sr-users July 2005

sr-users@lists.kamailio.org

199 participants
544 discussions

[Serusers] Setting tm module parameter "fr_inv_timer_avp" more than once during a call
by reticent 08 Aug '05

08 Aug '05

It seems that setting the "fr_inv_timer_avp" parameter more than once during a call is not possible. However there is a situation where this is necessary: 1. VoIP ATA to VoIP ATA call with call forwarding where call forwarded number is a PSTN destination To facilitate Call Forward No Answer, the fr_inv_timer_avp is used and set to a value (say 12 seconds), after this timer hits a failure_block is run, in this failure block i rewrite the destination and check if this destination is a PSTN number. If it is than I would like to change the fr_inv_timer_avp parameter to somthing higher due to the nature of PSTN termination but it does not seem to have any effect. Has anyone experienced this problem and possible found a workaround? Or maybe this is a known issue tavis

2 1

[Serusers] accounting
by harry gaillac 06 Aug '05

06 Aug '05

Hello, I use ser/serweb-0.9.3 I added tow accounts with numerical aliases. When I dial user1@domain to user2@domain it's ok. if i wish to forward to voicemail sems send me back voicemessages. However if i dial alias1@domain to alias2@domain I get the message below. How can I solve this problem ? Harry /////////////////////////////////////////////////////// Jul 24 14:58:15 serveur1 Sems[2535]: Error: (AmSession.cpp)(startSession)(497): 404 voicemail: no email address for user <84> Jul 24 14:58:15 serveur1 /usr/sbin/ser[7495]: ACC: call missed: method=INVITE, i-uri=sip:84@nxs.yi.org:5060;user=phone, o-uri=sip:h.gaillac@192.168.0.20, call_id=66219bb6-f9eef79c-bc2479d3(a)192.168.0.21, from="test" <sip:test@nxs.yi.org>;tag=CB323432-D0A509E1, code=404 voicemail: no email address for user <84>, userpart=84 /////////////////////////////////////////////////////// ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

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[Users] Stability, and -DF_MALLOC
by Kanakatti M. Subramanya 05 Aug '05

05 Aug '05

I'm running 0.9.5, on my system, and *have* to comment out -DF_MALLOC (and uncomment -DDBG_QM_MALLOC) to have the system be stable. It pretty much always crashes when trying to do save("location") Anyone else seen this? List of modules is 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/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/exec.so" loadmodule "/usr/local/lib/openser/modules/group.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/options.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/lcr.so"

3 5

[Users] via header not added when transport=TCP?
by Erich Nahum 02 Aug '05

02 Aug '05

Howdy, I'm playing with OPENSER 0.9.5 with sipp as the workload generator on a bunch of linux boxes. When the transport is UDP, openser adds a via header like it is supposed to as a proxy. But when I use TCP no via header is added. This is a UAC->proxy->UAS scenario. Here's the header as seen at the UAS: | UDP message received [633] bytes : | | INVITE sip:service@10.0.1.41:5060 SIP/2.0 | Record-Route: <sip:10.0.1.42;ftag=1;lr=on> | Via: SIP/2.0/UDP 10.0.1.42;branch=z9hG4bKe9cc.cc24a79.0 | Via: SIP/2.0/UDP bronxville1:5060;received=10.0.1.40;branch=z9hG4bK-1-0 | From: sipp <sip:sipp@bronxville1:5060>;tag=1 | To: sut <sip:service@10.0.1.41:5060> | Call-ID: 1-31891@bronxville1 | CSeq: 1 INVITE | Contact: sip:sipp@bronxville1:5060 | Max-Forwards: 16 | Subject: Performance Test | Content-Type: application/sdp | Content-Length: 132 | P-hint: outbound And the corresponding message received at the UAS when using TCP: | TCP message received [495] bytes : | | INVITE sip:service@10.0.1.41:5060 SIP/2.0 | Via: SIP/2.0/TCP bronxville1:5060;branch=z9hG4bK-1-0 | From: sipp <sip:sipp@bronxville1:5060>;tag=1 | To: sut <sip:service@10.0.1.41:5060> | Call-ID: 1-31865@bronxville1 | CSeq: 1 INVITE | Contact: sip:sipp@bronxville1:5060 | Max-Forwards: 70 | Subject: Performance Test | Content-Type: application/sdp | Content-Length: 132 Note the absence of the 2nd Via header. Openser complains about this on the way back to the UAC. Any suggestions appreciated. Thanks, -Erich -- Erich M. Nahum IBM T.J. Watson Research Center Research Staff Member P.O. Box 704 nahum(a)watson.ibm.com Yorktown Heights NY 10598

2 5

[Serusers] serctll add error
by chris 02 Aug '05

02 Aug '05

hello i tried running serctl add 1001 1001 chris_cleofe2yahoo.com but i got this error: awk: syntax error near line 1 awk: bailing out near line 1 awk: syntax error near line 1 awk: bailing out near line 1 HA1 calculation failed pls advise. thnks.

4 10

[Serusers] gw-pstn.3.07.cfg with Call-Transfer support
by Bastian Schern 01 Aug '05

01 Aug '05

Hello, I tried out the SER config "gw-pstn.3.07.cfg" from www.onsip.org. Nearly everything works fine but "Call-Transfer" is not working. Is it possible to integrate "Call-Transfer" support into the "gw-pstn.3.07.cfg"? Does anybody integrates this already? Regards Bastian debug=3 fork=yes log_stderror=no listen=192.0.2.13 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/mediaproxy.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" modparam("auth_db|permissions|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.0.2.13:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); break; }; if (uri==myself) { if (method=="ACK") { route(6); break; } else if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:\ +\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (uri=~"^sip:011[0-9]*@") { # International PSTN route(4); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { # Domestic PSTN route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (method=="CANCEL") { route(1); break; }; route(5); route(1); } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("192.0.2.245"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); route(5); route(1); } route[5] { # ----------------------------------------------------------------- # RTP Proxy Enabler # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } route[6] { # ------------------------------------------------------------------------ # ACK Handler # ------------------------------------------------------------------------ # ------------------------------------------------------------------------ # Aliases Section # ------------------------------------------------------------------------ lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; lookup("location"); route(1); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:\ +0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; }

2 2

[Serusers] UAC module status of CVS HEAD version
by Innocent Evil 01 Aug '05

01 Aug '05

Hello, Anybody know about the status of UAC module of CVS HEAD version. Does it work? Thanks,

4 4

[Users] Using permissions
by jayesh nambiar 01 Aug '05

01 Aug '05

hi there, Can you please tell me if I can use the permissions module to trust the requests coming towards my SIP server using source IP. But my problem is I want to use flat files to store the source IPs I trust. Is it possible And if the IPs are trusted I need to forward them without any authentication. I need to store the trusted IPs in flat files since I dont need the overhead of querying DB. Please help me out. Thanx --------------------------------- Start your day with Yahoo! - make it your home page

2 1

[Serusers] 1 way voice issue
by Sam Lee 01 Aug '05

01 Aug '05

Hi, I've recently encountered some problem with my SIP service whereby i call out to a specific number and i encounter a one way voice. If i'm the initiator, i cannot hear the other party but he can hear me. At first i thought it was a return route issue (as i'm going thru NAT) , so i switch my SIP to a public IP but i still face the same problem. Its really only that specific PSTN number that i have dialed facing this problem. The only difference that i can think of is that PSTN number is on a different route. I did a NGREP from my SIP server for the PSTN number that works (2-way voice) and the Number that doesn't work (1-way voice) . The only difference is there is an extra : NGW --> Proxy SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP Proxy --> SIP Device SIP/2.0 183 Session Progress..Via: SIP/2.0/UDP for the PSTN number that works (the one with 2-way voice). Anyone has idea what does the Session Progress is for ? Or what problem am i facing ? Thanks a mILLION ! Regards, Sam

2 1

[Serusers] STUN for NAT traversal...
by Roger Lewau 01 Aug '05

01 Aug '05

I'm trying to use STUN for NAT traversal for all clients not using 1 to 1 NATing, and mediaproxy for the rare few who does.But it seems that all traffic is proxied anyway, dont really understand why. I do call the routeblock for mediaproxy before relaying the call, but since STUN is used it should not detect that NAT is in place. Or am I missing some logic? Any Idea on how to accomplish this? Kind regards Roger - Getting one step closer to a really god service everyday ;)

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sr-users July 2005