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[Serusers] RE: [Sipforum-discussion] regarding Authentication...
by Paul van Schagen 21 Apr '05

21 Apr '05

-----Oorspronkelijk bericht----- Van: Paul van Schagen [mailto:paul.van.schagen@adaline.net] Verzonden: donderdag 21 april 2005 9:02 Aan: 'Josekutty K.K' Onderwerp: RE: [Sipforum-discussion] regarding Authentication... Methods you really want to authenticate are INVITE ( outgoing calls ) and REGISTER ( incoming calls ). You try the request first and the switch answers requesting authentication. Then the authentication is sent by the UA to the switch. Remember the authentication also verifies the IP address of the sender is correct. By doing a double request you make sure that the original request came from where they say it came from. Lets say somebody fakes an IP address and sents the packet to the switch. The switch then sends back an authentication requests which arrives at another destination. The call never gets authenticated correctly and is rejected. Hopes this helps Can anybody help me with my mediaproxy timeout ? Best regards Paul van Schagen -----Oorspronkelijk bericht----- Van: sipforum-discussion-bounces(a)lists.su.se [mailto:sipforum-discussion-bounces@lists.su.se] Namens Josekutty K.K Verzonden: donderdag 21 april 2005 8:47 Aan: sipforum-discussion(a)lists.su.se Onderwerp: [Sipforum-discussion] regarding Authentication... Hi all, I am a beginner in SIP. Let me explain my doubt. Once a SIP request authenticates itself, (usually a REGISTER request), all further requests MUST include the authorization details .. I mean I MUST send the authorization details to subsequent requests say OPTIONS, INVITE etc . Or Should I try each request without authentication first? Regards Josekutty K K ------ To unsubscribe from the Sipforum-discussion mailing list visit https://mbox.su.se/mailman/listinfo/sipforum-discussion

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[Serusers] Commercial SER Support
by Darren Nay 21 Apr '05

21 Apr '05

Hey All, Do any of you know of a reliable / knowledgeable source for commercial technical support for SER. If so, we may be interested. We have worked in the past directly with IPTel, and have discussed support options recently with Jiri. He suggested that we ask the fine folks of the SER Users mailing list for their suggestions. Thanks so much, Darren Nay Ionosphere, Inc dnay(a)ionosphere.net (864) 678-3158

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[Serusers] how to get call duration in radius accounting stop
by Kamran Ahmad 21 Apr '05

21 Apr '05

hi how to get call duration in radius. there should be callduration in accounting stop but here in my case there is no callduration Acct-Status-Type = Stop Service-Type = Sip-Session Sip-Response-Code = 200 Sip-Method = 8 User-Name = "4000@ser" Calling-Station-Id = "sip:4000@ser" Called-Station-Id = "sip:3000@ser" Sip-Translated-Request-URI = "sip:3000@ser" Acct-Session-Id = "CB13EC9C-9DB4-420F-BEF4-786E6398D714@UA" Sip-To-Tag = "as23df87de" Sip-From-Tag = "290015013442" Sip-Cseq = "2" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Acct-Delay-Time = 0 Client-IP-Address = 127.0.0.1 Acct-Unique-Session-Id = "ca12eca5fce40009" Timestamp = 1112912023 __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Re: [Serusers] Ser 0.9.0 and Cisco Voice Gateway
by Ahmad Cheikh-Moussa 21 Apr '05

21 Apr '05

Hi Steve! Now it works properly. Like you wrote the port command was missed in the dial peer. dial-peer voice 1 pots description Default-Dial-peer fuer ausgehende Anrufe preference 3 service session max-conn 25 destination-pattern 0T progress_ind alert enable 8 direct-inward-dial port 0/0/1:15 Now I have activated the www_authorize section. # Uncomment this if you want to use digest authentication if (!www_authorize("netuse.de", "subscriber")) { www_challenge("netuse.de", "1"); break; }; On my cisco I see following when I enable the ccsip debugging: Apr 21 09:24:52.933 MEST: //-1/xxxxxxxxxxxx/SIP/Info/HandleUdpSocketReads: Msg enqueued for SPI with IP addr: 192.168.254.30:5060 Apr 21 09:24:52.933 MEST: //-1/xxxxxxxxxxxx/SIP/Transport/sipTransportProcessNWNewConnMsg: context=0x00000000 Apr 21 09:24:52.933 MEST: //-1/xxxxxxxxxxxx/SIP/Msg/ccsipDisplayMsg: Received: SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.254.203:5060;branch=z9hG4bK2924EA From: "0*" <sip:0*@192.168.254.30>;tag=E7AF75C-13AE To: "0*" <sip:0*@192.168.254.30>;tag=b27e1a1d33761e85846fc98f5f3a7e58.2283 Call-ID: E1A23104-B0DA11D9-81B582BA-75F29624 CSeq: 42 REGISTER WWW-Authenticate: Digest realm="netuse.de", nonce="426756705993d4a9785e2e828158f76faa5234d4", qop="auth" Server: Sip EXpress router (0.9.0 (sparc64/solaris)) Content-Length: 0 Warning: 392 192.168.254.30:5060 "Noisy feedback tells: pid=16087 req_src_ip=192.168.254.203 req_src_port=52773 in_uri=sip:192.168.254.30:5060 out_uri=sip:192.168.254.30:5060 via_cnt==1" Apr 21 09:24:52.933 MEST: //-1/000000000000/SIP/Error/sipSPIHandleAuthChallenge: Error getting credentials Apr 21 09:24:52.933 MEST: //-1/000000000000/SIP/Info/sipSPIUdeleteCcbFromUACTable: ****Deleting from UAC table. Apr 21 09:24:52.933 MEST: //-1/000000000000/SIP/Info/sipSPIUdeleteCcbFromTable: Deleting from table. ccb=0x45D7009C key=E1A23104-B0DA11D9-81B582BA-75F29624 Apr 21 09:24:52.933 MEST: //-1/000000000000/SIP/Info/sipSPIFlushEventBufferQueue: There are 0 events on the internal queue that are going to be free'd Apr 21 09:24:52.933 MEST: //-1/000000000000/SIP/Info/sipSPIUfreeOneCCB: Freeing ccb 45D7009C It looks like the cisco tries to registering to the server, but the server decline this. How can I enable the registering of my gateway in ser, without to disable the www_authorize ? Regards, Ahmad On Apr 20, 05, Steve Blair wrote: > > > Ahmad Cheikh Moussa wrote: > > >Hi! > > > >> > >>>Can I ask you some questions? > >>>How do I know, that my ser server only act as registrar and > >>>redirect Server ? With redirect I mean, any other calls, which > >>>starts with 0, redirect to the gateway. > >> > >> > >> > >>I'm not sure I understand. Are you concerned about people outside > >>of your user community routing calls through your proxy and out to the > >>PSTN? If so then you need to use check_from and check_to to > >>determine if the caller is authenticated to your proxy. > > > >I have following line in ser.cfg > >t_relay_to_udp("192.168.254.203", "5060"); > >when I understand you right, then I have to change this paramater to > >sl_relay_to_udp("192.168.254.203", "5060"); > > > No, the t_relay is fine. > > >This would mean stateless forwarding to this gateway ? > >correct ? > > > >Regars, > > Ahmad > > > > -- > > ISC Network Engineering > The University of Pennsylvania > 3401 Walnut Street, Suite 221A > Philadelphia, PA 19104 > > > voice: 215-573-8396 > > 215-746-8001 > > fax: 215-898-9348 > > sip:blairs@upenn.edu > -- Ahmad Cheikh-Moussa NetUSE AG Dr.-Hell-Straße, 24107 Kiel, Germany Telefon: +49 431 2390 400 -- Telefax: +49 431 2390 499 Service: Service(a)NetUSE.DE -- http://NetUSE.DE/

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[Serusers] HELP Needed! SER Crashes.
by Aleksandr N/A 21 Apr '05

21 Apr '05

Hi! I have some problems with SER 0.8.14. When I'm trying AUTH modules, it crashes. (it says: --------------------------------------------------------------------------------------------------- Starting SER : cat: /var/run/ser.pid: No such file or directory started pid() --------------------------------------------------------------------------------------------------- My config is the following: # # $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/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" # 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") # -- 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("ignosya.dyndns.org", "subscriber")) { www_challenge("ignosya.dyndns.org", "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(); }; }

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[Serusers] (no subject)
by Clay Bryan 21 Apr '05

21 Apr '05

Hi, I loaded serweb on my gentoo box and when I open the my_account.php I get a error highlighted in red sorry - fifo reading error. Has anyone else encountered this problem and if so how can I fix it. Wercs Communications Clay Bryan Network Administrator Wercs Communications 400 East First Casper, WY 82601 <http://maps.yahoo.com/py/maps.py?Pyt=Tmap&addr=400+East+First&csz=Caspe r%2C+WY++82601&country=us> CBryan(a)wercs.com <mailto:CBryan@wercs.com> tel: fax: mobile: 307-233-8359 307-233-8701 307-258-7371 Add me to your address book... <https://www.plaxo.com/add_me?u=21475170292&v0=646557&k0=282754785> Want a signature like this? <http://www.plaxo.com/signature>

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[Serusers] SER & Mediaproxy. Sipura vs x-lite (again)
by Cameron Beattie 21 Apr '05

21 Apr '05

An earlier post discussed issues I had calling between Sipura and x-lite. After receiving a lot of help that issue was resolved ... sort of. For those interested, calls between two x-lite UAs through an incorrectly configured mediaproxy had two-way audio but calls between x-lite and sipura did not. I now get one-way audio between the Sipura and x-lite when the call goes over the internet and back into the same NAT, but get no audio when the calls goes out and back into another NAT. I am using: ser 0.9.1 mediaproxy 1.3.0 ser.cfg from onsip.org I have posted the relevant Invites from 4 scenarios below and would appreciate any insight: 1. Sipura to x-Lite on same NAT (one-way audio) 2. Sipura to x-Lite on different NAT (no audio) 3. x-Lite to x-Lite on same NAT (audio OK) 4. x-Lite to x-Lite on different NAT (audio OK) The problem may be that the c record is the local IP address for the Sipura with the Sipura calls whereas with the x-lite calls it is the public IP address. I've checked the settings on the sipura and the NAT settings = no (I think this means leave NAT to mediaproxy) whilst the Via settings = yes. *Scenario 1* The following invite is on a call from Sipura (beattiek) to x-Lite (beattiec) on the same NAT. Audio is heard in one direction (from x-lite to Sipura). Sipura -> Nat 1 -> Proxy -> Nat 1 -> x-lite U 147.202.44.XXX:5060 -> 60.234.199.XXX:1028 INVITE sip:beattiea@60.234.199.XXX:5060 SIP/2.0..Record-Route: <sip:alias1@147.202.44.XXX:5060;nat=yes;ftag=1df09ec95856c84do0;lr=on> ..Via: SIP/2.0/UDP 147.202.44.XXX;branch=z9hG4bK6799.73035531.0..Via: SIP/2.0/UDP 192.168.0.11:5060;received=60.234.199.XXX;branch=z9hG4b K-d4727cc1;rport=5060..From: Cameron's sipura 2000 <sip:beattiek@beta.mydomain.co.nz>;tag=1df09ec95856c84do0..To: <sip:alias1@beta. mydomain.co.nz>..Call-ID: 91508d79-f1c0295d@192.168.0.11..CSeq: 102 INVITE..Max-Forwards: 16..Contact: Cameron's sipura 2000 <sip:beatt iek@60.234.199.XXX:5060>..Expires: 240..User-Agent: Sipura/SPA2000-2.0.13(g)..Content-Length: 424..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: x-sipura..Content-Type: application/sdp....v=0..o=- 8343043 8343043 IN IP4 192.168.0.11..s=-..c=IN IP 4 192.168.0.11..t=0 0..m=audio 16454 RTP/AVP 0 2 4 8 18 96 97 98 100 101..a=rtpmap:0 PCMU/8000..a=rtpmap:2 G726-32/8000..a=rtpmap:4 G723/ 8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729a/8000..a=rtpmap:96 G726-40/8000..a=rtpmap:97 G726-24/8000..a=rtpmap:98 G726-16/8000..a=rtpma p:100 NSE/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. *Scenario 2* The following invite is on a call from Sipura (beattiek) to x-Lite (pearcej) on a separate NAT. No audio is heard in either direction. Sipura -> Nat 1 -> Proxy -> Nat 2 -> x-lite U 147.202.44.XXX:5060 -> 219.88.100.XX:5060 INVITE sip:pearcej@219.88.100.XX:5060 SIP/2.0..Record-Route: <sip:alias2@147.202.44.XXX:5060;nat=yes;ftag=3c35fa19eb73610fo0;lr=on>..Via : SIP/2.0/UDP 147.202.44.XXX;branch=z9hG4bK810a.ea51b996.0..Via: SIP/2.0/UDP 192.168.0.11:5060;received=60.234.199.XXX;branch=z9hG4bK-ca5 e9437;rport=5060..From: Cameron's sipura 2000 <sip:beattiek@beta.mydomain.co.nz>;tag=3c35fa19eb73610fo0..To: <sip:alias2@beta.mydomain .co.nz>..Call-ID: 92d1ee61-22aab967@192.168.0.11..CSeq: 102 INVITE..Max-Forwards: 16..Contact: Cameron's sipura 2000 <sip:beattiek@60.2 34.199.XXX:5060>..Expires: 240..User-Agent: Sipura/SPA2000-2.0.13(g)..Content-Length: 424..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: x-sipura..Content-Type: application/sdp....v=0..o=- 7997909 7997909 IN IP4 192.168.0.11..s=-..c=IN IP4 192.16 8.0.11..t=0 0..m=audio 16442 RTP/AVP 0 2 4 8 18 96 97 98 100 101..a=rtpmap:0 PCMU/8000..a=rtpmap:2 G726-32/8000..a=rtpmap:4 G723/8000..a= rtpmap:8 PCMA/8000..a=rtpmap:18 G729a/8000..a=rtpmap:96 G726-40/8000..a=rtpmap:97 G726-24/8000..a=rtpmap:98 G726-16/8000..a=rtpmap:100 NS E/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. *Scenario 3* The following invite is on a call from x-Lite (beattiec) to x-Lite (beattiea) on the same NAT. x-lite-> Nat 1 -> Proxy -> Nat 1 -> x-lite U 147.202.44.XXX:5060 -> 60.234.199.XXX:1028 INVITE sip:beattiea@60.234.199.XXX:5060 SIP/2.0..Record-Route: <sip:beattiea@147.202.44.XXX:5060;nat=yes;ftag=251677925;lr=on>..Via: SIP/ 2.0/UDP 147.202.44.XXX;branch=z9hG4bKd72b.189eef55.0..Via: SIP/2.0/UDP 60.234.199.XXX:5060;rport=1027;branch=z9hG4bK52D2A416DB7849C4BC88D 0A3B2B4B1D1..From: Cameron's laptop <sip:beattiec@beta.mydomain.co.nz>;tag=251677925..To: <sip:beattiea@beta.mydomain.co.nz>..Contact : <sip:beattiec@60.234.199.XXX:1027>..Call-ID: 0E31A4AE-E92F-46D1-8ACB-7CB7E2FF1641@192.168.0.15..CSeq: 18590 INVITE..Max-Forwards: 16..C ontent-Type: application/sdp..User-Agent: X-Lite release 1103m..Content-Length: 303....v=0..o=beattiec 169182050 169182181 IN IP4 60.234. 199.XXX..s=X-Lite..c=IN IP4 60.234.199.XXX..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.. *Scenario 4* The following invite is on a call from x-Lite (beattiec) to x-Lite (pearcej) on a separate NAT. x-lite -> Nat 1 -> Proxy -> Nat 2 -> x-lite U 147.202.44.XXX:5060 -> 219.88.100.XX:5060 INVITE sip:pearcej@219.88.100.XX:5060 SIP/2.0..Record-Route: <sip:alias2@147.202.44.XXX:5060;nat=yes;ftag=3274571863;lr=on>..Via: SIP/2. 0/UDP 147.202.44.XXX;branch=z9hG4bKa447.eadaf8e4.0..Via: SIP/2.0/UDP 60.234.199.XXX:5060;rport=1027;branch=z9hG4bKED04E2DC4F0846A5B616A13 47747365E..From: Cameron's laptop <sip:beattiec@beta.mydomain.co.nz>;tag=3274571863..To: <sip:alias2@beta.mydomain.co.nz>..Contact: <sip:beattiec@60.234.199.XXX:1027>..Call-ID: 60F12C88-B4A4-44D9-A207-FF7F6A97C3C8@192.168.0.15..CSeq: 7003 INVITE..Max-Forwards: 16..Cont ent-Type: application/sdp..User-Agent: X-Lite release 1103m..Content-Length: 303....v=0..o=beattiec 165963913 165964043 IN IP4 60.234.199 .XXX..s=X-Lite..c=IN IP4 60.234.199.XXX..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=rtpma p:3 gsm/8000..a=rtpmap:98 iLBC/8000..a=rtpmap:97 speex/8000..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15.. Many thanks for any help received. Regards Cameron Beattie

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[Serusers] Window Messenger and SER
by Hiep Trung Tran 21 Apr '05

21 Apr '05

Hi, After install SER on Solaris successfully then I use Window Messenger on 2 workstation to connect to the SER but neither sees the other is online. Could you please let me know how to fix this problem? Do I need to install mysql before using Window Messenger? Thanks a lot, Hiep

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[Serusers] need help please, v0.9.0 problems
by Noel Paul Evangelista 21 Apr '05

21 Apr '05

hi guys, i need help please. very new on SER. i just installed v0.9.0(ser-0.9.0_src.tar) using "make prefix=/usr/local all" then "make prefix=/usr/local install". looks like everything was installed. ser.cfg is /usr/local/etc/ser and executables are in /usr/local/sbin. the modules are in their correct directory as shown in the ser.cfg. my problem is when i do a "serctl start" i get this problem: [root@SIPserver sbin]# serctl start Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed secondly, if i do a "serctl moni" i get this message: [root@SIPserver sbin]# serctl moni Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config i am sure that the default ser.cfg has the line fifo=/tmp/ser_fifo. thirdly, if i do a "ser start" i get this message but it looks like ser is not still running: [root@SIPserver sbin]# ser start Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 146.82.219.99 [146.82.219.99]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 146.82.219.99 [146.82.219.99]:5060 Aliases: tcp: SIPserver.islandtel.com.ph:5060 tcp: localhost:5060 tcp: localhost.localdomain:5060 tcp: SIPserver:5060 udp: SIPserver.islandtel.com.ph:5060 udp: localhost:5060 udp: localhost.localdomain:5060 udp: SIPserver:5060 [root@SIPserver sbin]# other info about the server are the following: [root@SIPserver sbin]# ser -V version: ser 0.9.0 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $ main.c compiled on 14:40:36 Apr 19 2005 with gcc 3.2 [root@SIPserver sbin]# uname -a Linux SIPserver 2.4.20-8 #1 Thu Mar 13 17:54:28 EST 2003 i686 i686 i386 GNU/Linux [root@SIPserver sbin]# any help regarding my problem will be highly appreciated. best regards, Noel __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] Permissions module, database configuration.
by Rod Bacon 21 Apr '05

21 Apr '05

In the latest CVS, and in everything else I have read, I can see no mention of the permissions module using SQL as it's data source. I have added the options to my ser.cfg (as per other people's configuration), but SER insists on attempting to read the permissions.allow and permissions.deny files on startup. Is this normal? -- ========================================== Rod Bacon - VOIP Systems Engineer Empowered Communications Ground Floor, 102 York St. South Melbourne Victoria, Australia. 3205 Phone: +613 99401600 Fax: +613 99401650 ==========================================

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RE: [Serusers] Serious Problems with SER - Not responding anythin g.
by Ricardo Martinez 20 Apr '05

20 Apr '05

Thanks Jiri. I remember have been checked the typical services when the problem was happening. My Radius (i'm not use mysql) for example was working without problems... How can i log information of SER failures? Regards, Ricardo.- > -----Mensaje original----- > De: Jiri Kuthan [mailto:jiri@iptel.org] > Enviado el: Miércoles, 20 de Abril de 2005 13:12 > Para: Ricardo Martinez; 'serusers(a)lists.iptel.org' > Asunto: Re: [Serusers] Serious Problems with SER - Not responding > anything. > > > a typical reason is SER is blocked waiting for some external service. > It may be possibly unavailable mysql, DNS, or RADIUS... > > -jiri > > At 07:09 PM 4/20/2005, Ricardo Martinez wrote: > >Hello list. > > I'm experiencing problems with my SER. As i > mentioned in a past > >post > (http://lists.iptel.org/pipermail/serusers/2005-April/018324.html), > >from time to time (it is getting pretty often) my SER not > answering any > >request, when a REGISTER reach the server it is not > challenged, this happens > >too with the INVITE's messages. About two hours ago this > happened again. > >I asked in my last post how to LOG the internal errors or > messages from SER > >to see if i can get some idea about what's going on, but i > have no answer > >about it. > >This is the infornmation i was able to capture when this was > happening : > > > >the serctl moni command shows a lot of "failures". > > > >[cycle #: 11; if constant make sure server lives and fifo is on] > >Server: Sip EXpress router (0.8.14-3 (i386/linux)) > >Now: Wed Apr 20 12:56:29 2005 > >Up Since: Thu Apr 14 21:32:18 2005 > >Up time: 487451 [sec] > > > >Transaction Statistics > >Current: 6 (2612 waiting) Total: 38194 (0 local) > >Replied localy: 80752 > >Completion status 6xx: 4643, 5xx: 768, 4xx: 10664, 3xx: > 0,2xx: 22552 > > > >Stateless Server Statistics > >200: 5975004 202: 0 2xx: 0 > >300: 0 301: 0 302: 0 3xx: 0 > >400: 0 401: 205980 403: 0 404: 0 407: 22918 408: 0 483: 0 > 4xx: 19 > >500: 0 5xx: 0 > >6xx: 0 > >xxx: 0 > >failures: 24180 > > > >UsrLoc Stats > >Domain Registered Expired > >'location' 219 4552 > >-------------------------------------------------------------- > >[cycle #: 5; if constant make sure server lives and fifo is on] > >Server: Sip EXpress router (0.8.14-3 (i386/linux)) > >Now: Wed Apr 20 12:56:47 2005 > >Up Since: Thu Apr 14 21:32:18 2005 > >Up time: 487469 [sec] > > > >Transaction Statistics > >Current: 4 (2619 waiting) Total: 38219 (0 local) > >Replied localy: 80762 > >Completion status 6xx: 4643, 5xx: 770, 4xx: 10667, 3xx: > 0,2xx: 22552 > > > >Stateless Server Statistics > >200: 5975030 202: 0 2xx: 0 > >300: 0 301: 0 302: 0 3xx: 0 > >400: 0 401: 205995 403: 0 404: 0 407: 22922 408: 0 483: 0 > 4xx: 19 > >500: 0 5xx: 0 > >6xx: 0 > >xxx: 0 > >failures: 24380 > > > >UsrLoc Stats > >Domain Registered Expired > >'location' 220 4552 > >----------------------------------------------------------------- > >[cycle #: 2; if constant make sure server lives and fifo is on] > >Server: Sip EXpress router (0.8.14-3 (i386/linux)) > >Now: Wed Apr 20 12:56:52 2005 > >Up Since: Thu Apr 14 21:32:18 2005 > >Up time: 487474 [sec] > > > >Transaction Statistics > >Current: 4 (2620 waiting) Total: 38224 (0 local) > >Replied localy: 80762 > >Completion status 6xx: 4643, 5xx: 770, 4xx: 10667, 3xx: > 0,2xx: 22552 > > > >Stateless Server Statistics > >200: 5975034 202: 0 2xx: 0 > >300: 0 301: 0 302: 0 3xx: 0 > >400: 0 401: 206000 403: 0 404: 0 407: 22923 408: 0 483: 0 > 4xx: 19 > >500: 0 5xx: 0 > >6xx: 0 > >xxx: 0 > >failures: 24445 > > > >UsrLoc Stats > >Domain Registered Expired > >'location' 215 4557 > > > >What is happening to cause the "failures" counter grows that much?. > > > >The free command shows this : > >[root@root]# free > > total used free shared > buffers cached > >Mem: 1030888 626464 404424 0 > 86692 369848 > >-/+ buffers/cache: 169924 860964 > >Swap: 2040244 4864 2035380 > > > >The top command shows not much use of CPU. The proccess > were running (the > >SER and mediaproxy) > >Despite of all this, the SER was not answering anything. > >Could this be a bug? or maybe a problem with my ser.cfg file?. I'm > >attaching my ser.cfg file too to see if someone can tell me > if there is > >something wrong about it. > >I really hope that someone could help me, this is getting > really complicated > >every time it happens. > > > >Thanks > >Ricardo Martinez.- > > > >************************************************** > >SER.CFG > >************************************************** > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++++++++++++++++++++++++ > > > > > > > > > ># ----------- global configuration parameters > ------------------------ > > > >debug=0 # debug level (cmd line: -dddddddddd) > >fork=yes > >log_stderror=yes # (cmd line: -E) > > > >#Uncomment these lines to enter debugging mode > >#fork=no > >#log_stderror=yes > > > >listen=ip.of.my.ser > > > >alias=sip.mydomain.com > >alias=sip2.mydomain.net > >alias=ip.of.my.ser > > > > > >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" > >fifo_mode=0666 > > > > > ># ------------------ 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/textops.so" > >loadmodule "/usr/local//lib/ser/modules/exec.so" > > > ># ++++++++++ Para trabajar con NAT > > > >loadmodule "/usr/local//lib/ser/modules/mediaproxy.so" > >loadmodule "/usr/local//lib/ser/modules/domain.so" > >loadmodule "/usr/local//lib/ser/modules/dbtext.so" > >loadmodule "/usr/local//lib/ser/modules/nathelper.so" > > > ># Uncomment this if you want digest authentication > ># mysql.so must be loaded ! > >loadmodule "/usr/local/lib/ser/modules/auth.so" > > > ># ++++++++++ Modulos de Accounting y Autorizacion via Radius > > > >loadmodule "/usr/local/lib/ser/modules/auth_radius.so" > >loadmodule "/usr/local/lib/ser/modules/group_radius.so" > >loadmodule "/usr/local/lib/ser/modules/acc.so" > > > ># ----------------- setting module-specific parameters > --------------- > ># -- rr params -- > ># add value to ;lr param to make some broken UAs happy > >modparam("rr", "enable_full_lr", 1) > > > ># +++++ module authorization > >modparam("auth_radius","radius_config","/usr/local/etc/radius > client/radiuscl > >ient.conf") > >modparam("auth_radius","service_type",15) > > > > > ># +++++ module accounting > >modparam("acc","radius_config","/usr/local/etc/radiusclient/r > adiusclient.con > >f") > >modparam("acc","log_level",1) > >modparam("acc", "service_type", 15) > >modparam("acc", "radius_flag", 1) > >modparam("acc", "radius_missed_flag", 3) > >modparam("acc", "failed_transactions", 1) > >modparam("acc", "report_ack", 0) > > > > > ># +++++ group radius > >modparam("group_radius","radius_config","/usr/local/etc/radiu > sclient/radiusc > >lient.conf") > >modparam("group_radius", "use_domain", 1) > > > > > ># +++++ module mediaproxy > >modparam("mediaproxy", "natping_interval", 60) > >modparam("mediaproxy", "mediaproxy_socket", > "/var/run/mediaproxy.sock") > >#modparam("mediaproxy", "sip_asymmetrics", > >"/usr/local/etc/ser/sip-asymmetrics-clients") > >#modparam("mediaproxy", "rtp_asymmetrics", > >"/usr/local/etc/ser/rtp-asymmetrics-clients") > >modparam("registrar", "nat_flag", 5) > > > ># +++++ module domain > >modparam("domain", "db_url", "/usr/local/etc/ser/domaintables") > >modparam("domain", "domain_table", "domain") > >modparam("domain", "domain_col", "domain") > > > ># +++++ module tm > >modparam("tm", "fr_timer", 15) > >modparam("tm", "fr_inv_timer", 22) > >modparam("tm", "wt_timer", 5) > > > ># +++++ module registrar (Con estos parametro haces que el > location lookup > >priorize solo los mas recientes contacts) > >modparam("registrar", "append_branches", 0) > >modparam("registrar", "desc_time_order", 1) > > > > > ># ------------------------- request routing logic > ------------------- > > > ># main routing logic > > > >route { > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > ++++++++++++++ > > # Llamada a Rutina Externa "translate" para cambio de numero > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > ++++++++++++++ > > > > if (method=="INVITE" || method=="CANCEL") { > > exec_dset("/usr/local/etc/ser/translate"); > > }; > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > ++++++++++++++ > > # 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; > > }; > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > > # Marcamos todos los mensajes para que pasen por > nuestro servidor > > # a traves del record route. > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > > > > record_route(); > > > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > > # Permitimos el uso del campo Route en el ruteo de la llamada > > # > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > > > > if (method=="BYE" || method=="CANCEL") { > > log(1, "NAT: BYE o CANCEL recibido --> > terminando la sesion > >de media\n"); > > end_media_session(); > > setflag(1); > > }; > > > > if (loose_route()) { > > t_relay(); > > break; > > }; > > > > > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+ > > # Manejo de los mensajes NOTIFY para Keep Alive de NAT > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+ > > > > if ((method=="NOTIFY") && search("^Event: keep-alive")) { > > sl_send_reply("200", "OK"); > > break; > > }; > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+ > > # Marcamos las llamadas para accounting > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+ > > > > setflag(1); > > > > > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+ > > # Test para clientes NATeados > > # > > # 1 : Chequea si el cliente tiene IP privada en el > campo Contact. > > # 2 : Chequea si el cliente se contacto con una IP > distinta a la del > >campo VIA. > > # 3 : Chequea si el cliente tiene IP privada en el > ultimo campo VIA. > > # > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+ > > > > > > if (client_nat_test("3")) { > > log(1, "NAT: Requerimiento de IP privada --> > fixed contact > >(en rutina principal)\n"); > > setflag(5); > > force_rport(); > > > > if (method=="REGISTER") { > > fix_nated_contact(); > > } else { > > fix_contact(); > > }; > > append_hf("P-hint: fixed NAT contact for > request\r\n"); > > }; > > > > > > if (uri==myself) { > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++ > > # Chequeo de REGISTRO para equipos del dominio 1 > > # Dominio Voiss : sip.mydomain.com > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++ > > > > if (method=="REGISTER" && > uri=~"^sip:.*sip.mydomain.com") { > > # Challenge/Response > > if ( > !radius_www_authorize("sip.mydomain.com")) { > > > www_challenge("sip.mydomain.com", "1"); > > break; > > }; > > > > # Mandamos un mensaje de Log si es > que un cliente > >NATEADO se registro. > > if (isflagset(5)) { > > log(1, "NAT: Cliente > NAT'eado registrado > >(dentro del chequeo de REGISTER)\n"); > > }; > > > > save("location"); > > break; > > }; # Fin chequeo de Registro para Dominio : > sip.mydomain.com > > > > > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++ > > # Chequeo de REGISTRO para equipos del dominio 2 > > # Dominio Mayoristas : sip2.mydomain.net > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++ > > > > if (method=="REGISTER" && > uri=~"^sip:.*sip2.mydomain.net") { > > # Challenge/Response > > if ( > !radius_www_authorize("sip2.mydomain.net")) { > > > www_challenge("sip2.mydomain.net", "1"); > > break; > > }; > > > > # Mandamos un mensaje de Log si es > que un cliente > >NATEADO se registro. > > if (isflagset(5)) { > > log(1, "NAT: Cliente > NAT'eado registrado > >(dentro del chequeo de REGISTER)\n"); > > }; > > > > save("location"); > > break; > > }; # Fin chequeo de Registro para Dominio : > >sip2.mydomain.net > > > > > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++ > > # CHEQUEO DEL INVITE PARA TODAS LAS LLAMADAS > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++ > > > > if (method=="INVITE" && src_ip!=xx.xx.xx.xx) { > > if ( !radius_proxy_authorize("")) { > > proxy_challenge("", "1"); > > log (1, "LOG: Llamada no > AUTORIZADA\n"); > > break; > > }; > > }; > > > > > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++++ > > # Llamadas hacia dominio H.323 > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++++ > > > > if ( (uri=~"^sip:717.*@.*") | > (uri=~"^sip:707.*@.*") | > >(uri=~"^sip:777.*@.*") | (uri=~"^sip:333.*@.*") | (uri=~"^sip > >:0.*@.*") ){ > > log(1,"FORWARDING : Llamada hacia plataforma > >H.323\n"); > > route(1); > > break; > > }; > > > > > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++ > > # Llamadas SIP locales > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++ > > > > if (!lookup("location")) { #Usuarios que no estan en > >"location" database. > > route(4); > > break; > > }; > > > > > > setflag(3); > > > > > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++ > > # Manejo para llamadas NAT'das > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++ > > > > if (method=="INVITE") { > > log(1, "ROUTING: Reply processing > (online user) > >enabled to handle NAT'd resonses\n"); > > t_on_reply("1"); > > if ( (isflagset(5)) && > (method=="INVITE" || > >method=="ACK") ) { > > log(1, "NAT: Invite > received --> > >enabling media proxy (else del lookup 2do if)\n"); > > use_media_proxy(); > > append_hf("P-hint: > request forced to > >media proxy\r\n"); > > }; > > }; > > > > # Do it ? (Ricardo Martinez) > > }; > > > > append_hf("P-hint: USRLOC\r\n"); > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+++++ > > # Efectuamos la llamada > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+++++ > > > > if (!t_relay()) { > > sl_reply_error(); > > break; > > }; > > > >} /* end of initial routing logic */ > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++++++++++++ > ># COMIENZO DE LA SECCION DE RUTAS > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++++++++++++ > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >++++++++++++++++++ > ># Route 1 : Llamadas con destino plataforma > >H.323#+++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+++++++++++++++++++++++ > >route[1] > >{ > > # send it out now; use stateful forwarding as it > works reliably > > # even for UDP2TCP > > > > # Esta NATEADO? > > if (isflagset(5)) { > > log(1, "NAT: At least one participant NAT'd > --> enable reply > >processing (hacia GW)\n"); > > t_on_reply("1"); > > if (method=="INVITE") { > > log(1, "NAT: Invite received --> > enabling proxied > >media session (hacia GW)\n"); > > setflag(1); > > use_media_proxy(); > > }; > > }; > > > > rewritehostport("mygw.mydomain.com:5060"); > > append_hf("P-hint: GATEWAY\r\n"); > > > > if (!t_relay()) { > > sl_reply_error(); > > break; > > }; > > > >} > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+++++++++++++++++ > ># Route 4 : Llamadas a usuarios Off-line > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+++++++++++++++++ > > > >route[4] { > > > > if (!t_newtran()) { > > sl_reply_error(); > > }; > > > > if (!t_reply("404", "Not Found IT!")) { > > sl_reply_error(); > > }; > > break; > >} > > > > > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+++++++++++++++++ > ># On Reply Route 1 : Equipos NATeados!! > >#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ > +++++++++++++++ > >+++++++++++++++++ > > > >onreply_route[1] { > > if ((isflagset(5) || client_nat_test("3")) && > >(status=~"(180)|(183)|2[0-9][0-9]")) { > > if (!search("^Content-Length:\ 0")) { > > use_media_proxy(); > > }; > > }; > > > > if (client_nat_test("1")) { > > fix_nated_contact(); > > }; > > > > > >} > > > > > > > > > >Ricardo Martinez.- > > > >_______________________________________________ > >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] Precall Speech with SER and SEMS
by matt morris 20 Apr '05

20 Apr '05

Hello List, I would like to implement a precall speech feature with SER with possibly SEMS. Basically for a certain of group of our user we would like them to hear an audio before establishing their calls with the callee. However, I don't have much clues on how to do this. I have tried the Asterisk B2BUA by Mr. Mike Tkachuk (which worked) but I don't want the media stream to be handled by the B2BUA. Unfortunately, I do not know how to configure a signalling-only B2BUA with the playback - seems whenever I used the playback function in Asterisk and subsequent call media will be have to be handled by Asterisk. That is why I want to explore the possibility of another approach with SEMS. Would it be possible for the call to be first forwarded to the SER+SEMS box to play the announcement AND THEN forward the call to the actual called number? Please shed some light on how I could make this happen. Thank you very much. _________________________________________________________________ Scan and help eliminate destructive viruses from your inbound and outbound e-mail and attachments. http://join.msn.com/?pgmarket=en-ca&page=byoa/prem&xAPID=1994&DI=1034&SU=ht… Start enjoying all the benefits of MSN® Premium right now and get the first two months FREE*.

1 0

[Serusers] resolv problem ser 0.8.14
by Bruno Oliveira 20 Apr '05

20 Apr '05

hi, when beginning the server to be appears the message in log: localhost ser: WARNING: could not rev. resolve 192.168.1.100 anyone knows because? -- []s, Bruno "Niggas" Oliveira Belo Horizonte - MG Msn: n1gg4s(a)gmail.com Icq: 176314647 "Todo o nosso descontentamento por aquilo que nos falta procede da nossa falta de gratidão por aquilo que temos."

1 0

[Serusers] Re: Strange error on registration - Still a Problem in a Certain Scenario
by reticent 20 Apr '05

20 Apr '05

I was looking back through the mailing list archives and i found this email regarding failed inserts into the location table of a database due to duplicate entries and there is still a situation where this problem will occur. For the full original Thread, this is where is starts http://lists.iptel.org/pipermail/serusers/2004-April/007151.html http://lists.iptel.org/pipermail/serusers/2004-April/thread.html#7173 Basically the issue is that the database has an entry for a user that SER does not have in memory so when a registration comes in SER uses an INSERT statement instead of an UPDATE which fails because a key (username, contact, domain) already exists for the user in the database. The situation where it still occurs is when the database goes away and during the hiatus a registration expires, SER wipes the registration from memory and fails on the database delete. SER will recover from this situation only when it is stopped and started, this complicates load balanced situations where it is expected that SER can recover from certain failures (database failures in this case) without any administrive intervention

1 0

[Serusers] mediaproxy timeout problem
by Paul van Schagen 20 Apr '05

20 Apr '05

Hi I am testing mediaproxy with ser. I have a config with a Grandstream handytone 486 as a UA and a CISCO 5350 as a gateway. My sip server is SER 0.9.0 . I find that when setting up a phone call and disconnecting the Grandstream ( power off or disconnecting LAN cable ) the mediaproxy doesn't timeout and disconnect the call. This is what I have done ... Grandstream 486 -------- SER 0.9.0 with mediaproxy ------------CISCO 5350 I setup a call from a Grandstream Handytone 486 to the PSTN. and see this on the proxy server. root# ./sessions.py Caller Via Called Status Duration Codec Type Traffic ------------------------------------------------------------------------ --------------------------------- ------------ 212.4.174.186:18400 - 217.194.194.181:35008 - 83.157.4.99:56364 active 0'34" G723 Audio 35.81k/53.59k/18.71k Total traffic: 4.09kbps/352bps/4.44kbps (in1/in2/out) Session count: 1 Now I disconnect the grandstream from the LAN cable in the middle of the conversation to test the timeout function of the mediaproxy. now I get this. root # ./sessions.py Caller Via Called Status Duration Codec Type Traffic ------------------------------------------------------------------------ --------------------------------------------- 212.4.174.186:16652 - 217.194.194.181:35010 - 83.157.4.99:56364 active 0'36" G723 Audio 62.92k/74.53k/12.77k Total traffic: 16.53kbps/0bps/15.47kbps (in1/in2/out) Session count: 1 The disconnect is reflected in in2 (0bps ), the connection stays active because I keep on receiving UDP packets ( background noise maybe or somebody talking ) from the PSTN gateway. Is it correct to assume there is no end to end communication anymore as soon as one of both RTP streams is in 0bps ? or could it be that in that case there is silence from one end ? If it is correct to assume the connection to the UA has been lost as soon as there is no UDP data coming from it , how can I modify mediaproxy in a way it will disconnect on either one of the inputs being in 0bps? any help highly appreciated . Paul van Schagen

1 0

[Serusers] SEMS conference module (no sound)
by Oscar Patricio 20 Apr '05

20 Apr '05

Hi! I'm having trouble setting up SEMS conference calls! When i call the conference number everything goes ok. As soon as the second user joins, we enter in conference, but there is no sound!! I see udp trafic to and from both clients, but i hear no sound! As anyone ever had a similar problem? Thanx! Óscar

1 0

RE: [Serusers] LCR failure syntax
by Matt Schulte 20 Apr '05

20 Apr '05

FYI, I *finally* got around to doing this. A debug shows that load_gw is in fact loading both gateways.. 8(30980) DEBUG:load_gws(): Added AVP <lcr_gw_addr,374650ce> 8(30980) qm_malloc(0xf4edf000, 31) called from usr_avp.c: add_avp(112) 8(30980) qm_malloc(0xf4edf000, 32) returns address 0xf509a3ec frag. 0xf509a3d4 (size=32) on 1 -th hit 8(30980) DEBUG:load_gws(): Added AVP <lcr_gw_port,0> 8(30980) qm_malloc(0xf4edf000, 31) called from usr_avp.c: add_avp(112) 8(30980) qm_malloc(0xf4edf000, 32) returns address 0xf509a43c frag. 0xf509a424 (size=32) on 1 -th hit 8(30980) DEBUG:load_gws(): Added AVP <lcr_gw_addr,511f7d1> 8(30980) qm_malloc(0xf4edf000, 31) called from usr_avp.c: add_avp(112) 8(30980) qm_malloc(0xf4edf000, 32) returns address 0xf509a48c frag. 0xf509a474 (size=32) on 1 -th hit 8(30980) DEBUG:load_gws(): Added AVP <lcr_gw_port,0> Thoughts? Thanks. -----Original Message----- From: Juha Heinanen [mailto:jh@tutpro.com] Sent: Tuesday, April 12, 2005 11:59 PM matt, you could first check which gws get loaded by turning debugging on: DBG("load_gws(): DEBUG: Added gw_addr_port_avp <%x, %d>\n", *((unsigned int *)(val.s->s)), *((unsigned short *)(val.s->s + 4))); -- juha

1 0

RE: [Serusers] failure_route vs. onreply
by Matt Schulte 20 Apr '05

20 Apr '05

Thanks for clearing that up... -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Wednesday, April 20, 2005 9:17 AM To: Matt Schulte Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] failure_route vs. onreply They are quite different. onreply_route is executed for each branch. If your proxy server forks an INVITE to 3 destinations in parallel then onreply_route will be executed 3 times -- for each branch. onreply_route sections process replies (all operations within them will modify replies). After receiving final replies from all the branches, the proxy server decides which of them it should forward upstream. If the selected message is a negative final reply (3xx, 4xx, 5xx, 6xx) then it executes the failure_route section before forwarding it upstream. But -- the failure_route section does not work with the reply being forwarded upstream, but with the original request ! (in this case the original INVITE). That's quite important. There are two reasons why it is this way: 1) The most common operation performed in failure_route sections is serial forking (you take the original message and forward it somewhere else) -- and for that you need the original request. You may also want to change the reply being forwarded upstream and for this the original request will be needed as well. 2) (implementation related) -- it is likely that the reply was received by process A, but will be forwarded upstream by process B (where B is the process which received the reply from the last unfinished branch). To make it possible, the message is stored in shared memory and most functions that can be called from the script cannot handle it. +-----+ (2) 301 301 (1) | |<-------- <---------| SER | (2) 486 | |<-------- +-----+ (1) -- failure_route gets called here (2) -- onreply_route gets called here Jan. On 20-04 07:31, Matt Schulte wrote: > Ok, are these the same thing? I remember reading that onreply was > renamed to failure due to confusion? Now I'm confused again ;-) > > Matt > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Serious Problems with SER - Not responding anything.
by Ricardo Martinez 20 Apr '05

20 Apr '05

Hello list. I'm experiencing problems with my SER. As i mentioned in a past post (http://lists.iptel.org/pipermail/serusers/2005-April/018324.html), from time to time (it is getting pretty often) my SER not answering any request, when a REGISTER reach the server it is not challenged, this happens too with the INVITE's messages. About two hours ago this happened again. I asked in my last post how to LOG the internal errors or messages from SER to see if i can get some idea about what's going on, but i have no answer about it. This is the infornmation i was able to capture when this was happening : the serctl moni command shows a lot of "failures". [cycle #: 11; if constant make sure server lives and fifo is on] Server: Sip EXpress router (0.8.14-3 (i386/linux)) Now: Wed Apr 20 12:56:29 2005 Up Since: Thu Apr 14 21:32:18 2005 Up time: 487451 [sec] Transaction Statistics Current: 6 (2612 waiting) Total: 38194 (0 local) Replied localy: 80752 Completion status 6xx: 4643, 5xx: 768, 4xx: 10664, 3xx: 0,2xx: 22552 Stateless Server Statistics 200: 5975004 202: 0 2xx: 0 300: 0 301: 0 302: 0 3xx: 0 400: 0 401: 205980 403: 0 404: 0 407: 22918 408: 0 483: 0 4xx: 19 500: 0 5xx: 0 6xx: 0 xxx: 0 failures: 24180 UsrLoc Stats Domain Registered Expired 'location' 219 4552 -------------------------------------------------------------- [cycle #: 5; if constant make sure server lives and fifo is on] Server: Sip EXpress router (0.8.14-3 (i386/linux)) Now: Wed Apr 20 12:56:47 2005 Up Since: Thu Apr 14 21:32:18 2005 Up time: 487469 [sec] Transaction Statistics Current: 4 (2619 waiting) Total: 38219 (0 local) Replied localy: 80762 Completion status 6xx: 4643, 5xx: 770, 4xx: 10667, 3xx: 0,2xx: 22552 Stateless Server Statistics 200: 5975030 202: 0 2xx: 0 300: 0 301: 0 302: 0 3xx: 0 400: 0 401: 205995 403: 0 404: 0 407: 22922 408: 0 483: 0 4xx: 19 500: 0 5xx: 0 6xx: 0 xxx: 0 failures: 24380 UsrLoc Stats Domain Registered Expired 'location' 220 4552 ----------------------------------------------------------------- [cycle #: 2; if constant make sure server lives and fifo is on] Server: Sip EXpress router (0.8.14-3 (i386/linux)) Now: Wed Apr 20 12:56:52 2005 Up Since: Thu Apr 14 21:32:18 2005 Up time: 487474 [sec] Transaction Statistics Current: 4 (2620 waiting) Total: 38224 (0 local) Replied localy: 80762 Completion status 6xx: 4643, 5xx: 770, 4xx: 10667, 3xx: 0,2xx: 22552 Stateless Server Statistics 200: 5975034 202: 0 2xx: 0 300: 0 301: 0 302: 0 3xx: 0 400: 0 401: 206000 403: 0 404: 0 407: 22923 408: 0 483: 0 4xx: 19 500: 0 5xx: 0 6xx: 0 xxx: 0 failures: 24445 UsrLoc Stats Domain Registered Expired 'location' 215 4557 What is happening to cause the "failures" counter grows that much?. The free command shows this : [root@root]# free total used free shared buffers cached Mem: 1030888 626464 404424 0 86692 369848 -/+ buffers/cache: 169924 860964 Swap: 2040244 4864 2035380 The top command shows not much use of CPU. The proccess were running (the SER and mediaproxy) Despite of all this, the SER was not answering anything. Could this be a bug? or maybe a problem with my ser.cfg file?. I'm attaching my ser.cfg file too to see if someone can tell me if there is something wrong about it. I really hope that someone could help me, this is getting really complicated every time it happens. Thanks Ricardo Martinez.- ************************************************** SER.CFG ************************************************** #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ ++++++++++++++++++++++++++++++ # ----------- global configuration parameters ------------------------ debug=0 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) #Uncomment these lines to enter debugging mode #fork=no #log_stderror=yes listen=ip.of.my.ser alias=sip.mydomain.com alias=sip2.mydomain.net alias=ip.of.my.ser 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" fifo_mode=0666 # ------------------ 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/textops.so" loadmodule "/usr/local//lib/ser/modules/exec.so" # ++++++++++ Para trabajar con NAT loadmodule "/usr/local//lib/ser/modules/mediaproxy.so" loadmodule "/usr/local//lib/ser/modules/domain.so" loadmodule "/usr/local//lib/ser/modules/dbtext.so" loadmodule "/usr/local//lib/ser/modules/nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" # ++++++++++ Modulos de Accounting y Autorizacion via Radius loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/group_radius.so" loadmodule "/usr/local/lib/ser/modules/acc.so" # ----------------- setting module-specific parameters --------------- # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # +++++ module authorization modparam("auth_radius","radius_config","/usr/local/etc/radiusclient/radiuscl ient.conf") modparam("auth_radius","service_type",15) # +++++ module accounting modparam("acc","radius_config","/usr/local/etc/radiusclient/radiusclient.con f") modparam("acc","log_level",1) modparam("acc", "service_type", 15) modparam("acc", "radius_flag", 1) modparam("acc", "radius_missed_flag", 3) modparam("acc", "failed_transactions", 1) modparam("acc", "report_ack", 0) # +++++ group radius modparam("group_radius","radius_config","/usr/local/etc/radiusclient/radiusc lient.conf") modparam("group_radius", "use_domain", 1) # +++++ module mediaproxy modparam("mediaproxy", "natping_interval", 60) modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") #modparam("mediaproxy", "sip_asymmetrics", "/usr/local/etc/ser/sip-asymmetrics-clients") #modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/ser/rtp-asymmetrics-clients") modparam("registrar", "nat_flag", 5) # +++++ module domain modparam("domain", "db_url", "/usr/local/etc/ser/domaintables") modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") # +++++ module tm modparam("tm", "fr_timer", 15) modparam("tm", "fr_inv_timer", 22) modparam("tm", "wt_timer", 5) # +++++ module registrar (Con estos parametro haces que el location lookup priorize solo los mas recientes contacts) modparam("registrar", "append_branches", 0) modparam("registrar", "desc_time_order", 1) # ------------------------- request routing logic ------------------- # main routing logic route { #++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ # Llamada a Rutina Externa "translate" para cambio de numero #++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ if (method=="INVITE" || method=="CANCEL") { exec_dset("/usr/local/etc/ser/translate"); }; #++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ # 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; }; #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ # Marcamos todos los mensajes para que pasen por nuestro servidor # a traves del record route. #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ record_route(); #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ # Permitimos el uso del campo Route en el ruteo de la llamada # #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ if (method=="BYE" || method=="CANCEL") { log(1, "NAT: BYE o CANCEL recibido --> terminando la sesion de media\n"); end_media_session(); setflag(1); }; if (loose_route()) { t_relay(); break; }; #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ + # Manejo de los mensajes NOTIFY para Keep Alive de NAT #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ + if ((method=="NOTIFY") && search("^Event: keep-alive")) { sl_send_reply("200", "OK"); break; }; #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ + # Marcamos las llamadas para accounting #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ + setflag(1); #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ + # Test para clientes NATeados # # 1 : Chequea si el cliente tiene IP privada en el campo Contact. # 2 : Chequea si el cliente se contacto con una IP distinta a la del campo VIA. # 3 : Chequea si el cliente tiene IP privada en el ultimo campo VIA. # #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ + if (client_nat_test("3")) { log(1, "NAT: Requerimiento de IP privada --> fixed contact (en rutina principal)\n"); setflag(5); force_rport(); if (method=="REGISTER") { fix_nated_contact(); } else { fix_contact(); }; append_hf("P-hint: fixed NAT contact for request\r\n"); }; if (uri==myself) { #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ # Chequeo de REGISTRO para equipos del dominio 1 # Dominio Voiss : sip.mydomain.com #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ if (method=="REGISTER" && uri=~"^sip:.*sip.mydomain.com") { # Challenge/Response if ( !radius_www_authorize("sip.mydomain.com")) { www_challenge("sip.mydomain.com", "1"); break; }; # Mandamos un mensaje de Log si es que un cliente NATEADO se registro. if (isflagset(5)) { log(1, "NAT: Cliente NAT'eado registrado (dentro del chequeo de REGISTER)\n"); }; save("location"); break; }; # Fin chequeo de Registro para Dominio : sip.mydomain.com #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ # Chequeo de REGISTRO para equipos del dominio 2 # Dominio Mayoristas : sip2.mydomain.net #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ if (method=="REGISTER" && uri=~"^sip:.*sip2.mydomain.net") { # Challenge/Response if ( !radius_www_authorize("sip2.mydomain.net")) { www_challenge("sip2.mydomain.net", "1"); break; }; # Mandamos un mensaje de Log si es que un cliente NATEADO se registro. if (isflagset(5)) { log(1, "NAT: Cliente NAT'eado registrado (dentro del chequeo de REGISTER)\n"); }; save("location"); break; }; # Fin chequeo de Registro para Dominio : sip2.mydomain.net #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ ++ # CHEQUEO DEL INVITE PARA TODAS LAS LLAMADAS #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ ++ if (method=="INVITE" && src_ip!=xx.xx.xx.xx) { if ( !radius_proxy_authorize("")) { proxy_challenge("", "1"); log (1, "LOG: Llamada no AUTORIZADA\n"); break; }; }; #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ ++++++++++ # Llamadas hacia dominio H.323 #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ ++++++++++ if ( (uri=~"^sip:717.*@.*") | (uri=~"^sip:707.*@.*") | (uri=~"^sip:777.*@.*") | (uri=~"^sip:333.*@.*") | (uri=~"^sip :0.*@.*") ){ log(1,"FORWARDING : Llamada hacia plataforma H.323\n"); route(1); break; }; #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ ++++++++ # Llamadas SIP locales #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ ++++++++ if (!lookup("location")) { #Usuarios que no estan en "location" database. route(4); break; }; setflag(3); #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ ++++++++ # Manejo para llamadas NAT'das #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ ++++++++ if (method=="INVITE") { log(1, "ROUTING: Reply processing (online user) enabled to handle NAT'd resonses\n"); t_on_reply("1"); if ( (isflagset(5)) && (method=="INVITE" || method=="ACK") ) { log(1, "NAT: Invite received --> enabling media proxy (else del lookup 2do if)\n"); use_media_proxy(); append_hf("P-hint: request forced to media proxy\r\n"); }; }; # Do it ? (Ricardo Martinez) }; append_hf("P-hint: USRLOC\r\n"); #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ +++++ # Efectuamos la llamada #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ +++++ if (!t_relay()) { sl_reply_error(); break; }; } /* end of initial routing logic */ #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ ++++++++++++++++++ # COMIENZO DE LA SECCION DE RUTAS #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ ++++++++++++++++++ #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ ++++++++++++++++++ # Route 1 : Llamadas con destino plataforma H.323#++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ +++++++++++++++++++++++ route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP # Esta NATEADO? if (isflagset(5)) { log(1, "NAT: At least one participant NAT'd --> enable reply processing (hacia GW)\n"); t_on_reply("1"); if (method=="INVITE") { log(1, "NAT: Invite received --> enabling proxied media session (hacia GW)\n"); setflag(1); use_media_proxy(); }; }; rewritehostport("mygw.mydomain.com:5060"); append_hf("P-hint: GATEWAY\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ +++++++++++++++++ # Route 4 : Llamadas a usuarios Off-line #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ +++++++++++++++++ route[4] { if (!t_newtran()) { sl_reply_error(); }; if (!t_reply("404", "Not Found IT!")) { sl_reply_error(); }; break; } #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ +++++++++++++++++ # On Reply Route 1 : Equipos NATeados!! #+++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++++ +++++++++++++++++ onreply_route[1] { if ((isflagset(5) || client_nat_test("3")) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ 0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } Ricardo Martinez.-

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Re: [Serusers] failure_route vs. onreply
by Jiri Kuthan 20 Apr '05

20 Apr '05

At 02:31 PM 4/20/2005, Matt Schulte wrote: >Ok, are these the same thing? no failure_route reprocesses requests which failed. onreply processes replies coming back. -jiri >I remember reading that onreply was >renamed to failure due to confusion? Now I'm confused again ;-) > > Matt > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[Serusers] Probles with serctl
by Edgar Vidal 20 Apr '05

20 Apr '05

I've read the install file and commented and uncommented the corresponding lines in order to have authorization and be able to use the database of serweb; however, after I changed the domain in ser.cfg and export the SIP_DOMAIN variable and instead of starting the server by typing just "ser", I typed serctl start but I got the next error message: cat: /var/run/ser.pid: No suche file or directory I have ser-0.8.14_linux_i386.tar.gz and serweb_2004-07-27.tar.gz Could anybody help me? Thanks in advance Edgar Vidal SAS Institute México, S.A. de C.V. Edgar.Vidal(a)sas.com Tel. (52) 55 5000 3813 SAS ... The Power to Know

2 1

[Serusers] failure_route vs. onreply
by Matt Schulte 20 Apr '05

20 Apr '05

Ok, are these the same thing? I remember reading that onreply was renamed to failure due to confusion? Now I'm confused again ;-) Matt

2 1

[Serusers] differences
by Llanos Serna García-Conde 20 Apr '05

20 Apr '05

2 1

[Serusers] Ser 0.9.0 and Cisco Voice Gateway
by Ahmad Cheikh-Moussa 20 Apr '05

20 Apr '05

Hi! I'am new in to this VoIP technolgie. I have a SER 0.9.0 wit Cisco IOS Gatewy. I only want to configure a registrar Server with a Cisco gateway, which is connected to the PSTN. Can someone give me an ser.cfg example ? How can I tell the ser server that every call, which starts with 0, route directly to the IOS Gateway ? I added two user to the ser db. The users got a login, password and an email. Additional to this I added to both user a userloc with serctl ul add <user> sip:1223@domain.de. Both users can not phone each other. I got an busy tone and my Cisco 7960 Phone shows "reorder". Did I forgot something ? Here an output of serctl ul show: eagle[admin] # ./serctl ul show Dumping all contacts may take long: are you sure you want to proceed? [Y|N] y ===Domain list=== ---Domain--- name : 'aliases' size : 512 table: fcda0dd8 d_ll { n : 0 first: 0 last : 0 } ---/Domain--- ---Domain--- name : 'location' size : 512 table: fcd9e8b8 d_ll { n : 2 first: fcda08e8 last : fcda0af8 } ...Record(fcda08e8)... domain: 'location' aor : 'ahmad02' ~~~Contact(fcda0978)~~~ domain : 'location' aor : 'ahmad02' Contact : 'sip:460@netuse.de' Expires : -60589 q : 1 Call-ID : 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything' CSeq : 42 replic : 0 User-Agent: 'SIP Express Router FIFO' received : '' State : CS_SYNC Flags : 128 next : 0 prev : 0 ~~~/Contact~~~~ .../Record... ...Record(fcda0af8)... domain: 'location' aor : 'ahmad' ~~~Contact(fcda0b88)~~~ domain : 'location' aor : 'ahmad' Contact : 'sip:459@netuse.de' Expires : -60655 q : 1 Call-ID : 'The-Answer-To-The-Ultimate-Question-Of-Life-Universe-And-Everything' CSeq : 42 replic : 0 User-Agent: 'SIP Express Router FIFO' received : '' State : CS_SYNC Flags : 128 next : 0 prev : 0 ~~~/Contact~~~~ .../Record... ---/Domain--- ===/Domain list=== Thanks in advance, Ahmad -- Ahmad Cheikh-Moussa NetUSE AG Dr.-Hell-Straße, 24107 Kiel, Germany Telefon: +49 431 2390 400 -- Telefax: +49 431 2390 499 Service: Service(a)NetUSE.DE -- http://NetUSE.DE/

3 4

[Serusers] [PATCH] add mysql reconnect option for ser-0.8.14
by matteo brancaleoni 20 Apr '05

20 Apr '05

Hi, as many of you knows, mysql module on ser-0.8.14 (latest) lacks the option to reconnect to the DB server if the server goes away (ie server restarts, crash, blah). With this simple patch, I backported the reconnect option of ser CVS, along with the ping server option. You can use these 2 module params: -> modparam("mysql","auto_reconnect",1) // can be 0 or 1 -> modparam("mysql","ping_interval",60) // in seconds. if the connection has been idle for more than ping_interval, // the server will be pinged before the next query There's another addition: since Mysql 5.03 the reconnect option in the MySQL struct is disabled, so this patch takes care also to enable this flag (this thing is not done by the CVS version). Hope can be of any help, Matteo.

1 0

[Serusers] search string, circumflex
by Che Sosa 20 Apr '05

20 Apr '05

Hello list, I am not so familiar with C syntax but I suspect that the circumflex changes the parsing method. What is the difference between these two: first-> search("^(Contact|m): .*(a)(192\.168\.|10\.|172\.16)") second-> search("(From|f):.*@foo.bar") Questions: 1. How does the circumflex affect the string inside ( ) 2. How is ( ) used? Thanks, Che

3 2

[Serusers] starting v0.9.0 problems
by Noel Paul Evangelista 20 Apr '05

20 Apr '05

hi guys, i need help please. newbie here. i just installed v0.9.0(ser-0.9.0_src.tar) using "make prefix=/usr/local all" then "make prefix=/usr/local install". looks like everything was installed. ser.cfg is /usr/local/etc/ser and executables are in /usr/local/sbin. the modules are in their correct directory as shown in the ser.cfg. my problem is when i do a "serctl start" i get this problem: [root@SIPserver sbin]# serctl start Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed secondly, if i do a "serctl moni" i get this message: [root@SIPserver sbin]# serctl moni Error opening ser's FIFO /tmp/ser_fifo Make sure you have line fifo=/tmp/ser_fifo in your config i am sure that the default ser.cfg has the line fifo=/tmp/ser_fifo. thirdly, if i do a "ser start" i get this message but it looks like ser is not still running: [root@SIPserver sbin]# ser start Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 146.82.219.99 [146.82.219.99]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 146.82.219.99 [146.82.219.99]:5060 Aliases: tcp: SIPserver.islandtel.com.ph:5060 tcp: localhost:5060 tcp: localhost.localdomain:5060 tcp: SIPserver:5060 udp: SIPserver.islandtel.com.ph:5060 udp: localhost:5060 udp: localhost.localdomain:5060 udp: SIPserver:5060 [root@SIPserver sbin]# other info about the server are the following: [root@SIPserver sbin]# ser -V version: ser 0.9.0 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, DBG_QM_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.197 2004/12/03 19:09:31 andrei Exp $ main.c compiled on 14:40:36 Apr 19 2005 with gcc 3.2 [root@SIPserver sbin]# uname -a Linux SIPserver 2.4.20-8 #1 Thu Mar 13 17:54:28 EST 2003 i686 i686 i386 GNU/Linux [root@SIPserver sbin]# any help regarding my problem will be highly appreciated. best regards, Noel __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Serusers] SER & Mediaproxy: XLite - Audio OK, Sipura - No audio
by Cameron Beattie 20 Apr '05

20 Apr '05

I am using the onsip ser.cfg for mediaproxy. All seems OK when I call between two X-Lite clients. However calls between X-Lite and a Sipura 2000 have no audio in either direction. Both the X-Lite clients and the Sipura are on the same network behind NAT and going out to the SER/Mediaproxy server which has a public IP. SER and Mediaproxy are on the same server. I've tested SJPhone to Sipura and there's no audio also, so it seems like a Sipura issue. I've set NAT mapping enable and NAT Keep Alive enable to yes on the Sipura. What am I missing? Regards Cameron

3 9

[Serusers] serctl to remote mysql database
by TCCHAN 20 Apr '05

20 Apr '05

Hi: I know that ser itself can work with a remote database through changing the ser.cfg but can serctl add/remove user to a remote mysql database. Thanks for any help.

2 1

[Serusers] problem with call
by Edgardo O. Gonzales II 20 Apr '05

20 Apr '05

Hi there! I was able to install ser 0.8.14 on my debian box and make a call. However, I can hear echo every time I speak. Is there something missing with my config that can filter / stop this problem. Another, mysql authentication is not working though my config file is ok as reported during compilation. Can somebody help me in building a simple ser config file that use mysql to authenticate user. thanks in advance. ed

2 1

[Serusers] problem with special user for SER running
by Grigory Fishilevich 20 Apr '05

20 Apr '05

Hi, i install SER 0.8.14 on my SUSE Linux Enterprise Server 9. it works. but i dont like to run SER under root-account, so i add a new system user "ser", i can start SER, but serctl have problems with access to fifo. After "serctl moni" i see in /var/log/messages following: Apr 20 09:31:57 xxx /sbin/ser[8669]: ERROR: open_reply_pipe: open error (/tmp/ser_receiver_8732): Permission denied Apr 20 09:31:57 xxx /sbin/ser[8669]: ERROR: fifo_reply: no reply pipe /tmp/ser_receiver_8732 the user "ser" is member of group "users". how can i allow the user "ser" to create and access the fifos? many thanks, Grigory Fishilevich

3 2

[Serusers] Registration Time out
by Ramakrishnan_S 20 Apr '05

20 Apr '05

Hello, Thanks for reading this mail. I have installed latest SER in linux. With serctl moni, it shows 0 failures.But when we try to register from a SIP client, it times out always . The ps -eaf | grep ser shows several process running. Any pointers on how to test this will be really helpful. Thanks Ramki DISCLAIMER: This email (including any attachments) is intended for the sole use of the intended recipient/s and may contain material that is CONFIDENTIAL AND PRIVATE COMPANY INFORMATION. Any review or reliance by others or copying or distribution or forwarding of any or all of the contents in this message is STRICTLY PROHIBITED. If you are not the intended recipient, please contact the sender by email and delete all copies; your cooperation in this regard is appreciated.

1 0

[Serusers] (no subject)
by ljzhang 20 Apr '05

20 Apr '05

Hi all, Here is my confusion about SER supporting NAT transversal. 1. With only RTPProxy, what kinds of types of NAT can SER with nathelper transverse? 2. What's the difference between MediaProxy and RTPProxy? 3. If UA supports STUN, do SER need to configure nathelper to support NAT transversal? Just because I knew STUN protocol only process a few kinds of types of NAT, I want to know what advantage it will add if I configure SER with nathelper. Thanks a lot! 　　　　　　　　ljzhang 　　　　　　　　xueyou38(a)163.com 　　　　　　　　　　2005-04-12

2 1

[Serusers] (no subject)
by ljzhang 20 Apr '05

20 Apr '05

Hi all, Here is my confusion about SER supporting NAT transversal. 1. With only RTPProxy, what kinds of types of NAT can SER with nathelper transverse? 2. What's the difference between MediaProxy and RTPProxy? 3. If UA supports STUN, do SER need to configure nathelper to support NAT transversal? Just because I knew STUN protocol only process a few kinds of types of NAT, I want to know what advantage it will add if I configure SER with nathelper. Thanks a lot! 　　　　　　　　ljzhang 　　　　　　　　xueyou38(a)163.com 　　　　　　　　　　2005-04-12

2 1

[Serusers] Bug # 2925
by Darryl H. Thomas 19 Apr '05

19 Apr '05

Hello everyone, I was wondering whether anyone knew what was going on with the following bug: http://developer.berlios.de/bugs/?func=detailbug&bug_id=2925&group_id=480 For those not wanting to follow the links, this is a bug submitted by someone back in December wherein ser puts 'branch=0' in the Via header it creates for an ACK to a 200 OK Invite response. I haven't seen any activity related to this, and it's affecting interoperability with one of our termination providers. 0.9.0 users, have you seen the same problem?? Cheers, Darryl

2 1

Re: [Serusers] inside calls and PSTN calls
by Llanos Serna García-Conde 19 Apr '05

19 Apr '05

2 1

[Serusers] ser 0.9.0 and Cisco SIP Gateway
by Ahmad Cheikh-Moussa 19 Apr '05

19 Apr '05

Hi! I want to test a sip configuration with SER and Cisco IOS. Can someone tell me how to configure this ? Can someone send me an example configuration ? How do I associate a user with a directory number ? For what is the Caller-ID in the user information ? My configuration should be: SIP ua -- SER Server -->Cisco Gateway --> PSTN I know that are a lot of questions, but I don't know where else to find this information. The admin Guide and install guide didn't helped. I searched a lot of pages, mailing list and cisco documentation, but nowhere I could find any examples. Thanks in advance, Ahmad -- Ahmad Cheikh-Moussa NetUSE AG Dr.-Hell-Straße, 24107 Kiel, Germany Telefon: +49 431 2390 400 -- Telefax: +49 431 2390 499 Service: Service(a)NetUSE.DE -- http://NetUSE.DE/

1 0

[Serusers] alias in serweb
by Rosario Pingaro 19 Apr '05

19 Apr '05

I'm running serweb installed from cvs a month ago. I'd like to create aliases from serweb but I didn't find the appropriate config, there is an alias.php but I dont have the alias tab. Does some one knows hoe to enable it? thanks Rosario

1 0

[Serusers] change info in message status
by deb ale 19 Apr '05

19 Apr '05

Hi, to connect LCS my pstn gateway through SER I have to change from sip:user1@pdc.local.aastra.net to from sip:user1@192.168.1.108 when I receive the status eg 180 ringing I have to make the reverse operation. For that I think I have to use t_on_reply("1"); onreply_route[1] {...} But I never go onreply_route[1]. I am wrong but don't understand why. Here after my ser.cfg file. Thanks for help, Alexis # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ #debug=3 # debug level (cmd line: -dddddddddd) #fork=no #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/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/exec.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" # 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" #Modif ADE chargement module loadmodule "/usr/local/lib/ser/modules/textops.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 ( 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 record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; #ajout alexis external parsing for alaexis.debreu(a)aastra.net if (method=="INVITE") { if (uri=~"^sip:[a-zA-Z\.]@*") { log (1, "got mail format uri lookup for tel number\n"); exec_dset ("/usr/local/etc/ser/siplookup"); }; rewritehostport("192.168.1.109"); replace ("<sip:user1@pdc.local.aastra.net>", "<sip:user1@192.168.1.108>"); t_relay_to_udp("192.168.1.109","5060"); }; t_on_reply("1"); 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("sip.pdc.local.aastra.net", "subscriber")) { www_challenge("sip.pdc.local.aastra.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(); }; } onreply_route[1] { log (1, "got msg from Nexpan on reply\n"); } __________________________________________________________________ Découvrez le nouveau Yahoo! Mail : 250 Mo d'espace de stockage pour vos mails ! Créez votre Yahoo! Mail sur http://fr.mail.yahoo.com/

2 1

[Serusers] Redundant MySQL Bank
by Felipe Martins 19 Apr '05

19 Apr '05

Hi everybody, In order to increase the security, performance, reliability and scalability of my VoIP Service I'm putting SER and Mysql at two separate Servers. I'm a kind of worried about my mysql server, because, if its comes to stop one day, none of my Users will connect. In order to solve this problem, I'm thinking of doing a Redundant MySQL DB Server. I'm not sure how am I going to do that. Has anyone done it already, that could share some information ? I'm trying to make all my 3 Sip Server use my Local Mysql DB Server Machine to auth, but another Mysql Server Machine to do the redudance. Any help will be great Best Regards Thanks in Advance -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

3 2

[Serusers] Accounting forwarded calls with Setflag
by Martin Kjeldsen 19 Apr '05

19 Apr '05

Hi I was wondering if anybody had some advice on howto account forwarded calls to the subscriber who farwarded his phone? ATM I account people by the From address, and I was thinking of adding a control prior to the routes in ser.cfg, which serched the message for any diversion header: if (method=="INVITE" & search("Diversion:")) { log(1, "Call is being forwarded and we need to account it to this person\n"); append_hf("P-hint: Call being forwarded from\r\n"); setflag(1); }; But I dont seem to be getting the information, that I seek, cause no 302 status gets recorded in my accouonting table. Any help on what I'm doing wrong? And is there a better way to use setflag, or more correct? Regards Martin

2 3

[Serusers] Sorry, newbie. Please help with serweb...
by Alexandros Hatziandreou 19 Apr '05

19 Apr '05

Hi! After reading through the instructions, I have finally managed to make ser work. After that, I also followed the instructions and mysql seems to be working fine as well. (at least I can see the users I added so that should be a good sign!) Now I am trying to configure serweb so that others can connect to my server and add themselves etc but try as I may, I can not get it to work... I have made all the changes to the php files as suggested by the manual but when I try to go to the webserver's address all I get is : Database error: pconnect(localhost,administrator(a)oug.gr, $Password) failed. MYSQL Error: () Session halted. What is this? Can someone please help? Thanks you in advance, Alexandros.

1 0

RE: [Serusers] Parallel Forking Query
by Aisling O'Driscoll 19 Apr '05

19 Apr '05

Hi Juha, Thanks for the reply. I undertsand now. However if I want to implement sequential forking i.e. by adding a failure route etc, won't this do sequential forking for every user?....Is there a way of making this user profile specific? Many Thanks, Aisling. > >---- Original Message ---- >From: jh(a)tutpro.com >To: ashling.odriscoll(a)cit.ie >Subject: RE: [Serusers] Parallel Forking Query >Date: Mon, 18 Apr 2005 18:37:42 +0300 > >>Aisling O'Driscoll writes: >> >> > I want to set up SER such that some users have their imcoming >calls >> > forked to all their devices and others have their incoming calls >> > processed as normal....Is this possible?? >> >>if a user has several contacts registered for an aor with the same q >>value, then it is "normal" for a proxy to parallel fork to all those >>contacts. >> >>-- juha >> >>-------------------Legal >Disclaimer--------------------------------------- >> >>The above electronic mail transmission is confidential and intended >only for the person to whom it is addressed. Its contents may be >protected by legal and/or professional privilege. Should it be >received by you in error please contact the sender at the above >quoted email address. Any unauthorised form of reproduction of this >message is strictly prohibited. The Institute does not guarantee the >security of any information electronically transmitted and is not >liable if the information contained in this communication is not a >proper and complete record of the message as transmitted by the >sender nor for any delay in its receipt. >> >> -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

2 1

[Serusers] inside calls and PSTN calls
by Llanos Serna García-Conde 19 Apr '05

19 Apr '05

1 0

[Serusers] Hi All. a newbie question about SER + auth_radius:
by itsik 19 Apr '05

19 Apr '05

Thank u Daniel for the reply. I'm using ser 0.8.14. I have already installed radiusclient-ng-0.5.0. I forgot to mention I also saw the advice about adding /usr/local/lib to the LD_LIBRARY_PATH environment variable. Is there a way for me to check if the installation of the libradiusclient / radiusclient-ng went ok? Thx in advance itsik

1 0

[Serusers] Hi All. a newbie question about SER + auth_radius: loadmodule fails
by itsik 19 Apr '05

19 Apr '05

I'm trying to enable radius auth, but the loadmodule fails. I installed the libradiusclient 0.3.2 and I c libradiusclient.so in /usr/local/lib. I saw the advice about adding /usr/local/lib to the /etc/ld.so.conf and running ldconfig -v. I even saw that it handles the libradiusclient.so. Then I tried to fool auth_radius by adding a link to libradiusclient.so named libradiusclient.so.2, but then it didn't find some symbol rc_dict.. The configuration file is the default file, except the loadlodule for auth_radius and replacing www_auth with radius_auth. The ser output is: 0(14859) loading module /usr/local/lib/ser/modules/sl.so 0(14859) loading module /usr/local/lib/ser/modules/tm.so 0(14859) loading module /usr/local/lib/ser/modules/rr.so 0(14859) loading module /usr/local/lib/ser/modules/maxfwd.so 0(14859) loading module /usr/local/lib/ser/modules/usrloc.so 0(14859) loading module /usr/local/lib/ser/modules/registrar.so 0(14859) loading module /usr/local/lib/ser/modules/auth.so 0(14859) loading module /usr/local/lib/ser/modules/auth_radius.so 0(14859) ERROR: load_module: could not open module </usr/local/lib/ser/modules/auth_radius.so>: libradiusclient.so.2: cannot open shared object file: No such file or directory 0(14859) parse error (38,13-55): failed to load module 0(14859) set_mod_param_regex: usrloc matches module usrloc 0(14859) set_mod_param_regex: found <db_mode> in module usrloc [/usr/local/lib/ser/modules/usrloc.so] 0(14859) set_mod_param_regex: rr matches module rr 0(14859) set_mod_param_regex: found <enable_full_lr> in module rr [/usr/local/lib/ser/modules/rr.so] 0(14859) find_export: found <mf_process_maxfwd_header> in module maxfwd_module [/usr/local/lib/ser/modules/maxfwd.so] 0(14859) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(14859) find_export: found <sl_send_reply> in module sl_module [/usr/local/lib/ser/modules/sl.so] 0(14859) find_export: found <record_route> in module rr [/usr/local/lib/ser/modules/rr.so] 0(14859) find_export: found <loose_route> in module rr [/usr/local/lib/ser/modules/rr.so] 0(14859) find_export: found <t_relay> in module tm [/usr/local/lib/ser/modules/tm.so] 0(14859) find_export: <radius_www_authorize> not found 0(14859) find_export: <radius_www_authorize> not found 0(14859) parse error (105,32-33): unknown command, missing loadmodule? . a. why does the auth_radius looking for 'libradiusclient.so.2' doesn't it seeks the 'libradiusclient.so'? b. what can I do in order to make it work? Thx in advance, Itsik

2 1

[Serusers] Serweb latest version
by Cameron Beattie 19 Apr '05

19 Apr '05

A very elementary question. How do I get a version of serweb that is compatible with CVS head version of SER? I can't see it on the ftp site and I'm a cvs bunny so it'll be something like this... cd /usr/srcexport CVSROOT=:pserver:anonymous@cvs.berlios.de:/cvsroot/serwebcvs logincvs co -r ummmdunno serwebAny help would be appreciated.RegardsCameron

2 2

[Serusers] Ser on Gentoo
by Clay Bryan 19 Apr '05

19 Apr '05

I do see it in the portage tree but what about the ser-mysql integration and all the other add-ons. I don't see them in the portage tree. Do I need them or does ser come with them already included in the package that comes from the portage tree distribution. Wercs Communications Clay Bryan Network Administrator Wercs Communications 400 East First Casper, WY 82601 <http://maps.yahoo.com/py/maps.py?Pyt=Tmap&addr=400+East+First&csz=Caspe r%2C+WY++82601&country=us> CBryan(a)wercs.com <mailto:CBryan@wercs.com> tel: fax: mobile: 307-233-8359 307-233-8701 307-258-7371 Add me to your address book... <https://www.plaxo.com/add_me?u=21475170292&v0=646557&k0=282754785> Want a signature like this? <http://www.plaxo.com/signature>

3 2

[Serusers] MAC OS X
by Daniel Corbe 19 Apr '05

19 Apr '05

Has anyone gotten SER 0.9.0 to sucessfully compile and operate on MAC OX X?

2 1

[Serusers] CDR - via perl
by Kristian Knudsen 19 Apr '05

19 Apr '05

Hi all, I am trying to setup some cdr generation from a ser server (version 0.8.14) and I am using a perl script like this: foreach $var (sort(keys(%ENV))) { $val = $ENV{$var}; if ( ${var} =~ /^SIP_/){ print FP "${var}=\"${val}\"\n"; } } executed from ser.cfg, but I need the sip_response_code (2xx if ok). I can see that it is in the output from xlog, is it possible to fetch the value from any environment parameter at all? And how to do that. /kk

2 1

[Serusers] Bug in XLOG: ERROR on non-existant User-Agent header field
by Martin Koenig 19 Apr '05

19 Apr '05

Hello, the XLOG-Module prints an error if the User-Agend %ua Parameter is called on a request that does not contain a User-Agent header: XLOG: xl_get_useragent: ERROR cannot parse User-Agent header This is probably a bug, because this error is not showing up on a non-existant Contact-Header, nor on a non-existant To-Tag. Version is rel_0_9_0 cvs. With best regards, Martin Koenig

2 1

Re: [Serusers] Credentials with given realm not found
by Alex 19 Apr '05

19 Apr '05

So Daniel like i understand the problem is my radius configuration, another thing is that my ATA sending the same stuff, i mean if i will change the sip server to different one where i installed freeradius with ser it's working fine. Daniel where i can start to fix that problem.? Thank you very much for your time. On 4/14/05, Alex <alexandergav(a)gmail.com> wrote: > So Daniel like i understand the problem is my radius configuration, > another thing is that my ATA sending the same stuff, i mean if i will > change the sip server to different one where i installed freeradius > with ser it's working fine. > > Daniel where i can start to fix that problem.? > > Thank you very much for your time. > > > On 4/14/05, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: > > The second REGISTER (the block 3) must contains the response to the > > authentication challenge carried by 401 reply (block 2). That means that > > the block 3 must contain an Authorization header with authentication > > credentials computed according to HTTP-Digest authentication mechanism > > (RFC2617). Also, see the section 22.Usage of HTTP Authentication in SIP > > RFC3261 for more about user authentication in SIP. > > > > Daniel > > > > On 04/14/05 13:16, Alex wrote: > > > > >Sorry Daniel , i didn't get that, I send here 4 blocks, 1 one is the > > >register request the 2 is the reply from the server, 3 is the register > > >request, 4 is the reply from the server. If you can please point me to > > >the problem. Because like i see the 2 register requests (1,3 blocks) > > >are the same. > > > > > > > > >On 4/14/05, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: > > > > > > > > >>As you can see, the second REGISTER does not contain the authentication > > >>credentials (No Authorization header) in response to 401 reply. So, > > >>either you didn't configure the phone to authenticate or the Grandstream > > >>HT286 1.0.5.18 is faulty. > > >> > > >>Daniel > > >> > > >> > > >>On 04/14/05 12:35, Alex wrote: > > >> > > >> > > >> > > >>>Daniel thanks > > >>>btw it's clean installation of Red Hat Enterprise Linux AS release 3 > > >>>ser-08.14 , freeradius-1.2 , radiusclient-4.8 > > >>> > > >>>i am sending ngrep port 5060 > > >>>i have here 2 requests of register and the replies to register. > > >>> > > >>> > > >>>xxx.xxx.xxx.xxx - sipserverip > > >>>telephoneip - ip where the call coming from > > >>>Phonenumber - phone number > > >>> > > >>>-------------------------------------------------------------------------------------------------- > > >>> > > >>>U telephoneip:10739 -> xxx.xxx.xxx.xxx:5060 > > >>> REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP > > >>>telephoneip:10000;branch=z9hG4bK98514c3b052d7df6..From: "Test Alex" < > > >>> sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=50673f1baca1958c..To: > > >>><sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>..Contact: <sip > > >>> :Phonenumber@telephoneip:10000;user=phone>..Call-ID: > > >>>1cff1b8955b8fa5c@10.0.0.4..CSeq: 106 REGISTER..Expires: > > >>>3600..User-Agent > > >>> : Grandstream HT286 1.0.5.18..Max-Forwards: 70..Allow: > > >>>INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Lengt > > >>> h: 0.... > > >>># > > >>>U xxx.xxx.xxx.xxx:5060 -> telephoneip:10000 > > >>> SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP > > >>>telephoneip:10000;branch=z9hG4bK98514c3b052d7df6..From: "Test Alex" > > >>><sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=50673f1baca1958c..To: > > >>><sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=b27e1a1d33761e85846fc9 > > >>> 8f5f3a7e58.f894..Call-ID: 1cff1b8955b8fa5c@10.0.0.4..CSeq: 106 > > >>>REGISTER..WWW-Authenticate: Digest realm="xxx.xxx.xxx.xxx", nonc > > >>> e="425e3ac34dc9509392435c11fb260f41420049c7"..Server: Sip EXpress > > >>>router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 > > >>> xxx.xxx.xxx.xxx:5060 "Noisy feedback tells: pid=1912 > > >>>req_src_ip=telephoneip req_src_port=10739 in_uri=sip:xxx.xxx.xxx.xxx > > >>> out_uri=sip:xxx.xxx.xxx.xxx via_cnt==1".... > > >>># > > >>> > > >>>U telephoneip:10740 -> xxx.xxx.xxx.xxx:5060 > > >>> REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP > > >>>telephoneip:10000;branch=z9hG4bK98514c3b052d7df6..From: "Test Alex" < > > >>> sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=50673f1baca1958c..To: > > >>><sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>..Contact: <sip > > >>> :Phonenumber@telephoneip:10000;user=phone>..Call-ID: > > >>>1cff1b8955b8fa5c@10.0.0.4..CSeq: 106 REGISTER..Expires: > > >>>3600..User-Agent > > >>> : Grandstream HT286 1.0.5.18..Max-Forwards: 70..Allow: > > >>>INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Lengt > > >>> h: 0.... > > >>># > > >>>U xxx.xxx.xxx.xxx:5060 -> telephoneip:10000 > > >>> SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP > > >>>telephoneip:10000;branch=z9hG4bK98514c3b052d7df6..From: "Test Alex" > > >>><sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=50673f1baca1958c..To: > > >>><sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=b27e1a1d33761e85846fc9 > > >>> 8f5f3a7e58.f894..Call-ID: 1cff1b8955b8fa5c@10.0.0.4..CSeq: 106 > > >>>REGISTER..WWW-Authenticate: Digest realm="xxx.xxx.xxx.xxx", nonc > > >>> e="425e3acb812b5b2e8aa023e3fcffc618dc4cf661"..Server: Sip EXpress > > >>>router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 > > >>> xxx.xxx.xxx.xxx:5060 "Noisy feedback tells: pid=1885 > > >>>req_src_ip=telephoneip req_src_port=10740 in_uri=sip:xxx.xxx.xxx.xxx > > >>> out_uri=sip:xxx.xxx.xxx.xxx via_cnt==1".... > > >>># > > >>> > > >>> > > >>>tell me if you need something else. > > >>> > > >>> > > >>>On 4/14/05, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: > > >>> > > >>> > > >>> > > >>> > > >>>>Could you post the network dump with REGISTER/401/REGISTER messages? I > > >>>>will take a look to see if something is wrong. > > >>>> > > >>>> > > >>>>On 04/14/05 12:16, Alex wrote: > > >>>> > > >>>> > > >>>> > > >>>> > > >>>> > > >>>>>Digest realm is the same for the register requests. > > >>>>>furthermore the realm in To tag is correct. > > >>>>> > > >>>>> > > >>>>> > > >>>>> > > >>>>> > > >>>>> > > >>>>Did you mean To URI instead of To tag? > > >>>> > > >>>>Daniel > > >>>> > > >>>> > > >>>> > > >>>> > > >>>> > > >>>>>what else it can be. > > >>>>>Thanks for any help. > > >>>>> > > >>>>>On 4/14/05, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: > > >>>>> > > >>>>> > > >>>>> > > >>>>> > > >>>>> > > >>>>> > > >>>>>>Watch the network traffic (ngrep or ethereal on port 5060) and check the > > >>>>>>realm from 401 is the same as the one from next REGISTER. Also, when > > >>>>>>you use the empty realm parameter to radius_ww_authorize() and > > >>>>>>www_challenge(), the realm is taken from To URI. > > >>>>>> > > >>>>>>Daniel > > >>>>>> > > >>>>>> > > >>>>>>On 04/14/05 08:08, Alex wrote: > > >>>>>> > > >>>>>> > > >>>>>> > > >>>>>> > > >>>>>> > > >>>>>> > > >>>>>> > > >>>>>>>Hi guys maybe someone can find the problem, i still can't see anything > > >>>>>>>going to radius authentication. (the radius logs are empty) > > >>>>>>> > > >>>>>>>the register request is coming but it's not going to authenticate > > >>>>>>>through the radius. > > >>>>>>>Any help will be appreciated. > > >>>>>>> > > >>>>>>>here is the debug from ser : > > >>>>>>>--------------------------------------------------------------------------------------------- > > >>>>>>>14(1036) parse_headers: flags=-1 > > >>>>>>>14(1036) check_via_address(62.219.158.191, 62.219.158.191, 1) > > >>>>>>>14(1036) DEBUG:destroy_avp_list: destroing list (nil) > > >>>>>>>14(1036) receive_msg: cleaning up > > >>>>>>>9(1012) SIP Request: > > >>>>>>>9(1012) method: <REGISTER> > > >>>>>>>9(1012) uri: <sip:xxx.xxx.xxx.xxx> > > >>>>>>>9(1012) version: <SIP/2.0> > > >>>>>>>9(1012) parse_headers: flags=1 > > >>>>>>>9(1012) Found param type 232, <branch> = <z9hG4bKfc5751413c832e6d>; state=16 > > >>>>>>>9(1012) end of header reached, state=5 > > >>>>>>>9(1012) parse_headers: Via found, flags=1 > > >>>>>>>9(1012) parse_headers: this is the first via > > >>>>>>>9(1012) After parse_msg... > > >>>>>>>9(1012) preparing to run routing scripts... > > >>>>>>>9(1012) REGISTER: Authenticating user > > >>>>>>>9(1012) parse_headers: flags=4 > > >>>>>>>9(1012) end of header reached, state=9 > > >>>>>>>9(1012) DEBUG: get_hdr_field: <To> [45]; > > >>>>>>>uri=[sip:phonenumber@xxx.xxx.xxx.xxx;user=phone] > > >>>>>>>9(1012) DEBUG: to body [<sip:phonenumber@xxx.xxx.xxx.xxx;user=phone> > > >>>>>>>] > > >>>>>>> > > >>>>>>>9(1012) parse_headers: flags=4096 > > >>>>>>>9(1012) get_hdr_field: cseq <CSeq>: <103> <REGISTER> > > >>>>>>>9(1012) DEBUG: get_hdr_body : content_length=0 > > >>>>>>>9(1012) found end of header > > >>>>>>>9(1012) pre_auth(): Credentials with given realm not found > > >>>>>>>9(1012) REGISTER: challenging user > > >>>>>>>9(1012) build_auth_hf(): 'WWW-Authenticate: Digest > > >>>>>>>realm="xxx.xxx.xxx.xxx", > > >>>>>>>nonce="425e063022afc1142ed6730d46da41692ff3ed57" > > >>>>>>> > > >>>>>>>_______________________________________________ > > >>>>>>>Serusers mailing list > > >>>>>>>serusers(a)lists.iptel.org > > >>>>>>>http://lists.iptel.org/mailman/listinfo/serusers > > >>>>>>> > > >>>>>>> > > >>>>>>> > > >>>>>>> > > >>>>>>> > > >>>>>>> > > >>>>>>> > > >>>>>>> > > >>>>>>> > > >>>>> > > >>>>> > > >>>>> > > >>> > > >>> > > >>> > > > > > > > > > > > >

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[Serusers] NAT and ser
by Serhan Sevim 19 Apr '05

19 Apr '05

Hello there, we use ser to forward incoming call requests to a remote PSTN. this system works fine if the client has a non-natted public IP address.However, I can't tell the same thing for the natted clients and it doesn't work so far. using nathelper/rtpproxy, the client (x-lite) can make the connection and ring the phone but picking up the phone on the other end.. hmm, well, no audio can be heard. As a matter of fact, x-lite client times out with error 408 "after" I hear the remote phone rings. it seems like the client side simply signaling for connection but no data(audio) transfer can be made. below is my ser.cfg, does anybody have an idea? Thanks in advance. Serhan. # ----------- global configuration parameters ------------------------ #debug=7 # debug level (cmd line: -dddddddddd) #fork=yes #log_stderror=yes # (cmd line: -E) # Uncomment these lines to enter debugging mode #debug=7 #fork=yes #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 ---------------------------------- loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.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/xlog.so" #Database URL is here! modparam("usrloc", "db_url", "sql://ser:pass123@localhost/ser") #nathelper params. modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 seconds modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT #mysql loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- #modparam("usrloc", "db_mode", 0) modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic alias="domain.com" alias="sipgw.domain.com" 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; }; # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { log(1, "REGISTER received\n"); } else { log(1, "non-REGISTER received\n"); }; if (uri=~"sip:.*[@:]sipgw.domain.com") { log(1, "request for sipgw.domain.com received\n"); } else { log(1, "request for other domain received\n"); }; 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 entitie # use different transport protocol record_route(); # loose-route processing if (loose_route()) { t_relay(); break; }; # PSTN Forwarding if (uri=~"^sip:1") { strip(1); # forward(pstngw.jondoe.com,5060); force_rtp_proxy(); t_relay_to_udp("pstngw.jondoe.com", "5060"); 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") { save("location"); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "not exists!"); 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(); }; }

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[Serusers] anybody using Teliann
by Mohammad Khan 19 Apr '05

19 Apr '05

Hello list: Did you guys try Teliann 3100 series phone? I am having some issue with domain name, auto answer, proxy_authorization and stun. If you anybody tried this phone, please share your experience. I will really appreciate. Thanks, Mohammad Khan

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[Serusers] migration 0.8.14 to 0.9.0 problems with mysql
by Paul van Schagen 18 Apr '05

18 Apr '05

Hi, I am migrating from 0.8.14 to 0.9.0 can anyone tell me how to compile in mysql for ser 0.9.0 first I found that mysql.so was not compiled in standard then I modified the Makefile and included mysql , nevertheless ser displays an error. Apr 18 20:04:20 /usr/local/sbin/ser[4734]: Maxfwd module- initializing Apr 18 20:04:20 /usr/local/sbin/ser[4734]: bind_dbmod: Module sql does not export db_use_table function Apr 18 20:04:20 /usr/local/sbin/ser[4734]: ERROR: mod_init(): Can't bind database module Apr 18 20:04:20 /usr/local/sbin/ser[4734]: init_mod(): Error while initializing module usrloc thanks Paul van Schagen

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[Serusers] How to limit the maxinum registrations and current calls?
by Charles Wang 18 Apr '05

18 Apr '05

Hi, ALL: How to limit the maxinum register of UACs and the maxinum current calls of UACs? I have no idea to limit these from ser.cfg. Does anybody know how to limit them? -- Best Regards Charles

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[Serusers] What exactly does "lookup" lookup?
by Brian McCrary 18 Apr '05

18 Apr '05

Hello, I hope I'm not asking a really stupid question here, but here goes... I'm setting up a new SER installation and have it running pretty smoothly now for authenticated outbound traffic. The problem comes when trying to route incoming traffic through. My routing plans are pretty simple, when a number comes into SER, see if it it's a registered SIP phone, if so send the call to it, and if not, then authenticate the user, and send it to the PSTN gateway. I thought this could easily be solved by having a lookup("location") as the first part of my routing. However, I'm not really sure what lookup looks up. I assume it's the user being dialed, such as if a call comes in wanting sip:1234567890@x.x.x.x, lookup would look into the database and see user "1234567890" matches a registered user, and go from there. However, using lookup never returns a match, thus the call goes into a loop since it's next stop is the PSTN gateway. I can see the phone has registered by running serctl ul show. Here's part of my routing I tried using, I may be incorrect in using t_relay, but right now lookup is never true: if (method=="INVITE") { if (lookup("location")) { log("LOG: SIP Phone found\n"); t_relay(); break; } else if (uri=~"sip:[0-9]+@.*") { if (!is_user_in("credentials", "local")) { sl_send_reply("403", "Forbidden"); break; }; rewritehostport("10.0.0.4:5060"); # PSTN GW Brian

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[Serusers] Installing Ser on Gentoo
by Clay Bryan 18 Apr '05

18 Apr '05

Hi, I am a newbie at this. Has anyone had any luck installing ser on Gentoo linux? Wercs Communications Clay Bryan Network Administrator Wercs Communications 400 East First Casper, WY 82601 <http://maps.yahoo.com/py/maps.py?Pyt=Tmap&addr=400+East+First&csz=Caspe r%2C+WY++82601&country=us> CBryan(a)wercs.com <mailto:CBryan@wercs.com> tel: fax: mobile: 307-233-8359 307-233-8701 307-258-7371 Add me to your address book... <https://www.plaxo.com/add_me?u=21475170292&v0=646557&k0=282754785> Want a signature like this? <http://www.plaxo.com/signature>

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RE: [Serusers] e164 rewrite script
by Matt Schulte 18 Apr '05

18 Apr '05

Thanks, that's a step in the right direction. I wasn't refering to 5551212 as direc assist (bad example on my part) though. I was just wondering for to prefix the +1636 to 7 digits. In subst_uri I don't think it can read avp variables, that's my problem since the ruri will always be different. I was doing this to elminate all external scripts. Thanks again. -----Original Message----- From: Java Rockx [mailto:javarockx@gmail.com] Sent: Monday, April 18, 2005 9:39 AM To: Matt Schulte Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] e164 rewrite script Matt, This is what I use to dial local directory assistance. When a user dials 411 we rewrite the R-URI. So if use account 4073191122 dials 411 the R-URI gets massaged to 14075551212 and then sent out the PSTN gateway. Regards, Paul avp_write("$from", "i:34"); avp_pushto("$ruri", "i:34"); strip_tail(7); subst_uri('/^sip:([0-9]+)@(.*)$/sip:\15551212@\2/i'); avp_delete("i:34"); On 4/18/05, Matt Schulte <mschulte(a)netlogic.net> wrote: Ok this is all fairly easy except for 7 digit dialing (usa), the problem I'm having is how do I get the area code added to the 7 digits. I was going to use the user@ to produce this but am unsure how, eg: from: +16365551212 dialed: 5551111 should look like dialed: +16365551111 Thoughts? -----Original Message----- From: Matt Schulte Sent: Friday, April 15, 2005 1:44 PM To: serusers(a)lists.iptel.org Subject: [Serusers] e164 rewrite script Does anyone have a config to convert phone numbers (user@) to e164 numbers? Mainly american numbers.. Would save me a bit of time, else I can try to write something out.. Thanks. _______________________________________________ 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] Parallel Forking Query
by Aisling O'Driscoll 18 Apr '05

18 Apr '05

Hi, I am just wondering what is the position with forking a call i.e. is it the case that all calls must be parallel forked or else none? I want to set up SER such that some users have their imcoming calls forked to all their devices and others have their incoming calls processed as normal....Is this possible?? Thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

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[Serusers] New to SER, got some questions
by Daniel Corbe 18 Apr '05

18 Apr '05

Hello, I have a problem as well as a couple of questions which don't seem to be covered in the documentation. 1) I successfully got an installation of SER running with a MySQL back-end and Digest Authentication. I've added a user with the following command: /usr/local/sbin/serctl add 2000 pl00p4f daniel.junkmail(a)gmail.com Ultimatly, 2000 is the username (phone number of the SIP UA) and pl00p4f is the password. I've verified that this user is in the database: mysql> select * from subscriber where username like '2000'\G *************************** 1. row *************************** phplib_id: 8ba7accc847208eb10eb6baa6373f89d username: 2000 domain: eglobalphone.com password: pl00p4f first_name: last_name: phone: email_address: dcorbe(a)gmail.com datetime_created: 2005-04-16 19:47:09 datetime_modified: 0000-00-00 00:00:00 confirmation: flag: o sendnotification: greeting: ha1: 8ba7accc847208eb10eb6baa6373f89d ha1b: 711817121ed6cae7e44837e3fddf78d4 perms: NULL allow_find: 0 timezone: NULL rpid: NULL 1 row in set (0.14 sec) However; I cannot log into my proxy server. I keep getting 401 Unauthorized back. Here's a libpcap capture of my session. Also attached is my ser config file. I'm not sure what to do or what 2) And speaking of the SER config file, I have a question about the userloc module What is the difference between www_authorize/www_challenge and proxy_authorize/proxy_challenge? Thank you for your help. -Daniel

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[Serusers] *bangs head* Struggling with simple SER+SEMS..
by Dave Bath 18 Apr '05

18 Apr '05

Hey guys.... I'm really struggling with this. It was working perfectly yesterday and now I must have made a small change and I just cannot find what I've done wrong. Bascially, this is a very very simple unauthenticated SER front end to SEMS... all I want to be able to do is use the conference facilities. Incidentally, when I say unauthenticated I mean also without registering... so all a user has to do is send an INVITE. To that end, I am trying to just redirect sip URI's beginning with 600. It seems that the initial redirect goes through... but then something screwy is happening because SER never appears to receive the ACKs correctly.. and it starts sending two sets of 200 messages.. one with no SDP descriptor and then a second batch with a valid SDP descriptor. Because of this weirdness with the ACKs, the client and SER fight for a while sending out millions of repeat 200s and ACKs. Meanwhile, because the initial redirect to SEMS worked, the conference has actually been created... The same story with BYE.. the client sends it, but SER doesn't even appear to receive it (note all the debugging log lines). In fact, there' something rather dodgy up, because I don't even get the debug lines inside the route[1] block.... It sort of points to me that perhaps I've done something very stupid with the transactions.. that for some reason they don't match up ... but I can't find it. Attached is my ser.cfg. Any help would be so very much appreciated. I don't think it's a basic connectivity problem, because as I said, a very similar file was working perfectly yesterday... I've just stupidly changed it slightly somewhere. Many many thanks in advance. Dave listen=<publicip> check_via=yes # (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 "/lib/ser/modules/mysql.so" loadmodule "/lib/ser/modules/sl.so" loadmodule "/lib/ser/modules/tm.so" loadmodule "/lib/ser/modules/rr.so" loadmodule "/lib/ser/modules/maxfwd.so" loadmodule "/lib/ser/modules/usrloc.so" loadmodule "/lib/ser/modules/registrar.so" loadmodule "/lib/ser/modules/textops.so" # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- 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 (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; if (!method=="REGISTER") record_route(); if (loose_route()) { log(1,"Loose routing block"); append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { log(1,"Got a request for an extenal domain"); # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; if (uri==myself) { # switch to stateful mode: if (!t_newtran()){ sl_send_reply("500","could not create transaction"); break; }; if (method=="REGISTER") { save("location"); break; }; if (method=="ACK") { # absorb ACKs log(1,"Absorbing ACK"); break; }; lookup("aliases"); if (!uri==myself) { log(1,"Resolving external request"); append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # prevent timeout on the other side: t_reply("100","Trying - just wait a minute !"); if (method=="INVITE") { save("location"); if (uri=~"sip:600.*@") { log(1,"-> Redirecting conference to SEMS"); if(!t_write_unix("/tmp/am_sock","conference")) { t_reply("500","error contacting sems"); }; break; } ; break; } else if (method=="BYE" || method=="CANCEL") { log(1,"-> Sending BYE to SEMS"); if(!t_write_unix("/tmp/am_sock","bye")) { t_reply("500","error contacting sems"); }; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP log(1, "Attempting stateful forward..."); if (!t_relay()) { log(1,"... failure"); sl_reply_error(); }; log(1, "... success"); } _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Serusers] Fwd: AS5300/E1 + SER -> PB : Cisco redirect calls on SER
by Alcée 18 Apr '05

18 Apr '05

> > Hello, > I try to use cisco AS5300 with SER, like this : > phones -> SER -> AS5300 -> PSTN > or > phones -> AS5300 -> PSTN > | > \/ > SER > But the second form does not work. > > > I don't want people to directly ask for AS5300. So I follow the > "Howto" on SER web page. > But nothing arrive on SER when I make a call directly to AS5300, The > call is forward to pstn. > So I make this configuration to minimize the problem. > But nothing at all. > The "debug ccsip error" tells "no route to destination". > > Have you any idea ? > > thanks :) > > Alcée > > ! Last configuration change at 14:17:05 UTC Mon Apr 11 2005 by admin > ! NVRAM config last updated at 12:57:01 UTC Mon Apr 11 2005 by admin > ! > version 12.3 > service timestamps debug uptime > service timestamps log uptime > no service password-encryption > ! > hostname VoIP > ! > boot-start-marker > boot-end-marker > ! > ! > ! > resource-pool disable > ! > aaa new-model > ! > ! > aaa authentication login h323 group radius > aaa authorization exec h323 group radius > aaa accounting delay-start > aaa accounting connection h323 start-stop group radius > aaa nas port voip > aaa session-id common > ip subnet-zero > ! > ! > isdn switch-type primary-net5 > ! > voice rtp send-recv > voice echo-canceller extended > ! > voice class codec 1 > codec preference 1 g729r8 bytes 40 > codec preference 2 g711ulaw > ! > voice class codec 2 > codec preference 1 g723ar63 bytes 48 > codec preference 2 g723r63 bytes 48 > codec preference 3 g729br8 bytes 40 > codec preference 4 g729r8 bytes 40 > codec preference 5 g723ar53 bytes 40 > codec preference 6 g723r53 bytes 40 > codec preference 7 g711ulaw > ! > fax interface-type modem > ! > ! > controller E1 0 > clock source line primary > pri-group timeslots 1-31 > ! > controller E1 1 > pri-group timeslots 1-31 > ! > controller E1 2 > clock source line secondary 4 > pri-group timeslots 1-31 > ! > controller E1 3 > pri-group timeslots 1-31 > gw-accounting syslog > gw-accounting aaa > attribute h323-remote-id resolved > acct-template callhistory-detail > suppress pots > ! > ! > ! > interface Ethernet0 > ip address 192.168.0.129 255.255.255.0 > ! > interface Serial0:15 > no ip address > isdn switch-type primary-net5 > isdn protocol-emulate network > no cdp enable > ! > interface Serial1:15 > no ip address > isdn switch-type primary-net5 > no cdp enable > ! > interface Serial2:15 > no ip address > isdn switch-type primary-net5 > no cdp enable > ! > interface Serial3:15 > no ip address > isdn switch-type primary-net5 > no cdp enable > ! > interface FastEthernet0 > ip address 172.19.20.2 255.255.255.0 > duplex full > speed 100 > ! > interface Dialer0 > no ip address > ! > ip classless > ip route 0.0.0.0 0.0.0.0 172.19.20.1 > no ip http server > ! > ! > logging facility local0 > logging 192.168.0.2 > ! > ! > radius-server host 192.168.0.2 auth-port 1812 acct-port 1813 > radius-server key test > radius-server vsa send accounting > radius-server vsa send authentication > ! > voice-port 0:D > ! > voice-port 1:D > ! > voice-port 2:D > ! > voice-port 3:D > ! > ! > ! > dial-peer voice 3777 voip > huntstop > preference 2 > application session > destination-pattern 3777#T > progress_ind setup enable 3 > progress_ind alert enable 8 > voice-class codec 2 > session protocol sipv2 > session target sip-server > dtmf-relay cisco-rtp h245-signal h245-alphanumeric > fax rate 9600 > icpif 50 > ! > ! > sip-ua > retry invite 3 > retry response 3 > retry bye 3 > retry cancel 3 > timers expires 300000 > timers connect 300 > sip-server ipv4:192.168.0.1 > ! > ! > line con 0 > line aux 0 > line vty 0 4 > password intermobile > ! > ntp server 192.168.0.2 > end

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RE: [Serusers] e164 rewrite script
by Matt Schulte 18 Apr '05

18 Apr '05

Ok this is all fairly easy except for 7 digit dialing (usa), the problem I'm having is how do I get the area code added to the 7 digits. I was going to use the user@ to produce this but am unsure how, eg: from: +16365551212 dialed: 5551111 should look like dialed: +16365551111 Thoughts? -----Original Message----- From: Matt Schulte Sent: Friday, April 15, 2005 1:44 PM To: serusers(a)lists.iptel.org Subject: [Serusers] e164 rewrite script Does anyone have a config to convert phone numbers (user@) to e164 numbers? Mainly american numbers.. Would save me a bit of time, else I can try to write something out.. Thanks. _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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Re: [Serusers] ser-0.9.0 and PA clients
by Didier Hung Wan Luk 18 Apr '05

18 Apr '05

Hi, I'm using eyebeam and xten lite with a Fedora core 3 and ser-0.9.1 from csv... The ser server in main office serving branch offices connected with frame relay 64kb. It's working fine (not yet tested conference) and presence is working only when I instruct eyebeam to use peer-to-peer presence. When I try to use Presence Agent mode presence does not work, can someone help me...? My ser.cfg # # $Id: ser.cfg,v 1.25 2004/11/30 16:28:24 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 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 alias=192.168.130.209 listen=192.168.130.209 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ 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/pa.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/textops.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) modparam("usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("usrloc", "use_domain", 1) modparam("usrloc", "desc_time_order", 1) # -- 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) # -- pa params modparam("pa", "db_url", "mysql:ser:heslo@localhost/ser") modparam("pa", "default_expires",50) modparam("pa", "pa_domain", "192.168.130.209") # -- registrar params -- modparam("registrar", "use_domain", 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 (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) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); 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("192.168.130.209", "subscriber")) { www_challenge("192.168.130.209", "0"); break; }; save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } DidierH. > -----Original Message----- > From: Jamey Hicks [mailto:jamey.hicks@hp.com] > Sent: Tue 1/11/2005 11:41 PM > To: Helge Waastad > Cc: serusers(a)lists.iptel.org > Subject: Re: [Serusers] ser-0.9.0 and PA clients Helge Waastad wrote: >> Hello, >> I've started to use the 0.9.0 release and to me it seems to work as a >> charm :-) >> >> However does anybody know which SIP clients which is compliant for the PA? >> I'm using kphone and it works fine, but I would like to try out some >> for use on Microsoft? >> Messenger 5.0 does not work. > Eyebeam from xten.com seems to be the farthest along in terms of > implementing SIMPLE presence support, according to their website. I > know Xten was testing against the SER PA this fall. That being said, > testing Eyebeam against SER 0.9 is on my todo list for this week. I > just need to set up a SER 0.9 PA on the same side of the HP firewall as > my Windows box. >> Jamey This mail has been scanned for known virus like activity on behalf of GALANA DISTRIBUTION PETROLIERE S.A. This E-mail is privileged, confidential and intended solely for the addressee(s).Any unauthorised dissemination or copying of this E-mail, and any use or disclosure of any information contained in it, is strictly prohibited and may be illegal. If you have received this E-mail in error please notify the sender immediately and delete it from your system. GALANA DISTRIBUTION PETROLIERE S.A. does not accept any liability for any errors, omissions, interceptions, corrupted mail, lost communications or late delivery which may arise as a result of this E-mail or for any virus that may be contained in it. Any views or opinions expressed in this E-mail are solely those of the author and do not necessarily reflect those of GALANA DISTRIBUTION PETROLIERE S.A. Ce courriel a fait l'objet, par GALANA DISTRIBUTION PETROLIERE S.A. de verifications le protegeant des virus connus. Ce courriel est personnel, confidentiel et adresse exclusivement a son ou ses destinataire(s). Toute diffusion ou copie non autorisee ainsi que l'usage partiel ou total de son contenu est strictement interdit et peut-etre illegal. Si vous avez recu ce courriel par erreur, merci d'en informer son emetteur et de l'effacer de votre ordinateur. GALANA DISTRIBUTION PETROLIERE S.A. refuse toutes responsabilites liees aux erreurs, omissions, interceptions, deformations, communications perdues, ou retards de transmissions dont ce courriel pourrait etre l'objet ou a l'origine ainsi que des virus qu'il pourrait contenir. 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2 1

[Serusers] database and x86_64
by Lol Zimmerli 18 Apr '05

18 Apr '05

Hi, I'm trying to get SER running on a x86_64 server and 'postgres' module does not compile. It starts with that: make[1]: Entering directory `/usr/local/src/ser-0.9.0/modules/postgres' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -falign-loops -DNAME='"ser"' -DVERSION='"0.9.0"' -DARCH='"x86_64"' -DOS='"linux"' -DCOMPILER='"gcc 3.3"' -D__CPU_x86_64 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -I/usr/local/include -I/usr/local/pgsql/include -I/usr/include/postgresql -c aug_alloc.c -o aug_alloc.o In file included from aug_std.h:62, from aug_alloc.c:57: aug_sysdep.h:327: error: parse error before ':' token In file included from aug_std.h:68, from aug_alloc.c:57: and stops like this, after 250 lines of warning and errors: aug_alloc.c:370: warning: statement with no effect aug_alloc.c: At top level: /usr/include/gconv.h:176: warning: array `__data' assumed to have one element make[1]: *** [aug_alloc.o] Error 1 make[1]: Leaving directory `/usr/local/src/ser-0.9.0/modules/postgres' I'm using SER 0.9.0 and postgres 8.01 is installed in /usr/local/pgsql, compiled from sources. Any hint is welcome! Best regards -- # Lol Zimmerli // S y s C o ® // http://www.sysco.ch/ Follow each decision as closely as possible with its associated action. - The Elements of Programming Style (Kernighan & Plaugher)

1 0

[Serusers] Can't solve the problem. Need help.
by Alex 18 Apr '05

18 Apr '05

This is the answer i got on the list : The second REGISTER (the block 3) must contains the response to the authentication challenge carried by 401 reply (block 2). That means that the block 3 must contain an Authorization header with authentication credentials computed according to HTTP-Digest authentication mechanism (RFC2617). U sourceip:26012 -> xxx.xxx.xxx.xxx:5060 REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP sourceip:22115;branch=z9hG4bK4913f67fbbcfb571..From: "Alex " <sip:phonenumber@xxx.xxx.xxx.xxx>;tag=94a44507e03df901..To: <sip:phonenumber@xxx.xxx.xxx.xxx>..Contact: <sip:phonenumber@sourceip:22115>..Call-ID: c9f64c0c2ef27cd1@10.0.0.6..CSeq: 100 REGISTER..Expires: 3600..User-Agent: Grandstream HT286 1.0.5.11..Max-F orwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0.... # U xxx.xxx.xxx.xxx:5060 -> sourceip:22115 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP sourceip:22115;branch=z9hG4bK4913f67fbbcfb571..From: "Alex " <sip:phonenumber@xxx.xxx.xxx.xxx>; tag=94a44507e03df901..To: <sip:phonenumber@xxx.xxx.xxx.xxx>;tag=b27e1a1d33761e85846fc98f5f3a7e58.eb04..Call-I D: c9f64c0c2ef27cd1@10.0.0.6..CSeq: 100 REGISTER..WWW-Authenticate: Digest realm="xxx.xxx.xxx.xxx", nonce="42612dd595b3558cfdd4 46b83b081d1e2d3cc480"..Server: Sip EXpress router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 xxx.xxx.xxx.xxx:5060 " Noisy feedback tells: pid=21363 req_src_ip=sourceip req_src_port=26012 in_uri=sip:xxx.xxx.xxx.xxx out_uri=sip:xxx.xxx.xxx.xxx via_cnt==1".... # U sourceip:26012 -> xxx.xxx.xxx.xxx:5060 REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP sourceip:22115;branch=z9hG4bK4913f67fbbcfb571..From: "Alex " <sip:phonenumber@xxx.xxx.xxx.xxx>;tag=94a44507e03df901..To: <sip:phonenumber@xxx.xxx.xxx.xxx>..Contact: <sip:phonenumber@sourceip:22115>..Call-ID: c9f64c0c2ef27cd1@10.0.0.6..CSeq: 100 REGISTER..Expires: 3600..User-Agent: Grandstream HT286 1.0.5.11..Max-F orwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0.... # U xxx.xxx.xxx.xxx:5060 -> sourceip:22115 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP sourceip:22115;branch=z9hG4bK4913f67fbbcfb571..From: "Alex " <sip:phonenumber@xxx.xxx.xxx.xxx>;tag=94a44507e03df901..To: <sip:phonenumber@xxx.xxx.xxx.xxx>;tag=b27e1a1d33761e85846fc98f5f3a7e58.eb04..Call-I D: c9f64c0c2ef27cd1@10.0.0.6..CSeq: 100 REGISTER..WWW-Authenticate: Digest realm="xxx.xxx.xxx.xxx", nonce="42612dd63c17bd54baef 0f620d6b9b066c23c8ce"..Server: Sip EXpress router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 xxx.xxx.xxx.xxx:5060 " Noisy feedback tells: pid=21360 req_src_ip=sourceip req_src_port=26012 in_uri=sip:xxx.xxx.xxx.xxx out_uri=sip:xxx.xxx.xxx.xxx via_cnt==1".... here is my ser debug: ------------------------------------- 12(21360) SIP Request: 12(21360) method: <REGISTER> 12(21360) uri: <sip:xxx.xxx.xxx.xxx> 12(21360) version: <SIP/2.0> 12(21360) parse_headers: flags=1 12(21360) Found param type 232, <branch> = <z9hG4bKf776a5d04027adc6>; state=16 12(21360) end of header reached, state=5 12(21360) parse_headers: Via found, flags=1 12(21360) parse_headers: this is the first via 12(21360) After parse_msg... 12(21360) preparing to run routing scripts... 12(21360) REGISTER message received 12(21360) REGISTER: Authenticating user 12(21360) parse_headers: flags=4 12(21360) end of header reached, state=9 12(21360) DEBUG: get_hdr_field: <To> [34]; uri=[sip:phonenumber@xxx.xxx.xxx.xxx] 12(21360) DEBUG: to body [<sip:phonenumber@xxx.xxx.xxx.xxx> ] 12(21360) parse_headers: flags=4096 12(21360) get_hdr_field: cseq <CSeq>: <100> <REGISTER> 12(21360) DEBUG: get_hdr_body : content_length=0 12(21360) found end of header 12(21360) pre_auth(): Credentials with given realm not found 12(21360) REGISTER: challenging user 12(21360) build_auth_hf(): 'WWW-Authenticate: Digest realm="xxx.xxx.xxx.xxx", nonce="42613540bb4462c045f7f3fe7714c3a1d6c0bca9" ' 12(21360) parse_headers: flags=-1 12(21360) check_via_address(62.219.160.40, 62.219.160.40, 1) 12(21360) DEBUG:destroy_avp_list: destroing list (nil) 12(21360) receive_msg: cleaning up For some reason that i can not figure out i don't receive anything on the radius logs. it's looks like my granstream ATA286 not going through authectication process. when i change the ip of the sip server to different one (2-server) it's working perfect. The grandstream ATA286 sending same packets, on the 2 server it's working, on 1- one nothing happens. on the problematic server i have clean installation of freeradius-1.02 radiusclient-4.8 ser-8.14 I tested the installation of the freeradius with the : radclient -f digest localhost auth testing123 Received response ID 134, code 2, length = 45 Reply-Message = "Hello, test with digest" radius logs: ---------------------------------------------- rad_recv: Access-Request packet from host 127.0.0.1:32844, id=134, length=140 User-Name = "test" Digest-Response = "631d6d73147add2f9e437f59bbc3aeb7" Digest-Attributes = "\001\013testrealm" Digest-Attributes = "\002\n1234abcd" Digest-Attributes = "\003\010INVITE" Digest-Attributes = "\004\034sip:5555551212@example.com" Digest-Attributes = "\006\005MD5" Digest-Attributes = "\n\006test" Processing the authorize section of radiusd.conf modcall: entering group authorize for request 0 modcall[authorize]: module "preprocess" returns ok for request 0 modcall[authorize]: module "chap" returns noop for request 0 modcall[authorize]: module "mschap" returns noop for request 0 rlm_digest: Converting Digest-Attributes to something sane... Digest-Realm = "testrealm" Digest-Nonce = "1234abcd" Digest-Method = "INVITE" Digest-URI = "sip:5555551212@example.com" Digest-Algorithm = "MD5" Digest-User-Name = "test" rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 0 rlm_realm: No '@' in User-Name = "test", looking up realm NULL rlm_realm: No such realm "NULL" modcall[authorize]: module "suffix" returns noop for request 0 rlm_eap: No EAP-Message, not doing EAP modcall[authorize]: module "eap" returns noop for request 0 users: Matched entry DEFAULT at line 152 users: Matched entry test at line 215 modcall[authorize]: module "files" returns ok for request 0 modcall: group authorize returns ok for request 0 rad_check_password: Found Auth-Type Digest auth: type "digest" Processing the authenticate section of radiusd.conf modcall: entering group authenticate for request 0 A1 = test:testrealm:test A2 = INVITE:sip:5555551212@example.com KD = 1e00d6dbd30441265df6064b9d9b7da9:1234abcd:675b8c827b388805aa252ea38bfb6804 modcall[authenticate]: module "digest" returns ok for request 0 modcall: group authenticate returns ok for request 0 radius_xlat: 'Hello, test with digest' Sending Access-Accept of id 134 to 127.0.0.1:32844 Reply-Message = "Hello, test with digest" Finished request 0 I need some help to figure out that's the problem with this server. And what can be a reason what i don't see any authentication process in the radius , when the packets coming from ATA286 to authenticate the the user. Thanks for any help.

4 6

[Serusers] rtpproxy is not getting coneection with ser.
by Deepak Dhiman 18 Apr '05

18 Apr '05

hi friends ! i have configured my ser according to the given configuration on the site www.voip-info.org/wiki+rtpproxy. but after running rtpproxy successfully my ser is not getting start. it is also not showing the generic socket for communication between ser and rtpproxy, the socket /var/run/rtpproxy.sock doesn`t exist not even at any other location, can anybody help me, why is it happening and suggest me something. thanks Deepak Dhiman

2 2

[Serusers] want help regarding rtpproxy
by Deepak Dhiman 18 Apr '05

18 Apr '05

hi friends ! i have configured my ser according to the given configuration on the site www.voip-info.org/wiki+rtpproxy. but after running rtpproxy successfully my ser is not getting start. it is also not showing the generic socket for communication between ser and rtpproxy, the socket /var/run/rtpproxy.sock doesn`t exist not even at any other location, can anybody help me, why is it happening and suggest me something. thanks Deepak Dhiman

1 0

Re: [Serusers] still no help - usrloc synchronization
by Tina 18 Apr '05

18 Apr '05

Hi Paul, pls help me I ran into the following problem: ser1 receives REGISTER from VIP interface; then have to replicate the message to ser2. So I need to switch between interfaces - send from real IP. Do you know some way doing this with ser? Thx in advance, Tina Java Rockx <javarockx(a)gmail.com> wrote: Tina, I wish I had more information on that, but that LVS stuff is a black art to me. One of our engineers whipped up an LVS configuration that seems quite happy with the Call-ID "stickness" you described. All I can say right now is that it is apparently possible to do this with 100% open source. (i'm keeping my fingers crossed) :-) Regards, Paul On Apr 10, 2005 11:03 PM, Tina <kramarv(a)yahoo.com> wrote:Paul, thanks a lot, it helps a lot. The thing I do not understand - how you are going to solve call-id stickness (see my another post). Since we're also use LVS I had to find something proprietary, I am afraid this is not a best solution, but the only I have today. KRs, Tina Java Rockx <javarockx(a)gmail.com> wrote: Tina, I really don't know how the LVS server is configured because our network guy that we acquired from RedHat set all that stuff up. I do know that we basically have this set up: I hope the formatting goes well :-) +-----------+ +----------+ +-------+ +---------+ +--------------------+ | internet |------| Cisco |------| LVS |-----| Cisco |----| Application | | cloud | | 3600 | | | | 3600 | | & DB Servers | +-----------+ +----------+ +--------+ +---------+ +--------------------+ So the "Application & DB Servers" box represents multiple servers (all are dedicated to their service). These include SER, MySQL, Apache, configuration management, monitoring, RTP proxies, etc. MySQL is active-active so we have two-way replication. We only have 2 MySQL servers. And this is all we plan on ever having as MySQL is more than capable of handling anything ser can throw at it (given the right hardware). Each ser server is 100% oblivious to other SER servers except for when handling REGISTER messages. Here is basically the ser.cfg for each SER server: listen=10.3.0.221 # this will be 10.3.0.222 on the peer sip proxy modparam("usrloc", "db_mode", 1) # write-through route { if (method=="REGISTER") { if (src_ip==10.3.0.221) { # ip of peer ser proxy save_memory("location"); break; } else { save("location"); t_replicate("10.3.0.222"); break; }; }; } We have identifed a deficiency in the usrloc module urecord.c file whereby when using write-though mode the save_memory() function still attempt to update MySQL. This is a very bad thing because it causes a primary key violation on the location table. Yesterday I posted a patch to serdev to fix this, but today we identified another case where save_memory() tried to insert in to the location table. I'll post a revised patch to serdev as soon as I get a chance. So here is what happens. Assume I have two sip routers and two MySQL servers as: sip-01 is 10.3.0.221 sip-02 is 10.3.0.222 db-01 is 10.2.0.21 db-02 is 10.2.0.22 Now let's just assume that sip-01 is connected to db-01 and sip-02 is connected to db-02. When a SIP UA connects to a sip router (via LVS), assume sip-01 in this example and REGISTERs, sip-01 will call save("location") which updates/inserts to the location table. sip-01 then calls t_replicate to sip-02. sip-02 recieves the REGISTER message and calls save_memory("location") to update its in-memory cache only. It should never hit db-02. Now db-01 eventually (within a second or two) replicates to db-02 and now sip-02 is good to go, even in the event of a restart. Our difficulties mostly surrounded the fact that ser was inserting the physical IP of eth0 on the server as the top VIA rather than the VIP as required. We solved this by using record_route_preset() and passing the VIP as the IP. We also had to bring up a dummy interface in order to get ser to honor the VIP. Hope this helps. Regards, Paul On Apr 8, 2005 7:19 PM, Tina <kramarv(a)yahoo.com> wrote: Paul, we are using LVS as well. It creates template connection according to source Ip (e.g. UAC) and forwards to one of the tunneled IP. I tested this a couple of weeks ago, ser was happy and inserted VIP into via header, the only issue is to route the calls into right server. I agree, there is no such thing as SIP-level clustering. However, how would you solve location problem by DB replication - 1) usrloc does not update from DB....2) even though, I am afraid it can become bottleneck and lock down users ... KRs, Tina Java Rockx <javarockx(a)gmail.com> wrote: See my inline comments. Regards, Paul On Apr 8, 2005 1:08 PM, Greger V. Teigre <greger(a)teigre.com> wrote: Hi Tina, > I enjoy reading your posts, thanks a lot for the great work you are > doing for ser users! Thanks :-) > I am going to consider that kind of scenario as well (load balancer > returning Virtual IP address). I think for such configuration server > side has to keep session information and always forward a call to the > right server. I believe the load balancer can be in front and transparantly translate addresses in the SIP messages: User agent always sees a.b.c.d | Load balancer with public IP a.b.c.d | | ser1 (10.0.0.2) ser2 (10.0.0.3) Paul: You have a similar load-balancing solution, right? Do you use a commerical load balancer that you can reveal the name of? :-) Our platform is 100% open source. Our load balancer is LVS. I have no idea how it actually works. One if our engineers from RedHat set all that networking stuff up and it's way beyond me. The load balancer need to understand SIP (sort of) and use something (ex. IP address of user agent) to select SER server to send to and rewrite incoming to 10.0.0.2 and 10.0.0.3 respectively and outgoing to a.b.c.d. ser1 and ser2 will happily believe they are alone and communicate directly with user agent using their own private addresses. Both should know about all users (usrloc). Paul just recently posted a small patch for a setup where they use two-way replication between two That small patch will need a slight revision. We discovered through further testing that that patch only fixed attempts to INSERT into the location table. We didn't cover the UPDATE route, unfortunately. I believe Andrew has that patch done and is testing it. So I'll post it to serdev as soon as we know it works - possibly later today. mysql servers and where each SER instance will save all REGISTERs it receives from user agents, but only save to memory those replicated from the other SER peer. Klaus pointed out that the same can be done by not doing mysql replication, but have one DB for each SER and just use t_replicate() and save_noreply() (for two servers, for more than two you need to use forward_tcp()). I have heard complaints that this approach will generate too much network traffic. Also, t_replicate() does not guarantee delivery of the replication (forward_tcp is better, but not fool proof). So, the "best" solution would probably be to hope that Paul's I fully agree. Replication at the sip proxy level, IMHO, is not a good idea. From a purist point of view, replication is not a function of a SIP router - nor should it be. It is a function of the database and we believe that the MySQL replication is a better approach from a speed and reliabilty point of view - not to mention that it simplifies the tasks of the sip router. recent TCP/UDP patch for network access to FIFO commands get included in the CVS. Then, a new replication module should use XMLRPC or whatever protocol the patch is using to forward (in a guaranteed mode) the usrloc information, which then would be injected directly into SER's memory, just like save() would, and of course in whatever db_mode you would like. The replicaton module would have to keep a queue (of course, flushed to a DB or something) in case of network problems, so that the replication would be done (sooner or later) regardless of problems. Of course, this is a major undertaking, so for most replication scenarioes, I would think the forward_tcp() + save_noreply() would work just fine. g-) > "Greger V. Teigre" <greger(a)teigre.com> wrote: > I was thinking about a load balancing scenario where the load > balancer will replace the IP addresses. > g-) > > ---- Original Message ---- > From: Tina > To: Greger V. Teigre > Sent: Wednesday, April 06, 2005 09:45 PM > Subject: Re: [Serusers] still no help - usrloc synchronization > >> Thank you for givingme the scenario with "restricted IP" NAT, I am >> starting to find some acceptable solution. >> Unfortunately, "one-public-IP" approach is not free from problems >> also. If your SIP router inserts this "one-public-IP" into the VIA >> header, the reply routing goes via wrong SIP server... >> If your SIP server inserts its real-IP-address - the scenario >> mentioned above is still not resolved. >> Any comments? >> Tina >> >> "Greger V. Teigre" <greger(a)teigre.com> wrote: >> See inline. >> >>> If you use DNS server for load balancing... the client receives one >>> of your domain IP addresses according to SRV. I don't see the >>> problem >>> with a call here, cause UAC asks t! he address only once (before >>> sending INVITE). UAC already has the IP for BYE/reINVITEs. So why >>> would you replicate INVITEs? >> >> I would never replicate INVITEs, I would just make sure that they are >> proxied through the correct SER server (i.e. IP). >> >> The problems depends on your setup. If you have SERs with different >> IPs, ex UA1 has registered with server A and UA2 has registered with >> server B: If UA2 wants to call UA1 and UA is behind an IP restricted >> NAT, server A is stored in the NAT table of the NAT in front of UA1. >> If server B sends an INVITE to UA1, the INVITE will be refused by >> UA1's NAT. >> This is why a "one public IP" in front of a load balancing >> cluster probably is a good way to go. >> >>> If you use IPVS/LVS... I believe you can force SER to insert it's >>> public IP into VIA,! so there is no problem with replies. With >>> regard >>> to another requests, I believe load balancer keeps connection >>> template, then when another request comes it would be forwarded to >>> the same ser. >> >> Yes. There are different "keys" to use to load balance SIP messages. >> One good way from a NAT point of view is to use originating IP >> address. What you must remember is that the problem is not on the >> server side, but on the client side. The NAT will in many situations >> stop incoming UDP packets if the originating ip:port is not already >> stored in the NAT table. The Via header does not matter for the NAT. >> g-) >> >>> Any comments? >>> >>> "Greger V. Teigre" <greger(a)teigre.com> wrote: >>> Yes, I believe that is so. But still you get a problem if the NAT is >>> restricted, port-restricted or symmetric... The best would be to >>> load>> balance and always make sure that a given client is handled >>> through a given >>> SER (REGISTER and INVITEs). That includes forwarding INVITEs from >>> one >>> SER to >>> another... OR you must load balance in front of your servers with >>> one >>> common >>> public IP. >>> g-) >>> >>> Matt Schulte wrote: >>>> Ack, I didn't even think about NAT. Would these be added before it >>>> gets sent off to the second proxy? ie: >>>> >>>> if (!src_ip==blah.netlogic.net) { >>>> add_rcv_param(); >>>> t_replicate("blah.netlogic.net", "999"); >>>> }; >>>> >>>> -----Original Message----- >>>> From: Greger V. Teigre [mailto:greger@teigre.com] >>>> Sent: Tuesday, April 05, 2005 7:49 AM >>>> To: Matt Schulte; kramarv(a)yahoo.com >>>> ! ; Cc: serusers(a)lists.iptel.org >>>> Subject: ! Re: [Serusers] still no help - usrloc synchronization >>>> >>>> >>>> Well, you still have the NAT issues unless you do load balancing >>>> and your >>>> SER servers have the same public IP. >>>> Have you looked at 0.9.0 nathelper function add_rcv_param() ? It >>>> will add received info to the contact header for the other SER to >>>> process. Haven't really tried yet... >>>> g-) >>>> >>>> Matt Schulte wrote: >>>>> I'm starting to lean this direction, using t_replicate and all. I >>>>> could never get usrloc (db mode) to function properly.. >>>>> t_replicate is >>>> >>>>> a dirty but very effective workaround. >>>>> >>>>> -----Original Message----- >>>>> From: Greger V. Teigre [mailto:greger@teigre.com] >>>>> Sent: Saturday, April 02,! 2005 1:33 AM >>>>> To: kramarv(a)yahoo.com >>>>> Cc: serusers(a)lists.iptel.org >>>>> Subject: Re: [Serusers] still ! no help - usrloc synchronization >>>>> >>>>> >>>>> Have a look at this thread: >>>>> http://lists.iptel.org/pipermail/serusers/2005-January/014669.html >>>>> g-) >>>>> >>>>> Java Rockx wrote: >>>>>> Tina, >>>>>> >>>>>> I thought I saw you post the other day that you did not want to >>>>>> use t_replicate(), however, this is probably your best bet to >>>>>> getting this >>>>> >>>>>> to work, IMHO. >>>>>> >>>>>> Regards, >>>>>> Paul >>>>>> >>>>>> On Apr 1, 2005 4:08 PM, Tina wrote: >>>>> ! >> >>>>>> ! ;> Hi, please help me, I'm stuck with it!!!!! >>>>>>> I am trying to set up several sers with a shared MySQL database >>>>>>> for location service. >>>>>>> >>>>>>> I set in each ser.cfg: >>>>>>> >>>>>>> modparam("usrloc", "db_mode", 2) >>>>>>> modparam("usrloc", >>>>>>> "db_url","sql://ser:heslo@192.168.25.163/ser") >>>>>>> >>>>>>> and the servers are not synchronized. >>>>>>> The I set >>>>>>> modparam("usrloc", "db_mode", 2) >>>>>>> >>>>>>> >>>>>>> made UAC (Xlite) register to one of the servers. >>>>>>> I see it via usrloc, but there is no record in "location" mySQL >>>>>>> table....So others do not see the client and I'm unable to make >>>>>>> calls.... >>>>>>> >>>>>>> >>>>>>> Please help how to work with usrloc and mySQL... >>>>>>> >>>>>>> Tina, >>>>>>> software engineer >>>>>>> >>>>>>> ________________________________ >>>>>>> Do you Yahoo!? >>>>>>> Better first dates. More second dates. Yahoo! Personals >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> 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 >>>>> >>>>> _______________________________________________ >>>>> Serusers mailing list >>>>> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>> >>> >>> Yahoo! Messenger >>> Show us what our next emoticon should look like. Join the fun. >> >> >> Do you Yahoo!? >> Better first dates. More second dates. Yahoo! Personals > > > Yahoo! Messenger > Show us what our next emoticon should look like. Join the fun. _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers --------------------------------- Do you Yahoo!? Make Yahoo! your home page __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com --------------------------------- Do you Yahoo!? Plan great trips with Yahoo! Travel: Now over 17,000 guides!

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[Serusers] t_uac_dlg to nated UAS
by Juha Heinanen 17 Apr '05

17 Apr '05

if i use t_uac_dlg fifo command to send an invite request to a UAS that is behind NAT, t_uac sends ACK to the private contact address of 200 OK. in order fix the contact address in 200 OK, i should be able to somehow tell to t_uac_dlg that it should set a t_on_reply for the transaction. i didn't find any t_uac_dlg parameter for that purpose. perhaps there is some other way to get the contact address fixed? -- juha

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[Serusers] Mediaproxy 1.2.1 problem
by Tina 16 Apr '05

16 Apr '05

Hello, I am using mediaproxy 1.2.1 package. I have mediaproxy process does not answer when proxydispatcher tries to create a session - . Then I ran rtpgenerator to test the problem and got: =================================================== ./rtpgenerator.py Generator error: No answer from mediaproxy. ./rtpgenerator.py --ip=<=remote IP> --proxy=<remote IP> --gsm Traceback (most recent call last): File "./rtpgenerator.py", line 109, in __findFreeSocket raise GeneratorError, why[1] GeneratorError: Can't assign requested address Traceback (most recent call last): File "./rtpgenerator.py", line 192, in ? sender = RTPGenerator('request') File "./rtpgenerator.py", line 46, in __init__ self.addr = sock.getsockname() AttributeError: 'NoneType' object has no attribute 'getsockname' ./rtpgenerator.py --proxy=<remote mediaproxy IP> --gsm error: uncaptured python exception, closing channel <__main__.RTPGenerator connected 127.0.0.1:10000 at 0x824b3ec> (socket.error:(49, "Can't assign requested address") [/usr/local/lib/python2.3/asyncore.py|write|77] [/usr/local/lib/python2.3/asyncore.py|handle_write_event|397] [./rtpgenerator.py|handle_write|89]) error: uncaptured python exception, closing channel <__main__.RTPGenerator connected 127.0.0.1:10002 at 0x824b28c> (socket.error:(49, "Can't assign requested address") [/usr/local/lib/python2.3/asyncore.py|write|77] [/usr/local/lib/python2.3/asyncore.py|handle_write_event|397] [./rtpgenerator.py|handle_write|89]) Can anyone help me - what can be the problem? I tried the same package on another machines, everything was ok.... Thx in advance, Tina --------------------------------- Do you Yahoo!? Plan great trips with Yahoo! Travel: Now over 17,000 guides!

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Re: [Serusers] how to configure mediaproxy to work with remote RTPserver - pls help!
by Tina 16 Apr '05

16 Apr '05

Ran with remote options, will see how it works when solve replication via internal-interface. Thx! Tina "Greger V. Teigre" <greger(a)teigre.com> wrote: Yes. I believe mediaproxy.so uses a socket to communicate with proxydispatcher and that proxydispatcher will use the default route for outgoing communications. g-) ---- Original Message ---- From: Tina To: Greger V. Teigre ; serusers(a)lists.iptel.org Sent: Saturday, April 16, 2005 09:23 PM Subject: Re: [Serusers] how to configure mediaproxy to work with remote RTPserver - pls help! > Greger, thank you, this helps. > So I would run mediaproxy.py like that: > ./mediaproxy.py --listen <proxyIP:25060> --allow <IP1>,<IP2>,... > > where <proxyIP:25060> - is TCP address/port of mediaproxy and > <IP1>,<IP2>... - list of addresses of proxydispatchers in the domain? > > I didn't find information how proxydispatcher picks up its IP > interface. If the machine has several IP devices, I think I should > list all of them. > > BRs, > Tina > > "Greger V. Teigre" <greger(a)teigre.com> wrote: > Tina, > As you see in the mediaproxy README: > The dispatcher implemented here is based on distributing the sessions > based > on the SIP domain of the caller/destination using DNS SRV records. If > a > domain doesn't define any mediaproxy server, a default will be used. > > Further: > The dispatcher will use the sender's domain if the sender is local, > else will > use the destination's domain if the destination is local. If none of > them is > local, it will refuse to use a proxy server at all. > > You must then create DNS SRV records for the domains. > http://www.voip-info.org/wiki-DNS+SRV > > The proxydispatcher.py should run locally. Each proxy server should > run the mediaproxy.py proxy server with the --listen any:25060 > (default port) --allow ip-of-proxydispatcher > > Hope this helps. > g-) > ----- Original Message ----- > From: Tina > To: serusers(a)lists.iptel.org > Sent: Saturday, April 16, 2005 03:36 AM > Subject: [Serusers] how to configure mediaproxy to work with remote > RTPserver - pls help! > > > Hello, > I've tried mediaproxy package, it worked fine running on the same > machine. I need the following configuation: > ser+proxydispatcher > - RTP proxy (mediaproxy) > - RTP proxy 2 ... > How can I make proxydispatcher retrieves RTP proxy address by DNS? > Thanks in advance, > really need help! > > BRs, > Tina > __________________________________________________ > Do You Yahoo!? > Tired of spam? Yahoo! Mail has the best spam protection around > http://mail.yahoo.com > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > > Do you Yahoo!? > Read only the mail you want - Yahoo! Mail SpamGuard. --------------------------------- Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone.

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Re: [Serusers] how to configure mediaproxy to work with remote RTPserver - pls help!
by Tina 16 Apr '05

16 Apr '05

Greger, thank you, this helps. So I would run mediaproxy.py like that: ./mediaproxy.py --listen <proxyIP:25060> --allow <IP1>,<IP2>,... where <proxyIP:25060> - is TCP address/port of mediaproxy and <IP1>,<IP2>... - list of addresses of proxydispatchers in the domain? I didn't find information how proxydispatcher picks up its IP interface. If the machine has several IP devices, I think I should list all of them. BRs, Tina "Greger V. Teigre" <greger(a)teigre.com> wrote: Tina, As you see in the mediaproxy README: The dispatcher implemented here is based on distributing the sessions based on the SIP domain of the caller/destination using DNS SRV records. If a domain doesn't define any mediaproxy server, a default will be used. Further: The dispatcher will use the sender's domain if the sender is local, else will use the destination's domain if the destination is local. If none of them is local, it will refuse to use a proxy server at all. You must then create DNS SRV records for the domains. http://www.voip-info.org/wiki-DNS+SRV The proxydispatcher.py should run locally. Each proxy server should run the mediaproxy.py proxy server with the --listen any:25060 (default port) --allow ip-of-proxydispatcher Hope this helps. g-) ----- Original Message ----- From: Tina To: serusers(a)lists.iptel.org Sent: Saturday, April 16, 2005 03:36 AM Subject: [Serusers] how to configure mediaproxy to work with remote RTPserver - pls help! Hello, I've tried mediaproxy package, it worked fine running on the same machine. I need the following configuation: ser+proxydispatcher - RTP proxy (mediaproxy) - RTP proxy 2 ... How can I make proxydispatcher retrieves RTP proxy address by DNS? Thanks in advance, really need help! BRs, Tina __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com --------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers --------------------------------- Do you Yahoo!? Read only the mail you want - Yahoo! Mail SpamGuard.

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[Serusers] Password not being sent during Radius Auth
by Lucas Aimaretto 16 Apr '05

16 Apr '05

Hi there, This is my ser.cfg configuration if(method=="REGISTER") { if (!radius_www_authorize("")) { www_challenge("", "0"); break; }; save("location"); break; }; And here is some sniffing done with ngrep ... U IP_UA:11006 -> IP_SER:5060 REGISTER sip:IP_SER SIP/2.0. Via: SIP/2.0/UDP 192.168.1.178:11006;rport;branch=z9hG4bK810E80344EB24AE5B8D5FD21043E78CE . From: Lucas <sip:1991006@IP_SER>. To: Lucas <sip:1991006@IP_SER>. Contact: "Lucas" <sip:1991006@192.168.1.178:11006>. Call-ID: FCA6F7DD4BA94FA090F446BCE4AAE5B9@IP_SER. CSeq: 57327 REGISTER. Expires: 1800. Authorization: Digest username="1991006@IP_SER",realm="IP_SER",nonce="425b03326e0f4f0071f1a766 4c8823f1271f1212",response="8b9ec4e8e633c5dd7d4aee4aef1ffdba",uri="sip:I P_SER". Max-Forwards: 70. User-Agent: X-PRO build 1082. Content-Length: 0. . # U IP_SER:5060 -> IP_UA:11006 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.178:11006;rport=11006;branch=z9hG4bK810E80344EB24AE5B8D5FD2104 3E78CE;received=IP_UA. From: Lucas <sip:1991006@IP_SER>. To: Lucas <sip:1991006@IP_SER>;tag=6f0d146d94c4cb042663ff3cf87e2e72.d766. Call-ID: FCA6F7DD4BA94FA090F446BCE4AAE5B9@IP_SER. CSeq: 57327 REGISTER. WWW-Authenticate: Digest realm="IP_SER", nonce="425b03326e0f4f0071f1a7664c8823f1271f1212". Content-Length: 0. Warning: 392 IP_SER:5060 "Noisy feedback tells: pid=23023 req_src_ip=IP_UA req_src_port=11006 in_uri=sip:IP_SER out_uri=sip:IP_SER via_cnt==1". The thing is that I'm not seeing the Password Attribute at the radius output ... Well, to be honest, I do not know wich is the attribute SER uses to send password, but, the truth is no Password Attribute is sent to RADIUS. Any ideas ? Regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.308 / Virus Database: 266.9.6 - Release Date: 11/04/2005

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RE: [Serusers] still no help - usrloc synchronization
by Matt Schulte 16 Apr '05

16 Apr '05

Forgive my ignorance, why is NAT a bottleneck? Are you refering to CPU load or managibility? With alot of load balancers, NAT is the only way to go. -----Original Message----- From: Tina [mailto:kramarv@yahoo.com] Sent: Monday, April 11, 2005 11:37 AM To: Java Rockx Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] still no help - usrloc synchronization Pretty good news, Paul. I've only tested IP tunneling. It worked for me. The problem I described was born in my thoughts. I've also read some complaints from SIP-LVS users. The only configuration where Call-Id stickness does not necessary is LVS-NAT. Unfortunately, NAT becomes bottleneck very fast. Java Rockx <javarockx(a)gmail.com> wrote: Tina, I wish I had more information on that, but that LVS stuff is a black art to me. One of our engineers whipped up an LVS configuration that seems quite happy with the Call-ID "stickness" you described. All I can say right now is that it is apparently possible to do this with 100% open source. (i'm keeping my fingers crossed) :-) Regards, Paul On Apr 10, 2005 11:03 PM, Tina <kramarv(a)yahoo.com> wrote: Paul, thanks a lot, it helps a lot. The thing I do not understand - how you are going to solve call-id stickness (see my another post). Since we're also use LVS I had to find something proprietary, I am afraid this is not a best solution, but the only I have today. KRs, Tina Java Rockx <javarockx(a)gmail.com> wrote: Tina, I really don't know how the LVS server is configured because our network guy that we acquired from RedHat set all that stuff up. I do know that we basically have this set up: I hope the formatting goes well :-) +-----------+ +----------+ +-------+ +---------+ +--------------------+ | internet |------| Cisco |------| LVS |-----| Cisco |----| Application | | cloud | | 3600 | | | | 3600 | | & DB Servers | +-----------+ +----------+ +--------+ +---------+ +--------------------+ So the "Application & DB Servers" box represents multiple servers (all are dedicated to their service). These include SER, MySQL, Apache, configuration management, monitoring, RTP proxies, etc. MySQL is active-active so we have two-way replication. We only have 2 MySQL servers. And this is all we plan on ever having as MySQL is more than capable of handling anything ser can throw at it (given the right hardware). Each ser server is 100% oblivious to other SER servers except for when handling REGISTER messages. Here is basically the ser.cfg for each SER server: listen=10.3.0.221 <http://10.3.0.221/> # this will be 10.3.0.222 <http://10.3.0.222/> on the peer sip proxy modparam("usrloc", "db_mode", 1) # write-through route { if (method=="REGISTER") { if (src_ip==10.3.0.221 <http://10.3.0.221/> ) { # ip of peer ser proxy save_memory("location"); break; } else { save("location"); t_replicate("10.3.0.222 <http://10.3.0.222/> "); break; }; }; } We have identifed a deficiency in the usrloc module urecord.c file whereby when using write-though mode the save_memory() function still attempt to update MySQL. This is a very bad thing because it causes a primary key violation on the location table. Yesterday I posted a patch to serdev to fix this, but today we identified another case where save_memory() tried to insert in to the location table. I'll post a revised patch to serdev as soon as I get a chance. So here is what happens. Assume I have two sip routers and two MySQL servers as: sip-01 is 10.3.0.221 <http://10.3.0.221/> sip-02 is 10.3.0.222 <http://10.3.0.222/> db-01 is 10.2.0.21 <http://10.2.0.21/> db-02 is 10.2.0.22 <http://10.2.0.22/> Now let's just assume that sip-01 is connected to db-01 and sip-02 is connected to db-02. When a SIP UA connects to a sip router (via LVS), assume sip-01 in this example and REGISTERs, sip-01 will call save("location") which updates/inserts to the location table. sip-01 then calls t_replicate to sip-02. sip-02 recieves the REGISTER message and calls save_memory("location") to update its in-memory cache only. It should never hit db-02. Now db-01 eventually (within a second or two) replicates to db-02 and now sip-02 is good to go, even in the event of a restart. Our difficulties mostly surrounded the fact th at ser was inserting the physical IP of eth0 on the server as the top VIA rather than the VIP as required. We solved this by using record_route_preset() and passing the VIP as the IP. We also had to bring up a dummy interface in order to get ser to honor the VIP. Hope this helps. Regards, Paul On Apr 8, 2005 7:19 PM, Tina <kramarv(a)yahoo.com> wrote: Paul, we are using LVS as well. It creates template connection according to source Ip (e.g. UAC) and forwards to one of the tunneled IP. I tested this a couple of weeks ago, ser was happy and inserted VIP into via header, the only issue is to route the calls into right server. I agree, there is no such thing as SIP-level clustering. However, how would you solve location problem by DB replication - 1) usrloc does not update from DB....2) even though, I am afraid it can become bottleneck and lock down users ... KRs, Tina Java Rockx <javarockx(a)gmail.com> wrote: See my inline comments. Regards, Paul On Apr 8, 2005 1:08 PM, Greger V. Teigre <greger(a)teigre.com> wrote: Hi Tina, > I enjoy reading your posts, thanks a lot for the great work you are > doing for ser users! Thanks :-) > I am going to consider that kind of scenario as well (load balancer > returning Virtual IP address). I think for such configuration server > side has to keep session information and always forward a call to the > right server. I believe the load balancer can be in front and transparantly translate addresses in the SIP messages: User agent always sees a.b.c.d | Load balancer with public IP a.b.c.d | | ser1 (10.0.0.2 <http://10.0.0.2/> ) ser2 (10.0.0.3 <http://10.0.0.3/> ) Paul: You have a similar load-balancing solution, right? Do you use a commerical load balancer that you can reveal the name of? :-) Our platform is 100% open source. Our load balancer is LVS. I have no idea how it actually works. One if our engineers from RedHat set all that networking stuff up and it's way beyond me. The load balancer need to understand SIP (sort of) and use something (ex. IP address of user agent) to select SER server to send to and rewrite incoming to 10.0.0.2 <http://10.0.0.2/> and 10.0.0.3 <http://10.0.0.3/> respectively and outgoing to a.b.c.d. ser1 and ser2 will happily believe they are alone and communicate directly with user agent using their own private addresses. Both should know about all users (usrloc). Paul just recently posted a small patch for a setup where they use two-way replication between two That small patch will need a slight revision. We discovered through further testing that that patch only fixed attempts to INSERT into the location table. We didn't cover the UPDATE route, unfortunately. I believe Andrew has that patch done and is testing it. So I'll post it to serdev as soon as we know it works - possibly later today. mysql servers and where each SER instance will save all REGISTERs it receives from user agents, but only save to memory those replicated from the other SER peer. Klaus pointed out that the same can be done by not doing mysql replication, but have one DB for each SER and just use t_replicate() and save_noreply() (for two servers, for more than two you need to use forward_tcp()). I have heard complaints that this approach will generate too much network traffic. Also, t_replicate() does not guarantee delivery of the replication (forward_tcp is better, but not fool proof). So, the "best" solution would probably be to hope that Paul's I fully agree. Replication at the sip proxy level, IMHO, is not a good idea. From a purist point of view, replication is not a function of a SIP router - nor should it be. It is a function of the database and we believe that the MySQL replication is a better approach from a speed and reliabilty point of view - not to mention that it simplifies the tasks of the sip router. recent TCP/UDP patch for network access to FIFO commands get included in the CVS. Then, a new replication module should use XMLRPC or whatever protocol the patch is using to forward (in a guaranteed mode) the usrloc information, which then would be injected directly into SER's memory, just like save() would, and of course in whatever db_mode you would like. The replicaton module would have to keep a queue (of course, flushed to a DB or something) in case of network problems, so that the replication would be done (sooner or later) regardless of problems. Of course, this is a major undertaking, so for most replication scenarioes, I would think the forward_tcp() + save_noreply() would work just fine. g-) > "Greger V. Teigre" <greger(a)teigre.com> wrote: > I was thinking about a load balancing scenario where the load > balancer will replace the IP addresses. > g-) > > ---- Original Message ---- > From: Tina > To: Greger V. Teigre > Sent: Wednesday, April 06, 2005 09:45 PM > Subject: Re: [Serusers] still no help - usrloc synchronization > >> Thank you for givingme the scenario with "restricted IP" NAT, I am >> starting to find some acceptable solution. >> Unfortunately, "one-public-IP" approach is not free from problems >> also. If your SIP router inserts this "one-public-IP" into the VIA >> header, the reply routing goes via wrong SIP server... >> If your SIP server inserts its real-IP-address - the scenario >> mentioned above is s till not resolved. >> Any comments? >> Tina >> >> "Greger V. Teigre" <greger(a)teigre.com> wrote: >> See inline. >> >>> If you use DNS server for load balancing... the client receives one >>> of your domain IP addresses according to SRV. I don't see the >>> problem >>> with a call here, cause UAC asks t! he address only once (before >>> sending INVITE). UAC already has the IP for BYE/reINVITEs. So why >>> would you replicate INVITEs? >> >> I would never replicate INVITEs, I would just make sure that they are >> proxied through the correct SER server (i.e. IP). >> >> The problems depends on your setup. If you have SERs with different >> IPs, ex UA1 has registered with server A and UA2 ha s registered with >> server B: If UA2 wants to call UA1 and UA is behind an IP restricted >> NAT, server A is stored in the NAT table of the NAT in front of UA1. >> If server B sends an INVITE to UA1, the INVITE will be refused by >> UA1's NAT. >> This is why a "one public IP" in front of a load balancing >> cluster probably is a good way to go. >> >>> If you use IPVS/LVS... I believe you can force SER to insert it's >>> public IP into VIA,! so there is no problem with replies. With >>> regard >>> to another requests, I believe load balancer keeps connection >>> template, then when another request comes it would be forwarded to >>> the same ser. >> >> Yes. There are different "keys" to use to load balance SIP messages. >> One good way from a NAT point of view is to use originating IP > > address. What you must remember is that the problem is not on the >> server side, but on the client side. The NAT will in many situations >> stop incoming UDP packets if the originating ip:port is not already >> stored in the NAT table. The Via header does not matter for the NAT. >> g-) >> >>> Any comments? >>> >>> "Greger V. Teigre" <greger(a)teigre.com> wrote: >>> Yes, I believe that is so. But still you get a problem if the NAT is >>> restricted, port-restricted or symmetric... The best would be to >>> load>> balance and always make sure that a given client is handled >>> through a given >>> SER (REGISTER and INVITEs). That includes forwarding INVITEs from >>> one >>> SER to >>> another... OR you must load balance in front of your servers with >>> one >>> common >>> public IP. >>> g-) >>> >>> Matt Schulte wrote: >>>> Ack, I didn't even think about NAT. Would these be added before it >>>> gets sent off to the second proxy? ie: >>>> >>>> if (!src_ip==blah.netlogic.net <http://blah.netlogic.net/> ) { >>>> add_rcv_param(); >>>> t_replicate("blah.netlogic.net <http://blah.netlogic.net/> ", "999"); >>>> }; >>>> >>>> -----Original Message----- >>>> From: Greger V. Teigre [mailto:greger@teigre.com <mailto:greger@tei gre.com> ] >>>> Sent: Tuesday, April 05, 2005 7:49 AM >>>> To: Matt Schulte; kramarv(a)yahoo.com >>>> ! ; Cc: serusers(a)lists.iptel.org >>>> Subject: ! Re: [Serusers] still no help - usrloc synchronization >>>> >>>> >>>> Well, you still have the NAT issues unless you do load balancing >>>> and your >>>> SER servers have the same public IP. >>>> Have you looked at 0.9.0 nathelper function add_rcv_param() ? It >>>> will add received info to the contact header for the other SER to >>>> process. Haven't really tried yet... >>>> g-) >>>> >>>> Matt Schulte wrote: >>>>> I'm starting to lean this direction, using t_replicate and all. I >>>>> could never get usrloc (db mode) to function properly.. >>>>> t_replicate is >>>> >>>>> a dirty but very effective workaround. >>>>> >>>>> -----Original Message----- >>>>> From: Greger V. Teigre [mailto:greger@teigre.com] >>>>> Sent: Saturday, April 02,! 2005 1:33 AM >>>>> To: kramarv(a)yahoo.com >>>>> Cc: serusers(a)lists.iptel.org >>>>> Subject: Re: [Serusers] still ! no help - usrloc synchronization >>>>> >>>>> >>>>> Have a look at this thread: >>>>> http://lists.iptel.org/pipermail/serusers/2005-January/014669.html >>>>> g-) >>>>> >>>>> Java Rockx wrote: >>>>>> Tina, >>>>>> >>>>>> I thought I saw you post the other day that you did not want to >>>>>> use t_replicate(), however, this is probably your best bet to >>>>>> getting this >>>>> >>>>>> to work, IMHO. >>>>>> >>>>>> Regards, >>>>>> Paul >>>>>> >>>>>> On Apr 1, 2005 4:08 PM, Tina wrote: >>>>> ! >> >>>>>> ! ;> Hi, please help me, I'm stuck with it!!!!! >>>>>>> I am trying to set up several sers with a shared MySQL database >>>>>>> for location service. >>>>>>> >>>>>>> I set in each ser.cfg: >>>>>>> >>>>>>> modparam("usrloc", "db_mode", 2) >>>>>>> modparam("usrloc", >>>>>>> "db_url","sql://ser:heslo@192.168.25.163/ser") >>>>>>> >>>>>>> and the servers are not synchronized. >>>>>>> The I set >>>>>>> modparam("usrloc", "db_mode", 2) >>>>>>> >>>>>>> >>>>>>> made UAC (Xlite) register to one of the servers. >>>>>>> I see it via usrloc, but there is no record in "location" mySQL >>>>>>> table....So others do not see the client and I'm unable to make >>>>>>> calls.... >>>>>>> >>>>>>> >>>>>>> Please help how to work with usrloc and mySQL... >>>>>>> >>>>>>> Tina, >>>>>>> software engineer >>>>>>> >>>>>>> ________________________________ >>>>>>> Do you Yahoo!? >>>>>>> Better first dates. More second dates. Yahoo! Personals >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> 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 >>>>> >>>>> _______________________________________________ >>>>> Serusers mailing list >>>>> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >>> >>> >>> Yahoo! Messenger >>> Show us what our next emoticon should look like. Join the fun. >> >> >> Do you Yahoo!? >> Better first dates. More second dates. Yahoo! Personals > > > Yahoo! Messenger > Show us what our next emoticon should look like. Join the fun. _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _____ Do you Yahoo!? Make Yahoo! your home page <http://us.rd.yahoo.com/my/navbar/sethp/*http://www.yahoo.com/r/hs> __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com <http://mail.yahoo.com/> __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

RE: [Serusers] Rewrite To Heaader
by Carlos Vicente 16 Apr '05

16 Apr '05

Take a look at the replace method in the textops module http://www.iptel.org/ser/doc/modules/html/textops.html You could do something like: replace("^To: <sip:", "To: acct# <sip:"); cv On Thu, 2005-04-14 at 20:37 +0200, serusers-request(a)lists.iptel.org wrote: > Message: 33 > Date: Wed, 13 Apr 2005 08:29:50 -0400 > From: Eric Dean <edean(a)purespeedplus.com> > Subject: [Serusers] Rewrite To Heaader > To: serusers(a)lists.iptel.org > Message-ID: <425D10BE.7090703(a)purespeedplus.com> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Our VoIP service provider routes calls based upon the To field within > the SIP header. While SER can easily rewrite the URI, I need to > actually prepend our account code to every outgoing call with a > prefix. > does anyone know how to prefix the To field

2 1

[Serusers] Sorry, Newbie
by Nucleo Wireless LSI 16 Apr '05

16 Apr '05

Hi every one, My name is Fabio and I work in a lab in my university, and I want to use ser as an VoIP server in the lab so every computer in our network is a SIP Client with softfones making calls from one to another. This is an test scenario, I know that the use of a SIP server is not necessary to implement such functionality, but we want to test SIP working under this circumstances. If anyone knows were to find documentation, HOW-TO's and step by step guides, I'd be very pleased. Tank You for the attention Fabio Luiz Newbie Yahoo! Acesso Grátis - Internet rápida e grátis. Instale o discador agora! http://br.acesso.yahoo.com/

3 2

[Serusers] how to configure mediaproxy to work with remote RTP server - pls help!
by Tina 16 Apr '05

16 Apr '05

Hello, I've tried mediaproxy package, it worked fine running on the same machine. I need the following configuation: ser+proxydispatcher - RTP proxy (mediaproxy) - RTP proxy 2 ... How can I make proxydispatcher retrieves RTP proxy address by DNS? Thanks in advance, really need help! BRs, Tina __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

2 1

[Serusers] pls help how to configure remote medisproxy....
by Tina 16 Apr '05

16 Apr '05

Hello, I've tried mediaproxy package, it worked fine running on the same machine. I need the following configuation: ser+proxydispatcher - RTP proxy (mediaproxy) - RTP proxy 2 ... How can I make proxydispatcher retrieves RTP proxy address by DNS? Thanks in advance, really need help! BRs, Tina __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] Remote ser management instead of FIFO
by Martin Rusnak 15 Apr '05

15 Apr '05

Dear all, I would like to connect my web portal based on java technology with the ser daemon in order to provide some administration functionality. The solution would be to use FIFO interface, but it has two major disadvantages: 1. The portal might be deployed on different host. 2. Although Java can communicate via FIFO pipeline, it cannot create a new pilepine. Is there any possibility to manage ser daemon remotely via a socket interface? Thanks, Martin

4 7

[Serusers] Status of pa module in CVS HEAD?
by Hendrik Scholz 15 Apr '05

15 Apr '05

Hi! I just had a closer look at RFC 3265 (6.3 SUBSCRIBE Bodies) and figured out that SUBSCRIBES may have empty bodies if the header fields are sufficient. SER from CVS replies with a 400 error saying 'Error while parsing headers'. this is how a sample packet looks like: # U xx.xx.xx.xx:5061 -> xx.xx.xx.xx:5060 SIP/2.0 400 Bad Request. Via: SIP/2.0/UDP xx.xx.xx.xx;branch=z9hG4bKea3f.82e1f753.0. Via: SIP/2.0/UDP xx.xx.xx.xx;branch=0. Via: SIP/2.0/UDP xx.xx.xx.xx:49449. From: "user" <sip:user@foobar.bla>;tag=17cda3ce319449249f909aba8e53d03e;epid=44df82f654. To: <sip:bbbbb@foobar.bla>;tag=95329101123423eab1637e9ad490b3a6-e941. Call-ID: db28e67d79614e44b55275b6c6a025bd(a)xx.xx.xx.xx. CSeq: 1 SUBSCRIBE. Error while parsing headersContent-Length: 0. The error message is misplaced from my point of view. ser -E says: 12(24674) SUBSCRIBE from:sip:user@foobar.bla to:sip:bbbb@foobar.bla ruri:sip:bbbbb@foobar.bla messageid:3 12(24674) handle_subscription() entered 12(24674) parse_hfs(): Error while parsing headers 12(24674) handle_subscription(): Error while parsing message header 12(24674) handle_subscription about to send_reply and return -2 I cannot get my clients using 'application/xpidf+xml' to successfully exchange presence information mainly due to the fact that SER doesn't seem to take care of it: 11(24673) SUBSCRIBE from:sip:bbbb@foobar.bla to:sip:user@foobar.bla ruri:sip:user@foobar.bla messageid:5 11(24673) handle_subscription() entered 11(24673) parsing accept header 11(24673) check_message -0- _m=0x8141170 11(24673) check_message -1-11(24673) pa check_message: accept=application/xpidf+xml parsed=:0x300ff 11(24673) check_message -2- accepts_mimes=0x81425e0 11(24673) check_message -3- 11(24673) check_message -4- parsed_event=0x81425a0 11(24673) check_message -4- eventtype=0x0001 11(24673) check_message -4a- eventtype=0x1 epm[i].event_type=0x111(24673) check_message -4c- eventtype=0x1 mimetype=0x30005 accepts_mimes[k]=0x300ff 11(24673) check_message -4c- eventtype=0x1 mimetype=0x30009 accepts_mimes[k]=0x300ff 11(24673) check_message -4c- eventtype=0x1 mimetype=0x30008 accepts_mimes[k]=0x300ff 11(24673) check_message -4a- eventtype=0x1 epm[i].event_type=0x211(24673) check_message(): Accepts application/xpidf+xml not valid for event package et=presence 11(24673) handle_subscription(): Error while checking message 11(24673) handle_subscription about to send_reply and return -2 What's the status on presence? I've got a bunch of small patches that fix some minor problems but wonder if it makes sense to send them upstream. Anybody running presence on CVS HEAD? Thanks, Hendrik -- freenet Cityline GmbH, Hamburger Chaussee 2-4, 24114 Kiel, Germany Phone: +49 (0)431 9020552, Fax: +49 (0)431 9020559 Internet: http://www.freenet.de, eMail: hendrik.scholz(a)freenet-ag.de

2 1

[Serusers] e164 rewrite script
by Matt Schulte 15 Apr '05

15 Apr '05

Does anyone have a config to convert phone numbers (user@) to e164 numbers? Mainly american numbers.. Would save me a bit of time, else I can try to write something out.. Thanks.

1 0

[Serusers] change Message body
by deno＠emitt.de 15 Apr '05

15 Apr '05

Hallo I'm a newbie. I am writing my diplom (germany) about building an sip application server. I will try it with ser. Now i got a problem. I want to change the body of a Message. First i will use exec_msg to translate a text out of the body. Now i got the problem to rewrite this text after translation. please can someone help me.

1 0

RE: [Serusers] SER hung - Log Information?
by Matt Schulte 15 Apr '05

15 Apr '05

I see, our script is a little more archaic, it runs a perl script and connects each time. Our alternative was to daemonize but as you clearly demonstrated it did no/little good. The purpose for our script is an aliases db, because I could never get usrloc to work correctly mainly. re: 1) Yes this has been my very problem, since my problem is periodic and not load related (??) leaving full debug on is very painful if not impossible especially when forked.. 2) My only alternative is to use ENUM lookups, most everything else would require an external process and/or writing a SER module (gah).. If you're using simple URI replacement then it's recommended of course to rewrite within SER, what kind of functions does your script perform? 3) I used sipp for testing everything, this was with and without the external hog process running. Matt -----Original Message----- From: Ricardo Martinez [mailto:rmartinez@redvoiss.net] Sent: Friday, April 15, 2005 12:30 PM To: Matt Schulte; serusers(a)lists.iptel.org Subject: RE: [Serusers] SER hung - Log Information? Interesting too. My "translate" script takes an average of 17ms to return the desired value. By HTTP requests it connect to another process in another machine, which have an "always open" connection with my database. So i don't need to open a new connection with my database every time an INVITE reach the server. BUT STILL DESPITE OF THIS, MY SER SEEMS TO FREEZE FOR SOME REASON. I have some questions. 1.-How to check if the bottle neck is in the exec_dset command?? 2.-If we decide to replace this command (exec_dset)for a function in another module, what module would be recommendable?, for example how the ENUM module could do this?. We have the "restriction" that we need to query our databes for the R-URI change, so the module that will acomplish this task needs to have connection with my database, maybe through radius, sql, etc.? 3.-You mention that you made some kind of "stress" test with your platform. You loaded it with 30000 call in one hour. How you did that? Does your exec_dset running at the same time?. Thanks. Any help will be highly appreciated. Ricardo.- > -----Mensaje original----- > De: Matt Schulte [mailto:mschulte@netlogic.net] > Enviado el: Viernes, 15 de Abril de 2005 11:20 > Para: Ricardo Martinez; serusers(a)lists.iptel.org > Asunto: RE: [Serusers] SER hung - Log Information? > > > Interesting, we also use an external process to connect to a DB and > retrieve URI replacement. We were thinking that it's running "too > slow" and thus hanging SER, even though the logs make it appear as if > the > process(es) was hanging at random points. We're in the process of > eliminating this script and switching over to ENUM or similar. When we > restarted our MySQL server it improved the query time (it was running > for 180 days heheh) and seems to have made SER hang less. How long > does your external script take to run on average? If it's more > than a second > then that seems to be unnacceptable for SER.. > > P.S. The crappy thing is this happens over a period of time, > when I load > tested the server it ran flawlessly. Pushed 30000 calls through it in > less than an hour with no problems. The MySQL server in question is in > the same switch segment/subnet as well. > > > Matt > > -----Original Message----- > From: Ricardo Martinez [mailto:rmartinez@redvoiss.net] > Sent: Thursday, April 14, 2005 4:32 PM > To: Matt Schulte; serusers(a)lists.iptel.org > Subject: RE: [Serusers] SER hung - Log Information? > > > Hello Matt. > Thanks for your answer, i checked the post "exec_dset fork limit?" > (http://lists.iptel.org/pipermail/serusers/2004-December/014150.html) > and i think i have the same problem. I'm using the exec_dset > command to > manipulate the RURI, we need to do this in order to add prefixes > according to certain user profiles. > > route { > if (method=="INVITE" || method=="CANCEL") { > exec_dset("/usr/local/etc/ser/translate"); > }; > > As you can see i'm using it for every INVITE and CANCEL, the > "translate" > file is a C program that connects to a DB and returns the > desired RURI. > Could this be the cause of SER freeze?. Can we have an > opinion from the > developers? Is there a way to do the same thing with maybe another > module?. For the record i'm using : > > OS : Linux Red Hat 9.0 > SER : 0.8.14 > Mediaproxy : 1.2.1 > > I would really appreciate any help here. > Thanks!! > > Best Regards, > > Ricardo.- > > > -----Mensaje original----- > > De: Matt Schulte [mailto:mschulte@netlogic.net] > > Enviado el: Jueves, 14 de Abril de 2005 16:11 > > Para: Ricardo Martinez; serusers(a)lists.iptel.org > > Asunto: RE: [Serusers] SER hung - Log Information? > > > > > > I've been posting about this for some time now,I have never been > > able to figure out the problem. Do you use external proccesses or > > anything? That's the only "weird" thing we really do with SER, > > everything else is > > pretty straight forward. > > > > Search back from my posts and see what you think. > > > > -----Original Message----- > > From: Ricardo Martinez [mailto:rmartinez@redvoiss.net] > > Sent: Thursday, April 14, 2005 11:16 AM > > To: 'serusers(a)lists.iptel.org' > > Subject: [Serusers] SER hung - Log Information? > > > > > > Hello list. > > I had been experimenting some weird behavior in my SER program. > A > > couple of times my SER seems to be hung. For example the > first time, > > all the REGISTER messages reaching the server were not > > challenged, even > > worst the server don't respond anything. The second time > it happened > > with the INVITE messages. The weird thing is that the SER > > process were > > still running. I don't know what could cause this problem, but my > > question is : Is there a way to log information about the "internal > > system process" to have a clue about what is happening?. I > > would really > > appreciate any help. Thanks > > > > > > Ricardo Martinez.- > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers > > >

1 0

[Serusers] Another Looping Question
by Java Rockx 15 Apr '05

15 Apr '05

Hi All. Below is a snippet of an INVITE and you can see the usual "trying - your call is important to us" in the dialog. Anyhow, for some reason SER occasionally attempts to sent the ACK to itself, which causes a looping condition. The last ACK shown is the start of the looping condition. It only happens on occasion. SER should have consumed the ACK rather than attempted to forward it. Can anyone explain this? Regards, Paul IP Legend ------------------- 10.3.0.221 <http://10.3.0.221> - SER (behind a Cisco 3600 so ALG rewrote the IPs) 69.36.46.178 <http://69.36.46.178> - NATed SIP UA U 2005/04/15 02:19:16.252614 69.35.46.178:5060 <http://69.35.46.178:5060> -> 10.3.0.221:5060 <http://10.3.0.221:5060> INVITE sip:8187768080@sipdev.mycompany.net SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102:5060 <http://192.168.1.102:5060> ;branch=z9hG4bK3915905414. From: Test User <sip:3215591439@sipdev.mycompany.net>;tag=1465261089. To: <sip:8187768080@sipdev.mycompany.net>. Call-ID: 2671219156(a)192.168.1.102. CSeq: 101 INVITE. Contact: <sip:3215591439@192.168.1.102:5060>. Proxy-Authorization: Digest username="3215591439", realm=" sipdev.mycompany.net <http://sipdev.mycompany.net>", nonce="425f5e0de25efdb3f5a6125dcf2147bf30fe9177", uri=" sip:8187768080@sipdev.mycompany.net", response="61996c3bdc10990fec7e245751777516", algorithm=MD5, cnonce="ae02e47a3355bada49ce344c92f48587", qop=auth, nc=00000002. max-forwards: 70. Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY. Content-Type: application/sdp. Content-Length: 180. . v=0. o=- 32014 3005 IN IP4 192.168.1.102 <http://192.168.1.102>. s=-. c=IN IP4 192.168.1.102 <http://192.168.1.102>. t=0 0. m=audio 13456 RTP/AVP 18 0 8 2 4 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. . # U 2005/04/15 02:19:16.276216 10.3.0.221:5060 <http://10.3.0.221:5060> -> 69.35.46.178:5060 <http://69.35.46.178:5060> SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.1.102:5060 <http://192.168.1.102:5060> ;branch=z9hG4bK3915905414;rport=5060;received=69.35.46.178<http://69.35.46.178> . From: Test User <sip:3215591439@sipdev.mycompany.net>;tag=1465261089. To: <sip:8187768080@sipdev.myvox.net>. Call-ID: 2671219156(a)192.168.1.102. CSeq: 101 INVITE. Content-Length: 0. . # U 2005/04/15 02:19:16.792795 69.35.46.178:5060 <http://69.35.46.178:5060> -> 10.3.0.221:5060 <http://10.3.0.221:5060> ACK sip:8187768080@sipdev.mycompany.net SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102:5060 <http://192.168.1.102:5060> ;branch=z9hG4bK3499714875. From: Test User <sip:3215591439@sipdev.mycompany.net>;tag=1465261089. To: <sip:8187768080@sipdev.mycompany.net>;tag= cdbfe46960d7566fcbe09f199f2b328d.c2bb. Call-ID: 2671219156(a)192.168.1.102. CSeq: 100 ACK. Content-Length: 0. . # U 2005/04/15 02:19:16.912146 10.3.0.221:5060 <http://10.3.0.221:5060> -> 10.3.0.221:5060 <http://10.3.0.221:5060> ACK sip:8187768080@sipdev.mycompany.net SIP/2.0. Max-Forwards: 10. Record-Route: <sip:10.3.0.221 <http://10.3.0.221>;ftag=1465261089;lr>. Via: SIP/2.0/UDP 10.3.0.221 <http://10.3.0.221>;branch=0. Via: SIP/2.0/UDP 192.168.1.102:5060 <http://192.168.1.102:5060>;received= 69.35.46.178 <http://69.35.46.178>;branch=z9hG4bK3499714875. From: Test User <sip:3215591439@sipdev.mycompany.net>;tag=1465261089. To: <sip:8187768080@sipdev.mycompany.net>;tag= cdbfe46960d7566fcbe09f199f2b328d.c2bb. Call-ID: 2671219156(a)192.168.1.102. CSeq: 100 ACK. Content-Length: 0. P-hint: Local Destination.

3 5

RE: [Serusers] SER hung - Log Information?
by Ricardo Martinez 15 Apr '05

15 Apr '05

Interesting too. My "translate" script takes an average of 17ms to return the desired value. By HTTP requests it connect to another process in another machine, which have an "always open" connection with my database. So i don't need to open a new connection with my database every time an INVITE reach the server. BUT STILL DESPITE OF THIS, MY SER SEEMS TO FREEZE FOR SOME REASON. I have some questions. 1.-How to check if the bottle neck is in the exec_dset command?? 2.-If we decide to replace this command (exec_dset)for a function in another module, what module would be recommendable?, for example how the ENUM module could do this?. We have the "restriction" that we need to query our databes for the R-URI change, so the module that will acomplish this task needs to have connection with my database, maybe through radius, sql, etc.? 3.-You mention that you made some kind of "stress" test with your platform. You loaded it with 30000 call in one hour. How you did that? Does your exec_dset running at the same time?. Thanks. Any help will be highly appreciated. Ricardo.- > -----Mensaje original----- > De: Matt Schulte [mailto:mschulte@netlogic.net] > Enviado el: Viernes, 15 de Abril de 2005 11:20 > Para: Ricardo Martinez; serusers(a)lists.iptel.org > Asunto: RE: [Serusers] SER hung - Log Information? > > > Interesting, we also use an external process to connect to a DB and > retrieve URI replacement. We were thinking that it's running > "too slow" > and thus hanging SER, even though the logs make it appear as if the > process(es) was hanging at random points. We're in the process of > eliminating this script and switching over to ENUM or similar. When we > restarted our MySQL server it improved the query time (it was running > for 180 days heheh) and seems to have made SER hang less. How > long does > your external script take to run on average? If it's more > than a second > then that seems to be unnacceptable for SER.. > > P.S. The crappy thing is this happens over a period of time, > when I load > tested the server it ran flawlessly. Pushed 30000 calls through it in > less than an hour with no problems. The MySQL server in question is in > the same switch segment/subnet as well. > > > Matt > > -----Original Message----- > From: Ricardo Martinez [mailto:rmartinez@redvoiss.net] > Sent: Thursday, April 14, 2005 4:32 PM > To: Matt Schulte; serusers(a)lists.iptel.org > Subject: RE: [Serusers] SER hung - Log Information? > > > Hello Matt. > Thanks for your answer, i checked the post "exec_dset fork > limit?" > (http://lists.iptel.org/pipermail/serusers/2004-December/014150.html) > and i think i have the same problem. I'm using the exec_dset > command to > manipulate the RURI, we need to do this in order to add prefixes > according to certain user profiles. > > route { > if (method=="INVITE" || method=="CANCEL") { > exec_dset("/usr/local/etc/ser/translate"); > }; > > As you can see i'm using it for every INVITE and CANCEL, the > "translate" > file is a C program that connects to a DB and returns the > desired RURI. > Could this be the cause of SER freeze?. Can we have an > opinion from the > developers? Is there a way to do the same thing with maybe another > module?. For the record i'm using : > > OS : Linux Red Hat 9.0 > SER : 0.8.14 > Mediaproxy : 1.2.1 > > I would really appreciate any help here. > Thanks!! > > Best Regards, > > Ricardo.- > > > -----Mensaje original----- > > De: Matt Schulte [mailto:mschulte@netlogic.net] > > Enviado el: Jueves, 14 de Abril de 2005 16:11 > > Para: Ricardo Martinez; serusers(a)lists.iptel.org > > Asunto: RE: [Serusers] SER hung - Log Information? > > > > > > I've been posting about this for some time now,I have never > > been able to > > figure out the problem. Do you use external proccesses or anything? > > That's the only "weird" thing we really do with SER, > > everything else is > > pretty straight forward. > > > > Search back from my posts and see what you think. > > > > -----Original Message----- > > From: Ricardo Martinez [mailto:rmartinez@redvoiss.net] > > Sent: Thursday, April 14, 2005 11:16 AM > > To: 'serusers(a)lists.iptel.org' > > Subject: [Serusers] SER hung - Log Information? > > > > > > Hello list. > > I had been experimenting some weird behavior in my SER program. > A > > couple of times my SER seems to be hung. For example the > first time, > > all the REGISTER messages reaching the server were not > > challenged, even > > worst the server don't respond anything. The second time > it happened > > with the INVITE messages. The weird thing is that the SER > > process were > > still running. I don't know what could cause this problem, but my > > question is : Is there a way to log information about the "internal > > system process" to have a clue about what is happening?. I > > would really > > appreciate any help. Thanks > > > > > > Ricardo Martinez.- > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers > > >

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RE: [Serusers] SER hung - Log Information?
by Matt Schulte 15 Apr '05

15 Apr '05

Interesting, we also use an external process to connect to a DB and retrieve URI replacement. We were thinking that it's running "too slow" and thus hanging SER, even though the logs make it appear as if the process(es) was hanging at random points. We're in the process of eliminating this script and switching over to ENUM or similar. When we restarted our MySQL server it improved the query time (it was running for 180 days heheh) and seems to have made SER hang less. How long does your external script take to run on average? If it's more than a second then that seems to be unnacceptable for SER.. P.S. The crappy thing is this happens over a period of time, when I load tested the server it ran flawlessly. Pushed 30000 calls through it in less than an hour with no problems. The MySQL server in question is in the same switch segment/subnet as well. Matt -----Original Message----- From: Ricardo Martinez [mailto:rmartinez@redvoiss.net] Sent: Thursday, April 14, 2005 4:32 PM To: Matt Schulte; serusers(a)lists.iptel.org Subject: RE: [Serusers] SER hung - Log Information? Hello Matt. Thanks for your answer, i checked the post "exec_dset fork limit?" (http://lists.iptel.org/pipermail/serusers/2004-December/014150.html) and i think i have the same problem. I'm using the exec_dset command to manipulate the RURI, we need to do this in order to add prefixes according to certain user profiles. route { if (method=="INVITE" || method=="CANCEL") { exec_dset("/usr/local/etc/ser/translate"); }; As you can see i'm using it for every INVITE and CANCEL, the "translate" file is a C program that connects to a DB and returns the desired RURI. Could this be the cause of SER freeze?. Can we have an opinion from the developers? Is there a way to do the same thing with maybe another module?. For the record i'm using : OS : Linux Red Hat 9.0 SER : 0.8.14 Mediaproxy : 1.2.1 I would really appreciate any help here. Thanks!! Best Regards, Ricardo.- > -----Mensaje original----- > De: Matt Schulte [mailto:mschulte@netlogic.net] > Enviado el: Jueves, 14 de Abril de 2005 16:11 > Para: Ricardo Martinez; serusers(a)lists.iptel.org > Asunto: RE: [Serusers] SER hung - Log Information? > > > I've been posting about this for some time now,I have never > been able to > figure out the problem. Do you use external proccesses or anything? > That's the only "weird" thing we really do with SER, > everything else is > pretty straight forward. > > Search back from my posts and see what you think. > > -----Original Message----- > From: Ricardo Martinez [mailto:rmartinez@redvoiss.net] > Sent: Thursday, April 14, 2005 11:16 AM > To: 'serusers(a)lists.iptel.org' > Subject: [Serusers] SER hung - Log Information? > > > Hello list. > I had been experimenting some weird behavior in my SER program. A > couple of times my SER seems to be hung. For example the first time, > all the REGISTER messages reaching the server were not > challenged, even > worst the server don't respond anything. The second time it happened > with the INVITE messages. The weird thing is that the SER > process were > still running. I don't know what could cause this problem, but my > question is : Is there a way to log information about the "internal > system process" to have a clue about what is happening?. I > would really > appreciate any help. Thanks > > > Ricardo Martinez.- > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers >

2 1

[Serusers] FW: Struggling with simple SER+SEMS..
by Dave Bath 15 Apr '05

15 Apr '05

Hey guys.... I'm really struggling with this. It was working perfectly yesterday and now I must have made a small change and I just cannot find what I've done wrong. Bascially, this is a very very simple unauthenticated SER front end to SEMS... all I want to be able to do is use the conference facilities. Incidentally, when I say unauthenticated I mean also without registering... so all a user has to do is send an INVITE. To that end, I am trying to just redirect sip URI's beginning with 600. It seems that the initial redirect goes through... but then something screwy is happening because SER never appears to receive the ACKs correctly.. and it starts sending two sets of 200 messages.. one with no SDP descriptor and then a second batch with a valid SDP descriptor. Because of this weirdness with the ACKs, the client and SER fight for a while sending out millions of repeat 200s and ACKs. Meanwhile, because the initial redirect to SEMS worked, the conference has actually been created... The same story with BYE.. the client sends it, but SER doesn't even appear to receive it (note all the debugging log lines). In fact, there' something rather dodgy up, because I don't even get the debug lines inside the route[1] block.... It sort of points to me that perhaps I've done something very stupid with the transactions.. that for some reason they don't match up ... but I can't find it. Attached is my ser.cfg. Any help would be so very much appreciated. I don't think it's a basic connectivity problem, because as I said, a very similar file was working perfectly yesterday... I've just stupidly changed it slightly somewhere. Many many thanks in advance. Dave listen=<publicip> check_via=yes # (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 "/lib/ser/modules/mysql.so" loadmodule "/lib/ser/modules/sl.so" loadmodule "/lib/ser/modules/tm.so" loadmodule "/lib/ser/modules/rr.so" loadmodule "/lib/ser/modules/maxfwd.so" loadmodule "/lib/ser/modules/usrloc.so" loadmodule "/lib/ser/modules/registrar.so" loadmodule "/lib/ser/modules/textops.so" # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- 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 (msg:len >= max_len ) { sl_send_reply("513", "Message too big"); break; }; if (!method=="REGISTER") record_route(); if (loose_route()) { log(1,"Loose routing block"); append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { log(1,"Got a request for an extenal domain"); # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; if (uri==myself) { # switch to stateful mode: if (!t_newtran()){ sl_send_reply("500","could not create transaction"); break; }; if (method=="REGISTER") { save("location"); break; }; if (method=="ACK") { # absorb ACKs log(1,"Absorbing ACK"); break; }; lookup("aliases"); if (!uri==myself) { log(1,"Resolving external request"); append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # prevent timeout on the other side: t_reply("100","Trying - just wait a minute !"); if (method=="INVITE") { save("location"); if (uri=~"sip:600.*@") { log(1,"-> Redirecting conference to SEMS"); if(!t_write_unix("/tmp/am_sock","conference")) { t_reply("500","error contacting sems"); }; break; } ; break; } else if (method=="BYE" || method=="CANCEL") { log(1,"-> Sending BYE to SEMS"); if(!t_write_unix("/tmp/am_sock","bye")) { t_reply("500","error contacting sems"); }; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP log(1, "Attempting stateful forward..."); if (!t_relay()) { log(1,"... failure"); sl_reply_error(); }; log(1, "... success"); }

1 0

[Serusers] SER replies to CANCEL too early
by Esteban Monturus 15 Apr '05

15 Apr '05

I am installing a SIP proxy in a Linux machine in the local network of my office. I have implemented a SIP proxy with SER. The function of the SIP proxy is to let the users talk each other without knowing the IP where each one is reachable. My problem appears when the caller wants to cancel the call because the callee does not answer. Since I want the calle's phone to stop ringing, and I haven't received a final response, the caller's phone has to send a CANCEL request. When the proxy server receives the CANCEL, it replies with a 487 "Request cancelled". SER does this before forwarding the CANCEL to the callee's phone. These 487 response generated by SER makes the INVITE client transaction of the caller finish. When the INVITE server transaction finishes in the callee's phone, it sends another 487 response, which is forwarded by SER, but the caller's transaction is already finished, so it doesn't receive the corresponding ACK. I would like to know how to solve this problem, since SER should wait for the callee's phone to send the 487 and then forward this response to the caller's phone. Best regards, Esteban. ------------------------------------------------------------------------ --------------- Esteban Monturus de Carandini (mailto:emonturus@teltronic.es) Enhanced Wireless Solutions (EWS Project) TELTRONIC, S.A.U. Parque Tecnologico Walqa Edificio Uno Ctra Zaragoza N-330a, Km 556 22197 Cuarte (Huesca) - Spain Phone/Fax: +34 974 215018 http://www.teltronic.es http://ptwalqa.com ------------------------------------------------------------------------ --------------- ***** AVISO LEGAL ***** Este mensaje es solamente para la persona a la que va dirigido. Puede contener informacion confidencial o legalmente protegida. La transmision erronea de este mensaje no supone renuncia a su confidencialidad o a cualquier privilegio. Si usted ha recibido este mensaje por error, le rogamos que borre de su sistema inmediatamente el mensaje asi como todas sus copias y que notifique al remitente. No debe, directa o indirectamente, usar, revelar, distribuir, imprimir o copiar ninguna de las partes de este mensaje si no es usted el destinatario. Cualquier opinion expresada en este mensaje proviene del remitente, excepto cuando el mensaje establezca lo contrario y el remitente este autorizado para establecer que dichas opiniones provienen de TELTRONIC. En el caso de que el destinatario de este mensaje no consienta la utilizacion del correo electronico via Internet, rogamos lo ponga en nuestro conocimiento de manera inmediata. ***** DISCLAIMER ***** This message is intended exclusively for the named person. It may contain confidential, propietary or legally privileged information. No confidentiality or privilege is waived or lost by any mistransmission. If you receive this message in error, please immediately delete it and all copies of it from your system, destroy any hard copies of it and notify the sender. Your must not, directly or indirectly, use, disclose, distribute, print, or copy any part of this message if you are not the intended recipient. Any views expressed in this message are those of the individual sender, except where the message states otherwise and the sender is authorised to state them to be the views of TELTRONIC. If the addressee of this message does not consent to the use of internet e-mail, please communicate it to us immediately. ------------------------------------------------------------------------ ---------------

1 0

[Serusers] parallel fork woes
by Daniel Poulsen 15 Apr '05

15 Apr '05

Hi- I am having a problem where if I parallel fork a call, my failure_route gets called after the caller hangs up. I thought adding t_check_status("487") to the failure_route would take care of that, but it still t_relays when I don't want it to. This only seems to be a problem if it parallel forks and one of the SIP targets is not available. If I dial an alias with a single [not forked] target the t_check_status("487") works fine. Here is my code.. any thoughts? Thanks much. Dan route { <<<< Registration code snipped for brevity >>>> if(lookup("aliases")) { xlog("L_NOTICE", "%ci: alias lookup changed uri to %ru\n"); # If alias points outbound, forward it without question. if (!(uri==myself)) { xlog("L_NOTICE", "%ci: outbound alias\n"); route(4); # relay with hunt on failure break; }; }; if ( (uri=~"^sip:911@.*") | (uri=~"^sip:011[0-9]+@.*") | (uri=~"^sip:1[0-9]{10}@.*") ) { route(3); break; }; # Local location route(4); # relay with hunt on failure } /* end of initial routing logic */ route[4] { xlog("L_INFO", "r4: Relay with Hunt\n"); # If an invitation, we want to hunt on failure if(method == "INVITE") { t_on_failure("1"); # first hunt }; if (!lookup("location")) { xlog("L_NOTICE", "%ci: no location for %ru\n"); }; append_hf("P-hint: relay\r\n"); if (!t_relay()) { sl_reply_error(); break; }; } failure_route[1] { xlog("L_INFO", "failure+route+1 %ru\n"); revert_uri(); setflag(9); # Check to see if the call was cancelled if( t_check_status("487") | method == "CANCEL") { xlog("L_INFO", "t_check_status is 487\n"); break; } prefix("h1-"); if(lookup("hunt")) { # Hunt table is an exact copy of aliases xlog("L_NOTICE", "%ci: fail_r_1: hunt changed URI to %ru, relaying\n"); append_branch(); append_hf("P-hint: pt-hunt\r\n"); t_relay(); break; } else { xlog("L_NOTICE", "%ci: r8: no further hunts, giving up\n"); break; }; }

1 0

[Serusers] Rewrite To Heaader
by Eric Dean 15 Apr '05

15 Apr '05

Our VoIP service provider routes calls based upon the To field within the SIP header. While SER can easily rewrite the URI, I need to actually prepend our account code to every outgoing call with a prefix. does anyone know how to prefix the To field

5 8

[Serusers] ERROR: No Digest-Nonce Cannot perform Digest authentication
by Alex 15 Apr '05

15 Apr '05

I have installed freeradius 1.0.2 and radiusclient-ng.0.5.0 after the installation i am trying to run the test : radtest test test localhost 1649 testing123 User-Name = "test" User-Password = "test" NAS-IP-Address = MAINSIP NAS-Port = 1649 rad_recv: Access-Reject packet from host 127.0.0.1:1812, id=134, length=45 Reply-Message = "Hello, test with digest" I have file digest: --------------------------------------- User-Name = "test", Digest-Response = "631d6d73147add2f9e437f59bbc3aeb7", Digest-Realm = "testrealm", Digest-Nonce = "1234abcd" , Digest-Method = "INVITE", Digest-URI = "sip:5555551212@example.com", Digest-Algorithm = "MD5", Digest-User-Name = "test" in the debug of the radius server i receive that error: ---------------------------------------------------------------------- rad_recv: Access-Request packet from host 127.0.0.1:32842, id=115, length=56 User-Name = "test" User-Password = "test" NAS-IP-Address = 255.255.255.255 NAS-Port = 1649 modcall: entering group authorize modcall[authorize]: module "preprocess" returns ok modcall[authorize]: module "chap" returns noop rlm_eap: EAP-Message not found modcall[authorize]: module "eap" returns noop modcall[authorize]: module "digest" returns noop rlm_realm: No '@' in User-Name = "test", looking up realm NULL rlm_realm: No such realm "NULL" modcall[authorize]: module "suffix" returns noop users: Matched DEFAULT at 152 users: Matched test at 215 modcall[authorize]: module "files" returns ok modcall[authorize]: module "mschap" returns noop modcall: group authorize returns ok rad_check_password: Found Auth-Type Digest auth: type "digest" modcall: entering group authenticate ******ERROR: No Digest-Nonce: Cannot perform Digest authentication****** modcall[authenticate]: module "digest" returns invalid modcall: group authenticate returns invalid auth: Failed to validate the user. Thanks for the help

1 0

[Serusers] Another Looping Question
by Java Rockx 15 Apr '05

15 Apr '05

Hi All. Below is a snippet of an INVITE and you can see the usual "trying - your call is important to us" in the dialog. Anyhow, for some reason SER occasionally attempts to sent the ACK to itself, which causes a looping condition. The last ACK shown is the start of the looping condition. It only happens on occasion. SER should have consumed the ACK rather than attempted to forward it. Can anyone explain this? Regards, Paul IP Legend ------------------- 10.3.0.221 <http://10.3.0.221> - SER (behind a Cisco 3600 so ALG rewrote the IPs) 69.36.46.178 <http://69.36.46.178> - NATed SIP UA U 2005/04/15 02:19:16.252614 69.35.46.178:5060 <http://69.35.46.178:5060> -> 10.3.0.221:5060 <http://10.3.0.221:5060> INVITE sip:8187768080@sipdev.mycompany.net SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102:5060 <http://192.168.1.102:5060> ;branch=z9hG4bK3915905414. From: Test User <sip:3215591439@sipdev.mycompany.net>;tag=1465261089. To: <sip:8187768080@sipdev.mycompany.net>. Call-ID: 2671219156(a)192.168.1.102. CSeq: 101 INVITE. Contact: <sip:3215591439@192.168.1.102:5060>. Proxy-Authorization: Digest username="3215591439", realm=" sipdev.mycompany.net <http://sipdev.mycompany.net>", nonce="425f5e0de25efdb3f5a6125dcf2147bf30fe9177", uri=" sip:8187768080@sipdev.mycompany.net", response="61996c3bdc10990fec7e245751777516", algorithm=MD5, cnonce="ae02e47a3355bada49ce344c92f48587", qop=auth, nc=00000002. max-forwards: 70. Allow: INVITE, ACK, CANCEL, BYE, REFER, NOTIFY. Content-Type: application/sdp. Content-Length: 180. . v=0. o=- 32014 3005 IN IP4 192.168.1.102 <http://192.168.1.102>. s=-. c=IN IP4 192.168.1.102 <http://192.168.1.102>. t=0 0. m=audio 13456 RTP/AVP 18 0 8 2 4 101. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. . # U 2005/04/15 02:19:16.276216 10.3.0.221:5060 <http://10.3.0.221:5060> -> 69.35.46.178:5060 <http://69.35.46.178:5060> SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 192.168.1.102:5060 <http://192.168.1.102:5060> ;branch=z9hG4bK3915905414;rport=5060;received=69.35.46.178<http://69.35.46.178> . From: Test User <sip:3215591439@sipdev.mycompany.net>;tag=1465261089. To: <sip:8187768080@sipdev.mycompany.net>. Call-ID: 2671219156(a)192.168.1.102. CSeq: 101 INVITE. Content-Length: 0. . # U 2005/04/15 02:19:16.792795 69.35.46.178:5060 <http://69.35.46.178:5060> -> 10.3.0.221:5060 <http://10.3.0.221:5060> ACK sip:8187768080@sipdev.mycompany.net SIP/2.0. Via: SIP/2.0/UDP 192.168.1.102:5060 <http://192.168.1.102:5060> ;branch=z9hG4bK3499714875. From: Test User <sip:3215591439@sipdev.mycompany.net>;tag=1465261089. To: <sip:8187768080@sipdev.mycompany.net>;tag= cdbfe46960d7566fcbe09f199f2b328d.c2bb. Call-ID: 2671219156(a)192.168.1.102. CSeq: 100 ACK. Content-Length: 0. . # U 2005/04/15 02:19:16.912146 10.3.0.221:5060 <http://10.3.0.221:5060> -> 10.3.0.221:5060 <http://10.3.0.221:5060> ACK sip:8187768080@sipdev.mycompany.net SIP/2.0. Max-Forwards: 10. Record-Route: <sip:10.3.0.221 <http://10.3.0.221>;ftag=1465261089;lr>. Via: SIP/2.0/UDP 10.3.0.221 <http://10.3.0.221>;branch=0. Via: SIP/2.0/UDP 192.168.1.102:5060 <http://192.168.1.102:5060>;received= 69.35.46.178 <http://69.35.46.178>;branch=z9hG4bK3499714875. From: Test User <sip:3215591439@sipdev.mycompany.net>;tag=1465261089. To: <sip:8187768080@sipdev.mycompany.net>;tag= cdbfe46960d7566fcbe09f199f2b328d.c2bb. Call-ID: 2671219156(a)192.168.1.102. CSeq: 100 ACK. Content-Length: 0. P-hint: Local Destination.

1 0

[Serusers] DNS Lookup and other issue
by info＠beeplove.com 15 Apr '05

15 Apr '05

If I setup an UA with URL sip:beeplove@mydomain.org my client is going to find its sip server from DNS SRV record, right? If I have these in nameserver _sip._udp SRV 0 0 5060 sip.mydomain.org. _sip._tcp SRV 0 0 5060 sip.mydomain.org. is that mean, sip.mydomain.org is my registar server? or proxy server or both? And If my UA have an option to specify proxy and registar server, UA is not going to lookup SRV record, right? Please correct me someone, if I am understading wrong. Thanks Mohammad -------------------------------------------------------------------- mail2web - Check your email from the web at http://mail2web.com/ .

2 1

[Serusers] Not ANOTHER silly NAT question...
by Rod Bacon 15 Apr '05

15 Apr '05

Well.. it probably is. I have had mixed success with SER 0.8.14 and RTPPROXY. I have RADIUS auth, accounting, MYSQL, blah, blah, blah, all working.... My question, however, surrounds NAThelper configuration. If I leave all clients behind NATs to attempt registration with private IPs, fix the IPs with NAThelper and proxy the streams, then everything works OK. If I inroduce a client that is on a LIVE IP address, I only get one-way audio to a natted client (the natted client can't hear the live client). I assume that what is happening is that the live client is attempting to send it's audio direct to the natted client, instead of via the proxy. I figure that I've missed something silly somewhere in the config, but I'm darned if I know where. I won't bother posting my entire config (unless specifically requested), rather I'd like a general pointer as to where I should be looking. Thanks in advance.

6 7

[Serusers] Ser Redirect Server !!!
by Felipe Martins 15 Apr '05

15 Apr '05

Hi everybody, I finally managed to configure and make 2 Ser Servers talk to each other, each one with a registered soft phone or hard phone, they are talking with no delay and nearly no noise. I've made a redirect configuration at the routing logics and for me it worked. I found a similar configuration at the link: (http://www.voip-info.org/wiki-SER+example+redirect), but it's totally diferent from what I've done. This is what i've done. -------------------------------------------------------------------------- if (method=="REGISTER") { log(1, "REGISTER message received\n"); # Uncomment this if you want to use digest authentication if (!www_authorize("192.168.4.11", "subscriber")) { www_challenge("192.168.4.11", "0"); break; }; save("location"); break; }; # Repassando Chamadas Internacionais para Asterisk if (uri=~"^sip:[2][0-9].*@") { log(1, "Forwarding to Another Gateway\n"); rewritehost("192.168.4.10"); forward(192.168.4.10,5060); t_relay(); break; } ------------------------------------------------------------------------- Is there some error in this configuration ? -- Felipe Martins Mundivox Communications Tecnologia e Projetos fmartins(a)mundivox.com Tel.: +55 +21 +3820 8839 Cel.: +55 +21 +9823 8602 Fax.: +55 +21 +3820 8844 www.mundivox.com

1 0

[Serusers] Strange problems
by Alex 15 Apr '05

15 Apr '05

Hi guys thanks for any help, i got a lot of useful information from this list. I will try to explain my problem. i have 2 linux servers : 1- server Red Hat Linux release 9 ser + freeradius 0.9.3 + radius client 2-server Red Hat Enterprise Linux AS release 3 ser + freeradius 1.2 + radius client 4.8 I trying to register my ip phone through ser + radius . On the 1 server it's working fine i receiving register requests, i can see that in the logs, and the request is going through the authentication process. On the 2 server i receiving register requests, but it's not going to the radius authentication, i don't see anything inside the radius logs. Here is the ngrep on port 5060: from the 2 server ------------------------------------------------------------------------------------------------- U telephoneip:10739 -> xxx.xxx.xxx.xxx:5060 REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP telephoneip:10000;branch=z9hG4bK98514c3b052d7df6..From: "Test Alex" < sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=50673f1baca1958c..To: <sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>..Contact: <sip :Phonenumber@telephoneip:10000;user=phone>..Call-ID: 1cff1b8955b8fa5c@10.0.0.4..CSeq: 106 REGISTER..Expires: 3600..User-Agent : Grandstream HT286 1.0.5.18..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Lengt h: 0.... # U xxx.xxx.xxx.xxx:5060 -> telephoneip:10000 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP telephoneip:10000;branch=z9hG4bK98514c3b052d7df6..From: "Test Alex" <sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=50673f1baca1958c..To: <sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=b27e1a1d33761e85846fc9 8f5f3a7e58.f894..Call-ID: 1cff1b8955b8fa5c@10.0.0.4..CSeq: 106 REGISTER..WWW-Authenticate: Digest realm="xxx.xxx.xxx.xxx", nonc e="425e3ac34dc9509392435c11fb260f41420049c7"..Server: Sip EXpress router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 xxx.xxx.xxx.xxx:5060 "Noisy feedback tells: pid=1912 req_src_ip=telephoneip req_src_port=10739 in_uri=sip:xxx.xxx.xxx.xxx out_uri=sip:xxx.xxx.xxx.xxx via_cnt==1".... # U telephoneip:10740 -> xxx.xxx.xxx.xxx:5060 REGISTER sip:xxx.xxx.xxx.xxx SIP/2.0..Via: SIP/2.0/UDP telephoneip:10000;branch=z9hG4bK98514c3b052d7df6..From: "Test Alex" < sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=50673f1baca1958c..To: <sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>..Contact: <sip :Phonenumber@telephoneip:10000;user=phone>..Call-ID: 1cff1b8955b8fa5c@10.0.0.4..CSeq: 106 REGISTER..Expires: 3600..User-Agent : Grandstream HT286 1.0.5.18..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Lengt h: 0.... # U xxx.xxx.xxx.xxx:5060 -> telephoneip:10000 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP telephoneip:10000;branch=z9hG4bK98514c3b052d7df6..From: "Test Alex" <sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=50673f1baca1958c..To: <sip:Phonenumber@xxx.xxx.xxx.xxx;user=phone>;tag=b27e1a1d33761e85846fc9 8f5f3a7e58.f894..Call-ID: 1cff1b8955b8fa5c@10.0.0.4..CSeq: 106 REGISTER..WWW-Authenticate: Digest realm="xxx.xxx.xxx.xxx", nonc e="425e3acb812b5b2e8aa023e3fcffc618dc4cf661"..Server: Sip EXpress router (0.8.14 (i386/linux))..Content-Length: 0..Warning: 392 xxx.xxx.xxx.xxx:5060 "Noisy feedback tells: pid=1885 req_src_ip=telephoneip req_src_port=10740 in_uri=sip:xxx.xxx.xxx.xxx out_uri=sip:xxx.xxx.xxx.xxx via_cnt==1".... # here is the debug from ser : --------------------------------------------------------------------------------------------- 14(1036) parse_headers: flags=-1 14(1036) check_via_address(62.219.158.191, 62.219.158.191, 1) 14(1036) DEBUG:destroy_avp_list: destroing list (nil) 14(1036) receive_msg: cleaning up 9(1012) SIP Request: 9(1012) method: <REGISTER> 9(1012) uri: <sip:xxx.xxx.xxx.xxx> 9(1012) version: <SIP/2.0> 9(1012) parse_headers: flags=1 9(1012) Found param type 232, <branch> = <z9hG4bKfc5751413c832e6d>; state=16 9(1012) end of header reached, state=5 9(1012) parse_headers: Via found, flags=1 9(1012) parse_headers: this is the first via 9(1012) After parse_msg... 9(1012) preparing to run routing scripts... 9(1012) REGISTER: Authenticating user 9(1012) parse_headers: flags=4 9(1012) end of header reached, state=9 9(1012) DEBUG: get_hdr_field: <To> [45]; uri=[sip:phonenumber@xxx.xxx.xxx.xxx;user=phone] 9(1012) DEBUG: to body [<sip:phonenumber@xxx.xxx.xxx.xxx;user=phone> ] 9(1012) parse_headers: flags=4096 9(1012) get_hdr_field: cseq <CSeq>: <103> <REGISTER> 9(1012) DEBUG: get_hdr_body : content_length=0 9(1012) found end of header 9(1012) pre_auth(): Credentials with given realm not found 9(1012) REGISTER: challenging user 9(1012) build_auth_hf(): 'WWW-Authenticate: Digest realm="xxx.xxx.xxx.xxx", nonce="425e063022afc1142ed6730d46da41692ff3ed57" Daniel and Alex send some explanations i really appreciate that, but i don't know where to start to fix that problem, i am trying to understand why on the first server it's working , but on the second it's not working . both of them have the same configuration only the versions are different. I tried to connect different ip phone and it's sending the packets to the 2 server and it's working, but the 1 ip phone not working with the 2 server, but with the 1 one it's working fine, i just changing the sip proxy ip. Thanks for any help.

2 3

[Serusers] B2BUA and NAT.
by Amos Nungu 15 Apr '05

15 Apr '05

Hi. I can't find enough info regarding B2BUA, and how it works. At least I know that it is not scalling and it breaks SIP peer2peer functionality. BUT, it is said that it can handle NAT, HOW? 1.Does it solve NAT because it stays in the middle and proxy everything (+ media) or 2.can it be involved in the call only when NAT is detected like rtp proxy ? 3. iF 2, THEN how does it work in corporation with SER or any SIP Server? 4. Which situation will rtp proxy (rtpproxy/mediaproxy) fail to solve NAT issues NB. I just need to expand my knowledge regarding options. I'm using rtpproxy, it works fine in every respect only if I start it in foreground Regards.

2 3

[Serusers] Append prefix to RURI via AVP
by Martin Koenig 15 Apr '05

15 Apr '05

Hello, is it possible to add a prefix to the RURI that is based on an AVP, similar to prefix("x")? Scenario is the following: Each user would have an assotiated avp called cc and ac. Depending on incoming RURI, Ser would first add the area code avp, then the country code avp. Is this possible? With best regards, Martin Koenig

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