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

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

19 years, 3 months

[Serusers] accounting

by harry gaillac

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

19 years, 3 months

[Users] Stability, and -DF_MALLOC

by Kanakatti M. Subramanya

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"

19 years, 3 months

[Users] via header not added when transport=TCP?

by Erich Nahum

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

19 years, 3 months

[Serusers] serctll add error

by chris

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.

19 years, 3 months

[Serusers] gw-pstn.3.07.cfg with Call-Transfer support

by Bastian Schern

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(); }; }

19 years, 3 months

[Serusers] UAC module status of CVS HEAD version

by Innocent Evil

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

19 years, 3 months

[Users] Using permissions

by jayesh nambiar

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

19 years, 3 months

[Serusers] 1 way voice issue

by Sam Lee

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

19 years, 3 months

[Serusers] STUN for NAT traversal...

by Roger Lewau

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 ;)

19 years, 3 months

Jump to page:

2024

2023

2022

2021

2020

2019

2018

2017

2016

2015

2014

2013

2012

2011

2010

2009

2008

2007

2006

2005

2004

2003

2002

sr-users July 2005