sr-users September 2007

sr-users@lists.kamailio.org

158 participants
248 discussions

[OpenSER-Users] XCAP setting problem?
by Adrian Georgescu 05 Sep '07

05 Sep '07

You may obtain the XCAP server and the configurations from http:// openxcap.org The latest OpenSER svn version works with it and the sample configuration. Adrian >>>>> I don't know whether or not the xcap_server is set correct. There is no external XCAP server integrated. BTW, I know presence support XCAP function, does it means this module has already included a XCAP server? Thank You! Kevin

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[OpenSER-Users] XCAP setting problem?
by KevinKinnan 05 Sep '07

05 Sep '07

The PUBLISH request works when using the following configurations(no XCAP): loadmodule "sl.so" loadmodule "mysql.so" loadmodule "presence.so" loadmodule "presence_xml.so" modparam("presence_xml", "db_url", "mysql://openser:openserrw@localhost:3306/openser") modparam("presence_xml", "force_active", 1) modparam("presence_xml", "pidf_manipulation", 1) It doesn't if starting XCAP by updating these, the server can run properly, but cann't handle PUBLISH. loadmodule "xcap_client.so" modparam("presence_xml", "xcap_table", "xcap_xml") modparam("presence_xml", "force_active", 0) modparam("presence_xml", "xcap_server", "localhost:5065") I don't know whether or not the xcap_server is set correct. There is no external XCAP server integrated. BTW, I know presence support XCAP function, does it means this module has already included a XCAP server? Thank You! Kevin _________________________________________________________________ Windows Live Spaces 中最年轻的成员！ http://miaomiaogarden2007.spaces.live.com/

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[OpenSER-Users] mi_xmlrpc compilation issues
by Purna Chandar M 05 Sep '07

05 Sep '07

Hi, Im trying to compile mi_xmlrpc module in openser but could not succeed. The following errors are displayed mi_xmlrpc.c: In function 'xmlrpc_process': mi_xmlrpc.c:161: error: 'xmlrpc_server_abyss_rpc2_handler' undeclared (first use in this function) mi_xmlrpc.c:161: error: (Each undeclared identifier is reported only once mi_xmlrpc.c:161: error: for each function it appears in.) mi_xmlrpc.c:167: error: 'xmlrpc_server_abyss_default_handler' undeclared (first use in this function) Im using Suse 10.1; The libraries xmlrpc-c-0.9.10-36, libxml-1.8.17-385, libxml2-devel-2.6.23-13, xmlrpc-c-devel-0.9.10-36 are properly installed. Few discussions on the mailing list concluded that the module mi_xmlrpc compiles well on Debian without any dependency issues. Is it true?? Can somebody tell me whether Im missing any dependencies?? Please let me know how to successfully compile this module. Thanks and regards, Purna Chandar M

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Re: [OpenSER-Users] mi_xmlrpc compilation issues
by Kriti Wadhwa 05 Sep '07

05 Sep '07

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[OpenSER-Users] NatPing stop working
by Marcello Lupo 05 Sep '07

05 Sep '07

Hi to all, i have openser 1.2.1 active from some months.. Suddenly on yesterday the natping stop to work. I have 140 contacts in user location almost all of them natted. There is some limit to user in location table at which the ping stops or a particular server load? There is a way to have something in the logs about it?? I didn't found anything in the logs. This is my config: modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) With a restart of openser all come back to work but i'm afraid if it will happen again? Thanks for help, Bye, Marcello

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[OpenSER-Users] Opnser rejecting BYE packet
by Dan 05 Sep '07

05 Sep '07

Scenario 1: U 212.34.169.220:5060 -> 78.45.146.34:5060 BYE sip:test@78.45.146.34 SIP/2.0. Route: <sip:78.45.146.34;ftag=aa313a3c;lr>. CSeq: 2 BYE. Via: SIP/2.0/UDP 212.34.169.220:5060. From: <sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. Call-ID: 8246bf729f62a754. To: 58.68.96.11<sip:test@78.45.146.34>;tag=aa313a3c. Content-Length: 0. # U 78.45.146.34:5060 -> 212.34.169.220:5060 SIP/2.0 404 User Not Found. CSeq: 2 BYE. Via: SIP/2.0/UDP 212.34.169.220:5060. From: <sip:1217773456@78.45.146.34>;tag=03095807065938997270316869. Call-ID: 8246bf729f62a754. To: 58.68.96.11<sip:test@78.45.146.34>;tag=aa313a3c. Server: OpenSer (1.0.1 (i386/linux)). Content-Length: 0. Warning: 392 78.45.146.34:5060 "Noisy feedback tells: pid=24525 req_src_ip=78.45.146.34 req_src_port=5060 in_uri=sip:78.45.146.34;ftag=aa313a3c;lr out_uri=sip:78.45.146.34;ftag=aa313a3c;lr via_cnt==2". Scenario 2: U 59.176.78.110:50496 -> 210.68.65.78:5060 BYE sip:210.68.65.78 SIP/2.0. Route: <sip:@209.67.171.10;transport=udp>. Via: SIP/2.0/UDP 59.176.78.110:50496. From: sip:user1@210.68.65.78;tag=4154163258. To: <sip:12818495027@210.68.65.78;user=phone>;tag=3005360702044301069847989. Call-ID: 4202237689(a)192.168.1.156. CSeq: 3 BYE. User-Agent: Cisco ATA v2.15 ata18x (020927a). Content-Length: 0. . # U 210.68.65.78:5060 -> 59.176.78.110:50496 SIP/2.0 478 Unresolvable destination (478/TM). Via: SIP/2.0/UDP 59.176.78.110:50496. From: sip:user1@210.68.65.78;tag=4154163258. To: <sip:12818495027@210.68.65.78;user=phone>;tag=3005360702044301069847989. Call-ID: 4202237689(a)192.168.1.156. CSeq: 3 BYE. Server: OpenSer (1.0.1 (i386/linux)). Content-Length: 0. Warning: 392 210.68.65.78:5060 "Noisy feedback tells: pid=2144 req_src_ip=59.176.78.110 req_src_port=50496 in_uri=sip:210.68.65.78. In both cases BYE is not getting accounted for. Can anyone help me on this ?? Regards Dan

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Re: [OpenSER-Users] Broken "BYE" returned from Asterisk on TLS implementation ?
by David Loh 05 Sep '07

05 Sep '07

Hi, Arrggghh .. that's one of my attempts to eliminate the broken "BYE" problem... that's ngrep was captured when I set "modparam("rr", "enable_double_rr", "0");", I've paste another ngrep to http://pastebin.ca/674450, this time the double RR header is enabled. And I've posted my .cfg to http://pastebin.ca/Nx0Ss4Fd (key to decrypt the post is "openser"). Even though double RR header is enabled, but for BYE it's still doesn't process properly :( For the .cfg file line #130 onward, I did tried t_relay, forward and force_send_socket, but none of this will do the trick (force_send_socket was complaining TLS error due to missing certificate (?) ) Would appreciate if anyone could enlighten me why is this happen ? Thanks, David Loh Klaus Darilion wrote: > But the INVITE you posted at http://pastebin.ca/673392 also has only > one Record-Route header. > > regards > klaus > > David Loh schrieb: >> Hi, >> >> Yea, OpenSER proxy was add 2 record-route header for the INVITE/ACK >> ...but when asterisk disconnected the call and send BYE back to OpenSER, >> the TLS RR header wasn't present, the only 2 RR header was >> "SIP/2.0/UDP <OpenSER_IP>" and "SIP/2.0/UDP <Client_WAN_IP>" .... >> I'm puzzled ... is there any command to 'fix' this? >> >> >> Regards, >> David Loh >> >> Klaus Darilion wrote: >>> The openser proxy should add 2 record-route header (TLS and UDP = >>> double record route). This is why it does not work. >>> >>> regards >>> klaus >>> >>> David Loh schrieb: >>>> Hi All, >>>> >>>> Greeting. >>>> >>>> I've been struggle with OpenSER TLS implementation for more than a >>>> week, since I've ported from UDP to TLS, everything work fine >>>> except the "BYE" request from Asterisk (loose route), my >>>> implementation was something like below: >>>> >>>> [Client] --> [Router] --> [Internet] --> [SIP] --> [Asterisk] >>>> >>>> My OpenSER.cfg already configured to listen on two port which is :- >>>> "tls:eth0:5061" and "udp:eth0:5060", client make p2p or PSTN (or >>>> even voicemail) having no problem, >>>> but when the callee disconnect the call, caller will never get hang >>>> up :( >>>> >>>> I've attached my ethereal trace/ngrep to pastebin, >>>> http://pastebin.ca/673392 >>>> >>>> Wondering if anyone can help me with the broken "BYE" that returned >>>> from Asterisk ? >>>> Line #131, supposedly this line should have contain 2 Via header, >>>> one was "SIP/2.0/UDP" and another "SIP/2.0/TLS", >>>> but somehow the TLS via header was gone !! (compare to previous ACK >>>> (Line #117) /INVITE (Line #51). >>>> Due to the missing TLS via header, OpenSER log file was complaining >>>> "protocol/port mis-match". >>>> >>>> The last BYE request (Line #256) is actually firing from Client, >>>> which contain the "TLS" via. >>>> >>>> >>>> I've even tried "force_send_socket" to port 5061 (instead of 5060) >>>> from loose route, but it complaining TLS certificate error, >>>> since Asterisk doesn't support TLS natively, I've no clue why is >>>> the ACK/INVITE/CANCEL work but not BYE. >>>> if (loose_route) { >>>> .... >>>> if(is_method("BYE")) { force_send_socket(IP:5061); } >>>> } >>>> >>>> >>>> Has any one gone through of this kinda OpenSER over TLS + Asterisk >>>> setup, >>>> I'm really appreciate if you can share your experience with me, or >>>> pin point what's the mistakes I made here. >>>> >>>> Thanks in advance. >>>> >>>> Regards, >>>> David Loh >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Users mailing list >>>> Users(a)openser.org >>>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >> >> > >

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[OpenSER-Users] OpenSER with PostgreSQL
by Cliff Flood 04 Sep '07

04 Sep '07

Hi, I have an Ubuntu 6.06 install with PostgreSQL 8.1 and OpenSER 1.2.2 which I am testing at the moment. I have tested OpenSER with MySQL and it worked fine but when when I want to switch to PostgreSQL I'm having some difficulty. I have used the postgressqldb.sh script to initialise the database and it has completed without error. When I start OpenSER I get the following lines in my PostgreSQL logs: LOG: connection received: host=127.0.0.1 port=37610 LOG: connection authorized: user=openser database=openser LOG: SSL SYSCALL error: EOF detected LOG: could not receive data from client: Connection reset by peer LOG: unexpected EOF on client connection I have looked at OpenSER's logs with debugging turned on but I don't see anything that looks like an error but I would be happy to provide them. `openserctl moni` returns: ERROR: Error opening OpenSER's FIFO /tmp/openser_fifo ERROR: Make sure you have the line 'modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo")' in your config ERROR: and also have loaded the mi_fifo module. My openser.cfg looks like this: debug=9 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=4 listen=192.168.20.151 dns=no rev_dns=no port=5060 mpath="/usr/local/lib/openser/modules/" loadmodule "mysql.so" loadmodule "postgres.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "auth.so" loadmodule "auth_db.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("usrloc", "db_mode", 2) modparam("usrloc", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "db_url", "postgres://openser:openserrw@localhost/openser") modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # 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); }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); # if you have some interdomain connections via TLS #if(uri=~"@tls_domain1.net") { # t_relay("tls:domain1.net"); # exit; #} else if(uri=~"@tls_domain2.net") { # t_relay("tls:domain2.net"); # exit; #} route(1); }; # 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("host.doman.tld", "subscriber")) { # www_challenge("host.domain.tld", "0"); # exit; # }; save("location"); exit; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; 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(); }; exit; } Any and all help appreciated. -- Cliff Flood Systems Administrator Jazinga Inc.

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[OpenSER-Users] Strange Provider SIP - DSP fields in trying
by Marc LEURENT 04 Sep '07

04 Sep '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 I have a SIP provider: telecomitalia that use a cirpack! The problem is that the cirpack sends TRYING and RINGING packets with SDP fields! (I don't know if it follows RFC...) When a natted contact place a call, I use a rtp_proxy, but only the 200ok with session description is modified. Both TRYING/RINGING with session description Connection Address/port are not replaced! OpenWengo understand the modification and change the IP where it sends RTP from the one in TRYING/RINGING SDP to the one in 200ok (proxy address because of force_rtp_proxy()) But other softphones like Twinkle don't understand it and still send RTP to the first IP! Do you know how I can delete or modify TRYING/RINGING SDP? Thanks Below an example of a RINGING with SDP # U 212.129.6.65:5060 -> 88.191.45.91:5060 SIP/2.0 180 Ringing. Allow: UPDATE,REFER. Call-ID: 866489712(a)192.168.95.47. Contact: <sip:212.129.6.65:5060>. Content-Type: application/sdp. CSeq: 21 INVITE. From: "Marc LEURENT" <sip:mleurent@sd-7501.dedibox.fr>;tag=542924903. Record-Route: <sip:88.191.45.91;lr;ftag=542924903>. Server: Cirpack/v4.39a (gw_sip). To: <sip:0614730696@sd-7501.dedibox.fr>;tag=01-07627-0003a50a-10152ba67. Via: SIP/2.0/UDP 88.191.45.91;received=88.191.45.91;branch=z9hG4bK2cf4.8d35c996.0,SIP/2.0/UDP 192.168.95.47:5060;received=81.57.0.22;rport=64726;branch=z9hG4bK20204963;xxx-nat-type=sym. Content-Length: 303. . v=0. o=cp10 118890098142 118890098144 IN IP4 212.129.46.35. s=SIP Call. c=IN IP4 212.129.47.194. t=0 0. m=audio 32896 RTP/AVP 8. b=AS:64. a=rtpmap:8 PCMA/8000/1. a=ptime:10. a=sendrecv. m=video 65534 RTP/AVP 34 31. a=rtpmap:34 H263/90000/1. a=fmtp:34 . a=rtpmap:31 H261/90000/1. a=fmtp:31 . a=inactive. -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.7 (GNU/Linux) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFG3TdoqjpLE0HiOBYRAjnkAKCgzB4p89oTKhxuvZcLlklTme9w3ACeP+TW FI+LkwgbaJ1XJTIm3FUSMTg= =Cp53 -----END PGP SIGNATURE-----

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[OpenSER-Users] MediaProxy 1.9.0
by Mike O'Connor 04 Sep '07

04 Sep '07

Hi Guys I upgraded my mediaproxy to 1.9.0 and found out sometime later that the data usage update at call end seems to have stopped. I'm using direct mysql (not radius) for the end of call updates. No error message in the logs and a manual test of mysql show nothing wrong. Any idea. Mike

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sr-users September 2007