October 2003 - sr-users

[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/

2 1

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/

1 0

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. *******************************************************************************

6 15

[Serusers] Re: Can you help me!
by 郝育鹏 07 Oct '03

07 Oct '03

Hello, How are you! I'am glad to receiving your letter! I'd give my question information.I beg your help! Thank you very much! 1,operating system: my Redhat7.3 (Linux 2.4.18-3) 2,SER distribution: ser 0.8.11 (i386/linux) 3, SER configuration file : # $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=8 # debug level (cmd line: -dddddddddd) #fork=no #log_stderror=yes # (cmd line: -E) alias="Lserver.net" # 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" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/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") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # 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; }; # 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") { # Uncomment this if you want to use digest authentication if (!www_authorize("Lserver.sict.net", "subscriber")) { www_challenge("Lserver.sict.net", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } 4, SER logs : When I apply the following configuration options: debug=8, log_stderror=yes, fork=no. show this information: ..... 0(0) DEBUG: udp_init: trying SO_RCVBUF: 262140 0(0) DEBUG: setting SO_RCVBUF; set=262140,verify=131070 0(0) DEBUG: setting SO_RCVBUF has no effect 0(0) DEBUG: udp_init: trying SO_RCVBUF: 133118 0(0) DEBUG: setting SO_RCVBUF; set=133118,verify=131070 0(0) DEBUG: setting SO_RCVBUF has no effect 0(0) INFO: udp_init: SO_RCVBUF is finally 131070 0(0) WARNING: using only the first listen address (no fork) 1(4499) DBG: open_uac_fifo: opening fifo... 1(4499) DEBUG: fifo /tmp/ser_fifo opened, mode=438 0(4497) DEBUG: init_mod_child (1): tm 0(4497) DEBUG: callid: '763e7fde-4497(a)127.0.0.1' 0(4497) DEBUG: init_mod_child (1): usrloc 1(4499) DEBUG: init_mod_child (-1): tm 1(4499) DEBUG: callid: '763e7fde-4499(a)127.0.0.1' 1(4499) DEBUG: init_mod_child (-1): usrloc 2(4500) INFO: fifo process starting: 4500 2(4500) DEBUG: init_mod_child (-2): tm 2(4500) DEBUG: callid: '763e7fde-4500(a)127.0.0.1' 2(4500) DEBUG: init_mod_child (-2): usrloc 1(4499) DEBUG: init_mod_child (-1): auth_db 0(4497) DEBUG: init_mod_child (1): auth_db 2(4500) DEBUG: init_mod_child (-2): auth_db 2(4500) SER: open_uac_fifo: fifo server up at /tmp/ser_fifo... It's no longer running,always stop! why? My SER log is : .... Oct 7 13:30:05 Lserver ser: WARNING: could not rev. resolve 210.72.128.44 Oct 7 13:30:05 Lserver 10月 7 13:30:05 ser: Listening on Oct 7 13:30:05 Lserver 10月 7 13:30:05 ser: 127.0.0.1 [127.0.0.1 ]:5060 Oct 7 13:30:05 Lserver 10月 7 13:30:05 ser: 192.168.0.179 [192.1 68.0.179]:5060 Oct 7 13:30:05 Lserver 10月 7 13:30:05 ser: 210.72.128.44 [210.7 2.128.44]:5060 Oct 7 13:30:05 Lserver 10月 7 13:30:05 ser: Aliases: Lserver.sict.net:5060 loc alhost:5060 localhost.localdomain:5060 Lserver:5060 Lserver.net:* Oct 7 13:30:06 Lserver 10月 7 13:30:05 ser: ser startup succeeded Oct 7 13:49:07 Lserver 10月 7 13:49:07 ser: ser shutdown failed Oct 7 13:50:42 Lserver ser: parse error (14,1-2): parse error Oct 7 13:50:42 Lserver ser: parse error (14,1-2): Oct 7 13:51:24 Lserver ser: WARNING: could not rev. resolve 210.72.128.44 [root@Lserver /]# ----- Original Message ----- From: "Daniel-Constantin Mierla" <Daniel-Constantin.Mierla(a)fokus.fraunhofer.de> To: "郝育鹏" <haoypeng(a)sict.ac.cn> Cc: <serusers(a)iptel.org> Sent: Monday, October 06, 2003 7:26 PM Subject: Re: [Serusers] Can you help me! Hello, did you add some users into database? Please follow the guidelines from http://www.iptel.org/ser/bugs/ and send us the result. We can't guess everything what happen into your system only from config file. You have commented some global parameters and that changed the behavior SER should have with default config file -- you even commented the number of children that means ser listens only on first network interface (I have to double check it, but it should be like that), and that is usually 127.0.0.1. We need the log messages, network dumps and so on ... Regards, "Daniel

3 2

[Serusers] Re:Can you help me
by 郝育鹏 07 Oct '03

07 Oct '03

Hello! How are you! Thank you! I have seen! But when I request SER,SER server start to continue running and show some question: ....... 5(5251) DEBUG: add_param: epid=78fc235044 5(5251) end of header reached, state=29 5(5251) parse_headers: flags=256 5(5251) end of header reached, state=9 5(5251) DEBUG: get_hdr_field: <To> [33]; uri=[sip:haoypeng@Lserver.sict.net] 5(5251) DEBUG: to body [<sip:haoypeng@Lserver.sict.net> 5(5251) get_hdr_field: cseq <CSeq>: <1> <REGISTER> 5(5251) DEBUG: get_hdr_body : content_length=0 5(5251) found end of header 5(5251) find_first_route(): No Route headers found 5(5251) loose_route(): There is no Route HF 5(5251) check_self - checking if host==us: 16==9 && [Lserver.sict.net] == [127 0.0.1] 5(5251) check_self - checking if host==us: 16==13 && [Lserver.sict.net] == [19 .168.0.179] 5(5251) check_self - checking if host==us: 16==13 && [Lserver.sict.net] == [21 .72.128.44] 5(5251) parse_headers: flags=4096 5(5251) pre_auth(): Credentials with given realm not found 5(5251) build_auth_hf(): 'WWW-Authenticate: Digest realm="Lserver.sict.net", no ce="3f82ae01bb9e5c5720b768839131805780b4fc95" 5(5251) parse_headers: flags=-1 5(5251) check_via_address(192.168.0.244, 192.168.0.244, 3) 5(5251) receive_msg: cleaning up Why happen that Above ? I try testing manny times. But always like above. ----- Original Message ----- From: "Daniel-Constantin Mierla" <Daniel-Constantin.Mierla(a)fokus.fraunhofer.de> To: "郝育鹏" <haoypeng(a)sict.ac.cn> Cc: <Serusers(a)iptel.org> Sent: Tuesday, October 07, 2003 4:57 PM Subject: Re: [Serusers] Re: Can you help me! Hello, On 10/7/2003 8:29 AM, 郝育鹏 wrote: > [...] > > 4, SER logs : > as I told you, only the first network interface is used and that is 127.0.0.1 > > > > It's no longer running,always stop! why? My SER log is : Are you sure is not running? It stops here until a SIP request is received and then displays other debug messages. Here you had some errors in your config file (line 14), ser didn't start at that moment. Regards, }Daniel

2 1

RE: [Serusers] caller-id with raius using sip-rpid
by Steve Dolloff 07 Oct '03

07 Oct '03

OK, I understand now. The caller-id works correctly when I call out to the PSTN through a Cisco AS5350 gateway. The caller-id does not work when I call another ATA. I have to assume that the ATA does not support the remote-party-id field. Anyone else have experience with this, since I don't see any documentation related to remote-party-id on an ATA? I'll also post this to the cisco message boards and re-post any response that I get. Stephen Subject: RE: [Serusers] caller-id with raius using sip-rpid Well, as you see, rpid is attached to a non-numeric subscriber's request, so that it can be processed by devices capable of doing so. What is the problem then? -jiri At 11:21 PM 10/3/2003, Steve Dolloff wrote: >I want caller-id to work for a subscriber who has a non-numeric >username. > >Subject: Re: [Serusers] caller-id with raius using sip-rpid > >There is a Remote-Party-ID header in your message. What else do you >want ? Please be more specific and provide some examples, your >terminology is >hard to follow for me, sorry. > > Jan. > >On 03-10 16:05, Steve Dolloff wrote: >> Calling from one ata to another, proxied and registered through ser, >> here is my ngrep port 5060 output. Remote-party-id is being set but >not >> the call-id. >> >> U 219.242.10.153:5060 -> 226.145.234.80:5060 >> INVITE sip:256@226.145.234.80:5060;user=phone;transport=udp >> SIP/2.0..Max-Fo >> rwards: 10..Record-Route: >> <sip:256@219.242.10.153;ftag=2441840520;lr=on>..V >> ia: SIP/2.0/UDP 219.242.10.153;branch=z9hG4bK41f3.9e2c3976.0..Via: >> SIP/2.0/ >> UDP 226.145.234.113:5060;rport=5060..From: >> sip:test@voip2.test.net;tag=24418 >> 40520..To: <sip:256@voip2.test.net;user=phone>..Call-ID: >> 2303820380(a)226.145. >> 234.113..CSeq: 2 INVITE..Contact: >> <sip:test@226.145.234.113:5060;transport= >> udp>..User-Agent: Cisco ATA 186 v2.16.2 ata18x >> (030909a)..Authorization: D >> igest >> username="test",realm="voip2.test.net",nonce="3f7de542de99d4e832372e6b >> >> >812a73562d346e6f",uri="sip:256@voip2.test.net",response="1a4552ce6c6533 3 >> 439b >> 6a97672f415a5"..Expires: 300..Content-Length: 271..Content-Type: >> applicatio >> n/sdp..Remote-Party-ID: >> 222;party=calling;id-type=subscriber;screen=no....v >> =0..o=test 5648 5648 IN IP4 226.145.234.113..s=ATA186 Call..c=IN IP4 >> 226.14 >> 5.234.113..t=0 0..m=audio 16384 RTP/AVP 0 4 8 101..a=rtpmap:0 >> PCMU/8000/1.. >> a=rtpmap:4 G723/8000/1..a=rtpmap:8 PCMA/8000/1..a=rtpmap:101 >> telephone-even >> t/8000..a=fmtp:101 0-15..a=direction:active.. >> # >> U 226.145.234.80:5060 -> 219.242.10.153:5060 >> SIP/2.0 100 Trying..Via: SIP/2.0/UDP >> 219.242.10.153;branch=z9hG4bK41f3.9e2c >> 3976.0..Via: SIP/2.0/UDP 226.145.234.113:5060;rport=5060..From: >> sip:test@vo >> ip2.test.net;tag=2441840520..To: >> <sip:256@voip2.test.net;user=phone>;tag=2077 >> 559661..Call-ID: 2303820380@226.145.234.113..CSeq: 2 INVITE..Server: >> Cisco >> ATA 186 v2.16.2 ata18x (030909a)..Content-Length: 0.... >> >> >> Subject: Re: [Serusers] caller-id with raius using sip-rpid >> >> Is there any Remote-Party-ID header field in outgoing INVITE ? If so >> then your gateway/whatever ignores it. >> >> Jan. >> >> On 02-10 15:18, Steve Dolloff wrote: >> > Any phone that supports caller-id. But I haven't gotten to work. >I'm >> > seeing the username instead. >> > >> > >> > Subject: RE: [Serusers] caller-id with raius using sip-rpid >> > >> > Which phone device should I use to see caller ID >> > information when using the ATA186? >> > >> > Alejandro >> > >> > --- Steve Dolloff <sdolloff(a)noc.test.net> escreveu: > >> > At the top of the config I have the following >> > > statement.... >> > > >> > > if (method=="INVITE") { >> > > record_route(); >> > > if (!radius_www_authorize("")) { >> > > log(1,"radius auth >> > > failure"); >> > > >> > > www_challenge("voip2.test.net","0"); >> > > break; >> > > }; >> > > append_rpid_hf(); >> > > }; >> > > >> > > If I call a phone with caller-id and my username is >> > > 256, it shows 256 on >> > > the phone, if I call with a username of test, it >> > > shows 4534 on the >> > > handset even though I have the sip-rpid set to 222. >> > > >> > > Stephen >> > > >> > > >> > > Subject: Re: [Serusers] caller-id with raius using >> > > sip-rpid >> > > >> > > Retrieving caller-id when users register doesn't >> > > work because REGISTER >> > > messages are processed by the server and the server >> > > generates a reply >> > > only. >> > > >> > > What you need is to insert Remote-Party-ID header >> > > field into INVITE >> > > message, to make it work you must authenticate also >> > > INVITE messages. >> > > My guess is that you do not authenticate INVITE >> > > messages and therefore >> > > there is nothing append_rpid_hf can add. >> > > >> > > Jan. >> > > >> > > On 02-10 11:35, Steve Dolloff wrote: >> > > > I want to retrieve the caller-id of a user from >> > > radius when they >> > > > register and use it when I redirect an invite >> > > request to the voicemail >> > > > system so that the voicemail system can route the >> > > call based on >> > > > caller-id. I also want to be able to send it to >> > > the sip gateway when >> > > I >> > > > am routing calls to the pstn. >> > > > >> > > > I have this in my config currently and it doesn't >> > > appear to set the >> > > > caller-id in the invite message. >> > > > >> > > > >> > > rewritehostport("219.242.10.153:5061"); >> > > > append_rpid_hf(); >> > > > t_relay(); >> > > > >> > > > >> > > > Subject: Re: [Serusers] caller-id with raius using >> > > sip-rpid >> > > > >> > > > append_rpid_hf has no parameters. Version with 2 >> > > parameters is in >> > > > unstable branch in the CVS only. What exactly do >> > > you want to do ? >> > > > >> > > > Jan. >> > > > >> > > > On 02-10 11:26, Steve Dolloff wrote: >> > > > > I found this example in a previous Serusers >> > > message. >> > > > > >> > > > > append_rpid_hf("<sip:+", >> > > > > >> > > > > >> > > > >> > > >> > >> >"@zettou.net>;party=calling;id-type=subscriber;privacy=off;screen=yes") ; >> > > > > >> > > > > I tried calling append_rpid_hf();, but it does >> > > nothing. >> > > > > >> > > > > I also tried >> > > > > >> > > > >> > > >> > >> >append_rpid_hf("party=calling;id-type=subscriber;privacy=off;screen=yes " >> > > > > ); but I get an unknown command presumably >> > > because I am not using >> > > the >> > > > > right number of parameters. >> > > > > >> > > > > I only want to modify the calling-id info. >> > > > > >> > > > > Can anyone provide an example? >> > > > > >> > > > > -----Original Message----- >> > > > > From: Jan Janak [mailto:jan@iptel.org] >> > > > > Sent: Thursday, October 02, 2003 11:09 AM >> > > > > To: Steve Dolloff >> > > > > Cc: serusers(a)lists.iptel.org >> > > > > Subject: Re: [Serusers] caller-id with raius >> > > using sip-rpid >> > > > > >> > > > > If you want to add Remote-Party-ID header field >> > > then you have to >> > > call >> > > > > append_rpid_hf function in your script. >> > > > > >> > > > > Jan. >> > > > > >> > > > > On 02-10 11:07, Steve Dolloff wrote: >> > > > > > OK, I am setting up a Voicemail system (using >> > > asterisk) and I am >> > > > > > currently doing a rewritehostport(ip:port) and >> > > then trelay() to >> > > send >> > > > > it >> > > > > > to the voicemail system if an invite fails. >> > > > > > >> > > > > > Should I change something? See my ser.cfg and >> > > output from the call >> > > > to >> > > > > > the vm. >> > > > > > >> > > > > > Here is the code from ser.cfg >> > > > > > >> > > > > > >> > > > > rewritehostport("219.242.10.153:5061"); >> > > > > > t_relay(); >> > > > > > >> > > > > > Here is the sip info from asterisk. >> > > > > > >> > > > > > INVITE sip:200@219.242.10.153:5061;user=phone >> > > SIP/2.0 >> > > > > > Max-Forwards: 10 >> > > > > > Record-Route: >> > > <sip:200@219.242.10.153;ftag=2236658534;lr=on> >> > > > > > Via: SIP/2.0/UDP >> > > 219.242.10.153;branch=z9hG4bK13de.31901234.0 >> > > > > > Via: SIP/2.0/UDP >> > > 226.145.234.113:5060;rport=5060 >> > > > > > From: sip:test@voip2.test.net;tag=2236658534 >> > > > > > To: <sip:200@voip2.test.net;user=phone> >> > > > > > Call-ID: 2218108971(a)226.145.234.113 >> > > > > > CSeq: 1 INVITE >> > > > > > Contact: >> > > <sip:test@226.145.234.113:5060;transport=udp> >> > > > > > User-Agent: Cisco ATA 186 v2.16.2 ata18x >> > > (030909a) >> > > > > > Expires: 300 >> > > > > > Content-Length: 271 >> > > > > > Content-Type: application/sdp >> > > > > > >> > > > > > >> > > > > > >> > > > > > Subject: Re: [Serusers] caller-id with raius >> > > using sip-rpid >> > > > > > >> > > > > > Remote-Party-ID header field is inserted into >> > > forwarded requests, >> > > > not >> > > > > > responses. >> > > > > > >> > > > > > Jan. >> > > > > > >> > > > > > On 02-10 10:53, Steve Dolloff wrote: >> > > > > > > Ser doesn't appear to be passing the >> > > Caller-id to the ata at >> > > auth >> > > > or >> > > > > I >> > > > > > > am doing something wrong. Can anyone point >> > > me in the right >> > > > > direction? >> > > > > > > >> > > > > > > Thanks, >> > > > > > > >> > > > > > > Stephen >> > > > > > > >> > > > > > > I have the following entry in my freeradius >> > > users file. >> > > > > > > >> > > > > > > test(a)voip2.test.net Auth-Type := Digest, >> > > User-Password == "test" >> > > > > > > Reply-Message = "Hello, test with >> > > digest", Sip-Rpid = >> > > > > > > "8472222222" >> > > > > > > >> > > > > > > When I run a radclient test I get the >> > > correct info.. >> > > > > > > >> > > > > > > radclient -f digest.test 219.242.10.153:1812 >> > > auth testing >> > > > > > > Received response ID 134, code 2, length = >> > > 57 >> > > >> > === message truncated === >> > >> > Yahoo! Mail - o melhor webmail do Brasil >> > http://mail.yahoo.com.br > >_______________________________________________ >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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[Serusers] CPL editor
by Kevin Chu 07 Oct '03

07 Oct '03

Hi Bogdan, I am interested for using the cpl-c module with ser. Is that necessary to use mySQL with cpl-c? Can I use dbtext instead? Could you provide me more instruction and sample ser.cfg for using cpl-c? Thanks! BR, Kevin ------------------------------------------------------------- Date: Fri, 19 Sep 2003 21:49:54 +0200 From: Bogdan-Andrei IANCU <iancu(a)fokus.fraunhofer.de> Subject: [Serusers] CPL editor To: sr(a)lists.iptel.org, serusers(a)lists.iptel.org Message-ID: <3F6B5DE2.1090301(a)fokus.fraunhofer.de> Content-Type: text/plain; charset=us-ascii; format=flowed A new release (1.2) of CPLEd is available now at: https://developer.berlios.de/projects/cpled/ Works great against ser with the new cpl-c modules ;-) Bogdan

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[Serusers] newbie to SIP Express Router
by Ajeet 07 Oct '03

07 Oct '03

Hi, I am newbie to ser, and wnat to install and configure ser on my computer. Still I am reading infomration on internet for the configuration and installation of ser, but any information provided on this forum would certainly help me a lot. Here are the details. My OS RedHat linux 9. i have downloaded ser-0.8.11-0.i386.rpm and installed it simply by command rpm -i ser-0.8.11-0.i386.rpm what to do next? Why i need ser? Actually i have to show working ip-telephony service as a requirement in our course. The scenario is that, we have our private LAN (intranet) and i have to install and configure the ip-telephony service so that one cleint on the intranet can call other client on the sane intranet, i will use either Kphone or BonePhone as a client. So if some one can help me in establishing this service. I dont need to call outside my intranet or even i dont need some advanced features like security/authentication or any other thing other thing. -ajeet. stockholm. -- Ajeet ajeetnk(a)fastmail.fm -- http://www.fastmail.fm - IMAP accessible web-mail

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Re: [Serusers] Multple SER servers on same domain
by Tjardick van der Kraan 07 Oct '03

07 Oct '03

sorry i forgot the list cc.. ----- Original Message ----- From: "Tjardick van der Kraan" <tjardick(a)vanderkraan.net> To: "Jan Janak" <jan(a)iptel.org> Sent: Tuesday, October 07, 2003 11:27 AM Subject: Re: [Serusers] Multple SER servers on same domain > Hi Jan, > > Well what i'm looking into is putting SER up on our wireless community > Network. > > But as that as different segments and every now and then a wireless link > drops out i was thinking about creating some sort of islands which each > having their local SER server (.e.g set the dns on to a local ip for that > area) but in case the interconnection between the 'islands' is up and > running it would be nice if i on one 'island' would dial someone on another > island that the connection still could be setup. > > e.g an INVITE would go to local server > > but the local server (incase that user isn't logged in there) would > go and try server1, server2, etc. > > Hope this explains a bit more what i mean. > > Greetings, > > Tjardick > ----- Original Message ----- > From: "Jan Janak" <jan(a)iptel.org> > To: "Tjardick van der Kraan" <tjardick(a)vanderkraan.net> > Cc: <serusers(a)lists.iptel.org> > Sent: Tuesday, October 07, 2003 11:13 AM > Subject: Re: [Serusers] Multple SER servers on same domain > > > > Hello, > > > > I am not sure I fully understood. Let me ask one question, what is this > > good for ? > > > > You can certainly have several ser servers, each of them could have it's > > own location table (i.e. it's own database of currently registered > > users) and all of them can share the same subscriber table (i.e. you > > will have one common source of usernames and passwords for all the > > servers). Is that what are you looking for ? > > > > What you want can be probably accomplished using one server too. > > > > Jan. > > > > On 06-10 14:38, Tjardick van der Kraan wrote: > > > Hello Everyone, > > > > > > Just a new face on the mailinglist (only been lurking so far) so just a > > > friendly hi from me to all of you. > > > > > > I was wondering if the following idea could be realised with the use of > SER > > > and some (advanced) routing rules. > > > > > > One main SER server with one or more 'slave' SER servers running at > their > > > own location with a synchronized user database, but with their own > presence > > > table. > > > > > > e.g. if someone on the main SER server asks for a connectiong with a > user on > > > a 'sub' ser server of the same domain, the main SER server looks up if > the > > > user is online on it's server if not it checks if the user is online on > one > > > (or more) sub servers. > > > > > > What i like to accomplish is multple local networks that run their own > ser > > > server and support their 'local' connections, but sometimes there is a > > > connection available to another local network, in which case more users > can > > > be reached. so on a request and fail of the user online on the local > server > > > it would try and see if it can connect to (any) other server and if so > query > > > if the user is online there. > > > > > > so everyone will be user(a)domain.com but it would point to the ip of the > > > local server. > > > > > > Any config examples on how to achive this if possible would really be > > > appreciated. > > > > > > Kind regards, > > > > > > Tjardick van der Kraan > > > > > > -- > > > Tjardick van der Kraan > > > tjardick(a)vanderkraan.net > > > > > > Tel +32 4 34 40 522 > > > Fax +32 4 34 40 525 > > > GSM +32 497 45 27 36 > > > > > > IAXtel: 1 700 344 0522 > > > FWD: 26322 > > > IPtel: 91331 > > > > > > Rue des Pocheteux 10 > > > 4020 Jupille > > > Belgium > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > >

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[Serusers] SMS device to test
by Soo Wei Kang 07 Oct '03

07 Oct '03

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

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[Serusers] Hello
by Hassan CHOUMAR 07 Oct '03

07 Oct '03

Hello, I have a problem in the recording of server SER, as soon as I put the personal information during the phase of recording about the serweb to assign an address personal SIP, it sends me a message to confirm me that a mail will be sent in my inbox mail. But I receive any mail in my inbox mail. A person know Where comes the problem. Hassan ------------------------------------------------- This mail sent through IMP: http://horde.org/imp/

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[Serusers] How can we connect calls of two independent unknown SIP domains.
by John Foster 07 Oct '03

07 Oct '03

Hi all, Here is a scenario, i m working on, If anyone can comment on it. Scenario: I have two SIP domains, abc.com and xvz.com sip.abc.com is running ser and has registered users with xxxx(a)abc.com similarly sip.xyz.com is serving users xxxx(a)xyz.com sip.abc.com and sip.xyz.com can be resolved through standard dns servers as SRV hosts for their respective domains. There is no explicitly defined routing procedure on ser.conf for other domain, in both SIP servers machines. Is it possible for xxxx(a)abc.com to register with sip.abc.com and place a call to user xxxx(a)xyz.com. While trying, i m getting no packet on sip.abc.com not sip.xyz.com for the subject call. Any comment? JF --------------------------------- Do you Yahoo!? The New Yahoo! Shopping - with improved product search

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[Serusers] Registrar.so error
by giAndrea 07 Oct '03

07 Oct '03

Hi, another problem. After compilation of new sources (0.8.12dev-17-cristian) i've found this problem in modules registrar.so during execution (/etc/init/ser start) Oct 6 16:19:09 webby ser: 0(0) fixing /usr/lib/ser/modules/registrar.so save Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x42134000, 16) called from dlist.c: new_dlist(156) Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x42134000, 16) returns address 0x422bea0c frag. 0x422be9f4 (size=16) on 1 -th hit Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x42134000, 8) called from dlist.c: new_dlist(163) Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x42134000, 8) returns address 0x422bea4c frag. 0x422bea34 (size=8) on 1 -th hit Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x42134000, 36) called from udomain.c: new_udomain(117) Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x42134000, 36) returns address 0x422bea84 frag. 0x422bea6c (size=36) on 1 -th hit Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x42134000, 8192) called from udomain.c: new_udomain(124) Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x42134000, 8192) returns address 0x422bead8 frag. 0x422beac0 (size=8192) on 1 -th hit Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x80c17c0, 9) called from db_con.c: use_table(53) Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x80c17c0, 12) returns address 0x80cb9dc frag. 0x80cb9c4 (size=12) on 1 -th hit Oct 6 16:19:09 webby ser: 0(0) submit_query(): Unknown column 'flags' in 'field list' Oct 6 16:19:09 webby ser: 0(0) submit_query(): Error while submitting query Oct 6 16:19:09 webby ser: 0(0) preload_udomain(): Error while doing db_query Oct 6 16:19:09 webby ser: 0(0) register_udomain(): Error while preloading domain 'location' Oct 6 16:19:09 webby ser: 0(0) qm_free(0x42134000, 0x422bead8), called from udomain.c: free_udomain(164) Oct 6 16:19:09 webby ser: 0(0) qm_free: freeing frag. 0x422beac0 alloc'ed from udomain.c: new_udomain(124) Oct 6 16:19:09 webby ser: 0(0) qm_free(0x42134000, 0x422bea84), called from udomain.c: free_udomain(168) Oct 6 16:19:09 webby ser: 0(0) qm_free: freeing frag. 0x422bea6c alloc'ed from udomain.c: new_udomain(117) Oct 6 16:19:09 webby ser: 0(0) qm_free(0x42134000, 0x422bea4c), called from dlist.c: register_udomain(218) Oct 6 16:19:09 webby ser: 0(0) qm_free: freeing frag. 0x422bea34 alloc'ed from dlist.c: new_dlist(163) Oct 6 16:19:09 webby ser: 0(0) qm_free(0x42134000, 0x422bea0c), called from dlist.c: register_udomain(219) Oct 6 16:19:09 webby ser: 0(0) qm_free: freeing frag. 0x422be9f4 alloc'ed from dlist.c: new_dlist(156) Oct 6 16:19:09 webby ser: 0(0) domain_fixup(): Error while registering domain Oct 6 16:19:09 webby ser: ERROR: error ffffffff while trying to fix configuration Oct 6 16:19:09 webby ser: 0(0) qm_free(0x80c17c0, 0x80c8558), called from sr_module.c: destroy_modules(303) Oct 6 16:19:09 webby ser: 0(0) qm_free: freeing frag. 0x80c8540 alloc'ed from sr_module.c: register_module(135) Oct 6 16:19:09 webby ser: 0(0) qm_free(0x80c17c0, 0x80cb93c), called from auth_mod.c: destroy(221) Oct 6 16:19:09 webby ser: 0(0) qm_free: freeing frag. 0x80cb924 alloc'ed from auth_mod.c: generate_random_secret(163) Oct 6 16:19:09 webby ser: 0(0) qm_free(0x80c17c0, 0x80c84c8), called from sr_module.c: destroy_modules(303) Oct 6 16:19:09 webby ser: 0(0) qm_free: freeing frag. 0x80c84b0 alloc'ed from sr_module.c: register_module(135) Oct 6 16:19:09 webby ser: 0(0) qm_free(0x80c17c0, 0x80cb89c), called from sdp_mangler.c: free_compiled_expresions(538) Oct 6 16:19:09 webby ser: 0(0) qm_free: freeing frag. 0x80cb884 alloc'ed from sdp_mangler.c: compile_expresions(499) Oct 6 16:19:09 webby ser: 0(0) qm_free(0x80c17c0, 0x80cb8ec), called from sdp_mangler.c: free_compiled_expresions(544) Oct 6 16:19:09 webby ser: 0(0) qm_free: freeing frag. 0x80cb8d4 alloc'ed from sdp_mangler.c: compile_expresions(515) Oct 6 16:19:09 webby ser: 0(0) qm_free(0x80c17c0, 0x80c8438), called from sr_module.c: destroy_modules(303) Oct 6 16:19:09 webby ser: 0(0) qm_free: freeing frag. 0x80c8420 alloc'ed from sr_module.c: register_module(135) Oct 6 16:19:09 webby ser: 0(0) qm_free(0x80c17c0, 0x80c83a8), called from sr_module.c: destroy_modules(303) Oct 6 16:19:09 webby ser: 0(0) qm_free: freeing frag. 0x80c8390 alloc'ed from sr_module.c: register_module(135) Oct 6 16:19:09 webby ser: 0(0) qm_free(0x80c17c0, 0x80c831c), called from sr_module.c: destroy_modules(303) Oct 6 16:19:09 webby ser: 0(0) qm_free: freeing frag. 0x80c8304 alloc'ed from sr_module.c: register_module(135) Oct 6 16:19:09 webby ser: 0(0) qm_free(0x80c17c0, 0x80c828c), called from sr_module.c: destroy_modules(303) Oct 6 16:19:09 webby ser: 0(0) qm_free: freeing frag. 0x80c8274 alloc'ed from sr_module.c: register_module(135) Oct 6 16:19:09 webby ser: 0(0) qm_free(0x80c17c0, 0x80c81f8), called from sr_module.c: destroy_modules(303) Oct 6 16:19:09 webby ser: 0(0) qm_free: freeing frag. 0x80c81e0 alloc'ed from sr_module.c: register_module(135) Oct 6 16:19:09 webby ser: 0(0) qm_free(0x80c17c0, 0x80cb57c), called from dbase.c: disconnect_db(128) Oct 6 16:19:09 webby ser: 0(0) qm_free: freeing frag. 0x80cb564 alloc'ed from dbase.c: connect_db(88) Oct 6 16:19:09 webby ser: 0(0) qm_free(0x80c17c0, 0x80cb9dc), called from dbase.c: db_close(321) Oct 6 16:19:09 webby ser: 0(0) qm_free: freeing frag. 0x80cb9c4 alloc'ed from db_con.c: use_table(53) Oct 6 16:19:09 webby ser: 0(0) qm_free(0x80c17c0, 0x80cb4e4), called from dbase.c: db_close(323) Oct 6 16:19:09 webby ser: 0(0) qm_free: freeing frag. 0x80cb4cc alloc'ed from dbase.c: db_init(286) Oct 6 16:19:09 webby ser: 0(0) qm_free(0x80c17c0, 0x80c8164), called from sr_module.c: destroy_modules(303) Oct 6 16:19:09 webby ser: 0(0) qm_free: freeing frag. 0x80c814c alloc'ed from sr_module.c: register_module(135) Oct 6 16:19:09 webby ser: 0(0) qm_free(0x80c17c0, 0x80c80d4), called from sr_module.c: destroy_modules(303) Oct 6 16:19:09 webby ser: 0(0) qm_free: freeing frag. 0x80c80bc alloc'ed from sr_module.c: register_module(135) Oct 6 16:19:09 webby ser: 0(0) qm_free(0x80c17c0, 0x80c8044), called from sr_module.c: destroy_modules(303) Oct 6 16:19:11 webby ser: 0(0) qm_free: freeing frag. 0x80c802c alloc'ed from sr_module.c: register_module(135) Oct 6 16:19:11 webby ser: 0(0) DEBUG: tm_shutdown : start Oct 6 16:19:11 webby ser: ser startup succeeded Oct 6 16:22:38 webby ser: WARNING: could not rev. resolve 217.57.90.117 Oct 6 16:22:38 webby ser: WARNING: could not rev. resolve 192.168.100.117 217.57.90.117 and 192.168.100.117 are my two eth interfaces any idea? bye, Andrea

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[Serusers] Multple SER servers on same domain
by Tjardick van der Kraan 07 Oct '03

07 Oct '03

Hello Everyone, Just a new face on the mailinglist (only been lurking so far) so just a friendly hi from me to all of you. I was wondering if the following idea could be realised with the use of SER and some (advanced) routing rules. One main SER server with one or more 'slave' SER servers running at their own location with a synchronized user database, but with their own presence table. e.g. if someone on the main SER server asks for a connectiong with a user on a 'sub' ser server of the same domain, the main SER server looks up if the user is online on it's server if not it checks if the user is online on one (or more) sub servers. What i like to accomplish is multple local networks that run their own ser server and support their 'local' connections, but sometimes there is a connection available to another local network, in which case more users can be reached. so on a request and fail of the user online on the local server it would try and see if it can connect to (any) other server and if so query if the user is online there. so everyone will be user(a)domain.com but it would point to the ip of the local server. Any config examples on how to achive this if possible would really be appreciated. Kind regards, Tjardick van der Kraan -- Tjardick van der Kraan tjardick(a)vanderkraan.net Tel +32 4 34 40 522 Fax +32 4 34 40 525 GSM +32 497 45 27 36 IAXtel: 1 700 344 0522 FWD: 26322 IPtel: 91331 Rue des Pocheteux 10 4020 Jupille Belgium

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[Serusers] multiple enum lookups
by Jakob Schlyter 06 Oct '03

06 Oct '03

is it possible to somehow change paramters during runtime? I need to look up users in two different enum domains - first in e164.arpa and, if not found, in e164.foo.bar (2nd enum is for internal provider routing purposes). if not, should we add a enum_query_domain(service, domain) that enables this? jakob

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Re: [Serusers] Registrar.so error
by giAndrea 06 Oct '03

06 Oct '03

At 17.32 06/10/2003 +0200, you wrote: >At 05:12 PM 10/6/2003, giAndrea wrote: > > >Hi, another problem. > > > >After compilation of new sources (0.8.12dev-17-cristian) i've found this > problem in modules registrar.so during execution (/etc/init/ser start) > > > >Oct 6 16:19:09 webby ser: 0(0) fixing > /usr/lib/ser/modules/registrar.so save > >Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x42134000, 16) called from > dlist.c: new_dlist(156) > >Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x42134000, 16) returns > address 0x422bea0c frag. 0x422be9f4 (size=16) on 1 -th hit > >Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x42134000, 8) called from > dlist.c: new_dlist(163) > >Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x42134000, 8) returns > address 0x422bea4c frag. 0x422bea34 (size=8) on 1 -th hit > >Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x42134000, 36) called from > udomain.c: new_udomain(117) > >Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x42134000, 36) returns > address 0x422bea84 frag. 0x422bea6c (size=36) on 1 -th hit > >Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x42134000, 8192) called from > udomain.c: new_udomain(124) > >Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x42134000, 8192) returns > address 0x422bead8 frag. 0x422beac0 (size=8192) on 1 -th hit > >Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x80c17c0, 9) called from > db_con.c: use_table(53) > >Oct 6 16:19:09 webby ser: 0(0) qm_malloc(0x80c17c0, 12) returns > address 0x80cb9dc frag. 0x80cb9c4 (size=12) on 1 -th hit > >Oct 6 16:19:09 webby ser: 0(0) submit_query(): Unknown column 'flags' > in 'field list > >See what it tells you :) I've seen that, but i supposed that i was an error on source and not in db. I provide to correct my db structure. thanks. >You are using development version in which database structures have not >been updated yet. I suppose you actually don't want to take the development >version. > >It takes now a new column, called "flags". It is an int(11). > >-jiri -------------------------------------------------------------------------------------- There is no world without Verona's walls but purgatory, torture, hell itself. Hence-banished is banished from the world, and world's exile is death [Shakespeare: Romeo and Juliet - Act III - Scene III]

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RE: [Serusers] voicemail question - dial plan example - pstngatewayexample
by Steve Dolloff 06 Oct '03

06 Oct '03

THANK YOU. That works. Stephen Subject: RE: [Serusers] voicemail question - dial plan example - pstngatewayexample Steve, Have you tried using "append_branch()" after "rewritehostport(<new uri>)"? Jaime From: Steve Dolloff <sdolloff(a)noc.dls.net> on 03/10/2003 15:04 To: Alexander Mayrhofer <axelm(a)nic.at>@SMTP@Exchange Jiri Kuthan <jiri(a)iptel.org>@SMTP@Exchange serusers@lists.iptel.org@SMTP@Exchange cc: Subject: RE: [Serusers] voicemail question - dial plan example - pstngatewayexample Any other suggestions? Subject: Re: [Serusers] voicemail question - dial plan example - pstngateway example On (02.10.03 23:28), Alexander Mayrhofer wrote: > replacing > > rewritehostport("219.242.10.153:5061"); > > with > > append_branch("sip:219.242.10.153:5061"); > > in the fialure_route should do the trick. err, i hate replying to my own mails. The above line will make you loose the user part of the URI, and is therefore not a working solution ... we'd need a function to reset the branch status value here, which afaik does currently not exist ... axelm _______________________________________________ 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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[Serusers] Can you help me!
by 郝育鹏 06 Oct '03

06 Oct '03

Dear Teacher: How are you! I'am in china.I install ser in my computer. But when I login with Windows message 4.6 ,Ser server don't authenticate my usename. How do I config? when I login server every time, Server can request authentication. Help me! thank you! Mr, hao How are you! Is file name of SER script ser.cfg ? my SER script be shown at last , Is it correct? Thank you very much! # # $Id: ser.cfg,v 1.21.2.1 2003/07/30 16:46:18 andrei 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=7 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" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/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") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # 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; }; # 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") { # Uncomment this if you want to use digest authentication if (!www_authorize("Lserver", "subscriber")) { www_challenge("Lserver", "0"); break; }; save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; } # Then end

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sr-users October 2003