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[Serusers] SER doesn't generate NOTIFY
by Alois Schafferhofer 19 Jun '06

19 Jun '06

Hi, I use Presence Snapshot 4.1 on Linux 9.3. SER doesn't generate a NOTIFY when status changes. In MySQL database, I can see the change: table: Presentity_Contact basic: online/offline status:"" I wonder why basic is online/offline and status is "". Is this correct? In my messages, i send basic=open/closed. In my config-file, I set the timer_interval to 5 seconds. What am I doing wrong? Regards, Alois SUBSCRIBE sip:PDA_User@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP 10.15.200.22:5060;rport;branch=z9hG4bK67417..Max-Forwards: 70..To: <sip:PDA_User@fh-joanneum.at>..From: <sip:esca@fh-joanneum.at>;tag=z9hG4bK64795000..Call-ID: 720365631452@10.15.200.22..CSeq: 1 SUBSCRIBE..Contact: <sip:esca@10.15.200.22>..Expires: 3600..User-Agent: mjsip stack 1.6..Event: presence..Accept: application/cpimpidf+xml..Content-Length: 0.... SIP/2.0 202 Accepted..Via: SIP/2.0/UDP 10.15.200.22:5060;rport=5060;branch=z9hG4bK67417..To:<sip:PDA_User@fhjoanneum.at>;tag=589230a54cd08b3768529724501fae28-4f39..From: <sip:esca@fh-joanneum.at>;tag=z9hG4bK64795000..Call-ID:720365631452@10.15.200.22..CSeq: 1 SUBSCRIBE..Expires: 3600..Contact: <sip:10.15.200.57:5060>..Server: Sip EXpress router (0.10.99-dev35-pa-4.1 (i386/linux))..Content-Length: 0..Warning: 392 10.15.200.57:5060 "Noisy feedba ck tells: pid=17547 req_src_ip=10.15.200.22 req_src_port=5060 in_uri=sip:PDA_User@fhjoanneum. at out_uri=sip:PDA_User@fh-joanneum.at via_cnt==1".... NOTIFY sip:esca@10.15.200.22 SIP/2.0..Via: SIP/2.0/UDP 10.15.200.57;branch=z9hG4bK720f.274f52a6.0..To:<sip:esca@fhjoanneum.at>;tag=z9hG4bK64795000..From: <sip:PDA_User@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28-4f39..CSeq:1 NOTIFY..Call-ID: 720365631452@10.15.200.22..Content-Length: 0.. User-Agent: Sip EXpress router(0.10.99-dev35-pa-4.1 (i386/linux))..Event: presence..Contact: <sip:10.15.200.57:5060>..Subscription-State: pending;expires=3600.... U 10.15.200.22:5060 -> 10.15.200.57:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.15.200.57;branch=z9hG4bK720f.274f52a6.0..To: <sip:esca@fh-joanneum.at>;tag=z9hG4bK64795000..From: <sip:PDA_User@fh -joanneum.at>;tag=589230a54cd08b3768529724501fae28-4f39..Call-ID: 720365631452@10.15.200.22..CSeq: 1 NOTIFY..Server: mjsip stack 1.6..Content-Length:0. PUBLISH sip:PDA_User@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK843798674..Max-Forwards: 70..From: <sip:PDA_User@fhjoanneum .at>;tag=850624414..To: <sip:PDA_User@fh-joanneum.at>..Call-ID: 1836804380-820133259- 1969969487..CSeq: 6 PUBLISH..Contact: <sip:PDA_User@fh-joanneum.a t>..Content-Type: application/pidf+xml..Content-Length: 226..Expires: 7200..Event: presence..User-Agent: SIP .NET 1.0 evaluation version, www.independ entsoft.com..SIP-If-Match: 0x409c3088x10cc41b0x4496b143....<?xml version="1.0" encoding="utf-8"?> <presence xmlns="urn:ietf:params:xml:ns:pidf" entity="pres:PDA_user@fh-joanneum.at"> <tuple id="4e9a7335-5f7f-4a3c-91a7-fd761e9138b0"> <status><basic>open</basic> </status> </tuple> </presence>.. SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.0.0.3:5060;branch=z9hG4bK843798674.. From:<sip:PDA_User@fh-joanneum.at>;tag=850624414..To: <sip:PDA_User@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28-ed6b..Call-ID: 1836804380-820133259- 1969969487..CSeq: 6 PUBLISH..Expires: 120..SIP-Tag:0x409c3088x10cc41b0x4496b143..Contact: <sip:10.0.0.11:5060>..Server: Sip EXpress router (0.10.99-dev35-pa-4.1(i386/linux))..Content-Length: 0..Warning: 392 10.0.0.11: 5060 "Noisy feedback tells: pid=17551 req_src_ip=10.0.0.3 req_src_port=5060 in_uri=sip:PDA_User@fh-joanneum.at out_uri=sip:PDA_User@fh-joanneum.at via_cnt==1" # modparam("msilo","registrar","sip:registrar@test-domain.com") modparam("msilo","use_contact",0) modparam("msilo","expire_time",7200) # -- usrloc params -- # -- 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) modparam("rls", "min_expiration", 200) modparam("rls", "max_expiration", 300) modparam("rls", "default_expiration", 300) modparam("rls", "auth", "none") #"implicit" modparam("rls", "xcap_root", "http://localhost/xcap") modparam("rls", "reduce_xcap_needs", 1) modparam("rls", "db_mode", 1) modparam("rls", "db_url", "mysql://ser:heslo@localhost:3306/ser") modparam("pa", "default_expires", 3600) modparam("pa", "use_db", 1) #modparam("pa", "pa_domain", "fh-joanneum.at") modparam("pa", "use_offline_winfo", 1) # allow storing authorization requests for offline users into database modparam("pa", "offline_winfo_timer", 600) # how often try to remove old stored authorization requests modparam("pa", "offline_winfo_expiration", 600) # how long stored authorization requests live modparam("pa", "db_url", "mysql://ser:heslo@localhost:3306/ser") modparam("pa", "auth", "xcap") # mode of PA authorization: none, implicit or xcap modparam("pa", "auth_xcap_root", "http://localhost/xcap") modparam("pa", "winfo_auth", "none") # do not authorize watcherinfo subscriptions modparam("pa", "use_callbacks", 1) # use only published information if set to 0 modparam("pa", "accept_internal_subscriptions", 1) # don't accept internal subscriptions from RLS, ... modparam("pa", "watcherinfo_notify", 1) modparam("pa", "max_subscription_expiration", 3600) # maximum value of Expires for subscriptions modparam("pa", "max_publish_expiration", 120) # maximum value of Expires for publications modparam("pa", "timer_interval", 5) modparam("presence_b2b", "presence_route", "<sip:127.0.0.1;transport=tcp;lr>") # route for generated SUBSCRIBE requests for presence modparam("presence_b2b", "on_error_retry_time", 60) # waiting time from error to new attepmt about SUBSCRIBE modparam("presence_b2b", "wait_for_term_notify", 33) # how long wait for NOTIFY with Subscription-Status=terminated after unsubscribe modparam("presence_b2b", "resubscribe_delta", 30) # how long before expiration send renewal SUBSCRIBE request modparam("presence_b2b", "min_resubscribe_time", 60) # minimal time to send renewal SUBSCRIBE request from receiving previous response modparam("presence_b2b", "default_expiration", 3600) # default expiration timeout modparam("presence_b2b", "handle_presence_subscriptions", 1) # process internal subscriptions to presence events modparam("usrloc", "db_mode", 1) modparam("domain", "db_mode", 1) modparam("domain|uri_db|acc|auth_db|usrloc|msilo", "db_url", "mysql://ser:heslo@localhost:3306/ser") modparam("domain", "domain_table", "domain") #modparam("domain", "domain_column", "domain") modparam("fifo", "fifo_file", "/tmp/ser_fifo") route{ # XML RPC if (method == "POST" || method == "GET") { create_via(); log(1, "Created via()"); dispatch_rpc(); log(1, "Dispatched rps()"); break; log(1, "break XML RPC"); } # 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 (msg:len >= 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 if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (uri==myself) { log(1, "URI == myself"); if (method=="SUBSCRIBE") { log(1, "SUBSCRIBE received"); if (!t_newtran()) { sl_reply_error(); break; }; # new subscription if (@to.tag=="") { log(1, "new SUBSCRIPTION"); if ((@msg.supported=~"eventlist")) { log(1, "msg.supported=eventlist"); # Supported header field # -> may be RLS subscription if (is_simple_rls_target("$uid-list")) { log(1, "it is simple subscription!\n"); # handle_rls_subscription("1"); # takes From UID and makes XCAP query # for user's list named "default" if (@to.tag=="") { # only for new subscriptions (with empty to tag if (!query_resource_list("default")) { t_reply("500", "XCAP query error"); break; } } } if (!have_flat_list()) { # query_resource_list failed or was not called # do standard RLS query acording to To/AOR query_rls_services(); } if (have_flat_list()) { handle_rls_subscription("1"); break; } } # SUBSCRIBE to existing user # xlog("L_ERR", "PA: handling subscription: %tu from: %fu\n"); log(1, "handle SUBSCRIBE request\n"); handle_subscription("registrar"); log(1, "handled subscription\n"); log(1, "break 1"); break; } else { # renewal subscription log(1, "else - renewal subscription"); if (!handle_rls_subscription("0")) { log(1, "!handle rls subscription"); handle_subscription("registrar"); log(1, "handled subscription 2"); } log(1, "break 2"); break; } } if (method=="REGISTER") { log(1, "REGISTER received"); # if (!www_authorize("fh-joanneum.at", "credentials")) { # www_challenge("fh-joanneum.at", "0"); # break; # }; save("location"); log(1, "saved location"); break; }; if (method=="PUBLISH") { log(1, "PUBLISH received"); if (!t_newtran()) { sl_reply_error(); break; }; log(1, "handle publish request\n"); handle_publish("registrar"); log(1, "handled publish\n"); break; }; if (method=="NOTIFY") { log(1, "NOTIFY received"); if (!t_newtran()) { log(1, "newtran error\n"); sl_reply_error(); break; }; if (!handle_notify()) { log(1, "unable to handle notification - lui"); t_reply("481", "Unable to handle notification"); } break; }; # message authorization if (method=="MESSAGE") { log(1, "MESSAGE authorization\n"); if (!authorize_message("http://localhost/xcap")) { sl_reply("403", "Forbidden"); break; } } # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { log(1, "!lookup location gö"); sl_send_reply("404", "Not Found"); break; }; }; # append_hf("P-hint: usrloc applied\r\n"); log(1, "route (1)"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { log(1, "!t_relay - sl reply error"); sl_reply_error(); }; }

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RE: [Serusers] How to send instant message programmatically
by srinivas.sriram＠wipro.com 19 Jun '06

19 Jun '06

Hi Greger I have checked with sipp , with that we can send messages. But What is my requirement is i have write the code for constructing SIP "MESSAGE" message for a given sip uri. This I have to call form one of the ser module. Total what I am trying is to send a IM message when there is a missed call for a particular SIP uri. In SER there is a code for writing into the data base of the sip uri , when there is a missed call for that Uri. At this point we have to send IM message. Thank you, Regards, Sriram Srinivas. ________________________________ From: Greger V. Teigre [mailto:greger@teigre.com] Sent: Monday, June 19, 2006 5:35 PM To: Srinivas Sriram (WT01 - IP-Multimedia Carrier & Ent Networks) Cc: serusers(a)iptel.org Subject: Re: [Serusers] How to send instant message programmatically You are better off generating the message outside SER. SER is not a UAS. Have a look at sipsak or sipp (both found at sourceforge, I think). g-) srinivas.sriram(a)wipro.com wrote: Hello, I am using SER-0.9.6. How to send instant message programmatically, how to build sip message and which function can by used(like t_uac, t_request). Can you help where and which to be changed in ser code. Thank you, Regards, Sriram Srinivas. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com ________________________________ _______________________________________________ Serusers mailing list Serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

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Re: [Users] SIP Clients Can't REGISTER
by Daniel-Constantin Mierla 19 Jun '06

19 Jun '06

Hello, On 06/19/06 14:38, Hamid Ali Asgari wrote: > Hi, > Thanks for the reply. > I commented out the entry you mentioned: > > #101(a)mydomain.com Auth-Type := Accept, Sip-Group == "suspended", > #Service-Type == "Group-Check" > > What happens now is that the radius will check for a group named SUSPENDED > and will reject because it won't match the 101 user: > > >From RADIUS debug: > users: Matched entry DEFAULT at line 152 > users: Matched entry DEFAULT at line 255 > ----------------------------------------------------------- > Users:152: DEFAULT Auth-Type = System > Fall-Through = 1 > Users:255: DEFAULT Auth-Type := Reject, Service-Type == "Group-Check" > > Any thoughts? > this is ok, it is the response that the user is not in the suspended group. The processing should go further in openser configuration script. Do you encounter other issues with calls/registrations? Cheers, Daniel > Hamid > > > > > -----Original Message----- > From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Sent: Monday, June 19, 2006 12:54 PM > To: Hamid Ali Asgari > Cc: 'Edson'; users(a)openser.org > Subject: Re: [Users] SIP Clients Can't REGISTER > > If look carefully in the configuration file of the tutorial, it checks > for users belonging to "suspended" group. As you have in 'users' file: > > 101(a)mydomain.com Auth-Type := Accept, Sip-Group == "suspended", > Service-Type == "Group-Check" > > Reply-Message = "Authorized" > > means that user 101 is in suspended group and it is not allowed to > register or initiate calls. So remove this line for your 'users' file. > > Also, when you test, just watch the network traffic on you sip server, > it will reveal the reply code so you can spot easily which statement was > hit in the config file: ngrep -d any port 5060 > > Cheers, > Daniel > > > > > On 06/18/06 12:23, Hamid Ali Asgari wrote: > >> Hi, >> >> I get the following: >> >> [root@localhost radiusclient-ng]# radiusclient -f >> /usr/local/etc/radiusclient-ng/radiusclient.conf >> 'User-Name=101(a)mydomain.com' 'Password=101';echo $? >> >> Reply-Message = 'Authenticated' >> >> 1 >> >> I guess the radiusclient is working fine since I am receiving the >> RADIUS packets on the freeradius debugs. >> >> I suspect the following entries in the debug: >> >> rad_recv: Access-Request packet from host 127.0.0.1:32901, id=197, >> length=64 >> >> User-Name = "101@mydomain" >> >> Sip-Group = "*suspended*" >> >> Service-Type = Group-Check >> >> NAS-Port = 0 >> >> NAS-IP-Address = 127.0.0.1 >> >> Why is openser setting the group to SUSPENDED???? Since there is no >> group named _suspended_ on the radius that may be the reason the >> radius is rejecting: >> >> RADIUS Users file: >> >> ### --- avps --- >> >> 101(a)mydomain.com Auth-Type := Accept, Service-Type == "SIP-Callee-AVPs" >> >> Sip-Avp += "#3#1", >> >> Sip-Avp += "#4:08:00", >> >> Sip-Avp += "#5:16:00", >> >> Sip-Avp += "#6:Mon,Wed,Thu,Fri" >> >> 102(a)mydomain.com Auth-Type := Accept, Service-Type == "SIP-Callee-AVPs" >> >> Sip-Avp += "#3#1", >> >> Sip-Avp += "#4:08:00", >> >> Sip-Avp += "#5:16:00", >> >> Sip-Avp += "#6:Mon,Wed,Thu,Free" >> >> DEFAULT Auth-Type := Accept, Service-Type == "SIP-Callee-AVPs" >> >> ### --- group checking --- >> >> ### --- user 101 --- >> >> 101(a)mydomain.com Auth-Type := Accept, Sip-Group == "voip", >> Service-Type == "Group-Check" >> >> Reply-Message = "Authorized" >> >> 101(a)mydomain.com Auth-Type := Accept, Sip-Group == "suspended", >> Service-Type == "Group-Check" >> >> Reply-Message = "Authorized" >> >> 101(a)mydomain.com Auth-Type := Accept, Sip-Group == "pstn", >> Service-Type == "Group-Check" >> >> Reply-Message = "Authorized" >> >> ### --- user 102 --- >> >> 102(a)mydomain.com Auth-Type := Accept, Sip-Group == "voip", >> Service-Type == "Group-Check" >> >> Reply-Message = "Authorized" >> >> DEFAULT Auth-Type := Reject, Service-Type == "Group-Check" >> >> ### --- user authentication --- >> >> 101(a)mydomain.com Auth-Type := Digest, User-Password == "101" >> >> Reply-Message = "Authenticated", >> >> Sip-Avp += "rpid:101", >> >> Sip-Avp += "#2:192.168.10.17", >> >> # Sip-Avp += "#2:192.168.2.11" >> >> 102(a)mydomain.com Auth-Type := Digest, User-Password == "102" >> >> Reply-Message = "Authenticated", >> >> Sip-Avp += "rpid:102", >> >> # Sip-Avp += "#2:192.168.2.12" >> >> ------------------------------------------------------------------------ >> >> *From:* Edson [mailto:4lists@gmail.com] >> *Sent:* Thursday, June 15, 2006 4:19 PM >> *To:* 'Hamid Ali Asgari'; users(a)openser.org >> *Subject:* RE: [Users] SIP Clients Can't REGISTER >> >> Take a look on the user credentials on Radius DB. Radius is rejecting >> the authentication, so there should be some mismatch between Your >> login credentials and what Radius has in its DB. >> >> Try to run this command and see what's the result (linux version): >> >> radiusclient -f <conf-dir-of-radiusclient-ng>/radiusclient.conf -p 123 >> 'User-Name=some(a)valid.user' 'Password=Guess';echo $? >> >> It should return: >> >> Reply-Message = 'Authenticated' >> >> Edson. >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > >

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Re: [Serusers] Unresolvable Destination
by Greger V. Teigre 19 Jun '06

19 Jun '06

Oh yes, you just have to lookup the uri with domain name and translate it to an actual location (i.e. using lookup("location") or lookup("aliases")) Have a look at the SER Getting Started document from http://onsip.org/ (that one was for Olivier ;-) It will guide you through setting up SER. g-) Alois Schafferhofer wrote: > Thank you. > > Therefore I have two questions. > > Does it work without DNS server? > What about a Pocket PC? It can't be member of Active Directory. > > I hope, these questions don't seem too stupid for you. > > Regards, > Alois > > >> From: "Greger V. Teigre" <greger(a)teigre.com> >> To: Alois Schafferhofer <schaffi_lui(a)hotmail.com> >> CC: serusers(a)lists.iptel.org >> Subject: Re: [Serusers] Unresolvable Destination >> Date: Sat, 17 Jun 2006 20:58:46 +0200 >> >> This means that fh-joanneum.at is not possible to DNS resolve. A sip >> uri needs to be resolvable. g-) >> >> Alois Schafferhofer wrote: >>> Hi, >>> >>> I'm using Presence Snapshot 4.1. >>> >>> If I want to REGISTER my client, I get the error: Unresolvable >>> Destination. >>> >>> I have following lines in my log message: >>> >>> /usr/sbin/ser[8941]: Unknown domain to: sip:PDA_User@fh-joanneum.at >>> from: sip:PDA_User@fh-joanneum.at >>> /usr/sbin/ser[8941]: ERROR: mk_proxy: could not resolve hostname: >>> "fhjoanneum.at" >>> /usr/sbin/ser[8941]: ERROR: uri2proxy: bad host name in URI >>> <sip:PDA_User@fhjoanneum.at> >>> /usr/sbin/ser[8941]: ERROR: t_forward_nonack: failure to add branches >>> /usr/sbin/ser[8941]: sl_reply error() >>> /usr/sbin/ser[8941]: ERROR: sl_reply_error used: I'm terribly sorry, >>> server error occurred (1/SL) >>> >>> >>> >>> U 10.0.0.3:5060 -> 10.0.0.11:5060 >>> REGISTER sip:PDA_User@fh-joanneum.at SIP/2.0..Via: SIP/2.0/UDP >>> 10.0.0.3:5060;branch=z9hG4bK479720758..Max-Forwards: 70..From: >>> <sip:PDA_User@fhjoanneu >>> m.at>;tag=1505826169..To: <sip:PDA_User@fh-joanneum.at>..Call-ID: >>> 927318080- >>> 406265317-1110482780..CSeq: 1 REGISTER..Contact: >>> <sip:PDA_User@10.0.0.3>.. >>> Content-Length: 0..User-Agent: SIP .NET 1.0 evaluation version, >>> www.independentsoft.com.... >>> # >>> U 10.0.0.11:5060 -> 10.0.0.3:5060 >>> SIP/2.0 478 Unresolvable destination (478/TM)..Via: SIP/2.0/UDP >>> 10.0.0.3:5060;branch=z9hG4bK479720758..From: <sip:PDA_User@fhjoanneum. >>> at>;tag=1505826 >>> 169..To: >>> <sip:PDA_User@fh-joanneum.at>;tag=589230a54cd08b3768529724501fae28- >>> 2449..Call-ID: 927318080-406265317-1110482780..CSeq: 1 >>> REGISTER..Server: S >>> ip EXpress router (0.10.99-dev35-pa-4.1 (i386/linux))..Content-Length: >>> 0..Warning: 392 10.0.0.11:5060 "Noisy feedback tells: pid=8941 >>> req_src_ip=10.0 >>> .0.3 req_src_port=5060 in_uri=sip:PDA_User@fh-joanneum.at >>> out_uri=sip:PDA_User@fh-joanneum.at via_cnt==1".... >>> # >>> U 10.0.0.11:5060 -> 10.0.0.3:5060 >>> SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL)..Via: >>> SIP/2.0/UDP >>> 10.0.0.3:5060;branch=z9hG4bK479720758..From: <sip:PDA_User@fh-joanneum. >>> at>;tag=1505826169..To: <sip:PDA_User@fhjoanneum. >>> at>;tag=56a704c9068c5b1b9228b761fe5fb2cc.2449..Call-ID: 927318080- >>> 406265317-1110482780..CSeq: 1 REGI >>> STER..Server: Sip EXpress router (0.10.99-dev35-pa-4.1 >>> (i386/linux))..Content- >>> Length: 0..Warning: 392 10.0.0.11:5060 "Noisy feedback tells: pid=8941 >>> req_src_ip=10.0.0.3 req_src_port=5060 >>> in_uri=sip:PDA_User@fh-joanneum.at >>> out_uri=sip:PDA_User@fh-joanneum.at via_cnt==1" >>> >>> I hope you can help me. >>> >>> Regards, >>> Alois >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> Serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >>> >>> > > >

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[Serusers] fr_inv_timer
by olivier.taylor 19 Jun '06

19 Jun '06

hi all, I have modparam("tm", "fr_inv_timer", 60) when a callee doesn't anwer, he often send a 487 (cancelled) before 60 seconds... It means timeout in fact or no answer... If the caller stop the call, I also get a 487... How to make the difference on a 487 generated by the caller and by the callee? It makes difference, if it's generated by the callee, I will forward the call to voicemail or another number, if it's generated by the caller, I will do nothing. When I set fr_inv_timer to 15, I always get a timeout and it's ok, but my customers dislikes that as well as the callees who needs to be better than Ben Jonhson for the 100meters :( Does exists a parameter to be send to the callee to tell him to wait 60 seconds + 1 before sending me a Cancel? Regards, Olivier

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[Serusers] voip peering
by Mindaugas Kuprys 19 Jun '06

19 Jun '06

Hi, Is there anyone running ser for peering with other voip providers using IPERX, Neustar and Arbinet provided services? I'm thinking to do that or should better use other solution like enum server http://www.ag-projects.com/ENUM.html? Regards, Mindaugas

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[Users] Sql possiblities of 1.0.1
by Fatih Cerit 19 Jun '06

19 Jun '06

Hi all Is it possible to do these with 1.0.1 SQL modules?? 1- is it possible to only insert the location queries instead of insert and update? 2- is it possible to lookup() from diffrent table (like a view) and save() location table? if possible which will queried from init_db. 3- is it possible to call function-select instead of a insert like select * from function_location('','','','','',''); thanks cerit

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[Users] planet VIP 152 T
by Joao Pereira 19 Jun '06

19 Jun '06

Hello to all Im testing a Planet 152 T phone and Im having some problems. Can someone tell me if this phone does URI dialing? And does it work behind NAT (does it need any special configuration on the SIP server)? Thanks Joao Pereira

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[Users] sip contact relay/forward/rewrite?
by Paul Smith 19 Jun '06

19 Jun '06

Hi, I am new to openser and trying to use it as a proxy i.e. I have a phone behind NAT, and another SIP server with all our account/registrar/routing details, I want to try mediaproxy as a far-end nat traversal solution to allow us to install phones without doing much (or anything) to the NAT firewalls.... The idea is that I set up phones to use Openser as Outbound proxy, and use symmetric RTP, and tell openser to nathelper to do regular keep alive pings. Then the phone should open SIP port to openser when registering, and OpenSER forwards the registration to the current sip server. Later invites from the current server should go back through Openser to the phone over the still open ip:port pair. MediaProxy on the openser should handle the RTP traffic between phone and old sip server. I have got ser and mediaproxy running and outgoing calls are fine. The problem is that incoming calls are not going through the proxy... the phone registers via the openser proxy, but when initiating an incoming call to the phone the registrar server sends the SIP invite straight to the NAT firewall, which rejects it... What do I need to do so that openSER tells the registrar to route SIP INVITES back through the openser server? I realise that at present I am looking at a very insecure open relay - I'll add more auth and domain checking when I get the basic outbound relay / media proxy bit working. PHONE ---- NAT ---- SER ----- SIP registrar Server with user details etc. Thanks.

2 1

[Users] Can't forward to PSTN gateway
by Kenny Chua 19 Jun '06

19 Jun '06

Hi all, I'm having trouble allowing calls to my pstn gateway. I get some errors. These are the errors I get: I get the 404 Not Found error Here is the information I got from ngrep: U 2006/06/17 02:58:31.372496 68.xxx.xxx.17:5060 -> 192.168.1.20:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;branch=z9hG4bK082837F1736B4E11832482B6D09012F5..From: K enny <sip:201@mydomain.myftp.org>;tag=2300590181..To: <sip:13373038435@mydomain.myftp.org>..Call-ID: 4B570DFD-0E2E-4F32-8B4B-05ADC419D0E7(a)192.168.1.20.. CSeq: 60451 INVITE..Server: OpenSer (1.0.1 (i386/linux))..Content-Length: 0..Warning: 392 192.168.1.5:5060 "Noisy feedback tells: pid=2874 req_src_ip=1 92.168.1.1 req_src_port=5060 in_uri=sip:13373038435@mydomain.myftp.org out_uri=sip:13373038435@pstn.gateway.net:5060 via_cnt==1".... U 2006/06/17 02:58:31.487474 192.168.1.5:5060 -> 65.xxx.xxx.xxx:5060 INVITE sip:13373038435@pstn.gateway.net:5060 SIP/2.0..Record-Route: <sip:192.168.1.5;ftag=2300590181;lr=on>..Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK 7217.285e6331.0..Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;branch=z9hG4bK082837F1736B4E11832482B6D09012F5..From: Kenny <sip:201@mydomain.myftp.org>;t ag=2300590181..To: <sip:13373038435@mydomain.myftp.org>..Contact: <sip:201@192.168.1.1:5060>..Call-ID: 4B570DFD-0E2E-4F32-8B4B-05ADC419D0E7(a)192.168.1.20 ..CSeq: 60451 INVITE..Max-Forwards: 69..Content-Type: application/sdp..User-Agent: X-Lite release 1105x..Content-Length: 304....v=0..o=201 10653130 1065 3152 IN IP4 68.226.254.17..s=X-Lite..c=IN IP4 192.168.1.1..t=0 0..m=audio 8000 RTP/AVP 0 8 3 98 97 101..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=rt pmap:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=sendrecv.. U 2006/06/17 02:58:31.545023 65.200.189.132:5060 -> 192.168.1.5:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 68.226.254.17:5060;branch=z9hG4bK7217.285e6331.0..Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;branch=z9hG4bK082837 F1736B4E11832482B6D09012F5..From: Kenny <sip:201@mydomain.myftp.org>;tag=2300590181..To: <sip:13373038435@mydomain.myftp.org>..Call-ID: 4B570DFD-0E2E-4F 32-8B4B-05ADC419D0E7@192.168.1.20..CSeq: 60451 INVITE..Content-Length: 0.... U 2006/06/17 02:58:31.661353 65.200.189.132:5060 -> 192.168.1.5:5060 SIP/2.0 500 Server Internal Error..To: <sip:13373038435@mydomain.myftp.org>;tag=3359519831-260907..From: Kenny <sip:201@mydomain.myftp.org>;tag=23005901 81..Call-ID: 4B570DFD-0E2E-4F32-8B4B-05ADC419D0E7@192.168.1.20..CSeq: 60451 INVITE..Contact: sip:13373038435@65.xxx.xxx.xxx:5060..Via: SIP/2.0/UDP 68.xx x.xxx.17:5060;branch=z9hG4bK7217.285e6331.0..Via: SIP/2.0/UDP 192.168.1.1:5060;rport=5060;branch=z9hG4bK082837F1736B4E11832482B6D09012F5..Content-Length : 0.... U 2006/06/17 02:58:31.661842 192.168.1.5:5060 -> xx.xxx.xxx.132:5060 ACK sip:13373038435@pstn.gateway.net:5060 SIP/2.0..Via: SIP/2.0/UDP 192.168.1.5;branch=z9hG4bK7217.285e6331.0..From: Kenny <sip:201@mydomain.myftp.org>; tag=2300590181..Call-ID: 4B570DFD-0E2E-4F32-8B4B-05ADC419D0E7@192.168.1.20..To: <sip:13373038435@mydomain.myftp.org>;tag=3359519831-260907..CSeq: 60451 ACK..User-Agent: OpenSer (1.0.1 (i386/linux))..Content-Length: 0.... Below is my openser.cfg # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu 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 fork=no log_stderror=yes */ # put your server IP address here listen=192.168.1.5 alias=mydomain.myftp.org check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("usrloc", "db_mode", 2) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); exit; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); exit; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { route(1); }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (method=="REGISTER") { xlog("Registering\n"); route(2); return; }; lookup("aliases"); if(method =="INVITE"){ if( uri=~"^sip:1[0-9]*@"){ xlog("International call\n"); route(4); exit; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- if (!t_relay()) { sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!www_authorize("mydomain.myftp.org","subscriber")) { www_challenge("mydomain.myftp.org","0"); exit; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; xlog("Registered Successfully\n"); } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; consume_credentials(); lookup("aliases"); if (uri!=myself) { route(1); }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); exit; }; route(1); } route[4]{ xlog("Before forward to host\n"); rewritehostport("pstn.gateway.net:5060"); xlog("After forward to host\n"); route(1); } What am I doin wrong? Why am I not able to dial a PSTN line? My PSTN gateway allows international calls. Any help is appreciated. Thank you. --------------------------------- Ring'em or ping'em. Make PC-to-phone calls as low as 1¢/min with Yahoo! Messenger with Voice.

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