sr-users January 2004

sr-users@lists.kamailio.org

142 participants
338 discussions

[Serusers] Re: Serusers Digest, Vol 9, Issue 25

by Adrian Georgescu

http://www.ag-projects.com/#CDRTool On 25 Jan 2004, at 12:00, serusers-request(a)lists.iptel.org wrote: > > Date: Sun, 25 Jan 2004 01:56:28 +0100 > From: Arnd Vehling <av(a)nethead.de> > Subject: [Serusers] Billing System Recommendation > To: serusers <serusers(a)lists.iptel.org> > Message-ID: <4013143C.559505D7(a)nethead.de> > Content-Type: text/plain; charset=us-ascii > > Hi, > > can anyone recommend a VOIP Billing System to use with SER. Right now > i am not > very happy with the ispbs from hostplus. > > best regards, > > Arnd > > > > ------------------------------ > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > > End of Serusers Digest, Vol 9, Issue 25 > ***************************************

20 years, 10 months

[Serusers] Billing System Recommendation

by Arnd Vehling

Hi, can anyone recommend a VOIP Billing System to use with SER. Right now i am not very happy with the ispbs from hostplus. best regards, Arnd

20 years, 10 months

[Serusers] Don't Want To Be Busted ? ? ? . . . . . . . . crankshaft

by Cindy.bowden

20 years, 10 months

[Serusers] Reg. Codec Setting in SER for SIP Phones

by Balaji Bapulal Thoguluva

Hi, I have explained my problem below and I have given my question related to SER at the end. Any suggestion for the question would be of great help to me. Also I would appreciate if there is any solution for the problem I have described. I have the following network. Cisco SIP Phones (3322) <-> SIP Express Router (SIP Proxy router) <-> SIPH323 Converter <-> Cisco router 2611XM (acting as IP to IP Gateway) <-> Cisco CallManager <-> Cisco Skinny Phones (1133). First Problem (SIP->Skinny): --------------------------- When I call from SIP phone to skinny phone, the skinny phone rings. But when I take the hook, it gives me busy tone. I see from ethereal that the router is sending H.225.0 cs: Release Complete message to CallManager. My router dial-peer config is dial-peer voice 300 voip dest pattern 1133 session target CM's IP address codec g711alaw and my CallManager configuartion is: I use my default device pool that has default region using codec g711. I could also see that cisco skinny IP phone config. in CallManager uses default device pool defined above. Second problem (Skinny->SIP): ----------------------------- The same network configuartion is assumed. When I call from Skinny to SIP phone, the SIP phone just rings once. The SIP phone doesn't seem to ring continuosly (could not hear the dail tone) but shows a missed call from skinny phone. When I traced the call flow using ethereal, I see again the following router CallManager |----------->| H.225.0 cs:Alerting |<-----------| H.245 TerminalCapabiltySet |----------->| H.245 Terminal CapabiltySet |----------->| H.245 MasterSlaveDetermination |<-----------| H.245 TerminalCapabaility Ack |----------->| H.225.0 cs:Release Complete My dial-peer config is dial-peer voice 201 voip dest patt 3322 sess tar. SIP Proxy's IP codec g711alaw So, the conclusion is in both cases the router is sending Release complete message to CallManager. So I guess there is some capability mismatch between router and CallManeger. I guess SIPH323 is flexible to use any codec. Question: I have a slight doubt that is there anyway I can set the codec used by the sip phones in SER router. If there is any way, please throw some light on this issue. Thanks, Balaji

20 years, 10 months

[Serusers] Grandstream firmware 1.0.4.39 subscribe.

by Friso Kuipers

Hello all, I've been experimenting with Grandstream beta firmware for my budgetone 102. Result: every version > 1.04.18 kills all ser processes except one that just hangs and needs to be terminated with SIGKILL. Ser itself is 0.8.12 on linux 2.4.24, gcc 3.2.2, libc 2.3.2. Only this shows up in syslog (debug level 3): [/usr/local/sbin/ser] BUG: tcp_main_loop: dead child 2 This is the request that chokes ser: U 192.168.0.3:5060 -> 212.238.149.10:5060 SUBSCRIBE sip:bandsla.org:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.0.3;branch=z9hG4bK446fd67e04e4a465 From: <sip:f.kuipers@bandsla.org:5060>;tag=d64e39684c2d5733 To: <sip:f.kuipers@bandsla.org:5060> Contact: <sip:f.kuipers@192.168.0.3> Call-ID: 48adbbd50b3eaa53(a)192.168.0.3 CSeq: 54360 SUBSCRIBE User-Agent: Grandstream 1.0.4.39 Max-Forwards: 70 Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE Event: message-summary Expires: 600 Accept: application/simple-message-summary Content-Length: 0 Hope this is usefull, Friso Kuipers.

20 years, 10 months

[Serusers] error starting ser daemon with persistency support

by Ziying Sherwin

We are trying to compile ser 0.8.12 on Solaris 2.8 using gcc 3.3. The compilation and installation go smoothly until we tried to start the ser daemon with persistancy support. We followed the instructions listed on the website. First, we modified the MySQL configuration file to increase the maximun connections allowed. Second we restart the MySQL daemon. Third, we modified the ser_mysql.sh file to set the username, password and path properly and used it to create the MySQL tables. Then we modifid ser.cfg file and uncommented the following lines: loadmodule "/depot/package/ser_0.8.12/lib/ser/modules/mysql.so" loadmodule "/depot/package/ser_0.8.12/lib/ser/modules/auth.so" loadmodule "/depot/package/ser_0.8.12/lib/ser/modules/auth_db.so" modparam("usrloc", "db_mode", 2) modparam("auth", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") uncommented and modified the following lines: if (!www_authorize("<our domain name>", "subscriber")) { www_challenge("<our domain name>", "0"); break; }; Finally, we tried to start the ser daemon by directly calling the binary "ser", and the got the following error message: >ser ERROR: bad config file (1 errors) How can we find where the error is? All the modules listed in the configuration file are present in the directory. What else could go wrong? Thank you very much in advance. Ziying Sherwin

20 years, 10 months

[Serusers] Presence module

by CURRO_DOMINGUEZ

Hi all I'm testing pa module on ser and I have some questions. I would thank you if you could help me. I'm not using Jabber, so can I use pa Module to manage presence of SIP users? I have added loadmodule "/usr/local/lib/ser/modules/pa.so" modparam("pa", "default_expires", 3600) Where have I to add handle_subscription ("registrar")? Thank you very much Curro I'll attach my ser.cfg # ------------------------- 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; }; # 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("cdshuesca.es", "subscriber")) { www_challenge("cdshuesca.es", "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(); }; #lookup("aliases"); #save("aliases");

20 years, 10 months

[Serusers] Re: Problem with 0.8.13-dev-23-merged

by Andrea Girardi

Hi when I start this version I can see in /var/log/messages that ser start correctly but when I try to check server status with serctl moni or simply by connect with MSN ser doesn't work. [cycle #: 1; if constant make sure server lives and fifo is on] I've tried to find some error on log but I didn't found nothing. I'm using a ser.cfg that correctly works with 0.8.12 Any idea? thanks, Andrea

20 years, 10 months

[Serusers] NEW USER...

by Sebastian Nocetti

Hi everybody, I am new on this of SER and I am really confused... I want to know if someone can contact me by MSN (gnocetti(a)hotmail.com) and give me some clues.... to configure and make it works. Really I tried a lot but never succesfully, please if someone can introduce me to this excelent tool, becaus I want to implement SER like a PROXY SIP and ROUTING... for Carrier.... TX Sebastian.

20 years, 10 months

[Serusers] About mangler and xlog

by giAndrea

Folks, In my cfg file I've this block. I call this block when I receive a request from a private IP (ex. 192.168.x.x) /* route block for mangling */ route[1] { log(1, "route-block 1 (mangling)\n"); # we mangle the contact address and ports in SDP part of the message if (method == "INVITE") { setflag(1); log(1, "mangle block: INVITE received and we will mangle ip \n"); sdp_mangle_ip("192.168.0.0/255.0.0.0", "192.168.0.99"); }; # rewrite Contact: header if (search("Contact: .*(a)192\.168\.")) { /* we seem to have a private address on a Contact which is not valid */ encode_contact("enc_prefix","192.168.0.99"); xlog("L_NOTICE", "mangle block: encoded Contact header: <%ct> \n"); }; } When I send an INVITE I can see in SDP: Content-Type: application/sdp.. Content-Length: 204..204Contact: Andrea Girardi <sip:enc_prefix*andrea**192.168.0.98*5060*@192.168.0.99>.. User-Agent: SCS/v3.1.12.33.. .. v=0.. o=AndreaGirardi13698577 0 IN IP4 192.168.0.98.. s=ScS Client.. c=IN IP4 217.57.90.117.. t=0 0.. m=audio 1154 RTP/AVP 0 8 4.. a=rtpmap:0 PCM U/8000.. a=rtpmap:8 PCMA/8000.. a=rtpmap:4 G723/8000.. a=ptime: 30.. This work correctly when USERA send an Invite to USERB, but when USERB send 200 ok this is the SDP: Content-Type: application/sdp.. Content-Length: 189.... v=0.. o=braies 0 0 IN IP4 192.168.0.98.. s=ScS Client.. c=IN IP4 192.168.0.98.. b=CT:1000.. t=0 0.. m=audio 57820 RTP/AVP 0 8 4.. a=rtpmap:0 PCMU/8000.. a=rtpmap:8 PCMA/8000.. a=rtpmap:4 G723/8000.. It's correct or User B has to change c field in SDP? How? Can I see a encoded contact (sip:enc_pref*username*ip*port*protocol@public_ip *) with xlog or simply in stderr? i've tried to use %ct but it return <sip:userA@192.168.0.97:5060> is it better to use nathelper or mangler? Thanks, Andrea ------------------------------------------------------- Andrea Girardi mobile +39 347 9624051 sip address: giandrea(a)sip.quellidelpirana.it http://www.giandrea.com

20 years, 10 months

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sr-users January 2004