sr-users February 2005

sr-users@lists.kamailio.org

215 participants
467 discussions

[Serusers] JAIN SI, javax.sip.PeerUnavailableException and Java 1.5
by Graham D Forsyth 24 Feb '05

24 Feb '05

Hi. I am using JAIN-SIP to create a SIP message client. When trying to create Message, Address and Header factories using SipFactory, my program throws a PeerUnavailableException. This also occurs when I try to create a SIP stack- I get the following exception: javax.sip.PeerUnavailableException: The Peer SIP Stack: gov.nist.javax.sip.SipStackImpl could not be instantiated. Ensure the Path Name has been set. I have set the path name to gov.nist. Does anyone know why I am getting this exception even though I have set the SIP stack path? I am using Java 1.5. Maybe because the JAR file has been compiled with Java 1.4.2 the byte code is incompatible? Anyway I don't know. If anybody has any ideas about this please let me know. Thanks, Graham Forsyth Graham Forsyth -- The University of Stirling is a university established in Scotland by charter at Stirling, FK9 4LA. Privileged/Confidential Information may be contained in this message. If you are not the addressee indicated in this message (or responsible for delivery of the message to such person), you may not disclose, copy or deliver this message to anyone and any action taken or omitted to be taken in reliance on it, is prohibited and may be unlawful. In such case, you should destroy this message and kindly notify the sender by reply email. Please advise immediately if you or your employer do not consent to Internet email for messages of this kind.

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[Serusers] how to start rtpproxy
by Mohammad Khan 24 Feb '05

24 Feb '05

Hello List, I am trying to run ser with rtpproxy. I am getting this: Feb 24 11:55:22 perch /usr/sbin/ser[31012]: ERROR: send_rtpp_command: can't connect to RTP proxy I have only loadmodule "/usr/lib/ser/modules/nathelper.so" in my ser.cfg, no other nathelper function. I did serch in list and found 'it was answered so many times' but again bad luck for me. Could anybody post me the answer. I will really appreciate. Thank, Mohammad

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[Serusers] t_on_reply
by Iqbal 24 Feb '05

24 Feb '05

Hi Anyone using t_on_reply actually debugged the sonfig to see if it actually goes there, I am debugging the config, and although I seem to be going to t_on_reply(1) it doesnt seem to get there, cause the xlog statement in that route doesnt display, I know this is a bit of a open question...:-) iqbal

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[Serusers] what does a break do
by Iqbal 24 Feb '05

24 Feb '05

Hi I was just debugging my ser.cfg, and wanted to know what happens on a break command. I presume that the break is there to stop you falling through, but where do you jump to EG other code route { blah blah blah route [2]; } route[2] { if (a=b) {do something} else { break;} } route [3] { } In route two what happens a) when a=b and b) when the break is set. in a) does it jump back to the start of ser.cfg and process again, or jump back to the line after what called it, and on the "break" again does it jump back to the calling command Iqbal

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[Serusers] Help when make a Call Busy Forward Failed? Does 'best practice' release??
by Charles Wang 24 Feb '05

24 Feb '05

Dear ALL: I use Paul's ser.cfg for 0.9.0 post last year and wanna implement a call busy forward. I think Paul is busy on testing 'Best Practice', so he doesn't answer too much mail. I try to make a call from UA1 to UA2, and UA2 is busy. So the call will forward to another Trunk 3. Ex: UA1 ==> UA2(busy) ==(failure_route[1])==> Trunk 3 PS: UA1 and UA2 behind different NATs. But when I watch the log, I find that miss the following log (diff with call from UA1 to UA2 and directly forward to Trunk 3 Ex: UA1 ==> UA2(callfwd) ==> Trunk 3 When I make a callfwd call, the call will run route[3] but not failure_route[1]. UA1==>UA2(callfwd)==>route[2]==>Trunk 3 route[2] ====> SIP to SIP call route[3] ====> SIP to PSTN call failure_route[1] ====> NoAnswer or Busy Forward logic And I make a fwdbusy call, the call will run route[2] then go to failure_route[1], and return to route[3]. UA1==>UA2(fwdbusy)==>failure_route[1]==>route[3] =XXX=> Trunk 3 ???? Why does the method be failed? Do I must "end_media_session()" before start a busy call? How can I modify it? My snippet ser.cfg : -------------------------------------------------------------------------------------------------------------- route[3] { log(1, "SER: International Call Off-Net section route(3)\n"); # All Domestic Calls Go To CISCO if (method=="INVITE") { if (!proxy_authorize("", "subscriber")) { proxy_challenge("", "0"); break; } else if (!check_from()) { log(1, "Spoofed SIP call attempt"); sl_send_reply("403", "Use From=ID"); break; } else if (!(is_from_local() || is_uri_host_local())) { sl_send_reply("403", "Please register to use our service"); break; }; # enable caller id blocking for PSTN calls if (isflagset(25)) { append_rpid_hf(); }; }; # SIP->PSTN calls get 45 seconds to timeout log(1, "SER: Connecting to PSTN.....\n"); avp_write("i:45", "inv_timeout"); rewritehost("xxx.xxx.190.243"); #### A Trunking Gateway if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")){ sl_send_reply("479", "We don't forward to private IP addresses"); break; }; if (method=="INVITE" || method=="ACK") { use_media_proxy(); }; if (isflagset(31)) { # is voice mail? t_on_failure("1"); }; t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } failure_route[1] { log(1, "SER: Failure Route section failure_route(1)\n"); # if caller hung up then don't sent to voicemail if (t_check_status("487")) { break; }; if (isflagset(26) && t_check_status("486")) { # forward busy is flag 26 if (avp_pushto("$ruri", "s:fwdbusy")) { log(1, "SER: fork to fwdbusy\n"); avp_delete("s:fwdbusy"); append_branch(); resetflag(26); # test for domestic PSTN gateway if (uri=~"^sip:0[0-9]{9}@") { # if (avp_check("$fwd_busy_type", "eq/dom/i")) { # test for domestic PSTN gateway log(1, "SER: Busy Failure and Jump to route(3)\n"); route(3); } else if (uri=~"^sip:002[1-9][0-9]*@") { # } else if (avp_check("$fwd_busy_type", "eq/int/i")) { # test for international PSTN gateway log(1, "SER: Busy Failure and Jump to route(6)\n"); route(6); } else { # default to sip call log(1, "SER: Busy Failure and Jump to route(2)\n"); route(2); }; break; }; }; # here we can have either voicemail __OR__ forward no answer if (isflagset(27) && t_check_status("408")) { # forward no answer is flag 27 if (avp_db_load("$ruri/username", "s:fwdnoanswer")) { avp_pushto("$ruri", "s:fwdnoanswer"); log(1, "SER: fork to fwdnoanswer\n"); avp_delete("s:fwdnoanswer"); append_branch(); resetflag(27); if (uri=~"^sip:0[0-9]{9}@") { # if (avp_check("$fwd_no_answer_type", "eq/dom/i")) { # test for domestic PSTN gateway log(1, "SER: No Answer Failure and Jump to route(3)\n"); route(3); } else if (uri=~"^sip:002[1-9][0-9]*@") { # } else if (avp_check("$fwd_no_answer_type", "eq/int/i")) { # test for international PSTN gateway log(1, "SER: No Answer Failure and Jump to route(6)\n"); route(6); } else { # default to sip call log(1, "SER: No Answer Failure and Jump to route(2)\n"); route(2); }; break; }; } else if (isflagset(31) && avp_pushto("$ruri", "$voicemail")) { avp_delete("$voicemail"); log(1, "SER: No Answer Failure and Jump to route(4)\n"); route(4); break; }; } In SysLog: ------------------------------------------------------------------------------------------------------ proxydispatcher[13265]: command request 06DC9503-1938-4D9D-8628-E648F112BEC5(a)192.168.11.7 192.168.11.7:8000:audio 61.217.225.225 ser.xxx.net.tw local xxx.xxx.190.243 remote X-PRO=20build=201082 info=from:1011@ser.xxx.net.tw,to:9012@ser.xxx.net.tw,fromtag:2069977709,totag: Feb 24 00:59:21 ser proxydispatcher[13265]: domain ser.xxx.net.tw doesn't define any mediaproxy. Feb 24 00:59:21 ser proxydispatcher[13265]: will use default mediaproxy for this call. Feb 24 00:59:21 ser mediaproxy[1087]: command request 06DC9503-1938-4D9D-8628-E648F112BEC5(a)192.168.11.7 192.168.11.7:8000:audio 61.217.225.225 ser.xxx.net.tw local xxx.xxx.190.243 remote X-PRO=20build=201082 info=from:1011@ser.xxx.net.tw,to:9012@ser.xxx.net.tw,fromtag:2069977709,totag:,dispatcher Feb 24 00:59:21 ser mediaproxy[1087]: command execution time: 0.35 ms Feb 24 00:59:21 ser proxydispatcher[13265]: forwarding to mediaproxy on /var/run/mediaproxy.sock: got: 'xxx.xxx.190.248 35030' Feb 24 00:59:21 ser proxydispatcher[13265]: command execution time: 311.69 ms

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Re[2]: [Serusers] Do ser + asterisk_b2bua work ?
by Mike Tkachuk 24 Feb '05

24 Feb '05

Hello Iqbal, Wednesday, February 23, 2005, 11:33:15 AM, you wrote: I> Hi Mike I> For those of us who are using asterisk, is it a replacement for the b2b I> part of asterisk, or a add-on to asterisk to make the b2b part easier. I I> am using asterisk for voicemail, however I need to now start to look at I> the billing, I was just going to pas the stream via asterisk then hit I> pstn and use the rate-engine and cdr modules for asterisk, how/where I> does this fit in I> tks I> Iqbal It's add-on that will allow to make b2bua from asterisk box, all asterisk features are unchanged and you can use asterisk same way as you do. All accounting and authorization is performed using radius. P.S. Please use b2bua maillist (https://lists.berlios.de/mailman/listinfo/b2bua-users) for questions. I think serusers list is bad place for it. -- Best regards, ~*-,._.,-*~'`^`'~*-,._.,-*~'`^`'~*-,. Mike Tkachuk, ph:380-3433-47067 YES ISP, fx:380-3433-47067 Valova 17, mike|a|yes.net.ua Kolomyia, www.yes.net.ua Ukraine 78200 FWD: 66518 24.02.2005 ICQ# 57698805 MSN: mike_tkachuk|a|hotmail.com ~*-,._.,-*~'`^`'~*-,._.,-*~'`^`'~*-,.

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RE: [Serusers] DNS SRV lookup issue
by Tan, Chiang Kang 24 Feb '05

24 Feb '05

I have some issues with DNS mapping too. I want to have a sip user with sip:user1@testdomain.com. In my location table, I however stored sip:user1@machine1.testdomain.com as machine1.testdomain.com maps to the actual IP address (I have such an entry in /etc/hosts). Hence although a DNS SRV lookup is performed, it tries to contact _sip._udp.machine1.testdomain.com, which does not exist (I only have SRV entry for _sip._udp.testdomain.com). I think I'm missing something, can somebody tell me where? Any help will be greatly appreciated. cheers Chiang -----Original Message----- From: Zeus Ng [mailto:zeus.ng@isquare.com.au] Sent: 24 February 2005 09:04 To: 'Gandhi, Chetan' Cc: serusers(a)lists.iptel.org Subject: RE: [Serusers] DNS SRV lookup issue Are you absolutely sure the contact address is stored as user(a)testdomain.com and not user(a)some.ip.address in the location table? I'm sure t_relay() will do SRV lookup, with one condition. The URI should be in the form sip:user@testdomain.com, with no port specified. If the URI does have the port included like sip:user@testdomain.com:5060, then SER will not do SRV lookup. Most UAs I saw use IP address in contact header and that means no SRV lookup. Better check yours with serctl ul show. You can verify the SRV lookup with the following script. loadmodule "/usr/local/lib/ser/modules/tm.so" route{ rewritehost("stupid.domain"); t_relay(); } The ethereal output will show SRV lookup. Zeus > -----Original Message----- > From: serusers-bounces(a)lists.iptel.org > [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Gandhi, Chetan > Sent: Wednesday, 23 February 2005 9:15 AM > To: serusers(a)lists.iptel.org > Subject: [Serusers] DNS SRV lookup issue > > > Hi All, > > > > I am facing problem to get SER to do a SRV lookup. I have the > following scenario. > > > > Step 1: > > SIP UA1 registers with SER (testdomain.com) with contact > address user(a)testdomain.com > > SIP UA2 registers with SER (testdomain.com) with contact > address user(a)testdomain.com > > > > Step 2: > > SIP UA3 calls user(a)testdomain.com > > > > Problem > > I would expect SER to do a DNS SRV lookup on testdomain.com > and get sip1.testdomain.com and sip2.testdomain.com. > Unfortunately, SER only does A and AAAA query. (verified via > ethereal) and does not do SRV lookup. How can I make it do > SRV lookup ? I am using the default ser.cfg file. > > > > > > DNS entries > > _sip._udp.testdomain.com. 3600 IN > SRV 0 1 5060 sip1.testdomain.com. > > _sip._udp.testdomain.com. 3600 IN > SRV 0 1 5060 sip2.testdomain.com. > > > > I am trying to do load balancing but the SRV lookup failure > has got me stuck. Any pointers would be really appreciated. > > > > Thanks, > > Chetan > > _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] error to start ser-0.9.0
by Mohammad Khan 24 Feb '05

24 Feb '05

Hello list: I was away for little while from playing with ser. I just switched from 0.8.14 to 0.9.0 I am getting these, when I tried to start ser. My main concern is "submit_query(): Unknown column 'table_version' in 'field list'" Anybody out there can help, please. Feb 22 18:29:46 perch /usr/sbin/ser[21332]: Maxfwd module- initializing Feb 22 18:29:47 perch /usr/sbin/ser[21332]: submit_query(): Unknown column 'table_version' in 'field list' Feb 22 18:29:47 perch /usr/sbin/ser[21332]: submit_query(): Error while submitting query Feb 22 18:29:47 perch /usr/sbin/ser[21332]: table_version(): Error in db_query Feb 22 18:29:47 perch /usr/sbin/ser[21332]: register_udomain(): Error while querying table version Feb 22 18:29:47 perch /usr/sbin/ser[21332]: domain_fixup(): Error while registering domain Thanks MOhammad

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[Serusers] Error: execvp No such file/dir - running "/etc/rc.d/init.d/ser restart"
by Pepe 24 Feb '05

24 Feb '05

Hi I suposse you ind the solution, but if not the problem is in /etc/init.d/ser You need to change this line ser=/usr/sbin/ser for this one ser=/usr/local/sbin/ser Cya Mail at Cannes wrote: > After mysql setup looks ok > > After ser.cfg setup believed ok > > > > When running "/etc/rc.d/init.d/ser restart" > > I get error message:- > > > > Stopping ser: > > Starting ser: execvp: No such file or directory > > > [FAILED] > > > > Please can anyone give advice to resolve > > Thanks > > Chris > > > --- > Outgoing mail is certified Virus Free. > Checked by AVG anti-virus system (http://www.grisoft.com <http://www.grisoft.com/> ). > Version: 6.0.788 / Virus Database: 533 - Release Date: 01/11/2004 > >

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[Serusers] server problem of firewall?
by inda imoet 24 Feb '05

24 Feb '05

Hallo.... I have made a SER(SIP express router) server. i have try it in local network and it work. the client can call each other. when i try the server use IP public. a have a problem. i try using 2 client.one client uses static ip and other client uses ip public. local client can call and comunicate with the public client. but when public client call local client, it doesn't work. there some error message "408 : request time out". for information local client was under a firewall. Can you tell me the couse of error? is this because of the firewall? or there some script or other setting that must be added to the server.? __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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sr-users February 2005