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37510 discussions

R: R: [Users] using AVPs to write into database
by D'Addelfio Davide 28 Aug '06

28 Aug '06

Thanks, I've looked into but it works only with 1.1.x version...!!! What's the different with avp_db_store command, may I use it to store something in my SQL DB? I'll try to do but I have some problems, he doesn't register... I have another question: I need to work over sdp messages, is possible to do a parsing, always using the scripts language...? Any thought? Thanks Davide -----Messaggio originale----- Da: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] Inviato: mercoledì 5 aprile 2006 19.15 A: D'Addelfio Davide Cc: Jayesh Nambiar; Bogdan-Andrei Iancu; openser Oggetto: Re: R: [Users] using AVPs to write into database Hello, On 04/05/06 18:01, D'Addelfio Davide wrote: > > Hi guys, I have a similar problem, maybe easier. I'm new in openser so > I'd like to have some help. > > I'd like openser create a table in MySQL db when receive an INVITE, a > row with from uri, method and something else, I've look into modules > docs but I've not understanding at all...could you help me, please? > if you are using openser development version, then take a look at avp_db_query() in avpops module http://openser.org/docs/modules/1.1.x/avpops.html#AEN239 Cheers, Daniel > Thanks for reply > > Davide > > ------------------------------------------------------------------------ > > *Da:* users-bounces(a)openser.org [mailto:users-bounces@openser.org] > *Per conto di *Jayesh Nambiar > *Inviato:* mercoledì 5 aprile 2006 15.53 > *A:* Bogdan-Andrei Iancu > *Cc:* openser > *Oggetto:* Re: [Users] using AVPs to write into database > > Hi Bogdan, > > Thanks for the reply and sorry for sending the earlier mail 4 times as > yahoo was not responding peoperly. > > When I do an avp_write(), why does the avp name get stored in the uuid > column? > > I am not using any uuid as of now. I also tried to add the username of > the person who is calling but it again creates two rows, one with the > value as > > username and other row with the value as forwarded number. > > I thought if there is a way to insert the username also, it wud be > easier to check, if the incoming call for that user is to be forwarded > or not. > > I get the following in my usr_preferences table after I dial 86 > followed by any number. > > +-----------+----------+--------+-----------+------+-----------------------------+---------------------+ > | uuid | username | domain | attribute | type | value | modified | > +-----------+----------+--------+-----------+------+-----------------------------+---------------------+ > | s:callfwd | | | callfwd | 0 | sip:5515551478@202.80.61.10 | > 2006-04-05 19:10:49 | > +-----------+----------+--------+-----------+------+-----------------------------+---------------------+ > > The script snippet is as follows: > > if(uri=~"^sip:86[0-9]*@") { > > > strip(2); > avp_write("$ruri", "s:callfwd"); > #avp_write("$from/username", "i:999"); > #avp_db_store("i:999", "i:/usr_preferences"); > avp_db_store("s:callfwd", "s:callfwd/usr_preferences"); > sl_send_reply("200", "OK"); > exit; > }; > > > */Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>/* wrote: > > Hi, > > you need two steps: > 1) write the ruri (after strip) into an AVP - use avp_write() > 2) write the avp into the db - use avp_db_store() > > regards, > bogdan > > Jayesh Nambiar wrote: > > > Hi all, > > I am trying to implement the call forwarding feature in openser. The > > forwarding part is fine. But I have the follwing scenario: > > Whenever a user needs to change his forwarding number, he should be > > able to do so from his end device only. > > For eg: He first presses 86 and then the 10 digit number to be > > forwarded. Here the 86 has to be stripped and the 10 digit number > > should be inserted into the call forward value column of the > > preferences table. > > Also if he wants to remove the forwarded number, he can do so by > > pressing say for eg: 87. on receiving this number openser should > > delete that value from the > > table. > > Is this possible. I tried to do the following, but somehow it > does not > > change the > > value. > > if(uri=~"^sip:86[0-9]*@") { > > if(avp_db_load("$from/username", "s:callfwd")) { #check > > if call-fwd feature is enabled for the user > > log(1,"AVP condition returned true"); > > strip(2); > > avp_write("$ruri", "s:callfwd"); > > avp_print(); > > log(1,"AVP written"); > > sl_send_reply("200", "OK"); > > exit; > > }; > > }; > > Is avp_write the proper method or I guess avp_db_store can also help > > me. The avp_print() function also does not show me anything in the > > log. Are there any > > logical mistakes or I have mis-interpreted the syntax of avpops > functions. > > Please help me in thsi regard. > > Thanks a lot in advance. > > Jayesh. > > > > ------------------------------------------------------------------------ > > Jiyo cricket on Yahoo! India cricket > > > > Yahoo! Messenger Mobile > > > > Stay in touch with your buddies all the time. > > > >------------------------------------------------------------------------ > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > ------------------------------------------------------------------------ > > Jiyo cricket on Yahoo! India cricket > <http://us.rd.yahoo.com/mail/in/mailcricket/*http:/in.sports.yahoo.com/crick…> > Yahoo! Messenger Mobile > <http://us.rd.yahoo.com/mail/in/mailmobilemessenger/*http:/in.mobile.yahoo.c…> > Stay in touch with your buddies all the time. > > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/users >

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[Serusers] SER with VoiceMail Problem
by ram 28 Aug '06

28 Aug '06

Hi i am new to ser, iam going through the list, i found some of the similar queries http://lists.iptel.org/pipermail/serusers/2003-October/003132.html I have used the config file changed according to my setup and iam able to pick up messages from Asterisk also But i have some problem here is 1. i have registered user 8888 in my SER, when i dial 8888 its going to voice mail, how can i set the caller==calee Busy even i have added this if (!uri==myself) { t_relay(); break; }; But still no use. 2. when i dial to PSTN number, when the PSTN also go to voice mail after certain rings I suppose to get the voice Menu of other Side Number, but the voice go Blank ( instead of going to VoiceMenu) when checked in my Log, i get this error in my syslog Aug 28 18:33:55 router ser[757]: no location or no user Aug 28 18:34:01 router ser[751]: 10 digit exp match w/leading 1 Aug 28 18:34:01 router ser[751]: route[2]:SIP-to-PSTN call routed Aug 28 18:34:10 router ser[757]: no location or no user Aug 28 18:34:25 router ser[758]: no location or no user Aug 28 18:34:28 router ser[791]: ACC: call missed: from=88888 < sip:88888@mydomain.com>;tag=1674409670, i-uri= sip:1xxxxx80369@mydomain.com, method=INVITE, o-uri=sip:1xxxxx80369@myprovider:5060, code=408 Request Timeout Aug 28 18:34:28 router ser[791]: failure_route[5] Aug 28 18:34:28 router ser[791]: route[5]: re-relay to PSTN Aug 28 18:34:28 router ser[751]: ERROR: t_should_relay: status rewrite by UAS: stored: 408, received: 487 Aug 28 18:34:37 router ser[758]: 10 digit exp match w/leading 1 Aug 28 18:34:37 router ser[758]: route[2]:SIP-to-PSTN call routed Aug 28 18:34:37 router ser[758]: 10 digit exp match w/leading 1 Aug 28 18:34:37 router ser[758]: route[2]:SIP-to-PSTN call routed Aug 28 18:34:40 router ser[751]: no location or no user any suggestions Ram

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[Serusers] Proxy Authentication
by Javier Oviedo 28 Aug '06

28 Aug '06

Hi all In my network schema I have a Sip Express Router to authenticate my outbound calls to a PSTN. I add a new equipment to a ser with "serctl add user pass e-mail" command. My softphone is registered in proxy but when I make a call to a PSTN I obtain the following error: *Call rejected: 407 Proxy Authentication Rquired *: I use the SJPhone software. Any idea about where is the problem Thanks in advance Joe My ser configuration is: debug=3 fork=no log_stderror=yes port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(1); break; }; if (method=="ACK") { route(1); break; } if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("localhost","subscriber")) { www_challenge("localhost","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("localhost","subscriber")) { proxy_challenge("localhost","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(1); break; }; #if (!lookup("location")) { # sl_send_reply("404", "User Not Found"); # break; #}; route(1); } SoftPhone LOG: 17:04:43.1 SENDING TO: 10.110.0.127:5060 INVITE sip:313812@sipserver.es SIP/2.0 To: <sip:313812@sipserver.es> From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport Call-ID: fc7b726c6545d126 CSeq: 1 INVITE Contact: <sip:313813@172.25.97.138:9528> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 274 v=0 o=- 32759240 32759303 IN IP4 172.25.97.138 s=eyeBeam c=IN IP4 172.25.97.138 t=0 0 m=audio 7262 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 8013E837 DD8642E8 172.25.97.138 7262 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv 17:04:43.1 RECEIVING FROM: 10.110.0.127:5060 SIP/2.0 407 Proxy Authentication Required To: <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.3909 From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport=9528 Call-ID: fc7b726c6545d126 CSeq: 1 INVITE Proxy-Authenticate: Digest realm="localhost", nonce="44f307373cf9fb29da3028442e23f976513fc91d" Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 req_src_ip=172.25.97.138 req_src_port=9528 in_uri=sip:313812@sipserver.es out_uri=sip:313812@sipserver.es via_cnt==1" 17:04:43.1 SENDING TO: 10.110.0.127:5060 ACK sip:313812@sipserver.es SIP/2.0 To: <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.3909 From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-366872507-1--d87543-;rport Call-ID: fc7b726c6545d126 CSeq: 1 ACK Content-Length: 0 17:04:43.1 SENDING TO: 10.110.0.127:5060 INVITE sip:313812@sipserver.es SIP/2.0 To: <sip:313812@sipserver.es> From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport Call-ID: fc7b726c6545d126 CSeq: 2 INVITE Contact: <sip:313813@172.25.97.138:9528> Max-Forwards: 70 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO Content-Type: application/sdp Proxy-Authorization: Digest username="313813",realm="localhost",nonce="44f307373cf9fb29da3028442e23f976513fc91d",uri="sip:313812@sipserver.es",response="bfd77c8cdc1066487b170b19e1e0fe61",algorithm=MD5 User-Agent: eyeBeam release 3004t stamp 16741 Content-Length: 274 v=0 o=- 32759240 32759303 IN IP4 172.25.97.138 s=eyeBeam c=IN IP4 172.25.97.138 t=0 0 m=audio 7262 RTP/AVP 100 6 0 8 3 18 5 101 a=alt:1 1 : 8013E837 DD8642E8 172.25.97.138 7262 a=fmtp:101 0-15 a=rtpmap:100 speex/16000 a=rtpmap:101 telephone-event/8000 a=sendrecv 17:04:43.1 RECEIVING FROM: 10.110.0.127:5060 SIP/2.0 100 trying -- your call is important to us To: <sip:313812@sipserver.es> From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport=9528 Call-ID: fc7b726c6545d126 CSeq: 2 INVITE Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 req_src_ip=172.25.97.138 req_src_port=9528 in_uri=sip:313812@sipserver.es out_uri=sip:313812@sipserver.es via_cnt==1" 17:04:43.1 RECEIVING FROM: 10.110.0.127:5060 SIP/2.0 407 Proxy Authentication Required To: <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.fcd1 From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport=9528 Call-ID: fc7b726c6545d126 CSeq: 2 INVITE Proxy-Authenticate: Digest realm="localhost", nonce="44f307373cf9fb29da3028442e23f976513fc91d" Server: Sip EXpress router (0.9.6 (i386/linux)) Content-Length: 0 Warning: 392 10.110.0.127:5060 "Noisy feedback tells: pid=7865 req_src_ip=10.110.0.127 req_src_port=5060 in_uri=sip:313812@sipserver.es out_uri=sip:313812@sipserver.es via_cnt==2" 17:04:43.1 SENDING TO: 10.110.0.127:5060 ACK sip:313812@sipserver.es SIP/2.0 To: <sip:313812@sipserver.es>;tag=ae8ae6963bd737be6ef2befd1ceaf249.fcd1 From: Joe<sip:313813@sipserver.es>;tag=52710612 Via: SIP/2.0/UDP 172.25.97.138:9528;branch=z9hG4bK-d87543-736589046-1--d87543-;rport Call-ID: fc7b726c6545d126 CSeq: 2 ACK Content-Length: 0 17:04:43.2 Call (l:'Joe' r:'sip:313812@sipserver.es') - Call being terminated. Reasons: "Proxy Authentication Required", (code: 407)

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Re: [Serusers] rewrite the From part of SIP INVITE messages
by rcarvalho＠iric.up.pt 28 Aug '06

28 Aug '06

I'm sceptic about the use of functions that replace parts of SIP messages, because that way, will Ser still keep record of transactions? Wont those replacements affect Ser to recognise to whom route subsequent messages? Regards, Ricardo. Alexandr Dubovikov wrote: > On Fri, Aug 25, 2006 at 05:33:52PM +0100, Ricardo Carvalho wrote: >> I need to rewrite the From part of SIP INVITE messages before sending >> them to my IP Telco, or else they wont accept my calls! >> Anyone knows how can I do that? > > replace or subst > > the sample you can find in README file of textops module. > > >> Thanks, >> >> Ricardo. >> _______________________________________________ >> Serusers mailing list >> Serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >

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[Users] An error with exec_msg
by Zenone 28 Aug '06

28 Aug '06

Hi! I have an error when I try to run this in my openser.cfg: exec_msg("TAG=$(echo $SIP_HF_FROM|cut -f2 -d=);FROM=$(echo $SIP_HF_FROM|cut -f2 -d\<|cut -f1 -d\>|cut -f1-2 -d:);/usr/local/openser-1.1.0-tls/uri-folder/record $FROM $SIP_USER $TAG"); The obtained error is: Aug 28 15:11:21 localhost /usr/local/sbin/openser[32748]: ERROR:xl_parse_spec: bad pvar name "echo $SIP_HF_FROM|cut -f2 -d=);FROM=$(echo $SIP_HF_FROM|cut -f2 -d\<|cut -f1 -d\>|cut -f1-2 -d:);/usr/local/openser-1.1.0-tls/uri-folder/record $FROM $SIP_USER $TAG" Thanks in advance for your help! Michel ________________________________________________ Message sent using UebiMiau 2.7.8

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Re: [Serusers] serweb gives a blank page
by Andy Thomas 28 Aug '06

28 Aug '06

Andrey, Thanks for the pointer - I turned on PHP logging and found the problem- PHP Fatal error: Smarty error: unable to write to $compile_dir '/tmp/serweb'. Be sure $compile_dir is writable by the web server user. in /home/serweb/smarty/Smarty.class.php on line 1102 Down to permissions on the /tmp/serweb directory - all now working, thanks!!

2 1

[Users] Please help me in Openser Admin
by Ferianto siregar 28 Aug '06

28 Aug '06

Dear all, Thanks to all of you for your kind-hearted to give me solution in solving my problem. Thank you very much. Thanks to openser developer team, who has released Openser Administrator in sourceforge.net I have some questions about the openser admin. I do anybody can help me to undestand what the openser admin use for and how to install it. These are my question: 1. Is openser adminstrator a web application? I mean, does it have same function like "serweb" in ser? 2. How to install the openser administrator in openser server? I use openser1.1.0-tls and what packets does it need? 3.Is the openser administrator just used by the administrator to manage his/her server? or The openser administrator can be used like a serweb?(the client also can use it) because as I read from the reply message in this forum, the serweb can not use for openser1.1.0-tls 4. If it is not like serweb, can anybody tell me what is the suitable web application for openser ? That is all my questions. I do hope anybody can help me. Thanks Regards, Ferianto --------------------------------- How low will we go? Check out Yahoo! Messengers low PC-to-Phone call rates.

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[Serusers] REGISTRATION high pace & build_contact() error
by Kostas Marneris 28 Aug '06

28 Aug '06

Hi all, using sipsak in usrloc mode to test the SER sip REGISTRAR with the following process : loop: ./sipsak -U -p host.foo.com -x 3600 -a passwd -s sip:user@foo.com goto loop I get the following mesg in LOG file after 2 mins : build_contact(): No memory left ... and after a while : ERROR: build_res_buf_from_sip_req: out of memory ; needs 28132 send_reply(): Error while sending 200 OK ... CRITICAL ERROR : Can't save Contact to MySQL Location DB (this is a log mesg in my ser.cfg in case : if (!save("location")) ) ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) although it seems that SER keeps going (?) to write in Location DB both SER's & MySQL (this seems strange). I have IBM eServer 336 x 2 : - 2 dual core Intel Xeon 3.6 GHz with 2MB Cache, EMT64 - 4 GB RAM - 2x72GB Ultra 320 Scsi disks # ser.cfg children=5 Could this be a memory leak problem in SER ? Is there any conf fine-tuning I can do in order to avoid that ? When I increase the 'children' up to 20, the problem happens again but within 5 mins. thanks for any help, Kostas

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[Serusers] serweb gives a blank page
by Andy Thomas 28 Aug '06

28 Aug '06

I have installed Serweb, but when I run it I just get a blank page, with just the serweb title at the top. Nothing else appears, no login box or anything! I have set PEAR LOG to debug mode and the log only shows this- Aug 28 12:04:37 serweb [info] Useing file from default domain for filename: separator.html Aug 28 12:04:40 serweb [info] Useing file from default domain for filename: styles.css Aug 28 12:04:40 serweb [info] Useing file from default domain for filename: prolog.html Aug 28 12:04:40 serweb [info] Useing file from default domain for filename: separator.html Aug 28 12:04:41 serweb [info] Useing file from default domain for filename: styles.css Aug 28 12:04:41 serweb [info] Useing file from default domain for filename: prolog.html Aug 28 12:04:41 serweb [info] Useing file from default domain for filename: separator.html Aug 28 12:04:42 serweb [info] Useing file from default domain for filename: styles.css Aug 28 12:04:42 serweb [info] Useing file from default domain for filename: prolog.html Aug 28 12:04:42 serweb [info] Useing file from default domain for filename: separator.html The HTML generated is as follows- <!DOCTYPE html PUBLIC "-//W3C//DTD XHTML 1.0 Strict//EN" "http://www.w3.org/TR/xhtml1/DTD/xhtml1-strict.dtd"> <html> <head> <title>VoIPgb.net - web interface</title> <meta http-equiv="Content-Type" content="text/html; charset=utf-8"> <meta name="Author" content="Karel Kozlik <karel at iptel dot org>"> <meta http-equiv="PRAGMA" content="no-cache"> <meta http-equiv="Cache-control" content="no-cache"> <meta http-equiv="Expires" content="Mon, 28 Aug 2006 11:04:42 GMT"> <LINK REL="StyleSheet" HREF="/serweb/domains/_default/styles.css" TYPE="text/css"> </head> <script language="JavaScript" src="/serweb/js/login_completion.js.php"></script><body><h1> voipgb.net admin interface</h1><hr> <div class="swMain"> For some reason, the rest of the page isn't generated! Any ideas ??????

2 1

[Users] Problem with SQL queries in 1.1.0?
by Federico Giannici 28 Aug '06

28 Aug '06

We recently upgraded from version 1.0.0 to 1.1.0 from CVS of a few days ago. After a couple of days of uptime, some SQL errors began to appear. First there was this one: Aug 21 10:07:07 eowyn OpenSER[19895]: ERROR: warning_builder: buffer size exceeded Aug 21 10:07:07 eowyn OpenSER[19895]: WARNING: warning skipped -- too big Aug 21 10:07:07 eowyn OpenSER[19895]: submit_query: Aug 21 10:07:07 eowyn OpenSER[19895]: db_insert: Error while submitting query Aug 21 10:07:07 eowyn OpenSER[19895]: ERROR:acc:acc_db_request: Error while inserting to database Then, some errors like this: Aug 21 15:41:28 eowyn OpenSER[19895]: submit_query: Aug 21 15:41:28 eowyn OpenSER[19895]: db_insert: Error while submitting query Aug 21 15:41:28 eowyn OpenSER[19895]: ERROR:acc:acc_db_request: Error while inserting to database Or the following: Aug 21 19:02:10 eowyn OpenSER[20604]: submit_query: Aug 21 19:02:10 eowyn OpenSER[20604]: db_query: Error while submitting query Aug 21 19:02:10 eowyn OpenSER[20604]: NM ERROR: nm_hash_load_domains(): error with dbf.query() The nm_hash_load_domains() function is a module of mine, that make an SQL query. No problem with version 1.0.0. Is there any SQL known problem in the 1.1.0 CVS version? Maybe the first error caused some memory garbage? Thanks. -- ___________________________________________________ __ |- giannici(a)neomedia.it |ederico Giannici http://www.neomedia.it ___________________________________________________

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