sr-users April 2007

sr-users@lists.kamailio.org

177 participants
273 discussions

[Users] SEMS: ivr API
by Helmut Kuper 14 Apr '07

14 Apr '07

-----BEGIN PGP SIGNED MESSAGE----- Hash: SHA1 Hi, I have some trouble to get redirect.py running as an ivr module, cause ivr module's api has changed. Unfortunately I wasn#t able to find any new documentaion about sems' ivr except some hints that documentation isn't actual. Does anybody has the documented ivr API of sems 0.10? Thanks in advance Helmut -----BEGIN PGP SIGNATURE----- Version: GnuPG v1.4.5 (MingW32) Comment: Using GnuPG with Mozilla - http://enigmail.mozdev.org iD8DBQFGH4QM4tZeNddg3dwRAhycAJ97h/WkjOGAmzr/KbmSnB0QNs8DngCfboEN +qKBWEbg95HlLMq55hPhbEE= =jMhC -----END PGP SIGNATURE-----

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[Serusers] SER, SEMS voicemail and
by Meidiana Ten 13 Apr '07

13 Apr '07

Hi there, I'm sorry for bothering all this time with same questions I still can't solve the problem because nobody reply it. I really don't have any idea about how SEMS works. I setup eyebeam to forward call to voicemail number which is 115@domain for more info, the voicemail address was one of the user in subscriber/aliases table of SER that I added. When the call forwarded, SER have to check to usrloc for the address. It automatically appear that User not found on the softphone. Will it be any way that I don't have to create the address for voicemail address but SER can forward the call for SEMS? Or maybe my softphone forward call setting isn't right?Btw, I'm using eyebeam. I hope that someone reply for my mail this time. I'm sorry if my english is bad. Regards, Meidiana --------------------------------- Sucker-punch spam with award-winning protection. Try the free Yahoo! Mail Beta.

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[Users] NEW FEATURE: support for multiple RTPproxies
by Bogdan-Andrei Iancu 13 Apr '07

13 Apr '07

Hi, thanks to Ancuta Onofrei, a new member of Voice System's team, NATHELPER module now supports multiple rtpproxies - actually you can define multiple sets of rtpproxies - nathelper performs load-balancing over a set and the user can select the set. For more details, see: http://www.openser.org/docs/modules/devel/nathelper.html#AEN76 The definition of the rtpproxy socket was extended to allow definition of multiple sets. See: http://www.openser.org/docs/modules/devel/nathelper.html#AEN140 Also, a new function was added to set from script what set of rtpproxies is to be used. See: http://www.openser.org/docs/modules/devel/nathelper.html#AEN298 Example: # multiple sets of multiple rtproxies modparam("nathelper", "rtpproxy_sock", "1 == udp:localhost:12221 udp:localhost:12222") modparam("nathelper", "rtpproxy_sock", "2 == udp:localhost:12225") ..... set_rtp_proxy_set("2"); force_rtp_proxy(); ..... regards, bogdan

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[Users] Module tm, function t_lookup_request
by Wolfgang Hottgenroth 13 Apr '07

13 Apr '07

Hi, what happens to the t_lookup_request function of the tm module in the 1.2.0 release? I was using this function in my openser script and no it disappears. Although the underlying t_check function still exists. What is the reason? Thanks, Wolfgang

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[Users] Why do not have 407 Proxy Authentication Required
by Mike Lee 13 Apr '07

13 Apr '07

Dear friends: I have installed openser-1.1.0-tls version, and it works fine with mysql, the sip flow is below: UA OpenSER invite----------------> <----------------100 trying <----------------101 dialog establishement <----------------180 ringing ..... My question is, why not have the 407 Proxy Authentication Required in the per call, see my openser.cfg file: # # $Id: openser.cfg,v 1.6 2006/02/15 18:23:46 bogdan_iancu Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=4 fifo="/tmp/openser_fifo" # # uncomment the following lines for TLS support #disable_tls = 0 #listen = tls:your_IP:5061 #tls_verify = 1 #tls_require_certificate = 0 #tls_method = TLSv1 #tls_certificate = "/usr/local/etc/openser/tls/user/user-cert.pem" #tls_private_key = "/usr/local/etc/openser/tls/user/user-privkey.pem" #tls_ca_list = "/usr/local/etc/openser/tls/user/user-calist.pem" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/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"); 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("localhost", "subscriber")) { www_challenge("localhost", "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; } ============================ thanks. Mike --------------------------------- Ahhh...imagining that irresistible "new car" smell? Check outnew cars at Yahoo! Autos.

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[Serusers] using SER
by gabrielyans 13 Apr '07

13 Apr '07

Hi I was looking for a softswitch and found that SER could be usefull for what I am going to implement. I want to have a Signalling GW (I think it will be SER) who will receive calls and then direct them according to his routing table. SER will transfer calls to Asterisk, and then Asterisk (together with A2billing) will make the accounting for that call. Is it the right choise to use SER? Actually, I cannot find any forum concerning SER to check if there are already someone who is using SER as I am planning to use. Also, concerning WEB interface - it seams that SERWeb gives only user-management interface and not administrator interface - am I right? Please advise. gabrielyans

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[Serusers] SER + SEMS problem with ACK
by laurent schweizer 13 Apr '07

13 Apr '07

Hello, I'm using sems with ser 0.9.7-pre6 and I have a problem with ACK, as you can see , the other part is not acceting them, what is wrong... ? just, the SIP is going via a IPSEC vpn. Thanks for your help Laurent ? 212.190.89.140 -> 212.249.14.236 ............s....Y...5..E.....@....d..Y...}9.......fSIP/2.0 486 Busy here..Via: SIP/2.0/UDP 195.129.125.57;branch=z9hG4bK8ec6.3148d226.0..From:417875185 51 <sip:+41787518551@bylltel.com>;tag=568D320F-461F45CB000A78E3-40807960..To: <sip:+41787518551@212.190.89.137>;tag=6083161C-10B1..Date: Fri, 13 Apr 2007 08:56:47 GMT..Call-ID: 5A1CFF5D-461F45CB000A78EE-40807960@212.249.14.236..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 10 INVITE..Allow-Events: telephone-ev ent..Content-Length: 0.... U 212.190.89.137:5060 -> 195.129.125.57:5060 SIP/2.0 486 Busy here..Via: SIP/2.0/UDP 195.129.125.57;branch= z9hG4bK8ec6.3148d226.0..From: 41787518551 <sip:+41787518551@bylltel.com >;tag=568D320F-461F4 5CB000A78E3-40807960..To: <sip:+41787518551@212.190.89.137>;tag=6083161C-10B1..Date: Fri, 13 Apr 2007 08:56:47 GMT..Call-ID: 5A1CFF5D-461F45CB000A78EE-40 807960@212.249.14.236..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 10 INVITE..Allow-Events: telephone-event..Content-Length: 0.... ? 212.249.14.236 -> 212.190.89.140 ................koG.B...E.....@.@..&..}9..Y.........ACK sip:+41787518551@212.190.89.137 SIP/2.0..Via: SIP/2.0/UDP 195.129.125.57 ;branch=z9hG4bK8ec6.3148d 226.0..From: 41787518551 <sip:+41787518551@bylltel.com>;tag=568D320F-461F45CB000A78E3-40807960..Call-ID: 5A1CFF5D-461F45CB000A78EE-40807960(a)212.249.14.23 6..To: <sip:+41787518551@212.190.89.137>;tag=6083161C-10B1..CSeq: 10 ACK..User-Agent: Sip EXpress router(0.9.7-pre6(x86_64/linux))..Content-Length: 0... . ? 212.190.89.140 -> 212.249.14.236 ................gT.[_...E.....@....c..Y...}9.......fSIP/2.0 486 Busy here..Via: SIP/2.0/UDP 195.129.125.57;branch=z9hG4bK8ec6.3148d226.0..From:417875185 51 <sip:+41787518551@bylltel.com>;tag=568D320F-461F45CB000A78E3-40807960..To: <sip:+41787518551@212.190.89.137>;tag=6083161C-10B1..Date: Fri, 13 Apr 2007 08:56:47 GMT..Call-ID: 5A1CFF5D-461F45CB000A78EE-40807960@212.249.14.236..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 10 INVITE..Allow-Events: telephone-ev ent..Content-Length: 0.... U 212.190.89.137:5060 -> 195.129.125.57:5060 SIP/2.0 486 Busy here..Via: SIP/2.0/UDP 195.129.125.57;branch= z9hG4bK8ec6.3148d226.0..From: 41787518551 <sip:+41787518551@bylltel.com >;tag=568D320F-461F4 5CB000A78E3-40807960..To: <sip:+41787518551@212.190.89.137>;tag=6083161C-10B1..Date: Fri, 13 Apr 2007 08:56:47 GMT..Call-ID: 5A1CFF5D-461F45CB000A78EE-40 807960@212.249.14.236..Server: Cisco-SIPGateway/IOS-12.x..CSeq: 10 INVITE..Allow-Events: telephone-event..Content-Length: 0.... ? 212.249.14.236 -> 212.190.89.140 ............|ovW. ..5.izE.....@.@..&..}9..Y.........ACK sip:+41787518551@212.190.89.137 SIP/2.0..Via: SIP/2.0/UDP 195.129.125.57 ;branch=z9hG4bK8ec6.3148d 226.0..From: 41787518551 <sip:+41787518551@bylltel.com>;tag=568D320F-461F45CB000A78E3-40807960..Call-ID: 5A1CFF5D-461F45CB000A78EE-40807960(a)212.249.14.23 6..To: <sip:+41787518551@212.190.89.137>;tag=6083161C-10B1..CSeq: 10 ACK..User-Agent: Sip EXpress router(0.9.7-pre6(x86_64/linux))..Content-Length: 0...

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[Serusers] Avpops'exported function
by chungyu 13 Apr '07

13 Apr '07

Dear all: avpops module' exported function avp_db_delete() can do delete one record? example: I use "test" as default table that has two fields. username | number ------------------- bob | 1 alice | 1 bob | 2 my configuration: modparam("avpops", "avp_url", "mysql://ser:heslo@localhost/ser") modparam("avpops", "avp_table", "test") modparam("avpops", "username_column", "username") modparam("avpops", "avp_aliases", "num=i:55") modparam("avpops", "db_scheme", "number_scheme:table=test;value_col=number;value_type=string") if (method=="INVITE"){ avp_db_load("$from/username", "$num/$number_scheme"); //load from table "test" ,number column as string value into AVP name '$num' avp_db_delete("$from/username", "s:test") //delete all bob' record }; if bob calling then bob'record has deleted username | number -------------------- alice | 1 But i want delete a record. ex: delete number value is 1 for bob username | number -------------------- alice | 1 --------------------- bob | 2 avp_db_delete() can do delete one record? Another question is avp_db_store() can do insert new data in table? ex: insert into test (username,number) values ("mary","3") username | number ------------------- alice | 1 ------------------- bob | 2 ------------------- mary | 3 avp_db_store() can do it? sorry,I still can't understand what it mean about avp_db_store() and avp_db_delete(). Can tell me? Thanks a lot in advance Chungyu

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[Serusers] fifo
by Javier 12 Apr '07

12 Apr '07

Hi, I have a problem, I use the ser 0.8.14, and I update to 0.10.99 dev 30 I have a module that send BYE to all part of comunication using fifo. but this don't work in 0.10.99 version can you help me ? best regards Lou __________________________________________________ Preguntá. Respondé. Descubrí. Todo lo que querías saber, y lo que ni imaginabas, está en Yahoo! Respuestas (Beta). ¡Probalo ya! http://www.yahoo.com.ar/respuestas

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[Users] MOH SIP UA - does such a program exist?
by Aaron Daubman 12 Apr '07

12 Apr '07

Greetings, I'm looking for a (IPv6 capable) SIP UA that functions basically as a Music-on-Hold type client. This will be used in a lab, and the basic desired functionality is to have this UA register its extension, and then have any of our soft/hard-phones be able to dial the extension and hear the music playing (WAV, MP3, whatever) - note that there is no need for it to be able to handle simultaneous calls... a SIP softphone that allowed auto-answer and auto-wav-playback would be exactly what I'm looking for... I really just want to remove the need to have a live person on the other end of a call to answer and speak to demonstrate VoIP functionality during lab demonstrations and quality experiments... Thanks, ~Aaron

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sr-users April 2007