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11 participants
37462 discussions

[OpenSER-Users] LCR Failover
by Douglas Garstang 23 Mar '08

23 Mar '08

I'd like to think I'm not stupid, but I've been dealing with OpenSER for 2-3 years now and it still frustrates me just as much as it did the very first time I looked at it. Can LCR do failover? I have this below. I can see that one of my gateways is selected, seemingly at random, and then when it doesn't reply (because I used a bogus IP), it doesn't time-out after 3s and go to the next gw. Is this possible? How do I do it? Btw... can anyone recommend a book on OpenSER, since the documentation is so lacking? modparam("tm", "fr_inv_timer_avp", "$avp(i:705)") route{ $avp(i:705) = "3"; ... if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); sl_send_reply("100","Trying"); if (!load_gws()) { sl_send_reply("500", "Internal server error, unable to load gateways"); exit; } if(!next_gw()){ sl_send_reply("503", "Service not available, no gateways found"); exit; }; route(1); } } route[1] { if (is_method("INVITE")) { t_on_branch("2"); t_on_reply("2"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); }; exit; } ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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[Serusers] How do you force SDP c= ipaddr to be a specific value that isn't either rtpproxy interface IP?
by Frank Durda IV 23 Mar '08

23 Mar '08

I have a problem with SER 2.0.0-RC1 and rtpproxy that needs some help. I have a PSTN gateway config and a SER box all in a 10.x.x.x address space. Directly connected to two interfaces on the connect to a Cisco router, which performs 1:1 NAT conversion to provide public Internet IPs to those interfaces, one for SIP and one for RTP. (They are separate for bandwidth, debugging and access-list reasons.) Another two interfaces on the SER connect to the PSTN switch, which has separate ports for SIP and RTP traffic. Across the Internet or direct connections to the Cisco router are from Asterisk boxes or similar devices that proxy all SIP/RTP data from their SIP phones. It looks like this: Asterisk SIP SIP SIP Server +-----+ +-+ +------+ 66.1.2.3--internet--208.4.5.6|Cisco|----10.1.1.2|S|10.2.1.1--.2|PSTN | | 1:1 | |E| |Switch| 208.4.5.7| NAT |----10.5.1.2|R|10.6.1.1--.2|TDM | +-----+ +-+ +------+ RTP RTP RTP 208.4.5.6 is converted to 10.1.1.2 and vice versa 208.4.5.7 is converted to 10.5.1.2 and vice versa (Cisco command) ip nat inside source static 10.1.1.2 208.4.5.6 ip nat inside source static 10.5.1.2 208.4.5.7 rtpproxy is in use, and has 10.5.1.2(em5) (toward Cisco) and 10.6.1.1(em4) (toward PSTN switch). SER listens on 10.1.1.2(em1) and 10.2.1.1(em0). Ser.cfg takes care of the small amount of SIP-related tweaking that must be done to make SIP work: if (dst_ip==10.1.1.2) { #Received on em1 force_send_socket("10.2.1.1"); #emit on em0 rewritehost("10.2.1.2"); #Send to PSTN Switch $gw_ip = "10.2.1.2"; #Send to PSTN Switch When the INVITE message finally reaches the PSTN switch, it says: U 2008/03/23 04:05:55.780069 10.2.1.1:5060 -> 10.2.1.2:5060 INVITE sip:4097679102@10.2.1.2 SIP/2.0. Via: SIP/2.0/UDP 10.2.1.1;branch=z9hG4bK7572.3545b131.0. Via: SIP/2.0/UDP 66.1.2.3:5060;branch=z9hG4bK39eee695;rport=5060. From: "VOIP TEST 1000" <sip:4697271700@66.1.2.3>;tag=as571000e7. To: <sip:4097679102@208.4.5.6>. Contact: <sip:4697271700@66.1.2.3>. Call-ID: 2bf456fe3016fd6b62a1649c7cb6cd75(a)66.1.2.3. CSeq: 102 INVITE. User-Agent: Asterisk PBX. Max-Forwards: 16. Date: Sun, 23 Mar 2008 04:05:50 GMT. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Supported: replaces. Content-Type: application/sdp. Content-Length: 490. . v=0. o=root 55522 55522 IN IP4 66.1.2.3. s=session. c=IN IP4 10.6.1.1. t=0 0. m=audio 35010 RTP/AVP 0 3 8 112 5 10 7 97 111 101. a=rtpmap:0 PCMU/8000. a=rtpmap:3 GSM/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:112 AAL2-G726-32/8000. a=rtpmap:5 DVI4/8000. a=rtpmap:10 L16/8000. a=rtpmap:7 LPC/8000. a=rtpmap:97 iLBC/8000. a=fmtp:97 mode=30. a=rtpmap:111 G726-32/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-16. a=silenceSupp:off - - - -. a=ptime:20. a=sendrecv. So the invite sent to the PSTN looks wonderful, correctly pointing SIP and RTP back to the SER box. The problem is on the Cisco side of the SER box, and shows up when the 183/200 messages are emitted by the SER. SER/rtpproxy rewrite the c=IN IP4 line to point to 10.5.1.2, which would be fine except that I need the public IP address 208.4.5.7 to appear in this position. I have tried using force_rtp_proxy("flags","208.4.5.7") in the reply_route section, and I know it is getting executed, but nothing changes. I've tried a variety of flag combinations, but obviously haven't tried all 720 combinations (although it feels like I have). I also tried fix_nated_sdp("2") by itself, with an empty force_rtp_proxy() and a force_rtp_proxy("","208.4.5.7") and all that did was create a "none of the above value" for c= of 10.6.1.110.5.1.2, pasting the two IP addresses together from the rtpproxy command line, and ignoring the value right value I'm trying to provide. So, the question is: How do you say to SER "Here is the value that should be used for c= on this call"? How do I set this? For complex reasons eliminated here to make the example simple, extending the public IP address over to the em5 interface so it is public already is not practical. I need a fix that changes the SDP c= value to one I will know based on whom the distant asterisk server (or equivalent) device will be. There may be other fields that also need fixing for RFC compliancy, but c= being wrong is the current problem. Thanks for suggestions.

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[OpenSER-Users] unixODBC and SQLite???
by Douglas Garstang 21 Mar '08

21 Mar '08

/etc/odbc.ini: [sqlite1] Description = SQLite Driver = SQLite Database = /root/test.db Timeout = 9000 Trace = Yes TraceFile = /tmp/sqlite.log /etc/odbcinst.ini: [SQLite] Description = SQLite ODBC Driver Driver = /usr/local/lib/libsqlite3odbc.so Setup = /usr/local/lib/libsqlite3odbc.so openser.cfg: modparam("avpops", "db_url", "unixodbc://localhost/sqlite1") What's the correct format for specifying the DSN when using SQLite with ODBC? SQLite doesn't use a user/password or host. I tried the above, "unixodbc://localhost/sqlite1" and openser barfs with: ERROR:unixodbc:new_connection: failed to connect I also tried: modparam("avpops", "db_url", "unixodbc://sqlite1") and openser barfed with: new_db_id: Error while parsing database URL: unixodbc://sqlite1 What's the correct format? Is it even supported in Openser 1.2.0? Thanks, Doug ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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[OpenSER-Users] AVP_TABLE.. Why is it required?
by Douglas Garstang 21 Mar '08

21 Mar '08

I put this in openser.cfg: modparam("avpops","avp_url","mysql://unixodbc:passwd@host/database") When I start openser it complains and then EXITS. ERROR:avpops_init: "AVP_DB" present but "AVP_TABLE" found empty Why do I need to specify an avp_table? I plan to use avp_db_query() to retrieve values from another table. That's if using avp_db_query against an odbc source is even possible. People say it is, but given that it's not documented, I'm not sure. Doug. ____________________________________________________________________________________ Be a better friend, newshound, and know-it-all with Yahoo! Mobile. Try it now. http://mobile.yahoo.com/;_ylt=Ahu06i62sR8HDtDypao8Wcj9tAcJ

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[OpenSER-Users] AVPops avp_url???
by Douglas Garstang 21 Mar '08

21 Mar '08

I'm trying to use avpops with unixodbc. What's the format of the avp_uri for unixodbc? The documentation has: modparam("avpops","avp_url","mysql://user:passwd@host/database") What format do I use? Is it this? Is it something else? modparam("avpops","avp_url","unixodbc://user:passwd@host/database") Why isn't this documented? Doug ____________________________________________________________________________________ Looking for last minute shopping deals? Find them fast with Yahoo! Search. http://tools.search.yahoo.com/newsearch/category.php?category=shopping

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[OpenSER-Users] O'SER behind http Proxy which requires username and Password
by luzango mfupe 21 Mar '08

21 Mar '08

Hi Mates, IM having my OpenSER in the university campus where all the communication have to go through a http Proxy which always asks for user name and password. Where in the O'SER config file should i place these credentials ( http://user:pass@proxyname:port";) Do i have to do the same in the Mediaproxy config file as well? WBR, LU.

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[OpenSER-Users] UA Get's Disconnected Once a call is initiated
by Ali Jawad 21 Mar '08

21 Mar '08

Dear All I am using Openser with TLS, the client connects successfully however once I press the call button the connection to the client get's closed "Actually the client stays open however I do check with netstat and the connection is not there anymore". As for the openser server The following is noted in the logs Mar 18 15:13:56 [1396] DBG:core:handle_ser_child: read response= b613c8d8, 1, fd -1 from 5 (1378) Mar 18 15:13:56 [1378] DBG:core:tcp_send: after receive_fd: c= 0xb613c8d8 n=4 fd=14 Mar 18 15:13:56 [1378] DBG:core:tcp_send: sending... Mar 18 15:13:56 [1378] DBG:core:tls_update_fd: New fd is 14 Mar 18 15:13:56 [1378] DBG:core:tls_write: write was successful (580 bytes) Mar 18 15:13:56 [1378] DBG:core:tcp_send: after write: c= 0xb613c8d8 n=580 fd=14 Mar 18 15:13:56 [1378] DBG:core:tcp_send: buf= SIP/2.0 408 Request Timeout Via: SIP/2.0/TLS 192.168.0.149:9071;rport=9071;branch=z9hG4bK30231106nh26;received=my.own .public.ip From: 9613041705 <sip:9613041705@server.ip>;tag=30231106 To: 9613041705 <sip:9613041705@server.ip>;tag=22c184cf1a1beaaaab06669b2c48f35d-e4d8 Call-ID: F96E5BE097AAtYP4(a)server.ip CSeq: 2 REGISTER Server: OpenSER (1.3.0-tls (i386/linux)) Content-Length: 0 Warning: 392 xx.yy.yy.yy:443 "Noisy feedback tells: pid=1378 req_src_ip=xx.xx.xx.xx req_src_port=9071 in_uri=sip:87.236.144.13 out_uri=sip:pstn.gw.ip.here:5060 via_cnt==0" Mar 18 15:13:56 [1378] DBG:tm:relay_reply: sent buf=0x816aed8: SIP/2.0 4..., shmem=0xb614fb38: SIP/2.0 4 Mar 18 15:13:56 [1378] DBG:tm:final_response_handler: done Mar 18 15:13:56 [1385] DBG:core:tls_update_fd: New fd is 20 Mar 18 15:13:56 [1385] DBG:core:_tls_read: connection closed cleanly Mar 18 15:13:56 [1385] DBG:core:tcp_read_req: EOF Mar 18 15:13:56 [1385] DBG:core:io_watch_del: io_watch_del (0x813d8c0, 20, -1, 0x10) fd_no=2 called Mar 18 15:13:56 [1385] DBG:core:release_tcpconn: releasing con 0xb613c8d8, state -1, fd=20, id=12 Mar 18 15:13:56 [1385] DBG:core:release_tcpconn: extra_data 0xb614c9f0 Mar 18 15:13:56 [1396] DBG:core:handle_tcp_child: reader response= b613c8d8, -1 from 0 Mar 18 15:13:56 [1396] DBG:core:tcpconn_destroy: destroying connection 0xb613c8d8, flags 0002 Mar 18 15:13:56 [1396] DBG:core:tls_close: closing SSL connection Mar 18 15:13:56 [1396] DBG:core:tls_update_fd: New fd is 24 Mar 18 15:13:56 [1396] DBG:core:tls_shutdown: shutdown successful Mar 18 15:13:56 [1396] DBG:core:tls_tcpconn_clean: entered Mar 18 15:13:58 [1378] DBG:tm:utimer_routine: timer routine:7,tl=0xb615f734 next=(nil), timeout=3056900000 Mar 18 15:14:01 [1378] DBG:tm:timer_routine: timer routine:2,tl=0xb615f630 next=(nil), timeout=3060 Mar 18 15:14:01 [1378] DBG:tm:wait_handler: removing 0xb615f5e8 from table Mar 18 15:14:01 [1378] DBG:tm:delete_cell: delete transaction 0xb615f5e8 Mar 18 15:14:01 [1378] DBG:tm:wait_handler: done Mar 18 15:14:41 [1378] DBG:presence:msg_watchers_clean: cleaning pending subscriptions Mar 18 15:14:41 [1378] DBG:core:db_free_rows: freeing 0 rows Mar 18 15:14:41 [1378] DBG:presence:msg_presentity_clean: cleaning expired presentity information Mar 18 15:14:41 [1378] DBG:core:db_free_rows: freeing 0 rows With Regards Ali Jawad System Administrator Splendor Telecom (www.splendor.net) Beirut, Lebanon Phone: +961 1 373725 Fax: + 961 1 375554

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Re: [OpenSER-Users] How configure Voice mail for multi users.
by Mark Sayer 21 Mar '08

21 Mar '08

Asif - Despite what the documentation says, I'd drop the context portion of your command. i.e. exten => s-NOANSWER,n,VoiceMail(${EXTEN}) Since you are clearly going to VM for different reasons you can also add an optional switch that will play a different vm-message. Check the docs for more info. Mark At 03:04 a.m. 20/03/2008, you wrote: >Hi All, >i want to configure voice mail on Asterisk 1.4 for multiple users. let >me explain you the scenario. > >i have 10 users with the name of >1000,2000,3000,4000,5000,6000,.......and these user can call to each >other. Now i want to configure separate voice mail box for separate >user. > >my extensions.conf ..... settings below.. >[voicemail] >exten => _X.,1,Dial(SIP/${EXTEN}) >exten => _X.,n,NoOp(Dial Status: ${DIALSTATUS}) >exten => _X.,n,Goto(s-${DIALSTATUS},1) ...........

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[OpenSER-Users] new CDRTool release 6.3.0
by Adrian Georgescu 21 Mar '08

21 Mar '08

The software can be downloaded from: http://download.dns-hosting.info/CDRTool/ Changelog since 6.1.0: cdrtool (6.3.0) unstable; urgency=low * Boost the speed of the rating engine from 100% to 1000% depending on the complexity of the rating tables * Fixed set of delimiter from global.inc setting when exporting csv files * Added GetEntityProfiles command to rating engine * Return json encoded for GetPrepaidHistory * Added script to create partial billing_rates tables * Moved $CDRTool['rating'] settings to $RatingEngine in global.inc * Added $RatingEngine['split_rating_table'] option to query rating data in multiple tables, each table name is autogenerated from the billing_rates.name field * Fixed set of timeout for each SOAP connection * Adjust some column lengths Must apply setup/mysql/alter_tables.mysql cdrtool (6.2.3) unstable; urgency=low * Print information about the normalize lock in syslog and to stdout * Log if rating engine takes more time to complete. Must set $RatingEngine['log_delay'] to a value in micro seconds in global.inc and restart ratingEngine * Small html changes in sip settings page * Renamed RatingEngine object to differ from the same setting name from global.inc cdrtool (6.2.2) unstable; urgency=low * Added setting in global.inc $RatingEngine['prepaid_lock'] to enable multiple parallel calls using the same prepaid account, this is done by disabling the lock (setting it to 0 or false). Disable the lock only if you accept that the balance might become negative * Added email filter to Sip->getAccounts() * Log the call start time in the head of rating info to eliminate the confusion when rating calls that span multiple profiles * Show total call duration in rate info head * Use <voice-mailbox> instead of absolute URIs for call diversions * Removed unnecessary code used in the past for replication of changes to remote soap engine * Fixed replication of customer account to remote platform * Add active_master to replicatedDatabases in global.inc * Added web link between sip target and sip_accounts * Rewrite of mysql monitoring functions * Added DeleteBalance and DeleteBalance History commands to rating engine * Use colors in mysql replication status * Added posibility to replicate customer entries in Customer Port * Added getPrepaidHistory function to the rating engine * Replaced false with 0 and true with 1 for the return values from rating engine * Show database ip address when replication monitor returns an error * Fixed name of sip_ports settings * Added setup/crontabs directory for non debian installations cdrtool (6.2.1) unstable; urgency=low * Correct update of radius records when mediaproxy field is NULL or !timeout You must reload the sql stored procedures from setup/radius/OpenSER/radius_accounting.proc or update Freeradius with setup/radius/OpenSER/sql.conf when not using the stored procedures * Log add balance to prepaid_history table cdrtool (6.1.9) unstable; urgency=low * Fixed memory leak related to calling the same soap function several times * Added separate access numbers for FUNV, FBUS, FNOL, FNOA * Fixed load of customer properties in ngnpro client * Support multiple clusters for mysql replication monitor * Fixed vulnerability related to missing BYE and MediaProxy radius Update action (reported by Inaki Baz Castil) You must reload the sql stored procedures from setup/radius/OpenSER/radius_accounting.proc or update Freeradius with setup/radius/OpenSER/sql.conf when not using the stored procedures * Specify if output of soap functions is html formatted or not cdrtool (6.1.8) unstable; urgency=low * Fixed mrtg buildStatistics call in cron.d (env LANG=C) * Added web page to show replications status and dynamic instructions about how to fix it * Replace $CDRTool['mysql_clusters'] in global.inc with $replicated_databases (see setup/global.inc.in for syntax) * Delete DB1 and DB2 definitions from global.inc cdrtool (6.1.7) unstable; urgency=low * Set mobile_number per SIP account * Add sip_accounts_lite to soap engine settings * Update voicemail full name when sip account firstname or lastname changes * Create templates_c dir required by smarty * Added templates directory for sending welcome note cdrtool (6.1.6) unstable; urgency=low * Fixed load of customer profiles for prepaid calls when no default customer defined cdrtool (6.1.5) unstable; urgency=low * Added lock icon when SIP UA registered using TLS contact * Added more information about renormalizing prepaid calls * Show port number and transport protocol in sip trace * Group trace message per IP instead of ip:port * Added welcome message per soap engine * Show error message when ngnpro client is not configured * Fixed link from sip domains to sip aliases page * Fixed ENUM soap port authentication in sip settings page * Replaced DB_ser with DB_openser Edit global.inc and replace DB_ser with DB_openser * Increased curl timeout for bulk SOAP operations * Return price in debit balance cdrtool (6.1.4) unstable; urgency=low * Remove sample mrtg files cdrtool (6.1.3) unstable; urgency=high * Fixed calculation of rating profiles when using having specific gateways/domains/subscribers * Fixed default port alocation for NGNPro version < 3 * Allow set of reseller 0.0 (admin) in login account * Show SipThor node in Sip settings title bar * Fixed management of presence rules * Fixed link from Sip accounts to sip settings page cdrtool (6.1.2) unstable; urgency=high * Fixed detection of CDR duration when minimumDuration was set cdrtool (6.1.1) unstable; urgency=low * Corrected default login account sql insert statement Regards, Adrian

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[OpenSER-Users] How configure Voice mail for multi users.
by Mian M Asif 20 Mar '08

20 Mar '08

Hi All, i want to configure voice mail on Asterisk 1.4 for multiple users. let me explain you the scenario. i have 10 users with the name of 1000,2000,3000,4000,5000,6000,.......and these user can call to each other. Now i want to configure separate voice mail box for separate user. my extensions.conf ..... settings below.. [voicemail] exten => _X.,1,Dial(SIP/${EXTEN}) exten => _X.,n,NoOp(Dial Status: ${DIALSTATUS}) exten => _X.,n,Goto(s-${DIALSTATUS},1) exten => s-NOANSWER,1,Background(vm-nobodyavail) exten => s-NOANSWER,n,VoiceMail(${EXTEN}@usersmail) exten => s-NOANSWER,n,Hangup() exten => s-CONGESTION,1,Background(vm-nobodyavail) exten => s-CONGESTION,n,VoiceMail(${EXTEN}@usersmail) exten => s-CONGESTION,n,Hangup() exten => s-CANCEL,1,Background(vm-nobodyavail) exten => s-CANCEL,n,VoiceMail(${EXTEN}@usersmail) exten => s-CANCEL,n,Hangup() exten => s-BUSY,1,Background(salesrep) exten => s-BUSY,n,VoiceMail(${EXTEN}@usersmail) exten => s-BUSY,n,Hangup() exten => s-CHANUNAVAIL,1,Background(vm-nobodyavail) exten => s-CHANUNAVAIL,n,VoiceMail(${EXTEN}@usersmail) exten => s-CHANUNAVAIL,n,Hangup() my voicemail.conf [usersmail] 1000 => 1212, userm, 1000(a)abc.com 2000 => 1212, userm, 2000(a)abc.com please help me how can i set calling number before send voice mail in users voicemail box. when i dial like VoiceMail(${EXTEN}@usersmail) voice mail not work and when i heard code user, like this VoiceMail(1000@usersmail) voicemail work fine. but i want to set if user dial 2000 or 3000 or 4000 it should be set automatically. thanks for your cooperations. regards, Asif

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