sr-users August 2007

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169 participants
287 discussions

[OpenSER-Users] Failure Route - 408 Timeout & Asterisk
by Kelvin Williams 13 Aug '07

13 Aug '07

I would like to advise my users as to the reason the phone stopped ringing (no, I'm not going to let it ring for 20 minutes as many of them wish that I would). My grand design is to send the call to Asterisk which plays a recording to the effect of, "the party you have dialed does not answer, please try your call later" after the 2 minute timeout. Upon 408 the call is sent to Asterisk, however there is no audio. The call stays alive until Asterisk hits the Hangup() in the dialplan. I've tested by adding extra audio (to make sure the call does stay up) and watching on the Asterisk console. In other situations, audio passes fine between the Asterisk and the endpoint (ie. Calling voicemail). Any advice would be greatly appreciated. Some relevant portions of the config: loadmodule "tm.so" modparam("tm", "fr_timer", 3) modparam("tm", "fr_inv_timer", 120) modparam("tm", "wt_timer", 10) modparam("tm", "fr_inv_timer_avp", "$avp(s:timeout)") route[ROUTE_TO_PSTN] { .SNIP. (RURI fixes, etc.) rewritehostport("PSTN_IP:PSTN_PORT"); if (!is_method("BYE")) { t_on_reply("ROUTE_TO_PSTN_ONREPLY"); t_on_failure("ROUTE_TO_PSTN_FAILURE"); if (isflagset(FLAG_DEBUG)) xlog("L_INFO", "Relaying to PSTN - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci - On Reply, On Failure Set\n"); }; if (!t_relay()) { if (isflagset(FLAG_DEBUG)) xlog("L_INFO", "Relaying to PSTN - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci - Sending Reply Error\n"); sl_reply_error(); }; exit; } failure_route[ROUTE_TO_PSTN_FAILURE] { .SNIP. (SQL, etc for updating the CDR) if (t_check_status("408")) { rewriteuri("sip:ANC-408@ANS_IP:ANS_PORT"); append_branch(); t_relay(); exit; }; exit; }

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[OpenSER-Users] remote IP address
by Victor Gamov 13 Aug '07

13 Aug '07

Hi All! Is it possible to know IP-address to which request will be forwarded ? I see that t_relay() actually know it but how get these knowledge in script? any pseudo-variable available for it? -- CU, Victor Gamov

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[OpenSER-Users] Dispatcher round robin....
by Francisco Ivandic 13 Aug '07

13 Aug '07

Hi All, I'm using dispatcher succesfully, with round robin alg (option 4), and also use ds_mark_dst and ds_next_dst to select other proxy in case of failure. It balances great when all servers are active, but when there are inactive servers, round robin alg selects the next in the round, even if it's marked as inactive. This results in going through ds_next_dst, and overloads the first active server in the dispatcher file until completing the round. Then, other active servers are called, and when passing to a group of inactive servers, starts going through the first active (as much times as inactive servers there are), and again completing the round. Does anyone know a way to implement a round robin aware of inactive servers? Thanks in advance, Francisco Ivandic

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[OpenSER-Users] subs_status is still active after unsubscription, is it right?
by KevinKinnan 13 Aug '07

13 Aug '07

Thank your Anca for subscribe tip!Notify request with terminated subscription status is got when unsubscribing. But the subs_status in watchers table is still active, is that right? Regards,Kevin _________________________________________________________________ 通过 Windows Live Spaces 与朋友轻松共享您的生活。 http://spaces.live.com/spacesapi.aspx?wx_action=create&wx_url=/friends.aspx…

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[OpenSER-Users] trunk version 2592: NOTIFY problem
by KevinKinnan 13 Aug '07

13 Aug '07

SUBSCRIBE request can not save in watchers table properly. Both p_user and w_user are subscriber's name!? In this case, they are both 'alice'. The SUBSCRIBE request is shown as follows:----------------------------------------SUBSCRIBE sip:alice@148.88.232.101 SIP/2.0Call-ID: c14d16bf8f758655b09588b764c3f0c9(a)148.88.232.101CSeq: 20 SUBSCRIBEFrom: "alice" <sip:alice@148.88.232.101:5061>;tag=z9hG4bK41052461To: "bob" <sip:bob@148.88.232.101:5062>Via: SIP/2.0/TCP 148.88.232.101:5061;branch=z9hG4bK57563Max-Forwards: 70Expires: 360000Event: presenceRoute: <sip:148.88.233.209:5065;lr>Contact: "alice" <sip:alice@148.88.232.101:5061>Content-Length: 0----------------------------------------HELP!Regards,Kevin _________________________________________________________________ 通过 Live.com 查看资讯、娱乐信息和您关心的其他信息！ http://www.live.com/getstarted.aspx

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[Serusers] preload_udomain: bad callid
by Steffen Waldmann 13 Aug '07

13 Aug '07

hi there, when i restart my ser i get the following error: 0(0) preload_udomain: ERROR: bad callid record in table location 0(0) preload_udomain: ERROR: for username 1000, contact sip:1000@192.168.2.117:30316;rinstance=80910b76433a8887 0(0) preload_udomain: ERROR: skipping... 0(0) preload_udomain: ERROR: bad callid record in table location 0(0) preload_udomain: ERROR: for username 1001, contact sip:1001@192.168.2.220:5060;transport=udp 0(0) preload_udomain: ERROR: skipping... ser doesnt start up again... what could be the problem?

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[OpenSER-Users] About openser+asterisk voicemail
by zze-WU Yin RD-MAPS-LAN 13 Aug '07

13 Aug '07

Hello everybody, I have a question aboute the configuration of openser. I want to add asterisk voicemail, but I know that asterisk use the database 'openser' in the mysql. But my system use a openser+freeradius+mysql and I don't use a database 'openser' but 'radius'. So I want to know if it is possible to do a voice mail in my system? Any advice is welcom. Thank you very much!!!

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[OpenSER-Users] trunk version 2592: NOTIFY problem
by KevinKinnan 13 Aug '07

13 Aug '07

BTW, My sip client is programmed by Jain-NIST 1.2 _________________________________________________________________ 探索 Windows Vista 的世界 http://search.live.com/results.aspx?q=windows+vista&mkt=zh-cn&FORM=LIVSOP

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[OpenSER-Users] trunk version 2592: NOTIFY problem
by KevinKinnan 13 Aug '07

13 Aug '07

Hi,I've upgraded to the latest trunk version, and still got this failure. Finally, I sort it out. The 1.2.0 version allows the Route Header without Lr parameter, but the latest version does not. If there(in the latest trunk) is no Lr parameter in the Route Header, the username column in 'presentity' table will be null, while the PUBLISH message is not different from the one in 1.2.0 version.Regards,Kevin _________________________________________________________________ 中国十大热门旅游景点 http://search.msn.com/results.aspx?q=%E4%B8%AD%E5%9B%BD%E5%8D%81%E5%A4%A7%E…

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[OpenSER-Users] FreeBSD Bug - mi_xmlrpc
by Jorge Guntanis 13 Aug '07

13 Aug '07

OS: FreeBSD openser 6.2-RELEASE FreeBSD 6.2-RELEASE #0: Fri Jan 12 10:40:27 UTC 2007 root@dessler.cse.buffalo.edu:/usr/obj/usr/src/sys/GENERIC i386 OpenSer Version root@openser# openser -V version: openser 1.3.0-dev10-notls (i386/freebsd) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 poll method support: poll, select, kqueue. svnrevision: 2:2588M @(#) $Id: main.c 2578 2007-08-07 15:17:07Z bogdan_iancu $ main.c compiled on 01:33:27 Aug 10 2007 with gcc 3.4.6 also tried with release 2327 having the exact same result. Dependencies Version: libxml2-2.6.29 xmlrpc-c-0.9.10_2 libwww-5.4.0_4 Config File: # Uncomment this if you want to use SQL database loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "mi_xmlrpc.so" loadmodule "presence.so" loadmodule "presence_xml.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" Result: root@openser# openser ERROR: load_module: could not open module </usr/local/lib/openser/modules/mi_xmlrpc.so>: /usr/local/lib/libxmlrpc_client.so.7: Undefined symbol "HTThroughLine" parse error (53,13-14): failed to load module ERROR: bad config file (1 errors) Has anyone seen this before, any idea on how to fix this? Any help is appreciated. Thanks Jorge

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sr-users August 2007