sr-users October 2003

sr-users@lists.kamailio.org

105 participants
230 discussions

[Serusers] radius logon problems
by Floris Roset 08 Oct '03

08 Oct '03

Hi, I wonder if there has been an answer to the email below. I do have the same problem. thanks _______________________________________________ Radius logon problems Jiri Kuthan jiri at iptel.org Sat Sep 6 11:40:03 CEST 2003 Previous message: [Serusers] Radius logon problems Next message: [Serusers] sip_msg_cloner Messages sorted by: [ date ] [ thread ] [ subject ] [ author ] --------------------------------- Joseph, can you send us perhaps the SIP messages (ngrepped, etc.)?It is hard to guess what happens otherwise.-jiriAt 08:33 AM 9/6/2003, Jan Janak wrote:>Hello,>>On 06-09 00:25, Jiri Kuthan wrote:>> From these logs:>> >> 10(28016) pre_auth(): Credentials with given realm not found>> 10(28016) REGISTER: challenging user>> 10(28016) build_auth_hf(): 'WWW-Authenticate: Digest realm="ford.com", nonce="3f578eaee00b29a57d6fb234024d112ecf485627">> "Credentials with given realm not found" means that the server received> a SIP message that contained no credentials with given realm (ford.com> in this case). > >> I infer that the radius database includes no user identified by>> the username in question and domain "ford.com".>> No, it didn't make any request to the radius server in this case> because it was unable to find proper credentials in the SIP message.>> Jan.>>_______________________________________________>Serusers mailing list>Serusers at iptel.org>http://lists.iptel.org/mailman/listinfo/serusers--Jiri Kuthan http://iptel.org/~jiri/ --------------------------------- Want to chat instantly with your online friends? Get the FREE Yahoo!Messenger

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[Serusers] Questions???
by Hassan CHOUMAR 08 Oct '03

08 Oct '03

Hello all, I installed the server sip SER and he works well, but when I changed the IP address of the PC to put in another network, I have the configured and I changed the necessary modifications. Now the server SER works but I managed to realize the registrar in this server. Anybody knows where is the problem Thank you in advance Hassan ------------------------------------------------- This mail sent through IMP: http://horde.org/imp/

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Re: [Serusers] Not been able to log on
by Annie Sasidar 08 Oct '03

08 Oct '03

Hi, The two machines interact for a short period ( everything works fine)and then the connection is lost. When i reboot the two machines, the connection is renewed again for a short period.Could anybody give me a pointer on what could be going wrong. Thanks, Annie

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RE: [Serusers] SMS device to test
by LeeCC, Kenny 08 Oct '03

08 Oct '03

Hi, We have tested Siemens MC35T GSM modem and it works fine. Rgds, kenny.leecc > -----Original Message----- > From: Jan Janak [mailto:jan@iptel.org] > Sent: Tuesday, October 07, 2003 18:32 > To: Soo Wei Kang > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] SMS device to test > > > Hello, > > we are using Falcom A2D GSM modem. But it should work with almost any. > > Jan. > > On 07-10 18:34, Soo Wei Kang wrote: > > Hello, > > > > I would like to test the SMS Gateway feature. May I > > know which device works with the SMS Gateway? > > Specifically, can you please tell me the Brand and the Model > > that works? > > > > Has anybody tested with the Nokia 30 or Siemens M20 Terminal > > before? > > > > What is the standard AT commands that the SMS Gateway support? > > I'm not too familiar with this, but, is it something like > ETS GSM 07.05 > > or maybe something else? > > > > Thanks. > > > > Regards, > > WK > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

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[Serusers] Serweb registration bug
by Rork, Joseph (J.P.) 07 Oct '03

07 Oct '03

After registering using serweb, I get an email with a link to follow, and a row appears in the pending table of MySQL. After clicking on the link, I get the following page/error: iptel.org User Management 400 Table 'aliases' not found in memory, use save("aliases") or lookup("aliases") in the configuration script first We regret but your iptel.org confirmation attempt failed. Please contact info(a)iptel.org <mailto:info@iptel.org> for further assistance. Back to login form <../index.php> . The pending table still contains the record of the user, but the subscriber table also now contains the new row. How do I fix the aliases table error? And is this a bug that needs to be fixed? --joe

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[Fwd: Re: [Serusers] Not been able to log on]
by Mathieu MALAISE 07 Oct '03

07 Oct '03

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Re: [Serusers] Not been able to log on
by Annie Sasidar 07 Oct '03

07 Oct '03

Yes ser is running. Now i am not able to ping between the two machines. Could that be one of the reasons? But even when i could ping between the machines i had problem logging onto the server through Windows Messenger. Please guide. Thanks, Annie

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RE: [Serusers] Implementing Voicemail-Round 2- t_newtran ERROR
by Steve Dolloff 07 Oct '03

07 Oct '03

Jiri, I think the issue here is that we need to be able to set different call timers based on the destination of the call. If we have a user subscribed to voicemail, we would want a call_timer of 8 seconds for example. If we have a user that isn't subscribed to voicemail or a call being placed to the pstn, we might want a 60 second call timer. I'm sure you can see the problem... We can't fail our outbound calls after 8 seconds, and we can't wait 60 seconds to route incoming calls to voicemail. Stephen Subject: Re: [Serusers] Implementing Voicemail-Round 2- t_newtran ERROR Gavin, I am not exactly sure what the problem is -- indeed, SER transactions time out after some period of time, that's normal behaviour. It is called C-timer in RFC3261, we call the "tm" timer "inv_fr_invite" and can change its value with modparam. When the timer fires, the transaction is cancelled and deleted from server's memory not to waste it for ever. That's it. If you wish for-ever ringing phones, without proxy cancelling expired transactions you can do so under some specific circumstances. Then proxy drops pending transactions silently, without explicit cancellation, and the transactions may complete statelessly. If my memory serves, this behaviour is actually default unless any of frequent conditions occurs which requires stateful completion. Such conditions are SIP forking, use of accounting, on_failure processing, explicitly disallowing silent transaction drop, etc. (You simply need stateful completion for any of these functions.) The error as displayed in your logs is caused by your attempt to create two transactions for a single request. You create the first transaction context in route[]. When failure_route is triggered, you follow failure_route[1] and route[3] in which you attempt to create yet another transaction context. Also, let me remind again: VM module was designed for use in a UAS which stands separately from a proxy server. We never tried to mix proxy with voicemail in a single server instance. I will be glad to look at it as time allows, but I doubt it works now. -jiri At 11:40 PM 9/25/2003, Gavin Bensom wrote: >Steve and all, >Did you ever get resolution on the issue of the timeout and t_on_failure commands killing or routing PSTN calls to voicemail? > >I looked through the mail archives and it isn't clear what the resolution was. I'm running into the same problem. If I set t_on_failure to occur after a certain timeout, outgoing PSTN calls fail after that timeout as well. > >In fact, it seems that calls fail after the timeout even if t_on_failure isn't set. > >I've successfully gotten outgoing PSTN calls being handled by a different t_relay than incoming or internal network calls. > >What did you do to resolve the issue? > >My config. > >RedHat 9.0 >kernel 2.4.20-20smp >ser 0.8.11 (i386/linux) >main.c, v 1.162.2.5 > >Also, when a Status: 486 (busy) is encountered on the recieving party side, or a timeout occurs due to fr_inv_timer, I'm getting this error in my log > >Sep 25 14:07:56 jiffypop /usr/local/sbin/ser[23164]: ERROR: t_newtran: transaction already in process 0x422c0b38 >Anyone have any ideas on what the problem is? >ser.cft and ngrep output attached. > >Thanks, >G > > >Steve Dolloff <sdolloff(a)noc.dls.net> wrote: > > >>> I did place this portion inside the myself check >>>and it still tries to transfer to vm after the time expires. > >>I'm puzzled -- did not you want to transfer to vm after the time >expires? > >I will try to make this clearer. I am behind an ATA with a SIP proxy of >209.242.10.153. If I call someone else registered on my domain and they >are not available, I want to go to voice mail. If I call 1-800-555-1212 >from my phone, I do not want my sip proxy to reroute the call to >voicemail after 10 seconds if no one answers(or ever for that matter). >Right now if I dial 18005551212 from my handset, I see the destination >as sip:18005551212@209.242.10.153 on the server which matches to myself >and ser tries to send it to voicemail. > >Someone calling into the network is not a problem. They will never hit >our server unless the destination is local. > >>>This is the part that I really need help with! When the call timer >>>fails, the call goes to the route[1]. How do I get it into voice mail >>>from that point? > >>See bellow, I think that should work. > >This is what I had originally, and I get the following syslog. > > >Sep 10 16:36:36 voip2 ser: parse error (127,37-38): Command cannot be >used in the block >Sep 10 16:36:36 voip2 ser: ERROR: bad config file (1 errors) >Sep 10 16:36:36 voip2 ser: ser startup failed > >Is says that vm is not valid in the block. According the admin guide, >only certain commands can be used within a failure block. I assume that >is the problem here. If not, please let me know as this is exactly what >I want to do. > >>THE SAME STUFF LIKE ABOVE, YOU DON'T WANT TO t_relay ANYTHING > >>if(!vm("/tmp/am_fifo","voicemail")){ >> t_reply("500", "SEMS >>error"); >> }; >> break; > > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers > > >Do you Yahoo!? ><http://shopping.yahoo.com/?__yltc=s%3A150000443%2Cd%3A22708228%2Cslk%3 Atext%2Csec%3Amail>The New Yahoo! Shopping - with improved product search ># ># $Id: ser.cfg,v 1.20 2003/05/31 21:12:19 jiri Exp $ ># ># simple quick-start config script ># > ># ----------- global configuration parameters ------------------------ > >debug=3 # debug level (cmd line: -dddddddddd) >fork=yes >log_stderror=no # (cmd line: -E) > >/* Uncomment these lines to enter debugging mode >debug=8 >fork=no >log_stderror=yes >*/ > >check_via=no # (cmd. line: -v) >dns=no # (cmd. line: -r) >rev_dns=no # (cmd. line: -R) >port=5060 >children=4 >fifo="/tmp/ser_fifo" >sip_warning=no ># ># ------------------ module loading ---------------------------------- ># ># Uncomment this if you want to use SQL database >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/vm.so" >loadmodule "/usr/local/lib/ser/modules/pa.so" >loadmodule "/usr/local/lib/ser/modules/msilo.so" >loadmodule "/usr/local/lib/ser/modules/acc.so" >loadmodule "/usr/local/lib/ser/modules/textops.so" ># >#loadmodule "/usr/local/lib/ser/modules/nathelper.so" >#loadmodule "/usr/local/lib/ser/modules/uri.so" >#loadmodule "/usr/local/lib/ser/modules/group.so" ># ># Uncomment this if you want digest authentication ># mysql.so must be loaded ! >loadmodule "/usr/local/lib/ser/modules/auth.so" >loadmodule "/usr/local/lib/ser/modules/auth_db.so" ># ># ----------------- setting module-specific parameters --------------- ># ># -- usrloc params -- ># >#modparam("usrloc", "db_mode", 0) ># ># Uncomment this if you want to use SQL database ># for persistent storage and comment the previous line >modparam("usrloc", "db_mode", 2) ># ># -- auth params -- ># Uncomment if you are using auth module ># >modparam("auth_db", "calculate_ha1", yes) ># ># If you set "calculate_ha1" parameter to yes (which true in this config), ># uncomment also the following parameter) ># >modparam("auth_db", "password_column", "password") ># ># >modparam("acc", "log_level", 1) >modparam("acc", "log_flag", 2) >modparam("acc", "log_missed_flag", 2) >modparam("acc", "log_fmt", "fimos") ># >#modparam("tm", "fr_inv_timer", 15) #INVITE timeout >#modparam("tm", "fr_timer", 5) #negative INVITE reply or no > #final reply for a request for ACK ># >modparam("voicemail", "db_url", "sql://ser:heslo@localhost/ser") ># >#modparam("acc", "db_url", "sql://ser:heslo@localhost/ser") >#modparam("acc", "db_flag", 2) >#modparam("acc", "db_missed_flag", 2) ># ># ------------------------- request routing logic ------------------- ># ># main routing logic ># >alias=10.10.10.49 #sip server IP address >alias=jiffypop #sip server name >alias=mydomain.com #sip domain/realm >alias=jiffypop.mydomain.com #sip server FQDN ># >route{ > log(1,"entering main route"); > #prevent strangers from claiming to belong to our domain; > #if sender claims to be in our domain in From header field, > #better authenticate him > # code not inserted yet :) > > # initial sanity checks -- messages with > # max_forwards==0, or excessively long requests > if (!mf_process_maxfwd_header("10")) { > sl_send_reply("483","Too Many Hops"); > break; > }; > if (len_gt( max_len )) { > sl_send_reply("513", "Message too big"); > break; > }; > > # we record-route all messages -- to make sure that > # subsequent messages will go through our proxy; that's > # particularly good if upstream and downstream entities > # use different transport protocol > record_route(); > # loose-route processing > if (loose_route()) { > t_relay(); > break; > }; > > setflag(2); #set flag for accounting > > # 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") { > # digest authentication > log(1,"request for registration"); > if (!www_authorize("mydomain.com", "subscriber")) { > www_challenge("mydomain.com", "0"); > break; > }; > save("location"); > break; > }; > >/* ********** Dial out to PSTN logic ************* */ > > #forward numerical 7 digit requests to gateway > if(uri=~"^sip:[0-9]{7}@(mydomain\.com|10\.10\.10\.49)"){ > rewritehostport("10.10.10.5:5060"); > log(1,"7 digit expression match"); > route(2); > break; > > }; > # strip 650 and forward to GW if user dials 650 before phone no. > if(uri=~"^sip:650[0-9]{7}@(mydomain\.com|10\.10\.10\.49)"){ > strip(3); > rewritehostport("10.10.10.5:5060"); > log(1,"650 area code dialed, 650 stripped"); > route(2); > break; > > }; > #forward numerical 10 digit requests to gateway, append a 1 first > if(uri=~"^sip:[0-9]{10}@(mydomain\.com|10\.10\.10\.49)"){ > prefix("1"); > rewritehostport("10.10.10.5:5060"); > log(1,"10 digit expression match, prefix 1"); > route(2); > break; > }; > #forward numerical 11 digit requests that start with a 1 to GW > if(uri=~"^sip:1[0-9]{10}@(mydomain\.com|10\.10\.10\.49)"){ > rewritehostport("10.10.10.5:5060"); > log(1,"10 digit exp match w/leading 1"); > route(2); > break; > }; > #forward international N digit requests to gateway > if(uri=~"^sip:011[0-9]+@(mydomain\.com|10\.10\.10\.49)"){ > rewritehostport("10.10.10.5:5060"); > log(1,"international expression match"); > route(2); > break; > }; > >/* ********** VOICEMAIL logic ************* */ > > if (uri=~"^sip:voicemail\+@"){ > log(1,"sip:voicemail uri match"); > route(3); > break; > }; > >/* ****** Find Aliases and Locations of users ********* */ ># It is very important to lookup "aliases" before looking up "locations" > > if(!lookup("aliases")){ > log(1,"Couldn't find any matching alias"); > sl_send_reply("404", "User does not exist"); > break; > }; > if(!lookup("location")) { > log(1,"unable to locate user"); > route(3); > break; > }; > > }; > > > # forward to current uri now; use stateful forwarding; that > # works reliably even if we forward from TCP to UDP > log(1,"routing at eof: should not occur for outgoing PSTN calls"); > t_on_failure("1"); > if (!t_relay()) { > sl_reply_error(); > }; > log(1,"eof"); >} > >route[2]{ > log(1,"route[2]:SIP-to-PSTN call routed"); > if(!t_relay()){ > sl_reply_error(); > }; >} > >route[3]{ > log(1,"route[3]: voicemail processing"); > if(method=="ACK" || method=="INVITE" || method=="BYE" || method=="REFER"){ > log(1,"1st if entered in route[3] *vm*"); > if(t_newtran()){ > t_reply("100","Trying -- just a second"); > if(method=="INVITE" || method=="REFER"){ > log(1,"route[3]:method==INVITE || REFER"); > if(uri =~ "conference" ){ > if(!vm("/tmp/am_fifo","conference")){ > log(1,"could not contact conference server"); > t_reply("500","could not contact conference server"); > }; > } > else if (uri =~"echo"){ > if(!vm("/tmp/am_fifo","echo")){ > log(1,"could not contact echo"); > t_reply("500","could not contact echo"); > }; > } > else{ > if(!vm("/tmp/am_fifo","voicemail")){ > log(1,"vm module called and failed"); > t_reply("500", "voicemail error"); > }; > }; > break; > }; > if(method=="BYE"){ > log(1,"vm end/refer - begin"); > if(!vm("/tmp/am_fifo","bye")){ > log(1,"could not contact the media server"); > t_reply("500" , "could not contact the media server"); > }; > break; > }; > } > else{ > log(1,"route[3]:could not create new transaction"); > sl_send_reply("500", "could not create new transaction"); > }; > log(1,"route[3]:end of first method== check"); > }; > >} > >failure_route[1]{ > log(1,"failure_route[1]:jump to vm: route[3]"); > route(3); >} > >interface: eth0 (10.10.10.0/255.255.255.0) >match: 5060 >### >U 10.10.10.5:53667 -> 10.10.10.49:5060 > INVITE sip:6609@10.10.10.49:5060 SIP/2.0..Via: SIP/2.0/UDP 10.10.10.5:5060..From: <sip:6502188884@10.10.10. > 5>;tag=4CCDE44-B57..To: <sip:6609@10.10.10.49>..Date: Mon, 01 Mar 1993 22:22:15 GMT..Call-ID: A74AE3D-15AA11 > CC-8071812E-8BB375E1@10.10.10.5..Supported: timer,100rel..Min-SE: 1800..Cisco-Guid: 175259037-363467212-215 > 4725678-2343794145..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, > COMET, REFER, SUBSCRIBE, NOTIFY, INFO..CSeq: 101 INVITE..Max-Forwards: 6..Remote-Party-ID: <sip:6502188884@1 > 0.10.10.5>;party=calling;screen=yes;privacy=off..Timestamp: 731024535..Contact: <sip:6502188884@10.10.10.5:5 > 060>..Expires: 180..Allow-Events: telephone-event..Content-Type: application/sdp..Content-Length: 185....v=0 > ..o=CiscoSystemsSIP-GW-UserAgent 6694 6444 IN IP4 10.10.10.5..s=SIP Call..c=IN IP4 10.10.10.5..t=0 0..m=audi > o 18640 RTP/AVP 0..c=IN IP4 10.10.10.5..a=rtpmap:0 PCMU/8000..a=ptime:20.. ># >U 10.10.10.49:5060 -> 10.10.10.5:5060 > SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 10.10.10.5:5060..From: <sip:6502188884 > @10.10.10.5>;tag=4CCDE44-B57..To: <sip:6609@10.10.10.49>..Call-ID: A74AE3D-15AA11CC-8071812E-8BB375E1(a)10.10. > 10.5..CSeq: 101 INVITE..Server: Sip EXpress router (0.8.11 (i386/linux))..Content-Length: 0.... ># >U 10.10.10.49:5060 -> 10.10.10.189:5060 > INVITE sip:esavelle@10.10.10.189 SIP/2.0..Record-Route: <sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..Via: SIP > /2.0/UDP 10.10.10.49;branch=z9hG4bKaa33.8c7a9b23.0..Via: SIP/2.0/UDP 10.10.10.5:5060..From: <sip:6502188884 > @10.10.10.5>;tag=4CCDE44-B57..To: <sip:6609@10.10.10.49>..Date: Mon, 01 Mar 1993 22:22:15 GMT..Call-ID: A74A > E3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..Supported: timer,100rel..Min-SE: 1800..Cisco-Guid: 175259037-363 > 467212-2154725678-2343794145..User-Agent: Cisco-SIPGateway/IOS-12.x..Allow: INVITE, OPTIONS, BYE, CANCEL, AC > K, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO..CSeq: 101 INVITE..Max-Forwards: 5..Remote-Party-ID: <sip:65 > 02188884(a)10.10.10.5>;party=calling;screen=yes;privacy=off..Timestamp: 731024535..Contact: <sip:6502188884@10 > .10.10.5:5060>..Expires: 180..Allow-Events: telephone-event..Content-Type: application/sdp..Content-Length: > 185....v=0..o=CiscoSystemsSIP-GW-UserAgent 6694 6444 IN IP4 10.10.10.5..s=SIP Call..c=IN IP4 10.10.10.5..t=0 > 0..m=audio 18640 RTP/AVP 0..c=IN IP4 10.10.10.5..a=rtpmap:0 PCMU/8000..a=ptime:20.. ># >U 10.10.10.189:5060 -> 10.10.10.49:5060 > SIP/2.0 100 trying..Via: SIP/2.0/UDP 10.10.10.49;branch=z9hG4bKaa33.8c7a9b23.0..Via: SIP/2.0/UDP 10.10.10.5 > :5060..Record-Route: <sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..From: <sip:6502188884@10.10.10.5>;tag=4CCDE > 44-B57..To: <sip:6609@10.10.10.49>..Call-ID: A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101 INVITE > ..User-Agent: Grandstream SIP UA 1.0.3.81..Content-Length: 0.... ># >U 10.10.10.189:5060 -> 10.10.10.49:5060 > SIP/2.0 486 ..Via: SIP/2.0/UDP 10.10.10.49;branch=z9hG4bKaa33.8c7a9b23.0..Via: SIP/2.0/UDP 10.10.10.5:5060. > .Record-Route: <sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..From: <sip:6502188884@10.10.10.5>;tag=4CCDE44-B57 > ..To: <sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-08924feebae5..Call-ID : A74AE3D-15AA11CC-8071812E-8B > B375E1@10.10.10.5..CSeq: 101 INVITE..User-Agent: Grandstream SIP UA 1.0.3.81..Content-Length: 0.... >## >U 10.10.10.49:5060 -> 10.10.10.5:5060 > SIP/2.0 500 could not create new transaction..Via: SIP/2.0/UDP 10.10.10.5:5060..From: <sip:6502188884@10.10 > .10.5>;tag=4CCDE44-B57..To: <sip:6609@10.10.10.49>;tag=b27e1a1d33761e85846fc98f5f3a7e58.c9a7..Call-I D: A74AE > 3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101 INVITE..Server: Sip EXpress router (0.8.11 (i386/linux)) > ..Content-Length: 0.... ># >U 10.10.10.49:5060 -> 10.10.10.5:5060 > SIP/2.0 486 ..Via: SIP/2.0/UDP 10.10.10.5:5060..Record-Route: <sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..F > rom: <sip:6502188884@10.10.10.5>;tag=4CCDE44-B57..To: <sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-089 > 24feebae5..Call-ID: A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101 INVITE..User-Agent: Grandstream > SIP UA 1.0.3.81..Content-Length: 0.... ># >U 10.10.10.49:5060 -> 10.10.10.5:5060 > SIP/2.0 486 ..Via: SIP/2.0/UDP 10.10.10.5:5060..Record-Route: <sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..F > rom: <sip:6502188884@10.10.10.5>;tag=4CCDE44-B57..To: <sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-089 > 24feebae5..Call-ID: A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101 INVITE..User-Agent: Grandstream > SIP UA 1.0.3.81..Content-Length: 0.... ># >U 10.10.10.5:53667 -> 10.10.10.49:5060 > ACK sip:6609@10.10.10.49:5060 SIP/2.0..Via: SIP/2.0/UDP 10.10.10.5:5060..From: <sip:6502188884@10.10.10.5>; > tag=4CCDE44-B57..To: <sip:6609@10.10.10.49>;tag=b27e1a1d33761e85846fc98f5f3a7e58.c9a7..Date: Mon, 01 Mar 199 > 3 22:22:15 GMT..Call-ID: A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..Max-Forwards: 6..Content-Length: 0.. > CSeq: 101 ACK.... ># >U 10.10.10.49:5060 -> 10.10.10.5:5060 > SIP/2.0 486 ..Via: SIP/2.0/UDP 10.10.10.5:5060..Record-Route: <sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..F > rom: <sip:6502188884@10.10.10.5>;tag=4CCDE44-B57..To: <sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-089 > 24feebae5..Call-ID: A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101 INVITE..User-Agent: Grandstream > SIP UA 1.0.3.81..Content-Length: 0.... >### >U 10.10.10.49:5060 -> 10.10.10.5:5060 > SIP/2.0 486 ..Via: SIP/2.0/UDP 10.10.10.5:5060..Record-Route: <sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..F > rom: <sip:6502188884@10.10.10.5>;tag=4CCDE44-B57..To: <sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-089 > 24feebae5..Call-ID: A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101 INVITE..User-Agent: Grandstream > SIP UA 1.0.3.81..Content-Length: 0.... >##### >U 10.10.10.49:5060 -> 10.10.10.5:5060 > SIP/2.0 486 ..Via: SIP/2.0/UDP 10.10.10.5:5060..Record-Route: <sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..F > rom: <sip:6502188884@10.10.10.5>;tag=4CCDE44-B57..To: <sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-089 > 24feebae5..Call-ID: A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101 INVITE..User-Agent: Grandstream > SIP UA 1.0.3.81..Content-Length: 0.... >########### >U 10.10.10.49:5060 -> 10.10.10.5:5060 > SIP/2.0 486 ..Via: SIP/2.0/UDP 10.10.10.5:5060..Record-Route: <sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..F > rom: <sip:6502188884@10.10.10.5>;tag=4CCDE44-B57..To: <sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-089 > 24feebae5..Call-ID: A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101 INVITE..User-Agent: Grandstream > SIP UA 1.0.3.81..Content-Length: 0.... >########### >U 10.10.10.49:5060 -> 10.10.10.5:5060 > SIP/2.0 486 ..Via: SIP/2.0/UDP 10.10.10.5:5060..Record-Route: <sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..F > rom: <sip:6502188884@10.10.10.5>;tag=4CCDE44-B57..To: <sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-089 > 24feebae5..Call-ID: A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101 INVITE..User-Agent: Grandstream > SIP UA 1.0.3.81..Content-Length: 0.... >######## >U 10.10.10.49:5060 -> 10.10.10.5:5060 > SIP/2.0 486 ..Via: SIP/2.0/UDP 10.10.10.5:5060..Record-Route: <sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..F > rom: <sip:6502188884@10.10.10.5>;tag=4CCDE44-B57..To: <sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-089 > 24feebae5..Call-ID: A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101 INVITE..User-Agent: Grandstream > SIP UA 1.0.3.81..Content-Length: 0.... >###### >U 10.10.10.49:5060 -> 10.10.10.5:5060 > SIP/2.0 486 ..Via: SIP/2.0/UDP 10.10.10.5:5060..Record-Route: <sip:6609@10.10.10.49;ftag=4CCDE44-B57;lr>..F > rom: <sip:6502188884@10.10.10.5>;tag=4CCDE44-B57..To: <sip:6609@10.10.10.49>;tag=c02cab98-902a-cad0-7e53-089 > 24feebae5..Call-ID: A74AE3D-15AA11CC-8071812E-8BB375E1@10.10.10.5..CSeq: 101 INVITE..User-Agent: Grandstream > SIP UA 1.0.3.81..Content-Length: 0.... >############################exit >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] Not been able to log on
by Jiri Kuthan 07 Oct '03

07 Oct '03

Are you sure your server is really running? -jiri At 07:15 PM 10/7/2003, Annie Sasidar wrote: >Hi, > I am using windows messenger 5.0. This version supports SIP but i am not >able to log on as a first time user. The error i get is : >You have been signed out of SIP communications service because that service >has been temporarily shut down. Please try again later. > >Is there any other free SIP client that experts could advice which works >with SER 0.8-11. > > >Thanks in advance, >Annie > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

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Re: [Serusers] Re: Serusers Digest, Vol 5, Issue 63
by jaime.gil＠orange.co.uk 07 Oct '03

07 Oct '03

Steven, I remember having the same problem. In my case, the problem seemed to be in the loose_route processing. I don't know what it has to do with the accounting module, but commenting out this part made the trick in my case. Jaime "Steven R. Bunin" <steve(a)solaas.com> on 26/09/2003 17:11:41 To: serusers(a)lists.iptel.org cc: (bcc: Jaime GIL/EN/HTLUK) Subject: [Serusers] Re: Serusers Digest, Vol 5, Issue 63 Hi all, I have successfully gotten radius authentication working and I started getting Radius Start records for accounting but I am not sure what I am doing wrong in regards to getting radius Stop records. Below is the area I believe has the most affect on Radius Acccounting from my Log File. Any suggestions would be appreciated and if it would help to see the full Config file I will send it as well. record_route(); if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save("location"); break; }; if (method =="INVITE") { log(1,"INVITE\n"); setflag(1); }; if (method=="MESSAGE") { log(1,"MESSAGE\n"); setflag(1); }; if (method=="BYE"){ log (1, "BYE or CANCEL\n"); setflag(1); }; if (method=="CANCEL"){ log (1, "BYE or CANCEL\n"); setflag(1); }; Thanks in advance, Steve _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ******************************************************************************* Important. Confidentiality: This communication is intended for the above-named person and may be confidential and/or legally privileged. Any opinions expressed in this communication are not necessarily those of the company. If it has come to you in error you must take no action based on it, nor must you copy or show it to anyone; please delete/destroy and inform the sender immediately. Monitoring/Viruses Orange may monitor all incoming and outgoing emails in line with current legislation. Although we have taken steps to ensure that this email and attachments are free from any virus, we advise that in keeping with good computing practice the recipient should ensure they are actually virus free. Orange PCS Limited is a subsidiary of Orange SA and is registered in England No 2178917, with its address at St James Court, Great Park Road, Almondsbury Park, Bradley Stoke, Bristol BS32 4QJ. *******************************************************************************

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sr-users October 2003