sr-users June 2004

sr-users@lists.kamailio.org

176 participants
387 discussions

by AJ Grinnell

Can someone please help before I lose my mind here. I have all calls pointing to a cisco as5350. When the calls come in from the cisco, they are routed through ser and back out the cisco, which in turn gets routed back to the cisco by our PRI provider. This keeps going in a loop untill all channels are busied out. I have tried every way I can think of to block calls incoming from the IP of the cisco, but nothing works. What am I doing wrong here? simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) listen=x listen=127.0.0.1 alias=x alias=x alias=x check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/ser_fifo" sip_warning=yes syn_branch=no server_signature=yes reply_to_via=no # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/etc/lib/ser/modules/sl.so" loadmodule "/etc/lib/ser/modules/tm.so" loadmodule "/etc/lib/ser/modules/rr.so" loadmodule "/etc/lib/ser/modules/maxfwd.so" loadmodule "/etc/lib/ser/modules/usrloc.so" loadmodule "/etc/lib/ser/modules/registrar.so" loadmodule "/etc/lib/ser/modules/auth.so" loadmodule "/etc/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) modparam("usrloc", "db_mode", 2) modparam("auth_db", "db_url", "sql://ser:heslo@localhost/ser") # -- auth params -- modparam("auth_db", "db_url", "sql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", yes) 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; }; 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("x.x.x.x", "subscriber")) { www_challenge("x.x.x.x", "0"); break; }; save("location"); break; }; if (method=="INVITE") { if (!(uri=="sip:.*@10\.200\.20\.10")){ #Have also tried if (!uri=~"sip:.*@10\.200\.20\.10") if (uri=~"^sip:[2-9]......@.*") { # and if (!ip_addr==10\.200\.20\.10") # attempt handoff to PSTN. log( "Forwarding to PSTN\n" ); sethost("10.200.20.10"); }; }; }; }; # 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(); }; } AJ Grinnell Network Operations Technician CRT/ Arialink Broadband 1223 Turner Street, Suite A Lansing, MI 48906 517.346.5041 517.492.1321 direct

20 years, 5 months

[Serusers] Asterisk supports now Remote-Party-ID and redirect 30X

by Adrian Georgescu

For those using * as a SIP/PSTN gateway, Since last night Remote-Party-ID support has been added to Asterisk cvs. Also SIP redirect support 30X should be fixed now. Regards, Adrian

20 years, 5 months

[Serusers] SUBSCRIBE method problem!

by andayesh

I want to add presence support to my UA. It is implemented using RTC 1.2. It's not functioning I think the problem exist in the contact header field, which RTC put maddr part to this field's value. Can anybody help me? Register time [Mon Jun 21 12:17:36 2004] method <REGISTER> r-uri <wcpserver.digitalclone.net> call-id <068595f62e1b43c9abcee7a76b6f8157(a)10.10.10.2 > cseq <1> contact header <<sip:10.10.10.2:9231;transport=tcp>; methods="INVITE, MESSAGE, INFO, SUBSCRIBE, OPTIONS, BYE, CANCEL, NOTIFY, ACK, REFER"> >From uri <sip:n21@wcpserver.digitalclone.net> >From tag <cdd3f5d9a11c4c71971a4a51efe52696> To uri <sip:n21@wcpserver.digitalclone.net> To tag <<null>> SIP message id <26> IP source address <10.10.10.2> time [Mon Jun 21 12:19:31 2004] method <SUBSCRIBE> r-uri <n22(a)wcpserver.digitalclone.net> call-id <09f299e5576b46bf95104a933c891fb8(a)10.10.10.2 > cseq <1> contact header <<sip:n21@wcpserver.digitalclone.net:9231;maddr=10.10.10.2;transport=tcp>> >From uri <sip:n21@wcpserver.digitalclone.net> >From tag <52d5cc9b2e714761b6e759a47f750f15> To uri <sip:n22@wcpserver.digitalclone.net> To tag <<null>> SIP message id <27> IP source address <10.10.10.2> ERROR: tcpconn_connect: connect: (111) Connection refused ERROR: tcp_send: connect failed msg_send: ERROR: tcp_send failed t_uac: Attempt to send to 'sip:n21@wcpserver.digitalclone.net:9231;maddr=10.10.10.2;transport=tcp' failed handle_subscription successfuly handle_subscription exit

20 years, 5 months

[Serusers] Asterisk supports now Remote-Party-ID and redirect 30X

by AG Projects support

For those using * as a SIP/PSTN gateway, Since last night Remote-Party-ID support has been added to Asterisk cvs. Also SIP redirect support 30X should be fixed now. Regards, Adrian

20 years, 5 months

R: [Serusers] How to intercept reply messages

by Zappasodi Daniele

Thanks for the answer. Now I am able to intercept the reply but I am not still able to stop these messages. Cn I change the type of the reply? For example I could substitute a "302 moved Temporarily" with a "403 Forbidden" in order to avoid the call divert. Daniele -----Messaggio originale----- Da: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Inviato: mercoledì 16 giugno 2004 17.29 A: Zappasodi Daniele Cc: serusers(a)lists.iptel.org Oggetto: Re: [Serusers] How to intercept reply messages you have to set up a reply route: route{ .... t_on_reply("3"); .... }; onreply_route[3]{ if ( (status=~"200" || status=~"183") ) { force_rtp_proxy(); append_hf("P-RTP-Proxy: YES\r\n"); }; log(1, "leaving onreply_route(1)\n"); } regards, klaus Zappasodi Daniele wrote: > Hi all, > I need to intercept the reply messages ("302 Moved Temporarily", "200 > Accepted", "405 Method not allow" etc. ) and execute different actions > for different status codes (for example I could block all the "Moved > Temporally" messages or, in the Failure_route block, I could route the > calls to different destinations for each different failure reason : > Busy, no response, not available ...). > > How can I do this thing with the ser.cfg file? > > I have tried with the "method" operand and with the "search" function > but I'm not able to intercept the reply messages with these functions. > > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

20 years, 5 months

[Serusers] problem with serweb

by Igor Cahoj

Hello I've installed ser with mysql module ... then I've run the sh script to create the ser DB with all required tables. I've modified the config.php but still can't make serweb to work. The problem is with field names in subscriber table. The script created table subscriber with field names like USER_ID or REALM but the login script has differnet fieldnames. Instead of USER_ID USERNAME and instead of REALM DOMAIN. What should I do to make serweb to work? Or am I doing something terribly wrong??? Igor

20 years, 5 months

[Serusers] Weird problem when, restarting ser.

by Atle Samuelsen

Hi Guys and Girls. I ran into a problem a couple of days ago, when trying to restart ser. The scenario was that ser was sending out ALOT of 500 "I'm sorry... ". I found it then best to restart ser, to see if it stoped. When I then was going to start ser again (via ser's rc scripts) It did'nt want to start. Here's the messages in /var/log/messages. (First it looked like ser started normaly) VOIP /usr/local/sbin/ser[1302]: mod_init(): Database connection opened successfuly VOIP ser: ser startup succeeded VOIP /usr/local/sbin/ser[1302]: convert_row(): No memory left VOIP /usr/local/sbin/ser[1302]: convert_rows(): Error while converting row #6256 VOIP /usr/local/sbin/ser[1302]: convert_result(): Error while converting rows VOIP /usr/local/sbin/ser[1302]: get_result(): Error while converting result VOIP /usr/local/sbin/ser[1302]: preload_udomain(): Error while doing db_query VOIP /usr/local/sbin/ser[1302]: register_udomain(): Error while preloading domain 'location' VOIP /usr/local/sbin/ser[1302]: domain_fixup(): Error while registering domain VOIP /usr/local/sbin/ser[1302]: ERROR: fix_expr : fix_actions error To get ser upandRunning again, I had to delete all the entries in mysql::ser::location. Have you guys ever seen anything like this before? - Atle

20 years, 5 months

[Serusers] CC-Diversion

by Steve Blair

Hello: Can anyone speak to how the CC-Diversion field is used by Cisco gateways? It is my understanding that adding this header field in SER will result in the Cisco gateway setting the calling party id to the value in the CC-Diversion header. Is this correct? I'm asking because we have an Octel 350 voice mail system that I would like to use for mailbox for IP phone users. The 350 expects a physical SMDI circuit for the call signaling path and a different circuit for the message body path. It would be great to allow the SIP signaling exiting our IP cloud through a Cisco gateway to be able to tell the Octel system to which subscriber mailbox the message should be delivered. Thanks,Steve -- ISC Network Engineering The University of Pennsylvania 3401 Walnut Street, Suite 221A Philadelphia, PA 19104 voice: 215-573-8396 215-746-7903 fax: 215-898-9348 sip:blairs@upenn.edu

20 years, 5 months

[Serusers] get_contact_uri: Error while parsing Contact body

by Carlo Pires

Hi, I'm getting this message when my grandstream ht286 trys to register. It seens ht286 isn't sending authorization header field to SER. Or, ht286 sending strange contact (*) disturbs SER to process REGISTER requests ? This message starts few days ago. As I changed my ser box to last cvs version these days I'm confuse if it's a grandstream's problem or a ser's problem. ngrep logs: attached U 193.126.48.33:5060 -> 200.193.194.142:5060 REGISTER sip:sip.televoip.net SIP/2.0..Via: SIP/2.0/UDP 192.168.0.111;branch=z9hG4bK4977acbbc428883a..From: "Reginaldo Paula Souza" <sip:reginaldo_ptt@sip.televoip.net>;tag=307110492b802983..To: <sip:reginaldo_ptt@sip.televoip.net>..Contact: *..Call-ID: 6221319848b4afe5@192.168.0.111..CSeq: 106 REGISTER..Expires: 0..User-Agent: Grandstream HT286 1.0.4.68..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0.... # U 200.193.194.142:5060 -> 193.126.48.33:5060 SIP/2.0 401 Unauthorized..Via: SIP/2.0/UDP 192.168.0.111;branch=z9hG4bK4977acbbc428883a;rport=5060;received=193.126.48.33..From: "Reginaldo Paula Souza" <sip:reginaldo_ptt@sip.televoip.net>;tag=307110492b802983..To: <sip:reginaldo_ptt@sip.televoip.net>;tag=a791a9ab0b94eb4c05eab9b15c0fec4f.f9b6..Call-ID: 6221319848b4afe5@192.168.0.111..CSeq: 106 REGISTER..P-NATed-Caller: Yes..WWW-Authenticate: Digest realm="sip.televoip.net", nonce="40d46cd7c8d155389bef79bb516a0a2b2a5524f8"..Server: Sip EXpress router (0.8.13-dev-30-db_api (i386/linux))..Content-Length: 0..Warning: 392 200.193.194.142:5060 "Noisy feedback tells: pid=11293 req_src_ip=193.126.48.33 req_src_port=5060 in_uri=sip:sip.televoip.net out_uri=sip:sip.televoip.net via_cnt==1"....

20 years, 5 months

[Serusers] Answers

by "Juan G. Castañeda"

Sirs: Where can I read the answers to SERuser's questions? Regards, Juan G. Castañeda

20 years, 5 months

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sr-users June 2004