- sr-users - kamailio.org

Fwd: Re: [Serusers] PRACK in ser
by Dana Turcanu 27 Jun '05

27 Jun '05

--- Dana Turcanu <dana_turcanu(a)yahoo.com> wrote: > Date: Mon, 27 Jun 2005 01:57:00 -0700 (PDT) > From: Dana Turcanu <dana_turcanu(a)yahoo.com> > Subject: Re: [Serusers] PRACK in ser > To: harry gaillac <gaillacharry(a)yahoo.fr> > > > > Hello, > > > Hello, > > > > Look at my ethereal file > > > In your file the PRACK reach the calling party > and > it respond with 200 OK. > > In my case,the PRACK never reach the calling > party > and the server is returning 400 error. > > Is this error returned because the request from > tha Cisco ATA 186 is malformed?? > > Can some one give me a log of a call between tow > Cisco ATA 186 ( version 3.2 - with PRACK method) > passing through a SIP Proxy? > > > > > --- Dana Turcanu <dana_turcanu(a)yahoo.com> a écrit > : > > > > > Hello, > > > > > > --- Andreas Granig <andreas.granig(a)inode.info> > > > wrote: > > > > > > > Dana Turcanu wrote: > > > > > Is PRACK method supported in ser? When an > > > > equipment > > > > > sends a PRACK the server responds with "400 > > Bad > > > > > Request" and the call is disconnected. > > > > > > > > Are you sure the "400" comes from SER and not > > from > > > > the other endpoint? > > > > Do you allow routing of PRACKs in your > > SER-Config? > > > > > > > > > > I logged the call from the tow Cisco ATA 186. > > The > > > calling ata is sending PRACK and the answer is > > > comming > > > from the server: > > > > > > > > > 050620|11:48:29 [0:0]Tx Msg to > ip_sip_server:5060 > > > 050620|11:48:29 > > > 050620|11:48:29 PRACK > > > sip:0332yyyy@domain.com;user=phone SIP/2.0 > > > 050620|11:48:29 Via: SIP/2.0/UDP > > > ip_ata_1:5060;branch=z9hG4bK2b0f7845ea7a8edb > > > 050620|11:48:29 From: > > > > <sip:332xxxx@domain.com;user=phone>;tag=2064987569 > > > 050620|11:48:29 To: > > > > <sip:0332yyyy@domain.com;user=phone>;tag=811613285 > > > 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 > > > 050620|11:48:29 CSeq: 3 PRACK > > > 050620|11:48:29 RAck: 1 2 INVITE > > > 050620|11:48:29 User-Agent: Cisco ATA 186 > v3.2.0 > > > atasip (041111A) > > > > > > > > > > > > > > > 050620|11:48:29 [0]Rx Msg from > ip_sip_server:5060 > > > 050620|11:48:29 > > > 050620|11:48:29 SIP/2.0 400 Bad request > > > 050620|11:48:29 Via: SIP/2.0/UDP > > > > > > ip_ata_1:5060;branch=z9hG4bK2b0f7845ea7a8edb;rport=5060;received=ip_ata_1_nat > > > 050620|11:48:29 From: > > > > <sip:332xxxx@domain.com;user=phone>;tag=2064987569 > > > 050620|11:48:29 To: > > > > <sip:0332yyyy@domain.com;user=phone>;tag=811613285 > > > 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 > > > 050620|11:48:29 CSeq: 3 PRACK > > > > > > > > > In the log from the called ATA there is no PRACK > > or > > > 400 error. > > > Can you tell me what do I have to change in > ser > > > configuration in order to allow routing of > PRACKs? > > > > > > Thanks. > > > > > > > __________________________________________________ > > > Do You Yahoo!? > > > Tired of spam? Yahoo! Mail has the best spam > > > protection around > > > http://mail.yahoo.com > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > > ___________________________________________________________________________ > > > > Appel audio GRATUIT partout dans le monde avec le > > nouveau Yahoo! Messenger > > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > > > ____________________________________________________ > > Yahoo! Sports > Rekindle the Rivalries. Sign up for Fantasy Football > > http://football.fantasysports.yahoo.com > __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] PSTN - SER Basic
by gunawan 27 Jun '05

27 Jun '05

hi, I'm newbiie in using ser ..... I am confused about how ser can connect to PSTN.... in my experiment i have two client using xlite.. the name are xlite1 and xlite2 with alias number 6276101 and 6276102. and I have two PSTN number provided by local communication provider one is +62(761)53803 and another one is +62(761)53939.. I use 53803 for my PSTN gateway... I use cisco gateway 2620XM and VIC2FXO... 1. If I want to call to PSTN number, what number should I dial from xlite1?????? Do I have to dial my PSTN Gateway first before I dial my destination number???? 2. If I want to make a call from PSTN to xlite1, what should I dial after I dial my PSTN gateway number??? 3. Do I have to add my destination PSTN number at subscriber table in my database????? Thanks before... ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com

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[Users] SER to PSTN problem, network blows away... :(
by Ozan Blotter 27 Jun '05

27 Jun '05

Dear List, SER at Public IP : 212.154.104.198 and operating good, there's no firewall or etc. PSTN Gateway located at Public IP : 212.154.59.48 and operating good, there's no firewall or etc. There are two ATA's for testing purpose, both are behind NAT (Zoom X4 ADSL Modem/Gateway) but please note that port forwarding is done from router so when a user dials another one, the voice traffce flows directly between users, not on SER and again no problem for parties while calling each other, they can hear/speak. Gateway already set up to accept calls from SER's IP. When i try to dial a cellular/landline number as i can see results from ngrep utility, SER routes the call to Gateway and step by step it's , INVITE, RINGING... i hear ringing tone, and then internet connection for computers/servers/voip devices in the ATA's sides are cut and ADSL modems halts, i have to switch it off and on. after a time, not more then 10 seconds my cellular rings, i open it and there's no voice since the connection was lost with ATA's... Finally, i can say that routing is working but why my internet connection is lost everytime when i try to make a cellular/landline call... I've tried to setup a STUN Server Setting in ATA's, i've tried with a public STUN Server 66.7.238.210:3478 again that did not solved my problem. Anyone have an idea ? Thanks, Ozan Blotter My SER version is 0.9.4 and below is my SER.CFG 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" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/textops.so" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 8192 ) { sl_send_reply("513", "Message too big"); break; }; if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); break; }; if (method=="REGISTER") { save("location"); break; }; if (lookup("aliases")) { if(!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; } } if (uri=~"^sip:9053[0-9]*@*") { rewritehost("212.154.XX.YY"); # prefix("201"); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; } This is NGREP log for a cellular call : [root@n212-154-104-198 openser]# ngrep 90532 port 5060 interface: eth0 (212.154.104.196/255.255.255.252) filter: ip and ( port 5060 ) match: 90532 ### U 85.96.192.140:5060 -> 212.154.104.198:5060 INVITE sip:905325123580@212.154.104.198 SIP/2.0..Via: SIP/2.0/UDP 85.96.192 .140:5060;branch=z9hG4bKaae6e7e772518fac..Max-Forwards: 70..To: <sip:905325 123580(a)212.154.104.198>..From: <sip:8334843600@212.154.104.198;user=phone>; tag=xIjNyIDMxID..Call-ID: 7626671D19D1212@85.96.192.140..CSeq: 1 INVITE..Co ntact: <sip:8334843600@85.96.192.140:5060>..Session-Expires: 100;refresher= uac..Supported: timer..Content-Type: application/sdp..Content-Length: 191.. ..v=0..o=8334843600 0 0 IN IP4 85.96.192.140..s=-..c=IN IP4 85.96.192.140.. t=0 0..m=audio 15034 RTP/AVP 18 18..a=rtpmap:18 G729/8000..a=sendrecv..a=rt pmap:18 G729/8000..a=sendrecv..a=ptime:20.. # U 212.154.104.198:5060 -> 85.96.192.140:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 85.96. 192.140:5060;branch=z9hG4bKaae6e7e772518fac..To: <sip:905325123580@212.154. 104.198>..From: <sip:8334843600@212.154.104.198;user=phone>;tag=xIjNyIDMxID ..Call-ID: 7626671D19D1212@85.96.192.140..CSeq: 1 INVITE..Server: OpenSer ( 0.9.4 (i386/linux))..Content-Length: 0..Warning: 392 212.154.104.198:5060 " Noisy feedback tells: pid=1969 req_src_ip=85.96.192.140 req_src_port=5060 in_uri=sip:905325123580@212.154.104.198 out_uri=sip:905325123580@212.154.59 .48 via_cnt==1".... # U 212.154.104.198:5060 -> 212.154.59.48:5060 INVITE sip:905325123580@212.154.59.48 SIP/2.0..Record-Route: <sip:212.154.1 04.198;ftag=xIjNyIDMxID;lr=on>..Via: SIP/2.0/UDP 212.154.104.198;branch=z9h G4bK1bbb.81b58062.0..Via: SIP/2.0/UDP 85.96.192.140:5060;branch=z9hG4bKaae6 e7e772518fac..Max-Forwards: 16..To: <sip:905325123580@212.154.104.198>..Fro m: <sip:8334843600@212.154.104.198;user=phone>;tag=xIjNyIDMxID..Call-ID: 76 26671D19D1212@85.96.192.140..CSeq: 1 INVITE..Contact: <sip:8334843600@85.96 .192.140:5060>..Session-Expires: 100;refresher=uac..Supported: timer..Conte nt-Type: application/sdp..Content-Length: 191....v=0..o=8334843600 0 0 IN I P4 85.96.192.140..s=-..c=IN IP4 85.96.192.140..t=0 0..m=audio 15034 RTP/AVP 18 18..a=rtpmap:18 G729/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrec v..a=ptime:20.. # U 212.154.59.48:5060 -> 212.154.104.198:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 212.154.104.198:5060;branch=z9hG4bK1bb b.81b58062.0..Via: SIP/2.0/UDP 85.96.192.140:5060;branch=z9hG4bKaae6e7e7725 18fac..Record-Route: <sip:212.154.104.198;ftag=xIjNyIDMxID;lr=on>..From: <s ip:8334843600@212.154.104.198;user=phone>;tag=xIjNyIDMxID..To: <sip:9053251 23580(a)212.154.104.198>;tag=ff0551002a01ff10ff0000ffffffff1e..Call-ID: 76266 71D19D1212@85.96.192.140..CSeq: 1 INVITE..Contact: <sip:905325123580@212.15 4.59.48;user=phone>..Server: MERA MSIP v.1.0.1..Content-Length: 0.... ## U 212.154.59.48:5060 -> 212.154.104.198:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 212.154.104.198:5060;branch=z9hG4bK1b bb.81b58062.0..Via: SIP/2.0/UDP 85.96.192.140:5060;branch=z9hG4bKaae6e7e772 518fac..Record-Route: <sip:212.154.104.198;ftag=xIjNyIDMxID;lr=on>..From: < sip:8334843600@212.154.104.198;user=phone>;tag=xIjNyIDMxID..To: <sip:905325 123580(a)212.154.104.198>;tag=ff0551002a01ff10ff0000ffffffff1e..Call-ID: 7626 671D19D1212@85.96.192.140..CSeq: 1 INVITE..Contact: <sip:905325123580@212.1 54.59.48;user=phone>..Server: MERA MSIP v.1.0.1..Content-Length: 0.... # U 212.154.104.198:5060 -> 85.96.192.140:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 85.96.192.140:5060;branch=z9hG4bKaae6 e7e772518fac..Record-Route: <sip:212.154.104.198;ftag=xIjNyIDMxID;lr=on>..F rom: <sip:8334843600@212.154.104.198;user=phone>;tag=xIjNyIDMxID..To: <sip: 905325123580(a)212.154.104.198>;tag=ff0551002a01ff10ff0000ffffffff1e..Call-ID : 7626671D19D1212@85.96.192.140..CSeq: 1 INVITE..Contact: <sip:905325123580 @212.154.59.48;user=phone>..Server: MERA MSIP v.1.0.1..Content-Length: 0... . ##

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[Users] AVPOPS module problem
by Alexandre Passito 27 Jun '05

27 Jun '05

Hi, I'm trying to implement a "black list" with avpops module, but I got some errors: This is my cfg file: # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei 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 #debug=7 #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" fifo_db_url="dbtext:///home/alexandre/Devel/openser-0.9.4/bin/dbtext" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/home/alexandre/Devel/ser-0.8.14/bin//lib/ser/modules/mysql.so" loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/dbtext.so" loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/sl.so" loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/tm.so" loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/rr.so" loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/maxfwd.so" loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/usrloc.so" loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/registrar.so" loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/avpops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/auth.so" loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_url", "dbtext:///home/alexandre/Devel/openser-0.9.4/bin/dbtext") modparam("usrloc", "db_mode", 2) modparam("auth_db", "db_url", "dbtext:///home/alexandre/Devel/openser-0.9.4/bin/dbtext") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("avpops", "avp_url", "dbtext:///home/alexandre/Devel/openser-0.9.4/bin/dbtext") modparam("avpops", "avp_table", "preferences") # 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"); 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("iptel.org", "subscriber")) { # www_challenge("iptel.org", "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 #Black-list!!! if (method=="INVITE" || method=="MESSAGE") { if ( avp_db_load("$to/username", "s:not_allow/preferences") && avp_check("s:not_allow", "eq/$from/gi") ) { sl_send_reply("403", "Forbidden - You are blocked!"); break; } } if (!t_relay()) { sl_reply_error(); }; lookup("aliases"); } My "preference" table looks like: uuid(str) username(str) domain(str,null) attribute(str) value(str) type(int) "":"peter":"200.129.163.139":"not_allow":"paul(a)200.129.163.212":"0" I'm using openser in to a local network. Its ip is 200.129.163.178. peter address is 200.129.163.139 paul address is 200.129.163.212 The result is that both UA call each one, despite peter blocks paul. Any help? Best regards -- Alexandre Passito Universidade Federal do Amazonas Departamento de Ciência da Computação LabVoIP - GRSD

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[Serusers] Serweb fatal error
by li wei 27 Jun '05

27 Jun '05

Hi there: CVS update from server to get the newest version of serweb. Tend to work with the SER version 0.9.X. Following the v 1.21 INSTALL document, when i type http://localhost/html/admin/index.php It returns me the following error: Fatal error: Call to undefined function: session_name() in /usr/local/www/serweb/phplib/session4.inc on line 204 Can anybody help? Thanks a lot. ____________________________________________________ Do you Yahoo!? Win a dream trip to Western Australia, courtesy of the NEW My Yahoo! http://au.my.yahoo.com/

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[Serusers] How can I see the status of SER?
by serusers 26 Jun '05

26 Jun '05

Hi all: I want to see the status of SER when it is running. How can I do? Thanks for any idea. Tommy

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RE: [Serusers] Problem with cancel a invite
by Toni Barata 26 Jun '05

26 Jun '05

Hi All, Another problem related with INVITE and CANCEL: A makes call to B. A cancels the call before B picks up the call. If the proxy receives the "100 trying" and "180 ringing" before A sends CANCEL, the CANCEL is sent by the proxy to B. But if the proxy only receives the "100 trying" before A sends CANCEL (a very fast cancel), The CANCEL is never sent by the proxy to B. Regards, Toni -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Marian Dumitru Sent: segunda-feira, 13 de Junho de 2005 11:24 To: Greger V. Teigre Cc: Anders Hansson; serusers(a)lists.iptel.org Subject: Re: [Serusers] Problem with cancel a invite Hi Greger, CANCEL has nothing to do with loose route - and it will never hit he loose route. The Route ser is set only when the dialog is established (200 ok); and CANCEL happens *before* the dialog completion. The big idea with CANCEL is to take care to route it *exactly* as its INVITE. As a hint, if you use TM, t_relay() all the time will take care to send the CANCEL to same destination as INVITE. regards, Marian Greger V. Teigre wrote: > Most likely you don't handle the CANCEL messages correctly. May be that > you don't do proper loose route handling. Take a look at the example > configs (and the Getting Started document) at http://onsip.org/ You > gain much be following the logic found there, they have been verified > for all kinds of situations... > g-) > > Anders Hansson wrote: > >> Hi all, >> I´ve got a problem when trying to cancel an invite. >> The situation: A makes a call to B (INVITE). B press the "hang up" >> button. An here comes the problem. >> From As side, everything is OK, but from Bs side, the phone is still >> ringing... >> How can a solve this problem? Anyone? >> >> Thanks, >> Anders -- Voice System http://www.voice-system.ro _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

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[Users] Problem starting Openser
by Andréa Giordanna 26 Jun '05

26 Jun '05

I am trying to start openser with command 'openserctl start' but i get the error: Starting SER : PID file /var/run/openser.pid does not exist -- SER start failed When I type just 'openser', with debug mode in ser.cfg it shows the following lines: ... 2(0) DEBUG: register_fifo_cmd: new command (pwd) registered 2(0) DEBUG: register_fifo_cmd: new command (arg) registered 2(0) DEBUG: register_fifo_cmd: new command (which) registered 2(0) DEBUG: register_fifo_cmd: new command (ps) registered 2(0) DEBUG: register_fifo_cmd: new command (kill) registered 2(0) WARNING: no fifo_db_url given - fifo DB commands disabled! 0(7904) DEBUG: init_mod_child (1): tm 0(7904) DEBUG: callid: '7ffdfefb-7904(a)127.0.0.1' 0(7904) DEBUG: init_mod_child (1): usrloc 0(7904) DBT:dbt_cache_get_db: looking for db /home/alexandre/Devel/openser-0.9.4/bin/dbtext! 0(7904) DBT:dbt_cache_get_db: db already cached! 0(7904) DEBUG: init_mod_child (1): avpops 0(7904) DEBUG: init_mod_child (1): auth_db 0(7904) DBT:dbt_cache_get_db: looking for db /home/alexandre/Devel/openser-0.9.4/bin/dbtext! 0(7904) DBT:dbt_cache_get_db: db already cached! It seems everything is alright, except for this line: 2(0) WARNING: no fifo_db_url given - fifo DB commands disabled! Is this the problem? If it is not, what the possible problem? Thanks, Andrea Giordanna _______________________________________________________ Yahoo! Acesso Grátis - Internet rápida e grátis. Instale o discador agora! http://br.acesso.yahoo.com/

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[Users] UAC REDIRECT - new module
by Bogdan-Andrei Iancu 26 Jun '05

26 Jun '05

Hi folks, new module is available in the development tree (CVS head, future OpenSER 0.10.x). UAC_REDIRECT module enhance OpenSER with the functionality of being able to handle (interpret, filter, account and follow) redirect responses ( 3xx replies class). The module provide a powerful mechanism for selecting and filtering the redirect contacts. For more info and docs, please see http://www.openser.org/docs/modules/0.10.x/uac_redirect.html any feedback - as testing or functionality- is greatly appreciated. regards, bogdan

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[Users] Problem with db_text module
by Alexandre Passito 26 Jun '05

26 Jun '05

I was trying to start db_text but I got the following error. I tried to put <socket> column where it requests after this error, but openser did not start. 0(0) DBT:dbt_cache_get_db: looking for db /home/alexandre/Devel/openser-0.9. 4/bin/dbtext! 0(0) DBT:dbt_cache_get_db: new db! 0(0) DBT:dbt_load_file: request for table [version] 0(0) DBT:dbt_load_file: db is [/home/alexandre/Devel/openser-0.9.4/bin/dbtex t] 0(0) DBT:dbt_load_file: loading file [/home/alexandre/Devel/openser-0.9.4/bi n/dbtext/version] 0(0) DBT: column[0] is STR! 0(0) DBT: column[1] is INT! 0(0) DTB:dbt_db_get_table: version Content of [version] table_name(str) table_version(int) subscriber:3 location:6 aliases:6 0(0) DBT:dbt_query: new res with 1 cols 0(0) DBT:dbt_result_new: new res with 1 cols Content of result table_version(int) 6 0(0) DBT:dbt_load_file: request for table [location] 0(0) DBT:dbt_load_file: db is [/home/alexandre/Devel/openser-0.9.4/bin/dbtex t] 0(0) DBT:dbt_load_file: loading file [/home/alexandre/Devel/openser-0.9.4/bi n/dbtext/location] 0(0) DBT: column[0] is STR! 0(0) DBT: column[1] is STR! 0(0) DBT: column[2] is STR! 0(0) DBT: column[3] is INT! 0(0) DBT: column[4] is DOUBLE! 0(0) DBT: column[5] is STR! 0(0) DBT: column[6] is INT! 0(0) DBT: column[7] is STR! 0(0) DBT: column[8] is INT! 0(0) DBT: column[9] is INT! 0(0) DBT: column[10] is INT! 0(0) DBT: column[11] is STR! 0(0) DBT: column[12] is STR! 0(0) DTB:dbt_db_get_table: location Content of [location] username(str) domain(str,null) contact(str,null) expires(int,null) q(double,n ull) callid(str,null) cseq(int,null) last_modified(str) replicate(int,null) s tate(int,null) flags(int) user_agent(str) received(str) 0(0) DBT:dbt_get_refs: ERROR column <socket> not found 0(0) DBT:dbt_query: error while querying table! 0(0) preload_udomain(): Error while doing db_query 0(0) register_udomain(): Error while preloading domain 'location' 0(0) domain_fixup(): Error while registering domain ERROR: error -1 while trying to fix configuration 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(0) DBT:destroy ... 0(7157) shm_mem_destroy 0(7157) destroying the shared memory lock My tables are configured as follow: version table_name(str) table_version(int) subscriber:3 location:6 aliases:6 subscriber username(str) domn(str) password(str) first_name(str) last_name(str) phone(str) email_address(str) datetime_created(int) datetime_modified(int) confirmation(str) flag(str) sendnotification(str) greeting(str) ha1(str) ha1b(str) perms(str) allow_find(str) timezone(str,null) rpid(str,null) uuid(str,null) location username(str) domain(str,null) contact(str,null) expires(int,null) q(double,null) callid(str,null) cseq(int,null) last_modified(str) replicate(int,null) state(int,null) flags(int) user_agent(str) received(str) aliases username(str) domain(str,null) contact(str,null) expires(int,null) q(double,null) callid(str,null) cseq(int,null) last_modified(str) replicate(int,null) state(int,null) flags(int) user_agent(str) received(str) Uou! It's hard to start... :-) Any help? Best regards -- Alexandre Passito Universidade Federal do Amazonas Departamento de Ciência da Computação LabVoIP - GRSD

2 1

[Serusers] SER Configuration
by Abdul Lateef 26 Jun '05

26 Jun '05

Hello Friends, I am newly in this list and also with SER. I installed the SER 0.8.13 in my Redhat 7.3 box. But i don't have any idea how to configure well. and what are the configurations files. i checked using following commond but i am cannot understand why it is not starting. [root@voipbilling root]# /usr/local/sbin/serctl start Starting SER : PID file /var/run/ser.pid does not exist -- SER start failed [root@voipbilling root]# /usr/local/sbin/serctl stop Stopping SER : No PID file found! SER probably not running [root@voipbilling root]# /usr/local/sbin/serctl restart Stopping SER : No PID file found! SER probably not running [root@voipbilling root]# /usr/local/sbin/ser Listening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 10.0.0.28 [10.0.0.28]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 tcp: 10.0.0.28 [10.0.0.28]:5060 Aliases: tcp: voipbilling:5060 tcp: SpeedTouch.lan:5060 udp: voipbilling:5060 udp: SpeedTouch.lan:5060 If i wnat to change the udp/TCP IP and Aliases from where i can change? Because i have multi IPs in the same server and i want to use the public IP. Please let me know if there is such as tutorial and configurations guide is available? Thank You Yours, Abdul Lateef Computer Programmer HATIF COM Mob: +974 - 5405022 Tel: +974 - 4883068 ICQ: 276994704 YM!: abdul_zu Fax: +974 - 4883063 Doha Qatar http://www.hatif.com __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

3 2

[Serusers] multiple domains and aliases to link them
by Rosario Pingaro 26 Jun '05

26 Jun '05

I have ser configured and workin fine with multiple domains. I assign an unique numeric alias for each user not realted to the domain. So If I have an alias 0823384(a)domain1.com can't have 0823384(a)domain2.com link to another user. Now I'd like to let the user of doamin1.com using the regular phone to call the user of domain2.com trough SER (I am using 0.9.2); I am not able to figure out how to modify the ser.cfg to accomplish this. Could I rewrite the uri if I don't find the alias and let them find again with the new domain? Something like this: if (!lookup("alias")){ if (uri=~"domain1,com") { rewritehost("domain2.com"); lookup("alias"); # if it is not about the other domain it come back if (!lookup("alias")){ rewritehost("domain1.com"); } } else if (uri=~"domain2,com") { rewritehost("domain1.com"); lookup("alias"); # if it is not about the other domain it come back if (!lookup("alias")){ rewritehost("domain2.com"); } } } Hope o receive some feedback. Thanks

2 2

[Serusers] INVITEs and 408 retransmissions
by Jorge Crichigno 26 Jun '05

26 Jun '05

Hello, I was testing ser-0.9.0 in a scenario with two ip phones. When I disconnect one of them from the network, and the other phone call to the disconnected one and then cancel the call before the timeout happens, the following packets are sent: A SER B (disconnected from the network) INVITE-> <-100 INVITE-> INVITE-> INVITE-> CANCEL-> <-200 ACK-> ACK-> INVITE-> INVITE-> INVITE-> INVITE-> INVITE-> INVITE-> <-408 <-408 <-408 <-408 <-408 <-408 <-408 <-408 <-408 I'm using t_relay to route the INVITEs. Thank in advance!! Jorge

2 1

[Serusers] PRACK in ser
by Dana Turcanu 25 Jun '05

25 Jun '05

Hello, Is PRACK method supported in ser? When an equipment sends a PRACK the server responds with "400 Bad Request" and the call is disconnected. Thanks. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

3 3

[Users] formatted printing - unifying specifiers
by Daniel-Constantin Mierla 25 Jun '05

25 Jun '05

Hello, I send this mail to users@ mailing list, maybe someone gets a good feedback how easy to use and intuitive is this proposal, since it is not a technical issue. We want to unify the format used by some modules to print specific items -- mainly was xlog and now acc (to print extra information) and tm (tw_append parameter -- to add extra info with t_write_fifo()). We want to add support in avpops to be able to compose a string from such specifiers and store it in an avp and to textops to be able to search/replace dynamic strings. Now, there are several ways to specify an item to be replaced by a value. - xlog - the first that introduced specifiers, started from "printf" model and uses "%" to mark the beginning of a specifier (%xx different items, %{hdr_name} headers and %{:avp_name} avps). - acc - started from avpops approach and uses avp/avp_name and hdr/hdr_name - tm - uses avp[avp_name] and hdr[hdr_name] It is time to bring them to same format to be easier and intuitive to use. Our proposal is to use the "array" model: - $avp[name] - to specify an avp via name or alias - $hdr[name] - to specify a header - $msg[name] - to specify specific parts of a SIP message (e.g., method, request uri, status, body ...) - $sys[name] - for anything that is not related to a sip message (e.g., time, pid ...) Since there are important changes, the migration will be made in two steps. 1) move headers and avps to this format and keep the other dedicated specifiers from xlog as they are ($ followed by two letters) 2) move the dedicated specifiers to the array the belong to The big advantage is that once a new specifier is added it will be visible in all modules, avoiding duplicated code and maintenance overhead. Any feedback will be appreciated. Daniel

2 2

[Users] Dbtext module problem
by Alexandre Passito 25 Jun '05

25 Jun '05

Hi everybody, I was trying to install dbtext module, but i got the following error: 0(0) unixsock_register_cmd: New command (ul_stats) registered 0(0) unixsock_register_cmd: New command (ul_rm) registered 0(0) unixsock_register_cmd: New command (ul_rm_contact) registered 0(0) unixsock_register_cmd: New command (ul_dump) registered 0(0) unixsock_register_cmd: New command (ul_flush) registered 0(0) unixsock_register_cmd: New command (ul_add) registered 0(0) unixsock_register_cmd: New command (ul_show_contact) registered 0(0) find_mod_export: <db_use_table> in module /home/alexandre/Devel/openser-0.9.4/bin/dbtext not found 0(0) bind_dbmod: Module /home/alexandre/Devel/openser-0.9.4/bin/dbtext does not export db_use_table function 0(0) ERROR: mod_init(): Can't bind database module 0(0) init_mod(): Error while initializing module usrloc ERROR: error while initializing modules 0(0) DEBUG: tm_shutdown : start 0(0) DEBUG: unlink_timer_lists : emptying DELETE list 0(0) DEBUG: tm_shutdown : emptying hash table 0(0) DEBUG: tm_shutdown : releasing timers 0(0) DEBUG: tm_shutdown : removing semaphores 0(0) DEBUG: tm_shutdown : destroying tmcb lists 0(0) DEBUG: tm_shutdown : done 0(0) DBT:destroy ... 0(24535) shm_mem_destroy 0(24535) destroying the shared memory lock the openser.cfg is: # # $Id: ser.cfg,v 1.21.4.1 2003/11/10 15:35:15 andrei 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 debug=7 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/ser_fifo" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database #loadmodule "/home/alexandre/Devel/ser-0.8.14/bin//lib/ser/modules/mysql.so" loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/dbtext.so" loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/sl.so" loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/tm.so" loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/rr.so" loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/maxfwd.so" loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/usrloc.so" loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/registrar.so" loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/avpops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/auth.so" loadmodule "/home/alexandre/Devel/openser-0.9.4/bin//lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_url", "/home/alexandre/Devel/openser-0.9.4/bin/dbtext") modparam("usrloc", "db_mode", 2) modparam("auth_db", "db_url", "/home/alexandre/Devel/openser-0.9.4/bin/dbtext") modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") # 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"); 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 (...) Any Help? Best regards, -- Alexandre Passito Universidade Federal do Amazonas Departamento de Ciência da Computação LabVoIP - GRSD

2 1

[Users] Module AVPOPS question
by Alexandre Passito 25 Jun '05

25 Jun '05

Hi everybody, I'm starting to use openser and I have some questions about avpops module. Why this module is not implemented together with SER? I was trying to compile it with SER, but I got some errors. Is this module usable with dbtext module? Which modules are not present at SER but are present at Openser? Best regards, -- Alexandre Passito Universidade Federal do Amazonas Departamento de Ciência da Computação LabVoIP - GRSD

2 1

Re: [Serusers] Symmetric NAT
by chiahuey＠genme.com 25 Jun '05

25 Jun '05

that means there's no way to differentiate which NAT the UA is from in the routing script alone? any other way beside using STUN? regards, Chia -----Original message----- From: Klaus Darilion klaus.mailinglists(a)pernau.at Date: Fri, 24 Jun 2005 19:44:28 +0800 To: Chia Huey Lim chiahuey(a)genme.com Subject: Re: [Serusers] Symmetric NAT > The NAT detection must be done by the client, e.g. using STUN. If the > client detects symmetric NAT, it has to use the private IP in Via. Thus, > the proxy can detect that this client is unable of NAT traversal (using > the NAT checks from nathelper) > > regards, > klaus > > > Chia Huey Lim wrote: > > Is there anyway to test if the UA is from symmetric NAT? > > > > > > > > Regards, > > > > Chia > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers >

1 0

[Serusers] AVPOPS module error
by Alexandre Passito 25 Jun '05

25 Jun '05

Hi everybody, I'm trying to use module avpops in to ser. When I try to load the module with: loadmodule "/usr/local/lib/ser/modules/avpops.so" I get the following error: 0(22059) loading module /usr/local/lib/ser/modules/avpops.so 0(22059) ERROR: load_module: could not open module </usr/local/lib/ser/modules/avpops.so>: /usr/local/lib/ser/modules/avpops.so: undefined symbol: get_avp_name 0(22059) parse error (38,13-77): failed to load module How can I fix it? Best regards, -- Alexandre Passito Universidade Federal do Amazonas Departamento de Ciência da Computação LabVoIP - GRSD

1 0

[Serusers] Load Balance & request URI
by a c 24 Jun '05

24 Jun '05

hello, I have SER doing load balance across app servers. I would like SER to change the request URI with the ipaddress of the server in the returned destination set{using dispatcher module to Load balance}. How would I do this? rewritehost(?) regards, ac __________________________________ Discover Yahoo! Find restaurants, movies, travel and more fun for the weekend. Check it out! http://discover.yahoo.com/weekend.html

4 9

[Serusers] t_lookup_request, t_relay
by Jorge Crichigno 24 Jun '05

24 Jun '05

Hello, I have the following problem using the below lines in ser.cfg: if (t_lookup_request()){ do some tasks, but not BREAK; }; if (!t_relay()) { sl_reply_error(); break; }; With this lines, the errors shows below are displayed: ERROR: t_newtran: transaction already in process 0x40216208 35(28756) ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (1/SL) I'm using SER 0.9.0. Thank you! Jorge

2 2

Re: [Serusers] SER losing SIP registrations
by chiahuey＠genme.com 24 Jun '05

24 Jun '05

na, i didn't hack it. it's in ser.cfg regards, Chia -----Original message----- From: Juan Carlos Castro y Castro jcastro(a)instant.com.br Date: Fri, 24 Jun 2005 23:08:13 +0800 To: serusers(a)lists.iptel.org Subject: Re: [Serusers] SER losing SIP registrations > Where did you change that, in ser.cfg? Or did you hack the source? > (Sorry, I'm still a newbie on this thing) > > Chia Huey Lim wrote: > > >Wow!! It works! As I replaced fix_nated_contact with fix_nated_register for > >REGISTER messages, the UA has no problem re-registering. Never thought that > >it could be the contact that is causing the problem. > >Thank you so much, Jan. > > >

1 0

[Serusers] error msg
by Jack Wei 24 Jun '05

24 Jun '05

what does the following error message mean? my system ran out of memory? 12222220002(a)10.1.201.191 is being t_relay()ed 10(3135) ERROR: sip_msg_cloner: cannot allocate memory 10(3135) ERROR: new_t: out of mem: 10(3135) ERROR: t_newtran: new_t failed 10(3135) 12222220002(a)10.1.201.191 t_relay() error 10(3135) ERROR: sl_reply_error used: I'm terribly sorry, server error occured (1/SL) 9(3134) 0(3124) child process 3134 exited by a signal 11 0(3124) core was not generated 0(3124) INFO: terminating due to SIGCHLD 5(3130) 10(3135) INFO: signal 15 received INFO: signal 15 received 7(3132) INFO: signal 15 received 14(3140) INFO: signal 15 received 12(3137) INFO: signal 15 received 11(3136) INFO: signal 15 received 1(3126) INFO: signal 15 received 2(3127) INFO: signal 15 received 3(3128) INFO: signal 15 received 4(3129) INFO: signal 15 received 6(3131) INFO: signal 15 received 8(3133) INFO: signal 15 received 13(3138) INFO: signal 15 received 15(3141) INFO: signal 15 received 16(3142) INFO: signal 15 received 17(3143) INFO: signal 15 received 19(3145) INFO: signal 15 received 18(3144) INFO: signal 15 received Jack Wei __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

Re: [Serusers] SER losing SIP registrations
by Juan Carlos Castro y Castro 24 Jun '05

24 Jun '05

Where did you change that, in ser.cfg? Or did you hack the source? (Sorry, I'm still a newbie on this thing) Chia Huey Lim wrote: >Wow!! It works! As I replaced fix_nated_contact with fix_nated_register for >REGISTER messages, the UA has no problem re-registering. Never thought that >it could be the contact that is causing the problem. >Thank you so much, Jan. >

2 1

RE: [Serusers]Portal for forking call to preferred end device-sequential ringing
by Aisling 24 Jun '05

24 Jun '05

Hi, My mind has changed about how I'm going to approach this a million times!! ;) Ok following on from the below email, CPL doesn't seem such a good option particularly if NAT is coming into the equation(which in my case it will be). So the approach I think might work at the moment is: 1) Using some language (java servlet in the web interface probably) every 20/30 seconds issue the "serctl ul show 2000" to read the contact addresses for user 2000.This loop will hopefully keep a recent enough record of the contact addresses so devices can register whenever. 2) This is the tricky step which I need advise on. I must somehow know which contact address to associate with the device the user has chosen as the preferred device (e.g. desktop sip phone). At the moment I'm thinking the user must either enter the device ip addresses manually so I can search the contact based on IP address (This sucks in terms of NAT, scalability and losing mobility over networks) OR I could perhaps stipulate that a user cannot have two of the same device e.g. cant have two BT100 hardphones and I could search the location table based on model. I think either of these options aren't great....Any better ideas? 3) Then once I (somehow) know the devices contact address, I'll get my interface to issues the "serctl fifo ul_add 2000@AOR 2000@contact address 3600 qvalue 0 0" command and set that preferred contact address to the highest q value. Bar the obvious big hole that is step 2, does that all sound feasible?....Or in yere opinions is the CPL route the best way to approach...I'm try to figure out which one leads to the least amount of problems in terms of nat, scalability, mobility etc......All options seem to have down-sides but its difficult to evaluate which option is the most "doable" Thanks again, Aisling -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: 24 June 2005 08:36 To: Aisling Cc: 'Greger V. Teigre'; jh(a)tutpro.com Subject: Re: [Serusers]CPL - Portal for forking call to preferred end device-sequential ringing Hi Aisling, Aisling wrote: >Hi Bogdon, > > > >>"even if you use CPL, you need to place in each location node the >>contacts (and not the AOR) of the user, contacts which are dynamically >>stored only in user location. >> >> > > > >>what you are describing below can be possible only if you a user (as >>person) has different sip accounts (SIP users) for each of his devices. >> >> > > > >>Other way I don't see how you can place in the CPL script the "phone1" >>and "phone2" addresses." >> >> > >I was thinking that the user would first need to manage their contact >addresses via the web interface. So they would have to enter the IP >address for each device. > >e.g. > >pda xxx.xxx.xxx.150 >pc xxx.xxx.xxx.151 >laptop xxx.xxx.xxx.152 > >Then when a user wants a particular sequential ring order of perhaps >pc->laptop->pda, I would provision the cpl script and retrieve the >addresses from the above manageContacts to provide the "phone1", >"phone2" addresses. i.e. "phone1" could be sip:2000@xxx.xxx.xxx.xxx151 >for the pc and "phone2" could be sip:2000@xxx.xxx.xxx.152 for the >laptop. > >Is this scenario possible? > > yes, the scenario is possible, but there are couple of drawback: - you will give up the SIP-like registration mechanism (users have to upload by hand the IPs), so loose the mobility part - if user moved the laptop in a different network, he will have to update manually the contact on server (update which is normally done transparently via registration). - choosing CPL sometime proves to be a real act of bravery :) - first the scripts are difficult to provisions, second to integrate cpl-c with nat traversal is quite tricky (or with other services). >Also (and forgive me if this question is ignorant) couldn't I just >retrieve the contact address (not AOR) from the user location? > > via CPL? yes you can do it - use the lookup("registration") node. Then you can proxy what you got in a sequential way (see the "ordering" parameter of proxy node). regards, bogdan >Thanks, >Aisling. > > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: 22 June 2005 12:45 >To: Aisling >Cc: 'Greger V. Teigre'; jh(a)tutpro.com >Subject: Re: [Serusers]CPL - Portal for forking call to >preferredenddevice-sequential ringing > >Hi Aisling, > >There are many ways to implement serial forking - like using CPL, LCR or > >AVPOPS. >But the main idea (if I getting right from your initial email) is to >allow the user to order his contacts for the same account (like account >userx that have contact_pda_x, contact_PC_x, contact_office_x, etc). So >you have all this contacts in user location, you have several mechanism >for serial forking, but you are missing the link - to get the contacts >from user location and to feed any of the serial forking mechanism. This > >is the big problem. > >even if you use CPL, you need to place in each location node the >contacts (and not the AOR) of the user, contacts which are dynamically >stored only in user location. > >what you are describing below can be possible only if you a user (as >person) has different sip accounts (SIP users) for each of his devices. >Other way I don't see how you can place in the CPL script the "phone1" >and "phone2" addresses. > >and just to answer you to the CPL- related questions (from >implementation point of view): >- each user can have only one script - if you load a new script, the >previous one will be lost (overwritten). If you can different services >via CPL (like screening and no-answer), you need to mix them in the same > >script during provisioning. >- if you want to delete a CPL script via FIFO, use the REMOVE_CPL >command: > serctl fifo REMOVE_CPL user@domain > > >regards, >bogdan > > > > >Aisling wrote: > > > >>Hello, >> >>Many thanks for the ideas so far. >>I looked at CPL Greger and I think that provides a very simple solution >>to this - a simple Call forward on no answer script. >> >>I have an included an example that I came across below. If a user wants >>to modify the order of the devices a call should be sent to, then I >>simply have to retrieve the information from the user via the web >>interface and provision a new cpl script. I think this solves the >>problem - Does anyone foresee any problems with this or think it has >>disadvantages? >> >>I do have two minor questions if I am to go ahead with this direction >>though: >>1) If a particular user already has a cpl script e.g. a call screening >>script uploaded to the database and they then upload this forward on no >>answer script, will it overwrite the original script? i.e. can there >>only be one cpl script per user? >> >>2)How can a cpl script be "undone" or deleted? Must it be overwritten >> >> >or > > >>is there a way of simply removing it(without using mysql commands)? >> >>Example CPL: Call Forward on No Answer >> >><?xml version="1.0"> >> >><cpl> >> <subaction id="phone2"> >> <location url="sip:2000@phone2"> >> <proxy /> >> </location> >> </subaction> >> >> <incoming> >> <location url="sip:2000@phone1"> >> <proxy timeout="8"> >> <noanswer> >> <sub ref="phone2"/> >> </noanswer> >> </proxy> >> </location> >> </incoming> >></cpl> >> >>Many Thanks, >>Aisling. >> >>-----Original Message----- >>From: Greger V. Teigre [mailto:greger@teigre.com] >>Sent: 22 June 2005 05:42 >>To: Aisling O'Driscoll; jh(a)tutpro.com >>Cc: samuel.osorio(a)nl.thalesgroup.com; ashling.odriscoll(a)cit.ie; >>serusers(a)lists.iptel.org >>Subject: Re: [Serusers] Portal for forking call to >>preferredenddevice-sequential ringing >> >>Aisling O'Driscoll wrote: >> >> >> >> >>>Ok, just to recap - cos Im getting a little bit confused ;) >>> >>>I have two choices(?) >>> >>>1. Somehow invoke sipsak to configure permanent addresses with a >>>particular q value. >>>2. Develop a FIFO method to change q value. >>> >>> >>> >>> >>Yes. Except that I don't know if anybody verified that q value cannot >> >> >be > > >>changed with today's FIFO. I just asked the question... >> >> >> >> >> >>>Am I correct in thinking directly modifying the usrloc table in the >>>database is out of the question because the changes cant be updated >>>except by SER itself in which case a reboot would required - Correct? >>> >>> >>> >>> >>Correct. "SER itself" is here either serctl, sipsak or FIFO. >> >> >> >> >> >>>Also lcr module (load_contacts() etc) isnt suitable for per user >>>configurable sequential forking? >>> >>> >>> >>> >>I don't know about that. There is definitely a q-value based >>functionality >>there. >> >> >> >> >> >>>Have others tried to implement similar functionality or is it usually >>>a generic site wide sequential forking policy? >>> >>> >>> >>> >>I think using CPL could be an option. Have you looked at CPL and the >> >> >cpl > > >>module? >>g-) >> >> >> >> >> >>>Many thanks for the opinions and help so far. >>>Aisling. >>> >>> >>> >>> >>> >>>>---- Original Message ---- >>>>From: jh(a)tutpro.com >>>>To: greger(a)teigre.com >>>>Subject: Re: [Serusers] Portal for forking call to >>>>preferredenddevice-sequential ringing >>>>Date: Tue, 21 Jun 2005 21:41:54 +0300 >>>> >>>> >>>> >>>> >>>> >>>>>Greger V. Teigre writes: >>>>> >>>>> >>>>> >>>>> >>>>> >>>>>>:-D Yes, that is true. But I didn't know that you could change >>>>>> >>>>>> >>>>>> >>>>>> >>>>q-value in >>>> >>>> >>>> >>>> >>>>>>FIFO? >>>>>> >>>>>> >>>>>> >>>>>> >>>>>if that is not possible, then you can always use sipsak to install >>>>>"permanent" registrations with a given q value. sipsak has an >>>>>additional advantage over fifo in that you can apply permissions >>>>> >>>>> >>>>> >>>>> >>>>check >>>> >>>> >>>> >>>> >>>>>to sipsak registration, but not to fifo registration. >>>>> >>>>>-- juha >>>>> >>>>> -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

1 0

[Serusers] Serweb + SER 0.9.0 ('perms' problem)
by ODC 24 Jun '05

24 Jun '05

Serweb requires 'perms' table in the Ser database, however 'perms' is not present in SER 0.9.0. Thus, current version of serweb does not work with SER 0.9.0. Is there a practical solution? Php code can not be changed easily; can 'perms' be added? Where is the newer version of Serweb stored (can the Beta release be downloaded)? Thanks in advance, Dev --------------------------------- Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football

2 5

[Serusers] SER to PSTN problem, network blows away... :(
by Ozan Blotter 24 Jun '05

24 Jun '05

Dear List, SER at Public IP : 212.154.104.198 and operating good, there's no firewall or etc. PSTN Gateway located at Public IP : 212.154.59.48 and operating good, there's no firewall or etc. There are two ATA's for testing purpose, both are behind NAT (Zoom X4 ADSL Modem/Gateway) but please note that port forwarding is done from router so when a user dials another one, the voice traffce flows directly between users, not on SER and again no problem for parties while calling each other, they can hear/speak. Gateway already set up to accept calls from SER's IP. When i try to dial a cellular/landline number as i can see results from ngrep utility, SER routes the call to Gateway and step by step it's , INVITE, RINGING... i hear ringing tone, and then internet connection for computers/servers/voip devices in the ATA's sides are cut and ADSL modems halts, i have to switch it off and on. after a time, not more then 10 seconds my cellular rings, i open it and there's no voice since the connection was lost with ATA's... Finally, i can say that routing is working but why my internet connection is lost everytime when i try to make a cellular/landline call... I've tried to setup a STUN Server Setting in ATA's, i've tried with a public STUN Server 66.7.238.210:3478 again that did not solved my problem. Anyone have an idea ? Thanks, Ozan Blotter My SER version is 0.9.4 and below is my SER.CFG 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" loadmodule "/usr/lib/openser/modules/sl.so" loadmodule "/usr/lib/openser/modules/tm.so" loadmodule "/usr/lib/openser/modules/rr.so" loadmodule "/usr/lib/openser/modules/maxfwd.so" loadmodule "/usr/lib/openser/modules/usrloc.so" loadmodule "/usr/lib/openser/modules/registrar.so" loadmodule "/usr/lib/openser/modules/textops.so" modparam("usrloc", "db_mode", 0) modparam("rr", "enable_full_lr", 1) route{ if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 8192 ) { sl_send_reply("513", "Message too big"); break; }; if (!method=="REGISTER") record_route(); if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { append_hf("P-hint: outbound\r\n"); route(1); break; }; if (method=="REGISTER") { save("location"); break; }; if (lookup("aliases")) { if(!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; } } if (uri=~"^sip:9053[0-9]*@*") { rewritehost("212.154.XX.YY"); # prefix("201"); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { if (!t_relay()) { sl_reply_error(); }; } This is NGREP log for a cellular call : [root@n212-154-104-198 openser]# ngrep 90532 port 5060 interface: eth0 (212.154.104.196/255.255.255.252) filter: ip and ( port 5060 ) match: 90532 ### U 85.96.192.140:5060 -> 212.154.104.198:5060 INVITE sip:905325123580@212.154.104.198 SIP/2.0..Via: SIP/2.0/UDP 85.96.192 .140:5060;branch=z9hG4bKaae6e7e772518fac..Max-Forwards: 70..To: <sip:905325 123580(a)212.154.104.198>..From: <sip:8334843600@212.154.104.198;user=phone>; tag=xIjNyIDMxID..Call-ID: 7626671D19D1212@85.96.192.140..CSeq: 1 INVITE..Co ntact: <sip:8334843600@85.96.192.140:5060>..Session-Expires: 100;refresher= uac..Supported: timer..Content-Type: application/sdp..Content-Length: 191.. ..v=0..o=8334843600 0 0 IN IP4 85.96.192.140..s=-..c=IN IP4 85.96.192.140.. t=0 0..m=audio 15034 RTP/AVP 18 18..a=rtpmap:18 G729/8000..a=sendrecv..a=rt pmap:18 G729/8000..a=sendrecv..a=ptime:20.. # U 212.154.104.198:5060 -> 85.96.192.140:5060 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 85.96. 192.140:5060;branch=z9hG4bKaae6e7e772518fac..To: <sip:905325123580@212.154. 104.198>..From: <sip:8334843600@212.154.104.198;user=phone>;tag=xIjNyIDMxID ..Call-ID: 7626671D19D1212@85.96.192.140..CSeq: 1 INVITE..Server: OpenSer ( 0.9.4 (i386/linux))..Content-Length: 0..Warning: 392 212.154.104.198:5060 " Noisy feedback tells: pid=1969 req_src_ip=85.96.192.140 req_src_port=5060 in_uri=sip:905325123580@212.154.104.198 out_uri=sip:905325123580@212.154.59 .48 via_cnt==1".... # U 212.154.104.198:5060 -> 212.154.59.48:5060 INVITE sip:905325123580@212.154.59.48 SIP/2.0..Record-Route: <sip:212.154.1 04.198;ftag=xIjNyIDMxID;lr=on>..Via: SIP/2.0/UDP 212.154.104.198;branch=z9h G4bK1bbb.81b58062.0..Via: SIP/2.0/UDP 85.96.192.140:5060;branch=z9hG4bKaae6 e7e772518fac..Max-Forwards: 16..To: <sip:905325123580@212.154.104.198>..Fro m: <sip:8334843600@212.154.104.198;user=phone>;tag=xIjNyIDMxID..Call-ID: 76 26671D19D1212@85.96.192.140..CSeq: 1 INVITE..Contact: <sip:8334843600@85.96 .192.140:5060>..Session-Expires: 100;refresher=uac..Supported: timer..Conte nt-Type: application/sdp..Content-Length: 191....v=0..o=8334843600 0 0 IN I P4 85.96.192.140..s=-..c=IN IP4 85.96.192.140..t=0 0..m=audio 15034 RTP/AVP 18 18..a=rtpmap:18 G729/8000..a=sendrecv..a=rtpmap:18 G729/8000..a=sendrec v..a=ptime:20.. # U 212.154.59.48:5060 -> 212.154.104.198:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 212.154.104.198:5060;branch=z9hG4bK1bb b.81b58062.0..Via: SIP/2.0/UDP 85.96.192.140:5060;branch=z9hG4bKaae6e7e7725 18fac..Record-Route: <sip:212.154.104.198;ftag=xIjNyIDMxID;lr=on>..From: <s ip:8334843600@212.154.104.198;user=phone>;tag=xIjNyIDMxID..To: <sip:9053251 23580(a)212.154.104.198>;tag=ff0551002a01ff10ff0000ffffffff1e..Call-ID: 76266 71D19D1212@85.96.192.140..CSeq: 1 INVITE..Contact: <sip:905325123580@212.15 4.59.48;user=phone>..Server: MERA MSIP v.1.0.1..Content-Length: 0.... ## U 212.154.59.48:5060 -> 212.154.104.198:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 212.154.104.198:5060;branch=z9hG4bK1b bb.81b58062.0..Via: SIP/2.0/UDP 85.96.192.140:5060;branch=z9hG4bKaae6e7e772 518fac..Record-Route: <sip:212.154.104.198;ftag=xIjNyIDMxID;lr=on>..From: < sip:8334843600@212.154.104.198;user=phone>;tag=xIjNyIDMxID..To: <sip:905325 123580(a)212.154.104.198>;tag=ff0551002a01ff10ff0000ffffffff1e..Call-ID: 7626 671D19D1212@85.96.192.140..CSeq: 1 INVITE..Contact: <sip:905325123580@212.1 54.59.48;user=phone>..Server: MERA MSIP v.1.0.1..Content-Length: 0.... # U 212.154.104.198:5060 -> 85.96.192.140:5060 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 85.96.192.140:5060;branch=z9hG4bKaae6 e7e772518fac..Record-Route: <sip:212.154.104.198;ftag=xIjNyIDMxID;lr=on>..F rom: <sip:8334843600@212.154.104.198;user=phone>;tag=xIjNyIDMxID..To: <sip: 905325123580(a)212.154.104.198>;tag=ff0551002a01ff10ff0000ffffffff1e..Call-ID : 7626671D19D1212@85.96.192.140..CSeq: 1 INVITE..Contact: <sip:905325123580 @212.154.59.48;user=phone>..Server: MERA MSIP v.1.0.1..Content-Length: 0... . ##

1 0

[Serusers] Problems making acc module with radius enabled
by Cameron Beattie 24 Jun '05

24 Jun '05

I am trying to enable radius accounting. I have uncommented the following lines in the acc Makefile: DEFS+=-DRAD_ACC -I$(LOCALBASE)/include LIBS=-L$(LOCALBASE)/lib -lradiusclient I then ran the following: make proper include_modules="mysql" make make modules include_modules="mysql" make install include_modules="mysql" But I get the following message when I try to run ser, which I think indicates that radius accounting hasn't compiled properly: parameter <radius_config> not found in module <acc> I noticed the following on the make install, which doesn't look promising. Any ideas would be appreciated: make[1]: Entering directory `/usr/src/sip_router/modules/acc' gcc -fPIC -DPIC -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.10.99-dev7"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.3"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local/etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -DRAD_ACC -I/usr/local/include -c acc.c -o acc.o acc.c:53:29: radiusclient-ng.h: No such file or directory acc.c:603: error: syntax error before "phrase2code" acc.c:604: warning: return type defaults to `int' acc.c: In function `phrase2code': acc.c:605: error: `UINT4' undeclared (first use in this function) acc.c:605: error: (Each undeclared identifier is reported only once acc.c:605: error: for each function it appears in.) acc.c:605: error: syntax error before "code" acc.c:609: error: `code' undeclared (first use in this function) acc.c: At top level: acc.c:618: error: syntax error before "rad_status" acc.c:619: warning: return type defaults to `int' acc.c: In function `acc_rad_request': acc.c:640: error: `VALUE_PAIR' undeclared (first use in this function) acc.c:641: error: `UINT4' undeclared (first use in this function) acc.c:641: error: syntax error before "av_type" acc.c:655: error: invalid lvalue in assignment acc.c:666: error: `av_type' undeclared (first use in this function) acc.c:667: warning: implicit declaration of function `rc_avpair_add' acc.c:764: warning: implicit declaration of function `rc_acct' acc.c:764: error: `OK_RC' undeclared (first use in this function) acc.c:768: warning: implicit declaration of function `rc_avpair_free' make[1]: *** [acc.o] Error 1 make[1]: Leaving directory `/usr/src/sip_router/modules/acc' make[1]: Entering directory `/usr/src/sip_router/modules/radius_acc' make[1]: *** No targets specified and no makefile found. Stop. make[1]: Leaving directory `/usr/src/sip_router/modules/radius_acc' Regards Cameron

3 6

[Serusers] Symmetric NAT
by Chia Huey Lim 24 Jun '05

24 Jun '05

Is there anyway to test if the UA is from symmetric NAT? Regards, Chia

3 2

Re: [serusers] calling problem
by Java Rockx 24 Jun '05

24 Jun '05

Michael, I'm quite certain that if the value of the "flags" column in the location table is set to 1 then the SER registrar module thinks that the REGISTER message is from a NATed client. By default, SER uses flag 4 to indicate NAT status in the ser.cfg file. So if you have called setflag(4) somewhere before save("location") then you are informing SER that the client is NATed. You can change the flag by using something like this modparam("registrar", "nat_flag", 2) The above line tells SER to use flag 2 as the NAT indicator. Regards, Paul On 6/24/05, michael p <mikep3000(a)hotmail.com> wrote: > > it's impossible that it's that > > first all my client are behind a NAT and some flags are 1 and others 0 > > second i 'm already used a stun server. > > > do you think that's something else? > > do you know why after a specific time some SIP clients are deconnected > alone? > is there a variable that i can change for that? > > thanks > > M. > > >From: Java Rockx <javarockx(a)gmail.com> > >Reply-To: Java Rockx <javarockx(a)gmail.com> > >To: michael p <mikep3000(a)hotmail.com> > >CC: serusers(a)lists.iptel.org > >Subject: Re: [serusers] calling problem > >Date: Thu, 23 Jun 2005 09:02:29 -0400 > > > >Flags = 0 indicates that the SIP client is not behind a NAT device. > > > >Flags = 1 indicates that the SIP client is behind a NAT device. > > > >In order to get audio (ie, RTP streams) working one one or more SIP > clients > >is behind a NAT device you must use either rtpproxy or mediaproxy. > > > >The Getting Started document at http://onsip.org/ fully describes how to > >use > >both. > > > >A third, but IMHO, less appealing option would be to use a STUN server, > >such > >as mystun which is availble at sourceforge.net <http://sourceforge.net> < > http://sourceforge.net>. > > > >Regards, > >Paul > > > >On 6/23/05, michael p <mikep3000(a)hotmail.com> wrote: > > > > > > hi > > > > > > in my table location of SER i have a field flags, someone knows for > what > > > is? > > > > > > now i have three users registered and two can call and their flags are > >"0" > > > and one can't call and is flags field is "1" > > > > > > is that the reason why it don't work? > > > > > > thanks > > > > > > M. > > > > > > > > > _______________________________________________ > > > Serusers mailing list > > > serusers(a)lists.iptel.org > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > >

1 0

[Users] Documentation for CVS head
by Bogdan-Andrei Iancu 24 Jun '05

24 Jun '05

Hi folks, the documentation repository http://www.openser.org/docs/ made place for the documentation for development branch (CVS head). For each documentation area you find links to each openser version. The docs for CVS head are up-to-date (includes all changes from the CVS). Still documentation for LCR and MEDIAPROXY modules must be uploaded - the modules have only text docs, no sgml files :(. regards, bogdan

1 0

Re: [Serusers] 400 ul_add: flags expected
by Samuel Osorio Calvo 24 Jun '05

24 Jun '05

There is one field missing after replicate, called flag. It is a boolean parameter that I think it determines wether the UA is natted or not. So, the right command is: bash#serctl fifo ul_add {table} {AoR} {Contact} {Expires} {q} {replicate} {flags} so, as an example: serctl fifo ul_add location bob(a)biloxy.com bob(a)192.168.2.45 3600 1 0 0 Other useful commands are: bash#serctl fifo ul_rm {table} {AoR}--> which will remove all contacts for the given AoR and bash#serctl fifo ul_rm_contact {table} {AoR} {Contact} --> remove only the given contact for the AoR Hope it helps, Samuel Unclassified. >>> "Aisling" <ashling.odriscoll(a)cit.ie> 06/23/05 05:12PM >>> Hello, Has anyone experienced the above message when adding a permanent user location via the serctl fifo interface? I did a search and another post suggested it was due to a mismatch in the database due to customizing SER or because of serweb but because I haven't done anything like that, does anyone have another explanation? Also would anyone know the exact format for setting the q value with this command? According to previous correspondence this might be possible ("I think the current usrloc fifo commands provide enough flexibility to modify the q value, I don't know now the exact spelling but something like: serctl ul add location user@domain user@ip {expires} {q} {replicate}") Many Thanks, Aisling. _______________________________________ Aisling O' Driscoll Research Student Centre for Adaptive Wireless Systems Cork Institute of Technology. Email: ashling.odriscoll(a)cit.ie -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

1 0

[Serusers] google summer of code for opensource
by Iqbal 24 Jun '05

24 Jun '05

seeing that we have been having lots of discussions about opensource , and forking etc, maybe we could get some resources from here if anyone is interested http://code.google.com/summerofcode.html Iqbal

2 1

[Serusers] 486 Busy Here
by Ramu Yadav 24 Jun '05

24 Jun '05

Hi all, I am facing a problem. i.e: When I dial a number ex: (from 9005 to 9010). In this case 9010 reject that call then that guy will send "486 Busy Here" or "603 Decline" . (This is the first phone) So far so good. When I reject the 2nd phone will ring immediately but it didn't, it is ringing after 20 seconds. Here my "fr_inv_timer", 20 seconds. Thanks -- Ramu Yadav

1 0

[Serusers] Onsip doc
by harry gaillac 24 Jun '05

24 Jun '05

Hello, May be a good link for ONSIP http://www.tech-invite.com/ Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

2 1

[Serusers] call test generator
by Ernesto Lucero 24 Jun '05

24 Jun '05

Hi Does anybody has or knows of a very good TEST CALL GENERATOR i am trying to make test to the network where i am deploying my sip service Any help will be great Thanks moyotlan

3 2

[Serusers] SER losing SIP registrations
by Juan Carlos Castro y Castro 24 Jun '05

24 Jun '05

Our company has a call center implementation on a client with up to 100 support personnel using X-Lite version 1050 softphones running under Windows 98. Until a few days ago, the softphones were logged in directly on our PBXs. Now, they log onto a separate SER 0.8.14 box and the PBXs forward calls to SER. That was needed to unify queue management. It works. But some softphones are being randomly kicked out of SER, it seems SER isn't receiving the refresh REGISTER messages from the softphones. The re-register timeout is set to 500 seconds on the softphones. There's a lot of "removing spare zombie" and "Binding '<user>','<url>' has expired" messages in /var/log/messages. For now, we're instructing the client to increase the timeout to 10 hours on the softphones in which the problem happens most often. I don't know if that's really the right thing to do, I think we should somehow make sure the re-registers are done in a timely fashion and retried, but I could not find ant SER configuration option related to that. What should I do? Another thing: the softphones take a horribly high amount of time to log in, but that may be because X-Lite sucks. Regards, Juan

7 9

[Serusers] forking stable, unstable....just a thought how it could be done
by Iqbal 23 Jun '05

23 Jun '05

hi without wanting to start up last week discussions i thought this might be useful ------------------------------------- HEAD and STABLE In the early development of Asterisk, there was a single branch. Releases were few and far between, and the generally accepted way to obtain and use Asterisk was via CVS. While this is fine for people developing Asterisk, the others who wished to run the software in a production environment sometimes had to contend with broken branches, and not really sure what they were getting from one day to the next. At Astricon (http://www.astricon.net), the first official STABLE version 1.0 was released. The STABLE branch was born. Now there were two separate branches of Asterisk. There was HEAD which the developers would use to test patches and new features, and there was STABLE, which people could run in production and where only bug fixes and proven features could be ported over. However this caused two independent versions of code. While this is good for stability, reliability and development, it also doubles the development overhead. Every bug fix which is submitted and resolved on the bug tracker (http://bugs.digium.com) must then be ported to the stable branch. And the longer that the development and stable branches co-exist, the more diverse their code bases become. This can cause patches which apply cleanly to HEAD to not apply to STABLE. This is where the Asterisk Development Proxy comes in. Russell Bryant is the maintainer of the STABLE branch, and thus, must port every single resolved bug from HEAD into STABLE. This is an enormous under taking, and the purpose of formation of the Asterisk Development Proxy. Many of these patches can be ported from HEAD to STABLE with basic or average coding skills. It is also an excellent opportunity for developers unfamiliar with the Asterisk code base to get involved and help with something while they are getting their feet wet. So join the Asterisk Development Proxy and Asterisk Documentation Project team and get involved with Asterisk today! Retrieved from "http://dev.asteriskdocs.org/index.php/HEAD_and_STABLE" ------------------------------------------------------------ Iqbal

1 0

[Users] unified pseudo-variables in avpops and xlog
by Daniel-Constantin Mierla 23 Jun '05

23 Jun '05

Hello, the avpops and xlog module are using now same format for specifiers to print formatted strings. avpops has a new function avp_printf(dest, format) which stores in the AVP 'dest' the string resulted after substitution of the specifiers from "format" parameter. The old marker for specifiers '%' has been replaced by '$' which is more common for variables. To specify a header or an AVP you have to use now $hdr(header_name[index]) or $avp(avp_id[index]). The list with all available pseudo-variables in OpenSER is posted at: http://openser.org/docs/pseudo-variables.html avp_printf() allow to compose a value from many parts of a SIP message, other avps and many other values taken from system. For example: avp_printf("i:20", "the time is $Tf, the request is a $rm and has the from header $hdr(from)"); The documentation for avpops and xlog module has been updated on CVS, too. Note that these features are available only on CVS head (development version - http://openser.org/index.php#download). Online documentation site for modules (http://openser.org/docs/) has been split in 0.9x (stable) and 0.10.x (devel) directories. Follows the integration of same format for pseudo-variables in textops, acc and tm. We will appreciate any feedback. Daniel

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[Serusers] AVP on subst_uri()
by Miguel Angel Villar 23 Jun '05

23 Jun '05

Hi There. I'm using subst_uri() to make some URI transformation and is working great: subst_uri('/^sip:([0-9]+)@(.*)$/sip:011\1@\2/i'); Now I'm trying to insert an AVP value in the substitution but is not working: avp_push("011","var"); subst_uri('/^sip:([0-9]+)@(.*)$/sip:$var\1@\2/i'); When I do this the replaced URI has the string $var instead of the value of it (011). Can I do something like this in subst_uri()? Looking at the Textops User's guide I can see that there is some kind of parameter association for subst_uri(). <<subst_uri().txt>> Regards. MaV.-

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[Users] two issues: ser.cfg or openser.cfg and mysql password?
by Jose Bertuzzi 23 Jun '05

23 Jun '05

Hello guys, first of all congrats for this new idea of openser. I completly agree with you. I tried on june 14, the debian stable package and found that after installing it the config file created is openser.cfg but the program calls ser.cfg. I've just renamed it and start working. The other issue is that i don't know what is the mysql password to add users using openserctl add. I tried with heslo, but failed. So I changed the passwords on mysql and used the command modparam("acc", "db_url", "mysql://user:password@localhost/openser") but did not work. I also tried with the FreeBSD port and compiling src, with same results. What can I do? Thanks a lot for your help. Pablo. __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

4 5

[Serusers] 400 ul_add: flags expected
by Aisling 23 Jun '05

23 Jun '05

Hello, Has anyone experienced the above message when adding a permanent user location via the serctl fifo interface? I did a search and another post suggested it was due to a mismatch in the database due to customizing SER or because of serweb but because I haven't done anything like that, does anyone have another explanation? Also would anyone know the exact format for setting the q value with this command? According to previous correspondence this might be possible ("I think the current usrloc fifo commands provide enough flexibility to modify the q value, I don't know now the exact spelling but something like: serctl ul add location user@domain user@ip {expires} {q} {replicate}") Many Thanks, Aisling. _______________________________________ Aisling O' Driscoll Research Student Centre for Adaptive Wireless Systems Cork Institute of Technology. Email: ashling.odriscoll(a)cit.ie -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

1 0

[Serusers] problems when going to pstn
by Lucas Aimaretto 23 Jun '05

23 Jun '05

Hi all, I'm writting to ask for suggestion. I'm able to communicate sip-to-sip using SER. But when one user wants to talk with somebody in the pstn, I receive two different errors, depending on the gateway. Look ... This first message came from a Mera sip server. U IP_MERA:5060 -> IP_SER:5060 SIP/2.0 415 Unsupported Media Type. v: SIP/2.0/UDP IP_SER:5060;branch=z9hG4bKad03.0a8597a1.0. v: SIP/2.0/UDP IP_UA:15172;branch=z9hG4bK2413941036. Record-Route: <sip:IP_MERA:5060;maddr=38.117.22.34>. Record-Route: <sip:IP_SER;ftag=2859935538;lr=on>. f: "1991117" <sip:1991117@IP_SER>;tag=2859935538. t: <sip:011543515684778@IP_SER>;tag=2eff4b002937ff10ff0000045a5b1432. i: 6ad7a895fbb61ed09f5b2ea36b6612ce(a)192.168.1.30. CSeq: 1 INVITE. P-Session-Timeout: 7200. Server: MERA MSIP v.1.0.1. l: 0. This one from a cisco AS5350 U IP_CISCO:5060 -> IP_SER:5060 SIP/2.0 400 Bad Request - 'Invalid SDP information'. v: SIP/2.0/UDP IP_SER;branch=z9hG4bK2817.1aa690e7.0. v: SIP/2.0/UDP IP_UA:15188;branch=z9hG4bK3914033240. f: "1991117" <sip:1991117@IP_SER>;tag=2962826014. t: <sip:3055383346@IP_SER>. i: 332ee2e17a14427caeb2feaf2da22a3a(a)192.168.1.30. CSeq: 2 INVITE. P-Session-Timeout: 7200. l: 0. I do not get it. Why could I communicate sip-to-sip just fine, but, when going to pstn, I receive this errors. And, moreover, they are different. I'm only having this error with this particular user, which is behind a router doing NAT. Do you have any ideas of what could I do to solve this ... ? Or what could be the problem ? Best regards, Lucas -- No virus found in this outgoing message. Checked by AVG Anti-Virus. Version: 7.0.323 / Virus Database: 267.7.11/26 - Release Date: 22/06/2005

1 0

[Serusers] rtpproxy + nathelper + radius + mediaproxy + voicemail + asterisk + serweb + ser-0.9.3-fc1.0 and it is running fine. (ito.)
by vm_8000＠voice.dyndns.tv 23 Jun '05

23 Jun '05

rtpproxy + nathelper + radius + mediaproxy + voicemail + asterisk + serweb + ser-0.9.3-fc1.0 and it is running fine. (ito.) -rw-r--r-- 1 root root 835211 6月 23 18:35 ser-0.9.3-fc1.0.i386.rpm -rw-r--r-- 1 root root 1909414 6月 23 18:35 ser-0.9.3-fc1.0.src.rpm -rw-r--r-- 1 root root 1902353 6月 23 18:41 ser-0.9.3_src.tar.gz -rw-r--r-- 1 root root 3470729 6月 23 18:35 ser-debuginfo-0.9.3-fc1.0.i386 .rpm -rw-r--r-- 1 root root 64601 6月 23 18:35 ser-jabber-0.9.3-fc1.0.i386.rpm -rw-r--r-- 1 root root 111690 6月 23 18:35 ser-mysql-0.9.3-fc1.0.i386.rpm -rw-r--r-- 1 root root 21406 6月 23 18:35 ser-radius-0.9.3-fc1.0.i386.rpm ///////////////////////////////////////////// [cycle #: 10; if constant make sure server lives and fifo is on] Server: Sip EXpress router (0.9.3-0 (i386/linux)) Now: Thu Jun 23 22:04:39 2005 Up Since: Thu Jun 23 21:34:44 2005 Up time: 1795 [sec] Transaction Statistics Current: 0 (0 waiting) Total: 0 (0 local) Replied localy: 0 Completion status 6xx: 0, 5xx: 0, 4xx: 0, 3xx: 0,2xx: 0 Stateless Server Statistics 200: 19 202: 0 2xx: 0 300: 0 301: 0 302: 0 3xx: 0 400: 0 401: 19 403: 0 404: 0 407: 0 408: 0 483: 0 4xx: 0 500: 0 5xx: 0 6xx: 0 xxx: 0 failures: 0 UsrLoc Stats Domain Registered Expired 'aliases' 19 0 'location' 10 1 ///////////////////////////////////////////// http://www.voice.dyndns.tv sip:admin at iptel.org Saitama Japan Sip EXpress router Support Team voice at wmail.plala.or.jp /////////////////////////////////////////////

1 0

[serusers] calling problem
by michael p 23 Jun '05

23 Jun '05

hi in my table location of SER i have a field flags, someone knows for what is? now i have three users registered and two can call and their flags are "0" and one can't call and is flags field is "1" is that the reason why it don't work? thanks M.

2 1

[Serusers] Implementing Module with Osip
by Alexandre Passito 23 Jun '05

23 Jun '05

Hi, I'm trying to implement a module to Openser that should use parts of the Osip (sip stack implementation). That module could be a "black list", "follow me" and etc. Did somebody implement it already? Does somebody have an ideia how to begin to implement? Where can I find some information about it? Best regards, -- Alexandre Passito Universidade Federal do Amazonas Departamento de Ciência da Computação LabVoIP - GRSD

1 0

[Serusers] Server header in SER responses
by Fermín Galán Márquez 23 Jun '05

23 Jun '05

Hello, SER includes a "Server" header in responses (for examples, in the 200 OK responding a REGISTER when SER acting as registrar server). Server: Sip Express router (0.8.14 (i386/linux)) Is there any way to modify or/and remove the header? I have looked through the SIP Admin Guide (http://www.iptel.org/ser/doc/seruser/seruser.html) but I haven't found nothing related with this. (I know I could modify the "Server" header editing the SERVER_HDR directive in config.h and recompile, but I think is too radical :) Thanks in advance! Best regards, ------ Fermín Agora Systems, S. A.

2 1

[Serusers] PRACK
by Dana Turcanu 23 Jun '05

23 Jun '05

Hello, I have a problem with tow Cisco ATA 186 both having version v3.2. When I try to make a call on my side I hear that the phone is ringing 3 times and then the call is disconnected. On the other side, the telephone is ringing 10 times. I belive that the problem is when one Cisco sends a PRACK packet and and it receive "420 Bad request" from de sip server. Then the call is disconnected with error "504 Gateway Time-out". Below I put the logs from the tow ATAs. Any help in solving this problem will be appreciated. Thanks. Call from: 3327xxxx to 33271yyyy (log from the calling ATA): ===================================================== 050620|11:48:28 Calling 0332yyyy 050620|11:48:28 SCC Cmd[0:0]CLIP 050620|11:48:28 SCC Cmd[0:0]CALL 050620|11:48:28 [0]INVITE Retry 0 050620|11:48:28 [0:0]Tx Msg to ip_sip_proxy:5060 050620|11:48:28 050620|11:48:28 INVITE sip:0332yyyy@domain.com;user=phone SIP/2.0 050620|11:48:28 Via: SIP/2.0/UDP ip_ata_1:5060;branch=z9hG4bK5ce290af84b14296 050620|11:48:28 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:28 To: <sip:0332yyyy@domain.com;user=phone> 050620|11:48:28 Call-ID: 2133411222@ip_ata_1 050620|11:48:28 CSeq: 1 INVITE 050620|11:48:28 Contact: <sip:332xxxx@ip_ata_1:5060;user=phone;transport=udp> 050620|11:48:28 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:28 Expires: 300 050620|11:48:28 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:28 Supported: 100rel,replaces 050620|11:48:28 Content-Length: 236 050620|11:48:28 Content-Type: application/sdp 050620|11:48:28 050620|11:48:28 v=0 050620|11:48:28 o=332xxxx 57793347 57793347 IN IP4 ip_ata_1 050620|11:48:28 s=ATA186 Call 050620|11:48:28 c=IN IP4 ip_ata_1 050620|11:48:28 t=0 0 050620|11:48:28 m=audio 16384 RTP/AVP 8 0 101 050620|11:48:28 a=rtpmap:8 PCMA/8000/1 050620|11:48:28 a=rtpmap:0 PCMU/8000/1 050620|11:48:28 a=rtpmap:101 telephone-event/8000 050620|11:48:28 a=fmtp:101 0-15 050620|11:48:28 050620|11:48:28 [0]Rx Msg from ip_sip_server:5060 050620|11:48:28 050620|11:48:28 SIP/2.0 407 Proxy Authentication Required 050620|11:48:28 Via: SIP/2.0/UDP ip_ata_1:5060;branch=z9hG4bK5ce290af84b14296;rport=5060;received=ip_ata_1_nat 050620|11:48:28 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:28 To: <sip:0332yyyy@domain.com;user=phone>;tag=fee886471600e72424ce85e67d30b332.6542 050620|11:48:28 Call-ID: 2133411222@ip_ata_1 050620|11:48:28 CSeq: 1 INVITE 050620|11:48:28 Proxy-Authenticate: Digest realm="domain.com", nonce="rrrr" 050620|11:48:28 Server: Sip EXpress router (0.8.14-5 (i386/linux)) 050620|11:48:28 Content-Length: 0 050620|11:48:28 050620|11:48:28 050620|11:48:28 Response=407; Proxy Authentication Required 050620|11:48:28 [0]Tx INVITE to ip_sip_server:5060 050620|11:48:28 050620|11:48:28 ACK sip:0332yyyy@domain.com;user=phone SIP/2.0 050620|11:48:28 Via: SIP/2.0/UDP ip_ata_1:5060;branch=z9hG4bK5ce290af84b14296 050620|11:48:28 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:28 To: <sip:0332yyyy@domain.com;user=phone>;tag=fee886471600e72424ce85e67d30b332.6542 050620|11:48:28 Call-ID: 2133411222@ip_ata_1 050620|11:48:28 CSeq: 1 ACK 050620|11:48:28 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:28 Content-Length: 0 050620|11:48:28 050620|11:48:28 050620|11:48:28 [0]INVITE Retry 0 050620|11:48:28 [0:0]Tx Msg to ip_sip_server:5060 050620|11:48:28 050620|11:48:28 INVITE sip:0332yyyy@domain.com;user=phone SIP/2.0 050620|11:48:28 Via: SIP/2.0/UDP ip_ata_1:5060;branch=z9hG4bKf33e54a2a402e527 050620|11:48:28 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:28 To: <sip:0332yyyy@domain.com;user=phone> 050620|11:48:28 Call-ID: 2133411222@ip_ata_1 050620|11:48:28 CSeq: 2 INVITE 050620|11:48:28 Contact: <sip:332xxxx@ip_ata_1:5060;user=phone;transport=udp> 050620|11:48:28 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:28 Proxy-Authorization: Digest username="332xxxx",realm="domain.com",nonce="rrrr",uri="sip:03 32yyyy(a)domain.com",response="ihuhuh" 050620|11:48:28 Expires: 300 050620|11:48:28 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:28 Supported: 100rel,replaces 050620|11:48:28 Content-Length: 236 050620|11:48:28 Content-Type: application/sdp 050620|11:48:28 050620|11:48:28 v=0 050620|11:48:28 o=332xxxx 57793355 57793355 IN IP4 ip_ata_1 050620|11:48:28 s=ATA186 Call 050620|11:48:28 c=IN IP4 ip_ata_1 050620|11:48:28 t=0 0 050620|11:48:28 m=audio 16384 RTP/AVP 8 0 101 050620|11:48:28 a=rtpmap:8 PCMA/8000/1 050620|11:48:28 a=rtpmap:0 PCMU/8000/1 050620|11:48:28 a=rtpmap:101 telephone-event/8000 050620|11:48:28 a=fmtp:101 0-15 050620|11:48:28 050620|11:48:28 [0]Rx Msg from ip_sip_server:5060 050620|11:48:28 050620|11:48:28 SIP/2.0 100 Trying 050620|11:48:28 Via: SIP/2.0/UDP ip_ata_1:5060;branch=z9hG4bKf33e54a2a402e527;rport=5060;received=ip_ata_1_nat 050620|11:48:28 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:28 To: <sip:0332yyyy@domain.com;user=phone> 050620|11:48:28 Call-ID: 2133411222@ip_ata_1 050620|11:48:28 CSeq: 2 INVITE 050620|11:48:28 Server: Sip EXpress router (0.8.14-5 (i386/linux)) 050620|11:48:28 Content-Length: 0 050620|11:48:28 050620|11:48:28 050620|11:48:28 Response=100; Trying 050620|11:48:29 [0]Rx Msg from ip_sip_server:5060 050620|11:48:29 050620|11:48:29 SIP/2.0 100 trying -- your call is important to us 050620|11:48:29 Via: SIP/2.0/UDP ip_ata_1:5060;branch=z9hG4bKf33e54a2a402e527;rport=5060;received=ip_ata_1_nat 050620|11:48:29 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:29 To: <sip:0332yyyy@domain.com;user=phone> 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 050620|11:48:29 CSeq: 2 INVITE 050620|11:48:29 Server: Sip EXpress router (0.8.14-5 (i386/linux)) 050620|11:48:29 Content-Length: 0 050620|11:48:29 050620|11:48:29 050620|11:48:29 Response=100; Trying 050620|11:48:29 [0]Rx Msg from ip_sip_server:5060 050620|11:48:29 050620|11:48:29 SIP/2.0 180 Ringing 050620|11:48:29 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:29 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:29 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 050620|11:48:29 CSeq: 2 INVITE 050620|11:48:29 Require: 100rel 050620|11:48:29 RSeq: 1 050620|11:48:29 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:29 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:29 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:29 Content-Length: 0 050620|11:48:29 050620|11:48:29 050620|11:48:29 Response=180; Ringing 050620|11:48:29 SCC:ev=10[0:0] 3 0 050620|11:48:29 [0]StartTone 3 050620|11:48:29 [0]PRACK Retry 0 050620|11:48:29 [0:0]Tx Msg to ip_sip_server:5060 050620|11:48:29 050620|11:48:29 PRACK sip:0332yyyy@domain.com;user=phone SIP/2.0 050620|11:48:29 Via: SIP/2.0/UDP ip_ata_1:5060;branch=z9hG4bK2b0f7845ea7a8edb 050620|11:48:29 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:29 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 050620|11:48:29 CSeq: 3 PRACK 050620|11:48:29 RAck: 1 2 INVITE 050620|11:48:29 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:29 Proxy-Authorization: Digest username="332xxxx",realm="domain.com",nonce="rrrr",uri="sip:03 32yyyy(a)domain.com",response="tttt" 050620|11:48:29 Content-Length: 0 050620|11:48:29 050620|11:48:29 050620|11:48:29 [0]Rx Msg from ip_sip_server:5060 050620|11:48:29 050620|11:48:29 SIP/2.0 400 Bad request 050620|11:48:29 Via: SIP/2.0/UDP ip_ata_1:5060;branch=z9hG4bK2b0f7845ea7a8edb;rport=5060;received=ip_ata_1_nat 050620|11:48:29 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:29 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 050620|11:48:29 CSeq: 3 PRACK 050620|11:48:29 Server: Sip EXpress router (0.8.14-5 (i386/linux)) 050620|11:48:29 Content-Length: 0 050620|11:48:29 050620|11:48:29 050620|11:48:29 [0]Rx Msg from ip_sip_server:5060 050620|11:48:29 050620|11:48:29 SIP/2.0 180 Ringing 050620|11:48:29 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:29 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:29 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 050620|11:48:29 CSeq: 2 INVITE 050620|11:48:29 Require: 100rel 050620|11:48:29 RSeq: 1 050620|11:48:29 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:29 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:29 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:29 Content-Length: 0 050620|11:48:29 050620|11:48:29 050620|11:48:29 Response=180; Ringing 050620|11:48:30 [0]Rx Msg from ip_sip_server:5060 050620|11:48:30 050620|11:48:30 SIP/2.0 180 Ringing 050620|11:48:30 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:30 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:30 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:30 Call-ID: 2133411222@ip_ata_1 050620|11:48:30 CSeq: 2 INVITE 050620|11:48:30 Require: 100rel 050620|11:48:30 RSeq: 1 050620|11:48:30 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:30 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:30 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:30 Content-Length: 0 050620|11:48:30 050620|11:48:30 050620|11:48:30 Response=180; Ringing 050620|11:48:32 [0]Rx Msg from ip_sip_server:5060 050620|11:48:32 050620|11:48:32 SIP/2.0 180 Ringing 050620|11:48:32 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:32 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:32 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:32 Call-ID: 2133411222@ip_ata_1 050620|11:48:32 CSeq: 2 INVITE 050620|11:48:32 Require: 100rel 050620|11:48:32 RSeq: 1 050620|11:48:32 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:32 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:32 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:32 Content-Length: 0 050620|11:48:32 050620|11:48:32 050620|11:48:33 Response=180; Ringing 050620|11:48:35 [0]Rx Msg from ip_sip_server:5060 050620|11:48:35 050620|11:48:35 SIP/2.0 180 Ringing 050620|11:48:35 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:35 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:35 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:35 Call-ID: 2133411222@ip_ata_1 050620|11:48:35 CSeq: 2 INVITE 050620|11:48:35 Require: 100rel 050620|11:48:35 RSeq: 1 050620|11:48:35 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:35 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:35 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:35 Content-Length: 0 050620|11:48:35 050620|11:48:35 050620|11:48:35 Response=180; Ringing 050620|11:48:37 [0]Rx Msg from ip_sip_server:5060 050620|11:48:37 050620|11:48:37 SIP/2.0 180 Ringing 050620|11:48:37 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:37 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:37 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:37 Call-ID: 2133411222@ip_ata_1 050620|11:48:37 CSeq: 2 INVITE 050620|11:48:37 Require: 100rel 050620|11:48:37 RSeq: 1 050620|11:48:37 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:37 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:37 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:37 Content-Length: 0 050620|11:48:37 050620|11:48:37 050620|11:48:37 Response=180; Ringing 050620|11:48:39 [0]Rx Msg from ip_sip_server:5060 050620|11:48:39 050620|11:48:39 SIP/2.0 180 Ringing 050620|11:48:39 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:39 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:39 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:39 Call-ID: 2133411222@ip_ata_1 050620|11:48:39 CSeq: 2 INVITE 050620|11:48:39 Require: 100rel 050620|11:48:39 RSeq: 1 050620|11:48:39 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:39 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:39 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:39 Content-Length: 0 050620|11:48:39 050620|11:48:39 050620|11:48:39 Response=180; Ringing 050620|11:48:41 [0]Rx Msg from ip_sip_server:5060 050620|11:48:41 050620|11:48:41 SIP/2.0 180 Ringing 050620|11:48:41 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:41 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:41 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:41 Call-ID: 2133411222@ip_ata_1 050620|11:48:41 CSeq: 2 INVITE 050620|11:48:41 Require: 100rel 050620|11:48:41 RSeq: 1 050620|11:48:41 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:41 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:41 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:41 Content-Length: 0 050620|11:48:41 050620|11:48:41 050620|11:48:41 Response=180; Ringing 050620|11:48:43 [0]Rx Msg from ip_sip_server:5060 050620|11:48:43 050620|11:48:43 SIP/2.0 504 Gateway Time-out 050620|11:48:43 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:43 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:43 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:43 Call-ID: 2133411222@ip_ata_1 050620|11:48:43 CSeq: 2 INVITE 050620|11:48:43 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:43 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:43 Content-Length: 0 050620|11:48:43 050620|11:48:43 050620|11:48:43 Response=504; Gateway Time-out 050620|11:48:43 [0:0]Tx Msg to ip_sip_server:5060 050620|11:48:43 050620|11:48:43 ACK sip:0332yyyy@domain.com;user=phone SIP/2.0 050620|11:48:43 Via: SIP/2.0/UDP ip_ata_1:5060;branch=z9hG4bK3504d15d5210ae37 050620|11:48:43 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:43 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:43 Call-ID: 2133411222@ip_ata_1 050620|11:48:43 CSeq: 2 ACK 050620|11:48:43 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:43 Proxy-Authorization: Digest username="332xxxx",realm="domain.com",nonce="tttt",uri="sip:03 32yyyy(a)domain.com",response="rrrr" 050620|11:48:43 Content-Length: 0 050620|11:48:43 050620|11:48:43 050620|11:48:43 change next hop[0]: c = 4 050620|11:48:43 All PriObProxy down 050620|11:48:43 No AltObProxy 050620|11:48:43 INVITE failed 050620|11:48:43 SCC:ev=7[0:0] 3 0 050620|11:48:43 [0]StartTone 2 050620|11:48:43 [0]Rx Msg from ip_sip_server:5060 050620|11:48:43 050620|11:48:43 SIP/2.0 504 Gateway Time-out 050620|11:48:43 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:43 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:43 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:43 Call-ID: 2133411222@ip_ata_1 050620|11:48:43 CSeq: 2 INVITE 050620|11:48:43 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:43 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:43 Content-Length: 0 050620|11:48:43 050620|11:48:43 050620|11:48:45 [0]Rx Msg from ip_sip_server:5060 050620|11:48:45 050620|11:48:45 SIP/2.0 504 Gateway Time-out 050620|11:48:45 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:45 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:45 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:45 Call-ID: 2133411222@ip_ata_1 050620|11:48:45 CSeq: 2 INVITE 050620|11:48:45 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:45 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:45 Content-Length: 0 050620|11:48:45 050620|11:48:45 050620|11:48:49 [0]Rx Msg from ip_sip_server:5060 050620|11:48:49 050620|11:48:49 SIP/2.0 504 Gateway Time-out 050620|11:48:49 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:49 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:49 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:49 Call-ID: 2133411222@ip_ata_1 050620|11:48:49 CSeq: 2 INVITE 050620|11:48:49 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:49 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:49 Content-Length: 0 050620|11:48:49 050620|11:48:49 050620|11:48:53 [0]Rx Msg from ip_sip_server:5060 050620|11:48:53 050620|11:48:53 SIP/2.0 504 Gateway Time-out 050620|11:48:53 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:53 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:53 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:53 Call-ID: 2133411222@ip_ata_1 050620|11:48:53 CSeq: 2 INVITE 050620|11:48:53 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:53 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:53 Content-Length: 0 050620|11:48:53 050620|11:48:53 050620|11:48:54 [0]StopTone 050620|11:48:54 SCC Cmd[0:0]RELEASE 050620|11:48:54 CCBS expired 050620|11:48:54 >>> SIP Soft Reset [0:0] <<< 050620|11:48:54 [0]MPT mode 0 ======================================================= Call from 3327xxxx to 3327yyyy ( log from the called ATA): ======================================================= 050620|11:48:29 INVITE sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp SIP/2.0 050620|11:48:29 Max-Forwards: 10 050620|11:48:29 Record-Route: <sip:0332yyyy@ip_sip_server;ftag=2064987569;lr=on> 050620|11:48:29 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:29 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:29 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:29 To: <sip:0332yyyy@domain.com;user=phone> 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 050620|11:48:29 CSeq: 2 INVITE 050620|11:48:29 Contact: <sip:332xxxx@ip_ata_1_nat:5060;user=phone;transport=udp> 050620|11:48:29 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:29 Expires: 300 050620|11:48:29 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:29 Supported: 100rel,replaces 050620|11:48:29 Content-Length: 235 050620|11:48:29 Content-Type: application/sdp 050620|11:48:29 050620|11:48:29 v=0 050620|11:48:29 o=332xxxx 57793355 57793355 IN IP4 ip_ata_1 050620|11:48:29 s=ATA186 Call 050620|11:48:29 c=IN IP4 81.180.201.50 050620|11:48:29 t=0 0 050620|11:48:29 m=audio 36540 RTP/AVP 8 0 101 050620|11:48:29 a=rtpmap:8 PCMA/8000/1 050620|11:48:29 a=rtpmap:0 PCMU/8000/1 050620|11:48:29 a=rtpmap:101 telephone-event/8000 050620|11:48:29 a=fmtp:101 0-15 050620|11:48:29 050620|11:48:29 AVT: payload=101 050620|11:48:29 [1:0]Tx Msg to ip_sip_server:5060 050620|11:48:29 050620|11:48:29 SIP/2.0 100 Trying 050620|11:48:29 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:29 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:29 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:29 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 050620|11:48:29 CSeq: 2 INVITE 050620|11:48:29 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:29 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:29 Content-Length: 0 050620|11:48:29 050620|11:48:29 050620|11:48:29 SCC:ev=21[1:0] 0 0 050620|11:48:29 SCC:ev=5[1:0] 13 0 050620|11:48:29 [1:0]RINGING 050620|11:48:29 SCC Cmd[1:0]ALERT 050620|11:48:29 [1]R2INV Retry 0 050620|11:48:29 [1:0]Re-Tx Msg to ip_sip_server:5060 050620|11:48:29 050620|11:48:29 SIP/2.0 180 Ringing 050620|11:48:29 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:29 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:29 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:29 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 050620|11:48:29 CSeq: 2 INVITE 050620|11:48:29 Require: 100rel 050620|11:48:29 RSeq: 1 050620|11:48:29 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:29 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:29 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:29 Content-Length: 0 050620|11:48:29 050620|11:48:29 050620|11:48:29 [1]R2INV Retry 1 050620|11:48:29 [1:0]Re-Tx Msg to ip_sip_server:5060 050620|11:48:29 050620|11:48:29 SIP/2.0 180 Ringing 050620|11:48:29 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:29 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:29 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:29 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:29 Call-ID: 2133411222@ip_ata_1 050620|11:48:29 CSeq: 2 INVITE 050620|11:48:29 Require: 100rel 050620|11:48:29 RSeq: 1 050620|11:48:29 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:29 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:29 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:29 Content-Length: 0 050620|11:48:29 050620|11:48:29 050620|11:48:30 [1]R2INV Retry 2 050620|11:48:30 [1:0]Re-Tx Msg to ip_sip_server:5060 050620|11:48:30 050620|11:48:30 SIP/2.0 180 Ringing 050620|11:48:30 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:30 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:30 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:30 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:30 Call-ID: 2133411222@ip_ata_1 050620|11:48:30 CSeq: 2 INVITE 050620|11:48:30 Require: 100rel 050620|11:48:30 RSeq: 1 050620|11:48:30 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:30 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:30 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:30 Content-Length: 0 050620|11:48:30 050620|11:48:30 050620|11:48:32 [1]R2INV Retry 3 050620|11:48:32 [1:0]Re-Tx Msg to ip_sip_server:5060 050620|11:48:32 050620|11:48:32 SIP/2.0 180 Ringing 050620|11:48:32 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:32 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:32 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:32 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:32 Call-ID: 2133411222@ip_ata_1 050620|11:48:32 CSeq: 2 INVITE 050620|11:48:32 Require: 100rel 050620|11:48:32 RSeq: 1 050620|11:48:32 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:32 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:32 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:32 Content-Length: 0 050620|11:48:32 050620|11:48:32 050620|11:48:34 [1]R2INV Retry 4 050620|11:48:34 [1:0]Re-Tx Msg to ip_sip_server:5060 050620|11:48:34 050620|11:48:34 SIP/2.0 180 Ringing 050620|11:48:34 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:34 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:34 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:34 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:34 Call-ID: 2133411222@ip_ata_1 050620|11:48:34 CSeq: 2 INVITE 050620|11:48:34 Require: 100rel 050620|11:48:34 RSeq: 1 050620|11:48:34 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:34 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:34 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:34 Content-Length: 0 050620|11:48:34 050620|11:48:34 050620|11:48:36 [1]R2INV Retry 5 050620|11:48:36 [1:0]Re-Tx Msg to ip_sip_server:5060 050620|11:48:36 050620|11:48:36 SIP/2.0 180 Ringing 050620|11:48:36 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:36 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:36 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:36 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:36 Call-ID: 2133411222@ip_ata_1 050620|11:48:36 CSeq: 2 INVITE 050620|11:48:36 Require: 100rel 050620|11:48:36 RSeq: 1 050620|11:48:36 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:36 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:36 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:36 Content-Length: 0 050620|11:48:36 050620|11:48:36 050620|11:48:38 [1]R2INV Retry 6 050620|11:48:38 [1:0]Re-Tx Msg to ip_sip_server:5060 050620|11:48:38 050620|11:48:38 SIP/2.0 180 Ringing 050620|11:48:38 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:38 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:39 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:39 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:39 Call-ID: 2133411222@ip_ata_1 050620|11:48:39 CSeq: 2 INVITE 050620|11:48:39 Require: 100rel 050620|11:48:39 RSeq: 1 050620|11:48:39 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:39 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:39 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:39 Content-Length: 0 050620|11:48:39 050620|11:48:39 050620|11:48:41 [1]R2INV Retry 7 050620|11:48:41 [1:0]Re-Tx Msg to ip_sip_server:5060 050620|11:48:41 050620|11:48:41 SIP/2.0 180 Ringing 050620|11:48:41 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:41 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:41 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:41 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:41 Call-ID: 2133411222@ip_ata_1 050620|11:48:41 CSeq: 2 INVITE 050620|11:48:41 Require: 100rel 050620|11:48:41 RSeq: 1 050620|11:48:41 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp> 050620|11:48:41 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:41 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:41 Content-Length: 0 050620|11:48:41 050620|11:48:41 050620|11:48:43 R2INV TO; NoAltProxy 050620|11:48:43 [1:0]Provisional Response Timeout 050620|11:48:43 [1:0]Tx Msg to ip_sip_server:5060 050620|11:48:43 050620|11:48:43 SIP/2.0 504 Gateway Time-out 050620|11:48:43 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:43 Via: SIP/2.0/UDP ip_ata_1:5060;rport=5060;received=ip_ata_1_nat;branch=z9hG4bKf33e54a2a402e527 050620|11:48:43 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:43 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:43 Call-ID: 2133411222@ip_ata_1 050620|11:48:43 CSeq: 2 INVITE 050620|11:48:43 Server: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:43 Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER, PRACK, UPDATE 050620|11:48:43 Content-Length: 0 050620|11:48:43 050620|11:48:43 050620|11:48:43 reiniting next hop 050620|11:48:43 Refresh DNS for sip.domain.com aaaaaaaa 0 0 0 0 050620|11:48:43 Trying 1st IP ADDR aaaaaaaa 050620|11:48:43 [1]Rx Msg from ip_sip_server:5060 050620|11:48:43 050620|11:48:43 ACK sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp SIP/2.0 050620|11:48:43 Via: SIP/2.0/UDP ip_sip_server;branch=z9hG4bK03f9.fd903bc.0 050620|11:48:43 From: <sip:332xxxx@domain.com;user=phone>;tag=2064987569 050620|11:48:43 Call-ID: 2133411222@ip_ata_1 050620|11:48:43 To: <sip:0332yyyy@domain.com;user=phone>;tag=811613285 050620|11:48:43 CSeq: 2 ACK 050620|11:48:43 User-Agent: Sip EXpress router 050620|11:48:43 Content-Length: 0 050620|11:48:43 050620|11:48:43 050620|11:48:43 RING: got ACK 050620|11:48:43 9852:00;2,2,2,0, 050620|11:48:44 Recv[0]: 60 2906 464960 0 94 0 0 0 050620|11:48:44 Tx[0]: 60 2999 479840 050620|11:48:50 [1]REGISTER Retry 0 050620|11:48:50 [1:0]Tx Msg to ip_sip_server:5060 050620|11:48:50 050620|11:48:50 REGISTER sip:domain.com SIP/2.0 050620|11:48:50 Via: SIP/2.0/UDP ip_ata_2:5060;branch=z9hG4bKb417bd8a1c3112b7 050620|11:48:50 From: <sip:332yyyy@domain.com;user=phone>;tag=2612509671 050620|11:48:50 To: <sip:332yyyy@domain.com;user=phone> 050620|11:48:50 Call-ID: 2749858814@ip_ata_2 050620|11:48:50 CSeq: 355 REGISTER 050620|11:48:50 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp>;expires=3600 050620|11:48:50 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:50 Authorization: Digest username="332yyyy",realm="domain.com",nonce="42b678c1c05ddd4c6ff925bea0c9bb923c5ec1ee",uri="sip:domain.c om",response="asdasdd" 050620|11:48:50 Content-Length: 0 050620|11:48:50 050620|11:48:50 050620|11:48:50 [1]Rx Msg from ip_sip_server:5060 050620|11:48:50 050620|11:48:50 SIP/2.0 401 Unauthorized 050620|11:48:50 Via: SIP/2.0/UDP ip_ata_2:5060;branch=z9hG4bKb417bd8a1c3112b7 050620|11:48:50 From: <sip:332yyyy@domain.com;user=phone>;tag=2612509671 050620|11:48:50 To: <sip:332yyyy@domain.com;user=phone>;tag=fee886471600e72424ce85e67d30b332.919f 050620|11:48:50 Call-ID: 2749858814@ip_ata_2 050620|11:48:50 CSeq: 355 REGISTER 050620|11:48:50 WWW-Authenticate: Digest realm="domain.com", nonce="rrrr", stale=true 050620|11:48:50 Server: Sip EXpress router 050620|11:48:50 Content-Length: 0 050620|11:48:50 050620|11:48:50 050620|11:48:50 [1]Reg Resp 401; Unauthorized 050620|11:48:50 [1]REGISTER Retry 0 050620|11:48:50 [1:0]Tx Msg to ip_sip_server:5060 050620|11:48:50 050620|11:48:50 REGISTER sip:domain.com SIP/2.0 050620|11:48:50 Via: SIP/2.0/UDP ip_ata_2:5060;branch=z9hG4bK99467a1aab648880 050620|11:48:50 From: <sip:332yyyy@domain.com;user=phone>;tag=2612509671 050620|11:48:50 To: <sip:332yyyy@domain.com;user=phone> 050620|11:48:50 Call-ID: 2749858814@ip_ata_2 050620|11:48:50 CSeq: 356 REGISTER 050620|11:48:50 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp>;expires=3600 050620|11:48:50 User-Agent: Cisco ATA 186 v3.2.0 atasip (041111A) 050620|11:48:50 Authorization: Digest username="332yyyy",realm="domain.com",nonce="rrrr",uri="sip:domain.c om",response="sdfsdfsdffd" 050620|11:48:50 Content-Length: 0 050620|11:48:50 050620|11:48:50 050620|11:48:50 [1]Rx Msg from ip_sip_server:5060 050620|11:48:50 050620|11:48:50 SIP/2.0 200 OK 050620|11:48:50 Via: SIP/2.0/UDP ip_ata_2:5060;branch=z9hG4bK99467a1aab648880 050620|11:48:50 From: <sip:332yyyy@domain.com;user=phone>;tag=2612509671 050620|11:48:50 To: <sip:332yyyy@domain.com;user=phone>;tag=fee886471600e72424ce85e67d30b332.6927 050620|11:48:50 Call-ID: 2749858814@ip_ata_2 050620|11:48:50 CSeq: 356 REGISTER 050620|11:48:50 Contact: <sip:332yyyy@ip_ata_2:5060;user=phone;transport=udp>;q=0.00;expires=3600 050620|11:48:50 Server: Sip EXpress router 050620|11:48:50 Content-Length: 0 050620|11:48:50 050620|11:48:50 050620|11:48:50 [1]Reg Resp 200; OK 050620|11:48:50 [1]Reg OK (3600) =========================================================== __________________________________ Yahoo! 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[Serusers] "ser-0.9.0.GettingStarted.1.1" Setup Problem !!
by Freeman 23 Jun '05

23 Jun '05

Hi, I have some problem when i tried to setup "ser-0.9.0.GettingStarted.1.1.tar.gz" in Redhat 9, i get some error detail as below : how can i setup in RedHat 9 ? ========== make_and_install_output.txt ========= Makefile.rules:81: action.d: file or folder not found. Makefile.rules:81: crc.d: file or folder not found. Makefile.rules:81: daemonize.d: file or folder not found. Makefile.rules:81: data_lump.d: file or folder not found. Makefile.rules:81: data_lump_rpl.d: file or folder not found. Makefile.rules:81: dprint.d: file or folder not found. Makefile.rules:81: dset.d: file or folder not found. Makefile.rules:81: error.d: file or folder not found. Makefile.rules:81: fifo_server.d: file or folder not found. Makefile.rules:81: flags.d: file or folder not found. Makefile.rules:81: forward.d: file or folder not found. Makefile.rules:81: hash_func.d: file or folder not found. Makefile.rules:81: ip_addr.d: file or folder not found. Makefile.rules:81: main.d: file or folder not found. Makefile.rules:81: md5.d: file or folder not found. Makefile.rules:81: md5utils.d: file or folder not found. Makefile.rules:81: modparam.d: file or folder not found. Makefile.rules:81: msg_translator.d: file or folder not found. Makefile.rules:81: pass_fd.d: file or folder not found. Makefile.rules:81: proxy.d: file or folder not found. Makefile.rules:81: qvalue.d: file or folder not found. Makefile.rules:81: re.d: file or folder not found. Makefile.rules:81: receive.d: file or folder not found. Makefile.rules:81: resolve.d: file or folder not found. Makefile.rules:81: route.d: file or folder not found. Makefile.rules:81: route_struct.d: file or folder not found. Makefile.rules:81: script_cb.d: file or folder not found. Makefile.rules:81: socket_info.d: file or folder not found. Makefile.rules:81: sr_module.d: file or folder not found. Makefile.rules:81: stats.d: file or folder not found. Makefile.rules:81: tcp_main.d: file or folder not found. Makefile.rules:81: tcp_read.d: file or folder not found. Makefile.rules:81: timer.d: file or folder not found. Makefile.rules:81: tsend.d: file or folder not found. Makefile.rules:81: udp_server.d: file or folder not found. Makefile.rules:81: unixsock_server.d: file or folder not found. Makefile.rules:81: usr_avp.d: file or folder not found. Makefile.rules:81: mem/f_malloc.d: file or folder not found. Makefile.rules:81: mem/mem.d: file or folder not found. Makefile.rules:81: mem/memtest.d: file or folder not found. Makefile.rules:81: mem/q_malloc.d: file or folder not found. Makefile.rules:81: mem/shm_mem.d: file or folder not found. Makefile.rules:81: mem/vq_malloc.d: file or folder not found. Makefile.rules:81: parser/hf.d: file or folder not found. Makefile.rules:81: parser/msg_parser.d: file or folder not found. Makefile.rules:81: parser/parse_allow.d: file or folder not found. Makefile.rules:81: parser/parse_content.d: file or folder not found. Makefile.rules:81: parser/parse_cseq.d: file or folder not found. Makefile.rules:81: parser/parse_disposition.d: file or folder not found. Makefile.rules:81: parser/parse_diversion.d: file or folder not found. Makefile.rules:81: parser/parse_event.d: file or folder not found. Makefile.rules:81: parser/parse_expires.d: file or folder not found. Makefile.rules:81: parser/parse_fline.d: file or folder not found. Makefile.rules:81: parser/parse_from.d: file or folder not found. Makefile.rules:81: parser/parse_hname2.d: file or folder not found. Makefile.rules:81: parser/parse_hostport.d: file or folder not found. Makefile.rules:81: parser/parse_methods.d: file or folder not found. Makefile.rules:81: parser/parse_nameaddr.d: file or folder not found. Makefile.rules:81: parser/parse_param.d: file or folder not found. Makefile.rules:81: parser/parser_f.d: file or folder not found. Makefile.rules:81: parser/parse_rpid.d: file or folder not found. Makefile.rules:81: parser/parse_rr.d: file or folder not found. Makefile.rules:81: parser/parse_sipifmatch.d: file or folder not found. Makefile.rules:81: parser/parse_to.d: file or folder not found. Makefile.rules:81: parser/parse_uri.d: file or folder not found. Makefile.rules:81: parser/parse_via.d: file or folder not found. Makefile.rules:81: parser/digest/digest.d: file or folder not found. Makefile.rules:81: parser/digest/digest_parser.d: file or folder not found. Makefile.rules:81: parser/digest/param_parser.d: file or folder not found. Makefile.rules:81: parser/contact/contact.d: file or folder not found. Makefile.rules:81: parser/contact/parse_contact.d: file or folder not found. Makefile.rules:81: db/db.d: file or folder not found. Makefile.rules:81: db/db_fifo.d: file or folder not found. Makefile.rules:81: lex.yy.d: file or folder not found. Makefile.rules:81: cfg.tab.d: file or folder not found. bison -d -b cfg cfg.y cfg.y contains 1 shift/reduce conflict. flex cfg.lex gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c action.c -o action.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c crc.c -o crc.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c daemonize.c -o daemonize.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c data_lump.c -o data_lump.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c data_lump_rpl.c -o data_lump_rpl.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c dprint.c -o dprint.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c dset.c -o dset.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c error.c -o error.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c fifo_server.c -o fifo_server.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c flags.c -o flags.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c forward.c -o forward.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c hash_func.c -o hash_func.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c ip_addr.c -o ip_addr.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c main.c -o main.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c md5.c -o md5.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c md5utils.c -o md5utils.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c modparam.c -o modparam.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c msg_translator.c -o msg_translator.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c pass_fd.c -o pass_fd.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c proxy.c -o proxy.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c qvalue.c -o qvalue.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c re.c -o re.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c receive.c -o receive.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c resolve.c -o resolve.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c route.c -o route.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c route_struct.c -o route_struct.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c script_cb.c -o script_cb.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c socket_info.c -o socket_info.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c sr_module.c -o sr_module.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c stats.c -o stats.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c tcp_main.c -o tcp_main.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c tcp_read.c -o tcp_read.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c timer.c -o timer.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c tsend.c -o tsend.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c udp_server.c -o udp_server.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c unixsock_server.c -o unixsock_server.o gcc -g -O9 -funroll-loops -Wcast-align -Wall -minline-all-stringops -malign-double -falign-loops -mcpu=athlon -DNAME='"ser"' -DVERSION='"0.9.2"' -DARCH='"i386"' -DOS='"linux"' -DCOMPILER='"gcc 3.2"' -D__CPU_i386 -D__OS_linux -DCFG_DIR='"/usr/local//etc/ser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DDNS_IP_HACK -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DDBG_QM_MALLOC -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETHOSTBYNAME2 -DHAVE_UNION_SEMUN -DHAVE_SCHED_YIELD -DHAVE_MSG_NOSIGNAL -DHAVE_MSGHDR_MSG_CONTROL -DHAVE_ALLOCA_H -c usr_avp.c -o usr_avp.o In file included from usr_avp.c:45: mem/shm_mem.h: In function `_shm_malloc': mem/shm_mem.h:135: too many arguments to function `fm_malloc' mem/shm_mem.h: In function `_shm_realloc': mem/shm_mem.h:146: too many arguments to function `fm_realloc' usr_avp.c: In function `destroy_avp': usr_avp.c:313: too many arguments to function `fm_free' usr_avp.c: In function `destroy_avp_list_unsafe': usr_avp.c:328: too many arguments to function `fm_free' usr_avp.c: In function `destroy_avp_list': usr_avp.c:343: too many arguments to function `fm_free' usr_avp.c: In function `add_avp_galias': usr_avp.c:420: too many arguments to function `fm_malloc' usr_avp.c:426: too many arguments to function `fm_malloc' usr_avp.c:437: too many arguments to function `fm_malloc' usr_avp.c:459: too many arguments to function `fm_free' usr_avp.c:461: too many arguments to function `fm_free' make: *** [usr_avp.o] Error 1 ................... ========== make_and_install_output.txt ========= Tks a Lot ! Freemane 有即時通知，收艂

1 0

[Serusers] Weird register issue
by Anthony Dean 23 Jun '05

23 Jun '05

Weird issue, only with registers. Sometimes when I start ser (and openser, same issue with both) my test phone (behind nat) will not register allways getting 401 unauthorized back. But wait a minute and it will register (without resetting ser or the phone). Is this a built in feature or a bug (it only happens when I reset ser/openser after 60 seconds the server and phone behave normally). Got me stumped as the configs on the phone and ser don't change. REGISTER sip:XX.XX.XX.XX SIP/2.0. Via: SIP/2.0/UDP 192.168.10.25:5060. From: <sip:2959770500@XX.XX.XX.XX;user=phone>;tag=1886287424. To: <sip:2959770500@XX.XX.XX.XX;user=phone>. Call-ID: 147690828(a)192.168.10.25. CSeq: 1 REGISTER. Contact: Nathan Gross <sip:2959770500@192.168.10.25:5060;user=phone;transport=udp>;expires=60. User-Agent: Cisco-CP7905/1.02-040406A. Allow: ACK, BYE, CANCEL, INVITE, NOTIFY, OPTIONS, REFER, REGISTER. Content-Length: 0. SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.10.25:5060;rport=62686;received=YY.YY.YY.YY. From: <sip:2959770500@XX.XX.XX.XX;user=phone>;tag=1886287424. To: <sip:2959770500@XX.XX.XX.XX;user=phone>;tag=264d52ef5b2f408fc315ea210b15c82e .5681. Call-ID: 147690828(a)192.168.10.25. CSeq: 1 REGISTER. WWW-Authenticate: Digest realm="XX.XX.XX.XX", nonce="42b9e43447292eaaa08c9b86dcc1e23344ac3c6a", qop="auth". Content-Length: 0. Thank You Anthony Dean Network Design & Support Tristar Communications This correspondence is considered confidential and any reproduction for the purpose of public disclosure is forbidden without written permission by the author signed above. If you are not the intended recipient, please immediately notify the sender and delete any copies. If you are responding to this e-mail, please append or attach all previous correspondence as a point of reference.

2 1

[Serusers] desc_time_order not found in registrar
by TCCHAN 23 Jun '05

23 Jun '05

Dear All, I am using ser 0.9.0 and found that the line modparam("registrar","desc_time_order",1) gives a error: set_mod_param_regex: parameter <desc_time_order> not found in module <registrar>. I look at the documentation, there is such a param is registrar. What's happening here? Thank for any help available. Regards, TC Chan

2 1

[Serusers] Load balancing with Mediaproxy
by peiyin 23 Jun '05

23 Jun '05

Hi everyone, I'm currently stuck at load balancing 2 ser with mediaproxy. I've setup the 2 DNS record in my master db.abc with ............ ser1.abc.net.my. IN A yy.yy.yy ser2.abc.net.my. IN A zz.zz.zz _mediaproxy._tcp.abc.net.my. ......... ser1.abc.net.my. _mediaproxy._tcp.abc.net.my. ......... ser2.abc.net.my. Both domains are pingable. What should be my next step? my present sip phones are all pointed directly to ser1.xx.xx.xx and ser1 is connected to remote db A. Do i still need to setup a virtual IP with heartbeat? What will be my sip setting at my sip phone? Also, what will be in my /etc/profile (export SIP_DOMAIN=??) Thanks in advance. peiyin

3 7

[Serusers] SER with Radius Authentication
by Rafael J. Risco G.V. 23 Jun '05

23 Jun '05

Hi, I´ve configured freeradius and SER according to the Radius HOW TO document, Accounting works very well but now I am doing some tests trying to do user authentication however all the authentication requests coming to the freeradius fails and X-lite sipphone is receiving an Unauthorized message from SER, please some advice, thanks rafael PS: config files... in /usr/local/etc/raddb/users : --------- test Auth-Type := Digest, User-Password == "test" Reply-Message = "Hello, test with digest" 6609876 Auth-Type := Digest User-Password := "9876", Digest-Response = "lalalalala", Reply-Message = "Hello, ibm1" 6604321 Auth-Type := Digest User-Password := "4321", Digest-Response = "lalalalala", Reply-Message = "Hello, ibm2" --------- Some relevant data in ser.cfg: ... modparam("group_radius", "use_domain", 0) .... if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!radius_www_authorize("")) { www_challenge("", "1"); break; }; if (!save("location")) { sl_reply_error(); }; break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # does the user wish redirection on no availability? (i.e., is he # in the voicemail group?) -- determine it now and store it in # flag 4, before we rewrite the flag using UsrLoc if (radius_is_user_in("Request-URI", "voicemail")) { log(1, "requested user is in voicemail group"); setflag(4); }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { # sl_send_reply("404", "Not Found"); log(1,"unable to locate user"); route(4); break; }; }; # End of "if(uri==myself)" .... ------------------RADIUSD -X Output ---------------------------: rad_recv: Access-Request packet from host 127.0.0.1:33187, id=79, length=311 User-Name = "6604321(a)10.0.1.22" Digest-Attributes = 0x0a0936363034333231 Digest-Attributes = 0x010b31302e302e312e3232 Digest-Attributes = 0x022a34323366333163373062336631643261643330383833633238303434316632663133643136613830 Digest-Attributes = 0x040f7369703a31302e302e312e3232 Digest-Attributes = 0x030a5245474953544552 Digest-Attributes = 0x050661757468 Digest-Attributes = 0x090a3030303030303162 Digest-Attributes = 0x08224433343132424232394131453131443939334232303035304241373836433642 Digest-Response = "a6a7812ac0331324f977453c228da2ed" Service-Type = IAPP-Register Sip-URI-User = "6604321" Cisco-AVPair = "call-id=D3412ADB9A1E11D993B20050BA786C6B(a)10.0.1.22" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Processing the authorize section of radiusd.conf modcall: entering group authorize for request 8 modcall[authorize]: module "preprocess" returns ok for request 8 modcall[authorize]: module "chap" returns noop for request 8 modcall[authorize]: module "mschap" returns noop for request 8 rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "6604321" Digest-Realm = "10.0.1.22" Digest-Nonce = "423f31c70b3f1d2ad30883c280441f2f13d16a80" Digest-URI = "sip:10.0.1.22" Digest-Method = "REGISTER" Digest-QOP = "auth" Digest-Nonce-Count = "0000001b" Digest-CNonce = "D3412BB29A1E11D993B20050BA786C6B" rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 8 rlm_realm: Looking up realm "10.0.1.22" for User-Name = "6604321(a)10.0.1.22" rlm_realm: No such realm "10.0.1.22" modcall[authorize]: module "suffix" returns noop for request 8 rlm_eap: No EAP-Message, not doing EAP modcall[authorize]: module "eap" returns noop for request 8 users: Matched DEFAULT at 152 modcall[authorize]: module "files" returns ok for request 8 modcall: group authorize returns ok for request 8 rad_check_password: Found Auth-Type DIGEST auth: type "digest" Processing the authenticate section of radiusd.conf modcall: entering group authenticate for request 8 rlm_digest: Configuration item "User-Password" is required for authentication. modcall[authenticate]: module "digest" returns invalid for request 8 modcall: group authenticate returns invalid for request 8 auth: Failed to validate the user. Delaying request 8 for 1 seconds Finished request 8 Going to the next request --- Walking the entire request list --- Waking up in 1 seconds... rad_recv: Access-Request packet from host 127.0.0.1:33188, id=80, length=311 User-Name = "6609876(a)10.0.1.22" Digest-Attributes = 0x0a0936363039383736 Digest-Attributes = 0x010b31302e302e312e3232 Digest-Attributes = 0x022a34323366333163373062336631643261643330383833633238303434316632663133643136613830 Digest-Attributes = 0x040f7369703a31302e302e312e3232 Digest-Attributes = 0x030a5245474953544552 Digest-Attributes = 0x050661757468 Digest-Attributes = 0x090a3030303030303163 Digest-Attributes = 0x08224433343132424235394131453131443939334232303035304241373836433642 Digest-Response = "50fa695654b20e2eec54a1003fe15d9f" Service-Type = IAPP-Register Sip-URI-User = "6609876" Cisco-AVPair = "call-id=D3412ADE9A1E11D993B20050BA786C6B(a)10.0.1.22" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Processing the authorize section of radiusd.conf modcall: entering group authorize for request 9 modcall[authorize]: module "preprocess" returns ok for request 9 modcall[authorize]: module "chap" returns noop for request 9 modcall[authorize]: module "mschap" returns noop for request 9 rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "6609876" Digest-Realm = "10.0.1.22" Digest-Nonce = "423f31c70b3f1d2ad30883c280441f2f13d16a80" Digest-URI = "sip:10.0.1.22" Digest-Method = "REGISTER" Digest-QOP = "auth" Digest-Nonce-Count = "0000001c" Digest-CNonce = "D3412BB59A1E11D993B20050BA786C6B" rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 9 rlm_realm: Looking up realm "10.0.1.22" for User-Name = "6609876(a)10.0.1.22" rlm_realm: No such realm "10.0.1.22" modcall[authorize]: module "suffix" returns noop for request 9 rlm_eap: No EAP-Message, not doing EAP modcall[authorize]: module "eap" returns noop for request 9 users: Matched DEFAULT at 152 modcall[authorize]: module "files" returns ok for request 9 modcall: group authorize returns ok for request 9 rad_check_password: Found Auth-Type DIGEST auth: type "digest" Processing the authenticate section of radiusd.conf modcall: entering group authenticate for request 9 rlm_digest: Configuration item "User-Password" is required for authentication. modcall[authenticate]: module "digest" returns invalid for request 9 modcall: group authenticate returns invalid for request 9 auth: Failed to validate the user. Delaying request 9 for 1 seconds Finished request 9 Going to the next request --- Walking the entire request list --- Waking up in 1 seconds... --- Walking the entire request list --- Sending Access-Reject of id 79 to 127.0.0.1:33187 Waking up in 1 seconds... rad_recv: Access-Request packet from host 127.0.0.1:33189, id=81, length=311 User-Name = "6609876(a)10.0.1.22" Digest-Attributes = 0x0a0936363039383736 Digest-Attributes = 0x010b31302e302e312e3232 Digest-Attributes = 0x022a34323366333163373062336631643261643330383833633238303434316632663133643136613830 Digest-Attributes = 0x040f7369703a31302e302e312e3232 Digest-Attributes = 0x030a5245474953544552 Digest-Attributes = 0x050661757468 Digest-Attributes = 0x090a3030303030303163 Digest-Attributes = 0x08224433343132424236394131453131443939334232303035304241373836433642 Digest-Response = "e4f68760f2b3eed0ad45942b32542c92" Service-Type = IAPP-Register Sip-URI-User = "6609876" Cisco-AVPair = "call-id=D3412ADE9A1E11D993B20050BA786C6B(a)10.0.1.22" NAS-IP-Address = 127.0.0.1 NAS-Port = 5060 Processing the authorize section of radiusd.conf modcall: entering group authorize for request 10 modcall[authorize]: module "preprocess" returns ok for request 10 modcall[authorize]: module "chap" returns noop for request 10 modcall[authorize]: module "mschap" returns noop for request 10 rlm_digest: Converting Digest-Attributes to something sane... Digest-User-Name = "6609876" Digest-Realm = "10.0.1.22" Digest-Nonce = "423f31c70b3f1d2ad30883c280441f2f13d16a80" Digest-URI = "sip:10.0.1.22" Digest-Method = "REGISTER" Digest-QOP = "auth" Digest-Nonce-Count = "0000001c" Digest-CNonce = "D3412BB69A1E11D993B20050BA786C6B" rlm_digest: Adding Auth-Type = DIGEST modcall[authorize]: module "digest" returns ok for request 10 rlm_realm: Looking up realm "10.0.1.22" for User-Name = "6609876(a)10.0.1.22" rlm_realm: No such realm "10.0.1.22" modcall[authorize]: module "suffix" returns noop for request 10 rlm_eap: No EAP-Message, not doing EAP modcall[authorize]: module "eap" returns noop for request 10 users: Matched DEFAULT at 152 modcall[authorize]: module "files" returns ok for request 10 modcall: group authorize returns ok for request 10 rad_check_password: Found Auth-Type DIGEST auth: type "digest" Processing the authenticate section of radiusd.conf modcall: entering group authenticate for request 10 rlm_digest: Configuration item "User-Password" is required for authentication. modcall[authenticate]: module "digest" returns invalid for request 10 modcall: group authenticate returns invalid for request 10 auth: Failed to validate the user. Delaying request 10 for 1 seconds Finished request 10 Going to the next request Sending Access-Reject of id 80 to 127.0.0.1:33188 Waking up in 1 seconds... --- Walking the entire request list --- Waking up in 1 seconds... --- Walking the entire request list --- Sending Access-Reject of id 81 to 127.0.0.1:33189 Waking up in 2 seconds... --- Walking the entire request list --- Cleaning up request 8 ID 79 with timestamp 423f309b Waking up in 1 seconds... --- Walking the entire request list --- Cleaning up request 9 ID 80 with timestamp 423f309c Waking up in 1 seconds... --- Walking the entire request list --- Cleaning up request 10 ID 81 with timestamp 423f309d Nothing to do. Sleeping until we see a request. -- rrgv

4 5

[Serusers] SER Server behind NAT and Record Route
by Matt morris 23 Jun '05

23 Jun '05

Hello List, Here's a problem I am currently having. My SER server is behind a router and its ip is an internal ip address. Hence, SER has to listen on its internal ip (would it be possible to configure it to listen on the WAN ip?). The call establishment is fine but for some reason when the user agent (I have tried XLite and another hw ATA) hangs up it will send the BYE to the server's interal ip instead of the its WAN ip. Thus the server never receives the BYE and the other side will still be connected. I am guessing it is because the Record Route header has the internal ip instead of the WAN one. How could I rewrite the ip in the record router header to the server's WAN ip? Thank you all in advance _________________________________________________________________ Scan and help eliminate destructive viruses from your inbound and outbound e-mail and attachments. http://join.msn.com/?pgmarket=en-ca&page=byoa/prem&xAPID=1994&DI=1034&SU=ht… Start enjoying all the benefits of MSN® Premium right now and get the first two months FREE*.

2 1

[Serusers] RTPProxy Problem !
by Freeman 23 Jun '05

23 Jun '05

Hi, I installed "Ser-0.8.14" + "nathelper" + "rtpproxy" in RedHat 9.0, and add "./rtpproxy" in startup script, when i tried both sipsoftphone in the internet behind NAT that can connect but "no voice" and auto hang-up, but when both sipsoftphone in local network that no problem ! Which port range i need release for rtpproxy in the firewall ? my ser.cfg config as below : # ----------- 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 #debug=4 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- 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/textops.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/usr/local/lib/ser/modules/dbtext.so" #loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" #loadmodule "/usr/local/lib/ser/modules/mysql.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) #modparam("auth_db", "db_url", "db:/var/dbtext") #modparam("auth_db", "user_column", "user") #modparam("auth_db", "domain_column", "domain") #modparam("auth_db", "password_column", "password") #modparam("auth_db", "calculate_ha1", 1) #modparam("auth_db", "password_column_2", "ha1_2") modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) # Ping interval 30 s modparam("nathelper", "ping_nated_only", 1) # Ping only clients behind NAT modparam("nathelper", "rtpproxy_sock","/var/run/rtpproxy.sock") # -- 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 #define NAT_UAC_TEST_C_1918 0x01 /* * test for occurences of RFC1918 addresses in Contact * header field */ #define NAT_UAC_TEST_RCVD 0x02 /* * test if source address of signaling is different from * address advertised in Via */ #define NAT_UAC_TEST_V_1918 0x04 /* * test for occurences of RFC1918 addresses in SDP body */ #define NAT_UAC_TEST_S_1918 0x08 /* * test for occurences of RFC1918 addresses top Via */ 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; }; if (nat_uac_test("3")) { append_hf("Alex-hint: NAThelper\r\n"); fix_nated_contact(); # Rewrite contact with source IP of signalling if (method == "REGISTER" || ! search("^Record-Route:")) { #if (www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "1"); # }; if (method == "INVITE") { append_hf("Alex-hint: SDP rewritten\r\n"); fix_nated_sdp("3"); # Add direction=active to SDP }; log("LOG: Someone trying to register from private IP, rewriting\n"); force_rport(); # Add rport parameter to topmost Via setflag(6); # Mark as NATed }; }; 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); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); 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") { save("location"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # !! Nathelper if (isflagset(6)) { force_rtp_proxy(); append_hf("NAT: ...\r\n"); }; # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } # !! Nathelper onreply_route[1] { # NATed transaction ? append_hf("NAT: tes\r\n"); if (isflagset(6) && status == "(200)") { fix_nated_contact(); fix_nated_sdp("3"); force_rtp_proxy(); append_hf("NAT: test\r\n"); # otherwise, is it a transaction behind a NAT and we did not # know at time of request processing ? (RFC1918 contacts) } else if (nat_uac_test("1")) { fix_nated_contact(); fix_nated_sdp("3"); force_rtp_proxy(); append_hf("NAT: test1\r\n"); }; } Please help ! Tks a Lot ! Freeman 伈鴃立即知

3 9

[Serusers] Problesm with Vovida and Mediaproxy
by Alberto Cruz 22 Jun '05

22 Jun '05

Hi guys: Does anybody knows if there is a problem using SER 0.9.2, Mediaproxy 1.3.1 and Vovida B2BUA Vocal 1.5.0? I'm having no audio when I try to place a call even if the call is established. I'm attaching my ser.cfg and a ngrep debug for reference The only weird thing that I can notice is that the B2BUA is using the localhost IP as the Via Address even when the invites was sent to the Public IP address. Did I make somtehing wrong at the ser.cfg file? Regards Alberto Cruz # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global configuration parameters ------------------------ debug=5 # debug level (cmd line: -ddddd) fork=yes log_stderror=no /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes # (cmd line: -E) debug=7 */ check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) listen=200.94.75.6 port=5060 alias=200.94.75.6 alias=sipsrv.conecta-te.com fifo_mode=0666 fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database 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_radius.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/uri_radius.so" loadmodule "/usr/local/lib/ser/modules/group.so" loadmodule "/usr/local/lib/ser/modules/group_radius.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/acc.so" loadmodule "/usr/local/lib/ser/modules/domain.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) modparam("domain", "db_mode", 1) modparam("usrloc|domain", "db_url", "mysql://ser:heslo@localhost/ser") #modparam("group", "use_domain", 0) # -- 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") ### Radius Configuration ### #modparam("auth_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") #modparam("auth_radius", "service_type", 15) #modparam("group_radius", "radius_config", "/usr/local/etc/radiusclient/radiusclient.conf") #modparam("group_radius", "use_domain", 0) # -- nathelper params -- modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) # -- mediprox params -- 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") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("registrar", "nat_flag", 6) # ------------------------- 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")) { if (method!="ACK") { sl_send_reply("483","Too Many Hops"); }; break; }; if (msg:len > max_len ) { if (method!="ACK") { 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 #---------------------------------------------------- if (method=="INVITE" && client_nat_test("3")){ record_route_preset("200.94.75.6:5060;nat=yes"); }else if (!method=="REGISTER"){ record_route(); }; #--------------------------------------------- # we tear down the call before entering to loose route #--------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); append_hf("P-hint: finishing call\r\n"); }; #------------------------------------------------- # subsequent messages withing a dialog should take the # path determined by record-routing #------------------------------------------------- if (loose_route()) { append_hf("P-hint: rr-enforced\r\n"); # The following lines are added due media proxy if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") && !search("^Route:.*;nat=yes")) { # Mark as NAT'ed setflag(6); use_media_proxy(); }; }; # end media session for BYE and CANCEL is done above # before entering the loose route. no need to call it here route(1); break; }; #---------------------------------------- # processing the most common sip messages # for all the calls #---------------------------------------- if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(5); route(1); break; }; if (uri==myself) { #--------------------------------- #all the calls from the sip uacs should #be handled by the b2bua for authorization #and accounting #--------------------------------- if (!(src_ip==200.94.75.6 || src_ip==200.94.75.1)and(method=="INVITE" || method=="BYE" || method=="CANCEL")) { log("Hey this request is NOT from B2BUA\n"); log("Forwarding to B2BUA\n"); if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; forward(200.94.75.6,5065); break; }; if (method=="ACK") { route(6); break; } else if (method=="CANCEL" || method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; route(4); if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } route[1] { #------------------------------- # Default SIP message handler # forward to current uri now; use stateful forwarding; that # works reliably even if we forward from TCP to UDP #------------------------------ t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); append_hf("P-hint: finishing call no-relay\r\n"); }; 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 (is_from_local() || uri==myself) { if (!radius_www_authorize("")) { www_challenge("", "0"); break; } else if (!check_to()){ sl_send_reply("403", "Username!=To not allowed"); break; }; if (!save("location")) { sl_reply_error(); }; } else { append_hf("P-hint: outbound alias\r\n"); sl_send_reply("403", "This domain is not served here"); }; } route[3] { #-------------------------- #"CANCEL" and "INVITE" message handler #-------------------------- if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if (method=="INVITE") { if (!(is_from_local() || uri==myself || is_uri_host_local())) { sl_send_reply("403", "Relaying is forbidden"); break; }; } else if (method=="CANCEL") { route(1); break; }; route(4); if (!lookup("location")) { if ((uri=~"sip:01[1-9][0-9]+@.*") || (uri=~"sip:00[1-9][0-9]+@.*")) { rewritehost("200.94.75.1"); }; } else { sl_send_reply("404", "User not found"); break; }; route(5); route(1); } route[4] { #--------------- #we need to check for the alias #--------------- lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; } route[5] { #------------------------------------- #RTP Proxy handler #------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); append_hf("P-hint: are you nated?\r\n"); }; } route[6] { #-------------------------- #ACK handler #-------------------------- route(4); lookup("location"); route(1); } failure_route[1] { end_media_session(); append_hf("P-hint: finishing call on failure\r\n"); } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|(2[0-9][0-9])")) { if (client_nat_test("1")) { fix_nated_contact(); }; if (!search("^Content-Length:\ +0")) { use_media_proxy(); append_hf("P-hint: use nat for call\r\n"); }; }; } interface: any filter: ip and ( port 5060 ) match: sip ##### U 2005/06/22 03:33:01.181845 200.67.33.247:61973 -> 200.94.75.6:5060 INVITE sip:018183324166@sipsrv.conecta-te.com SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-a8bb0d6a. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 101 INVITE. Max-Forwards: 70. Contact: 8183651723 <sip:8183651723@172.31.253.127:5060>. Expires: 240. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 314. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 2188429 2188429 IN IP4 172.31.253.127. s=-. c=IN IP4 172.31.253.127. t=0 0. m=audio 16470 RTP/AVP 18 0 4 8 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 2005/06/22 03:33:01.184060 200.94.75.6:5060 -> 200.94.75.6:5065 INVITE sip:018183324166@sipsrv.conecta-te.com SIP/2.0. Record-Route: <sip:018183324166@200.94.75.6:5060;nat=yes;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=0. Via: SIP/2.0/UDP 172.31.253.127:5060;rport=61973;received=200.67.33.247;branch=z9hG4bK-a8bb0d6a. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 101 INVITE. Max-Forwards: 16. Contact: 8183651723 <sip:8183651723@200.67.33.247:61973>. Expires: 240. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 314. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Supported: x-sipura. Content-Type: application/sdp. . v=0. o=- 2188429 2188429 IN IP4 172.31.253.127. s=-. c=IN IP4 172.31.253.127. t=0 0. m=audio 16470 RTP/AVP 18 0 4 8 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 2005/06/22 03:33:01.195680 200.94.75.6:5065 -> 200.94.75.6:5060 INVITE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 127.0.0.1:5065. To: <sip:018183324166@sipsrv.conecta-te.com>. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. CSeq: 101 INVITE. Max-Forwards: 16. Expires: 240. Contact: <sip:8183651723@127.0.0.1:5065>. Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER. Supported: x-sipura. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Type: application/sdp. Content-Length: 314. . v=0. o=- 2188429 2188429 IN IP4 172.31.253.127. s=-. c=IN IP4 172.31.253.127. t=0 0. m=audio 16470 RTP/AVP 18 0 4 8 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=fmtp:101 0-15. a=ptime:30. # U 2005/06/22 03:33:01.201069 200.94.75.6:5060 -> 200.94.75.6:5065 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 127.0.0.1:5065;rport=5065;received=200.94.75.6. To: <sip:018183324166@sipsrv.conecta-te.com>. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. CSeq: 101 INVITE. Server: Sip EXpress router (0.9.2 (i386/linux)). Content-Length: 0. Warning: 392 200.94.75.6:5060 "Noisy feedback tells: pid=22827 req_src_ip=200.94.75.6 req_src_port=5065 in_uri=sip:018183324166@200.94.75.6:5060 out_uri=sip:018183324166@200.94.75.1:5060 via_cnt==1". . # U 2005/06/22 03:33:01.201175 200.94.75.6:5060 -> 200.94.75.1:5060 INVITE sip:018183324166@200.94.75.1:5060 SIP/2.0. Record-Route: <sip:018183324166@200.94.75.6:5060;nat=yes;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK6512.acf311c7.0. Via: SIP/2.0/UDP 127.0.0.1:5065;rport=5065;received=200.94.75.6. To: <sip:018183324166@sipsrv.conecta-te.com>. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. CSeq: 101 INVITE. Max-Forwards: 15. Expires: 240. Contact: <sip:8183651723@200.94.75.6:5065>. Allow: ACK,BYE,CANCEL,INFO,INVITE,NOTIFY,OPTIONS,REFER. Supported: x-sipura. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Type: application/sdp. Content-Length: 311. P-hint: are you nated?. . v=0. o=- 2188429 2188429 IN IP4 172.31.253.127. s=-. c=IN IP4 200.94.75.6. t=0 0. m=audio 35024 RTP/AVP 18 0 4 8 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. a=fmtp:101 0-15. a=ptime:30. # U 2005/06/22 03:33:01.216824 200.94.75.1:5060 -> 200.94.75.6:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK6512.acf311c7.0,SIP/2.0/UDP 127.0.0.1:5065;rport=5065;received=200.94.75.6. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. Date: Wed, 22 Jun 2005 08:33:22 GMT. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U 2005/06/22 03:33:06.105293 200.94.75.1:5060 -> 200.94.75.6:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK6512.acf311c7.0,SIP/2.0/UDP 127.0.0.1:5065;rport=5065;received=200.94.75.6. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. Date: Wed, 22 Jun 2005 08:33:22 GMT. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 210. . v=0. o=CiscoSystemsSIP-GW-UserAgent 133 5493 IN IP4 200.94.75.1. s=SIP Call. c=IN IP4 200.94.75.1. t=0 0. m=audio 18168 RTP/AVP 18. c=IN IP4 200.94.75.1. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. # U 2005/06/22 03:33:06.108419 200.94.75.6:5060 -> 200.94.75.6:5065 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 127.0.0.1:5065;rport=5065;received=200.94.75.6. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. Date: Wed, 22 Jun 2005 08:33:22 GMT. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow-Events: telephone-event. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 210. P-hint: use nat for call. . v=0. o=CiscoSystemsSIP-GW-UserAgent 133 5493 IN IP4 200.94.75.1. s=SIP Call. c=IN IP4 200.94.75.6. t=0 0. m=audio 35024 RTP/AVP 18. c=IN IP4 200.94.75.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. # U 2005/06/22 03:33:11.985671 200.94.75.1:5060 -> 200.94.75.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK6512.acf311c7.0,SIP/2.0/UDP 127.0.0.1:5065;rport=5065;received=200.94.75.6. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. Date: Wed, 22 Jun 2005 08:33:22 GMT. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:018183324166@200.94.75.1:5060>. Record-Route: <sip:018183324166@200.94.75.6:5060;nat=yes;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Type: application/sdp. Content-Length: 210. . v=0. o=CiscoSystemsSIP-GW-UserAgent 133 5493 IN IP4 200.94.75.1. s=SIP Call. c=IN IP4 200.94.75.1. t=0 0. m=audio 18168 RTP/AVP 18. c=IN IP4 200.94.75.1. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. # U 2005/06/22 03:33:11.988704 200.94.75.6:5060 -> 200.94.75.6:5065 SIP/2.0 200 OK. Via: SIP/2.0/UDP 127.0.0.1:5065;rport=5065;received=200.94.75.6. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. Date: Wed, 22 Jun 2005 08:33:22 GMT. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 101 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO. Allow-Events: telephone-event. Contact: <sip:018183324166@200.94.75.1:5060>. Record-Route: <sip:018183324166@200.94.75.6:5060;nat=yes;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Type: application/sdp. Content-Length: 210. P-hint: use nat for call. . v=0. o=CiscoSystemsSIP-GW-UserAgent 133 5493 IN IP4 200.94.75.1. s=SIP Call. c=IN IP4 200.94.75.6. t=0 0. m=audio 35024 RTP/AVP 18. c=IN IP4 200.94.75.6. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=ptime:20. # U 2005/06/22 03:33:12.082614 200.67.33.247:61973 -> 200.94.75.6:5060 ACK sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-c5ebdced. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 101 ACK. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. Contact: 8183651723 <sip:8183651723@172.31.253.127:5060>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:12.085346 200.94.75.6:5060 -> 127.0.0.1:5065 ACK sip:018183324166@127.0.0.1:5065 SIP/2.0. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-c5ebdced. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 101 ACK. Max-Forwards: 16. Contact: 8183651723 <sip:8183651723@172.31.253.127:5060>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. P-hint: rr-enforced. . # U 2005/06/22 03:33:12.087437 200.94.75.6:5065 -> 200.94.75.6:5060 ACK sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 127.0.0.1:5065. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. CSeq: 101 ACK. Max-Forwards: 70. Route: <sip:018183324166@200.94.75.1:5060>. Content-Length: 0. . # U 2005/06/22 03:33:12.090626 200.94.75.6:5060 -> 200.94.75.1:5060 ACK sip:018183324166@200.94.75.1:5060 SIP/2.0. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=0. Via: SIP/2.0/UDP 127.0.0.1:5065;received=200.94.75.6. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. CSeq: 101 ACK. Max-Forwards: 16. Content-Length: 0. P-hint: rr-enforced. . # U 2005/06/22 03:33:20.813952 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:20.816802 200.94.75.6:5060 -> 127.0.0.1:5065 BYE sip:018183324166@127.0.0.1:5065 SIP/2.0. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK47f1.7c391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 16. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. P-hint: finishing call. P-hint: rr-enforced. . # U 2005/06/22 03:33:20.818924 200.94.75.6:5065 -> 200.94.75.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.6:5060;branch=z9hG4bK47f1.7c391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:20.820445 200.94.75.6:5065 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060;lr=on SIP/2.0. Via: SIP/2.0/UDP 127.0.0.1:5065. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@200.94.75.1:5060>. Content-Length: 0. . # U 2005/06/22 03:33:20.823726 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:20.826479 200.94.75.6:5060 -> 200.94.75.1:5060 BYE sip:018183324166@200.94.75.1:5060 SIP/2.0. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK3512.5c7c5523.0. Via: SIP/2.0/UDP 127.0.0.1:5065;received=200.94.75.6. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. CSeq: 102 BYE. Max-Forwards: 16. Content-Length: 0. P-hint: finishing call. P-hint: rr-enforced. . # U 2005/06/22 03:33:20.833064 200.94.75.1:5060 -> 200.94.75.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK3512.5c7c5523.0,SIP/2.0/UDP 127.0.0.1:5065;received=200.94.75.6. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. Date: Wed, 22 Jun 2005 08:33:41 GMT. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. Server: Cisco-SIPGateway/IOS-12.x. Content-Length: 0. CSeq: 102 BYE. . # U 2005/06/22 03:33:20.834554 200.94.75.6:5060 -> 200.94.75.6:5065 SIP/2.0 200 OK. Via: SIP/2.0/UDP 127.0.0.1:5065;received=200.94.75.6. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=1B2C9420-70E. Date: Wed, 22 Jun 2005 08:33:41 GMT. Call-ID: 7493ff1a89cef56a809686afb229e4ac(a)127.0.0.1. Server: Cisco-SIPGateway/IOS-12.x. Content-Length: 0. CSeq: 102 BYE. . # U 2005/06/22 03:33:21.314404 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:21.317081 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:22.312811 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:22.315462 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:24.315246 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:24.317882 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . ##### U 2005/06/22 03:33:28.313369 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:28.316301 200.94.75.6:5060 -> 127.0.0.1:5065 BYE sip:018183324166@127.0.0.1:5065 SIP/2.0. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK47f1.8c391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 16. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. P-hint: finishing call. P-hint: rr-enforced. . # U 2005/06/22 03:33:28.317393 200.94.75.6:5065 -> 200.94.75.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.6:5060;branch=z9hG4bK47f1.8c391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:28.320084 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:32.313366 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:32.316045 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:36.311492 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:36.314360 200.94.75.6:5060 -> 127.0.0.1:5065 BYE sip:018183324166@127.0.0.1:5065 SIP/2.0. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK47f1.9c391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 16. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. P-hint: finishing call. P-hint: rr-enforced. . # U 2005/06/22 03:33:36.315488 200.94.75.6:5065 -> 200.94.75.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.6:5060;branch=z9hG4bK47f1.9c391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:36.318179 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:40.312612 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:40.315279 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:44.312361 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:44.315262 200.94.75.6:5060 -> 127.0.0.1:5065 BYE sip:018183324166@127.0.0.1:5065 SIP/2.0. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK47f1.ac391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 16. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. P-hint: finishing call. P-hint: rr-enforced. . # U 2005/06/22 03:33:44.316366 200.94.75.6:5065 -> 200.94.75.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.6:5060;branch=z9hG4bK47f1.ac391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:44.319075 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:48.312358 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:48.315043 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:52.312107 200.67.33.247:61973 -> 200.94.75.6:5060 BYE sip:018183324166@200.94.75.6:5060 SIP/2.0. Via: SIP/2.0/UDP 172.31.253.127:5060;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 70. Route: <sip:018183324166@127.0.0.1:5065>. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. . # U 2005/06/22 03:33:52.315065 200.94.75.6:5060 -> 127.0.0.1:5065 BYE sip:018183324166@127.0.0.1:5065 SIP/2.0. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Via: SIP/2.0/UDP 200.94.75.6;branch=z9hG4bK47f1.bc391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Max-Forwards: 16. User-Agent: Linksys/PAP2-2.0.12(LS). Content-Length: 0. P-hint: finishing call. P-hint: rr-enforced. . # U 2005/06/22 03:33:52.316179 200.94.75.6:5065 -> 200.94.75.6:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 200.94.75.6:5060;branch=z9hG4bK47f1.bc391b44.0. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . # U 2005/06/22 03:33:52.318886 200.94.75.6:5060 -> 200.67.33.247:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP 172.31.253.127:5060;received=200.67.33.247;branch=z9hG4bK-eb281eb1. To: <sip:018183324166@sipsrv.conecta-te.com>;tag=7f9ab200. From: 8183651723 <sip:8183651723@sipsrv.conecta-te.com>;tag=cc6b17aa14d1536ao0. Call-ID: f4ac9caa-c775926a(a)172.31.253.127. CSeq: 102 BYE. Record-Route: <sip:200.94.75.6;ftag=cc6b17aa14d1536ao0;lr=on>. Content-Length: 0. . ############exit 89 received, 0 dropped

1 0

[Serusers] Re: [Sems] sems module
by harry gaillac 22 Jun '05

22 Jun '05

Stefan, May be you read my emails about asterisk+ser+mwi. i sent questions to find a sip media server for ser but asterisk don't seem to be the right way for ser (just pstn gateway). how many devoloppers work on sems project ? I don't understand ivr plugin this one can provide IVR UMS/mwi and more but sems conf file provide one file *.py for this module !? Based on IVR plugin is it possible to write some dedicated modules : UMS+mwi module MOH module IVR module Harry --- Stefan Sayer <sayer(a)cs.tu-berlin.de> a écrit : > harry gaillac wrote: > > hello, > > > > > > If i add this code will it play music as in xmms > > playlist for example. > > > > what i expect from moh module is to look in the > moh > > directory for one or more wav file (-this > directory is > > defined in sems.cfg). > > > > if a least one file is found moh play the same > file > > (boucle) . > > > > If more than one file is found all playlist is > palyed > > (boucle) > > > > i expect from moh module it provide something like > > xmms > > if you want to listen the same music. > > > > I agree i know what the moh should do but > translate my > > > > need into C++ is a problem for me. > it is way easier to write a short python script: > import ivr > import os > > ... > > def pushFiles(): > print "pushing files in the media queue..." > for file in os.listdir("/path/to/mymusic"): > fullname = os.path.join(directory, > file) > print "enqueueing " + fullname > ivr.enqueueMediaFile(fullname, false) > > ... > > This could of course be optimized; it will open all > files when the media > queue is empty...just to get you started. > > see e.g. http://effbot.org/librarybook/os-path.htm > or > http://docs.python.org/lib/os-file-dir.html > > Stefan > > > > > > Thanks for help > > > > Harry > > > > > > --- Stefan Sayer <sayer(a)cs.tu-berlin.de> a écrit : > > > > > >>pardon: > >>Stefan Sayer wrote: > >> > >>>and some python documentation ( > >> > >>www.python.org/docs ). > >>www.python.org/doc/ > >> > >>>... > >> > >>>If you prefer c++ you can simply extend the > >> > >>announcement module > >> > >>>// somewhere in onSessionStart > >>>while (!getSession()->sess_stopped.get()) { > >>> getSession()->rtp_str.play("file1.wav"); > >>> getSession()->rtp_str.play("file2.wav"); > >>> getSession()->rtp_str.play("file3.wav"); > >>>} > >> > >>vector<string> files; > >>files.push_back("file1.wav"); > >>files.push_back("file2.wav"); > >>files.push_back("file3.wav"); > >> > >>vector<string>::iterator it = files.begin(); > >>while (!getSession()->sess_stopped.get()) { > >> getSession()->rtp_str.play(*it); > >> it++; > >> if (it == files.end()) > >> it = files.begin(); > >>} > >> > >> > >>>Stefan > >>> > >>> > >>> > >>>_______________________________________________ > >> > >>Sems mailing list > >>sems(a)lists.iptel.org > >>http://lists.iptel.org/cgi-bin/mailman/listinfo/sems > >> > > > > > > > > > > > > > > > > > > > ___________________________________________________________________________ > > > Appel audio GRATUIT partout dans le monde avec le > nouveau Yahoo! Messenger > > Téléchargez cette version sur > http://fr.messenger.yahoo.com > > > > > > ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

2 1

[Serusers] 5060 Firewalled
by Daniel Poulsen 22 Jun '05

22 Jun '05

Hello- I have a client in a building where the IT staff refuse to open port 5060 for him citing 'security concerns'. Is there a way to get SER to listen on multiple ports? or do I have to start another instance of SER with a different ser.cfg? Thanks for your input. Dan

2 1

RE: [Serusers] 5060 Firewalled
by Giudice, Salvatore 22 Jun '05

22 Jun '05

Where is this SER going to be deployed? at the customer location or in your rack? If you wanted the same installation to listen on two different ports, you could install a second instance or configure port forwarding. All this is sort of useless though and creates something else to support. Maybe they will be willing to buy an SBC like an ingate or a jasomi and deploy it at the customer location instead of opening the ominous scary port 5060... Either way, good luck. _____ From: Daniel Poulsen [mailto:dpoulsen@gmail.com] Sent: Wednesday, June 22, 2005 12:20 PM To: serusers(a)lists.iptel.org Subject: [Serusers] 5060 Firewalled Hello- I have a client in a building where the IT staff refuse to open port 5060 for him citing 'security concerns'. Is there a way to get SER to listen on multiple ports? or do I have to start another instance of SER with a different ser.cfg? Thanks for your input. Dan

1 0

RE: [Serusers] pa cannot work as a B2BUA?
by Simith Nambiar 22 Jun '05

22 Jun '05

-----Original Message----- From: Jamey Hicks [mailto:jamey.hicks@hp.com] Sent: Monday, June 13, 2005 11:23 PM To: mma Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] pa cannot work as a B2BUA? mma wrote: > hi, > > i read in pa module that it cannot subsribe to presence of users > using SIP, > so how can i do to manage the presence of my contact list > PA still does not have support to SUBSCRIBE to contacts. What you can do is set up PA so that it only handles presence for users who register against and publish presence information to your SER server, and to forward subscribe requests belonging to other registrars or UAs. > i added pa module to SER and a had this errors in response to a > SUBSCRIBE request > > parse_hfs(): Error while parsing headers > I would have to see the message to be able to tell what is wrong. Which UA was generating the subscription? Which version of SER? -Jamey Hi Jamey, I have a similar Error , when using MSN Messenger Ver 4.6, and Ser-0.9.0 , Please let me know if you have any hints to look for problems ! Cheers. Simith _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers ______________________________________________________________________ This email has been scanned via a message filtering facility. If you have any issues with your email, please contact the IT Dept.

5 8

[serusers] pa module
by michael p 22 Jun '05

22 Jun '05

excuse me i have this errors too : 12(942) parse_hfs(): Error while parsing headers 12(942) handle_subscription(): Error while parsing message header 12(942) ERROR: t_reply: cannot send a t_reply to a message for which no T-state has been established 12(942) send_reply(): Error while sending 400 Bad Request someone can help me? thanks M.

3 3

RE: [Serusers] append_to_reply and save behaviour
by Fermín Galán Márquez 22 Jun '05

22 Jun '05

Ok. Thanks! I thought that the response was sent when the route block ends (that is, when the 'break' action is reached). Best regards, ------ Fermín Agora Systems, S. A. -----Mensaje original----- De: Samuel Osorio Calvo [mailto:samuel.osorio@nl.thalesgroup.com] Enviado el: miércoles, 22 de junio de 2005 13:15 Para: fermin.galan(a)agora-2000.com; serdev(a)lists.iptel.org; serusers(a)lists.iptel.org CC: sd-ims(a)agws.agora-2000.com Asunto: Re: [Serusers] append_to_reply and save behaviour The reason is that the function save sends the response and all further processing can not affect this response (which is already travelling back). I don't know if it can be called bug, feature, or just design... ;) SER config file executes the commands sequentially (besides route blocks) and you have to be careful with the order. Samuel. Unclassified. >>> Fermín Galán Márquez <fermin.galan(a)agora-2000.com> 06/22/05 12:47PM >>> Hello, I've found and estrange behaviour in the 'append_to_reply' action when used in conjunction with 'save' action during REGISTER processing. In particular, if I use in ser.cfg: ... append_to_reply("Service-Route: <sip:orig@scscf.domain1.com;lr>\r\n"); save("domain1"); ... Then the Service-Route header is added to the "200 OK". But if changing the relative order of the actions: ... save("domain1"); append_to_reply("Service-Route: <sip:orig@scscf.domain1.com;lr>\r\n"); ... Then the 'append_to_reply' is ignored: no header is added to the response. I think is a weird behaviour... I would like to know if this is a bug or a feature (I have read the textops documentation at http://www.iptel.org/ser/doc/modules/html/textops.html#AEN104 and nothing about problems with 'save' are mentioned). Thank you in advance! Best regards, ------ Fermín Agora Systems, S. A. _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] stuck while using fork=no
by Chia Huey Lim 22 Jun '05

22 Jun '05

Hi. I am using ser 0.9.0. The moment I change from fork = yes to fork = no to enter debug mode And I attempt to "serctl restart" I was stuck in [root@localhost ser-0.9.0]# serctl start Starting SER : The moment I change back to fork = yes, I can start SER again. Any further adjustment I need to do when I switch from fork = yes to fork = no? Thanks in advance. Regards, Chia

2 1

[Serusers] SEMS with multiple SERs
by Andreas Granig 22 Jun '05

22 Jun '05

Hi all, I gave the SEMS-conference module a try and it looks quite promising. But one thing comes in my mind: We've multiple SERs balanced via DNS-SRV, so a call to a conference room can end up in any of the SEMS instances where SER is running, which results in different conference rooms for one single number. To solve this issue, the communication between SER and SEMS has to be made networking enabled (assuming a simple case where one SEMS-server serves many SERs; for multiple SEMS-Instances maybe a common DB or a dispatcher has to be used to find the correct SEMS-Server). Some possibilites come in my mind: 1. Adding a t_write_udp/tcp() function to SER's tm-module and add a UDP/TCP-Listener to SEMS. 2. Enhance the new xmlrpc-module of SER to allow sending XMLRPC-requests from inside the SER-config and add an XMLRPC-server to SEMS. Is there already anybody working on this? Or any comments on the approaches? Andy

2 2

[Serusers] Re: [Sems] SEMS crashes, fifo errors
by Llanos Serna García-Conde 22 Jun '05

22 Jun '05

1 0

[Serusers] ASTERISK+SER+MWI
by harry gaillac 22 Jun '05

22 Jun '05

Hello, I wish to use asterisk as voicemail system with ser My phones are registered on SER and asterisk provide conference ivr and voicemail system. I want to use Asterisk Realtime Architecture to store voicemessages and configuration . Michael Shuler has written a patch for mwi. http://bugs.digium.com/bug_view_page.php?bug_id=0002980 I need help to set up voicemail storage and how to send mwi to clients registered on ser Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

4 7

Re: [Serusers] SER act as a 3cpp server
by Samuel Osorio Calvo 22 Jun '05

22 Jun '05

I don't think SER can act as a 3pcc because it does not have a full SIP stack and it is not able to act as a b2bua(which is what a 3pcc mainly is). Moreover, since it is not a call statefull proxy is hardly impossible to add several functionalities for a SIP entityy to work as a 3pcc. my 2 cents, Samuel. Unclassified. >>> Iqbal <iqbal(a)gigo.co.uk> 06/22/05 02:54PM >>> I havent looked at the code below, but th MWI works if the phone is registered in asterisk, if the phone is registered in SER, then MWI in asterisk needs to send a message back via ser to the phone to make its light flash. As for the call being in asterisk, what I mean is if the rtp stream is in asterisk, then when you press keys asterisk can detect these, and based on keystrokes be able to do what you wish. Doin this in SER I dont think is possible since it works on SIP messages as opposed to keystrokes (which come as part of the RTP stream, having said that I do believe some IP phones can be set to send these outside using SIP INFO (I havent tested this)), and hence when you use hold/transfer and all the rest they should send a MESSAGE which SER can recognise, else it will just be buttons you are pressing which do little, if you have transferred the cvall to asterisk, asterisk should recognise and act on it. Iqbal harry gaillac wrote: >What do you mean "if the call is in asterisk" >what about MWI_Version_11.tar.gz > >some examples help me > >Regards >harry >--- Iqbal <iqbal(a)gigo.co.uk> a écrit : > > > >>if the call is in asterisk, then yes you should be >>able to do this, >>since asterisk should be able to detect your >>keystrokes, and based on >>that direct you to whatever >> >>Iqbal >> >>harry gaillac wrote: >> >> >> >>>Here is my problem >>> >>>I use 2 polycom ip phones (see posted mail) >>>asterisk+ser . >>> >>>When i press hold key i wish caller listen music >>> >>> >>via >> >> >>>asterisk MOH >>>Polycom told me you need a 3pcc server >>>Does asterisk is able to act as a 3pcc server ? >>> >>>caller controller callee MOH >>> >>>Regards >>>--- Quang Minh PHAN <quang-minh.phan(a)wengo.fr> a >>> >>> >>écrit >> >> >>>: >>> >>> >>> >>> >>> >>>>You can find a shell script (examples/ctd.sh) >>>> >>>> >>which >> >> >>>>implement Click To Dial >>>>example by using SER FIFO. However, you shouldn't >>>>use SER to implement a >>>>3PCC server as it's not designed for that purpose. >>>> >>>>Minh >>>> >>>>-----Message d'origine----- >>>>De : harry gaillac [mailto:gaillacharry@yahoo.fr] >>>>Envoyé : mercredi 22 juin 2005 12:25 >>>>À : Quang Minh PHAN >>>>Objet : RE: [Serusers] SER act as a 3cpp server >>>> >>>>NO, >>>> >>>>I mean 3pcc . >>>> >>>>Harry >>>> >>>> >>>> >>>> >>>> >>>> >>http://www.iptel.org/info/players/ietf/callprocessing/3pcc/draft-ietf-sippin >> >> >>> >>> >>> >>> >>>>g-3pcc-03.txt >>>>--- Quang Minh PHAN <quang-minh.phan(a)wengo.fr> a >>>>écrit >>>>: >>>> >>>> >>>> >>>> >>>> >>>>>Hi, >>>>> >>>>> Did you mean 3GPP? In that case, the answer is >>>>> >>>>> >>>>> >>>>> >>>>no. >>>> >>>> >>>> >>>> >>>>>Regards, >>>>> >>>>>Minh >>>>> >>>>>-----Message d'origine----- >>>>>De : serusers-bounces(a)lists.iptel.org >>>>>[mailto:serusers-bounces@lists.iptel.org] De la >>>>>part de harry gaillac >>>>>Envoyé : mercredi 22 juin 2005 11:41 >>>>>À : serusers(a)lists.iptel.org >>>>>Objet : [Serusers] SER act as a 3cpp server >>>>> >>>>>Hello, >>>>> >>>>>Does SER is able to cat as a 3cpp server ? >>>>> >>>>>Regards >>>>>harry >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>___________________________________________________________________________ >> >> >>> >>> >>> >>> >>>>>Appel audio GRATUIT partout dans le monde avec le >>>>>nouveau Yahoo! Messenger >>>>>Téléchargez cette version sur >>>>>http://fr.messenger.yahoo.com >>>>> >>>>>_______________________________________________ >>>>>Serusers mailing list >>>>>serusers(a)lists.iptel.org >>>>>http://lists.iptel.org/mailman/listinfo/serusers >>>>> >>>>> >>>>> >>>>> >>>>> >>>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>___________________________________________________________________________ >> >> >>> >>> >>> >>> >>>>Appel audio GRATUIT partout dans le monde avec le >>>>nouveau Yahoo! Messenger >>>>Téléchargez cette version sur >>>>http://fr.messenger.yahoo.com >>>> >>>>_______________________________________________ >>>>Serusers mailing list >>>>serusers(a)lists.iptel.org >>>>http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>> >>>> >>>> >>>> >>> >>> >>> >>> >>> >>> >>> >>___________________________________________________________________________ >> >> >> >>>Appel audio GRATUIT partout dans le monde avec le >>> >>> >>nouveau Yahoo! Messenger >> >> >>>Téléchargez cette version sur >>> >>> >>http://fr.messenger.yahoo.com >> >> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >>>. >>> >>> >>> >>> >>> > > > > > > > >___________________________________________________________________________ >Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger >Téléchargez cette version sur http://fr.messenger.yahoo.com > >. > > > _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] (no subject)
by lucape＠inwind.it 22 Jun '05

22 Jun '05

can you help me to configure lcs2005 with asterisk... I use SER to resolve the problem that there is for communication protocol...LCS uses tcp, Asterik UDP. Someone, knows how to do the configuration beetwen LCS and SER , SER and Asterisk? the function of asterisk is SIP-PSTN Gateway for the LCS PC-phone communication thanks ____________________________________________________________ Navighi a 4 MEGA e i primi 3 mesi sono GRATIS. Scegli Libero Adsl Flat senza limiti su http://www.libero.it

1 0

[Serusers] SER act as a 3cpp server
by harry gaillac 22 Jun '05

22 Jun '05

Hello, Does SER is able to cat as a 3cpp server ? Regards harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

3 5

[Serusers] variables in ser.cfg
by Jack Wei 22 Jun '05

22 Jun '05

is there anyway to set temp variables in ser.cfg? if so, how to set/access? Jack Wei __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

3 2

[Serusers] Does SER may act as a 3pcc server not 3cpp !
by harry gaillac 22 Jun '05

22 Jun '05

Hello, Does SER is able to cat as a 3pcc server ? http://www.iptel.org/info/players/ietf/callprocessing/3pcc/draft-ietf-sippi… Regards harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

2 1

[Users] failover and presence/IM service
by jimmy way 22 Jun '05

22 Jun '05

Hi all, 1st question is does openser support Failover and load balance of openser server. That means we can use two or more openser server to handle the SIP request/response and use hot backup. How to do that 2nd question is that I find openser support both Jabber and SIMPLE. It look like Jabber is stable and support and many people from open source world, many plus GOOGLE. But SIMPLE is still unstable. I want to use Jabber for providing presence/IM service, plus file transfer service.(I want to using reSIProcate stack from sipfoundry as client's stack) Does openser support that all? And which is better, Jabber or SIMPLE? Regards, Jimway __________________________________ Yahoo! Mail Stay connected, organized, and protected. Take the tour: http://tour.mail.yahoo.com/mailtour.html

2 3

Re: [Serusers] append_to_reply and save behaviour
by Samuel Osorio Calvo 22 Jun '05

22 Jun '05

The reason is that the function save sends the response and all further processing can not affect this response (which is already travelling back). I don't know if it can be called bug, feature, or just design... ;) SER config file executes the commands sequentially (besides route blocks) and you have to be careful with the order. Samuel. Unclassified. >>> Fermín Galán Márquez <fermin.galan(a)agora-2000.com> 06/22/05 12:47PM >>> Hello, I've found and estrange behaviour in the 'append_to_reply' action when used in conjunction with 'save' action during REGISTER processing. In particular, if I use in ser.cfg: ... append_to_reply("Service-Route: <sip:orig@scscf.domain1.com;lr>\r\n"); save("domain1"); ... Then the Service-Route header is added to the "200 OK". But if changing the relative order of the actions: ... save("domain1"); append_to_reply("Service-Route: <sip:orig@scscf.domain1.com;lr>\r\n"); ... Then the 'append_to_reply' is ignored: no header is added to the response. I think is a weird behaviour... I would like to know if this is a bug or a feature (I have read the textops documentation at http://www.iptel.org/ser/doc/modules/html/textops.html#AEN104 and nothing about problems with 'save' are mentioned). Thank you in advance! Best regards, ------ Fermín Agora Systems, S. A. _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] append_to_reply and save behaviour
by Fermín Galán Márquez 22 Jun '05

22 Jun '05

Hello, I've found and estrange behaviour in the 'append_to_reply' action when used in conjunction with 'save' action during REGISTER processing. In particular, if I use in ser.cfg: ... append_to_reply("Service-Route: <sip:orig@scscf.domain1.com;lr>\r\n"); save("domain1"); ... Then the Service-Route header is added to the "200 OK". But if changing the relative order of the actions: ... save("domain1"); append_to_reply("Service-Route: <sip:orig@scscf.domain1.com;lr>\r\n"); ... Then the 'append_to_reply' is ignored: no header is added to the response. I think is a weird behaviour... I would like to know if this is a bug or a feature (I have read the textops documentation at http://www.iptel.org/ser/doc/modules/html/textops.html#AEN104 and nothing about problems with 'save' are mentioned). Thank you in advance! Best regards, ------ Fermín Agora Systems, S. A.

1 0

Re: [Serusers] Portal for forking call to preferredenddevice-sequential ringing
by Aisling O'Driscoll 22 Jun '05

22 Jun '05

Ok, just to recap - cos Im getting a little bit confused ;) I have two choices(?) 1. Somehow invoke sipsak to configure permanent addresses with a particular q value. 2. Develop a FIFO method to change q value. Am I correct in thinking directly modifying the usrloc table in the database is out of the question because the changes cant be updated except by SER itself in which case a reboot would required - Correct? Also lcr module (load_contacts() etc) isnt suitable for per user configurable sequential forking? Have others tried to implement similar functionality or is it usually a generic site wide sequential forking policy? Many thanks for the opinions and help so far. Aisling. > >---- Original Message ---- >From: jh(a)tutpro.com >To: greger(a)teigre.com >Subject: Re: [Serusers] Portal for forking call to >preferredenddevice-sequential ringing >Date: Tue, 21 Jun 2005 21:41:54 +0300 > >>Greger V. Teigre writes: >> >> > :-D Yes, that is true. But I didn't know that you could change >q-value in >> > FIFO? >> >>if that is not possible, then you can always use sipsak to install >>"permanent" registrations with a given q value. sipsak has an >>additional advantage over fifo in that you can apply permissions >check >>to sipsak registration, but not to fifo registration. >> >>-- juha >> >>-------------------Legal >Disclaimer--------------------------------------- >> >>The above electronic mail transmission is confidential and intended >only for the person to whom it is addressed. Its contents may be >protected by legal and/or professional privilege. Should it be >received by you in error please contact the sender at the above >quoted email address. Any unauthorised form of reproduction of this >message is strictly prohibited. The Institute does not guarantee the >security of any information electronically transmitted and is not >liable if the information contained in this communication is not a >proper and complete record of the message as transmitted by the >sender nor for any delay in its receipt. >> >> -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

5 4

[Serusers] SEMS crashes, fifo errors
by Llanos Serna García-Conde 22 Jun '05

22 Jun '05

1 0

[serusers] calling problem
by michael p 22 Jun '05

22 Jun '05

hi i have a problem : i have 2 same softphone , 1 can call the other and the other cannot call the previous. it's the same configuration for both in the softphone. i did test about this that can't call: i change the softphone, i change the laptop, i use the user that this can call use and change with mine... in basis ser.cfg what do the difference between 2 users ? nothing i guess i'm using a stun to pass over nat and it works in one direction (from one user to the other) it didn't work someone has an idea thanks M.

1 0

[serusers] pa module
by michael p 22 Jun '05

22 Jun '05

hi i'm wroting you cause i saw that you wrote a lot about pa module and you finally succeed to install it. congratulations! i 'm still trying and only one person answered me... please can you help me? my problems and questions are : 1. did you find more docs about pa module cause the README is not very long.? 2. how install it? what the difference between handle_subscription ("registrar") and publish ? 3. how i have to use the pa module and mysql ? i saw in one of your message your ser.cfg but i don't know where you found all the informations to write the lines about pa modules... 4. pa module works with all the softphone (i'm using a softphone from Xten (eyebeam))? how do they know to communicate with this module of SER? how can i do test? please help me it's very important thanks a lot Michael

2 1

Re: [Serusers] Portal for forking call to preferredenddevice-sequential ringing
by Samuel Osorio Calvo 22 Jun '05

22 Jun '05

>> 2. Develop a FIFO method to change q value. > >Yes. Except that I don't know if anybody verified that q value cannot be >changed with today's FIFO. I just asked the question... I think the current usrloc fifo commands provide enough flexibility to modify the q value, I don't know now the exact spelling but womething like: serctl ul add location user@domain user@ip {expires} {q} {replicate} I recall that there where a few values representing expiration, replicate, q value and maybe something else. From the source code, it seems that if the contact already exists, serctl ul add will update the record. If you don't have the user's record values you can use serctl ul show. I'm not 100% sure, but if you use sipsack, you should be carefull to not modifying other parameters just as received, User-Agent...am I right? Samuel. Unclassified. >>> "Greger V. Teigre" <greger(a)teigre.com> 06/22/05 06:42AM >>> Aisling O'Driscoll wrote: > Ok, just to recap - cos Im getting a little bit confused ;) > > I have two choices(?) > > 1. Somehow invoke sipsak to configure permanent addresses with a > particular q value. > 2. Develop a FIFO method to change q value. Yes. Except that I don't know if anybody verified that q value cannot be changed with today's FIFO. I just asked the question... > Am I correct in thinking directly modifying the usrloc table in the > database is out of the question because the changes cant be updated > except by SER itself in which case a reboot would required - Correct? Correct. "SER itself" is here either serctl, sipsak or FIFO. > Also lcr module (load_contacts() etc) isnt suitable for per user > configurable sequential forking? I don't know about that. There is definitely a q-value based functionality there. > Have others tried to implement similar functionality or is it usually > a generic site wide sequential forking policy? I think using CPL could be an option. Have you looked at CPL and the cpl module? g-) > Many thanks for the opinions and help so far. > Aisling. > >> >> ---- Original Message ---- >> From: jh(a)tutpro.com >> To: greger(a)teigre.com >> Subject: Re: [Serusers] Portal for forking call to >> preferredenddevice-sequential ringing >> Date: Tue, 21 Jun 2005 21:41:54 +0300 >> >>> Greger V. Teigre writes: >>> >>>> :-D Yes, that is true. But I didn't know that you could change >> q-value in >>>> FIFO? >>> >>> if that is not possible, then you can always use sipsak to install >>> "permanent" registrations with a given q value. sipsak has an >>> additional advantage over fifo in that you can apply permissions >> check >>> to sipsak registration, but not to fifo registration. >>> >>> -- juha >>> >>> -------------------Legal >> Disclaimer--------------------------------------- >>> >>> The above electronic mail transmission is confidential and intended >> only for the person to whom it is addressed. Its contents may be >> protected by legal and/or professional privilege. Should it be >> received by you in error please contact the sender at the above >> quoted email address. Any unauthorised form of reproduction of this >> message is strictly prohibited. The Institute does not guarantee the >> security of any information electronically transmitted and is not >> liable if the information contained in this communication is not a >> proper and complete record of the message as transmitted by the >> sender nor for any delay in its receipt. >>> >>> > > > > -------------------Legal > Disclaimer--------------------------------------- > > The above electronic mail transmission is confidential and intended > only for the person to whom it is addressed. Its contents may be > protected by legal and/or professional privilege. Should it be > received by you in error please contact the sender at the above > quoted email address. Any unauthorised form of reproduction of this > message is strictly prohibited. The Institute does not guarantee the > security of any information electronically transmitted and is not > liable if the information contained in this communication is not a > proper and complete record of the message as transmitted by the > sender nor for any delay in its receipt.

1 0

[Serusers] Tr: FW: Polycom soundpoint IP
by harry gaillac 22 Jun '05

22 Jun '05

Remarque : message transféré en pièce jointe. Hi all, I asked to polycom how to place on hold with music caller when i press hold key. I use SER and asterisk to provide MOH. >From Polycom: We don't support this because when in BroadSoft and Sylantro environments, the servers act as 3pcc's so the phone doesn't need to do anything for the system to get MOH (i.e., we don't need 7155 in these environments. Does Ser is able to act as a 3pcc to provide MOH when hold key is pressed ? http://www.iptel.org/info/players/ietf/callprocessing/3pcc/draft-ietf-sippi… ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

[Serusers] ===== MediaProxy Setup Problem =====
by Freeman 22 Jun '05

22 Jun '05

Hi, I have some problem when i setup the mediaproxy for ser, OS : Redhat 9 Ser : Ser 0.8.14 MediaProxy : MediaProxy 1.3.1 python : python-2.2.2-26 python-optik : python-optik-1.4-2 I tried uppack the mediaproxy to "/usr/local/" and copied the start script from "/usr/local/mediaproxy/boot/mediaproxy.other" to "/etc/init.d/mediaproxy", when i run the start script "/etc/init.d mediaproxy start", i get some error as below : Starting SER MediaProxy server: mediaproxyTraceback (most recent call last): File "/usr/local/mediaproxy/mediaproxy.py", Line 21, in ? from optparse import OptionParser ImportError: No module named optparse Traceback (most recent call last): File "/usr/local/mediaproxy/proxydispatcher.py", line 35, in ? from optparse import OptionParser ImportError: No module named optparse . How can i fix the problem ! How can i config ser.cfg for MediaProxy ? Please help ! Tks a Lot ! Freeman _______________________________________ 想即時收到新 email 通知？下載 Yahoo! Messenger http://messenger.yahoo.com.hk

1 0

[Serusers] ASTERISK+SER+MWI
by harry gaillac 22 Jun '05

22 Jun '05

Hello, I wish to use asterisk as voicemail system with ser My phones are registered on SER and asterisk provide conference ivr and voicemail system. I want to use Asterisk Realtime Architecture to store voicemessages and configuration . Michael Shuler has written a patch for mwi. http://bugs.digium.com/bug_view_page.php?bug_id=0002980 I need help to set up voicemail storage and how to send mwi to clients registered on ser Regards Harry ___________________________________________________________________________ Appel audio GRATUIT partout dans le monde avec le nouveau Yahoo! Messenger Téléchargez cette version sur http://fr.messenger.yahoo.com

1 0

Re: [Serusers] 30s calls
by Greger V. Teigre 22 Jun '05

22 Jun '05

Jerlique Ban wrote: >> In order for the call to not disconnect, the final ACK from >> caller (the ACK to the OK sent by the callee) must be >> received by the callee (Asterisk). So, your problem may be >> caused by your ser.cfg not handling ACKs properly. I have > > The ACK is most certainly being received. I have confimed this by > watching the 'sip debug' in asterisk. I neglected to mention in my > first email that another phone through a different router (with NAT) > work without issue. Its only this one phone. > > What I do see at the time of losing audio, is another INVITE. So, did the OK go through? And did Asterisk send the 100 Trying first? Probably notm because when the UA receices a 100 Trying, it should not resend the INVITE. You could try adding: sl_send_reply("100", "Trying"); Pretty early in your INVITE handling and before forwarding to asterisk. This should at least cause the UA to keep shut for awhile. > If I register the phone direct to Asterisk, it works without issue, > its only having issues through SER. > > I've got it working now, but I'm not sure what I did.... > >> If not, have a look at the >> reference configs at http://onsip.org/ to see how ACKs are handled. > > My config is based on one of their samples :) Excellent! ;-) > PS: I've added a bunch of log() statements to the file, and can see > the output on my terminal, however how do I record the src ip, or URI > with the log function? Use xlog (0.9.x). I have this right after all the "not supported" and integrity checks: # Log all messages xlog("L_INFO","%is sends %rm, r-uri:%ru, from:%fu, to:%tu, status:%rs\n"); g-)

1 0

[Serusers] how does a b2bua help billing
by Iqbal 22 Jun '05

22 Jun '05

Hi I am curious as to how a b2bua does the billing side of things, I know it sits in the middle and the media goes through it, but how does it keep count of how long the session has been open for, does it keep a continously running count of time after a INVITE or use some other non-evasive method. If it keeps a continous count (I presume in memory), how does is initiate a disconnect, does it compare the time used with that left (which is determined at call start) Any links , pointers, Iqbal

3 4

[Serusers] ser.cfg scripting error
by Jack Wei 21 Jun '05

21 Jun '05

my setup is 2 ser servers & 1 mysql server. i want to look up a user in table "location" to see if one of the two ser servers registered the user, so i've decided to follow the example in the admin guide and use exec_dset(). however, i'm getting an error output. anyone knows what's wrong with the following? if ( !exec_dset( 'QUERY="select contact from location where username=\"$SIP_OUSER\""; DEST_URI=`mysql -user -pheslo -h10.1.201.107 -Bs -e"$QUERY" ser`; if [ -z "$DEST_URI" ] then exit 1 fi; echo "$DEST_URI"; echo > /dev/null;' ) ) { xlog("L_INFO", "\n\n%ru is not online (in db)\n"); sl_send_reply( "404", "User does not exist" ); break; }; the error output is: sh: -c: line 4: syntax error: unexpected end of file 12(14932) ERROR:exec_str: no uri from QUERY="select contact from location where username=\"$SIP_OUSER\""; DEST_URI=`mysql -user -pheslo -h10.1.201.107 -Bs -e"$QUERY" ser`; if [ -z "$DEST_URI" ] then exit 1 fi; echo "$DEST_URI"; echo > /dev/null; sip:12222220002@10.1.201.191 12(14932) i'm not sure why 'sip:12222220002@10.1.201.191' is there, but i think that might be part of the problem. when i tried the following: if ( !exec_dset( 'echo sip:12222220002@10.1.201.65; echo > /dev/null' ) ) it works fine. Jack Wei __________________________________ Yahoo! Mail Mobile Take Yahoo! Mail with you! Check email on your mobile phone. http://mobile.yahoo.com/learn/mail

3 2

Re: [Serusers] Portal for forking call to preferred enddevice-sequential ringing
by Greger V. Teigre 21 Jun '05

21 Jun '05

Samuel Osorio Calvo wrote: > In fact, SER has a model for usrloc changes outside itself: FIFO > interface. Using the ul command of this interface it is possible to > add/update/remove users' information during runtime without having to > restart SER. In fact, these commands update SER's cache and this > information is immediately used by SER. It will be propagated to the > permament DB storage depending on the mode selected. :-D Yes, that is true. But I didn't know that you could change q-value in FIFO? Never really thought about that, it's logical, though :-) Kind of double work to build something that uses FIFO for 0.9.x, when head (hopefully) will get the xmlrpc. It looks like Jan plans to remove the fifo code when replacing with xmlrpc. > I would suggest to add a chapter of the FIFO/UNIX/TCP/XMLRPC > interface to the famous Getting Started Document ;) Good idea :-) However, it's sort of a moving target so far, it's quite an effort to keep the document updated, and we would rather suffer in coverage/depth than suffer from outdated info... g-) > Samuel. > > > Unclassified. >>>> "Greger V. Teigre" <greger(a)teigre.com> 06/21/05 04:53PM >>> > You get into even worse problems as SER does not really have a model > for > usrloc changes outside itself. Thus, dependent on your db mode > (memory-only, > flush, write-through etc) SER will potentially overwrite your > externally set > q-value before stopping. If you use write-through, you probably don't > get > that problem, and yes, you could probably theoretically restart SER to > reload the usrloc with new q-values (never validated that, though). > However, with a lot of locations, it's not really feasible due to > SER's > start-up loading of locations (and being unavailable while doing so). > Also, > you will probably restart in the middle of a transaction and calls > may fail (unless you use the restart safe parameter). > g-) > > Aisling wrote: >> Thank you for the replys. I will look up the post you suggested for >> further info on the lcr module. One thing I just want to clarify: >> >> " DB-based changes is the way to go - Updating the usrloc table >> outside >> SER does not work as SER loads usrloc info into memory."...... >> >> Does this mean I must modify the database and then restart SER, >> thereby >> disrupting the service for other users? >> >> Thanks, >> Aisling. >> >> -----Original Message----- >> From: Greger V. Teigre [mailto:greger@teigre.com] >> Sent: 21 June 2005 13:33 >> To: Samuel Osorio Calvo; ashling.odriscoll(a)cit.ie; serusers(a)lists.iptel.org >> Subject: Re: [Serusers] Portal for forking call to preferred end >> device-sequential ringing >> >> I believe the lcr module will do q-value based sequential forking. >> There >> was a thread on that a while back. Search for q-value and Juha. LCR >> can >> be >> found in head and as a backported 0.9.x module. >> DB-based changes is the way to go! Doing dynamic ser.cfg changes is >> not >> feasible as long you have to restart ser (and thus reload contacts, >> which >> may take a long time). Updating the usrloc table outside SER does >> not work >> as SER loads usrloc info into memory. >> g-) >> >> Samuel Osorio Calvo wrote: >>> I guess a sequential forking ordered with the q value of the contact >>> header field value would do what you are asking for. The user has >>> just to configure the appropriate value in his/her UA to the >>> preferred ringing sequence. This method is fully SIP compliant but >>> what I am not sure is if SER does sequential forking in the q order >>> (it wasn't a few months ago but some modifications might have added >>> it...someone can tell you better than me). >>> If you still want users to modify SER's database with a web >>> interface, you might try to modify the q value in the usrloc >>> database, of course if SER does q-ordered sequential forking, but I >>> just think it adds lots of complexity to a feature which SER is >>> supposed to provide together with minum user configuration in the >>> endpoints. >>> >>> Hope it helps, >>> >>> Samuel. >>> >>> >>> Unclassified. >>>>>> "Aisling" <ashling.odriscoll(a)cit.ie> 06/21/05 12:54PM >>> >>> Hello all, >>> >>> I'm hoping someone will offer an opinion as to how I should approach >>> the >>> following and if I am thinking along the correct lines: >>> >>> I am creating a web application where a user can dictate which >>> device they want a call delivered to. So if I have a user with sip >>> url "sip:2000@server" and they have registered with the SER server >>> from a pda, pc and laptop, SER will currently parallel fork a call >>> to all these >>> destinations causing all softphones to ring (correct?). However I >>> want a >>> user to choose from a drop down menu in their browser (which I'm >>> developing with servlets and JSP's) their preferred phone e.g. pda >>> and then SER will fork the call to the softphone on the pda first. >>> >>> So basically my current plan is to retrieve the information from the >>> user about their preferred phone (which will be associated with a >>> particular IP address) and then dynamically modify the ser.cfg for a >>> sequential forking rule for that user. I would appreciate opinions >>> as to >>> whether I am approaching this correctly or if there's an alternative >>> method for such functionality (perhaps such as forking the call to >>> the device which last registered or something)? >>> >>> Many Thanks, >>> Aisling. >>> >>> >>> >>> >>> -------------------Legal >>> Disclaimer--------------------------------------- >>> >>> The above electronic mail transmission is confidential and intended >>> only for the person to whom it is addressed. Its contents may be >>> protected by legal and/or professional privilege. Should it be >>> received by you in error please contact the sender at the above >>> quoted email address. Any unauthorised form of reproduction of this >>> message is strictly prohibited. The Institute does not guarantee the >>> security of any information electronically transmitted and is not >>> liable if the information contained in this communication is not a >>> proper and complete record of the message as transmitted by the >>> sender nor for any delay in its receipt. >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >> >> >> -------------------Legal >> Disclaimer--------------------------------------- >> >> The above electronic mail transmission is confidential and intended >> only >> for the person to whom it is addressed. Its contents may be protected >> by >> legal and/or professional privilege. Should it be received by you in >> error please contact the sender at the above quoted email address. >> Any unauthorised form of reproduction of this message is strictly >> prohibited. The Institute does not guarantee the security of any >> information electronically transmitted and is not liable if the >> information contained in this communication is not a proper and >> complete >> record of the message as transmitted by the sender nor for any delay >> in >> its receipt. >> >> >> -------------------Legal >> Disclaimer--------------------------------------- >> >> The above electronic mail transmission is confidential and intended >> only for the person to whom it is addressed. Its contents may be >> protected by legal and/or professional privilege. Should it be >> received by you in error please contact the sender at the above >> quoted email address. Any unauthorised form of reproduction of this >> message is strictly prohibited. The Institute does not guarantee the >> security of any information electronically transmitted and is not >> liable if the information contained in this communication is not a >> proper and complete record of the message as transmitted by the >> sender nor for any delay in its receipt.

3 2

[Serusers] Tr: Re: [Sems] sems module
by harry gaillac 21 Jun '05

21 Jun '05

Remarque : message transféré en pièce jointe. ___________________________________________________________________________ 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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[Serusers] Problem with Voice in SER0.9.0
by Bruno Oliveira 21 Jun '05

21 Jun '05

hi when I make one call enters two kphoner i don't have voice but the message functions. anyone know why ? thanks for your time -- []s, Bruno "Niggas" Oliveira Belo Horizonte - MG Msn: n1gg4s(a)gmail.com Icq: 176314647 "Todo o nosso descontentamento por aquilo que nos falta procede da nossa falta de gratidão por aquilo que temos."

2 3

Re: [Serusers] Portal for forking call to preferred end device-sequential ringing
by Samuel Osorio Calvo 21 Jun '05

21 Jun '05

In fact, SER has a model for usrloc changes outside itself: FIFO interface. Using the ul command of this interface it is possible to add/update/remove users' information during runtime without having to restart SER. In fact, these commands update SER's cache and this information is immediately used by SER. It will be propagated to the permament DB storage depending on the mode selected. I would suggest to add a chapter of the FIFO/UNIX/TCP/XMLRPC interface to the famous Getting Started Document ;) Samuel. Unclassified. >>> "Greger V. Teigre" <greger(a)teigre.com> 06/21/05 04:53PM >>> You get into even worse problems as SER does not really have a model for usrloc changes outside itself. Thus, dependent on your db mode (memory-only, flush, write-through etc) SER will potentially overwrite your externally set q-value before stopping. If you use write-through, you probably don't get that problem, and yes, you could probably theoretically restart SER to reload the usrloc with new q-values (never validated that, though). However, with a lot of locations, it's not really feasible due to SER's start-up loading of locations (and being unavailable while doing so). Also, you will probably restart in the middle of a transaction and calls may fail (unless you use the restart safe parameter). g-) Aisling wrote: > Thank you for the replys. I will look up the post you suggested for > further info on the lcr module. One thing I just want to clarify: > > " DB-based changes is the way to go - Updating the usrloc table > outside > SER does not work as SER loads usrloc info into memory."...... > > Does this mean I must modify the database and then restart SER, > thereby > disrupting the service for other users? > > Thanks, > Aisling. > > -----Original Message----- > From: Greger V. Teigre [mailto:greger@teigre.com] > Sent: 21 June 2005 13:33 > To: Samuel Osorio Calvo; ashling.odriscoll(a)cit.ie; serusers(a)lists.iptel.org > Subject: Re: [Serusers] Portal for forking call to preferred end > device-sequential ringing > > I believe the lcr module will do q-value based sequential forking. > There > was a thread on that a while back. Search for q-value and Juha. LCR > can > be > found in head and as a backported 0.9.x module. > DB-based changes is the way to go! Doing dynamic ser.cfg changes is > not > feasible as long you have to restart ser (and thus reload contacts, > which > may take a long time). Updating the usrloc table outside SER does not > work > as SER loads usrloc info into memory. > g-) > > Samuel Osorio Calvo wrote: >> I guess a sequential forking ordered with the q value of the contact >> header field value would do what you are asking for. The user has >> just to configure the appropriate value in his/her UA to the >> preferred ringing sequence. This method is fully SIP compliant but >> what I am not sure is if SER does sequential forking in the q order >> (it wasn't a few months ago but some modifications might have added >> it...someone can tell you better than me). >> If you still want users to modify SER's database with a web >> interface, you might try to modify the q value in the usrloc >> database, of course if SER does q-ordered sequential forking, but I >> just think it adds lots of complexity to a feature which SER is >> supposed to provide together with minum user configuration in the >> endpoints. >> >> Hope it helps, >> >> Samuel. >> >> >> Unclassified. >>>>> "Aisling" <ashling.odriscoll(a)cit.ie> 06/21/05 12:54PM >>> >> Hello all, >> >> I'm hoping someone will offer an opinion as to how I should approach >> the >> following and if I am thinking along the correct lines: >> >> I am creating a web application where a user can dictate which device >> they want a call delivered to. So if I have a user with sip url >> "sip:2000@server" and they have registered with the SER server from a >> pda, pc and laptop, SER will currently parallel fork a call to all >> these >> destinations causing all softphones to ring (correct?). However I >> want a >> user to choose from a drop down menu in their browser (which I'm >> developing with servlets and JSP's) their preferred phone e.g. pda >> and then SER will fork the call to the softphone on the pda first. >> >> So basically my current plan is to retrieve the information from the >> user about their preferred phone (which will be associated with a >> particular IP address) and then dynamically modify the ser.cfg for a >> sequential forking rule for that user. I would appreciate opinions as >> to >> whether I am approaching this correctly or if there's an alternative >> method for such functionality (perhaps such as forking the call to >> the device which last registered or something)? >> >> Many Thanks, >> Aisling. >> >> >> >> >> -------------------Legal >> Disclaimer--------------------------------------- >> >> The above electronic mail transmission is confidential and intended >> only for the person to whom it is addressed. Its contents may be >> protected by legal and/or professional privilege. Should it be >> received by you in error please contact the sender at the above >> quoted email address. Any unauthorised form of reproduction of this >> message is strictly prohibited. The Institute does not guarantee the >> security of any information electronically transmitted and is not >> liable if the information contained in this communication is not a >> proper and complete record of the message as transmitted by the >> sender nor for any delay in its receipt. >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers > > > -------------------Legal > Disclaimer--------------------------------------- > > The above electronic mail transmission is confidential and intended > only > for the person to whom it is addressed. Its contents may be protected > by > legal and/or professional privilege. Should it be received by you in > error please contact the sender at the above quoted email address. Any > unauthorised form of reproduction of this message is strictly > prohibited. The Institute does not guarantee the security of any > information electronically transmitted and is not liable if the > information contained in this communication is not a proper and > complete > record of the message as transmitted by the sender nor for any delay > in > its receipt. > > > -------------------Legal > Disclaimer--------------------------------------- > > The above electronic mail transmission is confidential and intended > only for the person to whom it is addressed. Its contents may be > protected by legal and/or professional privilege. Should it be > received by you in error please contact the sender at the above > quoted email address. Any unauthorised form of reproduction of this > message is strictly prohibited. The Institute does not guarantee the > security of any information electronically transmitted and is not > liable if the information contained in this communication is not a > proper and complete record of the message as transmitted by the > sender nor for any delay in its receipt.

1 0

[Users] avpop request and lcr ideas
by Kristin Galway 21 Jun '05

21 Jun '05

Thanks for creating this branch away from the seemingly stagnant SER project. I have in the past seen that requests for help or features have fallen on empty replies as evidenced while searching the mailing lists. I hope that this will be different here. I hope that be able to contribute back to this project once I become more versed in the inner workings as I see a very strong need for the following: Route lookups based on OCN/LERG data and carrier cost tables. (I already have the queries to do the but not the means to implement yet) This function would set say a defined number of flags to indicate outbound carriers based on cost. Having a module to do this would allow LCR lookups with fallover capabilities but based on OCN's and carrier data. Something else I would like to see is a way of using avops to set a fr_inv_timeout value. For instance: if (avp_db_load("$ruri/username", "s:voicemail")) { if (avp_check("s:voicemail", "eq/y/i")) { xlog("L_N", "\033[1m\033[36m ROUTE[0] - s:voicemail = y \033[0m\n"); # recipient has voicemail enabled. set flag(31) setflag(31); # test for a user set timeout if (avp_db_load("$ruri/username", i:vmtimeout")) { avp_write("$vmtimeout", s:callee_fr_inv_timer"); }; }; }; Thanks! Kristin ____________________________________________________ Yahoo! Sports Rekindle the Rivalries. Sign up for Fantasy Football http://football.fantasysports.yahoo.com

2 1

[Serusers] Re: [Serdev] rel_0_9_0 release critical bugs/missing things
by Greger V. Teigre 21 Jun '05

21 Jun '05

As far I can see (using a filter on Matches versions: 0.9.3), there are no issues. The same search, but using HEAD, gives me 22 issues. Jan stated that everything he is aware of has been registered at http://bugs.sip-router.org/. I would like to make a call for submissions on http://bugs.sip-router.org/ of lacking documentation that you either have identified and started on (ref. earlier calls for documentation) or just identified (so that others may pick up the documentation effort). Let's try to make 0.9.x documentation as complete as possible now leading up to the release!! g-) Klaus Darilion wrote: > maybe on the bugtracker? > > Juha Heinanen wrote: >> is there somewhere a list of rel_0_9_0 release critical bugs/missing >> things? it would be high time to release the next stable version >> and if the lists exists, perhaps people could help to fix the loose >> ends. -- juha >> >> _______________________________________________ >> Serdev mailing list >> serdev(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serdev >> >> > > _______________________________________________ > Serdev mailing list > serdev(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serdev

6 12

Re: [Serusers] Portal for forking call to preferred end device -sequential ringing
by Samuel Osorio Calvo 21 Jun '05

21 Jun '05

I guess a sequential forking ordered with the q value of the contact header field value would do what you are asking for. The user has just to configure the appropriate value in his/her UA to the preferred ringing sequence. This method is fully SIP compliant but what I am not sure is if SER does sequential forking in the q order (it wasn't a few months ago but some modifications might have added it...someone can tell you better than me). If you still want users to modify SER's database with a web interface, you might try to modify the q value in the usrloc database, of course if SER does q-ordered sequential forking, but I just think it adds lots of complexity to a feature which SER is supposed to provide together with minum user configuration in the endpoints. Hope it helps, Samuel. Unclassified. >>> "Aisling" <ashling.odriscoll(a)cit.ie> 06/21/05 12:54PM >>> Hello all, I'm hoping someone will offer an opinion as to how I should approach the following and if I am thinking along the correct lines: I am creating a web application where a user can dictate which device they want a call delivered to. So if I have a user with sip url "sip:2000@server" and they have registered with the SER server from a pda, pc and laptop, SER will currently parallel fork a call to all these destinations causing all softphones to ring (correct?). However I want a user to choose from a drop down menu in their browser (which I'm developing with servlets and JSP's) their preferred phone e.g. pda and then SER will fork the call to the softphone on the pda first. So basically my current plan is to retrieve the information from the user about their preferred phone (which will be associated with a particular IP address) and then dynamically modify the ser.cfg for a sequential forking rule for that user. I would appreciate opinions as to whether I am approaching this correctly or if there's an alternative method for such functionality (perhaps such as forking the call to the device which last registered or something)? Many Thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

4 4

[Serusers] iptel is hiring
by Jiri Kuthan 21 Jun '05

21 Jun '05

iptel is hiring developers who are familiar with programming in C, C++, and understand TCP/IP, SIP and ideally SER+SEMS. iptel is present in two locations: Berlin, Germany and Prague, Czech Republic. In some cases, we may consider remote work as well. Independent of the location, we are having nice offices with scenic views over the city and rooms crowded with seazoned SER developers :) I will be happy to take inquiries and applications off-line. Please include CV with work experience. Thank you! -jiri PS -- I am continously travelling, so don't be surprised if I cannot respond in real time. -- Jiri Kuthan http://iptel.org/~jiri/

1 0

[serusers] pa module
by michael p 21 Jun '05

21 Jun '05

hi all i have some problems to include the pa module in SER. in the README it's a very quick config. i don't understand where exactly i hvae to put the function "handle_subscription("registrar");" ? i don't understand how it works tha pa module to create the tables for pa module ? (i don't find the script modules/pa/pa.sql ---- how can i find it and how i have to run it? ) thanks for the help Mike

1 0

Re: [Serusers] Is there an AVP to alter "expires=" in REGISTERmessages?
by Samuel Osorio Calvo 21 Jun '05

21 Jun '05

I totally agree on the solution, luck with the vendor.... Samuel. Unclassified. >>> Java Rockx <javarockx(a)gmail.com> 06/21/05 01:40PM >>> Thanks for the feedback all. I've been considering what to do with this issue, and I decided that it is better to (but more painful) to have the vendor fix their firmware than it is for me to alter SER's functionality in a way as to move it away from RFC compliance. The save_extended_expires suggestion would be an easy fix, but the right thing to do is fix the problem rather than the symptom. Regards, Paul On 6/21/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > > I was taking a more detailled look at the code and it seems that the > Contact header will not be added into the reply if you use > save_noreply+sl_send_reply because the Contact buffer is added inside the > registrar module....but I recall that it worked. I suggest you to give it a > try and see the answer with ethereal|ngrep|* maybe it is taken after when > constructing the answer from the sipmsg. > Anyhow it would be easy to add a save_extended_expires command that makes > what you need but, as I said, people can start using these SIP patches to > build even more non-compliant proxies with SER.... > > Samuel. > > > Unclassified. > >>> "Greger V. Teigre" <greger(a)teigre.com> 06/21/05 07:19AM >>> > You're right of course :-) > Samuel's approach is more correct, but I'm not sure you will be able to > generate the correct reply using save_noreply and sl_reply (with Contact) ?! > Does save_noreply generate the reply? > g-) > > ---- Original Message ---- > From: Java Rockx > To: Greger V. Teigre > Cc: serusers > Sent: Monday, June 20, 2005 02:21 PM > Subject: Re: [Serusers] Is there an AVP to alter "expires=" in > REGISTERmessages? > > > Greger, > > > > That would be possible, but the way SER generates the 200OK reply, it > > would contain the altered "expires=" value. By doing so, the ATA > > would take that new value and use it, so if the ATA sent the REGISTER > > with an expires=300 and SER replied with a 200OK and a contact header > > containing expires=360, the ATA would not send another REGISTER for > > 360 seconds. > > > > I was fooling around with sip_msg.c and reply.c were I padded the > > expires with 60 seconds before saving to usrloc and then removing the > > pad before sending the reply and it works, but I would just feel > > better by not altering the source code. > > > > I guess I'd just feel more comfortable if SER had some sort of AVP or > > modparam for the registrar module that did this automagically. > > > > Regards, > > Paul > > > > > > On 6/20/05, Greger V. Teigre <greger(a)teigre.com> wrote: > > It was supposed to be short, but maybe too short. I meant doing the > > subst on the REGISTER message before save. > > g-) > > ---- Original Message ---- > > From: Greger V. Teigre > > To: Java Rockx ; serusers > > Sent: Monday, June 20, 2005 09:01 AM > > Subject: Re: [Serusers] Is there an AVP to alter "expires=" in > > REGISTERmessages? > > > >> Hi Paul, > >> Would it be possible to "do it upside down" ? ;-) I.e. do a > >> subst("/^Expires:(.*)[0-9]$/ Expires: yourvalue+60/gi"); > >> g-) > >> > >> ---- Original Message ---- > >> From: Java Rockx > >> To: serusers > >> Sent: Sunday, June 19, 2005 04:28 PM > >> Subject: [Serusers] Is there an AVP to alter "expires=" in REGISTER > >> messages? > >> > >>> Hi All. > >>> > >>> We've got some ATAs that don't quite re-REGISTER in a timely fashion > >>> and therefore SER expires the usrloc record a few seconds before the > >>> ATA sends another REGISTER message. > >>> > >>> Is there an AVP or something to add, for example, 60 seconds to the > >>> "expires=" value in the REGISTER message prior to calling save()? By > >>> doing so, these "broken" ATAs would re-REGISTER before SER removes > >>> the usrloc record from cache. > >>> > >>> Regards, > >>> Paul > >>> > >>> > >>> > >>> > >>> _______________________________________________ > >>> Serusers mailing list > >>> serusers(a)lists.iptel.org > >>> http://lists.iptel.org/mailman/listinfo/serusers > >> > >> > >> > >> _______________________________________________ > >> Serusers mailing list > >> serusers(a)lists.iptel.org > >> http://lists.iptel.org/mailman/listinfo/serusers > >

1 0

[Users] Re: no solution yet :( that's why i've decided to write to you
by Daniel-Constantin Mierla 21 Jun '05

21 Jun '05

You have to create a dialing plan for openser also. In your script, after the block: if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; (Just four your information, you don't need "if (uri==myself)" because will be true all the time) Back to dialing plan, you have to add some rules like: if(uri=~"^sip:05[0-9]+@") { # call to mobile rewritehost("192.168.1.35 <mailto:20105353490056@192.168.1.35%29>"); prefix("201"); route(1); break; } Daniel On 06/20/05 11:00, Ozan Blotter wrote: > Dear Team, > > I know this is annoying to post nearly same things but still no answer > :( Is it really so hard or can anybody tell me why this is not working. > > > Basically the system is as : > > ATA 1 : 12345 (registered and can call 54321, no problem) This is > Zyxel P2002. > ATA 2 : 54321 (registered and can call 12345, no problem) This is > Zyxel P2002 too. > ANALOG FXO : 201 (registered as 201, for test purposes i only > registered one port of it) This is Eusso 7104-S 4 Port Analog FXO and > i can add 3 more ports as available. > > With Brekeke's Ondo SIP Proxy, i created a simple dial-plan and it > worked, even i can add more prefixes no problem. But in SER i cannot > :( I'm using OpenSER. Normally it's like when i want to call a GSM, > proxy simply adds a prefix 201 which is same with registered sip > number for regarding analog port and forwards it (looks like > sip:20105353490056@192.168.1.35) > <mailto:20105353490056@192.168.1.35%29> to Gateway and Gateway simply > strips 201 from it and dials 05353490056, yes it connects. But what > this same does not work on SER, any ideas because it's very boring to > play all day and night ? > > Thanks.... > > This is my current SER.CFG as it comes with OPENSER. Can you please > help me to add lines where or/and add remove somethings... > > [root@localhost openser]# cat ser.cfg > # > # $Id: openser.cfg,v 1.1.1.1 2005/06/13 16:47:30 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" > > # ------------------ module loading ---------------------------------- > > # Uncomment this if you want to use SQL database > #loadmodule "/usr/local/lib/ser/modules/mysql.so" > > loadmodule "/usr/lib/openser/modules/sl.so" > loadmodule "/usr/lib/openser/modules/tm.so" > loadmodule "/usr/lib/openser/modules/rr.so" > loadmodule "/usr/lib/openser/modules/maxfwd.so" > loadmodule "/usr/lib/openser/modules/usrloc.so" > loadmodule "/usr/lib/openser/modules/registrar.so" > loadmodule "/usr/lib/openser/modules/textops.so" > > # Uncomment this if you want digest authentication > # mysql.so must be loaded ! > #loadmodule "/usr/lib/openser/modules/auth.so" > #loadmodule "/usr/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"); > break; > }; > if (msg:len >= 2048 ) { > 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 > 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); > break; > }; > > if (!uri==myself) { > # mark routing logic in request > append_hf("P-hint: outbound\r\n"); > route(1); > 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("openser.org", "subscriber")) { > # www_challenge("openser.org", "0"); > # break; > # }; > > save("location"); > break; > }; > > lookup("aliases"); > if (!uri==myself) { > append_hf("P-hint: outbound alias\r\n"); > route(1); > break; > }; > > # native SIP destinations are handled using our USRLOC DB > if (!lookup("location")) { > sl_send_reply("404", "Not Found"); > break; > }; > }; > 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(); > }; > } > > [root@localhost openser]# > >------------------------------------------------------------------------ > >_______________________________________________ >Team mailing list >Team(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/team > >

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Re: [Serusers] Is there an AVP to alter "expires=" in REGISTERmessages?
by Java Rockx 21 Jun '05

21 Jun '05

Thanks for the feedback all. I've been considering what to do with this issue, and I decided that it is better to (but more painful) to have the vendor fix their firmware than it is for me to alter SER's functionality in a way as to move it away from RFC compliance. The save_extended_expires suggestion would be an easy fix, but the right thing to do is fix the problem rather than the symptom. Regards, Paul On 6/21/05, Samuel Osorio Calvo <samuel.osorio(a)nl.thalesgroup.com> wrote: > > I was taking a more detailled look at the code and it seems that the > Contact header will not be added into the reply if you use > save_noreply+sl_send_reply because the Contact buffer is added inside the > registrar module....but I recall that it worked. I suggest you to give it a > try and see the answer with ethereal|ngrep|* maybe it is taken after when > constructing the answer from the sipmsg. > Anyhow it would be easy to add a save_extended_expires command that makes > what you need but, as I said, people can start using these SIP patches to > build even more non-compliant proxies with SER.... > > Samuel. > > > Unclassified. > >>> "Greger V. Teigre" <greger(a)teigre.com> 06/21/05 07:19AM >>> > You're right of course :-) > Samuel's approach is more correct, but I'm not sure you will be able to > generate the correct reply using save_noreply and sl_reply (with Contact) ?! > Does save_noreply generate the reply? > g-) > > ---- Original Message ---- > From: Java Rockx > To: Greger V. Teigre > Cc: serusers > Sent: Monday, June 20, 2005 02:21 PM > Subject: Re: [Serusers] Is there an AVP to alter "expires=" in > REGISTERmessages? > > > Greger, > > > > That would be possible, but the way SER generates the 200OK reply, it > > would contain the altered "expires=" value. By doing so, the ATA > > would take that new value and use it, so if the ATA sent the REGISTER > > with an expires=300 and SER replied with a 200OK and a contact header > > containing expires=360, the ATA would not send another REGISTER for > > 360 seconds. > > > > I was fooling around with sip_msg.c and reply.c were I padded the > > expires with 60 seconds before saving to usrloc and then removing the > > pad before sending the reply and it works, but I would just feel > > better by not altering the source code. > > > > I guess I'd just feel more comfortable if SER had some sort of AVP or > > modparam for the registrar module that did this automagically. > > > > Regards, > > Paul > > > > > > On 6/20/05, Greger V. Teigre <greger(a)teigre.com> wrote: > > It was supposed to be short, but maybe too short. I meant doing the > > subst on the REGISTER message before save. > > g-) > > ---- Original Message ---- > > From: Greger V. Teigre > > To: Java Rockx ; serusers > > Sent: Monday, June 20, 2005 09:01 AM > > Subject: Re: [Serusers] Is there an AVP to alter "expires=" in > > REGISTERmessages? > > > >> Hi Paul, > >> Would it be possible to "do it upside down" ? ;-) I.e. do a > >> subst("/^Expires:(.*)[0-9]$/ Expires: yourvalue+60/gi"); > >> g-) > >> > >> ---- Original Message ---- > >> From: Java Rockx > >> To: serusers > >> Sent: Sunday, June 19, 2005 04:28 PM > >> Subject: [Serusers] Is there an AVP to alter "expires=" in REGISTER > >> messages? > >> > >>> Hi All. > >>> > >>> We've got some ATAs that don't quite re-REGISTER in a timely fashion > >>> and therefore SER expires the usrloc record a few seconds before the > >>> ATA sends another REGISTER message. > >>> > >>> Is there an AVP or something to add, for example, 60 seconds to the > >>> "expires=" value in the REGISTER message prior to calling save()? By > >>> doing so, these "broken" ATAs would re-REGISTER before SER removes > >>> the usrloc record from cache. > >>> > >>> Regards, > >>> Paul > >>> > >>> > >>> > >>> > >>> _______________________________________________ > >>> Serusers mailing list > >>> serusers(a)lists.iptel.org > >>> http://lists.iptel.org/mailman/listinfo/serusers > >> > >> > >> > >> _______________________________________________ > >> Serusers mailing list > >> serusers(a)lists.iptel.org > >> http://lists.iptel.org/mailman/listinfo/serusers > >

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[Serusers] new voicamail.so
by Llanos Serna García-Conde 21 Jun '05

21 Jun '05

1 0

[Users] OpenSER Web Site hosting
by Daniel-Constantin Mierla 21 Jun '05

21 Jun '05

Hello everybody, currently the openser.org web site is hosted by Voice-System server. It has to be moved on a separate hosting server, so that developers and contributors can have full access to it. Are you aware of a good hosting service? It has to be a Linux/Unix server, allow ssh access, provide a good bandwidth and the price to be as much as possible. I have opened a topic in openser forum, too, -- you can use what you like more to reply. Daniel

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[Serusers] Portal for forking call to preferred end device - sequential ringing
by Aisling 21 Jun '05

21 Jun '05

Hello all, I'm hoping someone will offer an opinion as to how I should approach the following and if I am thinking along the correct lines: I am creating a web application where a user can dictate which device they want a call delivered to. So if I have a user with sip url "sip:2000@server" and they have registered with the SER server from a pda, pc and laptop, SER will currently parallel fork a call to all these destinations causing all softphones to ring (correct?). However I want a user to choose from a drop down menu in their browser (which I'm developing with servlets and JSP's) their preferred phone e.g. pda and then SER will fork the call to the softphone on the pda first. So basically my current plan is to retrieve the information from the user about their preferred phone (which will be associated with a particular IP address) and then dynamically modify the ser.cfg for a sequential forking rule for that user. I would appreciate opinions as to whether I am approaching this correctly or if there's an alternative method for such functionality (perhaps such as forking the call to the device which last registered or something)? Many Thanks, Aisling. -------------------Legal Disclaimer--------------------------------------- The above electronic mail transmission is confidential and intended only for the person to whom it is addressed. Its contents may be protected by legal and/or professional privilege. Should it be received by you in error please contact the sender at the above quoted email address. Any unauthorised form of reproduction of this message is strictly prohibited. The Institute does not guarantee the security of any information electronically transmitted and is not liable if the information contained in this communication is not a proper and complete record of the message as transmitted by the sender nor for any delay in its receipt.

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