16802]: route[12] Base Route Invite - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16802]: Proxy authentication failed - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: New request - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[12] Base Route Invite - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[3] Usr Preferences Caller - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[1] Clear Usr Preferences Caller - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: Clear caller preferences - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: Load caller preferences for uuid '1' - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[13] Invite Find Callee - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[2] Clear Usr Preferences Callee - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: Clear callee preferences - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: Callee is not local - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[15] Invite To External - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: Loading PSTN gateways - M=INVITE RURI= sip:00862123456789@openser.domain.name F=sip:101@openser.domain.name T= sip:00862123456789@openser.domain.name IP=61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[11] Base Outbound - M=INVITE RURI= sip:02123456789@asterisk.ip.addr:5060;transport=udp F= sip:101@openser.domain.name T=sip:00862123456789@openser.domain.name IP= 61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 openser[16801]: route[8] Clir - M=INVITE RURI= sip:02123456789@asterisk.ip.addr:5060;transport=udp F= sip:101@openser.domain.name T=sip:00862123456789@openser.domain.name IP= 61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 localhost mediaproxy[1905]: command request 25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 192.168.11.9:57538:audio 61.217.xxx.xxx openser.domain.name local asterisk.ip.addr remote X-PRO=20build=201082 info=from:101@openser.domain.name, to:00862123456789@openser.domain.name,fromtag:1111281943,totag: localhost mediaproxy[1905]: session 25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9: started. listening on 60.250.132.204:35076 localhost mediaproxy[1905]: command execution time: 3.04 ms openser[16801]: Request leaving server - M=INVITE RURI= sip:02123456789@asterisk.ip.addr:5060;transport=udp F= sip:101@openser.domain.name T=sip:00862123456789@openser.domain.name IP= 61.217.xxx.xxx ID=25C0CC42-016D-4FEC-9B60-A0D1076A8816(a)192.168.11.9 ################# my asterisk CLI ######################## [root@asterisk ~]# asterisk -rvvvvvvvvvvvvvv == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found == Binding iaxusers to mysql/asterisk/iaxfriends == Binding iaxpeers to mysql/asterisk/iaxfriends == Binding queues to mysql/asterisk/queue_table == Binding queue_members to mysql/asterisk/queue_member_table Asterisk 1.2.16, Copyright (C) 1999 - 2006 Digium, Inc. and others. Created by Mark Spencer <markster(a)digium.com> Asterisk comes with ABSOLUTELY NO WARRANTY; type 'show warranty' for details. This is free software, with components licensed under the GNU General Public License version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'show license' for details. ========================================================================= Connected to Asterisk 1.2.16 currently running on asterisk (pid = 26311) Verbosity is at least 14 -- Remote UNIX connection asterisk*CLI> sip debug SIP Debugging re-enabled asterisk*CLI> ######### my command running on asterisk machine: "ngrep -t -W byline -d any port 5060" #################### interface: any filter: (ip) and ( port 5060 ) # U 2007/05/17 13:31:35.908163 my.openser.ip.addr:5060 -> my.asterisk.ip.addr :5060 INVITE sip:03939749001@my.asterisk.ip.addr:5060;transport=udp SIP/2.0. Record-Route: <sip:my.openser.ip.addr;lr;ftag=3840196923;nat=yes>. Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0. Via: SIP/2.0/UDP 192.168.11.9:57536 ;received=61.217.xxx.xxx ;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4. From: 101 <sip:101@my.openser.domain.name>;tag=3840196923. To: < sip:0028863939749001@my.openser.domain.name>. Contact: <sip:101@61.217.xxx.xxx:57536>. Call-ID: 18875244-8D15-416B-92B7-DD24DA4630E2(a)192.168.11.9 . CSeq: 4807 INVITE. Max-Forwards: 69. Content-Type: application/sdp. User-Agent: X-LITE build 1082. Content-Length: 321. . v=0. o=101 45727796 45727796 IN IP4 192.168.11.9 . s=X-LITE. c=IN IP4 my.openser.ip.addr. t=0 0. m=audio 35066 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 2007/05/17 13:31:36.325713 my.openser.ip.addr:5060 -> my.asterisk.ip.addr :5060 INVITE sip:03939749001@my.asterisk.ip.addr:5060;transport=udp SIP/2.0. Record-Route: <sip:my.openser.ip.addr;lr;ftag=3840196923;nat=yes>. Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0. Via: SIP/2.0/UDP 192.168.11.9:57536;received=61.217.xxx.xxx;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4. From: 101 <sip:101@my.openser.domain.name>;tag=3840196923. To: <sip:0028863939749001@my.openser.domain.name >. Contact: <sip:101@61.217.xxx.xxx:57536>. Call-ID: 18875244-8D15-416B-92B7-DD24DA4630E2(a)192.168.11.9. CSeq: 4807 INVITE. Max-Forwards: 69. Content-Type: application/sdp. User-Agent: X-LITE build 1082. Content-Length: 321. . v=0. o=101 45727796 45727796 IN IP4 192.168.11.9. s=X-LITE. c=IN IP4 my.openser.ip.addr . t=0 0. m=audio 35066 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 2007/05/17 13:31:37.325722 my.openser.ip.addr:5060 -> my.asterisk.ip.addr :5060 INVITE sip:03939749001@my.asterisk.ip.addr:5060;transport=udp SIP/2.0. Record-Route: <sip:my.openser.ip.addr;lr;ftag=3840196923;nat=yes>. Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0. Via: SIP/2.0/UDP 192.168.11.9:57536;received=61.217.xxx.xxx;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4. From: 101 <sip:101@my.openser.domain.name>;tag=3840196923. To: <sip:0028863939749001@my.openser.domain.name >. Contact: <sip:101@61.217.xxx.xxx:57536>. Call-ID: 18875244-8D15-416B-92B7-DD24DA4630E2(a)192.168.11.9. CSeq: 4807 INVITE. Max-Forwards: 69. Content-Type: application/sdp. User-Agent: X-LITE build 1082. Content-Length: 321. . v=0. o=101 45727796 45727796 IN IP4 192.168.11.9. s=X-LITE. c=IN IP4 my.openser.ip.addr . t=0 0. m=audio 35066 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 2007/05/17 13:31:39.325425 my.openser.ip.addr:5060 -> my.asterisk.ip.addr :5060 INVITE sip:03939749001@my.asterisk.ip.addr:5060;transport=udp SIP/2.0. Record-Route: <sip:my.openser.ip.addr;lr;ftag=3840196923;nat=yes>. Via: SIP/2.0/UDP my.openser.ip.addr;branch=z9hG4bKab01.ae06a6e5.0. Via: SIP/2.0/UDP 192.168.11.9:57536;received=61.217.xxx.xxx;rport=57536;branch=z9hG4bK834BA777F3C7439EBBC7C4DAECC52FD4. From: 101 <sip:101@my.openser.domain.name>;tag=3840196923. To: <sip:0028863939749001@my.openser.domain.name >. Contact: <sip:101@61.217.xxx.xxx:57536>. Call-ID: 18875244-8D15-416B-92B7-DD24DA4630E2(a)192.168.11.9. CSeq: 4807 INVITE. Max-Forwards: 69. Content-Type: application/sdp. User-Agent: X-LITE build 1082. Content-Length: 321. . v=0. o=101 45727796 45727796 IN IP4 192.168.11.9. s=X-LITE. c=IN IP4 my.openser.ip.addr . t=0 0. m=audio 35066 RTP/AVP 0 8 3 18 98 97 101. a=rtpmap:0 pcmu/8000. a=rtpmap:8 pcma/8000. a=rtpmap:3 gsm/8000. a=rtpmap:18 G729/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:97 speex/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. -- Best Regards Charles -- Best Regards Charles

17 years, 6 months

[Serusers] Problem running presence modules - SER won't start

by Mahatma Sarasvati

Hello All, Running SER 2.0.0 rc1 from CVS (updated yesterday) on CentOS 5. I'm trying to test/use the presence functionality in SER. I've compiled and installed with make group_include="standard standard-dep" all I've set the environment variable: export LD_LIBRARY_PATH=/usr/local/lib/ser [root@vx1 modules]# echo $LD_LIBRARY_PATH /usr/local/lib/ser My supporting modules are here: [root@vx1 modules]# ls -l /usr/local/lib/ser/ total 828 -rwxr-xr-x 1 root root 40768 May 14 15:19 libbinrpc.so -rwxr-xr-x 1 root root 179044 May 14 15:19 lib_ser_cds.so -rwxr-xr-x 1 root root 213028 May 14 15:19 lib_ser_presence.so -rwxr-xr-x 1 root root 371056 May 14 15:19 lib_ser_xcap.so drwxr-xr-x 2 root root 4096 May 14 15:19 modules [root@vx1 modules]# Load module commands are in ser.cfg: loadmodule "/usr/local/lib/ser/modules/xmlrpc.so" loadmodule "/usr/local/lib/ser/modules/xcap.so" loadmodule "/usr/local/lib/ser/modules/pa.so" loadmodule "/usr/local/lib/ser/modules/presence_b2b.so" loadmodule "/usr/local/lib/ser/modules/rls.so" loadmodule "/usr/local/lib/ser/modules/dialog.so" libXML2 and libXML2-devel are installed. Yet, when I try to start SER I get this: May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/pa.so>: lib_ser_xcap.so: cannot open shared object file: No such file or directory May 15 13:09:24 vx1 ser: parse error (81,13-14): failed to load module May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/presence_b2b.so>: lib_ser_xcap.so: cannot open shared object file: No such file or directory May 15 13:09:24 vx1 ser: parse error (82,13-14): failed to load module May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/rls.so>: lib_ser_xcap.so: cannot open shared object file: No such file or directory May 15 13:09:24 vx1 ser: parse error (83,13-14): failed to load module May 15 13:09:24 vx1 ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/dialog.so>: lib_ser_cds.so: cannot open shared object file: No such file or directory Can anyone help? I think I've done everything right. Why won't SER start? Thanks, M Here's my modules directory: [root@vx1 modules]# ls -l total 7264 -rwxr-xr-x 1 root root 67000 May 14 15:19 acc_db.so -rwxr-xr-x 1 root root 73248 May 14 15:19 acc_syslog.so -rwxr-xr-x 1 root root 69946 May 14 15:19 auth_db.so -rwxr-xr-x 1 root root 58071 May 14 15:19 auth.so -rwxr-xr-x 1 root root 37345 May 14 15:19 avp_db.so -rwxr-xr-x 1 root root 168768 May 14 15:19 avpops.so -rwxr-xr-x 1 root root 79110 May 14 15:19 avp.so -rwxr-xr-x 1 root root 279385 May 14 15:19 ctl.so -rwxr-xr-x 1 root root 72896 May 14 15:19 db_ops.so -rwxr-xr-x 1 root root 218639 May 14 15:18 dbtext.so -rwxr-xr-x 1 root root 97287 May 14 15:19 dialog.so -rwxr-xr-x 1 root root 63003 May 14 15:19 dispatcher.so -rwxr-xr-x 1 root root 21604 May 14 15:19 diversion.so -rwxr-xr-x 1 root root 114536 May 14 15:19 domain.so -rwxr-xr-x 1 root root 45704 May 14 15:19 enum.so -rwxr-xr-x 1 root root 109624 May 14 15:19 eval.so -rwxr-xr-x 1 root root 84357 May 14 15:19 exec.so -rwxr-xr-x 1 root root 81398 May 14 15:19 fifo.so -rwxr-xr-x 1 root root 53442 May 14 15:19 flatstore.so -rwxr-xr-x 1 root root 43414 May 14 15:19 gflags.so -rwxr-xr-x 1 root root 87574 May 14 15:19 lcr.so -rwxr-xr-x 1 root root 35883 May 14 15:19 maxfwd.so -rwxr-xr-x 1 root root 112635 May 14 15:19 mediaproxy.so -rwxr-xr-x 1 root root 87485 May 14 15:19 msilo.so -rwxr-xr-x 1 root root 106696 May 14 15:19 mysql.so -rwxr-xr-x 1 root root 187822 May 14 15:19 nathelper.so -rwxr-xr-x 1 root root 21203 May 14 15:19 options.so -rwxr-xr-x 1 root root 741640 May 14 15:19 pa.so -rwxr-xr-x 1 root root 131893 May 14 15:19 pdt.so -rwxr-xr-x 1 root root 272113 May 14 15:19 permissions.so -rwxr-xr-x 1 root root 122158 May 14 15:19 pike.so -rwxr-xr-x 1 root root 232133 May 14 15:19 presence_b2b.so -rwxr-xr-x 1 root root 26172 May 14 15:19 print.so -rwxr-xr-x 1 root root 47597 May 14 15:19 ratelimit.so -rwxr-xr-x 1 root root 152843 May 14 15:19 registrar.so -rwxr-xr-x 1 root root 402652 May 14 15:19 rls.so -rwxr-xr-x 1 root root 114896 May 14 15:19 rr.so -rwxr-xr-x 1 root root 63276 May 14 15:19 sanity.so -rwxr-xr-x 1 root root 66942 May 14 15:19 sl.so -rwxr-xr-x 1 root root 222033 May 14 15:18 sms.so -rwxr-xr-x 1 root root 34415 May 14 15:19 speeddial.so -rwxr-xr-x 1 root root 120807 May 14 15:19 textops.so -rwxr-xr-x 1 root root 43276 May 14 15:19 timer.so -rwxr-xr-x 1 root root 848962 May 14 15:19 tm.so -rwxr-xr-x 1 root root 117593 May 14 15:19 uac.so -rwxr-xr-x 1 root root 38793 May 14 15:19 unixsock.so -rwxr-xr-x 1 root root 38979 May 14 15:19 uri_db.so -rwxr-xr-x 1 root root 38603 May 14 15:19 uri.so -rwxr-xr-x 1 root root 351468 May 14 15:19 usrloc.so -rwxr-xr-x 1 root root 41867 May 14 15:19 xcap.so -rwxr-xr-x 1 root root 100354 May 14 15:19 xlog.so -rwxr-xr-x 1 root root 98456 May 14 15:19 xmlrpc.so [root@vx1 modules]# ____________________________________________________________________________________Boardwalk for $500? In 2007? Ha! Play Monopoly Here and Now (it's updated for today's economy) at Yahoo! Games. http://get.games.yahoo.com/proddesc?gamekey=monopolyherenow

17 years, 6 months

[Users] Help (handle status)

by Jeferson Prevedello

Hi guys, I able the control of simultaneous users with the function [max_contacts], but whenever the user is blocked the softphone (X-lite) show the following message: [Registration error: 503 - Service Unavailable]. I tried to adapt the message conform code below, but not resolve: if (t_check_status("503")) { sl_send_reply("503","my new reply"); exit; } Any sugestion ? Regards Jeferson

17 years, 6 months

[Users] changing log level of certain messages

by Dimo

Hi guys, I was wondering if it is possible to change the log level of some messages, namely: ERROR:auth_radius:radius_authorize_sterman: rc_auth failed rc_send_server: no reply from RADIUS server xxxx.:1812 I have alot of clutter from this one and i would like to lower its severity so it does not get logged, but i still need other messages from radius to be logged. Apart from going into the code and changing the log level do you know of any other way to do it? Best. Dimo

17 years, 6 months

[Users] Mediaproxy

by Mhmd

Greeting , I have downloaded Media Proxy , but unfortunately I could not find mediaproxy.iso in you package , in you readme , you have explain to how I install it , you mentioned to mediaproxy.iso file which is missing in the package , can you please urgently advise where from can I get the full mediaproxy server . waiting your reply a.s.a.p . Best Regards Mohammad Network Engineer GCS Amman, Jordan

17 years, 6 months

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sr-users May 2007