sr-users August 2006

sr-users@lists.kamailio.org

184 participants
360 discussions

by Shaun Hofer

When I try to use avp_pushto() in one section of ser.cfg it works and in another section it doesn't. When I use it to out right forward a call it works fine (callfwd). When it's used in the failure_route[1] it doesn't seem to push a value onto ruri (I have made sure that avp_db_load() has been called/used) . I'm wondering if avp_pushto() one of those methods that can't be called from a failed route ? I get the following error when I use it: ERROR: t_forward_nonack: no branched for forwarding ERROR: w_t_relay (failure mode): forwarding failed ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) I tried taking away voicemail section in failure_route[1] and made sure I was phoning extension with a number instead of 'voicemail' in value column of the database. That made no difference. When I tried adding append_branch() after avp_pushto() it just ends up ringing the original phone number again. Thanks in advance -Shaun route[1] { # ---------------------------------------------------------- # Default Message Handler # ---------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[3] { # ---------------------------------------------------------- # INVITE Message Handler # ---------------------------------------------------------- if (!allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (avp_db_load("$ruri/username", "s:callfwd")) { setflag(22); avp_pushto("$ruri", "s:callfwd"); route(6); break; }; if (!lookup("location")) { if (uri=~"^sip:0[1-9][0-9]{8}") { # Domestic PSTN route(4); route(5); break; }; sl_send_reply("404", "User Not Found"); break; }; if (avp_db_load("$ruri/username", "s:fwdnoanswer")) { if (!avp_check("s:fwdnoanswer", "eq/$ruri/i")) { setflag(27); }; }; log(1,"t on fail \n"); t_on_failure("1"); route(4); route(1); } route[4] { # ---------------------------------------------------------- # NAT Traversal Section # ---------------------------------------------------------- if (isflagset(6) || isflagset(7) || isflagset(11)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } route[6] { # ---------------------------------------------------------- # Call Forwarding Handler # ---------------------------------------------------------- lookup("aliases"); if (uri!=myself) { if (!isflagset(22)) { append_branch(); }; route(4); route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:88[0-9]{8}") { # Offline VoIP Network user sent to * for voicemail route(4); route(5); break; }; sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } failure_route[1] { if (t_check_status("487")) { break; }; if (isflagset(26) && t_check_status("408")) { if (avp_check("s:fwdnoanswer","eq/voicemail/I")) { avp_delete("s:fwdnoanswer"); resetflag(26); revert_uri(); rewritehostport("x.x.x.x:5060");#goto voicemail server append_branch(); t_relay_to_udp("x.x.x.x", "5060"); break; } else { avp_pushto("$ruri", "s:fwdnoanswer"); avp_delete("s:fwdnoanswer"); resetflag(26); route(6); break; }; }; end_media_session(); }

18 years, 3 months

[Serusers] Problem with fr_inv_timer_avp

by Michel Zenone

Hi! Does somebody know why 'fr_inv_timer_avp' does nothing when I use it as follow: ... modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("avpops","avp_url","mysql://root@localhost/ser") modparam("avpops","avp_table","avptimer") # 'avptimer' is like 'usr_preferences' ... avp_db_load("$ruri","inv_timeout/avptimer"); Actually I try to set a fr_inv_timer different from an user to another. 'avptimer' timer defines, for each users, their proper fr_inv_timer. If somebody can help... Thanks in advance, Michel

18 years, 3 months

[Serusers] xmlrpc

by Victoria Cortez

Hello everybody, Can anybody tell me about xmlrpc module? I've found it in the CVS version. I could not find any docs about it yet. What is this module suppose to do, exactly? Thanks in advance, Victoria

18 years, 3 months

[Serusers] Looking for Call termination service provider

by John Kim

Hi, I am looking for a VoIP wholesaler who will allow us to operate our own SIP server infrastructure while we can forward all PSTN destined calls to their gateway. I really didn't think that it would be hard to find one but I have not been successful. I have checked so far with iCallGlobe.com and Deltathree.com. Can any of you recommend us some VoIP wholesaler with quality service and competitive rate who allows this business setup? Below is logical business relation setup: Our subscriber <--->ATA setup to connect sip.ours.com<----->PSTN gateway of wholesaler.com<--->PSTN Basically we are looking for good "wholesaler.com". Any comments will be appreciated. John K.

18 years, 3 months

Re: [Serusers] SS7 Signaling & Gateways

by Max Clark

Thanks Victor - unfortunately Cirpack does not support North American networks. On 8/9/06, Victor Stanescu <victor.stanescu(a)gtstelecom.ro> wrote: > I don't know what is reasonable for you, but I really advise you to take > a look at Cirpack (www.cirpack.com). It's the full option gateway/switch > (isdn, r2, ss7, v5, mgcp, sip, h.323, h.248), and if you take time to > study their hardware, you can start from very small and pay as you grow. > > I am using Cirpack's for 3 years and I was never more happier about > other telco equipment :-) > > Max Clark wrote: > > Hi all, > > > > I am looking for a _resonable_ cost SS7 solution to bundle with a TDM > > <-> SIP media gateway. I know the Cisco 5350 supports SS7 when bundled > > with an external solution - is there anything reasonably priced that > > will give me SS7 capability? > > > > Thanks in advance, > > Max > > > -- Max Clark http://www.clarksys.com

18 years, 3 months

[Serusers] Trying to use a remote mysql....

by Roger Lewau

I have been running SER quite successfully for a long time now. So tonight I dared to replicate the database to a remote server and changed the ser.cfg as follows. This is the two lines I changed in my ser.cfg filen #fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_db_url="mysql://ser:heslo@192.168.12.140/ser" #modparam("auth_db|permissions|uri_db|usrloc|acc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db|permissions|uri_db|usrloc|acc", "db_url", "mysql://ser:heslo@192.168.121.140/ser") Also updated serctl as follows: SQL_HOST=192.168.121.140 I then Stopped Ser, MySQL and restarted SER. It takes a long time then I get these line in the error log.. and ser fails to start. Aug 9 22:36:08 ns2 ser[43517]: new_connection(): Can't connect to local MySQL server through socket '/tmp/mysql.sock' (2) Aug 9 22:36:08 ns2 ser[43517]: db_init(): Could not create a connection Aug 9 22:36:08 ns2 ser[43517]: ERROR: group_db_ver: unable to open database connection Aug 9 22:36:08 ns2 ser[43517]: group:mod_init(): Error while querying table version Aug 9 22:36:08 ns2 ser[43517]: init_mod(): Error while initializing module group Then I start the old local mySQL server again and restart ser and it springs to life. Why is that? I am not pointing to the local mysql server anymore... Have I missed something in the configuration? Even more strange, it seems that all authentication and accounting gets queried in the new server as I see can see the data in the new server alse sees the client connection from SER. Any one who can help with an explanation on this behaviour? Kind regards Roger

18 years, 3 months

[Users] I can`t add new client by using openserctl add. Please help...

by Ferianto siregar

Dear all, Now, I try to test my openser configuration by using xlite. I tried to call from client to test the openser configuration that I have built working sucessfully. So, I must create a new user by using "openserctl add" command. But when i tried to do that, I got an error message. The error message said that " no rows effected". I think that this message said that I can not add a new user. For example: #openserctl add aldi aldi aldi(a)pcr.ac.id What can i do for solve this problem? I do hope anybody can give me a sugesstion.Please help me...Please Note: I use mysql for the database, and it can work goodly. ( I can show all the table that there is in openser database) Thanks with cheers, Ferianto --------------------------------- Do you Yahoo!? Next-gen email? Have it all with the all-new Yahoo! Mail Beta.

18 years, 3 months

[Users] Q:ABOUT:DELAY THE INVITE

by Alejandro Sanchez

MY scenario is the next. I have 2 video telephones that are conected to the PSTN and they has access trougth a dial up conection to the IP network and the problem is that the 2 video telephones shuld be register before one to other geretate the INVITE then i need make that when the first REGISTER arrive the sip server generate a delay for wait for the REGISTER of the other video telephone and both VT state register for the INVITE message.. Someone knows how can i do this? Thank's Alex. ___________________________________________________________ Do You Yahoo!? La mejor conexión a Internet y <b >2GB</b> extra a tu correo por $100 al mes. http://net.yahoo.com.mx

18 years, 3 months

[Serusers] Call limiting ?

by Alexandre Snarskii

Hi! As far as I know, SER does not have functionality of limiting number of calls on per-user basis. It was discussed years ago, and the reason for that was "We don't have ability to know is call still alive" (http://lists.iptel.org/pipermail/serusers/2004-November/013035.html) But, in April 2005, RFC 4028 (Session Timers in SIP) appeared, and it (at least theoretically) can be utilised to achieve knowlege of active dialogs, and, as result - on number of active calls. And this RFC looks like supported on Cisco gateways: U X.X.X.X:59104 -> X.X.X.X:5060 INVITE sip:aaaaaaa@X.X.X.X:5060 SIP/2.0. Supported: 100rel,timer,replaces. ^^^^^ Min-SE: 1800. ^^^^^^^^^^^^^^ User-Agent: Cisco-SIPGateway/IOS-12.x. Question: is this RFC/functionality planned to be supported in SER ? If not - are there any good intro on "How to write SER module by yourself" ? :)

18 years, 3 months

[Serusers] SS7 Signaling & Gateways

by Max Clark

Hi all, I am looking for a _resonable_ cost SS7 solution to bundle with a TDM <-> SIP media gateway. I know the Cisco 5350 supports SS7 when bundled with an external solution - is there anything reasonably priced that will give me SS7 capability? Thanks in advance, Max -- Max Clark http://www.clarksys.com

18 years, 3 months

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sr-users August 2006