- sr-users - kamailio.org

[Users] How to check Displayname in From Tag
by martin＠campus-merseburg.de 02 Feb '06

02 Feb '06

Hi, i want to change the "From:" tag with uac_replace_from when the displayname is "anonymous". with from_uri i can only check the uri. is there a way to check the displayname in the "From: " tag? for example: From: "Anonymous <sip:1234@blah.net>" i want to change it to: From: "Anonymous <sip:Anonymous@Anonymous>" Ciao

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[Serusers] Serweb
by Ismael 02 Feb '06

02 Feb '06

hello, i have followed serveb installation instructions, and know i am on the last step : "Open the file http://<your-host>/<your-install-dir>/admin/index.php" But i don't understand what i have to put on "your-host" ? Thanks, Ismaël

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[Serusers] Install and configure SER
by Ismael 02 Feb '06

02 Feb '06

Hello, I want to install SER on my debian. Could you explain me step by step how to install and configure ser server on my computer ?? How attach mysql to SER server ? Thanks to advance, Ismaël

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[Serusers] Need help to install SER and Serweb
by Ismael 02 Feb '06

02 Feb '06

Hello, I installed Debian on my computer, and now i want to install ser proxy ans serweb. To install SER, i did : // installation of apache, mysql, and phpmyadmin apt-get install apache mysql-server phpmyadmin // installation of SER dpkg -i ser_0.9.6-0.1_i386.deb // restart of SER service /etc/init.d/ser restart To install serweb, i just did : tar -xvzf serweb-0.9.4.tar.gz What do i have to do now ?? Thanks for your help, Ismaël

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[Serusers] TCP sockets and NATs
by Teemu Harju 02 Feb '06

02 Feb '06

Hi, This is a question that has been bugging me for some time now. If I use SER and the clients connected to it use only TCP, will there be a problem with the number of sockets that can be open at the same time, if these clients are behind NAT? I mean that for NAT bindings to stay open, each client has to keep open TCP connection to SER all the time after registration. What is the maximum number of open sockets that Linux can handle at the same time? I've changed the maximum number of open files to 65535 (ulimit -n 65535). Doesn't this also mean the maximum number of sockets? Is this the absolute maximum? Also, does this mean the maximum number of clients that can connect to my proxy? Regards, Teemu -- Teemu Harju http://www.teemuharju.net

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[Serusers] SER RTP QoS Monitoring
by Aisling O'Driscoll 02 Feb '06

02 Feb '06

Hi, I want to be able to extract QoS parameters such as jitter, delay and packet loss from VoIP calls made using my SER(both peer to peer for my public clients and also for clients where the voice will be routed with mediaproxy). I would like to develop something as opposed to purchasing a software package. >From searching the Internet the last few days I have seen that the following options seem to be available: 1) RTCP RTCP stream will contain this information. The problem is that sniffers cannot clearly distinguish RTCP packets from other UDP packets unless something like an IP address or specific UDP port is supplied. I had considered using TCPDUMP but then where do I position this in the network when voice is going peer to peer between client and I'm not sure how this can work to monitor all calls when I might not be aware of the end user IP addresses. 2) SIP INFO extension This also seem to be an option but most off the shelf phones don't seem to support this and this would required modification of a SIP ua. 3) SNMP I thought maybe SNMP might be an option but the SER snmp module no longer exists... Does anyone have any comments on the above? Are the statements that I made correct or can anyone think of other ways to monitor the voice QoS? I am trying to understand how commercial applications have accomplished this. Many thanks in advance for advice. Aisling. >---- Original Message ---- >From: clona(a)cyberhouse.no >To: ashling.odriscoll(a)cit.ie >Subject: Re: [Serusers] Status of SER SNMP module >Date: Sun, 29 Jan 2006 20:30:02 +0100 > >> >>Hi Aisling, >> >>The SNMP module is dead as a killed turtle :( >> >>-Atle >>* Aisling O'Driscoll <ashling.odriscoll(a)cit.ie> [060129 20:26]: >>> Hello, >>> >>> I'm just wondering what the current status of the SER SNMP module >is? >>> Is it currently supported and is there somewhere I can find >>> documentation on it? >>> >>> 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)iptel.org >>> http://mail.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.

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[Serusers] high-availibility setup using f5 bigip
by Jack Wei 02 Feb '06

02 Feb '06

hi, I'm trying to set up 2 SER and 2 Asterisks boxes using a bigip switch to do load-balancing. I'm using Asterisk as a voicemail application only and have successfully integrated SER with Asterisk without the switch. But when I try to use the switch as a load-balancer, I get lots of NAT problems. Does anyone know how to setup the switch and SER/Asterisk properly? Thanks, Jack __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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RE: [Serusers] vm.so
by Toygun Mavinil 01 Feb '06

01 Feb '06

Then where do i have to insert the codes in ser.cfg, the timeout for forward to voicemail? I tried some places but not successfull My ser.cfg is from onsip ser mediaproxy examle; Here, # $Id: nat-mediaproxy.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ debug=3 fork=yes log_stderror=no listen=192.0.2.13 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.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" modparam("auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.0.2.13:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } Toygun >Read the more recent SEMS documentation. vm.so was deprecated. >You now use other methods (depending on whether you're using a fifo or a unix socket in SEMS). >For instance, I use: >if(!t_write_unix("/tmp/am_sock","voicemail")) { > t_reply("500","error contacting sems"); >}; On Wed, 1 Feb 2006 20:44:36 +0200, Toygun Mavinil wrote > Hi, > I installed ser 0.9.6 on fc4 > Compiled and installed with all modules but vm.so module is missing. > Looked at the source modules directory, there is no vm directory or file. > I installed sems from cvs > Sems is working > > Also > Do i have to add down lines to ser.cfg > > loadmodule "/home/srouter/sip_router/modules/vm/vm.so" > modparam("vm", "auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") > modparam("vm", "email_column", "email") > modparam("vm", "subscriber_table", "subscriber") > modparam("vm", "user_column", "username") > modparam("vm", "domain_column", "domain") > > if(method=="ACK" || method=="INVITE" || method=="BYE"){ > > if (!t_newtran()) { > log("could not create new transaction\n"); > sl_send_reply("500","could not create new transaction"); > break; > }; > > t_reply("100","Trying -- just wait a minute !"); > if(method=="INVITE"){ > log("**************** vm start - begin ******************\n"); > if(!vm("/tmp/am_fifo","voicemail")){ > log("could not contact the answer machine\n"); > t_reply("500","could not contact the answer machine"); > }; > log("**************** vm start - end ******************\n"); > } else if(method=="BYE"){ > log("**************** vm end - begin ******************\n"); > if(!vm("/tmp/am_fifo","bye")){ > log("could not contact the answer machine\n"); > t_reply("500","could not contact the answer machine"); > }; > log("**************** vm end - end ******************\n"); > }; > break; > }; > if (method=="CANCEL") { > sl_send_reply("200", "cancels are junked here"); > break; > }; > sl_send_reply("501", "method not understood here"); > > > > > > > Toygun >

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FW: [Serusers] vm.so
by Toygun Mavinil 01 Feb '06

01 Feb '06

Then where do i have to insert the codes in ser.cfg, the timeout for forward to voicemail? I tried some places but not successfull My ser.cfg is from onsip ser mediaproxy examle; Here, # $Id: nat-mediaproxy.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ debug=3 fork=yes log_stderror=no listen=192.0.2.13 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.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" modparam("auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.0.2.13:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } Toygun >Read the more recent SEMS documentation. vm.so was deprecated. >You now use other methods (depending on whether you're using a fifo or a unix socket in SEMS). >For instance, I use: >if(!t_write_unix("/tmp/am_sock","voicemail")) { > t_reply("500","error contacting sems"); >}; On Wed, 1 Feb 2006 20:44:36 +0200, Toygun Mavinil wrote > Hi, > I installed ser 0.9.6 on fc4 > Compiled and installed with all modules but vm.so module is missing. > Looked at the source modules directory, there is no vm directory or file. > I installed sems from cvs > Sems is working > > Also > Do i have to add down lines to ser.cfg > > loadmodule "/home/srouter/sip_router/modules/vm/vm.so" > modparam("vm", "auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") > modparam("vm", "email_column", "email") > modparam("vm", "subscriber_table", "subscriber") > modparam("vm", "user_column", "username") > modparam("vm", "domain_column", "domain") > > if(method=="ACK" || method=="INVITE" || method=="BYE"){ > > if (!t_newtran()) { > log("could not create new transaction\n"); > sl_send_reply("500","could not create new transaction"); > break; > }; > > t_reply("100","Trying -- just wait a minute !"); > if(method=="INVITE"){ > log("**************** vm start - begin ******************\n"); > if(!vm("/tmp/am_fifo","voicemail")){ > log("could not contact the answer machine\n"); > t_reply("500","could not contact the answer machine"); > }; > log("**************** vm start - end ******************\n"); > } else if(method=="BYE"){ > log("**************** vm end - begin ******************\n"); > if(!vm("/tmp/am_fifo","bye")){ > log("could not contact the answer machine\n"); > t_reply("500","could not contact the answer machine"); > }; > log("**************** vm end - end ******************\n"); > }; > break; > }; > if (method=="CANCEL") { > sl_send_reply("200", "cancels are junked here"); > break; > }; > sl_send_reply("501", "method not understood here"); > > > > > > > Toygun >

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[Serusers] vm.so
by Toygun Mavinil 01 Feb '06

01 Feb '06

Hi, I installed ser 0.9.6 on fc4 Compiled and installed with all modules but vm.so module is missing. Looked at the source modules directory, there is no vm directory or file. I installed sems from cvs Sems is working Also Do i have to add down lines to ser.cfg loadmodule "/home/srouter/sip_router/modules/vm/vm.so" modparam("vm", "auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("vm", "email_column", "email") modparam("vm", "subscriber_table", "subscriber") modparam("vm", "user_column", "username") modparam("vm", "domain_column", "domain") if(method=="ACK" || method=="INVITE" || method=="BYE"){ if (!t_newtran()) { log("could not create new transaction\n"); sl_send_reply("500","could not create new transaction"); break; }; t_reply("100","Trying -- just wait a minute !"); if(method=="INVITE"){ log("**************** vm start - begin ******************\n"); if(!vm("/tmp/am_fifo","voicemail")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; log("**************** vm start - end ******************\n"); } else if(method=="BYE"){ log("**************** vm end - begin ******************\n"); if(!vm("/tmp/am_fifo","bye")){ log("could not contact the answer machine\n"); t_reply("500","could not contact the answer machine"); }; log("**************** vm end - end ******************\n"); }; break; }; if (method=="CANCEL") { sl_send_reply("200", "cancels are junked here"); break; }; sl_send_reply("501", "method not understood here"); Toygun

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[Users] Forward requests based on userid
by Mouncif Benziane 01 Feb '06

01 Feb '06

Hi folks, I wanted to route sip requests based on user id with it's appropriate Gateway IP address which is stored in Mysql database , is this possible? Database structure: userid,gateway 617978XXX 64.80.XXX.XXX 78189XXXX 64.100.XXX.XXXX example: if someone call 617978XXX he will be routed to it's approriate gateway at present I am using static route like: if (uri=~"^sip:617978XXX@") { rewritehost("64.80.XXX.XXX"); } if (uri=~"^sip:78189XXXX@") { rewritehost("664.100.XXX.XXXX"); } etc,.... I am afraid the config file gonna get larger if I add more than 500 users(DIDs), I am looking for an alternative to store DID numbers in database (Mysql or text) and route them accordingly with the appropriate Gateway address. Thank you so much and I will appreciate if you can help me out! Ben __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Users] check if transaction exists
by Andreas M. 01 Feb '06

01 Feb '06

Hi all, i want to introduce following in my configuration: CallerA calls CallerB -> ringing CallerC calls CallerB -> reject during this transaction, without that CallerB picks up the phone, CallerC should not be able to call CallerB during this transaction, After this transactions, CallerC can call CallerB also. Any ideas how to check if there is an ongoing transaction ? regards, Andreas M.

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[Serusers] Re: 200 OK/SDP and ACK/SDP problem
by Carla Ares 01 Feb '06

01 Feb '06

> > Hi all, > I have a problem with audio, when the destination of a call is natted and > the other side is public. This problem is cause because the SDP info only is > present in the 200 OK and in the ACK message. > > I'm using ser (8.14) with mediaproxy (1.2.1). > > I have an H323 to SIP translator (and SIP to H323 too) connected to the > PSTN and to the SIP Proxy (SER). > H323 is the origin of the call (public client), and SIP UAC is the > destination (Natted client). > My proxy is working in state full mode and has a public IP. > > The H323 side is using slow start, so when it starts the transaction with > an INVITE, it will not send the SDP info (it will send it in the ACK > message). > The proxy forwards this message to the destination. > SIP UAC, starts ringing. When we hooks off, the SIP UAC send an 200 OK > with SDP message to the H323 side. > > This is the 200 OK / SDP that the proxy receives from the SIP side. > > No. Time Source Destination Protocol > Info > 488 28.488811 200.68.89.2 200.68.89.12 SIP/SDP > Status: 200 OK, with session description > Message body > Session Description Protocol > Session Description Protocol Version (v): 0 > Owner/Creator, Session Id (o): 700600 6015 6015 IN IP4 > 192.168.0.101 > Owner Username: 700600 > Session ID: 6015 > Session Version: 6015 > Owner Network Type: IN > Owner Address Type: IP4 > Owner Address: 192.168.0.101 > Session Name (s): AddPac Gateway SDP > Connection Information (c): IN IP4 192.168.0.101 > Connection Network Type: IN > Connection Address Type: IP4 > Connection Address: 192.168.0.101 > Time Description, active time (t): 0 0 > Session Start Time: 0 > Session Stop Time: 0 > Media Description, name and address (m): audio 23018 RTP/AVP > 18 8 0 101 > Media Type: audio > Media Port: 23018 > Media Proto: RTP/AVP > Media Format: ITU-T G.729 > Media Format: ITU-T G.711 PCMA > Media Format: ITU-T G.711 PCMU > Media Format: 101 > > Following the ser.cfg, this 200 OK / SDP will make the proxy to use > mediaproxy (use_media_proxy()), because the destination user is natted. So > the mediaproxy module, will generate a lookup command to the > proxydispatcher.py. > > proxydispatcher[30535]: command lookup 2281401749(a)200.68.89.10 > 192.168.0.101:23018:audio 200.68.89.2 200.68.89.10 remote 200.68.89.10unknown AddPac=20SIP=20Gateway info= > from:1150316660@200.68.89.10,to:1152464490@200.68.89.10 > ,fromtag:3512844671,totag:7b07717a4 > proxydispatcher[30535]: warning: trying to lookup session with > non-existent id: '2281401749(a)200.68.89.10' > > The proxydispatcher does not recognize that command because it has not > generate the session before. So, the proxy forwards the 200 OK /SDP message > without changing the SDP info. When the H323 sides receives that info, it > thinks that it has to send RTP to the private IP. > > Then the H323 sends the ACK with SDP. > > Then, we cannot ear audio in both sides. > > Here is the problem, because, I didn't receive an INVITE with SDP that can > create the session into the dispatcher. > > > A is public, B is natted > > side A Proxy side B > > INVITE without SDP > ----------> > 100 Trying > <---------- > > INVITE without SDP > ----------> > > 100 Trying > <---------- > 180 Ringing > <---------- > 180 Ringing > <---------- > > 200 OK SDP (B private IP) > <---------- > *1* > 200 OK SDP (B private IP) > <---------- > > ACK SDP (A public IP) > ----------> > *2* > ACK SDP (mediaproxy public IP) > ----------> > > ...................................................... > RTP from B > <---------------------- > *3* > ...................................................... > > > *1* The proxy must replace the private IP of the SDP, with the ip of the > mediaproxy. The looku prequest does not work, because no session was found > *2* the proxy generates the session in the mediaproxy. So B thinks that > the RTP must go to the mediaproxy. But A never knows the mediaproxy address. > *3* RTP from A goes to anywhere, because A does not know where it is the > private adrress of B. > > Do you know a solution to this? Is the sequence of the message all right? Regards. Carla

2 5

[Serusers] Here I go again...Call type processing Section problem?
by Juan 01 Feb '06

01 Feb '06

Hi. As I had no answers to this question, here it is again. Till now I was happy using SER handling communications between UAs using nathelper and RPT proxy. Now I changed the ser.cfg script to the one with PSTN support and none of the UAs get registered. I am getting a 403 Forbidden. Analyzing the script I understood that this 403 was coming from the Call Type Processing Section so I changed if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { and so on to if (uri!=myself) and now the UAs are registering again. But of course I can't make calls. So I would like to know how this< is_uri_host_local and is_from _local> work and what should I do to make them allow my UAs to register. I am attaching the ser.cfg that doesn't work. Thanks in advance Juan Ferrari

2 2

[Users] TM bug?
by Greg Fausak 01 Feb '06

01 Feb '06

I am running a proxy pushing 2 million calls a day. Every now and then, I drop 5 or 10 BYE messages. I've attached the sip_scenario log file, F16,F17,F20 all send BYE messages to 12.46.104.62. F21 is sending the BYE form 12.46.104.62 to 12.46.104.252. The OK comes back, but is never relayed. BYEs continue to retry until 12.46.104.62 finally gives up and sends a Request timeout 408 in frame F38. So, F38 is evidence that the transaction started. Right? Why are all of the BYEs ignored, and why isn't the OK relayed? I have xlog output from 12.46.104.58 indicating each time a BYE is loose-routed to 12.46.104.62. The syslog file shows each attempt (F15,F18,F19,F23, etc). I also have xlog output from 12.46.104.62, however, the syslog on that machine never shows receiving or loose-routing a BYE with this callid. It seems like receiving more than one BYE in a short period of time is screwing up the t_relay()??? The calls are being generated with sipp, and 999,990 times out of a million it works. ---greg -- Greg Fausak greg(a)thursday.com

2 2

[Serusers] [VOIPSEC] any information regarding adding DTLS using OpenSSL
by Pjothi 01 Feb '06

01 Feb '06

Dear all, I am a student and am trying to setup a demonstration with TLS and DTLS support between a SIP client and a Proxy. Has anyone some information regarding adding DTLS support for a SIP client/Proxy or a more generic one using OpenSSL. I just need it for demonstration purposes and so error handling's if any need not be complete. Just a simple support and a demonstration of both would do fine. If anyone has any information regarding this, may be a how to start or any introductory article would be of great help, kindly let me know. This would be of great help. Any suggestions regarding the same are most welcome. kindly let me know and thank you very much. regards, Pjothi _______________________________________________ Voipsec mailing list Voipsec(a)voipsa.org http://voipsa.org/mailman/listinfo/voipsec_voipsa.org --9B095B5ADSN=_01C62670194DF0AE000000EForinoco-- ______________________________________________________________________ This email has been scanned by the MessageLabs Email Security System. For more information please visit http://www.messagelabs.com/email ______________________________________________________________________

1 0

Re: [Serusers] failover and VRRP
by hgaillac-sip＠yahoo.fr 01 Feb '06

01 Feb '06

> I use NAGIOS to > monitor SER to know when a server of applications on > that server have ... Can you tell me more ? Harry --- Steve Blair <blairs(a)isc.upenn.edu> a écrit : > > > hgaillac-sip(a)yahoo.fr wrote: > > >Hello, > > > >Ok SRV records point to proxy sip server but if > this > >one is down before or during connection !!! > > > > > > > During a connection doesn't matter in my > environment. I am running SER > in a stateless > mode. Once the call is setup media flows between > endpoints and the proxy > is out of the > connection. I am not using NAT either. > > > SER1 > >DNS servers=== [?] =====sip agents > > SER2 > > > >How [?] can it provide H-A (heartbeat soft) between > >ser boxes. > > > > > > > Neither the SRV or VRRP can provide a heartbeat > between SER boxes. You'd > need an application running on the SER boxes to do > that. I rely upon > t_replicate to update the backup during > registration, I use NAGIOS to > monitor SER to know when a server of applications on > that server have > failed and I use an SRV record defined in each phone > to identify the > proxy. This way the phone will failover on it's own > during call setup. > Of course this doesn't help during a call but as I > said I am running SER > in a stateless mode so the proxy "steps out" once > the call is setup. > > >If one of these sip proxy failed during a call can > we > >redirect the call to the other proxy ? > > > > > I'm not sure how you would accomplish this. > > >Regards > >Harry > > > >--- Steve Blair <blairs(a)isc.upenn.edu> a écrit : > > > > > > > >>Sorry. I don't do diagrams :-) Which piece of this > >>explanation is unclear? > >> > >>-Steve > >> > >>hgaillac-sip(a)yahoo.fr wrote: > >> > >> > >> > >>>Could you please send me a diagram ? > >>> > >>>Harry > >>>--- Steve Blair <blairs(a)isc.upenn.edu> a écrit : > >>> > >>> > >>> > >>> > >>> > >>>>VRRP for layer 3. H-A woud depend upon your > >>>>availability requirements. > >>>>In our case we use an SRV record to identify > >>>>multiple proxy servers each > >>>>with their own priority and weight. Phones > >>>> > >>>> > >>register > >> > >> > >>>>using the SRV name > >>>>as the proxy address. Each proxy uses > t_replicate > >>>> > >>>> > >>to > >> > >> > >>>>replicate > >>>>registration data. So far this works for us. > >>>> > >>>>_Steve > >>>> > >>>>hgaillac-sip(a)yahoo.fr wrote: > >>>> > >>>> > >>>> > >>>> > >>>> > >>>>>Ok so vrrp for layer 2/3 but which solution for > >>>>> > >>>>> > >>H-A > >> > >> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>? > >>>> > >>>> > >>>> > >>>> > >>>>>Harry > >>>>>--- Steve Blair <blairs(a)isc.upenn.edu> a écrit > : > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>> > >>>>>>Harry: > >>>>>> > >>>>>> Assuming you are talking about running VRRP > in > >>>>>>your routers and not > >>>>>>on the Asterisk host itself then the answer is > a > >>>>>>qualified yes. VRRP by > >>>>>>itself won't give you the transparency of an > SRV > >>>>>>record for identifying > >>>>>>your proxy servers. It will allow you to have > >>>>>>automatic failover to > >>>>>>backup gateways should the primary gateway > >>>>>> > >>>>>> > >>to/from > >> > >> > >>>>>>your "SER subnet" fail. > >>>>>> > >>>>>>-Steve > >>>>>> > >>>>>>hgaillac-sip(a)yahoo.fr wrote: > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>>>hello, > >>>>>>> > >>>>>>>Can we use VRRP when a SER is down ? > >>>>>>> > >>>>>>> > >>>>>>> SER1 (master) > >>>>>>>sip agents == || === Asterisk Farm==sip/pstn > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>gateway > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>> > >>>>>>> SER2 (slave) > >>>>>>> > >>>>>>>Regards > >>>>>>>Harry > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>>>> > >>>>___________________________________________________________________________ > >>>> > === message truncated === ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

3 3

[Serusers] SIP message bodies sometimes corrupted
by Jon Burke 01 Feb '06

01 Feb '06

Hello, I'm having trouble with SER (0.9.4) sometimes corrupting the bodies of SIP messages that it forwards. I'm using binary message bodies, with appropriate Accept, Content-Type and Content-Length headers which the proxy server should forward without modification. Sometimes the messages are forwarded correctly with all headers and the message body intact. Other times, one byte has gone missing from the end of the message body, so that the value in the Content-Length header is one more than the actual number of bytes in the message body. The receiving UA then rejects this message because it is malformed. From repeated testing, it looks as though this problem happens whenever the number of bytes in the SIP message (going out from SER) prior to the body is an even number. When there are an odd number of bytes, the message body is always uncorrupted. Anybody have any ideas what's happening here or how to fix it? Jon.

2 1

[Serusers] is SER UDP capable ?
by Pjothi 01 Feb '06

01 Feb '06

Dear all, Is SER proxy udp capabale, can I use an UDP client application to communicate with SER ? thanks and regards, Pjothi

2 1

[Users] pdt module enhanced with multidomain support
by Elena Ramona Modroiu 01 Feb '06

01 Feb '06

Hello all, the pdt module was enhanced to support multidomain translation of the prefixes, i.e. a new R-URI domain is searched by prefix and domain of FromURI. Until now the search was performed by prefix only. ************** This means the database table will have three columns (sdomain, prefix, domain), where sdomain stands for a source domain to which a set of (prefixes,domain) pairs could be assinged. Database creation example: DROP DATABASE IF EXISTS pdt; CREATE DATABASE pdt; USE pdt; -- create table CREATE TABLE pd_multidomain ( sdomain VARCHAR(255) NOT NULL, prefix VARCHAR(32) NOT NULL, domain VARCHAR(255) NOT NULL DEFAULT "", PRIMARY KEY (sdomain, prefix) ); ************** The module exports one function prefix2domain(mode) to be used from script to update the RURI if this is in a PSTN format. For details about the 'mode' parameter please refer to the README file. The module parameter called 'prefix' is available to define which URIs must be translated. Here it is an example how to set it in the script file modparam("pdt", "prefix", "00"). This means only the RURIs in the format 00xxxxx@yyyyy will be updated if predix2domain() is called in the script. By default is set to NULL, and all RURIs will be updated if prefix2domain() is called. An example of translation: -- Input: modparam("pdt", "prefix", "00") Database entries (sdomain - prefix - domain): (siphub.org, 123, alpha.org) (siphub.org, 124, beta.org) (siphub.org, 126, gamma.org) (openser.org, 123, one.org) (openser.org, 124, two.org) (openser.org, 125, three.org) Message: RURI: 00123456789(a)mydomain.com FromURI: alice(a)siphub.org -- Translation via prefix2domain(0) -- Output RURI: 456789(a)alpha.org ************** Three fifo commands are available for adding, deleting and listing the (sdomain, prefix, domain) entries in the database. /* * How to add a new entry - fifo command example * * --- * :pdt_add:[response_file]\n * sdomain\n * prefix\n * domain\n * \n * -- * * - '.' (dot) means NULL value and is not allowed */ /* * How to delete existent entries - fifo command example: * * --- * :pdt_delete:[response_file]\n * sdomain * domain\n * \n * -- * * - the specified (sdomain, *, domain) entry will be deleted * - '.' (dot) means NULL value and is not allowed */ /* * How to see already defined entries - fifo command example: * * --- * :pdt_list:[response_file]\n * sdomain\n * prefix\n * domain\n * \n * -- * * - '.' (dot) means NULL value * - the comparison operation is 'START WITH' -- if domain is 'a' then * all domains starting with 'a' are listed */ In the module directory under ./doc/ there are available example shell scripts for these commands. Any feedback is highly appreciated. Regards, Ramona

1 0

RE: [Serusers] Serweb does not send mail on registration
by Bart J. Smit 01 Feb '06

01 Feb '06

Hi Karel, Fixed! 8-) I did 'yum -y install php-imap' which fixed the imap_8bit() function call. I then got "sorry -- cannot open write fifo" on the registration confirmation page. This file is not writable by my httpd user (user apache on CentOS) so I set the ownership to root.apache. This is reset to root.root when ser starts, so I added 'chown root.apache /tmp/ser_fifo' to the start section of the /etc/init.d/ser script, just before the 'return $RETVAL' line. Serweb now works a treat. Thanks again for your code and your help, Bart... -----Original Message----- From: Karel Kozlik [mailto:karel@iptel.org] Sent: 01 February 2006 10:30 To: Bart J. Smit Cc: SER Users Subject: Re: [Serusers] Serweb does not send mail on registration well, you need the imap extension for php. It should be sufficient to add this line to your php.ini: extension=imap.so Karel Bart J. Smit napsal(a): > Thanks Karel, > > Good thinking; I forgot to check the apache logs. > > The error log gave a vital clue: > > PHP Fatal error: Call to undefined function: imap_8bit() in > /var/www/serweb-0.9.4/html/functions.php on line 310. > > I'll go and check the PHP to see if I need additional PEAR modules or a > patch to the script. > > Bart... > > -----Original Message----- > From: Karel Kozlik [mailto:karel@iptel.org] > Sent: 01 February 2006 07:55 > To: Bart J. Smit > Cc: SER Users > Subject: Re: [Serusers] Serweb does not send mail on registration > > Hello, > is there an error message in PHP log. Or in apache error log? > > Karel > > Bart J. Smit napsal(a): > >>I have serweb-0.9.4 installed on CentOS 4.2 with MySQL 4.1.12-3 and >>ser-0.9.6-1 compiled from the source RPM at >>http://ftp.upjs.sk/pub/users/sal/Fedora/4/voip. When I register a new >>user, the PHP script goes to an empty page without title and the email >>never gets sent to the address entered on the registration form. >> >>The database gets the correct user details inserted into the pending >>table. The sendnotification and greeting fields remain empty. >> >>I have modified the line in config_data_layer.php for the collation >>character set to read: >>$config->data_sql->collation = "latin1_swedish_ci"; >>This got rid of a DB error in serweb which I traced through pear >>logging: >>Illegal mix of collations (latin1_swedish_ci,IMPLICIT) and >>(utf8_general_ci,COERCIBLE) for operation '='] >> >>I have sent a test email with the debugging/mail.php script and it >>arrives fine. I have tested the registration page with local > > recipients, > >>recipients on the LAN and external recipients but the behaviour stays >>the same. >> >>All the pages I have googled on serweb registration problems always >>state that the emails are sent out correctly, so I'm likely missing >>something obvious. >> >>Can anybody help me get the registration to work? >> >>Thanks, >> >>Bart... >> >>_______________________________________________ >>Serusers mailing list >>serusers(a)lists.iptel.org >>http://lists.iptel.org/mailman/listinfo/serusers

1 0

RE: [Serusers] Serweb does not send mail on registration
by Bart J. Smit 01 Feb '06

01 Feb '06

Thanks Karel, Good thinking; I forgot to check the apache logs. The error log gave a vital clue: PHP Fatal error: Call to undefined function: imap_8bit() in /var/www/serweb-0.9.4/html/functions.php on line 310. I'll go and check the PHP to see if I need additional PEAR modules or a patch to the script. Bart... -----Original Message----- From: Karel Kozlik [mailto:karel@iptel.org] Sent: 01 February 2006 07:55 To: Bart J. Smit Cc: SER Users Subject: Re: [Serusers] Serweb does not send mail on registration Hello, is there an error message in PHP log. Or in apache error log? Karel Bart J. Smit napsal(a): > I have serweb-0.9.4 installed on CentOS 4.2 with MySQL 4.1.12-3 and > ser-0.9.6-1 compiled from the source RPM at > http://ftp.upjs.sk/pub/users/sal/Fedora/4/voip. When I register a new > user, the PHP script goes to an empty page without title and the email > never gets sent to the address entered on the registration form. > > The database gets the correct user details inserted into the pending > table. The sendnotification and greeting fields remain empty. > > I have modified the line in config_data_layer.php for the collation > character set to read: > $config->data_sql->collation = "latin1_swedish_ci"; > This got rid of a DB error in serweb which I traced through pear > logging: > Illegal mix of collations (latin1_swedish_ci,IMPLICIT) and > (utf8_general_ci,COERCIBLE) for operation '='] > > I have sent a test email with the debugging/mail.php script and it > arrives fine. I have tested the registration page with local recipients, > recipients on the LAN and external recipients but the behaviour stays > the same. > > All the pages I have googled on serweb registration problems always > state that the emails are sent out correctly, so I'm likely missing > something obvious. > > Can anybody help me get the registration to work? > > Thanks, > > Bart... > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

2 1

Re: [Fwd: [Users] enum patch]
by Greg Fausak 01 Feb '06

01 Feb '06

Juha, I've updated the documentation. I don't know how to produce output, so I don't know if it works. I haven't done a patch before either, so, let me know if this is the preferred format or if you want to see it a different way. By the way, when I was reading the documentation I noticed that the enum query puts the highest priority query in the ruri, then the rest of them are stored in new branches with a q value. I haven't actually tried this. Is it the plan that these branches will be looped through serially upon failure without having a retry route block? The new patch is attached. Thanks, -g On Jan 29, 2006, at 4:27 AM, Bogdan-Andrei Iancu wrote: > To be more precisely, the file is modules/enum/doc/enum_user.sgml > > bogdan > > Juha Heinanen wrote: > >>> is the documentation the README file? >>> >> >> you have to edit the xml file from which README is generated >> automatically. >> >> -- juha >> >> >

4 3

[Users] dispatcher module
by Jayesh Nambiar 01 Feb '06

01 Feb '06

Hi all, I am trying to use the dispatcher module. I have two gateway address in the dispatcher.list file. But when it reaches that particular gateway, it does not rewrite the host with that gateway. Is dispatcher not supposed to do it?? One more thing.... when I used the function forward(); for forwarding the call to set of gateways, the openser shows an error saying that it is "bad forwardarguement". openser needs the ip addr and the port in the forward function, but then how will it look at the destination set. It worked when I used t_relay(), but I need to forward it statelessly. I just used the function as it was shown in the example of the dispatcher module in the docs.. can someone please help me on this.. Thanks in advance. jayesh. --------------------------------- Jiyo cricket on Yahoo! India cricket

2 1

[Users] Using uac_replace_from to munge username between ser and Asterisk?
by Barry Flanagan 01 Feb '06

01 Feb '06

Hi, I am trying to get a multi-domain setup working between OpenSER and Asterisk (PSTN gateway and Voicemail). It appears that I need to munge the username before it goes to Asterisk so that it includes the domain element, otherwise there is clashing in the username namespace on Asterisk. It looks like I should be able to use uac_replace_from to change the From URI from user(a)domain.com to user_domain(a)domain.com so that then I can set up the users in Asterisk as [user_domain] Does this sound sane, and could someone tell me how I can accomplish this in OpenSER? I have tried a few things but they all end up as bad headers. Thanks in advance. -- -Barry Flanagan

4 6

Re: [Serusers] [serweb]Problem with SerWeb
by mkumar＠mantragroup.com 01 Feb '06

01 Feb '06

Hi All, I tried installing 0.9.2 and 0.9.4 but both behave in the same way, only if I hit 5 to 10 times on login then one can login and in admin only ACl changes and user preferences are working. Server monitoring and admin privileges are not working. Things which are working will work only if I hit 5 to 10 times on them only. Please help me, I do not know where I am going wrong, I followed all the steps in serweb INSTALL doc. Thanks, Manoj. Quoting Karel Kozlik <karel(a)iptel.org>: > Hi, > I looked at your web server but it's behaviour is very strange. It > sometimes don't reply, sometimes not return whole html page. For > example the page > http://www.mantragroup.com/serweb/html/admin/users.php is often cut > off in the middle. > > Try examine your logs. Access and error log of apache and log of php. > > Karel > > mkumar(a)mantragroup.com napsal(a): >> Hi Karel Kozlik, >> >> Thanks for the reply. Initially I tried using serweb admin mode but >> for me only >> the initial page http://www.mantragroup.com/serweb/html/admin/users.php is >> displayed from there if I hit on admin privileges or server >> monitoring or user >> preferences nothing opens, it still shows the same initial page of >> admin login. >> >> Thanks, >> Manoj. >> >> Quoting Karel Kozlik <karel(a)iptel.org>: >> >>> Hi, >>> please (CC) the forum in your replies in order to others can read >>> the answers. >>> >>> To your problem: >>> the correct values to allow change ACL are: >>> >>> priv_name: acl_control >>> priv_value: int >>> >>> priv_name: acl_control >>> priv_value: ld >>> >>> priv_name: acl_control >>> priv_value: local >>> >>> But you realy do not need to know this! >>> >>> Instead of changeing values in DB you can simply login to serweb as admin: >>> - click to tab 'admin privileges' >>> - click to link 'changes privilges' for user 'admin' >>> - mark the checkboxes in 'ACL control' area >>> - click to 'save' >>> >>> Now you should be able to change ACL of other users. >>> >>> Karel >>> >>> >>> >>> mkumar(a)mantragroup.com napsal(a): >>> >>>> Hi Karel Kozlik, >>>> >>>> I really do not know what are the correct values that must be given under >>>> priv_name, for admin I gave is_admin, change_privileges and for giving acl >>>> permissions I tried giving local,ld,acl_ld,acl_local but none worked and I >>>> still get "You haven't any privileges to control ACL" when I log >>>> in to admin, >>>> can you please give me syntax which grants acl change privilges to >>>> admin. I >>>> tried searching on net and doc which came with installation but >>>> none helped me. >>>> Please spare a minute and help me configure serweb. >>>> >>>> Thanks, >>>> Manoj. >>>> >>>> Quoting Karel Kozlik <karel(a)iptel.org>: >>>> >>>>> Hi, >>>>> that is OK. You can have more admins in serweb. And you can >>>>> define privileges for changeing ACL for each admin. To do that, >>>>> click to tab 'admin privileges' and select 'change privileges' >>>>> for some admin. Here you may define which values of ACL this >>>>> admin may change. >>>>> >>>>> Karel >>>>> >>>>> mkumar(a)mantragroup.com napsal(a): >>>>> >>>>>> Hi All, >>>>>> >>>>>> I was able to install serweb and login to it both from admin and >>>>>> user end. But >>>>>> in admin end I have a problem, when I hit on ACL of any user >>>>>> that appered when >>>>>> I first login as admin, I get a error "You haven't any >>>>>> privileges to control >>>>>> ACL". In the DataBase I gave the following privileges to admin >>>>>> >>>>>> username admin >>>>>> domain mantragroup.com >>>>>> priv_name change_privileges >>>>>> priv_value 1 >>>>>> >>>>>> username admin >>>>>> domain mantragroup.com >>>>>> priv_name change_privileges >>>>>> priv_value 1 >>>>>> >>>>>> Someone please help me and guide me to inderstand where I am >>>>>> going wrong. >>>>>> >>>>>> Thanks, >>>>>> Manoj >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Serusers mailing list >>>>>> serusers(a)lists.iptel.org >>>>>> http://lists.iptel.org/mailman/listinfo/serusers >>>>> >>>>> >>>>> >>>> >>>> >>>> >>> >> >> >> >

2 2

[Serusers] About Serweb installation
by Guty 01 Feb '06

01 Feb '06

> I'm running my SER on a debian 3.1. > I've installed (i think so) all necesary stuff to run SER whit mysql > support and serweb interface. > when I try to acces it, i receive the following message: > > Warning: main(DB.php): failed to open stream: No such file or directory in > /home/gustavo/serweb/html/main_prepend.php on line 38 > > Fatal error: main(): Failed opening required 'DB.php' > (include_path='.:/usr/share/php:/usr/share/pear') in > /home/gustavo/serweb/html/main_prepend.php on line 38 > > > I'll preciate if anyone could help me.... > (as you could see, I'm new to all this stuff... ) > Thank you.....

2 1

[Serusers] Presence + XCAP root problem with multiple domains
by Adrian Georgescu 01 Feb '06

01 Feb '06

Hello Vaclav, I cannot manage to handle the same username under different domain names using the same XCAP tree. Data location for user1(a)example1.com will conflict with data location for user1(a)example2.com on the filesystem is the same: /xcap-root/pres-rules/users/user1/presence-rules.xml My XCAP server can generate separate trees for each domain and create the XCAP structure under each domain (like /xcap-root/example1/pres-rules/users/ ) but in this case SER should be able to build a dynamic root like: https://xcap.example.com/$domain$/ In Eyebeam I can configure the xcap-root statically but obviously if I do this for multiple domains, SER should be able to build it from information available in the PUBLISH/SUBSCRIBE request so that it can locate the right tree on the XCAP server. Do you see any other solution for this problem? Regards, Adrian

2 1

[Serusers] Serweb does not send mail on registration
by Bart J. Smit 01 Feb '06

01 Feb '06

I have serweb-0.9.4 installed on CentOS 4.2 with MySQL 4.1.12-3 and ser-0.9.6-1 compiled from the source RPM at http://ftp.upjs.sk/pub/users/sal/Fedora/4/voip. When I register a new user, the PHP script goes to an empty page without title and the email never gets sent to the address entered on the registration form. The database gets the correct user details inserted into the pending table. The sendnotification and greeting fields remain empty. I have modified the line in config_data_layer.php for the collation character set to read: $config->data_sql->collation = "latin1_swedish_ci"; This got rid of a DB error in serweb which I traced through pear logging: Illegal mix of collations (latin1_swedish_ci,IMPLICIT) and (utf8_general_ci,COERCIBLE) for operation '='] I have sent a test email with the debugging/mail.php script and it arrives fine. I have tested the registration page with local recipients, recipients on the LAN and external recipients but the behaviour stays the same. All the pages I have googled on serweb registration problems always state that the emails are sent out correctly, so I'm likely missing something obvious. Can anybody help me get the registration to work? Thanks, Bart...

2 1

[Serusers] windows messenger 5.1 and IM
by Yi Zheng 01 Feb '06

01 Feb '06

Note: forwarded message attached.

2 1

[Serusers] [serweb] Problem with admin privileges, account, accounting, aliases
by mkumar＠mantragroup.com 01 Feb '06

01 Feb '06

Hi All, I installed serweb and able to login both as admin and user. But some features are not at all working for me and also I could not see any error message. The following tabs in admin mode are not working account,accounting,aliases in users admin privileges server monitoring after enabling cron jobs The above tabs common to user mode like accounting are also not working. I have mysql version 3.23.58. I am getting these problems only because mysql version is <4.0 or should I update or add anything to my ser.cfg or any other place? Someone please help me. I do not know where I am going wrong or what to do and there are no errors as well. Thanks, Manoj.

1 0

[Serusers] SER and Cisco ATA 186
by R2 01 Feb '06

01 Feb '06

Hi all, How I can supress of uri user=phone ? This ATA receive call from PSTN, but, when I send calls to PSTN, i don't obtain response. The integration to PSTN is on Asterisk. Thanks R2

2 1

[Serusers] Help on SER with Mediaproxy
by Muhammad Asif Ali 01 Feb '06

01 Feb '06

Hi All I have a scenario in which I want to deploy SER with mediaproxy & allow everyone/anyone connecting with my SER to talk to each other. I am using ONSIP gettingStarted document. In order to allow everyone to connect to my SER without authentication I commented the following portion in ser.cfg: in Route[2] #if (!www_authorize("","subscriber")) { # www_challenge("","0"); # break; #}; #if (!check_to()) { # sl_send_reply("401", "Unauthorized"); # break; #}; # consume_credentials(); and in Route[3] # if (!proxy_authorize("","subscriber")) { # proxy_challenge("","0"); # break; #} else if (!check_from()) { # sl_send_reply("403", "Use From=ID"); # break; #}; #consume_credentials(); Now when I try to connect 2 UA (Simpleopal) from behind a same NAT, they do get registered, & records also appear in location table of DataBase. But when I place call from UA1 to UA2 like "call sip:UA2@aa.bb.cc.dd", where aa.bb.cc.dd is the public IP of my SER machine (mediaproxy is also running on same machine), it ends up showing transport error. On SER side: A) SER logs say: on lookup() "UA2 not found in usrloc" and on is_local() "Realm 'ab.cc.dd.ee' is not local". the is_lcoal appears again saying "Realm 192.168.0.241 is not local" where 192.168.0.241 is UA2's local IP. B)Mediaproxy log says: command request c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1 203.82.51.9:5000:audio 203.82.51.9 aa.bb.cc.dd remote 192.168.0.241 remote OPAL/2.0 info= from:UA1@aa.bb.cc.dd,to:UA2@aa.bb.cc.dd ,fromtag:b09168c6-fbf4-1810-80b4-0050badb4a03,totag: session c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1: started. listening on aa.bb.cc.dd:35000 command execution time: 9.35 ms command request c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1 203.82.51.9:5000:audio 203.82.51.9 aa.bb.cc.dd remote 192.168.0.241 remote OPAL/2.0 info= from:UA1@aa.bb.cc.dd,to:UA2@aa.bb.cc.dd ,fromtag:b09168c6-fbf4-1810-80b4-0050badb4a03,totag: command execution time: 0.60 ms session c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1: 0/0/0 packets, 0/0/0 bytes (caller/called/relayed) session c78d68c6-fbf4-1810-80b4-0050badb4a03@UA1: ended (did timeout). (203.82.51.9) is my LAN's public IP aa.bb.cc.dd is the public IP of SER machine C) using ngrep I found out that it sends an "SIP/2.0 408 Request Timeout" cant figure out whats going on, anybody having any idea please help. Thanx in advance Regards Muhammad Asif Ali

2 4

[Serusers] any info reg adding DTLS using OpenSSL
by Pjothi 01 Feb '06

01 Feb '06

Dear all, I am a student and am trying to setup a demonstration with TLS and DTLS support between a SIP client and a Proxy. Has anyone some information regarding adding DTLS support for a SIP client/Proxy or a more generic one using OpenSSL. I just need it for demonstration purposes and so error handling's if any need not be complete. Just a simple support and a demonstration of both would do fine. If anyone has any information regarding this, may be a how to start or any introductory article would be of great help, kindly let me know. This would be of great help. Any suggestions regarding the same are most welcome. kindly let me know and thank you very much. regards, Pjothi

2 1

[Serusers] STUN and symmetric nat again
by Istvan De 01 Feb '06

01 Feb '06

Hi! I've read about the above topic in the arcives and now I've got the same problem. User A is behind a symmetric NAT -> a new port is bound for every different (external ip, external port) pair. User B has a connection to the net without any NAT or firewall. User A wants to have a SIP+RTP session with user B. How? I guess basically what is needed is forming of a SIP message like "send the your reply to the same ip and same port from where you have received this" and user A can send this to user B. For the RTP session, a pretty similar thing would be needed like, A says to B: "I send you the RTP stream on your x port, send your rtp stream back to the same ip and port from where you receive mine". How to setup such a SIP packet? Or maybe there is a standard solution for this already? (STUN wont help in this case I guess, and TURN is not an option for me.) I've read about "symmetric SIP + symmetric RTP" maybe that is the same as I've described above? Thanks Istvan

3 2

[Serusers] adress book
by Joao Pereira 01 Feb '06

01 Feb '06

Hello to all Im using X-lite, eyeBeam and Cisco 79XX phones and I would like to know the best way of implement a centralized address book system. Maybe the solution is LDAP, but these clients doesnt seem to support LDAP.Who should contact the LDAP directory? the SIP clients or the SIP server? Thanks Joao Pereira

4 4

Re: [Serusers] Monitoring
by hgaillac-sip＠yahoo.fr 01 Feb '06

01 Feb '06

Do you Know HSRP from Cisco System ? http://www.cisco.com/univercd/cc/td/doc/cisintwk/ics/cs009.htm Harry --- Hendrik Scholz <hendrik.scholz(a)freenet-ag.de> a écrit : > Hi! > > hgaillac-sip(a)yahoo.fr wrote: > > What about VRRP for failover ? > > > > SER1 (master) > > sip agents == || === Asterisk Farm==sip/pstn > gateway > > SER2 (slave) > > What do you mean by that? > > Somewhat confused, > Hendrik > > -- > freenet Cityline GmbH, Hamburger Chaussee 2-4, 24114 > Kiel, Germany > Phone: +49 (0)431 9020552, Fax: +49 (0)431 9020559 > Internet: http://www.freenet.de, eMail: > hendrik.scholz(a)freenet-ag.de > ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

2 1

[Serusers] About Serweb installation
by Gustavo Moreira 01 Feb '06

01 Feb '06

> I'm running my SER on a debian 3.1. > I've installed (i think so) all necesary stuff to run SER whit mysql > support and serweb interface. > when I try to acces it, i receive the following message: > > Warning: main(DB.php): failed to open stream: No such file or directory in > /home/gustavo/serweb/html/main_prepend.php on line 38 > > Fatal error: main(): Failed opening required 'DB.php' > (include_path='.:/usr/share/php:/usr/share/pear') in > /home/gustavo/serweb/html/main_prepend.php on line 38 > > > I'll preciate if anyone could help me.... > (as you could see, I'm new to all this stuff... ) > Thank you.....

1 0

[Serusers] About Serweb installation
by Guty 01 Feb '06

01 Feb '06

I'm running my SER on a debian 3.1. I've installed (i think so) all necesary stuff to run SER whit mysql support and serweb interface. when I try to acces it, i receive the following message: Warning: main(DB.php): failed to open stream: No such file or directory in /home/gustavo/serweb/html/main_prepend.php on line 38 Fatal error: main(): Failed opening required 'DB.php' (include_path='.:/usr/share/php:/usr/share/pear') in /home/gustavo/serweb/html/main_prepend.php on line 38 I'll preciate if anyone could help me.... (as you could see, I'm new to all this stuff... ) Thank you.....

1 0

[Serusers] windows messenger 5.1 and IM
by Yi Zheng 31 Jan '06

31 Jan '06

I have two window messenger 5.1 clients on the same subnet as the SER server w/ pa module. They subscribe/notify to each other ok. HOwever they cannot IM each other. The callee, after receiveing INVITE, sent back a 100 trying, then 200 OK, and then immediately BYE. I searched thru archives and looks like this is a known issue with wm. What is the solution or workaround to this? Here is the messages I captured: U 10.1.104.251:1172 -> 10.1.104.254:5060 INVITE sip:104pc1@10.1.104.254 SIP/2.0..Via: SIP/2.0/UDP 10.1.104.251:8053 ..Max-Forwards: 70..From: "104pc2(a)10.1.104.254" <sip:104pc2@10.1.104.254>; tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..To: <sip:104pc1@10.1 .104.254>..Call-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..Cont act: <sip:10.1.104.251:8053>..User-Agent: RTC/1.3..Content-Type: applicati on/sdp..Content-Length: 119....v=0..o=- 0 0 IN IP4 10.1.104.251..s=session ..c=IN IP4 10.1.104.251..t=0 0..m=message 5060 sip sip:104pc2@10.1.104.254 .. # U 10.1.104.254:5060 -> 10.1.104.251:8053 SIP/2.0 100 trying -- your call is important to us..Via: SIP/2.0/UDP 10.1. 104.251:8053..From: "104pc2(a)10.1.104.254" <sip:104pc2@10.1.104.254>;tag=d4 6c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..To: <sip:104pc1@10.1.104.2 54>..Call-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..Server: Si p EXpress router (0.10.99-dev30-tm-timers-pa-3 (i386/linux))..Content-Leng th: 0..Warning: 392 10.1.104.254:5060 "Noisy feedback tells: pid=23866 re q_src_ip=10.1.104.251 req_src_port=1172 in_uri=sip:104pc1@10.1.104.254 out _uri=sip:10.1.104.248:13018 via_cnt==1".... # U 10.1.104.254:5060 -> 10.1.104.248:13018 INVITE sip:10.1.104.248:13018 SIP/2.0..Record-Route: <sip:10.1.104.254;fta g=d46c61cfc8bf4ae4942b0e9c2a2f38d7;lr=on>..Via: SIP/2.0/UDP 10.1.104.254;b ranch=z9hG4bK2216.fea17c01.0..Via: SIP/2.0/UDP 10.1.104.251:8053..Max-Forw ards: 16..From: "104pc2(a)10.1.104.254" <sip:104pc2@10.1.104.254>;tag=d46c61 cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..To: <sip:104pc1@10.1.104.254>. .Call-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..Contact: <sip: 10.1.104.251:8053>..User-Agent: RTC/1.3..Content-Type: application/sdp..Co ntent-Length: 119..P-hint: usrloc applied....v=0..o=- 0 0 IN IP4 10.1.104. 251..s=session..c=IN IP4 10.1.104.251..t=0 0..m=message 5060 sip sip:104pc 2(a)10.1.104.254.. # U 10.1.104.248:1334 -> 10.1.104.254:5060 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.1.104.254;branch=z9hG4bK2216.fea17 c01.0..Via: SIP/2.0/UDP 10.1.104.251:8053..From: "104pc2(a)10.1.104.254" <si p:104pc2@10.1.104.254>;tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d136 2..To: <sip:104pc1@10.1.104.254>;tag=36f889ef11354d12890edd7a068dc893..Cal l-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..User-Agent: RTC/1. 3..Content-Length: 0.... # U 10.1.104.248:1334 -> 10.1.104.254:5060 SIP/2.0 200 OK..Via: SIP/2.0/UDP 10.1.104.254;branch=z9hG4bK2216.fea17c01. 0..Via: SIP/2.0/UDP 10.1.104.251:8053..From: "104pc2(a)10.1.104.254" <sip:10 ;tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..T">4pc2(a)10.1.104.254>;tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=69200d1362..T o: <sip:104pc1@10.1.104.254>;tag=36f889ef11354d12890edd7a068dc893..Call-ID : 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 INVITE..Record-Route: <sip:10. 1.104.254;ftag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;lr=on>..Contact: <sip:10.1 .104.248:13018>..User-Agent: RTC/1.3..Content-Type: application/sdp..Conte nt-Length: 119....v=0..o=- 0 0 IN IP4 10.1.104.248..s=session..c=IN IP4 10 .1.104.248..t=0 0..m=message 5060 sip sip:104pc1@10.1.104.254.. # U 10.1.104.248:1334 -> 10.1.104.254:5060 BYE sip:10.1.104.254;ftag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;lr=on SIP/2.0.. Via: SIP/2.0/UDP 10.1.104.248:13018..Max-Forwards: 70..From: <sip:104pc1@1 0.1.104.254>;tag=36f889ef11354d12890edd7a068dc893..To: "104pc2(a)10.1.104.25 4" <sip:104pc2@10.1.104.254>;tag=d46c61cfc8bf4ae4942b0e9c2a2f38d7;epid=692 00d1362..Call-ID: 7dc26db0eb374cedbfa098d8fb7e2796..CSeq: 1 BYE..Route: <s ip:10.1.104.251:8053>..User-Agent: RTC/1.3..Content-Length: 0.... ##

1 0

[Users] Parsing SIP responses
by Ovidiu Sas 31 Jan '06

31 Jan '06

Hi, I configured openser to handle redirect responses and re-route calls based on the Contact header from the 3xx response using the uac_redirect module. In addition to that, I would like to transfer some headers from the 3xx response into the redirected INVITE (see diagram): UAC openser UAS1 UAS2 | | | | |---INVITE--->| | | | |---INVITE--->| | | |<--3XX-------| | | | with Diversion header | | |---ACK------>| | | |---------------INVITE------->| | | | with Diversion header | | | from 3xx response from UAS1 Is there an API that will allow me to search for headers inside a response and save them in to an AVP? Thx, -ovi

2 3

Re: [Serusers] no SRV query after rewriting R-URI?
by alan chiu 31 Jan '06

31 Jan '06

great, thanks for the clarification...so simply, rewrite("host") would trigger SRV lookup. --- Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > Hi Alan! > > That's the correct behaviour. Please read RFC 3263. > If the URI contains a > port, e.g. inside.acme.com:5060 there must not be an > SRV lookup. Because > the SRV lookup leads to a port, which is already > defined. > > regards > klaus > > On Tue, January 31, 2006 20:28, alan chiu said: > > Server: Sip EXpress router (0.8.12 > (i86pc/solaris)) > > > > when endpoint sends 18005551212(a)sip.acme.com, SER > > (without rewriting of R-URI, looks it up with SRV > > *************** > > > > 386.656875 192.168.1.246 -> 10.100.0.111 SIP/SDP > > Request: INVITE sip:18005551212@sip.acme.com, with > > session description > > 386.657307 10.100.0.111 -> 192.168.1.246 SIP > Status: > > 100 trying -- your call is important to us > > 88.811658 10.100.0.111 -> 192.168.1.11 DNS > Standard > > query SRV _sip._udp.sip.acme.com > > 98.825208 192.168.1.11 -> 10.100.0.111 DNS > Standard > > query response, No such name > > 98.825496 10.100.0.111 -> 192.168.1.11 DNS > Standard > > query A sip.acme.com > > 108.845136 192.168.1.11 -> 10.100.0.111 DNS > Standard > > query response, No such name > > 108.845468 10.100.0.111 -> 192.168.1.11 DNS > Standard > > query AAAA sip.acme.com > > 118.865215 192.168.1.11 -> 10.100.0.111 DNS > Standard > > query response, No such name > > 416.711408 10.100.0.111 -> 192.168.1.246 SIP > Status: > > 478 Unresolveable destination (478/TM) > > 407.921364 192.168.1.246 -> 10.100.0.111 SIP > Request: > > ACK sip:18005551212@sip.acme.com > > > > > *********************************************************************************** > > > > > > ************* > > after rewritehostport("inside.acme.com:5060"), SER > > only does an A lookup > > *************** > > > > 424.851557 192.168.1.246 -> 10.100.0.111 SIP/SDP > > Request: INVITE sip:18005551212@nat.foo.com, with > > session descripti > > on > > 415.971953 10.100.0.111 -> 192.168.1.246 SIP > Status: > > 100 trying -- your call is important to us > > 127.006508 10.100.0.111 -> 192.168.1.11 DNS > Standard > > query A inside.foo.com > > 127.007557 192.168.1.11 -> 10.100.0.111 DNS > Standard > > query response A 192.168.1.211 > > 424.853464 10.100.0.111 -> 192.168.1.211 SIP/SDP > > Request: INVITE > sip:18005551212@inside.foo.com:5060, > > with session descriptio > > n > > 415.973466 10.100.0.111 -> 192.168.1.211 SIP/SDP > > Request: INVITE > sip:18005551212@inside.foo.com:5060, > > with session descriptio > > n > > > > > ****************************************************************** > > > > How can I get SER to consistently do SRV lookup > after > > rewriting of R-URI? > > > > _______________________________________________ > > Serusers mailing list > > serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > >

1 0

Re: [Users] Path
by Bogdan-Andrei Iancu 31 Jan '06

31 Jan '06

forgot to cc the list Bogdan-Andrei Iancu wrote: > Hi Andreas, > > you are right - the problem was not the string itself, but the str > itself. Anyhow, hopefully is fixed now. > > regards, > bogdan > > Andreas Granig wrote: > >> Bogdan, >> >>> However, another problem accured. >>> >>> I see in my logs after an INVITE: >>> >>> set_dst_uri: Not enough memory >>> lookup(): Failed to set dst_uri of Path >> >> >> >> In my original patch I made a copy of "route->nameaddr.uri" in >> get_path_dst_uri() of modules/registrar/path.c and then free'd the >> parsed route, but your optimization just points to that uri and also >> free's that route, which invalidates the pointer to that uri, as far >> as I understand. >> >> Can you please check that again, because if I comment out the >> "free_rr(&route)", it does work (although it creates a mem-leak). >> >> Andy > > > >

1 0

[Users] grp reverse function??
by Marcello Lupo 31 Jan '06

31 Jan '06

Hi, there is a way to accomplish the reverse function that group module do? As i know the group module allow me to check if a user is inside a specified group. How can i know in which group is a user without querying the system to know if the user is in all the groups one by one? Actually i'm doing it using avpops and setting an avp that tell me which is the group of the user. I was wandering if there is a more efficient (more performant) way. Thanks, bye, Marcello

2 1

[Users] Is it possiable to rewriting sorc UA name?
by Steve Xu 31 Jan '06

31 Jan '06

Hi All, I found that PSTN GW usually put the "From: xxxx@ipaddress" as ANI (caller ID) when it receive a SIP call. There is a function "rewritingURI" for called URI, I am wandering if there was a function can do rewriting on the the sorc "From" also. For example: user FROM:alice@abc.com make call to hugen(a)abf.com, can I rewrite the alice(a)abc.com to 661234223(a)abc.com when I send INVITE to hugen(a)abf.com ? Any help will be appreciated. Steve

3 3

[Users] privacy=full and contact hf
by Wolfgang Hottgenroth 31 Jan '06

31 Jan '06

Hi, before forwarding a call (INVITE) from the PSTN with flag "privacy=full" in the Remote-Party-ID hf to a none-voice-licensed carrier I need to make sure that the calling party number is completely wiped out from the SIP requests. I see this number at three places: the Remote-Party-ID hf itself, the >From hf and the Contact hf. The Remote-Party-ID hf can simply be dropped or replaced by something anonymous. The From hf can be rewritten using uac_replace_from. But what about the Contact hf? This field is, as far as I can see, required for routing purposes, to tell the called party side where to send replies. So, if I would manage it to rewrite the field, I would have to make sure, the replies sent to the rewritten address once again will be rewritten to target for the meant destination. Is there a best practice for this item? Thanks, Wolfgang

3 7

[Fwd: Re: [Serusers] no SRV query after rewriting R-URI?]
by Klaus Darilion 31 Jan '06

31 Jan '06

I've forgot to cc the list. klaus -------------------------------------------------------------------------- Hi Alan! That's the correct behaviour. Please read RFC 3263. If the URI contains a port, e.g. inside.acme.com:5060 there must not be an SRV lookup. Because the SRV lookup leads to a port, which is already defined. regards klaus On Tue, January 31, 2006 20:28, alan chiu said: > Server: Sip EXpress router (0.8.12 (i86pc/solaris)) > > when endpoint sends 18005551212(a)sip.acme.com, SER > (without rewriting of R-URI, looks it up with SRV > *************** > > 386.656875 192.168.1.246 -> 10.100.0.111 SIP/SDP > Request: INVITE sip:18005551212@sip.acme.com, with > session description > 386.657307 10.100.0.111 -> 192.168.1.246 SIP Status: > 100 trying -- your call is important to us > 88.811658 10.100.0.111 -> 192.168.1.11 DNS Standard > query SRV _sip._udp.sip.acme.com > 98.825208 192.168.1.11 -> 10.100.0.111 DNS Standard > query response, No such name > 98.825496 10.100.0.111 -> 192.168.1.11 DNS Standard > query A sip.acme.com > 108.845136 192.168.1.11 -> 10.100.0.111 DNS Standard > query response, No such name > 108.845468 10.100.0.111 -> 192.168.1.11 DNS Standard > query AAAA sip.acme.com > 118.865215 192.168.1.11 -> 10.100.0.111 DNS Standard > query response, No such name > 416.711408 10.100.0.111 -> 192.168.1.246 SIP Status: > 478 Unresolveable destination (478/TM) > 407.921364 192.168.1.246 -> 10.100.0.111 SIP Request: > ACK sip:18005551212@sip.acme.com > > *********************************************************************************** > > > ************* > after rewritehostport("inside.acme.com:5060"), SER > only does an A lookup > *************** > > 424.851557 192.168.1.246 -> 10.100.0.111 SIP/SDP > Request: INVITE sip:18005551212@nat.foo.com, with > session descripti > on > 415.971953 10.100.0.111 -> 192.168.1.246 SIP Status: > 100 trying -- your call is important to us > 127.006508 10.100.0.111 -> 192.168.1.11 DNS Standard > query A inside.foo.com > 127.007557 192.168.1.11 -> 10.100.0.111 DNS Standard > query response A 192.168.1.211 > 424.853464 10.100.0.111 -> 192.168.1.211 SIP/SDP > Request: INVITE sip:18005551212@inside.foo.com:5060, > with session descriptio > n > 415.973466 10.100.0.111 -> 192.168.1.211 SIP/SDP > Request: INVITE sip:18005551212@inside.foo.com:5060, > with session descriptio > n > > ****************************************************************** > > How can I get SER to consistently do SRV lookup after > rewriting of R-URI? > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

1 0

[Serusers] no SRV query after rewriting R-URI?
by alan chiu 31 Jan '06

31 Jan '06

Server: Sip EXpress router (0.8.12 (i86pc/solaris)) when endpoint sends 18005551212(a)sip.acme.com, SER (without rewriting of R-URI, looks it up with SRV *************** 386.656875 192.168.1.246 -> 10.100.0.111 SIP/SDP Request: INVITE sip:18005551212@sip.acme.com, with session description 386.657307 10.100.0.111 -> 192.168.1.246 SIP Status: 100 trying -- your call is important to us 88.811658 10.100.0.111 -> 192.168.1.11 DNS Standard query SRV _sip._udp.sip.acme.com 98.825208 192.168.1.11 -> 10.100.0.111 DNS Standard query response, No such name 98.825496 10.100.0.111 -> 192.168.1.11 DNS Standard query A sip.acme.com 108.845136 192.168.1.11 -> 10.100.0.111 DNS Standard query response, No such name 108.845468 10.100.0.111 -> 192.168.1.11 DNS Standard query AAAA sip.acme.com 118.865215 192.168.1.11 -> 10.100.0.111 DNS Standard query response, No such name 416.711408 10.100.0.111 -> 192.168.1.246 SIP Status: 478 Unresolveable destination (478/TM) 407.921364 192.168.1.246 -> 10.100.0.111 SIP Request: ACK sip:18005551212@sip.acme.com *********************************************************************************** ************* after rewritehostport("inside.acme.com:5060"), SER only does an A lookup *************** 424.851557 192.168.1.246 -> 10.100.0.111 SIP/SDP Request: INVITE sip:18005551212@nat.foo.com, with session descripti on 415.971953 10.100.0.111 -> 192.168.1.246 SIP Status: 100 trying -- your call is important to us 127.006508 10.100.0.111 -> 192.168.1.11 DNS Standard query A inside.foo.com 127.007557 192.168.1.11 -> 10.100.0.111 DNS Standard query response A 192.168.1.211 424.853464 10.100.0.111 -> 192.168.1.211 SIP/SDP Request: INVITE sip:18005551212@inside.foo.com:5060, with session descriptio n 415.973466 10.100.0.111 -> 192.168.1.211 SIP/SDP Request: INVITE sip:18005551212@inside.foo.com:5060, with session descriptio n ****************************************************************** How can I get SER to consistently do SRV lookup after rewriting of R-URI?

1 0

[Serusers] how to get raw sip logs from SER natively?
by alan chiu 31 Jan '06

31 Jan '06

Is there a good way to write raw sip logs processed by SER to a file? (not live captures)

1 0

[Serusers] how to get raw sip logs from SER natively?
by alan chiu 31 Jan '06

31 Jan '06

Is there a good way to write raw sip logs processed by SER to a file? (not live captures)

1 0

[Serusers] witch RTP solution is the best?
by Pol 31 Jan '06

31 Jan '06

Hi Users, I should implement any RTP proxy solution for our SER deployment to avoid NAT problem. I have tried to look for in google some non subjective reasons to use Mediaproxy or to use RTPproxy but I have not found anything. I would like to know which solution is the best one, which one is most common or witch one are you using, how it works and if its possible some examples or documentation links. Thxs in advance.

2 1

[Serusers] any information regarding adding DTLS using OpenSSL
by Pjothi 31 Jan '06

31 Jan '06

Dear all, I am a student and am trying to setup a demonstration with TLS and DTLS support between a SIP client and a Proxy. Has anyone some information regarding adding DTLS support for a SIP client/Proxy or a more generic one using OpenSSL. I just need it for demonstration purposes and so error handling's if any need not be complete. Just a simple support and a demonstration of both would do fine. If anyone has any information regarding this, may be a how to start or any introductory article would be of great help, kindly let me know. This would be of great help. Any suggestions regarding the same are most welcome. kindly let me know and thank you very much. regards, Pjothi

1 0

[Serusers] features-callfwd.5.0.cfg produces syntax errors running latest CVS
by Kim Culhan 31 Jan '06

31 Jan '06

Are there any example configuration files which will work with ser from the latest CVS ? I have been using a configuration based on features-callfwd.5.0.cfg This appears to be unuseable with ser from CVS as it returns: Jan 28 16:23:13 test ser: parse error (16,1-5): syntax error Jan 28 16:23:13 test ser: parse error (16,5-6): unknown config variable Jan 28 16:23:13 test ser: parse error (17,12-13): unknown config variable Jan 28 16:23:13 test ser: parse error (19,1-11): Any help fixing the syntax errors and config vars is greatly appreciated, how can I trace the sources of these errors in the config file ? -kim -- w8hdkim(a)gmail.com

3 6

Re: [Serusers] failover and VRRP
by hgaillac-sip＠yahoo.fr 31 Jan '06

31 Jan '06

Is it the best way ? Which solution would be able to check the ser availability bfore estabilsh the connections SER2 sip agents===DNS servers==[the solutions ??]== SER1 Harry --- sip <sip(a)arcdiv.com> a écrit : > We use Coyote Point load balancers (good, relatively > cheap, hardware > load-balancers) that monitor the connectivity to the > SER server. The problem > is that it ONLY monitors connectivity, so if your > SER is still accepting > connections to the socket, but then not working > correctly, it doesn't know the > difference. > > Our SER works off a centrally-shared DB instead of a > replicated one (mysql > running from a SAN connected to both machines > through a SAN switch). That way, > if one of the servers goes down, the other one > picks up as if nothing ever > happened. > > SER does not have stateful connections from the UA, > so if the server gets > switched in the middle of a conversation, it will > cut the conversation off... > but at least there will still be a server active for > additional connections. > > N. > > > On Tue, 31 Jan 2006 13:24:00 +0100 (CET), > hgaillac-sip wrote > > Do you know a solution !? > > > > Harry > > --- sip <sip(a)arcdiv.com> a écrit : > > > > > Right. VRRP is for layer 2/3 but it's for > routing > > > purposes. > > > > > > If you want to handle the application layer with > > > failover, you need a > > > load-balancer of some sort. That way, if one SER > > > dies, the load balancer sends > > > data on to the other one. > > > > > > N. > > > > > > On Tue, 31 Jan 2006 12:37:56 +0100 (CET), > > > hgaillac-sip wrote > > > > Hello, > > > > > > > > I think vrrp handle layer 2/3 not sip > application > > > what > > > > do you advise me ? > > > > > > > > Harry > > > > > > > > --- sip <sip(a)arcdiv.com> a écrit : > > > > > > > > > Sure. You can even use VRRP if there's no > SER in > > > > > your network. I'm not sure > > > > > what that has to do with SER, though. > > > > > > > > > > VRRP is router-side. > > > > > > > > > > N. > > > > > > > > > > > > > > > On Tue, 31 Jan 2006 10:32:03 +0100 (CET), > > > > > hgaillac-sip wrote > > > > > > hello, > > > > > > > > > > > > Can we use VRRP when a SER is down ? > > > > > > > > > > > > SER1 (master) > > > > > > sip agents == || === Asterisk > Farm==sip/pstn > > > > > gateway > > > > > > SER2 (slave) > > > > > > > > > > > > Regards > > > > > > Harry > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ___________________________________________________________________________ > > > > > Nouveau : téléphonez moins cher avec Yahoo! > > > > > Messenger ! Découvez les tarifs > > > > > exceptionnels pour appeler la France et > > > > > l'international. Téléchargez sur > > > > > http://fr.messenger.yahoo.com > > > > > > > > > > > > > > > _______________________________________________ > > > > > > Serusers mailing list > > > > > > serusers(a)lists.iptel.org > > > > > > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ___________________________________________________________________________ > > > Nouveau : téléphonez moins cher avec Yahoo! > > > Messenger ! Découvez les tarifs > > > exceptionnels pour appeler la France et > > > l'international. Téléchargez sur > > > http://fr.messenger.yahoo.com > > > > > > > > > > > > > > > > > > > ___________________________________________________________________________ > Nouveau : téléphonez moins cher avec Yahoo! > Messenger ! Découvez les tarifs > exceptionnels pour appeler la France et > l'international. Téléchargez sur > http://fr.messenger.yahoo.com > > ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

1 0

[Serusers] ser and asterisk
by preeti j 31 Jan '06

31 Jan '06

hello, I am able to issue 302 redirect from ser to asterisk.is it possible to redirect calls to a realtime peer registerd on asterisk. please help me with config requirement.cldn't find anything in the docs relating to realtime. Thanks, Preeti --------------------------------- Do you Yahoo!? With a free 1 GB, there's more in store with Yahoo! Mail.

2 1

[Serusers] SER Presence
by Akeru Ikena 31 Jan '06

31 Jan '06

Hi Vaclav, > In the config in handling PUBLISH request call handle_publish instead of > handle_subscription. Woops, right, thanks alot. Here are some details, I'm using SER 0.9.6 Unstable from debian package (http://ftp.iptel.org/pub/ser/0.9.6/packages/debian/unstable/) If I add one of these parameters in ser.cfg file, modparam("pa", "auth", "none") modparam("pa", "winfo_auth", "none") modparam("pa", "auth_xcap_root", "http://localhost/xcap-root") I get this error : ERROR: bad config file (1 errors) Erreur de segmentation And when using PUA.pl to test this, I always received a closed presence state. Do you have any idea ? Cheers, Akeru.

2 3

[Serusers] failover and VRRP
by hgaillac-sip＠yahoo.fr 31 Jan '06

31 Jan '06

hello, Can we use VRRP when a SER is down ? SER1 (master) sip agents == || === Asterisk Farm==sip/pstn gateway SER2 (slave) Regards Harry ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

3 2

[Serusers] Monitoring
by hgaillac-sip＠yahoo.fr 31 Jan '06

31 Jan '06

Hi asterisk and ser users, Is there a solution to monitor asterisk and ser with snmp ? Regards Harry ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

4 5

Re: [Serusers] Problem with SerWeb
by mkumar＠mantragroup.com 31 Jan '06

31 Jan '06

Hi Karel Kozlik, Thanks for the reply. Initially I tried using serweb admin mode but for me only the initial page http://www.mantragroup.com/serweb/html/admin/users.php is displayed from there if I hit on admin privileges or server monitoring or user preferences nothing opens, it still shows the same initial page of admin login. Thanks, Manoj. Quoting Karel Kozlik <karel(a)iptel.org>: > Hi, > please (CC) the forum in your replies in order to others can read the > answers. > > To your problem: > the correct values to allow change ACL are: > > priv_name: acl_control > priv_value: int > > priv_name: acl_control > priv_value: ld > > priv_name: acl_control > priv_value: local > > But you realy do not need to know this! > > Instead of changeing values in DB you can simply login to serweb as admin: > - click to tab 'admin privileges' > - click to link 'changes privilges' for user 'admin' > - mark the checkboxes in 'ACL control' area > - click to 'save' > > Now you should be able to change ACL of other users. > > Karel > > > > mkumar(a)mantragroup.com napsal(a): >> Hi Karel Kozlik, >> >> I really do not know what are the correct values that must be given under >> priv_name, for admin I gave is_admin, change_privileges and for giving acl >> permissions I tried giving local,ld,acl_ld,acl_local but none worked and I >> still get "You haven't any privileges to control ACL" when I log in >> to admin, >> can you please give me syntax which grants acl change privilges to admin. I >> tried searching on net and doc which came with installation but none >> helped me. >> Please spare a minute and help me configure serweb. >> >> Thanks, >> Manoj. >> >> Quoting Karel Kozlik <karel(a)iptel.org>: >> >>> Hi, >>> that is OK. You can have more admins in serweb. And you can define >>> privileges for changeing ACL for each admin. To do that, click to >>> tab 'admin privileges' and select 'change privileges' for some >>> admin. Here you may define which values of ACL this admin may >>> change. >>> >>> Karel >>> >>> mkumar(a)mantragroup.com napsal(a): >>> >>>> Hi All, >>>> >>>> I was able to install serweb and login to it both from admin and >>>> user end. But >>>> in admin end I have a problem, when I hit on ACL of any user that >>>> appered when >>>> I first login as admin, I get a error "You haven't any privileges >>>> to control >>>> ACL". In the DataBase I gave the following privileges to admin >>>> >>>> username admin >>>> domain mantragroup.com >>>> priv_name change_privileges >>>> priv_value 1 >>>> >>>> username admin >>>> domain mantragroup.com >>>> priv_name change_privileges >>>> priv_value 1 >>>> >>>> Someone please help me and guide me to inderstand where I am going wrong. >>>> >>>> Thanks, >>>> Manoj >>>> >>>> >>>> _______________________________________________ >>>> Serusers mailing list >>>> serusers(a)lists.iptel.org >>>> http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >> >> >> >

2 1

[Users] loose_route()
by JF 31 Jan '06

31 Jan '06

Hi, One of my (Open)SER machines (P1) is acting as registrar and so I have alias=mydomain.tld in the script; it is used in the (uri == myself) check to handle REGISTER requests. I have a setup like this: UA1 (user1(a)mydomain.tld) - P2 - P1 - P3 - P1 - P2 - UA2 (user2(a)mydomain.tld) (all proxies are record-routing except P3) For initial requests (INVITE from UA1 to UA2) all is fine. But I'm having problems with the loose_route() function, because when an in-dialog (subsequent) request comes into P1 (from UA1, P2) with a Route header containing (<P1>, <P1>, <P2>), the after_strict() function is called, when I would expect the after_loose() function to be called since P1 is a loose router. Within loose_route(), is_preloaded() returns 0, which is correct. But then is_myself returns true because of the alias... Is this a bug in loose_route() which doesn't account for this kind of setup (Routeing in-dialog requests through the registrar)? Maybe this could be avoided, if there was a parameter forcing loose_route to treat previous hop as a loose router... Or am I just doing something wrong? Thanks in advance, JF

2 2

[Users] Path
by Helge Waastad 31 Jan '06

31 Jan '06

Hi, just a few questions regarding the path implementation (if someone cares to answer, taht is ;-)) I've read through the RFC3327 (I hope) If I have a scenario: UAC->OBP->Registrar (th OB is a dispatcher) In the current version, should I append_hf both: Supported: path and a Path: <sip:[OBP];lr> to make it work? I tried. I get a P-Registrar-Error: Path parse error and in my log: ERROR: build_path_vector(): First Path URI is not a loose-router, not supported I guess I might have a configuration error, but do not really know. br hw

3 14

[Users] Can openser put a Contact header in a response when behave as UAS?
by Jose Antonio Garvayo 31 Jan '06

31 Jan '06

Hi, I want to use the openser as a UAS on behalf of a user that's not registered. There's a proxy in the middle that receives an INVITE and find that the user is not registered, and rely the INVITE to the openser. I want the openser to send a 200 OK response, but also to put a Contact in the response so the final ACK will be send also to the openser. I have tried the following: append_hf("Contact: xxxxx:pp"); if (t_newtran()) { xlog("UAS logic"); append_hf("Contact: xxxxx:pp"); t_reply("200","OK"); append_hf("Contact: xxxxx:pp"); } else sl_reply_error(); But still no Contact appears in the 200 OK response. Thanks

2 2

Re: [Serusers] Problem with SerWeb
by Karel Kozlik 31 Jan '06

31 Jan '06

Hi, please (CC) the forum in your replies in order to others can read the answers. To your problem: the correct values to allow change ACL are: priv_name: acl_control priv_value: int priv_name: acl_control priv_value: ld priv_name: acl_control priv_value: local But you realy do not need to know this! Instead of changeing values in DB you can simply login to serweb as admin: - click to tab 'admin privileges' - click to link 'changes privilges' for user 'admin' - mark the checkboxes in 'ACL control' area - click to 'save' Now you should be able to change ACL of other users. Karel mkumar(a)mantragroup.com napsal(a): > Hi Karel Kozlik, > > I really do not know what are the correct values that must be given under > priv_name, for admin I gave is_admin, change_privileges and for giving acl > permissions I tried giving local,ld,acl_ld,acl_local but none worked and I > still get "You haven't any privileges to control ACL" when I log in to > admin, > can you please give me syntax which grants acl change privilges to admin. I > tried searching on net and doc which came with installation but none > helped me. > Please spare a minute and help me configure serweb. > > Thanks, > Manoj. > > Quoting Karel Kozlik <karel(a)iptel.org>: > >> Hi, >> that is OK. You can have more admins in serweb. And you can define >> privileges for changeing ACL for each admin. To do that, click to tab >> 'admin privileges' and select 'change privileges' for some admin. Here >> you may define which values of ACL this admin may change. >> >> Karel >> >> mkumar(a)mantragroup.com napsal(a): >> >>> Hi All, >>> >>> I was able to install serweb and login to it both from admin and user >>> end. But >>> in admin end I have a problem, when I hit on ACL of any user that >>> appered when >>> I first login as admin, I get a error "You haven't any privileges to >>> control >>> ACL". In the DataBase I gave the following privileges to admin >>> >>> username admin >>> domain mantragroup.com >>> priv_name change_privileges >>> priv_value 1 >>> >>> username admin >>> domain mantragroup.com >>> priv_name change_privileges >>> priv_value 1 >>> >>> Someone please help me and guide me to inderstand where I am going >>> wrong. >>> >>> Thanks, >>> Manoj >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org >>> http://lists.iptel.org/mailman/listinfo/serusers >> >> > > >

1 0

[Serusers] [serweb]Grant acl privilege to admin
by mkumar＠mantragroup.com 31 Jan '06

31 Jan '06

Hi All, I was able to install serweb and able to login both as admin and user. After I log in as admin when I hit on ACL of any user that appeared, I get "You haven't any privileges to control ACL". Can someone please tell me what are the permissions that must be granted to admin for enabling ACL change privilege, currently admin is like this username admin domain mantragroup.com priv_name is_admin priv_value 1 username admin domain mantragroup.com priv_name change_privileges priv_value 1 Any help will be sincerely appreciated. Thanks, Manoj.

1 0

[Serusers] failure_route problem
by Walter Luis 31 Jan '06

31 Jan '06

Dear Friends, The script works? []'s Walter On 10/20/05, Klaus Darilion < klaus.mailinglists(a)pernau.at> wrote: > > > failure_route[1]{ > > > > if(t_check_status("487")) { > break; > } > > klaus > > log(1,"!!!!!!!!!!! Timeout!!!!!!!!!!!!!!!!"); > > > > revert_uri(); > > > > rewritehostport(" 10.1.3.11:5061"); > > > > append_branch(); > > > > t_relay(); > > > > break(); > > > > } > > > > > > > > > > > > But my problem is that SER still rewrites the url and forwards it to > > asterisk even that the calling person has hung up. > > > > Does anyone know how to do it that SER will forward the call only in > > case of time out and will not forward when the caller hang up. > > > > Any help will be welcome. > > > > > > > > Best regards > > > > Jarek > > > > > > > > > > > > > > > > > > > > > > ------------------------------------------------------------------------ > > > > _______________________________________________ > > 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 > -- Walter Luis -- Walter Luis

1 0

[Serusers] ser uses mediaproxy?
by Toygun Mavinil 30 Jan '06

30 Jan '06

Hi, I am using ser 0.9.6 on fc4 Just installed mediaproxy and started it as service Then same thing on ser Now how can i be sure that ser uses mediaproxy My ser.cfg file is from onsip example # $Id: nat-mediaproxy.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ debug=3 fork=yes log_stderror=no listen=public ip # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.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" modparam("auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.0.2.13:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("FQDN","subscriber")) { www_challenge("FQDN","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } Toygun

1 0

[Serusers] mediaproxy
by Toygun Mavinil 30 Jan '06

30 Jan '06

Hi, I am using ser 0.9.6 on fc4 Just installed mediaproxy and started it as service Then same thing on ser Now how can i be sure that ser uses mediaproxy My ser.cfg file is from onsip example # $Id: nat-mediaproxy.cfg 9 2005-08-19 15:30:55Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ debug=3 fork=yes log_stderror=no listen=public ip # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/domain.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" modparam("auth_db|domain|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("192.0.2.13:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3") || search("^Route:.*;nat=yes")) { setflag(6); use_media_proxy(); }; }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("FQDN","subscriber")) { www_challenge("FQDN","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } Toygun

1 0

[Serusers] failover and VRRP
by hgaillac-sip＠yahoo.fr 30 Jan '06

30 Jan '06

hello, Can we use VRRP when a SER is down ? SER1 (master) sip agents ==== || === Asterisk Farm==sip/pstn gateway SER2 (slave) Regards Harry ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

1 0

[Serusers] Problem with SerWeb
by mkumar＠mantragroup.com 30 Jan '06

30 Jan '06

Hi All, I was able to install serweb and login to it both from admin and user end. But in admin end I have a problem, when I hit on ACL of any user that appered when I first login as admin, I get a error "You haven't any privileges to control ACL". In the DataBase I gave the following privileges to admin username admin domain mantragroup.com priv_name change_privileges priv_value 1 username admin domain mantragroup.com priv_name change_privileges priv_value 1 Someone please help me and guide me to inderstand where I am going wrong. Thanks, Manoj

2 1

Re: [Users] How to check Displayname in From Tag
by Bogdan-Andrei Iancu 30 Jan '06

30 Jan '06

martin(a)campus-merseburg.de wrote: >Am 30.1.2006 schrieb "Bogdan-Andrei Iancu" <bogdan(a)voice-system.ro>: > > > > >>>>regarding the pseudo variables: have you tried the $fn? >>>> >>>> >>>in the 1.0.0 cvs tree it is null. >>> >>> >>indeed, $fn is not present in 1.0.0 - only on 1.1.0 >> >> > >avp_printf("i:20", "$hdr(from[0])"); >if (avp_check("i:20", "re/Anonymous.*/g")) >{ > uac_replace_from("Anonymous","sip:Anonymous@Anonymous"); >} > >is working fine for me. :) > > ok great ! :) regards, bogdan

1 0

[Serusers] Using MySQL as routing backend
by Florian Fainelli 30 Jan '06

30 Jan '06

Hello all ! I would like to use my MySQL database as backend to store routing rules, is it posible or should I add this functionnality myself ? Thank you very much for your answer. -- Cordialement, Florian Fainelli --------------------------------------- 5, rue Charles Fourier Chambre 1511 91011 Evry http://www.alphacore.net ---------------------------------------- Institut National des TÃ©lÃ©comnications http://www.int-evry.fr/telecomint ----------------------------------------

2 1

[Users] FEATURE: new enum extensions
by Bogdan-Andrei Iancu 30 Jan '06

30 Jan '06

Hi, thanks to Greg's patch, new functionalities are available in the enum module. Here is the initial description of the patch: <quote> is_from_user_enum: I find this extremely useful. This function uses the 'From' uri and does a enum lookup based on that number. A true or false is returned. I route to my customers using enum, so when the customer does an outbound call i can use this to verify that they are indeed a valid customer before routing the call. enum_fquery: Carry the previous function another step. You can do 'source routing' of a call using the from address. I do a enum lookup based on the from number, and then apply the ruri to number to the naptr that was found. Using this technique you can 'source route' individual numbers. In this way, for example, customer A can elect (pay for) routing out 2 or 3 gateways, while customer B elects to just have one provider. Another interesting feature is to temporarily reroute a customer to an intercept stating that his account is suspended. </quote> for details, please refer to the on-line documentation: http://openser.org/docs/modules/1.1.x/enum.html regards, bogdan

1 0

[Users] tcp_read error
by Klaus Darilion 30 Jan '06

30 Jan '06

Hi! I have the following problem openser1 <-----TLS----> openser2 -----INVITE------------> <------403-------------- openser1 does not like the 403 response (generated with sl_send_reply()). tcp_read_headers: ERROR: no clen, p=A ERROR: tcp_read_req: bad request, state=4, error=4 buf: SIP/2.0 403 do not reuse this TLS connection ^M Via: SIP/2.0/TLS 10.10.0.41:5063;branch=z9hG4bK2ac3.a346b0f3.0^M Via: SIP/2.0/UDP 10.10.0.41;branch=z9hG4bK2ac3.54290e8.0^M Via: SIP/2.0/UDP 10.10.0.205:7836;branch=z9hG4bK-d87543-4f7f2765412a2d12-1--d87543-;rport=7836^M To: <sip:+43201@proxy1.ienum.labs.nic.at>;tag=10ad1f9ed36b57d334f8c84f393c0304.7a7e^M From: <sip:user101@proxy1.ienum.labs.nic.at>;tag=ca36697e^M Call-ID: 6c71b8428247c758@d2lnd2Ft^M CSeq: 2 INVITE^M server_header: BP 2^M Content-Length: 0^M Warning: 392 10.10.0.42:5063 "Noisy feedback tells: pid=2661 req_src_ip=10.10.0.41 req_src_port=32918 in_uri=sip:+43201@bp2.ienum.labs.nic.at;transport=tls out_uri=sip:+43201@bp2.ienum.labs.nic.at;transport=tls via_cnt==3"^M ^M parsed: SIP/2.0 403 do not reuse this TLS connection ^M Does someone knows this problem? If not, I will debug next week :-( regards klaus

2 3

[Serusers] radius and md5
by Zoran Milic 30 Jan '06

30 Jan '06

Hi, Is it possible to use radius WITHOUT MD5 hashing? Instead, I wish to send user and pass as plain text. Zoran

2 4

[Serusers] SIP gateways authentication
by Viktor Rachovic 30 Jan '06

30 Jan '06

Hello all, we started using ISDN-SIP gateways where you can specify one digest-id in authentification header for multiple caller ids. Now, there seems to be a problem with security if you don't bind caller-id with ip address or domain. As we cannot know from wich IP (domain) our call will come from we cannot bind it to certain IP (domain). There should be some way to specify which caller-ids (usernames) should relate to auth-id. Similar direction i found in ser admin's guide: "There may be a need for a more complex relationship between From/To username and digest id. For example, providers with an established user/password database may wish to keep using it, whereas permitting users to claim some telephone numbers in From. To address such needs generally, there needs to be a 1:N mapping between digest id and all usernames that are acceptable for it. This is being addressed in a newly contributed module "domain", which also addresses more generally issues of domain matching for multidomain scenarios." Still, i didn't found how how could i use domain module for this. What approach other people use in such case? Viktor

1 0

RE: [Serusers] Unable to configure serweb-0.9.4
by Bart J. Smit 30 Jan '06

30 Jan '06

Hi Manoj, I downloaded the same tarball but I followed the directions from the INSTALL file within (these are usually more up to date) This is what I did on CentOS 4.2: Start with setting your SIP_DOMAIN environment variable in ~/.bash_profile: SIP_DOMAIN=your.server.domain.com export SIP_DOMAIN I found that many ser utilities (serctl particularly) expect this to be set. Extract the tarball under /var/www/ and add a link to serweb for easy upgrading to later versions: cd /var/www tar xzf serweb-0.9.4.tar.gz ln -s serweb-0.9.4 serweb Create /etc/httpd/conf.d/serweb.conf to avoid messing up your main apache conf file: Alias /serweb /var/www/serweb/html <Location /serweb> Order deny, allow Deny from all Allow from 127.0.0.1 Allow from 192.168.0.0/16 </Location> Edit /var/www/serweb/config/config_data_layer.php and set your MySQL credentials for admin as you entered them with the ser_mysql.sh script. If you don't know what they are set to, execute this query: 'select username, domain, password from ser.subscriber' in your favourite MySQL client. If you used the default ser/heslo, then these are set correctly in the PHP file already. Confirm that the domain is the same as your SIP_DOMAIN variable. The /var/www/serweb/config/config_paths.php is where you set the root path. You need to change this if you set a path in serweb.conf that is different from /serweb. You will have to have a good read through /var/www/serweb/config/config.php to see if you want any of the stuff mentioned there. I didn't need to change it. (re)start apache and browse to http://your.server.domain.com/serweb/admin. If you enter admin it will automagically be expanded to admin(a)your.sip.domain and you can enter the password as heslo (or whatever you changed it to). Hope this helps, Bart... -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of mkumar(a)mantragroup.com Sent: 30 January 2006 11:37 To: serusers(a)lists.iptel.org Subject: [Serusers] Unable to configure serweb-0.9.4 Hi All, I downloaded serweb from ftp://ftp.berlios.de/pub/ser/0.9.4/contrib/serweb-0.9.4.tar.gz and I am trying to install serweb from instructions given in http://www.iptel.org/ser/doc/ser-howto/ser-Howto.html#AEN256. I executed tar -xvzf serweb-0.9.4.tar.gz and afterwards as per the instructions my next step is to modify /user_interface/prepend.php, but I could not find user_interface directory itself, I searched for it and I could not find it in my UNIX box itself. Is there any bug with serweb-0.9.4.tar.gz or we do not need user_interface directory itself for serweb-0.9.4? Can someone please help me and guide me to configure serweb-0.9.4. Thanks, Manoj. _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

Tr: RE: [Serusers] failover again VRRP
by hgaillac-sip＠yahoo.fr 30 Jan '06

30 Jan '06

Remarque : message transféré en pièce jointe. ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

1 0

[Serusers] Unable to configure serweb-0.9.4
by mkumar＠mantragroup.com 30 Jan '06

30 Jan '06

Hi All, I downloaded serweb from ftp://ftp.berlios.de/pub/ser/0.9.4/contrib/serweb-0.9.4.tar.gz and I am trying to install serweb from instructions given in http://www.iptel.org/ser/doc/ser-howto/ser-Howto.html#AEN256. I executed tar -xvzf serweb-0.9.4.tar.gz and afterwards as per the instructions my next step is to modify /user_interface/prepend.php, but I could not find user_interface directory itself, I searched for it and I could not find it in my UNIX box itself. Is there any bug with serweb-0.9.4.tar.gz or we do not need user_interface directory itself for serweb-0.9.4? Can someone please help me and guide me to configure serweb-0.9.4. Thanks, Manoj.

2 1

[Users] [Fwd: [Serdev] TLS status update]
by Klaus Darilion 30 Jan '06

30 Jan '06

FYI klaus -------- Original Message -------- Subject: [Serdev] TLS status update Date: Mon, 30 Jan 2006 11:50:26 +0100 From: Jan Janak <jan(a)iptel.org> To: serdev(a)iptel.org Hello, here is an interim update on the state of the TLS code: Current status -------------- There is a new module called TLS in the cvs repository. The module attempts to merge the TLS code from experimental cvs module (maintained by Cesc) and Andrei's implementation, to get the best of both. Furthermore the module contains some new functions for script based certificate verification. The TLS module is much more stable than the original implementation. The original (experimental) code works only when there are no simultaneous TLS connections/attempts. The original code would either crash or block SER in seconds under heavy load so it is not really suitable for production use. The code in TLS module is much more stable, it contains several fixes (icluding fix of openssl) and does not seem to crash even under load Unfortunately we discovered a problem during testing of TLS code which can lead to SER being blocked for the duration of tls_send_timeout interval. After that time it would recover. Andrei can probably elaborate more on this issue, but as far as I know there is no easy way to fix it. In conclusion the TLS code in the tls module is more stable than the original code from experimental but is not yet ready for production deployments (unless you can risk SER being blocked for some time). Installation ------------ To use the module you need to patch SER with tls-core.patch. The patch is in tls module directory. After applying the patch compile SER with TLS=1 appended to the make command. This is a temporary inconvenience, we are working on eliminating the need for the patch. TLS select functions -------------------- The current HEAD SER version contains a mechanism which we internaly call the "select" framework. The select framework makes it possible to register functions from modules which can extract pieces of information from the SIP message, transports or IP datagrams. Select functions are denoted by @ character in the configuration file. The name of the select consists of tokens (optionally followed by a number in square brackes) and denoted by ., for example: @contact[2].uri.params.transport Would return the value of transport URI parameter from the second Contact URI in the SIP message. TLS module makes use of this mechanism and contains several functions that can be used to retrieve TLS related information in the configuration script. Currently implemented are the following functions: @tls -- String description of the TLS layer @tls.version -- Protocol version being used @tls.desc -- The same as @tls @tls.cipher -- Cipher name being used @tls.cipher.bits -- Number of bits used for encryption @tls.peer -- Peer certificate subject common name @tls.me -- Local certificate subject common name @tls.peer.subject -- same as @tls.peer @tls.peer.issuer -- Peer certificate issuer common name @tls.peer.verified -- True if peer cert has been verified @tls.{peer,my}.version -- Peer/local certificate version @tls.{peer,my}.sn -- Peer/local certificate number @tls.{peer,my}.not_before -- Certificate validity start @tls.{peer,my}.not_after -- Certificate validity end @tls.{peer,my}.email -- Email address from subj alternative name @tls.{peer,my}.host -- DNS anme from subj alternative name @tls.{peer,my}.uri -- URI from subj alternative name @tls.{peer,my}.ip -- IP address from subj alternative name @tls.{peer,my}.{subj,issuer}.locality -- locality component @tls.{peer,my}.{subj,issuer}.country -- Issuer/subject country @tls.{peer,my}.{subj,issuer}.state - Issuer/subject state @tls.{peer,my}.{subj,issuer}.organization -- Subject/issuer organization @tls.{peer,my}.{subj,issuer}.unit - Subject/issuer organizational unit I will send details description of the functions in a separate email. SSL certificate based authentication ------------------------------------ The functions can be used to implement SSL certificate based authentication in the configuration script: if (proto == TLS && @tls.peer != "Bob") { sl_reply("403", "Forbidden"); break; } This will reject any request received through TLS unless the client provides a client certificate which has subject common name set to "Bob". Select @tls is useful for storing information about the TLS connection (the string can be stored in accounting table, for example, appended to the outgoing message as a header, and so on). Multi-domain configuration -------------------------- The original (experimental) code contained the support for multi-domain configuration. This mechanism was extended a little bit in the tls module. In addition to server based TLS domains there is also preliminary support for TLS client domains. TLS client domains are used when SER attempts to connect to a remote party using TLS. This makes it possible to configure different TLS client certificates for different destinations. In addition to that almost all TLS parameters can be now set on per-domain basis. This includes: - Certificate file - Private key file - Enable/disable certificate verification - Certificate verification depth - CA file - Request/do not request certificate from peer - Cipher list - TLS method TLS domains are identified using IP and port, this is the destination IP and port of the TLS connections, in server TLS domains this must be one of the IP and port configured through listen directives. In TLS client domains this is the IP and port of the remote peer. There is always one default server domain and one default client domain. These are used when no matching server/client domain can be found for IP/port pair. TLS domains are configured through modparams, most tls modparams accept the parameter value with the following syntax: <ip>:<port>=<value. If no IP and port has been specified then the parameters of the default domain are set: modparam("tls", "method", "1.2.3.4:5090=SSLv23") modparam("tls", "verify_certificate", "1") modparam("tls", "require_certificate", "0") modparam("tls", "private_key", "/usr/local/etc/ser/alice_prik.key") modparam("tls", "private_key", "1.2.3.4:5090=key.pem") modparam("tls", "ca_list", "/usr/local/etc/ser/rootca.cert.pem") modparam("tls", "certificate", "/usr/local/etc/ser/alice_cert.crt") modparam("tls", "handshake_timeout", 120) modparam("tls", "send_timeout", 120) modparam("tls", "tls_log", 2) This configuration is somewhat suboptimal and may change yet. Currently there is no possibility to configure TLS client domains (it will be added soon). Example configuration --------------------- Here is a configuration example which demonstrates a couple of interesting things that can be done with the TLS module, XMLRPC management interface and new serctl tools. This configuration file snippet makes it possible to run serctl remotely. It uses the XMLRPC procotol which is encrypted using TLS. Futhermore TLS certificate authentication is used so only people having a correct client certificate are allowed to execute the kill command (this command will stop ser): loadmodule "./modules/tls/tls.so" loadmodule "./modules/sl/sl.so" loadmodule "./modules/xmlrpc/xmlrpc.so" listen=tls:1.2.3.4:5061 modparam("tls", "method", "SSLv23") modparam("tls", "verify_certificate", "1") modparam("tls", "require_certificate", "0") modparam("tls", "private_key", "/usr/local/etc/ser/alice_prik.key") modparam("tls", "ca_list", "/usr/local/etc/ser/rootca.cert.pem") modparam("tls", "certificate", "/usr/local/etc/ser/alice_cert.crt") route { if (proto == TLS && (method == "POST" || method == "GET")) { create_via(); # XMLRPC requests do not contain via, create it if (!(a)tls.peer.verified) { # Client did not provide certificate or it is not valid xmlrpc_reply("400", "Unauthorized"); break; } if (@xmlrpc.method == "core.kill") { # Make sure the client has the permission to execute the command if (@tls.peer != "SER-Killer") { xmlrpc_reply("400", "Access to core.kill denied"); break; } } dispatch_rpc(); break; } } To be done ---------- There is still few things that are not yet done: - @Cesc: Do we need the function set as verify_callback ? As far as I know there is an internal representation in openssl. If we need it then where do we get the verify_depth from (it can be different in different domains) - @Cesc: A couple of openssl parameters have different values, could you check that nothing important is missing ? - Currently only keys without passhprase would work, because SER has no controlling terminal anymore when executing mod_init and child_init module functions (this is where private keys are loaded) Is this feature or bug ? Should we have support for private keys protected by passphrase (requires human intervention when starting ser) - The modparam syntax needs changes to support client domains - TLS can block SER - Support for revocation lists - Should we generate SER CA in a script (possibly called from package pre-init scripts) ? How do we make management of thousands of client certificates easy ? Jan. _______________________________________________ Serdev mailing list Serdev(a)iptel.org http://mail.iptel.org/mailman/listinfo/serdev

1 0

Re: [Serusers] Registering 1 SIP phone with multiple VoIP controllers
by Peter Hudec 30 Jan '06

30 Jan '06

Hi Suvendu, yes, it works, but what about the autorization ? In your case any softphone can register wich your proxy. In real world it is a bad idea. I tired to forward the REGISTER request only it the client is correctly authorized. -- cut --- if (is_method("REGISTER")) { xlog("L_INFO", "REGISTER <$rm> r-uri <$ru>\n"); if (!www_authorize("voip.globaltel.sk", "subscriber")) { www_challenge("voip.globaltel.sk", "1"); return; } if (!save("location")) { sl_reply_error(); return; } # t_relay_to_udp("10.1.1.115", "5060"); t_replicate_udp("10.1.1.115", "5060"); return; } else { --- cut --- To the server 10.1.1.115 is send REGOSTER REQ with full Digest informations. The 10.1.1.115 reject this with "401 unanuthorized". Can any one tell me how to redistibute REGISTER REQ (successfull) accross network? I can't share SER database or have *(realtime). I have to do it only using SIP signaling. best regards hudecof Suvendu Sethi wrote: > Hi Jose, > > IP phone also allows you to configure for multiple > VoIP controller servers but what I wanted was > configuring 1 channel for multiple servers. > > It works fine now with the code which I pasted > earlier. > > Cheers, > Suvendu > > > --- Voipers Portugal <voipers(a)gmail.com> wrote: > > >>Not sure about that IP phone, but a Softphone allows >>you to add multiple >>server, so you can register with as many as you wish >>(and the softphone >>supports). >> >>Jose Simoes >> >> >>On 1/17/06, Suvendu Sethi <suvenduji(a)yahoo.co.in> >>wrote: >> >>>Hi All, >>> >>>Is it possible to register 1 IP phone with >> >>multiple >> >>>VoIP controllers? >>> >>>I want to configure 1 port on Cisco 7940 phone to >>>register with SER proxy server and then forward >> >>the >> >>>registration request to 2 other Aspect VoIP >>>controllers for registration. >>> >>>I am running SER 0.8.14 on linux. >>> >>>I tried the following option but it is registering >>>with only 1 VoIP controller. >>> >>>if (method=="REGISTER") >>>{ >>>save("location"); >>>rewritehost("10.248.13.77"); >>>forward(10,248,13,77,5060); >>>rewritehost("10.248.13.66"); >>>forward(10,248,13,66,5060); >>>break; >>>}; >>> >>>Can you help me on this please? >>> >>>Many thanks in advane. >>> >>>Cheers, >>>Suvenduji >>> >>> >>>Send instant messages to your online friends >> >>http://in.messenger.yahoo.com >> >>>_______________________________________________ >>>Serusers mailing list >>>serusers(a)lists.iptel.org >>>http://lists.iptel.org/mailman/listinfo/serusers >>> >> > > > Send instant messages to your online friends http://in.messenger.yahoo.com > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > -- "The box said 'Requires Windows 95, NT, or better,' so I installed Linux."

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[Users] FW: [Devel] Rtpproxy coredumps :(
by Elton Machado 30 Jan '06

30 Jan '06

Hi as sugested here it goes ;) Sorry for doubling posting. Elton Machado wrote: Hi, I'm getting a problem with nathelper and rtpproxy, somethimes happens that rtpproxy core dumps, and I'm trying to understand why. Maybe I have some mistake in my nathelper configuration with the force rtp proxy. I'm using the cvs version of rtpproxy. The result of dumps is: GNU gdb 6.1.1 [FreeBSD] Copyright 2004 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type "show copying" to see the conditions. There is absolutely no warranty for GDB. Type "show warranty" for details. This GDB was configured as "i386-marcel-freebsd"... Core was generated by `rtpproxy'. Program terminated with signal 11, Segmentation fault. Reading symbols from /lib/libc.so.6...done. Loaded symbols for /lib/libc.so.6 Reading symbols from /libexec/ld-elf.so.1...done. Loaded symbols for /libexec/ld-elf.so.1 #0 ishostseq (ia1=0xbfbfa3f0, ia2=0x0) at rtpp_util.c:48 48 if (ia1->sa_family != ia2->sa_family) Another thing how is possible from ser or openser to control, in case of rtproxy borked out I realive the process again? Any help will be appreciated. Regards, Elton _______________________________________________ > Devel mailing list > Devel(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/devel > > -----Mensagem original----- De: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Enviada: segunda-feira, 30 de Janeiro de 2006 13:19 Para: Elton Machado Cc: 'devel' Assunto: Re: [Devel] Rtpproxy coredumps :( Hi Elton! You should post your bugreport also to the ser users list. This is were the developer (Maxim) is active. regards klaus

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[Serusers] FW: [Devel] Rtpproxy coredumps :(
by Elton Machado 30 Jan '06

30 Jan '06

Hi as sugested here it goes ;) Sorry for doubling posting. Elton Machado wrote: Hi, I'm getting a problem with nathelper and rtpproxy, somethimes happens that rtpproxy core dumps, and I'm trying to understand why. Maybe I have some mistake in my nathelper configuration with the force rtp proxy. I'm using the cvs version of rtpproxy. The result of dumps is: GNU gdb 6.1.1 [FreeBSD] Copyright 2004 Free Software Foundation, Inc. GDB is free software, covered by the GNU General Public License, and you are welcome to change it and/or distribute copies of it under certain conditions. Type "show copying" to see the conditions. There is absolutely no warranty for GDB. Type "show warranty" for details. This GDB was configured as "i386-marcel-freebsd"... Core was generated by `rtpproxy'. Program terminated with signal 11, Segmentation fault. Reading symbols from /lib/libc.so.6...done. Loaded symbols for /lib/libc.so.6 Reading symbols from /libexec/ld-elf.so.1...done. Loaded symbols for /libexec/ld-elf.so.1 #0 ishostseq (ia1=0xbfbfa3f0, ia2=0x0) at rtpp_util.c:48 48 if (ia1->sa_family != ia2->sa_family) Another thing how is possible from ser or openser to control, in case of rtproxy borked out I realive the process again? Any help will be appreciated. Regards, Elton _______________________________________________ > Devel mailing list > Devel(a)openser.org > http://openser.org/cgi-bin/mailman/listinfo/devel > > -----Mensagem original----- De: Klaus Darilion [mailto:klaus.mailinglists@pernau.at] Enviada: segunda-feira, 30 de Janeiro de 2006 13:19 Para: Elton Machado Cc: 'devel' Assunto: Re: [Devel] Rtpproxy coredumps :( Hi Elton! You should post your bugreport also to the ser users list. This is were the developer (Maxim) is active. regards klaus

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[Serusers] Sip-Rpid?
by miliczo＠sezampro.yu 30 Jan '06

30 Jan '06

Hi, In "SIP Express Router RADIUS HOWTO" by Jan Janak I've read that by adding the Sip-Rpid field in my freeradius user file "SIP Express Router can be configured to put the phone number into Remote-Party-ID header field of the SIP message" Well, my question is, can I use the Rpid field and put it in the Caller id field and how.? If anybody has done this or even thinks he has a clue, please help. Thanks

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[Users] Sending stored messages based on time
by Daniel-Constantin Mierla 30 Jan '06

30 Jan '06

Hello, I introduced the reminder functionality submitted by your patch in the development version of msilo. I changed the patch to integrate better with the module and I introduced several other functionalities. Here is the description. There is a new column "snd_time" in the table silo which specifies the time after which the message can be delivered. If the value is 0, then the message is not delivered based on timer. If the message is not successfully delivered at the specified time, then the snd_time is set to 0, to avoid flooding the network, if the destination is not online. It will be delivered first time when he registers. The time for delivery can be set via AVP, the AVP must contain a value in the format YYYYMMDDHHMMSS. The id of the avp can be set via module parameter. Here is a cfg example which can be used to set a reminder service: modparam("msilo","reminder","sip:reminder@test.org") modparam("msilo","send_time",10) modparam("msilo","userid_avp",34) modparam("msilo","snd_time_avp",35) route { .... if(method=="MESSAGE") { if(uri=~"sip:4442[0-9]{13}@") { strip(3); avp_write("$ruri/username", "i:35"); avp_write("$from", "i:34"); m_store("0", "0" /*do not send info message back*/); sl_send_reply("200", "ok"); exit; }; }; ..... } The documentation has been updated: http://openser.org/docs/modules/1.1.x/msilo.html Please note that you have to add the snd_time column to table silo (see the definition in readme) and increase the version to 4 in table 'version'. Testing and feedback are welcome. Cheers, Daniel

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[Serusers] Forking based on URI. Please help.
by Leo 30 Jan '06

30 Jan '06

I am running ser 0.9.4 on FC4 with MySQL and all is working. I want to implement forking based on SIP URI. When someone dials 004888(which is a SIP URI just used for this matching), I want to fork the call to 004990 and 004100 in parallel. So I put this in my ser.cfg: if (uri==myself) { if (uri=~"^sip:004888@~*") { rewriteuri("sip:004990@sip.tcto.net:49300"); append_branch("sip:004100@sip.tcto.net:49200"); route(1); break; }; }; Route[1] { if (!t_relay()) { sl_reply_error(); }; } Can anyone tell me what I am doing wrong? Or point me to some documentation on forking. I have searched ONSIP and module documentation already. Leo Papadopoulos leo(a)ltcjp.com

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Re: [Serusers] SER RTP QoS Monitoring
by Voipers Portugal 30 Jan '06

30 Jan '06

Well, actually, ethereal supports Real Time. There is an option when you start doing the capture that allows to show the packets in real time. Try to look at that. Jose Simoes On 1/30/06, Aisling <ashling.odriscoll(a)cit.ie> wrote: > > Hi Jose, > > > > I looked at ethereal but I don't think it's possible to use that in real > time i.e. you have to start the capture and when you think the call is > over stop the capture and examine the results. I would like something in > real-time…that makes me think tethereal perhaps might be an option…. > > > > Thanks, > > Aisling. > > > > -----Original Message----- > *From:* Voipers Portugal [mailto:voipers@gmail.com] > *Sent:* 30 January 2006 11:47 > *To:* Aisling > *Cc:* John Todd; serusers(a)iptel.org; clona(a)cyberhouse.no > *Subject:* Re: [Serusers] SER RTP QoS Monitoring > > > > Ethereal does that automatically, so I think you can start there. As for > RTPproxy, as far as I remember, it does not support RTCP (at least it didn´t > when I tested a few months ago). > > > > Jose Simoes > > > > On 1/30/06, *Aisling* <ashling.odriscoll(a)cit.ie> wrote: > > > Hi, > > Thanks to John, Nils and Atle for the replies. So if I am to proceed > with extracting QoS info from RTCP it is now my understanding that I > would either have to have my QoS gathering application at: > > 1) the media proxy (used primarily for nat situations) > 2) where the voice is going peer to peer, it would need to be placed at > the end points. However what I'm wondering about here is would it be > possible to have a QoS collector application running on another machine > on the same network as the two endpoints (obviously presuming they're > all on the same network) which could intercept the sender and receiver > reports? And if so does anyone have any idea if there's another way to > distinguish RTCP packets without knowing the IP address or port? > > Many thanks, > Aisling. > > -----Original Message----- > From: John Todd [mailto: jtodd(a)loligo.com] > Sent: 29 January 2006 22:33 > To: Aisling O'Driscoll; serusers(a)iptel.org > Subject: Re: [Serusers] SER RTP QoS Monitoring > > Hello - > Your method #1 (RTCP) is the best solution, it would seem. Method > #2 (INFO) is sort-of supported in some devices, like the Cisco 79xx > series with SIP software - they support insertion of some basic call > element data into the BYE message. Method #3 is complex, as it would > require the SNMP collector to have full knowledge of all call states > on the devices that were queried (though possibly this is not more > complex than the RTCP collection process, either.) In any of these > three cases, SER does not collect the call quality data - the > endpoints or media proxies are the only places that see this data. > If you run a mediaproxy in conjunction with SER, then it may be > possible to collect and report on that data, but there are drawbacks > to a design of that nature. > > For the long term, I would suggest taking a look at this draft, > which describes the use of RTCP-XR (or "vq-rtcpxr") as set of data to > be transmitted via SIP messages between media-handling entities and > other reporting systems: > > > http://www.ietf.org/internet-drafts/draft-johnston-sipping-rtcp-summary- > 08.txt > > While we are still a while away from seeing vendor (or open source) > implementations, this seems like a reasonable proposal. It allows > for SIP-based publication or subscription of call quality events by > various network elements, and it will permit arbitrary querying of > this data at any time interval (not just at the end of the call.) > > JT > > > > >Hi, > > > >I want to be able to extract QoS parameters such as jitter, delay and > >packet loss from VoIP calls made using my SER(both peer to peer for > >my public clients and also for clients where the voice will be routed > >with mediaproxy). I would like to develop something as opposed to > >purchasing a software package. > > > >>From searching the Internet the last few days I have seen that the > >following options seem to be available: > > > >1) RTCP > >RTCP stream will contain this information. The problem is that > >sniffers cannot clearly distinguish RTCP packets from other UDP > >packets unless something like an IP address or specific UDP port is > >supplied. I had considered using TCPDUMP but then where do I position > >this in the network when voice is going peer to peer between client > >and I'm not sure how this can work to monitor all calls when I might > >not be aware of the end user IP addresses. > > > >2) SIP INFO extension > >This also seem to be an option but most off the shelf phones don't > >seem to support this and this would required modification of a SIP ua. > > > >3) SNMP > >I thought maybe SNMP might be an option but the SER snmp module no > >longer exists... > > > >Does anyone have any comments on the above? Are the statements that I > >made correct or can anyone think of other ways to monitor the voice > >QoS? I am trying to understand how commercial applications have > >accomplished this. > > > >Many thanks in advance for advice. > >Aisling. > > > >>---- Original Message ---- > >>From: clona(a)cyberhouse.no > >>To: ashling.odriscoll(a)cit.ie > >>Subject: Re: [Serusers] Status of SER SNMP module > >>Date: Sun, 29 Jan 2006 20:30:02 +0100 > >> > >>> > >>>Hi Aisling, > >>> > >>>The SNMP module is dead as a killed turtle :( > >>> > >>>-Atle > >>>* Aisling O'Driscoll < ashling.odriscoll(a)cit.ie> [060129 20:26]: > >>>> Hello, > >>>> > >>>> I'm just wondering what the current status of the SER SNMP module > >>is? > >>>> Is it currently supported and is there somewhere I can find > >>>> documentation on it? > >>>> > >>>> Many thanks, > > >>> Aisling. > > > >_______________________________________________ > >Serusers mailing list > >Serusers(a)iptel.org > >http://mail.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. > > > _______________________________________________ > Serusers mailing list > Serusers(a)iptel.org > http://mail.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

FW: [Serusers] SER RTP QoS Monitoring
by Aisling 30 Jan '06

30 Jan '06

Hi Jose, I looked at ethereal but I dont think its possible to use that in real time i.e. you have to start the capture and when you think the call is over stop the capture and examine the results. I would like something in real-timethat makes me think tethereal perhaps might be an option. Thanks, Aisling. -----Original Message----- From: Voipers Portugal [mailto:voipers@gmail.com] Sent: 30 January 2006 11:47 To: Aisling Cc: John Todd; serusers(a)lists.iptel.org; clona(a)cyberhouse.no Subject: Re: [Serusers] SER RTP QoS Monitoring Ethereal does that automatically, so I think you can start there. As for RTPproxy, as far as I remember, it does not support RTCP (at least it didn´t when I tested a few months ago). Jose Simoes On 1/30/06, Aisling <ashling.odriscoll(a)cit.ie> wrote: Hi, Thanks to John, Nils and Atle for the replies. So if I am to proceed with extracting QoS info from RTCP it is now my understanding that I would either have to have my QoS gathering application at: 1) the media proxy (used primarily for nat situations) 2) where the voice is going peer to peer, it would need to be placed at the end points. However what I'm wondering about here is would it be possible to have a QoS collector application running on another machine on the same network as the two endpoints (obviously presuming they're all on the same network) which could intercept the sender and receiver reports? And if so does anyone have any idea if there's another way to distinguish RTCP packets without knowing the IP address or port? Many thanks, Aisling. -----Original Message----- From: John Todd [mailto: jtodd(a)loligo.com] Sent: 29 January 2006 22:33 To: Aisling O'Driscoll; serusers(a)lists.iptel.org Subject: Re: [Serusers] SER RTP QoS Monitoring Hello - Your method #1 (RTCP) is the best solution, it would seem. Method #2 (INFO) is sort-of supported in some devices, like the Cisco 79xx series with SIP software - they support insertion of some basic call element data into the BYE message. Method #3 is complex, as it would require the SNMP collector to have full knowledge of all call states on the devices that were queried (though possibly this is not more complex than the RTCP collection process, either.) In any of these three cases, SER does not collect the call quality data - the endpoints or media proxies are the only places that see this data. If you run a mediaproxy in conjunction with SER, then it may be possible to collect and report on that data, but there are drawbacks to a design of that nature. For the long term, I would suggest taking a look at this draft, which describes the use of RTCP-XR (or "vq-rtcpxr") as set of data to be transmitted via SIP messages between media-handling entities and other reporting systems: http://www.ietf.org/internet-drafts/draft-johnston-sipping-rtcp-summary- 08.txt While we are still a while away from seeing vendor (or open source) implementations, this seems like a reasonable proposal. It allows for SIP-based publication or subscription of call quality events by various network elements, and it will permit arbitrary querying of this data at any time interval (not just at the end of the call.) JT >Hi, > >I want to be able to extract QoS parameters such as jitter, delay and >packet loss from VoIP calls made using my SER(both peer to peer for >my public clients and also for clients where the voice will be routed >with mediaproxy). I would like to develop something as opposed to >purchasing a software package. > >>From searching the Internet the last few days I have seen that the >following options seem to be available: > >1) RTCP >RTCP stream will contain this information. The problem is that >sniffers cannot clearly distinguish RTCP packets from other UDP >packets unless something like an IP address or specific UDP port is >supplied. I had considered using TCPDUMP but then where do I position >this in the network when voice is going peer to peer between client >and I'm not sure how this can work to monitor all calls when I might >not be aware of the end user IP addresses. > >2) SIP INFO extension >This also seem to be an option but most off the shelf phones don't >seem to support this and this would required modification of a SIP ua. > >3) SNMP >I thought maybe SNMP might be an option but the SER snmp module no >longer exists... > >Does anyone have any comments on the above? Are the statements that I >made correct or can anyone think of other ways to monitor the voice >QoS? I am trying to understand how commercial applications have >accomplished this. > >Many thanks in advance for advice. >Aisling. > >>---- Original Message ---- >>From: clona(a)cyberhouse.no >>To: ashling.odriscoll(a)cit.ie >>Subject: Re: [Serusers] Status of SER SNMP module >>Date: Sun, 29 Jan 2006 20:30:02 +0100 >> >>> >>>Hi Aisling, >>> >>>The SNMP module is dead as a killed turtle :( >>> >>>-Atle >>>* Aisling O'Driscoll < <mailto:ashling.odriscoll@cit.ie> ashling.odriscoll(a)cit.ie> [060129 20:26]: >>>> Hello, >>>> >>>> I'm just wondering what the current status of the SER SNMP module >>is? >>>> Is it currently supported and is there somewhere I can find >>>> documentation on it? >>>> >>>> Many thanks, > >>> Aisling. > >_______________________________________________ >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. _______________________________________________ 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

[Serusers] Whiskey & Beer for SNMP module
by Thomas Nilsen 30 Jan '06

30 Jan '06

Hello! Two cases of beer or two bottles of whiskey to first developer that release a simple snmp module that provide equal status equal to the info accessible through the fifo. Deadline for first sneak preview is next Friday! :-) - Cheers Thomas

3 3

[Serusers] failover again
by hgaillac-sip＠yahoo.fr 30 Jan '06

30 Jan '06

Hi all, I wish to set ser (outbound proxy) with failover support : SER1 sip agents ======= ======= Asterisk Farm ==sip/pstn gateway SER2 SER1 and SER2 use SQL DB if one of the SERs is down how can we get the other available for the sip agents for example we need to reboot a SER when the ser.cfg has changed . I look at dispatcher module ! Regards Harry ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

3 3

[Serusers] Ser server
by Ofer Zaarur 29 Jan '06

29 Jan '06

Hi All, I'm trying to connect X-lite to ser server that installed as local server. The test configuration is running in local network and the IP is fix. please let me know if you have any idea what is going on. thanks, Ofer log from the Xlite © 2004 Xten Networks, Inc. All rights reserved. X-Lite release 1105d build stamp 99999 License key: 3FA27C9F993AC5240ADFBC4466EBEEA5 Attempting SIP protocol listen on: 192.168.1.45:5060 Attempting SIP protocol listen on: 192.168.1.45:5060 Attempting SIP protocol listen on: 192.168.1.45:5060 Attempting SIP protocol listen on: 192.168.1.45:5060 Attempting SIP protocol listen on: 192.168.1.45:5060 Attempting SIP protocol listen on: 192.168.1.45:5060 Attempting SIP protocol listen on: 192.168.1.45:5060 Attempting SIP protocol listen on: 192.168.1.45:5061 Established SIP protocol listen on: 192.168.1.45:5061 Discovered Single Mapped Port Symmetric NAT Firewall SIP: 192.168.1.45:5061 RTP: 192.168.1.45:8000 NAT: 151.203.64.98 Discovering external SIP port on symmetric firewall... Discovered external SIP port on symmetric firewall: 50402 RECEIVE TIME: 405492948 RECEIVE << 64.69.76.23:3478 PROXY#0: 192.168.1.45:5060 SEND TIME: 405493076 SEND >> 192.168.1.45:5060 REGISTER sip:192.168.1.45 SIP/2.0 v: SIP/2.0/UDP 192.168.1.45:5061;rport;branch=z9hG4bK11776FC28025ABB2C29C1BD89E88795E f: 1234 <sip:1234@192.168.1.45>;tag=1752773568 t: 1234 <sip:1234@192.168.1.45> m: "1234" <sip:1234@192.168.1.45:5061> i: 7F1AD919BE44D24B89E73FAA0668BBDD(a)192.168.1.45 CSeq: 5434 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: X-Lite release 1105d l: 0 RECEIVE TIME: 405493262 RECEIVE << 192.168.1.45:5060 SIP/2.0 483 Too Many Hops v: SIP/2.0/UDP 192.168.1.45:5061;rport=5061;branch=z9hG4bK11776FC28025ABB2C29C1BD89E88795E f: 1234 <sip:1234@192.168.1.45>;tag=1752773568 t: 1234 <sip:1234@192.168.1.45>;tag=b27e1a1d33761e85846fc98f5f3a7e58.bee7 i: 7F1AD919BE44D24B89E73FAA0668BBDD(a)192.168.1.45 CSeq: 5434 REGISTER Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.45:5060 "Noisy feedback tells: pid=12541 req_src_ip=192.168.1.45 req_src_port=5060 in_uri=sip:192.168.1.45 out_uri=sip:192.168.1.45 via_cnt==71" SEND TIME: 405513508 SEND >> 192.168.1.45:5060 REGISTER sip:192.168.1.45 SIP/2.0 v: SIP/2.0/UDP 192.168.1.45:5061;rport;branch=z9hG4bK705C4BCDC57F933D959A1C991F8A4DB0 f: 1234 <sip:1234@192.168.1.45>;tag=1752773568 t: 1234 <sip:1234@192.168.1.45> m: "1234" <sip:1234@192.168.1.45:5061> i: 7F1AD919BE44D24B89E73FAA0668BBDD(a)192.168.1.45 CSeq: 5435 REGISTER Expires: 1800 Max-Forwards: 70 User-Agent: X-Lite release 1105d l: 0 RECEIVE TIME: 405513686 RECEIVE << 192.168.1.45:5060 SIP/2.0 483 Too Many Hops v: SIP/2.0/UDP 192.168.1.45:5061;rport=5061;branch=z9hG4bK705C4BCDC57F933D959A1C991F8A4DB0 f: 1234 <sip:1234@192.168.1.45>;tag=1752773568 t: 1234 <sip:1234@192.168.1.45>;tag=b27e1a1d33761e85846fc98f5f3a7e58.6822 i: 7F1AD919BE44D24B89E73FAA0668BBDD(a)192.168.1.45 CSeq: 5435 REGISTER Server: Sip EXpress router (0.8.12 (i386/linux)) Content-Length: 0 Warning: 392 192.168.1.45:5060 "Noisy feedback tells: pid=12541 req_src_ip=192.168.1.45 req_src_port=5060 in_uri=sip:192.168.1.45 out_uri=sip:192.168.1.45 via_cnt==71"

1 0

[Serusers] rtpproxy
by Toygun Mavinil 29 Jan '06

29 Jan '06

Hi there, I try to use ser 0.9.6 on fc4 with mysql rtpproxy because of nat No errors starting ser On local, no problen betweeb ip phones and gateways but one way voice problen on sjphone On wan, always voice goes one way, caller can nor hear but callee can hear and can not talk FC4 gets wan ip directly, all firewalls are off. I tried with mediaproxy module, nothing changed, Here is ser.cfg (from configs.GettingStarted); # $Id: nat-rtpproxy.cfg 9 2005-08-19 debug=3 fork=yes log_stderror=no listen=wan ip port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("FQDN","subscriber")) { proxy_challenge("FQDN","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } ------------ Toygun

1 0

[Serusers] 2 Problems with SER
by mkumar＠mantragroup.com 29 Jan '06

29 Jan '06

I was able to successfully install SER and I can make calls. But I have two problems with it, they are 1) Even unregistered Users are able to call out. 2) We have few computers connected via LAN and we are connected to a router. We have a UNIX box somewhere else on which we installed SER. My problem is from computers which have indiviual IP are able to make and receive calls but in my internal network we are having one way problems or sometimes we are unable to make or recieve calls. I am attaching my ser.cfg, please check it and help me to solve my problems. Thanks, Manoj.

3 3

Re: [Serusers] Status of SER SNMP module
by Aisling O'Driscoll 29 Jan '06

29 Jan '06

:( Ok thanks for letting me know. ---- Original Message ---- From: clona(a)cyberhouse.no To: ashling.odriscoll(a)cit.ie Subject: Re: [Serusers] Status of SER SNMP module Date: Sun, 29 Jan 2006 20:30:02 +0100 > >Hi Aisling, > >The SNMP module is dead as a killed turtle :( > >-Atle >* Aisling O'Driscoll <ashling.odriscoll(a)cit.ie> [060129 20:26]: >> Hello, >> >> I'm just wondering what the current status of the SER SNMP module >is? >> Is it currently supported and is there somewhere I can find >> documentation on it? >> >> 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)iptel.org >> http://mail.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

[Serusers] Status of SER SNMP module
by Aisling O'Driscoll 29 Jan '06

29 Jan '06

Hello, I'm just wondering what the current status of the SER SNMP module is? Is it currently supported and is there somewhere I can find documentation on it? 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.

2 1

[Users] REGISTER and location/username
by Tyler 29 Jan '06

29 Jan '06

Hello list, How can I make it so when a client registers, their auth name is what gets stored in the username field of the location table, and not the 'display name' from the From field? I've tried to do some regexp stuff with subst command but it seems like the save() function doesn't pay any attention to the changes I make. Any suggestions? Thanks tf.

2 1

[Serusers] Hunt groups?
by Chris Roberts 28 Jan '06

28 Jan '06

Heya, I've seen very little discussion searching the archive for hunt groups coded in SER. There seems to be a way of providing this functionality through the lookup("") function, but I'm not entirely clear how to use this and whether I need some kind of serial fork etc? Does anyone have any examples of hunt groups implemented with SER? Would I be better to use lookup("") with Q values or maybe code this myself in AVPops...? How have other people handled this? Cheers, Chris. -- |=========----- -------=======| | Chris Roberts (croberts(a)bongle.co.uk) | |=======------- -----=========|

1 0

[Serusers] t_relay vs forward ?
by Corey 28 Jan '06

28 Jan '06

What exactly is the difference between 't_relay' and 'forward'? If I'm trying to route/forward sip-to-pstn-gateway calls, should I be using 't_relay' or 'forward' after I 'rewritehost'?

1 0

[Serusers] rtpproxy
by Toygun Mavinil 28 Jan '06

28 Jan '06

Hi there, I try to use ser 0.9.6 on fc4 with mysql rtpproxy because of nat No errors starting ser On local, no problen betweeb ip phones and gateways but one way voice problen on sjphone On wan, always voice goes one way, caller can nor hear but callee can hear and can not talk FC4 gets wan ip directly, all firewalls are off. I tried with mediaproxy module, nothing changed, Here is ser.cfg (from configs.GettingStarted); # $Id: nat-rtpproxy.cfg 9 2005-08-19 debug=3 fork=yes log_stderror=no listen=wan ip port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/auth.so" loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" modparam("auth_db|uri_db|usrloc", "db_url", "mysql://ser:heslo@localhost/ser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) route { # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { sl_send_reply("483", "Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message Overflow"); break; }; # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method!="REGISTER") { record_route(); }; if (method=="BYE" || method=="CANCEL") { unforce_rtp_proxy(); } # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { if (has_totag() && (method=="INVITE" || method=="ACK")) { if (nat_uac_test("19")) { setflag(6); force_rport(); fix_nated_contact(); }; force_rtp_proxy("l"); }; route(1); break; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(4); route(1); break; }; if (method=="CANCEL") { route(1); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(1); } route[1] { # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); }; } route[2] { # ----------------------------------------------------------------- # REGISTER Message Handler # ---------------------------------------------------------------- if (!search("^Contact:[ ]*\*") && nat_uac_test("19")) { setflag(6); fix_nated_register(); force_rport(); }; sl_send_reply("100", "Trying"); if (!www_authorize("","subscriber")) { www_challenge("","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- if (!proxy_authorize("FQDN","subscriber")) { proxy_challenge("FQDN","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); if (nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (uri!=myself) { route(4); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; route(4); route(1); } route[4] { # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; } onreply_route[1] { if (isflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if (!search("^Content-Length:[ ]*0")) { force_rtp_proxy(); }; }; if (nat_uac_test("1")) { fix_nated_contact(); }; } ------------ Toygun

1 0

[Users] Mysql connection timeout issue
by Marcello Lupo 28 Jan '06

28 Jan '06

Hi to all, i'm experiencing strange behavior and want to ask a clue. After some time of inactivity (around 15 hour more or less) or few activity on my Openser i start getting message in my logs: submit_query: MySQL server has gone away db_query: Error while submitting query is_local(): Error while querying database and obviously openser is no more working. I have to restart openser to have reopened the connection. I'm using mysql 5.0.15-max and was thinking on the wait_timeout or interactive_timeout variables in mysql that by default are at 28800 (8 hour). I raised both variables to 1000000 but still getting the problem. Did you have any ideas? Thanks, Bye, Marcello

3 4

Re: [Serusers] TLS support in SER
by Klaus Darilion 28 Jan '06

28 Jan '06

Yi Zheng wrote: > Hi, > > I am wondering whether SER has any support for TLS as a security > mechanism? Thanks, Yes. It is in the experimental tree: http://cvs.berlios.de/cgi-bin/viewcvs.cgi/ser/experimental/tls/ You can also try openser, which has TLS integrated in the stable version 1.0.0 regards klaus

5 9

[Serusers] group call pickup
by Joao Pereira 28 Jan '06

28 Jan '06

Hello have anyone implemented Group Call PickUp in SER? I have a SER / Asterisk setup and I only found documentation for Asterisk. But my SIP - SIP calls dont pass trough Asterisk... It need this to be done by SER. Can this be done in SER? Thanks Joao Pereira www.fccn.pt

2 2

RE: [Serusers] Monitoring
by hgaillac-sip＠yahoo.fr 28 Jan '06

28 Jan '06

Hello monigroup, Does Monit provide monitoring for Sip Express Router (www.iptel.org) ? Regards harry --- Frank Fischer <frank.fischer(a)digitalnomads.ch> a écrit : > > Just take a look at it. > It's a general monitoring tool to manage and monitor > processes under linux, > no matter what app it is. > > > > -----Original Message----- > > From: hgaillac-sip(a)yahoo.fr > [mailto:hgaillac-sip@yahoo.fr] > > Sent: Friday, January 27, 2006 2:19 PM > > To: Frank Fischer > > Subject: RE: [Serusers] Monitoring > > > > Does it support SER ? > > --- Frank Fischer <frank.fischer(a)digitalnomads.ch> > a > > écrit : > > > > > > > > I dunno if snmp is supported (i guess not), but > have > > > a look at monit > > > http://www.tildeslash.com/monit/ > > > > > > > > > > -----Original Message----- > > > > From: serusers-bounces(a)lists.iptel.org > > > > [mailto:serusers-bounces@lists.iptel.org] On Behalf > Of > > > hgaillac-sip(a)yahoo.fr > > > > Sent: Friday, January 27, 2006 1:04 PM > > > > To: serusers(a)lists.iptel.org > > > > Cc: asterisk-users(a)lists.digium.com > > > > Subject: [Serusers] Monitoring > > > > > > > > Hi asterisk and ser users, > > > > > > > > Is there a solution to monitor asterisk and > ser > > > with > > > > snmp ? > > > > > > > > Regards > > > > Harry > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ______________________________________________________________ > > > > _____________ > > > > Nouveau : téléphonez moins cher avec Yahoo! > > > Messenger ! > > > > Découvez les tarifs exceptionnels pour appeler > la > > > France et > > > > l'international. > > > > Téléchargez sur http://fr.messenger.yahoo.com > > > > > > > > > _______________________________________________ > > > > Serusers mailing list > > > > serusers(a)lists.iptel.org > > > > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > > > > > > > > > > > > > > > > > > > > > > > > > > ______________________________________________________________ > > _____________ > > Nouveau : téléphonez moins cher avec Yahoo! > Messenger ! > > Découvez les tarifs exceptionnels pour appeler la > France et > > l'international. > > Téléchargez sur http://fr.messenger.yahoo.com > > > > > ___________________________________________________________________________ Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international. Téléchargez sur http://fr.messenger.yahoo.com

2 1

[Serusers] SER / LCR / Dynamic routing / Wondering....
by Jean-Michel Hiver 27 Jan '06

27 Jan '06

Hi List, Currently I use the Perl module Asterisk::LCR to handle the least cost routing. So I basically have SER forward PSTN calls to my Asterisk box, and Asterisk handles the routing... I'm thinking that the LCR part should really be handled by SER. Because, well, it's dubbed the SIP Express ROUTER, so I'm thinking I ought to get it to actually ROUTE stuff, including LCR. And I would really like to keep Asterisk for what it's good at, i.e. voicemail, IVR menus, and AGI. Now, there's a couple of ways I could do this. 1) Do some static routing, i.e. have a script which writes a (really, really big) ser.cfg configuration file. Now I wonder how SER would handle a multi-megabyte configuration file. Another issue is what happen to SIP traffic when you do a ser restart? I would assume, bad things... 2) Do dynamic routing, but I have no idea on how to achieve this with SER. Any tips? Let me know what I can do... I would just *love* to upload a SER::LCR module on CPAN as well :) Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

2 1

[Serusers] syslog+stderr logging
by Ramin Nikaeen 27 Jan '06

27 Jan '06

Hi everyone, Does anyone know if I can tell ser to log to both stderr and syslog at the same time? If there is not, can someone tell me where I need to make the changes to enable such feature? Is there any way of telling ser to toggle between different logging modes, let's say stderr versus syslog, while ser is running without shutdowing and restarting ser?! Thanks ramin

1 0

[Serusers] Postgres database
by Voipers Portugal 27 Jan '06

27 Jan '06

Hi, I am am having a problem with Postgres database in ser 0.9.6. Does anyone knows where i can find the SQL to build the table correctly? Because all I can get is the script for my sql database. Jose Simoes

1 0

[Serusers] serweb -> redirection limit problem
by Zoran Milic 27 Jan '06

27 Jan '06

Hi, I have installed webser, and when I try to start it in my browser (firefox) I get the folowing message: Redirection limit for this URL exceeded. Unable to load requested page. This may be caused by coockies that are blocked. I have double checked my config.php file and my browser for coockies. My Apache server has this in his acces.log GET /serweb/admin/index.php?phplib_Session=db77590f25d30089130cda6e4dcd3092&phplib_Session=5092037156d439be7dd59e7e9ffbc867&phplib_Session=8441b33d9f333f90c6495368984a7ff6&phplib_Session=3a6e115edce6ca7c65b5f492e6bdbcd0&phplib_Session=79120eb3b77d090a58b9f0172c696f9d&phplib_Session=9b5f244b6bfe537ce2bd59e711f0bfe6&phplib_Session=51c79bdf3e623b85c8590611512664a3&phplib_Session=cbd7daf25a83af79d9660e16ef4fba8a&phplib_Session=3e1876e770613ca251a9afb32094f90f&phplib_Session=08ffe0736fdb43dab0f5c1b83e443e81&phplib_Session=570815505da88a372144254079a30960&phplib_Session=2b45a276e228e1ec638ec3b81d503056&phplib_Session=c908736f3c02bffd27c1078e522111ee&phplib_Session=87ea8cdd7634368189d6fb2b3d0497a3&phplib_Session=5c949caf202c02ab2a1207098b94cc60&phplib_Session=74cdea68be840e7e51f4293040f68e85&phplib_Session=16a2121208f6198b073f3d4ba326ec72&phplib_Session=399f0e1900dfb75674fcea4584a0f10d&phplib_Session=c6bcc733b60c6275b3ddfa403d946cba&phplib_Session=096eccdaaa2bf18409b471a29d795b18 HTTP/1.1" 302 I am using php 5.1.2. Any ideas? best regards, Zoran

2 4

[Serusers] Starting SER with user rights
by Ladislav Andel 27 Jan '06

27 Jan '06

Hi, i would like to start and restart SER with common user rights. if I do /etc/init.d/ser start as user I get touch: cannot touch `/var/lock/subsys/ser': Permission denied The same question would be for starting rtpproxy. I guess it's through suid but if set suid bit to my ser script (chmod u+s ser) I get the same Permission denied Could you help me please? Thank you Ladislav

1 0

[Serusers] 2 Problems with SER
by mkumar＠mantragroup.com 27 Jan '06

27 Jan '06

Hi All, I was able to successfully install SER and I can make calls. But I have two problems with it, they are 1) Even unregistered Users are able to call out. 2) We have few computers connected via LAN and we are connected to a router. We have a UNIX box somewhere else on which we installed SER. My problem is from computers which have indiviual IP are able to make and receive calls but in my internal network we are having one way problems or sometimes we are unable to make or recieve calls. I am attaching my ser.cfg, please check it and help me to solve my problems. Thanks, Manoj.

2 1

[Serusers] problem with SIP message greater than 1500 bytes
by zaghouani hamdi 27 Jan '06

27 Jan '06

hi, I m using ser-0.8.14. i m trying to send a SIP message from an SER server to another SER server. The length of this message is greater than 1500 Bytes(around 1600 Bytes) , so it is fragmented before being sent. (in my case it's fragmented to 2 packets.) then I check the SIP flow using ethereal I found that : In the IP header of the first packet, the "more fragment" flag is set to "0" so at the reception the second packet is ignored. And we get at the receiver a SIP message which is not completed.( we don t receive all the SIP message). is it SER who force that the sent message being less than 1500 bytes ? thinks a lot Hamdi --------------------------------- Nouveau : téléphonez moins cher avec Yahoo! Messenger ! Découvez les tarifs exceptionnels pour appeler la France et l'international.Téléchargez la version beta.

3 4

[Serusers] Does SER has TLS support
by Praveen Jothi 27 Jan '06

27 Jan '06

Hello all, Does SER has support for TLS ? I see some TLS functions in SER source. kindly enlighten me on this. regards, Pjothi

2 1

[Serusers] onreply_route
by George Lambson 27 Jan '06

27 Jan '06

Hi there, What does the onreply_route section in ser.cfg do? If anyone could elighten me that would be great. Thanks, George

1 0

Re: [Serusers] setflag command ~
by George Lambson 27 Jan '06

27 Jan '06

Nevermind, I figured it out. >>> George Lambson <LambsonGE(a)mtc.byu.edu> 1/26/2006 8:50 PM >>> Hi all, Is the setflag command used only with the acocunting module? I noticed that it is present in the mediaproxy config file from onsip.org. I was wondering if it is necessary? Thanks, George _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Serusers] setflag command ~
by George Lambson 27 Jan '06

27 Jan '06

Hi all, Is the setflag command used only with the acocunting module? I noticed that it is present in the mediaproxy config file from onsip.org. I was wondering if it is necessary? Thanks, George

1 0

[Serusers] Where are my BYE messages gone?
by Jean-Michel Hiver 26 Jan '06

26 Jan '06

Hi again list, I'm trying to get acc module to work so that I can write a billing script for SER (which I'll probably stick on CPAN just like what I've done for Asterisk::LCR). Thanks to Olivier Taylors' help now the module works (well, at least SER starts). But I see no trace of my BYE messages. Which is strange since I do record route (well, I think I do). I have pasted my ser.cfg on http://pastebin.ca/38650 Can you see what's going wrong? I would really like to avoid using Asterisk *just* to be able to bill stuff. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

2 2

[Serusers] Test/Lab environment setup
by Jeroen Berning 26 Jan '06

26 Jan '06

Hello, I am pretty new to all of this, so I hope somebody give me some help. I am trying to get SER working with a Cisco AS5350 and some Cisco SIP phones, can somebody help me with some configs of the AS5350 and how ser.cfg should look like. The SIP phones can already call each other but it is the forward to the as5350 that is giving me problems. I am thinking I am experimenting with to much things that could be wrong, so it would be nice to have some working configs. Regards, Jeroen

1 0

[Serusers] Need CANCEL before failure_route called on branch timeout...
by Phil D'Amore 26 Jan '06

26 Jan '06

Hi all... Sorry this is a bit long, but this problem is driving me nuts... I have a problem where I need to try a sequence of numbers on the PSTN, where the one is tried if the previous one fails or times out. My problem is handling the timeout (408) case: 1. First INVITE sent to PSTN gw 2. final response timer fires, fake 408 generated 3. failure_route in scripts called, new branch created, next INVITE sent to PSTN gw 4. CANCEL is sent (I realize this is not sync'd to step 3 but it typically happens this way that that causes my problem) 5. PSTN GW sends 482 Loop Detected back to me for the *second* branch created in 3. Of course the second branch fails and that number is never tried at that point. I understand from looking at the code in timer.c that this sequence is by design. What I need is for the CANCEL to come out *before* the next INVITE, and I need to be able to reasonably ASSume that the CANCEL was received by the PSTN gw and my invite will not trigger loop detection. I've fielded two different ways of doing this, both of which make me feel uneasy. Both of these involved mucking with final_response_handler in tm/timer.c. After the line: DBG("DEBUG: final_response_handler:stop retr. and send CANCEL (%p)\n", t); I changed the rest of the function first to read as follows (try not to laugh too hard): cancel_branch(t, r_buf->branch); sleep(1); fake_reply(t, r_buf->branch, 408 ); DBG("DEBUG: final_response_handler : done\n"); I'm a little uneasy about scheduling that cancel while the reply is locked, but it *seems* safe. I also didn't like the fact that I was hard-coding the time, I'd rather have it linked a bit more to the gateway processing the CANCEL, so... Then I tried this: if(should_cancel_branch(t, r_buf->branch)) { int timer = 2; DBG("DEBUG: final_response_handler: cancel hack (%p)\n", t); UNLOCK_REPLIES(t); cancel_branch(t, r_buf->branch); fr_tl->timer_list= NULL; /* set to NULL so that set_timer will work */ set_timer( fr_tl, FR_INV_TIMER_LIST, &timer ); } else { DBG("DEBUG: final_response_handler: timeout (%p)\n", t); fake_reply(t, r_buf->branch, 408 ); } DBG("DEBUG: final_response_handler : done\n"); This gives me the link between the final from the cancel being received, and the send of the second invite, but it has the unwanted effect of causing the status when the failure_route script is called to always be 487, so I'd need a way to communicate to scripts that the 487 is really the result of a local cancel triggered by the fr_timer firing. I could theoretically get that by re-running branch picking and checking the uac for that branch for a local cancel, but that doesn't seem clean. Presumably if I put a hack in place like this, it would default to off, and I'd use a flag or something on the message before t_relay()ing the original request to signal to tm that I want this treatment for this transaction, but I digress. What I really need is a way to make everything look normal to the scripts (i.e. t_check_status will still match to a 408), but gets me the CANCEL sequence I need. Any ideas? Am I a bad person for trying the code above? ;) Thanks, Phil __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Serusers] Message to moderator
by Jean-Michel Hiver 26 Jan '06

26 Jan '06

Why did I receive a "Your message to Serusers awaits moderator approval" message? I thought I /was/ subscribed to this list... Cheers, Jean-Michel.

4 3

[Serusers] auth_radius error
by Nhadie 26 Jan '06

26 Jan '06

Hi All, What does this error mean? Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: Maxfwd module- initializing Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: ERROR: auth_radius: can't get code for the Sip-Session attribute value Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: init_mod(): Error while initializing module auth_radius I'm using ser 0.9.5 and radiusclient-ng 0.5.2. Thank You Regards, Nhadie

2 1

[Serusers] SER + ser_acc_db_module
by Jean-Michel Hiver 26 Jan '06

26 Jan '06

Hi, I apt-get installed ser_acc_db_module and tried to configure it. After restarting SER, SER simply doesn't reply to any REGISTER command. It seems to just ignore them so my phones cannot register. And I have no NAT issues either since I don't use NAT with SIP. Here's what I did at the configuration level: Firstly, I have added the following config to load and configure the acc_db module. <bunch of loadmodule statements> # ACC #loadmodule "/usr/lib/ser/modules/acc_db.so" #modparam("acc", "db_url", "sql://ser:heslo@localhost/ser") #modparam("acc", "db_flag", 1) #modparam("acc", "db_missed_flag", 1) #modparam("acc", "failed_transactions", 1) <bunch of modparam statements> Then, just after the "Record route" section and the "Loose route section", I have added this: # ----------------------------------------------------------------------- # Sets the acc accounting flag 4 billing info. # ----------------------------------------------------------------------- if( (method=="INVITE") || (method=="ACK") || (method=="BYE")) { setflag(1); }; Below I have my PSTN hookup, such as: # ----------------------------------------------------------------- # Pass on stuff going to PSTN to Asterisk # ----------------------------------------------------------------- if (src_ip==<trusted_customer_ip>) { if (lookup("location")) { t_relay(); break(); }; rewritehostport ("<asterisk_gw_ip>:5060"); if (!t_relay()) { # sl_send_reply ("403", "prout"); sl_reply_error(); }; break; }; Any ideas of what can be going on? As soon as I comment out acc-db related config and restart SER everything works smoothly again. Cheers, Jean-Michel.

2 1

[Serusers] Call type processing Section problem?
by Juan 26 Jan '06

26 Jan '06

Hi. Till now I was happy using SER handling communications between UAs using nathelper and RPT proxy. Now I changed the ser.cfg script to the one with PSTN support and none of the UAs get registered. I am getting a 403 Forbidden. Analyzing the script I understood that this 403 was coming from the Call Type Processing Section so I changed if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { and so on to if (uri!=myself) and now the UAs are registering again. But of course I can't make calls. So I would like to know how this< is_uri_host_local and is_from _local> work and what should I do to make them allow my UAs to register. Thanks in advance Juan Ferrari

1 0

[Users] typo in TLS tutorial
by Klaus Darilion 26 Jan '06

26 Jan '06

http://www.openser.org/docs/tls.html#AEN343 tls_method="tlsv1" should be (without "") tls_method=tlsv1 regards klaus

2 1

[Serusers] re : Is SER a B2BUA
by Yeung OnTai-q16645 26 Jan '06

26 Jan '06

Hi all, Is SER a B2BUA for IM, voice etc? i.e. If user A sends an IM message(via SIP MESSAGE) to SER, and SER sends the message to user B. Seems like it is not, because the Call-IDs are the same on both call legs.... Is there any special configuration which can turn it into a B2BUA? Regards, Dennis

2 1

[Serusers] auth_radius error
by Nhadie 26 Jan '06

26 Jan '06

Hi All, What does this error mean? Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: Maxfwd module- initializing Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: ERROR: auth_radius: can't get code for the Sip-Session attribute value Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: init_mod(): Error while initializing module auth_radius I'm using ser 0.9.5 and radiusclient-ng 0.5.2. Thank You Regards, Nhadie

1 0

[Serusers] auth_radius error
by Nhadie 26 Jan '06

26 Jan '06

Hi All, What does this error mean? Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: Maxfwd module- initializing Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: ERROR: auth_radius: can't get code for the Sip-Session attribute value Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: init_mod(): Error while initializing module auth_radius I'm using ser 0.9.5 and radiusclient-ng 0.5.2. Thank You Regards, Nhadie

1 0

[Serusers] auth_radius error
by Nhadie 26 Jan '06

26 Jan '06

Hi All, What does this error mean? Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: Maxfwd module- initializing Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: ERROR: auth_radius: can't get code for the Sip-Session attribute value Jan 26 17:42:13 sip01 /usr/local/sbin/ser[23745]: init_mod(): Error while initializing module auth_radius I'm using ser 0.9.5 and radiusclient-ng 0.5.2. Thank You Regards, Nhadie

1 0

[Serusers] INVITE transaction timer vs. 'Proceeding' timer
by Arek Bekiersz 26 Jan '06

26 Jan '06

Hello, This time I've got a question :) I think it will be useful for everybody. I have discussion with SIP device manufacturer about expiration time of call attempt. When call is made from UA it sends INVITE with "Expires: 65". It is controlled by device's timer setting "ring_time_limit". This timer can be set to anything between 10 and 600 seconds, we set it to 65 seconds. Fine. However at the same time it starts a special timer, say "max_response_time_invite". It can be set to anything between 8 and 20 seconds. We set it to 20 sec. Fine so far. CASE 1: Imagine we call from UA to PSTN number, using one of VoIP->PSTN providers. We have more than one provider, so we setup SER with "fr_timer" value of, let's say 25 seconds and "fr_inv_timer" of 60. We prepare failure routes in case when those timers hits in PSTN call. When "fr_timer" hits, we simply reroute to another gateway. Great. SER forwards INVITE from UA to remote PSTN gateway and at the same time sends back "100 Trying" to UA. Fine. Imagine now that it is busy-hour and it takes PSTN provider 21 seconds to send back "180 Ringing" (or "183 Session Progress") and after 8 seconds more remote callee picks up ("200 OK"). Unfortunately device wil send CANCEL to SER, because it hasn't received "180 Ringing" or "183 Session Progress" within the "max_response_time_invite" setting of 20 seconds. It only received "100 Trying" at the beginning. This ruined any failure route attempts from SER, as call failed. CASE 2: Imagine now a second call. This time PSTN provider send "183 Session Progress" after 14 seconds and remote callee picked up after further 15 seconds. This time everything went good. UA did not hit "max_response_time_invite" timer - it waited patiently for whole 29 seconds for remote callee to pick up. I think that when UA receives provisional response (like "100 Trying") from SER it should stop timer "max_response_time_invite" and use 'ring_time_limit'. What device does now, it uses: * "max_response_time_invite" when "100 Trying" received * "ring_time_limit" when "180 Ringing" or "183 Session Progress" received I haven't really found explanation of this in RFC3261. When you look at state machine on page 128, it is not explicitly said which timer controls behaviour of UA when in "Proceeding" state ("Proceeding" is entered oin receipt of ANY provisional response). QUESTION 1: what timer controls behaviour of UA when in "Proceeding" state? Especially regarding timeout. Is this still the 'timer B' started at the beginning of INVITE transaction? If it is 'timer B', why UA is CANCELing transaction after 20 seconds, if it informed by "Expires: 65" field in INVITE that it set 'timer B' to 65 seconds. QUESTION 2: What should I propose to manufacturer for maximum value for "max_response_time_invite"? Something like 120 seconds ? -- Thanks, Arek Bekiersz

4 3

RE: [Serusers] re : presence module with ser
by Yeung OnTai-q16645 26 Jan '06

26 Jan '06

Hi I'd like to use PA, XCAP with eyeBeam. Are you talking about the "ser-0.10.99-dev30-tm-timers-pa-3", I've downloaded the tar file and tried "make install". It seems like presence works fine, XCAP is also working(except RLS)... About the RLS, do I need to create a directory in "<xcap-root>/rls-services/global/index" where all the XML (for RLS) files are stored? How can I configure the eyeBeam accordingly? By default it's configured as "subscribe to contact list : $username$-list@$domain$". It seems like SER does not consider it as a list, it only considers it just another URI. Did I do something wrong? Or do I need to change anything in SER? Regards, Dennis -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Tuesday, January 24, 2006 11:11 PM To: Yeung OnTai-q16645 Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] re : presence module with ser On 18-01-2006 09:07, Yeung OnTai-q16645 wrote: > Hi all, > > I need to build SER with presence and XCAP. Does anyone know which > version of SER contain a relatively "stable" version of "pa" and "rls". > I've tried 0.9.6 and the cvs stable build, but I ran into different > problems(error in loading modules, compling errors, etc) in making the > source code. > > Can anyone point me to the right direction please? Try the version from http://ftp.iptel.org/pub/ser/presence That one should be relatively easy to build. Note that the XCAP server is not included. Jan. > > Thank you very much! > > Regards, > Dennis > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

1 0

[Users] CPL-C module and PostgreSQL database
by Lenir 25 Jan '06

25 Jan '06

Guys, Does the CPL-C module work with PostgreSQL or any other database module other then MySQL? Regards, Lenir

2 2

Re: [Serusers] Error 599 : Message too big
by Akeru Ikena 25 Jan '06

25 Jan '06

OK, after some investigations here are my findings. I post them here so if some one get the same error he might get hints. It seems if the the other end of the call has his soft phone closed, this error is returned. I tried to increase the max size to 4096 and now I get the "Too many hops" error. Which means, the message is forwarded way too much time, beeing increased over the 2048 limit. Now, why do I get a route loop when a user is offline...I suspect this "...if (uri==myself) {..." of beeing the culprit. I added various alias to the proxy ip adress, to the proxy name and even the domain name bu nothing worked out. Any ideas ?

2 1

[Serusers] Can anyone define the 'flags' column in the location database?
by Leo Papadopoulos 25 Jan '06

25 Jan '06

Dear SER users, I cannot get any information on the column definition of the 'location' table. However, all I really need is the definition of the 'flags' column. I know if you set flags to 128 you insert a permanent entry into the location table, but I cannot figure out what the rest of the flags are for. Can anyone help? I have searched: - seruser archives - location module readme file - onsip website Leo Papadopoulos Web site: www.telecomCTO.com

1 0

[Serusers] handle_publish crashes in PA
by sip 25 Jan '06

25 Jan '06

Okay... I'll admit, this isn't the most recent PA module, but I thought someone might have a hint nonetheless. I'm running SER 0.9.6. I do NOT wish to upgrade to the somewhat haphazard presence snapshot as it's not entirely in stable/usable condition yet. I was, however, under the impression that simple PUBLISH/SUBSCRIBE methods would work okay on this version of the module. Am I mistaken? Usual stuff in the ser.cfg if(method=="SUBSCRIBE") { if(!t_newtran()) { log(1, "SUBSCRIBE newtran error\n"); sl_reply_error(); }; handle_subscription("registrar"); break; }; if(method=="PUBLISH") { if(!t_newtran()) { log(1, "PUBLISH newtran error\n"); sl_reply_error(); }; handle_publish("registrar"); break; }; When my Snom phone sends a publish, it looks like this: U 66.112.115.10:2057 -> 66.112.115.34:5060 PUBLISH sip:11019552444@my.domain.com:5060 SIP/2.0. Via: SIP/2.0/UDP 66.112.115.10:2057;branch=z9hG4bK-nc00a965o8y5;rport. From: "Death" <sip:11019552444@my.domain.com:5060>;tag=o6bezw24ge. To: "Death" <sip:11019552444@my.domain.com:5060>. Call-ID: 3c267063afc8-19wbm3ssr4y3@snom190. CSeq: 1 PUBLISH. Max-Forwards: 70. Event: proxy-config. Content-Type: application/text. Content-Length: 0. . It sends this (with slightly different From: tags) 9 times and then ser coredumps without a word to the log. Any ideas? Give up presence for now and just tell my users to get over it? :) N.

2 1

[Users] tls_domain section for outoing TLS connections
by Klaus Darilion 25 Jan '06

25 Jan '06

Hi! Reading the TLS docs I understand that using the tls_domain section, I can specify different certificates and CAs which will be used according to the socket on which the incoming TLS request is received. But is there also a way to use the "TLS domains" also for outgoing TLS connections? If yes, how do I trigger using a certain "TLS domain" for outgoing TLS connections? regards klaus

1 0

Re: [Serusers] about RTP Proxy
by Voipers Portugal 25 Jan '06

25 Jan '06

Are you sure that the problem is the RTP proxy? Have you tried to analyse the traffic to see if there is packet loss on the network? RTPproxy doesn´t affect the traffic, only corrects the packets address. (at least is what i think). Regards, Jose Simoes On 1/12/06, Anthony Chapellier <anthony(a)mbdsys.com> wrote: > Hi, > > I tested RTP proxy a few days ago with two SIP UAs using G711 codec. RTP > proxy did RTP relay right but the sound is very bad quality and seems to > be chopped.... Does anyone got the same problem ? > > I used the last version of RTP proxy from the CVS source files. Maybe > it's coming from the version, in this case does somebody could send me a > working version of RTP proxy ? or know how I could resolve this ? > > Thanks, > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

1 0

[Serusers] SER.CFG errors
by Voipers Portugal 25 Jan '06

25 Jan '06

Hi, I have recently installed the latest stable version of SER for debian at: http://ftp.iptel.org/pub/ser/latest/packages/debian/stable/ But i am having some problems running it with my old config file. Here are the errors that appear in my log: 0(6841) set_mod_param_regex: parameter <db_url> not found in module <acc> 0(6841) parse error (52,3-4): Can't set module parameter 0(6841) parse error (118,46-47): unknown command, missing loadmodule? 0(6841) parse error (124,46-47): unknown command, missing loadmodule? 0(6841) parse error (210,38-39): unknown command, missing loadmodule? ERROR: bad config file (4 errors) And the lines that correspond to the errors are: 52: modparam("usrloc|uri", "db_url", "sql://seruser:serpass@ip:5432/ser") and 118, 124, 210 are all the same: acc_db_request("Connected", "acc"); I have all modules loaded without any problem. Do you know if any of the functions or attributes have change their names recently? Regards, Jose Simoes

1 0

[Serusers] presence
by Barta János 25 Jan '06

25 Jan '06

Hello! I have a problem again :-). I would like to make a special service. I would like to make the SER get the presence info not only from the own DB but from an external LDAP database too. For example: The ser receives a location request. First it checks its own database, and if the SER can not found the user in the USERDB, then it looks it up from a script. I made a script, but a have a problem with this. if (!lookup("location")) { if (!exec_dset("/etc/ser/ldap.scr")) { sl_send_reply("404", "Not Found"); break; } else { log(1," sipldap call"); }; }; If the script found the user, then the answer will be (f.e.): sip:user@mydomain.hu exit 0 else it answers: exit 1 The contact person in the contact list will not be active (not will be green), if the script answers:sip:user@mydomain.hu. Why? (I need a special format in the answer?) Or it will not work with a script? I need to use XCAP? Thanks, Jani ________________________________________________________________ Harry Potter és a Félvér Herceg! Garantált szállítás a megjelenés napján! (február 10. ) Jegyezze elő most! http://www.bookline.hu/control/news?newsid=322&affiliate=frehp6kar1482

1 0

[Serusers] SER Presence
by Akeru Ikena 25 Jan '06

25 Jan '06

Hi all, Now that my routing problems seems to be gone, I ran into some presence issue. It seems the presence mechanism is working well, meaning I created the DB tables to store the informations, I added some lines in my conf (see below) and now I'd like my users to be able to send presence info each other WITHOUT an xcap server. I naturally wanted to add modparam("pa", "auth", "none"). But, as I understood, it seems this option is no longer exported in the lastest SER Version. So, how can I do ? Is it now mandatory to have an xcap server running ? Thanks alot. if (method == "SUBSCRIBE") { # create transaction state; abort if error occured xlog("L_INFO", "\n[SER]: --SUBSCRIBE--: Time: [%Tf] Method: <%rm> R-uri: <%ru> Contact Header: <%ct> From: <%fu> To <%tu>\n\n"); if ( !t_newtran()) { sl_reply_error(); break; }; if(!handle_subscription("registrar")) { sl_reply_error(); break; }; break; }; if (method == "PUBLISH") { # create transaction state; abort if error occured xlog("L_INFO", "\n[SER]: --PUBLISH--: Time: [%Tf] Method: <%rm> R-uri: <%ru> Contact Header: <%ct> From: <%fu> To <%tu> \n\n"); if ( !t_newtran()) { sl_reply_error(); break; }; if(!handle_subscription("registrar")) { sl_reply_error(); break; }; break; };

2 1

[Serusers] ser+radius + accounting + matching caller with its number
by miliczo 25 Jan '06

25 Jan '06

Hallo, Does anybody know how to match username with a telephone number so that one can only register with his own number. I can't use mysql database, only a database behind radius. I tried with avp_load_radius("digest"), but it doesn't work. I dont know what is wrong with it, or how it is ment to be used. I understand that this function imports all other data that radius returns, but I dont know where to? Best regards, Zoran

1 0

[Users] Re: uac_replace_from Core
by H Quintana 25 Jan '06

25 Jan '06

Hi Bogdan, It's working fine right now, thank you very much. In my case, the problem was with the Eyebeam softphone version: 1.1 3007n stamp 17816. Best regards, Humberto >Humberto, > >I found the bug in to/from parser - it is fixed on CVS. Please >let me know if works on now. > >regards, >bogdan > >Bogdan-Andrei Iancu wrote: > >> Hi Humberto, >> >> do you have the core file? can you reproduce the crash? if so, >>can you provide the guilty request? >> >> Thanks for report, >> bogdan >> >> >> H Quintana wrote: >> >>> Hi, >>> >>> I'd like to have some ideas for the solution of this >>> problem: In OpenSer 1.0., when I use the function >>> uac_replace_from("","me(a)here.com") I'm getting the >>> following error: >>> Jan 24 15:19:59 server1 >>> /usr/local/sbin/openser[22100]: BUG: del_lump: offset >>> + len exceeds message size (79 + -135152687 > 946) >>> Jan 24 15:20:00 server1 >>> /usr/local/sbin/openser[22084]: child process 22100 >>> exited by a signal 6 >>> Jan 24 15:20:00 server1 >>> /usr/local/sbin/openser[22084]: core was generated >>> Jan 24 15:20:00 server1 >>> /usr/local/sbin/openser[22084]: INFO: terminating due >>> to SIGCHLD >>> >>> >>> Thanks for your help. Best regards, >>> >>> Humberto __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

1 0

[Users] loose_route() (misbehaviour?)
by samuel 25 Jan '06

25 Jan '06

Hi all, Might be related to the mail I just sent about the username in the record-route or to another problem but the thing is that I can not close the session from the callee side...i'll try to explain: There are two users registered to two different openser instances. In the session initiation, the record route headers contain the usernames of both users and are therefore stored in the end points. When the caller sends the BYE, the callee's openser receives the BYE and rewrites the URI with the record-route (shouldn't it consider the request for itself and check the Req-URI instead??). NExt is the diagram and some more info... Alice oser1 oser2 Bob | | | | |--F1---> |--F2--->|--F3-->| | | | | | | | | | | | | |--F4--> |--F5-->|--F6--> | | | | | | | | | | | | | | |<--F8---|<--F9--| F1: INVITE Bob@oser2 F2: INVITE Bob@oser2 Record-Route:<sip:Bob@oser1;nat=yes;lr=on> F3: INVITE Bob@bob_contact_ip Record-Route:<sip:Bob@oser2;nat=yes;lr=on> Record-Route:<sip:Bob@oser1;nat=yes;lr=on> OK,ACK not shown.. F4: BYE Bob@nat_fixed_ip Route:<sip:Bob@oser1;nat=yes;lr=on>,<sip:Bob@oser2;nat=yes;lr=on> F5: BYE Bob@nat_fixed_ip Route:<sip:Bob@oser1;nat=yes;lr=on> F6: BYE Bob@bob_fixed_ip Everything ok, session closed. But, and here comes the problem: F7: BYE Alice@alice_fixed_ip Route:<sip:Bob@oser2;nat=yes;lr=on>,<sip:Bob@oser1;nat=yes;lr=on> When this message arrives to ser1, it sees the route header and, make a strict routing process and rewrites the header in the Req-URI to Bob@ser1. oser1 resends itself the message, and when it receives it again looks for use Bob in the local bindings which obviously does not exists (Bob is in oser2). Log: Jan 25 13:17:07 localhost openser[25501]: is_preloaded: No Jan 25 13:17:07 localhost openser[25501]: grep_sock_info - checking if host==us: 13==13 && [ip_oser1]== [ip_oser1] Jan 25 13:17:07 localhost openser[25501]: grep_sock_info - checking if port 5060 matches port 5060 Jan 25 13:17:07 localhost openser[25501]: after_strict: Next hop: 'sip:Bob@oser1_ip;nat=yes;lr=on' is loose router Jan 25 13:17:07 localhost openser[25501]: parse_headers: flags=ffffffffffffffff Jan 25 13:17:07 localhost openser[25501]: DEBUG: get_hdr_body : content_length=0 Jan 25 13:17:07 localhost openser[25501]: found end of header Jan 25 13:17:07 localhost openser[25501]: rewrite_uri: Rewriting Request-URI with 'sip:Bob@oser1_ip:5060;nat=yes;lr=o n' Jan 25 13:17:07 localhost openser[25501]: after_strict: The last route URI: 'sip:Bob@oser1_ip:5060;nat=yes;lr=on' Jan 25 13:17:07 localhost openser[25501]: DEBUG: has_totag: totag found Jan 25 13:17:07 localhost openser[25501]: DEBUG: t_newtran: msg id=12 , global msg id=11 , T on entrance=0xffffffff Jan 25 13:17:07 localhost openser[25501]: parse_headers: flags=ffffffffffffffff Jan 25 13:17:07 localhost openser[25501]: parse_headers: flags=78 Jan 25 13:17:07 localhost openser[25501]: t_lookup_request: start searching: hash=3891, isACK=0 Jan 25 13:17:07 localhost openser[25501]: DEBUG: RFC3261 transaction matching failed Jan 25 13:17:07 localhost openser[25501]: DEBUG: t_lookup_request: no transaction found Thanks a lot!!! Sam.

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[Users] Question concerning nathelper and rtpproxy
by Wolfgang Hottgenroth 25 Jan '06

25 Jan '06

Hi, I'm trying to get nathelper and rtpproxy to work, following the guide found on onsip.org, but unfortunately, it does not work correctly for. Don't know why. My setup is the following: SIP-Phone (170.127.120.189) -> Proxy1 (62.191.185.30) -> Proxy2 (62.191.185.147) -> PSTN-GW (139.4.200.6) Proxy1 supports authentication and handles the NAT travseral stuff. Since in the current testbed, SIP-Phone is not actual behind a NAT device I bypassed the test in the openser configuration to recognize this phone as "nated". What I see is, for a not nated phone, that the signaling goes the chain as drawn above and the media stream goes from the phone to the ngw and vice-versa, as I've expected. Everything is fine in this case. For a nated phone, the signaling again goes through the chain, but the media stream from the phone goes to the rtpproxy on Proxy1, which tries to send it to Proxy2, where no rtpproxy is running, and thereby an "ICMP port unreachable" is returned (which actual does prevent the rtpproxy from continously sending the media stream). Digging into the SIP and SDP is see the following: SIP-Phone sends an INVITE to Proxy1, in the SDP it has "Connection Information: 170.127.120.189" (IP of the phone) Proxy1 replies with "100 Trying" and sends the INVITE to Proxy2, in the SDP it has now "Connection Information: 62.191.185.30" (IP of Proxy1), as I've expected. Proxy2 replied with "100 Trying" and sends the INVITE to the ngw, in the SDP it still has "Connection Information: 62.191.185.30". Now the ngw sends "183 Session Progress" to Proxy2, which has in the SDP "Connection Information: 139.4.200.6" (IP of the ngw). Proxy2 sends "183 Session Progress" to Proxy1, still with "Connection Information: 139.4.200.6" in the SDP. Finally Proxy1 sends "183 Session Progress" to the phone, now with "Connection Information: 62.191.185.30" (IP of Proxy1) in the SDP, as I've expected, since the phone should send the media stream through the rtpproxy running on Proxy1. The phone sends the media stream as expected to Proxy1. But, and this is the problem, the rtpproxy sends the media stream not to the ngw, as I've expected looking into "SDP Connection Information" received by this box, but to Proxy2, which never was mentioned in a Connection Information. When starting rtpproxy with -f, I saw the it was asked by openser/nathelper to proxy between the phone and Proxy2, so the rtpproxy seems to work correctly. But why asks the nathelper on Proxy1 to proxy the media between the phone and Proxy2 and not, as I've expected between the phone and the ngw? My openser configuration is attached, if someone would be so nice to have a look and send a hint for me. Thanks, Wolfgang # # $Id: openser.cfg,v 1.5 2005/10/28 19:45:33 bogdan_iancu Exp $ # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (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 ---------------------------------- loadmodule "/opt/openser/lib/openser/modules/sl.so" loadmodule "/opt/openser/lib/openser/modules/tm.so" loadmodule "/opt/openser/lib/openser/modules/rr.so" loadmodule "/opt/openser/lib/openser/modules/maxfwd.so" loadmodule "/opt/openser/lib/openser/modules/usrloc.so" loadmodule "/opt/openser/lib/openser/modules/registrar.so" loadmodule "/opt/openser/lib/openser/modules/textops.so" loadmodule "/opt/openser/lib/openser/modules/auth.so" loadmodule "/opt/openser/lib/openser/modules/auth_radius.so" loadmodule "/opt/openser/lib/openser/modules/xlog.so" loadmodule "/opt/openser/lib/openser/modules/nathelper.so" loadmodule "/opt/openser/lib/openser/modules/uri.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) #modparam("rr", "enable_full_lr", 1) modparam("xlog", "buf_size", 8192) modparam("auth_radius", "radius_config", "/opt/openser/etc/openser/radiusclient.conf") modparam("auth", "rpid_prefix", "<sip:") modparam("auth", "rpid_suffix", "@62.191.185.30>;screen=yes;privacy=off") modparam("auth", "rpid_avp", "rpid") modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 30) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_sock", "unix:/opt/rtpproxy/sock/rtpproxy.sock") # ------------------------- request routing logic ------------------- # main routing logic route{ xlog("L_NOTICE", "REQUEST: $rm, $ru, $si\n"); if (method != "REGISTER") { record_route(); } if (method == "BYE" || method == "CANCEL") { unforce_rtp_proxy(); } if (src_ip == "170.127.120.189") { xlog("L_NOTICE", "*** FOUND THAT PHONE\n"); setflag(6); force_rport(); } if (loose_route()) { xlog("L_NOTICE", "*** loose routing enforced\n"); if (has_totag() && (method == "INVITE" || method == "ACK")) { if (nat_uac_test("3")) { xlog("L_NOTICE", "*** found NAT (1) !\n"); setflag(6); force_rport(); fix_nated_contact(); } force_rtp_proxy("l"); } route(1); return; } if (uri != myself) { xlog("L_NOTICE", "*** uri != myself\n"); sl_send_reply("483", "will not handle this"); return; } if (method == "CANCEL") { xlog("L_NOTICE", "*** cancel\n"); route(1); return; } else if (method == "INVITE") { xlog("L_NOTICE", "*** invite\n"); route(3); return; } else if (method == "REGISTER") { xlog("L_NOTICE", "*** register\n"); route(2); return; } xlog("L_NOTICE", "*** at the end: $rm\n"); route(1); return; } route[1] { t_on_reply("1"); if (!t_relay()) { if (method == "INVITE" && isflagset(6)) { unforce_rtp_proxy(); }; sl_reply_error(); } } route[2] { if (!search("^Contact: [ ]*\*") && nat_uac_test("3")) { xlog("L_NOTICE", "*** found NAT (2) !\n"); if (nat_uac_test("1")) { xlog("L_NOTICE", " --> 1\n"); } if (nat_uac_test("2")) { xlog("L_NOTICE", " --> 2\n"); } setflag(6); fix_nated_register(); force_rport(); } if (! radius_www_authorize("uu.net")) { www_challenge("uu.net", "0"); return; } consume_credentials(); if (!save("location")) { sl_reply_error(); } return; } route[3] { if (src_ip == "62.191.185.147") { xlog("L_NOTICE", "*** from upstream proxy\n"); if (uri =~ "^sip:99") { xlog("L_NOTICE", "*** outbreak prefix from outside, reject\n"); sl_send_reply("484", "no pstn outbreak for you"); return; } } else { if (! radius_www_authorize("uu.net")) { www_challenge("uu.net", "0"); return; }; consume_credentials(); if (nat_uac_test("3")) { xlog("L_NOTICE", "*** found NAT (3) !\n"); setflag(6); } append_rpid_hf(); if (uri=~"^sip:99") { xlog("L_NOTICE", "*** pstn outbreak call, uri: $ru\n"); strip(2); xlog("L_NOTICE", "*** stripped uri: $ru\n"); if (uri=~"^sip:[1-9]") { xlog("L_NOTICE", "*** local call\n"); prefix("+49231"); } else if (uri=~"^sip:0[1-9]") { xlog("L_NOTICE", "*** national call\n"); strip(1); prefix("+49"); } else if (uri=~"^sip:00[1-9]") { xlog("L_NOTICE", "*** international call\n"); strip(2); prefix("+"); } else { xlog("L_NOTICE", "*** something wrong\n"); sl_send_reply("488", "wrong telephone number"); return; } rewritehostport("62.191.185.147:5060"); route(4); route(1); return; } } xlog("L_NOTICE", "*** looking up ...\n"); if (!lookup("location")) { xlog("L_NOTICE", " --> not found\n"); sl_send_reply("404", "Not Found"); return; } xlog("L_NOTICE", " --> found\n"); route(4); route(1); return; } route[4] { if (isflagset(6)) { xlog("L_NOTICE", "*** handle NAT (1)\n"); force_rport(); fix_nated_contact(); force_rtp_proxy(); } return; } onreply_route[1] { xlog("L_NOTICE", "*** onreply_route\n"); if (isflagset(6) && status =~ "(180)|(183)|2[0-9][0-9]") { if (! search("^Content-Length:[ ]*0")) { xlog("L_NOTICE", "*** handle NAT (2)\n"); force_rtp_proxy(); } } if (nat_uac_test("1")) { fix_nated_contact(); xlog("L_NOTICE", "*** handle NAT (3)\n"); } return; }

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[Users] Re: [Devel] web interface development
by Bogdan-Andrei Iancu 25 Jan '06

25 Jan '06

I put also "users" at CC - more feedback is welcomed ;). wiki location: http://openser.org/dokuwiki/doku.php?id=openser_web_interface_documentation regards, bodgan Mike Williams wrote: >I've started posting a few design for the web interface plans on the Doku Wiki, they are in an early stage and I will continue to add to them. Please discuss or ask questions on the development list. Also, I would like to see people start adding wanted features and goals to a Wiki Page. > >Thanks. > >---Mike > >_______________________________________________ >Devel mailing list >Devel(a)openser.org >http://openser.org/cgi-bin/mailman/listinfo/devel > > >

1 0

[Users] enum patch
by Greg Fausak 25 Jan '06

25 Jan '06

Openser users, ------ is_from_user_enum: I find this extremely useful. This function uses the 'From' uri and does a enum lookup based on that number. A true or false is returned. I route to my customers using enum, so when the customer does an outbound call i can use this to verify that they are indeed a valid customer before routing the call. enum_fquery: Carry the previous function another step. You can do 'source routing' of a call using the from address. I do a enum lookup based on the from number, and then apply the ruri to number to the naptr that was found. Using this technique you can 'source route' individual numbers. In this way, for example, customer A can elect (pay for) routing out 2 or 3 gateways, while customer B elects to just have one provider. Another interesting feature is to temporarily reroute a customer to an intercept stating that his account is suspended. -------- Both of these functions assume that the openser admin is operating private enum domain databases. I haven't created a patch file before. This file is a context diff output for a few files in the modules/enum directory. -g

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Re: [Users] Crash (core)
by Bogdan-Andrei Iancu 25 Jan '06

25 Jan '06

Hi Lars, it took some time, but thanks to your details debug output I was able to trace the problem and fix the bug. As I was expecting, in was in the to/from parser. Please update from CVS and give it a try. Thanks again for the help, bogdan Lars Sundqvist wrote: > I wish I could, but the INVITE's are not logged and I have not been > able to reproduce the problem on the test system :-( so i realize this > isn't very easy to "fix" I was mainly checking if this was a known > problem. > > Version is 1.0.0-tls (disable_tls=0) compiled from source. > > uac is called like: > uac_replace_from("SomeText: $avp(s:display)","") > > The INVITE was most likely produced by a "Aastra Intelligate" SIP UA > since the particular customer has one of these registered right now. > > /Lasse > > Bogdan-Andrei Iancu wrote: > >> Hi Lars, >> >> can you please post the complete INVITE request that produces the >> crash and how the uac_replace_from() function is called from script? >> and what version are you using? >> >> regards, >> bogdan >> >> Lars Sundqvist wrote: >> >>> I have switched from SER to OpenSER in our production enviroment >>> since I >>> need a few of the "new" modules, I am however having troubles with >>> OpenSER hanging and crashing, so much so, that I had to change back. >>> >>> There is no clues in the logs to why this is happening but i have a >>> coredump from a crash: >>> >>> (gdb) where >>> #0 0x0052f7a2 in _dl_sysinfo_int80 () from /lib/ld-linux.so.2 >>> #1 0x005707d5 in raise () from /lib/tls/libc.so.6 >>> #2 0x00572149 in abort () from /lib/tls/libc.so.6 >>> #3 0x08054863 in del_lump (msg=0x0, offset=186, len=-135181242, >>> type=HDR_OTHER_T) at data_lump.c:288 >>> #4 0x00f8bd20 in replace_from (msg=0x8130d98, from_dsp=0xbff977a0, >>> from_uri=0x0) at from.c:207 >>> #5 0x00f8e57b in w_replace_from2 (msg=0x8130d98, dsp=0x812f700 >>> "\030W\022\b\v", uri=0x0) at uac.c:311 >>> #6 0x08051703 in do_action (a=0x8126658, msg=0x8130d98) at >>> action.c:677 >>> #7 0x080529e3 in do_action (a=0x8126680, msg=0x8130d98) at >>> action.c:773 >>> #8 0x0805297b in do_action (a=0x8126a58, msg=0x8130d98) at >>> action.c:773 >>> #9 0x0805297b in do_action (a=0x8126f80, msg=0x8130d98) at >>> action.c:773 >>> #10 0x0805297b in do_action (a=0x8127058, msg=0x8130d98) at >>> action.c:773 >>> #11 0x0805297b in do_action (a=0x812b6d0, msg=0x8130d98) at >>> action.c:773 >>> #12 0x080529e3 in do_action (a=0x812b6f8, msg=0x8130d98) at >>> action.c:773 >>> #13 0x080529e3 in do_action (a=0x812b720, msg=0x8130d98) at >>> action.c:773 >>> #14 0x0805297b in do_action (a=0x812b7f8, msg=0x8130d98) at >>> action.c:773 >>> #15 0x08052b42 in run_actions (a=0x8122da0, msg=0x8130d98) at >>> action.c:773 >>> #16 0x080753e9 in receive_msg ( >>> buf=0x80eb300 "INVITE sip:987654321@xxx.xxx.xxx SIP/2.0\r\nVia: >>> SIP/2.0/UDP >>> 192.168.0.17:5060;rport;branch=z9hG4bK0ffd9a6274aeb347617736a6da35e918\r\nTo: >>> >>> 987654321<sip:987654321@xxx.xxx.xxx>\r\nFrom: 123456789<si"..., >>> len=943, >>> rcv_info=0xbff98b80) >>> at receive.c:160 >>> #17 0x0808dd1c in udp_rcv_loop () at udp_server.c:464 >>> #18 0x08064e7f in main_loop () at main.c:1033 >>> #19 0x08065615 in main (argc=3, argv=0xbff98d34) at main.c:1570 >>> >>> (Numbers and sip domain edited) >>> Anyone who has a clue whats going wrong? >>> >>> Lars Sundqvist >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >> >

2 2

SV: [Serusers] Setting up ser (configuring static forwarding)
by Robert Zettinger 25 Jan '06

25 Jan '06

Hi, I am running everything on machines on the same private network. My softphone is on a windows machine with the local ip address 192.168.1.183, ser is on a Linux machine with the local ip address 192.168.1.147, and my friends softphone is on a windows machine with the local ip address 192.168.1.192. And what I am trying to do is: 1. Having my softphone use 192.168.1.147 (the ser machine) as proxy. 2. Setting ser to forward a call to cooldude(a)company.se to 192.168.1.192 (my friend with a softphone) Robert -----Ursprungligt meddelande----- Från: Ladislav Andel [mailto:ladia6@centrum.cz] Skickat: den 25 januari 2006 13:51 Till: Robert Zettinger; serusers(a)lists.iptel.org Ämne: Re: [Serusers] Setting up ser (configuring static forwarding) Hi, you should better tell us where is your SER running . Inside of private network? public internet? Is your softphone in the same private network as your friend's softphone? Could you draw your network endpoints eg. SIP UA---(public internet)---- SER --- (private network) --- your friend's SIP UA If we know the location of your network entities then it would help us a lot to understand what you are trying to do. Ladislav Robert Zettinger wrote: > > I have problems setting up Sip Express Router (SER). I have installed > it, and it seems to run fine, and my softphone seems to able to > connect to it. Just to see if it works, I want to set up ser to > connect my softphone to my friend's (his IP is 192.168.1.192). > Therefore I change the ser.cfg file to look like this: > > ---------- > > # > # $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=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/ser_fifo" > > # ------------------ 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/textops.so" > > # Uncomment this if you want digest authentication > # mysql.so must be loaded ! > #loadmodule "../usr/local/lib/ser/modules/auth.so" > #loadmodule "../usr/local/lib/ser/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=="sip:cooldude@company.se") { > forward( 192.168.1.192, 5060); > } > > 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("iptel.org", "subscriber")) { > # www_challenge("iptel.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(); > }; > } > > > ---------- > > Where > > if (uri=="sip:cooldude@company.se") { > forward( 192.168.1.192, 5060); > } > > is the only thing I have added to the original default config file > that comes with the distribution. I now want to enter > "sip:cooldude@company.se" in my softphone and get forwarded to > 192.168.1.192. But nothing happens. Ideas? > > > > I am using Linux 2.4.18-14 and ser 0.9.6. > > ------------------------------------------------------------------------ > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

2 1

[Users] rr add_username
by samuel 25 Jan '06

25 Jan '06

Hi all!!!! My openSER does not honour the add_username parameter of the rr module and always insert the user part in the record route headers. Maybe it is already corrected in the CVS... I downloaded the next snapshot openser-1.0.0-cvs-13Jan.tgz and the output from the -V is version: openser 1.0.0 (i386/linux) flags: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, DNS_IP_HACK, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT ADAPTIVE_WAIT_LOOPS=1024, MAX_RECV_BUFFER_SIZE 262144, MAX_LISTEN 16, MAX_URI_SIZE 1024, BUF_SIZE 65535 @(#) $Id: main.c,v 1.7 2005/10/28 14:55:23 klaus_darilion Exp $ main.c compiled on 16:28:45 Jan 16 2006 with gcc 3.3 Thanks a lot, Sam.

2 2

[Serusers] Setting up ser (configuring static forwarding)
by Robert Zettinger 25 Jan '06

25 Jan '06

I have problems setting up Sip Express Router (SER). I have installed it, and it seems to run fine, and my softphone seems to able to connect to it. Just to see if it works, I want to set up ser to connect my softphone to my friend's (his IP is 192.168.1.192). Therefore I change the ser.cfg file to look like this: ---------- # # $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=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/ser_fifo" # ------------------ 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/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "../usr/local/lib/ser/modules/auth.so" #loadmodule "../usr/local/lib/ser/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=="sip:cooldude@company.se") { forward( 192.168.1.192, 5060); } 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("iptel.org", "subscriber")) { # www_challenge("iptel.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(); }; } ---------- Where if (uri=="sip:cooldude@company.se") { forward( 192.168.1.192, 5060); } is the only thing I have added to the original default config file that comes with the distribution. I now want to enter "sip:cooldude@company.se" in my softphone and get forwarded to 192.168.1.192. But nothing happens. Ideas? I am using Linux 2.4.18-14 and ser 0.9.6.

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[Users] uac_replace_from Core
by H Quintana 25 Jan '06

25 Jan '06

Hi, I'd like to have some ideas for the solution of this problem: In OpenSer 1.0., when I use the function uac_replace_from("","me(a)here.com") I'm getting the following error: Jan 24 15:19:59 server1 /usr/local/sbin/openser[22100]: BUG: del_lump: offset + len exceeds message size (79 + -135152687 > 946) Jan 24 15:20:00 server1 /usr/local/sbin/openser[22084]: child process 22100 exited by a signal 6 Jan 24 15:20:00 server1 /usr/local/sbin/openser[22084]: core was generated Jan 24 15:20:00 server1 /usr/local/sbin/openser[22084]: INFO: terminating due to SIGCHLD Thanks for your help. Best regards, Humberto __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] rtpproxy
by Olivier Taylor 25 Jan '06

25 Jan '06

Hi all, I have a ser(0.9.4/Freebsd) server well working, but in some cases, I have a strang behavior. I use rtpproxy for nat traversal,all my users can call pstn, but when we try to call one of our users from internal or from pstn, Ser sends many invites and callee never answer to these invite. I must conclude that they don't receive it... All the users having that problems are under a nated network. Any idea ? Olivier

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[Serusers] Billing - Call Forward
by Bjorn Asmul 25 Jan '06

25 Jan '06

Has anyone successfully been able to set the "From"-field to the original "To"-field after changing the r-uri to the forwarded destination? A calls B which forwards to C Currently A is billed because that's the callerID that goes through. I want to bill B. -- Bjorn

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[Serusers] Radius AVOPS
by Deepak Singhal 25 Jan '06

25 Jan '06

Can the caller party IP Address be added in radius authentication and accounting packets for SER ? Perhaps using avops module ? Regards Deepak Singhal

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RE: [Serusers] Billing - Call Forward
by Bjorn Asmul 25 Jan '06

25 Jan '06

Don't think so. append_rpid_hf() maybe, but I haven't had any luck with it... -----Original Message----- From: Paul Hazlett [mailto:javarockx@gmail.com] Sent: Tuesday, January 24, 2006 11:28 PM To: Bjorn Asmul Cc: seruser Subject: Re: [Serusers] Billing - Call Forward I'm not too familiar with the UAC module, but perhaps it will do the trick. Cheers, Paul On 1/24/06, Bjorn Asmul <bjorn(a)atlasvoice.com> wrote: > > Has anyone successfully been able to set the "From"-field to the > original "To"-field after changing the r-uri to the forwarded destination? > > A calls B which forwards to C > > Currently A is billed because that's the callerID that goes through. > I want to bill B. > > -- Bjorn > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers > > >

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RE: [Serusers] re : presence module with ser
by Yeung OnTai-q16645 25 Jan '06

25 Jan '06

Hi Jan, I'd like to use PA, XCAP with eyeBeam. Are you talking about the "ser-0.10.99-dev30-tm-timers-pa-3", I've downloaded the tar file and tried "make install". It seems like presence works fine, XCAP is also working(except RLS)... About the RLS, do I need to create a directory in "<xcap-root>/rls-services/global/index" where all the XML (for RLS) files are stored? How can I configure the eyeBeam accordingly? By default it's configured as "subscribe to contact list : $username$-list@$domain$". It seems like SER does not consider it as a list, it only considers it just another URI. Did I do something wrong? Or do I need to change anything in SER? Regards, Dennis -----Original Message----- From: Jan Janak [mailto:jan@iptel.org] Sent: Tuesday, January 24, 2006 11:11 PM To: Yeung OnTai-q16645 Cc: serusers(a)lists.iptel.org Subject: Re: [Serusers] re : presence module with ser On 18-01-2006 09:07, Yeung OnTai-q16645 wrote: > Hi all, > > I need to build SER with presence and XCAP. Does anyone know which > version of SER contain a relatively "stable" version of "pa" and "rls". > I've tried 0.9.6 and the cvs stable build, but I ran into different > problems(error in loading modules, compling errors, etc) in making the > source code. > > Can anyone point me to the right direction please? Try the version from http://ftp.iptel.org/pub/ser/presence That one should be relatively easy to build. Note that the XCAP server is not included. Jan. > > Thank you very much! > > Regards, > Dennis > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

1 0

Re: Fwd: [Serusers] using dbtext module for authentication
by Jan Janak 25 Jan '06

25 Jan '06

It seems that you do not have "version" table. that table contains version of each table in your database, this is used for sanity checking. You can take a look into ser_mysql.sh and copy the contents of version table from corresponding sql table. Jan. On 20-01-2006 16:08, Yi Zheng wrote: > Hi, > > I have the following authentication related stuff in my ser.cfg file. However ser server failed to start with the following messages logged. I have a directory /root/dbtext/ser to store tables (such as subscriber). What could be wrong? Thanks, > > log messages > -------------------- > Jan 20 13:44:08 my ./ser[4424]: table_version(): Error in db_query > Jan 20 13:44:08 my ./ser[4424]: authdb_fixup: Error while querying table > version > Jan 20 13:44:08 my ./ser[4424]: ERROR: fix_expr : fix_actions error > > > > My ser.cfg > --------------- > loadmodule "/home/ser/presence/ser-0.10.99-dev30-tm-timers-pa-3/lib/ser/modules/dbtext.so" > loadmodule "/home/ser/presence/ser-0.10.99-dev30-tm-timers-pa-3/lib/ser/modules/auth.so" > loadmodule "/home/ser/presence/ser-0.10.99-dev30-tm-timers-pa-3/lib/ser/modules/auth_db.so" > > # -- 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) > modparam("usrloc", "db_mode", 0) > modparam("pa", "use_db", 0) > modparam("auth_db", "db_url", "dbtext:///root/dbtext/ser") > > /root/dbtext/ser/subscriber > ------------------------------------- > username(str) password(str) ha1(str) domain(str) ha1b(str) > suser:supasswd:xxx:iptel.org:xxx > test:test123::my.cisco.com:: > > Note: forwarded message attached. > Date: Fri, 20 Jan 2006 11:36:18 -0800 (PST) > From: Yi Zheng <yizheng(a)sbcglobal.net> > Subject: [Serusers] using dbtext module for authentication > To: serusers(a)lists.iptel.org > > Hi, > > I don't have enough space for MySql and hence want to try to use a plain text file (dbtext module) for authentication purpose. Just wondering whether it works properly with SER and what are the limitations besides limited space. Thanks, > > - ming > _______________________________________________ > 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

2 1

Re: [Serusers] re : add subscriber issue in ser-0.10.99
by Jan Janak 24 Jan '06

24 Jan '06

On 19-01-2006 17:43, Yeung OnTai-q16645 wrote: > When I use "#serctl add <username> <password> <email>", it does not work > anymore, because it complains that subscriber table is not found. serctl (the shell script) will be deprecated. > Whatever database table(subscriber or other) is being used now, there > should be a place where I can provision a new client with password. > Otherwise how can I authenticate it when the client register to SER? The table is called credentials, there is a command called ser_cred in sip_router/tools/serctl which can be used to provision the table. Note that an interface similar to the original serctl is being worked on, it will be available soon. > It seems like in ser-0.10.99, clients can just register to ser without > any provisioning.... No, but the newer SER version has much cleaner data model and the meaning of tables changed. credentials table maps digest credentials to users (identified by uid). uri table maps SIP URIs to users, and so on. Jan.

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[Serusers] avp_radius - how to start?
by Zoran Milic 24 Jan '06

24 Jan '06

Hello, Sorry for asking such a dumb question but I'm in a kind of a hurry. How do I declare variables in which I wish to store data I get from avp_load_radius() ? How do I assign a value to a variable and then check it up to some other value? Basicly my problem comes to avp+radius basic usage. Best regards, Zoran

1 0

[Serusers] NAT travesal: force_rport() what does it do?
by Yi Zheng 24 Jan '06

24 Jan '06

Hi, I am reading to understand the different solutions for NAT travesal. For nathelper+RTP proxy solution, what does the force_rport() function do? Is it rfc3581 compliant? Does it mean that the client needs to understand "received" and "rport" in the Via header? Thanks, - ming

2 1

[Serusers] SER version
by Voipers Portugal 24 Jan '06

24 Jan '06

I would like to know what's the most stable version of SER to use RADIUS, POSTGRES, LDAP, ENUM, LRC, NATHELPER and ACC modules? And what´s the name that i should use to get it from the CVS server. Regards, Jose Simoes

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RE: [Serusers] msilo, voicemail and serweb
by Alejandro Mellado 24 Jan '06

24 Jan '06

Yes, On serweb acount tab message store. I've seen the voicemail plugin code, and this doesn't write on mysql silo table. How I do put voicemail on serweb? Regards Alejandro. El sáb, 23-07-2005 a las 11:46 +0200, harry gaillac escribió: > Hello, > > You mean forward to voicemail on account tab ? > > harry > --- Alejandro Mellado <amellado(a)uct.cl> a écrit : > > > Hi > > > > > > I'm trying to activate the voicemail in serweb. > > I've working msilo with IM successfully. > > > > Somebody Can help me? > > > > > > Regards > > Saluda Atte. > > Alejandro Mellado <amellado(a)uct.cl>. > > Escuela de Informática > > Universidad Católica de Temuco > > > > _______________________________________________ > > 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 Saluda Atte. Alejandro Mellado <amellado(a)uct.cl>. Escuela de Informática Universidad Católica de Temuco

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Re: [Serusers] using dbtext module for authentication
by Jan Janak 24 Jan '06

24 Jan '06

On 20-01-2006 11:36, Yi Zheng wrote: > Hi, > > I don't have enough space for MySql and hence want to try to use a plain text file (dbtext module) for authentication purpose. Just wondering whether it works properly with SER and what are the limitations besides limited space. Thanks, Not all modules have been tested with dbtext, you will probably have to try it for yourself. dbtext is slow, thus it is not suitable for larger setups. Updates of dbtext files are not realtime, thus you would not see changes immediately. Jan.

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Re: [Serusers] re : presence module with ser
by Jan Janak 24 Jan '06

24 Jan '06

On 18-01-2006 09:07, Yeung OnTai-q16645 wrote: > Hi all, > > I need to build SER with presence and XCAP. Does anyone know which > version of SER contain a relatively "stable" version of "pa" and "rls". > I've tried 0.9.6 and the cvs stable build, but I ran into different > problems(error in loading modules, compling errors, etc) in making the > source code. > > Can anyone point me to the right direction please? Try the version from http://ftp.iptel.org/pub/ser/presence That one should be relatively easy to build. Note that the XCAP server is not included. Jan. > > Thank you very much! > > Regards, > Dennis > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

1 0

Re: [Serusers] I have a problem: undefined symbol: dbf
by Jan Janak 24 Jan '06

24 Jan '06

It looks like you are trying to call the functions of DB API but do not have db_func_t defined anywhere in the module. Check, for example, the sources of auth_db module, look for db_func_t Jan. On 17-01-2006 14:47, jvr_78(a)yahoo.com.ar wrote: > Hi, > I'm new in this group. > I have work with ser 0.8.14 and now I need presence module, so actualice the > ser version to 0.10.99, and now don't work my module. > > the error that show is: > > #ser > 0(3981) ERROR: load_module: could not open module > </usr/local/lib/ser/modules/control.so>: /usr/local/lib/ser/modules/control.so: > undefined symbol: dbf > 0(3981) parse error (47,13-14): failed to load module > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (62,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (63,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (66,21-22): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (67,19-20): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (68,19-20): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (69,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (70,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (71,21-22): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (72,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (73,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (74,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (75,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (76,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (77,20-21): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (78,19-20): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (79,19-20): Can't set module parameter > 0(3981) set_mod_param_regex: No module matching control found > | 0(3981) parse error (80,20-21): Can't set module parameter > 0(3981) parse error (130,28-29): unknown command, missing loadmodule? > > 0(3981) parse error (134,20-21): unknown command, missing loadmodule? > > 0(3981) parse error (150,33-34): unknown command, missing loadmodule? > > 0(3981) parse error (159,35-36): unknown command, missing loadmodule? > > 0(3981) parse error (161,25-26): unknown command, missing loadmodule? > > 0(3981) parse error (169,24-25): unknown command, missing loadmodule? > > 0(3981) parse error (184,42-43): unknown command, missing loadmodule? > > 0(3981) parse error (215,26-27): unknown command, missing loadmodule? > > 0(3981) parse error (219,24-25): unknown command, missing loadmodule? > > 0(3981) parse error (230,24-25): unknown command, missing loadmodule? > > 0(3981) parse error (263,28-29): unknown command, missing loadmodule? > > 0(3981) parse error (267,20-21): unknown command, missing loadmodule? > > 0(3981) parse error (281,24-25): unknown command, missing loadmodule? > > ERROR: bad config file (31 errors) > > > > > > ___________________________________________________________ > 1GB gratis, Antivirus y Antispam > Correo Yahoo!, el mejor correo web del mundo > http://correo.yahoo.com.ar > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers

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FW: [Serusers] ser live database and mysql
by Richard C. Thompson 24 Jan '06

24 Jan '06

Frank Thanks for the reply I thought that setting "modparam("usrloc", "db_mode", 2)" to 1 did the same thing. ie it updates the table whenever anything changes. I had it set to 1 but the "location" sql database swelled to over 9000 and ser wouldn't restart until I truncated the table. My concern is that ser can't delete records in sql even though I checked the grants and privileges for user ser and they seem to be ok. Also serctl alias rm doesn't remove the aliases record from the table, again an issue with grants. But serctl alias add works. | Grants for ser@localhost | +--------------------------------------------------------------------------- --------+ | GRANT USAGE ON *.* TO 'ser'@'localhost' IDENTIFIED BY PASSWORD '*******************************' | | GRANT ALL PRIVILEGES ON `ser`.* TO 'ser'@'localhost' | +--------------------------------------------------------------------------- --------+ Got any ideas? Rick _____ From: Frank Fischer [mailto:frank.fischer@digitalnomads.ch] Sent: Tuesday, January 24, 2006 3:25 AM To: RThompson(a)vir2com.com Subject: RE: [Serusers] ser live database and mysql ser keeps the location table in memory for performance reasons and does synchronisation with the backend database sometimes. As far as i understand, this could lead to differences between in-memory data and in database-data. To avoid this situation and force SER to write all data to the database, you could use the experimental module "usrloc-cl". It works fine for me. - Frank _____ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Richard C. Thompson Sent: Monday, January 23, 2006 9:55 PM To: serusers(a)lists.iptel.org Subject: [Serusers] ser live database and mysql Hi all Hopefully someone can answer this. I am running ser with mysql and I have noticed that serctl moni shows a smaller number in "location" than the mysql "location" table. Eg serctl moni shows 404 in location mysql shows 556 in location I have modparam("usrloc", "db_mode", 2) even when I refresh the sql query many times over a long time, the tables never match. It seems that ser can't delete some of the entries Any help or hints would be appreciated Rick

1 0

[Users] trusting gateways through radius
by Jayesh Nambiar 24 Jan '06

24 Jan '06

Hi all, I have configured openser(1.0.0) to authenticate and account via freeradius. It is working properly and fine. But one problem that I am facing is trusting the gateways. I dont want to authenticate the gateways but trust them through their IP address. Is there a way I can do it in freeradius? If avp_radius can be used, pls explain where do i store the IPs that I trust in freeradius. If anyone can help in this regard it would be appreciated. Thanks a lot in advance. Jayesh. Send instant messages to your online friends http://in.messenger.yahoo.com

2 2

[Users] SER as loadbalancer and NAT
by Andreas Granig 24 Jan '06

24 Jan '06

Hi, A question to the NAT experts: I wonder if there are known side effects if I use fix_nated_contact() in combination with nat_flag in the registrar module instead of fix_nated_register() to save NATed contacts? What I want to do is checking for NAT on a stateless SER acting as load balancer, then forwarding it to a proxy in the LAN, like: # remove P-Nated-HF here if present... if(method == REGISTER && !search("^Contact:[ ]*\*") && client_nat_test("7")) { fix_nated_contact(); append_hf("P-Nated: 1\r\n"); force_rport(); } # forwarding to proxy using dispatcher module And in the proxy: modparam("registrar", "nat_flag", 1) # ... if(method == REGISTER) { if(is_present_hf("P-Nated")) setflag(1); save("location"); } # ... if(method == INVITE) { # a request for a local user... lookup("location"); setdsturi("sip:<address of loadbalancer>"); t_relay(); } Of course a mechanism has to be provided to let the loadbalancer or an external application NAT-ping the address provided in the rewritten contact. As far as I have seen, the difference is that this approach would save the NAT-adress of the contact in usrloc with an empty received-param, while fix_nated_register() saves the private adress of the contact with the NAT-adress in the received-param. I would assume (not verified yet), that in the case of using fix_nated_contact(), the NAT-address is set as r-uri after a lookup, with an empty dst-uri, which will have to be set to the load balancer's address to traverse the client-side NAT. In case of using fix_nated_register(), I assume the r-uri is set to the client's private address, and the received-address as dst-uri. Traversing the client's NAT wouldn't be that easy in this case because you would have to save the received-address somewhere in the message and forward the request to the load-balancer, which in turn has to forward it to the client's NAT address. So would this approach using fix_nated_contact() for REGISTERs make sense for using a SER-based load balancer? Or are there any limitations I haven't considered? Thanks, Andy

3 9

[Users] cancel method problem
by unplug 24 Jan '06

24 Jan '06

When UA1 makes a call to UA2, both ringing. Then UA1 cancels the call, a strange log message (from ngrep) will be got below. After that UA2 will ring 3 times itself. You can see in line 718, there is a cancel method. After that, there are many Record-Route, Via in the sip CANCEL method (line 763, line 1578 and many). Does anyone know what the problem is? http://meerkat.no-ip.com/openser/cancel.log

3 4

[Serusers] Help require to install SER in RH9
by Dinesh TJ 24 Jan '06

24 Jan '06

Hello All, Am a newbee to SER and interested to know how it works. I am tryin to install SER in RH9 box and facing some issues. If somebody can send the detailed step by step procedure on how to install then it will be really helpful for me. I am facing problem in Mysql part, first of all am not familiar with Mysql moreover i need to connect mysql with SER which is quite difficult for me. Please help me on this regard Regards Dinesh

1 0

[Serusers] ser live database and mysql
by Richard C. Thompson 23 Jan '06

23 Jan '06

Hi all Hopefully someone can answer this. I am running ser with mysql and I have noticed that serctl moni shows a smaller number in "location" than the mysql "location" table. Eg serctl moni shows 404 in location mysql shows 556 in location I have modparam("usrloc", "db_mode", 2) even when I refresh the sql query many times over a long time, the tables never match. It seems that ser can't delete some of the entries Any help or hints would be appreciated Rick

1 0

[Users] Call-specific AVP DBS storage for OpenSER
by Joachim Fabini 23 Jan '06

23 Jan '06

Hi all, Unfortunately the previous thread on this topic ended without a definitive statement or conclusion, probably because of the New Year holiday (btw, Happy New Year to all of you!). So here's another try: OpenSER is currently session stateful but not call stateful. This is valid also/especially for the AVP module: You can store AVPs on a per-contact base but (imho) _not_ specific to a call or dialog. The reason for this is that the AVP module does not allow/ implement the use of call-dependent variables as AVP names. Adding this feature to OpenSER doesn't require a change in interfaces (implementation effort is another topic ;). It's just about enabling the use of AVP or header values (like, e.g. call-id) as the _name_ of another AVP for DBS storage. I attach a specific example below. Can some of the developers and/or maintainers please comment and/or give some hints on the feasibility of this approach? Is it possible to add this feature to OpenSER? What effort is required to add this enhancement to the AVP module? Please come back if some details are unclear, I might have missed something important. Many thanks in advance for your reply and help, Best regards --Joachim PS: Example: The task is to process any SIP message that passes through our proxy. We must compute a unique ID specific to the call on the first message belonging to this call and store it in DBS. Whenever another message that is part of this call/dialog arrives, we retrieve the unique ID from the DBS and, e.g., append it as a specific header field value. The only way how this can be done is to use some call- specific AVP name (e.g. call-id), as we do not have any influence on the structure of the AVP value (so regexp- search based on AVP value does not help). # This is the AVP name that is required to retrieve # our specific, call-dependent AVP value from the # database whenever a message that belongs to this # call passes our OpenSER proxy. avp_printf("s:myTmpHeaderName", "$from/username$hdr(call-id)"); # This is the value we'd like to store. Could be also # only "$Ts", i.e., we can _not_ rely on regular # expression search for later retrieving the AVP # from the DBS based on its value. avp_printf("s:myTmpHeaderValue", "$hdr(call-id)-$Ts"); # This storing of the call-dependent info in DBS does # currently not work. OpenSER complains at start-up # about the variable AVP name/key on both of the # following lines. The former variant seems much more # flexible to me than the latter one. avp_db_store("s:myTmpHeaderName","s:myTmpHeaderValue"); avp_db_store("$hdr(call-id)","s:myTmpHeaderValue"); # This load obviously does also not work. avp_db_load("$hdr(call-id)","s:myTmpHeaderValue");

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[Serusers] Rtpproxy again
by Olivier Taylor 23 Jan '06

23 Jan '06

Another strange thing. When I call one of our users, it rings ok, but if I hangup and retry the same number, Ser seems not finding that user, if the user reset his phone, no problems, it rings again... Olivier

1 0

[Serusers] Push a specific entry in /etc/hosts
by Walter Luis 23 Jan '06

23 Jan '06

Hi, Friends, Is there a way (line in the server.conf) to push a especif entry in /etc/hosts to openvpn clients? I mean, something like: "command" domain.com 20.x.x.x Then, from the start to the finish of the openvpn, the client always knows that domain.com is 200.X.X.X and won't consult the dns. []'s -- Walter Luis

1 1

[Serusers] rtpproxy
by Olivier Taylor 23 Jan '06

23 Jan '06

Hi all, I have a ser(0.9.4/Freebsd) server well working, but in some cases, I have a strang behavior. I use rtpproxy for nat traversal,all my users can call pstn, but when we try to call one of our users from internal or from pstn, Ser sends many invites and callee never answer to these invite. I must conclude that they don't receive it... All the users having that problems are under a nated network. Any idea ? Olivier

1 0

[Serusers] Test
by sip 23 Jan '06

23 Jan '06

Having received zero messages on the ser list since the 20th, I thought I'd see if there's an issue with the list or if it's just been bizarrely quiet. N.

1 0

[Serusers] Asterisk and SER
by Jim McIver 23 Jan '06

23 Jan '06

Gentlemen, If I was using SER as a proxy server to deal with clients behind NATs, I can understand the concept - the sip proxy rewrites the header information to correct for the NATed client. Where I have a large "hole" in my understanding is concerning inbound calls from PSTN. Currently, these arrive at our Asterisk box which has no knowledge of the clever things SER should be doing to deal with the NATed clients. Can anyone please fill in the gaps in my understanding ? Better still.....can anyone let me have a look at their Asterisk and SER config files on a system that works ??? Jim -- --- www.yakara.com

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Re: [Serusers] Fwd: No RTP
by Voipers Portugal 23 Jan '06

23 Jan '06

SER does not handle RTP messages, only SIP messages. Jose Simoes On 1/16/06, samuel <samu60(a)gmail.com> wrote: > > www.onsip.org > > 2006/1/16, mohd suhaimi <see2belief(a)yahoo.com>: > > Hi, > > > > Any help on below is appreciated. > > > > Thanks > > > > --- mohd suhaimi <see2belief(a)yahoo.com> wrote: > > > > > Date: Thu, 12 Jan 2006 02:17:15 -0800 (PST) > > > From: mohd suhaimi <see2belief(a)yahoo.com> > > > Subject: No RTP > > > To: seruser <serusers(a)lists.iptel.org> > > > > > > Hi, > > > > > > when i make SIP to SIP calls thru my ser, no rtp > > > packets is processed. Especially when in a NATed > > > environment. > > > > > > Any idea? > > > > > > Thanks > > > > > > __________________________________________________ > > > Do You Yahoo!? > > > Tired of spam? Yahoo! Mail has the best spam > > > protection around > > > http://mail.yahoo.com > > > > > > > > > __________________________________________________ > > 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 > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org > http://lists.iptel.org/mailman/listinfo/serusers >

1 0

[Users] openserctl doesnt work
by martin＠campus-merseburg.de 23 Jan '06

23 Jan '06

Hi, when starting openserctl i get this message: root@sip:~ # /usr/local/sbin/openserctl /usr/local/sbin/openserctl: syntax error at line 1024: `GW_IP=$' unexpected OS is Solaris OpenSER Version: 1.0.0 CVS Bye

2 2

[Users] Asterisk & Openser
by Ronald Wiplinger 23 Jan '06

23 Jan '06

I want to use Openser to register all phones. If the phone is allowed (paid) than it should be able to reach a certain gateway (flat rate to certain numbers) Distinguished via mysql database. If the destination number is not in the flat rate, than the call should be forwarded to Astersik to handle the call (prepaid card) If features are requested, than should be also Asterisk handle it. Is there a starting point for this set-up?

2 1

[Users] connection between openser and mysql both in different machine
by unplug 23 Jan '06

23 Jan '06

How does openser connect to mysql which is not in the same machine (say openser is in 10.0.0.1 and mysql is in 10.0.0.2)? I have configured all mysql statement in the configuration file as "mysql://openser:openserrw@10.0.0.2/openser" but openser failed to start. Below is the error message from the log. Jan 21 14:25:51 owthkems01 /usr/local/sbin/openser[12298]: new_connection: Can't connect to local MySQL server through socket '/var/lib/mysql/mysql.sock' (2) Jan 21 14:25:51 owthkems01 /usr/local/sbin/openser[12298]: ERROR:domain_db_init: cannot initialize database connection Jan 21 14:25:51 owthkems01 /usr/local/sbin/openser[12298]: ERROR: domain:child_init(): Unable to connect to the database Jan 21 14:25:51 owthkems01 /usr/local/sbin/openser[12298]: ERROR:init_mod_child: Error while initializing module domain, rank 1

2 2

[Users] question about avp
by unplug 23 Jan '06

23 Jan '06

Say I have a record in the table usr_preferences with username, attribute=function, value=voicemail. I use avp_db_load("$from/username", "s:function") to get the value. I can print out the value in xlog function by using $avp(s:function). How can I put that value in the header by using append_hf? As I found that it doens't work if I use append_hf("P-hint: $avp(s:function)\r\n"). In other words, how can I get back the value of the variable stored in avp for other functions processing?

3 3

RE: [Serusers] SER frontend
by Barta János 23 Jan '06

23 Jan '06

Hello! I'm again :-). If I use the next config in my ser.cfg, then the authentication works fine, but the presence does not. if (!www_authorize("domain.hu", "subscriber")) { forward( csps.domain.hu, 5060); break; }; But if I insert the save("location") line in my config file, then the presence info will be good, but the authentication will be wrong. if (!www_authorize("domain.hu", "subscriber")) { forward( csps.domain.hu, 5060); save("location"); break; }; Can I mix these two features? Thanks, Jani -----Original Message----- From: serusers-bounces(a)lists.iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of Barta János Sent: Friday, January 20, 2006 4:45 PM To: serusers(a)lists.iptel.org Subject: [Serusers] SER frontend Hello! I would like to make a frontend SIP perver. For this server will be only one task, to provide the presence information. The architecture: SIP client -> SER ->CSPS. I have a problem with the registration method. I'm using http digest auth. If I forward the authentication messages (to the CSPS), then the SIP client logs in automatically, and in this case I don't have the possibility to type the username/pwd. What is wrong? And I have a second question: Can I use W$ Messenger 5.1 presence under SER? (The 5.0 works fine, but the 5.1 does not.) This is my config file: sip:/etc/ser# cat ./ser.cfg # # $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=3 # debug level (cmd line: -dddddddddd) #fork=yes log_stderror=yes # (cmd line: -E) /* Uncomment these lines to enter debugging mode fork=no log_stderror=yes */ alias=domain.hu alias=100.22.2.180 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" fifo_db_url="mysql://ser:heslo@localhost/ser" # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/pa.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" loadmodule "/usr/lib/ser/modules/exec.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- ####################### # -- usrloc params -- # ####################### /* 0 -- dont use mysql, 1 -- write_through, 2--write_back */ modparam("usrloc","db_mode",0) #modparam("usrloc","db_url","mysql://xxxxxxx:xxxx@localhost/ser") #modparam("usrloc", "use_domain", 1) #modparam("usrloc", "timer_interval", 10) ##################### # -- auth params -- # ##################### #modparam("auth_db","calculate_ha1", yes) #modparam("auth_db","db_url","mysql://xxxxx:xxxxx@localhost/ser") #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) ######################### # -- presence params -- # ######################### #modparam("pa", "db_url", "mysql://xxxxxxxx:xxxxxxxx@localhost/ser") modparam("pa", "default_expires",50) modparam("pa", "pa_domain", "domain.hu") # ------------------------- 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 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"); if (!t_relay()) { sl_reply_error(); }; break; }; if (uri==myself) { if (method=="REGISTER") { # if (!www_authorize("domain.hu", "subscriber")) { # forward( csps.domain.hu, 5060); # www_challenge("domain.hu", "0"); # break; # }; save("location"); break; }; lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; forward( csps.domain.hu, 5060); break; }; sl_send_reply("404", "Not Found"); # append_hf("P-hint: usrloc applied\r\n"); } sip:/etc/ser# Thanks, Jani ________________________________________________________________ Harry Potter és a Félvér Herceg! Garantált szállítás a megjelenés napján! (február 10. ) Jegyezze elő most! http://www.bookline.hu/control/news?newsid=322&affiliate=frehp6kar1482

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[Serusers] AW: Serusers Digest, Vol 33, Issue 26, RE: Running mediaproxy
by "Töx, Reinhold" 23 Jan '06

23 Jan '06

>Gentlemen, >I tried running media proxy using the scripts given in the getting started document (putting the scripts into /etc/init.d/ as >shown in the document). >However, when I try to run it, I get the following error - >[root@itrhost2 init.d]# ./mediaproxy start >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 >Any ideas what's wrong here ? Have you checked Python yet. Which version of Python do you use? Run 'python' from command line and type 'import optparse' to see what happens. If there is an error, you do not have the Python module 'optparse'. Upgrade to a newer version of Python or install the module by hand. Regards, Reinhold

1 0

[Users] substring function
by unplug 22 Jan '06

22 Jan '06

Does openser provide any substring function to get a substring within a string? For example, I have a string "1234abc5678" in avp s:gw, and I want to get "abc" from s:gw.

2 1

[Users] uac_replace_from function
by unplug 20 Jan '06

20 Jan '06

I am using uac_replace_from function as below. # replace only display and do not touch uri uac_replace_from("35418474",""); After running uac_replace_from function, I get the following message. U 203.193.4.242:5060 -> 203.193.4.226:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 203.193.4.226;branch=z9hG4bKaed7.46cc4101.0,SIP/2.0/UDP 10.0.0.52:5060;rport=5060;received=210.184.4.31;branch=z9hG4bKHNHpldVsKca6IYUS. From: "35418474"<sip:882754853589@ow01.owtel.com>;tag=sBl2gLH6O1myyqL5. To: "92588819" <sip:92588819@ow01.owtel.com>;tag=47C61DF4-26CD. Date: Fri, 06 Jan 2006 10:41:17 GMT. Call-ID: yTCQvS56ZCZXCHgt(a)10.0.0.52. Server: Cisco-SIPGateway/IOS-12.x. In the normal message, sip message, there is a space between display name and uri. ie. From: "35418474" <sip:882754853589@ow01.owtel.com> However, after I apply uac_replace_from function, there is a space there. How can I add a space such that it looks like a normal sip message? Does the space will affect the caller display?

3 10

[Users] centralized speed dial
by Reto Kortas 20 Jan '06

20 Jan '06

Hi! I am planning to set up a centralized speed dial service for a 2-layer proxy hierarchy. The design should look like these: PSTN | | UA --> LOCAL_SP --> MAIN-PROXY <-- LOCAL_SP <-- UA | | UA --> LOCAL_SP ------ | | PSTN ... and many many more local SIP Proxys (LOCAL_SP). Any branch has a local SIP-Proxy which functions also as a PSTN-Gateway and will route local calls, whether they are from inside or from the PSTN. The users are configured on the local SIP-Proxy, which will also register these user on a REGISTER-Request also on the centralized SIP-Proxy. The Speed dial database should contain internal as well as external (PSTN) numbers. The tricky thing is, that the users have to use their local PSTN Gateway, when they want to make off-net calls. The call flow must like these: UA -INVITE-> LOCAL_SP -INVITE-> MAIN-PROXY -> DB-Lookup for speed dial-alias MAIN-PROXY --301_RESPONSE--> LOCAL_SP --301_RESPONSE--> UA -> Setup new call with info from new contact header Does anybody now how to setup OpenSER, that he will execute a DB lookup for the user part of the R-URI and answers with a 301 response and the new address in the contact header? May I think the routing must look like this: Route[x] { NEW_URI = get_alias_for(1234); append_branch(NEW_URI); sl_send_reply("301" , "Number found"); } But how to set up and querie the database? Thx Reto

1 0

[Users] Question on re-writing From uri when ser forwards to Asterisk
by Barry Flanagan 20 Jan '06

20 Jan '06

Hi, I am using asterisk as PSTN gateway for users registered with OpenSER. Everything is working well for a single domain, but I will have problems with this when I add extra domains. I will have multiple domains using the OpenSER server, all using the same set of Asterisk servers for PSTN. My problem is that when user@domain1 gets sent to Asterisk, they are authenticated as just "user" (i.e. the domain portion is stripped off) and they get registered under the [user] sip name in Asterisk. Now, when I add the second domain, and user@domain2 goes to Asterisk, they also will appear to Asterisk to be [user] I think what I need to do is rewrite the From: URI to be user_domain1 and user_domain2 and then set them up in Asterisk as [user_domain1] and [user_domain2] so that they are unique So, my questions are: is this the correct/only way to do this, and if so, what do I need to do to achieve this? Many thanks. -- -Barry Flanagan

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[Users] only one call from duplicate register
by Ibrahim TUNALI 20 Jan '06

20 Jan '06

i know i can multiple register on a sip proxy with same account at same time from different locations. And both of them can call different numbers. But i want to only one client can call, other one will block during first one's conversation and second one will recive "not acceptable" response. is it possible and can you give a configuration example. regards, itunali

1 0

[Users] NAT+uac_replace_from
by unplug 20 Jan '06

20 Jan '06

Actually, I am replacing the username of the uri with the alias that stored in the database. In my configuration file, it is using mediaproxy for NAT function (features-callfwd.5.0.cfg from getting started). I also add the following codes in the very first of the route routine for alias replacing purpose. route { ... if (!has_totag() && method=="INVITE") { if (avp_db_load("$from/uri","s:alias")) { xlog("L_INFO","sip408: have alias - [$avp(s:alias)]\n"); uac_replace_from("anonymous","sip:$avp(s:alias)@$si"); } else { xlog("L_INFO","sip411: no alias\n"); }; }; ... } When I make a call from a phone to the PSTN phone, the caller drops the ring when the callee rings. Callee hangs up and callee rings again few seconds after. You can find an error message "SIP/2.0 481 Call Leg/Transaction Does Not Exist" in line 180. I wonder if there is any thing wrong with the above code or my concept is wrong. Please help. Below is the sip message. interface: any filter: (ip) and ( port 5060 ) # U 10.0.0.46:5060 -> 10.0.0.27:5060 INVITE sip:34163634@10.0.0.27 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bKeKunMgjqbhggzOxi. Max-Forwards: 70. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-2d. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Contact: <sip:9000@10.0.0.46:5060>. CSeq: 1 INVITE. Supported: 100rel, replaces. Content-Type: application/sdp. Content-Length: 286. . v=0. o=9000 58685689 12166570 IN IP4 10.0.0.46. s=SIP CALL. c=IN IP4 10.0.0.46. t=0 0. m=audio 6000 RTP/AVP 18 3 0 8 4 101. a=rtpmap:18 G729/8000. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:4 G723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 10.0.0.27:5060 -> 10.0.0.46:5060 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bKeKunMgjqbhggzOxi;rport=5060. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. CSeq: 1 INVITE. Server: OpenSer (1.0.0 (i386/linux)). Content-Length: 0. Warning: 392 10.0.0.27:5060 "Noisy feedback tells: pid=3353 req_src_ip=10.0.0.46 req_src_port=5060 in_uri=sip:34163634@10.0.0.27 out_uri=sip:934163634@10.30.0.248 via_cnt==1". . # U 10.0.0.27:5060 -> 10.30.0.248:5060 INVITE sip:934163634@10.30.0.248 SIP/2.0. Record-Route: <sip:34163634@10.0.0.27:5060;nat=yes;ftag=l7473818dCOs9ZMG;lr=on>. Via: SIP/2.0/UDP 10.0.0.27;branch=z9hG4bK0d2f.18572a04.0. Via: SIP/2.0/UDP 10.0.0.46:5060;rport=5060;branch=z9hG4bKeKunMgjqbhggzOxi. Max-Forwards: 69. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-2d. From: "anonymous"<sip:36418473@10.0.0.46>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Contact: <sip:9000@10.0.0.46:5060>. CSeq: 1 INVITE. Supported: 100rel, replaces. Content-Type: application/sdp. Content-Length: 287. . v=0. o=9000 58685689 12166570 IN IP4 10.0.0.46. s=SIP CALL. c=IN IP4 10.0.0.27. t=0 0. m=audio 35540 RTP/AVP 18 3 0 8 4 101. a=rtpmap:18 G729/8000. a=rtpmap:3 GSM/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:4 G723/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. # U 10.30.0.248:5060 -> 10.0.0.27:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 10.0.0.27;branch=z9hG4bK0d2f.18572a04.0,SIP/2.0/UDP 10.0.0.46:5060;rport=5060;branch=z9hG4bKeKunMgjqbhggzOxi. From: "anonymous"<sip:36418473@10.0.0.46>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Date: Wed, 04 Jan 2006 09:15:35 GMT. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U 10.30.0.248:5060 -> 10.0.0.27:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 10.0.0.27;branch=z9hG4bK0d2f.18572a04.0,SIP/2.0/UDP 10.0.0.46:5060;rport=5060;branch=z9hG4bKeKunMgjqbhggzOxi. From: "anonymous"<sip:36418473@10.0.0.46>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Date: Wed, 04 Jan 2006 09:15:35 GMT. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Require: 100rel. RSeq: 1250. Allow-Events: telephone-event. Contact: <sip:934163634@10.30.0.248:5060>. Record-Route: <sip:34163634@10.0.0.27:5060;nat=yes;ftag=l7473818dCOs9ZMG;lr=on>. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 177. . v=0. o=CiscoSystemsSIP-GW-UserAgent 584 9780 IN IP4 10.30.0.248. s=SIP Call. c=IN IP4 10.30.0.248. t=0 0. m=audio 17340 RTP/AVP 18. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. # U 10.0.0.27:5060 -> 10.0.0.46:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 10.0.0.46:5060;rport=5060;branch=z9hG4bKeKunMgjqbhggzOxi. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Date: Wed, 04 Jan 2006 09:15:35 GMT. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Require: 100rel. RSeq: 1250. Allow-Events: telephone-event. Contact: <sip:934163634@10.30.0.248:5060>. Record-Route: <sip:34163634@10.0.0.27:5060;nat=yes;ftag=l7473818dCOs9ZMG;lr=on>. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 175. . v=0. o=CiscoSystemsSIP-GW-UserAgent 584 9780 IN IP4 10.30.0.248. s=SIP Call. c=IN IP4 10.0.0.27. t=0 0. m=audio 35540 RTP/AVP 18. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes. # U 10.0.0.46:5060 -> 10.0.0.27:5060 PRACK sip:934163634@10.30.0.248:5060 SIP/2.0. Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bK8UeWxCfGrMJSO66f. Route: <sip:34163634@10.0.0.27:5060;nat=yes;ftag=l7473818dCOs9ZMG;lr=on>. Max-Forwards: 70. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-2d. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Contact: <sip:9000@10.0.0.46:5060>. CSeq: 2 PRACK. RAck: 1250 1 INVITE. Content-Length: 0. . # U 10.0.0.27:5060 -> 10.30.0.248:5060 PRACK sip:934163634@10.30.0.248:5060 SIP/2.0. Record-Route: <sip:10.0.0.27;ftag=l7473818dCOs9ZMG;lr=on>. Via: SIP/2.0/UDP 10.0.0.27;branch=z9hG4bKdc2f.dbee15d1.0. Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bK8UeWxCfGrMJSO66f. Max-Forwards: 69. User-Agent: Koncept KE10XX v4.32.05 00-09-45-0a-fc-2d. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Contact: <sip:9000@10.0.0.46:5060>. CSeq: 2 PRACK. RAck: 1250 1 INVITE. Content-Length: 0. . #----------------------------------line 180 -------------------------- U 10.30.0.248:5060 -> 10.0.0.27:5060 SIP/2.0 481 Call Leg/Transaction Does Not Exist. Via: SIP/2.0/UDP 10.0.0.27;branch=z9hG4bKdc2f.dbee15d1.0,SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bK8UeWxCfGrMJSO66f. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. CSeq: 2 PRACK. Content-Length: 0. . # U 10.0.0.27:5060 -> 10.0.0.46:5060 SIP/2.0 481 Call Leg/Transaction Does Not Exist. Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bK8UeWxCfGrMJSO66f. From: "9000" <sip:9000@10.0.0.27>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. CSeq: 2 PRACK. Content-Length: 0. . # U 10.30.0.248:5060 -> 10.0.0.27:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 10.0.0.27;branch=z9hG4bK0d2f.18572a04.0,SIP/2.0/UDP 10.0.0.46:5060;rport=5060;branch=z9hG4bKeKunMgjqbhggzOxi. From: "anonymous"<sip:36418473@10.0.0.46>;tag=l7473818dCOs9ZMG. To: "34163634" <sip:34163634@10.0.0.27>;tag=1EBAD768-B8F. Date: Wed, 04 Jan 2006 09:15:35 GMT. Call-ID: wVhSdbheOYfdmQmf(a)10.0.0.46. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 1 INVITE. Require: 100rel. RSeq: 1250. Allow-Events: telephone-event. Contact: <sip:934163634@10.30.0.248:5060>. Record-Route: <sip:34163634@10.0.0.27:5060;nat=yes;ftag=l7473818dCOs9ZMG;lr=on>. Content-Type: application/sdp. Content-Disposition: session;handling=required. Content-Length: 177. . v=0. o=CiscoSystemsSIP-GW-UserAgent 584 9780 IN IP4 10.30.0.248. s=SIP Call. c=IN IP4 10.30.0.248. t=0 0. m=audio 17340 RTP/AVP 18. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=yes.

3 5

[Users] 481 - Call leg Transaction
by unplug 20 Jan '06

20 Jan '06

In what case will cause the following error response. Does it happen in call forwarding? 481 - Call Leg Transcation does Not Exist Do I need to handle it or just ignore it?

2 4

[Users] only one call from duplicate register
by Ibrahim TUNALI 19 Jan '06

19 Jan '06

i know i can multiple register on a sip proxy with same account at same time from different locations. And both of them can call different numbers. But i want to only one client can call, other one will block during first one's conversation and second one will recive "not acceptable" response. is it possible and can you give a configuration example. regards, itunali

1 0

[Users] textops regular expression matching = subst to replace from field
by Neill Wilkinson 18 Jan '06

18 Jan '06

Hi all, Looking to get a way to replace the From field in a REGISTER message to the forward on to another Registrar, as a third party registration I've tried something along the lines of: If (subs("/(^From:)([ ]*)/\1 sip:newaddress@new.server/I")) log(1,"Match!\n"); else log(1,"No match\n"); I basically need help with the Regex syntax and a good means of checking - is egrep or sed a good means of testing? Thanks..... Neill...;o) --- avast! Antivirus: Outbound message clean. Virus Database (VPS): 0603-2, 17/01/2006 Tested on: 1/18/2006 11:16:18 avast! - copyright (c) 1988-2004 ALWIL Software. http://www.avast.com

2 4

[Users] call forward
by unplug 18 Jan '06

18 Jan '06

I am using the sample configuration file called "features-callfwd.5.0.cfg" with some modification. UA dials a 8 digits number and if it is not in the location table, prefix('9') will be appended in the number and route to the PSTN gateway. According to the configuration below, route 6 is responsible for call forward. With appropriate records in usr_preferences, openser can perform direct forward and forward on busy. However, in foward on no answer, one of the party can't hear any voice from the other party. I think the flow is something like this. UA1 openser UA5 UA6 ------INVITE----> --------INVITE------------> ... <-------time out/no answer- ------------------INVITE------------------> ... <--------------RTP established----------------------------------> result: UA6 can hear UA1, but UA1 can't get any sound from UA6 Anyone can help if there is any mistable in my configuration file to fix the above problem. Thanks! ------------start configuration------------- # $Id: features-callfwd.cfg 15 2005-09-15 17:37:10Z /CN=Greger V. Teigre/emailAddress=greger(a)onsip.org $ debug=3 fork=yes log_stderror=no listen=203.193.16.226 # INSERT YOUR IP ADDRESS HERE port=5060 children=4 dns=yes rev_dns=no fifo="/tmp/openser_fifo" fifo_db_url="mysql://openser:openserrw@localhost/openser" #fifo_mode=0666 #fifo permissions can be changed here log_facility=LOG_LOCAL0 alias=01.ol.com loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/sl.so" loadmodule "/usr/local/lib/openser/modules/tm.so" loadmodule "/usr/local/lib/openser/modules/rr.so" loadmodule "/usr/local/lib/openser/modules/maxfwd.so" loadmodule "/usr/local/lib/openser/modules/usrloc.so" loadmodule "/usr/local/lib/openser/modules/registrar.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" loadmodule "/usr/local/lib/openser/modules/textops.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/permissions.so" loadmodule "/usr/local/lib/openser/modules/acc.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/uac.so" loadmodule "/usr/local/lib/openser/modules/group.so" modparam("auth_db|permissions|uri_db|usrloc|acc|group", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy","natping_interval", 30) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/usr/local/etc/ser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/usr/local/etc/ser/rtp-clients") modparam("usrloc", "db_mode", 2) #minimize write back window - default is 60 seconds modparam("usrloc", "timer_interval", 10) #use deomain information for user #modparam("usrloc", "use_domain", 1) modparam("registrar", "nat_flag", 6) #The maximum expires value of a Contact modparam("registrar", "max_expires", 120) #all contacts will be ordered in descending modification time order modparam("registrar", "desc_time_order", 1) #modparam("registrar", "use_domain", 1) modparam("rr", "enable_full_lr", 1) #Timer which hits if no final reply for an INVITE modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("avpops", "avp_url", "mysql://openser:openserrw@localhost/openser") modparam("avpops", "avp_table", "usr_preferences") modparam("acc", "log_level", 1) modparam("acc", "log_flag", 1) modparam("acc", "db_flag", 1) modparam("uac", "from_restore_mode", "auto") route { xlog("L_INFO","---[$fU-MR]---\n"); # ----------------------------------------------------------------- # Sanity Check Section # ----------------------------------------------------------------- if (!mf_process_maxfwd_header("10")) { xlog("L_INFO","---[$fU-main(91)]: 483 Too Many Hops---\n"); sl_send_reply("483", "Too Many Hops"); return; }; if (msg:len > max_len) { xlog("L_INFO","---[$fU-main(97)]: 483 Message Overflow---\n"); sl_send_reply("513", "Message Overflow"); return; }; #----------------------------------------------------------------- # detect the anonymous incoming call and add anonymous to the uri #----------------------------------------------------------------- if (!has_totag() && (method=="INVITE"||method=="ACK"||method=="CANCEL"||method=="BYE")) { route(7); } #------------------------------------------------------------------ # Accoutnng Section #------------------------------------------------------------------ setflag(1); #t_relay(); # ----------------------------------------------------------------- # Record Route Section # ----------------------------------------------------------------- if (method=="INVITE" && client_nat_test("3")) { # INSERT YOUR IP ADDRESS HERE record_route_preset("203.193.16.226:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); }; # ----------------------------------------------------------------- # Call Tear Down Section # ----------------------------------------------------------------- if (method=="BYE" || method=="CANCEL") { end_media_session(); }; # ----------------------------------------------------------------- # Loose Route Section # ----------------------------------------------------------------- if (loose_route()) { xlog("L_INFO","---[$fU-main(137)]: loose_route()---\n"); if (has_totag() && (method=="INVITE" || method=="ACK")) { if (client_nat_test("3")||search("^Route:.*;nat=yes")){ setflag(6); use_media_proxy(); }; }; route(1); return; }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- #domian module if (!is_uri_host_local()) { if (is_from_local() || allow_trusted()) { route(4); route(1); } else { xlog("L_INFO","---[$fU-main(159)]: 403 Forbidden---\n"); sl_send_reply("403", "Forbidden"); }; return; }; if (method=="CANCEL") { route(1); return; } else if (method=="INVITE") { route(3); return; } else if (method=="REGISTER") { route(2); return; } else if (method=="ACK") { route(1); return; }; lookup("aliases"); #xlog("L_INFO","SIP line 191: [$fu] to [$tu]\n"); if (uri!=myself) { route(4); route(1); return; }; if (!lookup("location")) { xlog("L_INFO","---[$fU-main(188)]: 404 User Not Found---\n"); sl_send_reply("404", "User Not Found"); return; }; route(1); } route[1] { xlog("L_INFO","---[$fU-R1]---\n"); # ----------------------------------------------------------------- # Default Message Handler # ----------------------------------------------------------------- t_on_reply("1"); if (!t_relay()) { if (method=="INVITE" || method=="ACK") { end_media_session(); }; sl_reply_error(); }; #xlog("L_INFO","SIP Request line 220: [$rm] from [$fu] to [$tu]\n"); } route[2] { xlog("L_INFO","---[$fU-R2]---\n"); # ----------------------------------------------------------------- # REGISTER Message Handler # ----------------------------------------------------------------- sl_send_reply("100", "Trying"); if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if (!www_authorize("","subscriber")) { www_challenge("","0"); return; }; if (!check_to()) { xlog("L_INFO","---[$fU-route2(237)]: 401 Unauthorized---\n"); sl_send_reply("401", "Unauthorized"); return; }; #------------check if the account is activated ---------- if (!is_user_in("From", "active")) { xlog("L_INFO","---[$fU-route2(244)]: 402 Payment Required---\n"); sl_send_reply("402", "Payment Required"); #append_hf("P-hint: PAYMANT REQUIRED\r\n"); return; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { xlog("L_INFO","---[$fU-R3]---\n"); # ----------------------------------------------------------------- # INVITE Message Handler # ----------------------------------------------------------------- #if (!allow_trusted()) { #if (!proxy_authorize("","subscriber")) { #proxy_challenge("","0"); #return; #} else if (!check_from()) { #sl_send_reply("403", "Use From=ID"); #return; #}; #consume_credentials(); #}; if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; #if (uri=~"^sip:1[0-9]{10}@") { #strip(1); #}; lookup("aliases"); #xlog("L_INFO","SIP line 292: [$fu] to [$tu]\n"); if (uri!=myself) { route(4); route(1); return; }; #if (uri=~"^sip:011[0-9]*@") { #route(4); #route(5); #return; #}; if (avp_db_load("$ruri/username", "s:callfwd")) { xlog("L_INFO","---[$fU-route3(305)]: destination uri: [$ru]---\n"); setflag(22); avp_pushto("$ruri", "s:callfwd"); xlog("L_INFO","---[$fU-route3(308)]: destination uri: [$ru]---\n"); route(6); return; }; if (!lookup("location")) { xlog("L_INFO","---[$fU]-(route3:315) destination uri: [$du]---\n"); if (uri=~"^sip:[0-9]*@") { xlog("L_INFO","---[$fU-route3(317)]: sip:(0-9*@)---\n"); prefix("9"); route(4); route(5); return; }; xlog("L_INFO","---[$fU-route3(324)]: 404 User Not Found---\n"); sl_send_reply("404", "User Not Found"); return; }; if (avp_db_load("$ruri/username", "s:fwdbusy")) { if (!avp_check("s:fwdbusy", "eq/$ruri/i")) { xlog("L_INFO","---[$fU-route3(331)]: setflag(26)---\n"); setflag(26); }; }; if (avp_db_load("$ruri/username", "s:fwdnoanswer")) { if (!avp_check("s:fwdnoanswer", "eq/$ruri/i")) { xlog("L_INFO","---[$fU-route3(338)]: setflag(27)---\n"); setflag(27); }; }; t_on_failure("1"); route(4); route(1); } route[4] { xlog("L_INFO","---[$fU-R4]---\n"); # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- if (isflagset(6) || isflagset(7)) { xlog("L_INFO","---[$fU-route4(357)]: $ru---\n"); if (!isflagset(8)) { xlog("L_INFO","---[$fU-route4(359)]: $ru---\n"); setflag(8); use_media_proxy(); }; }; } route[5] { xlog("L_INFO","---[$fU-R5]---\n"); # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehost("203.193.16.242"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); t_on_failure("1"); route(4); route(1); } route[6] { xlog("L_INFO","---[$fU-R6]---\n"); # ------------------------------------------------------------------------ # Call Forwarding Reply Route Handler # # This must be done as a route block because sl_send_reply() cannot be # called from the failure_route block # ------------------------------------------------------------------------ #if (uri=~"^sip:1[0-9]{10}@") { #strip(1); #}; lookup("aliases"); if (!isflagset(22)) { append_branch(); }; if (uri!=myself) { xlog("L_INFO","---[$fU-route6(405)]: (uri!=myyself)---\n"); route(4); route(1); return; }; #if (uri=~"^sip:011[0-9]*@") { } #if (uri=~"^sip:[0-9]*@") { #xlog("L_INFO","---[$fU-route6(415)]: (app prefix(9))---\n"); #prefix("9"); #route(4); #route(5); #return; #}; if (!lookup("location")) { if (uri=~"^sip:[0-9]*@") { xlog("L_INFO","---[$fU-route6(425)]: (sip:(0-9*@)---\n"); route(4); route(1); return; }; xlog("L_INFO","---[$fU-route6(432)]: 404 User Not Found---\n"); sl_send_reply("404", "User Not Found"); }; route(4); route(1); } route[7] { xlog("L_INFO","---[$fU-R7]---\n"); #if (method=="INVITE"||method=="ACK"||method=="CANCEL"||method=="BYE") { #if (is_user_in("from","")) { #xlog("L_INFO","SIP is_user_in=true\n"); #uac_replace_from("anonymous","sip:anonymous@s01.ol.com"); #} #} if (avp_db_load("$from/username","s:alias")) { xlog("L_INFO","---[$fU-route7(450)]: have alias from aliases- [$avp(s:alias)]---\n"); #uac_replace_from("$avp(s:alias)","sip:$avp(s:alias)@$si"); #uac_replace_from("$avp(s:alias) " ,""); } else { xlog("L_INFO","---[$fU-route7(454)]: no alias from aliases---\n"); }; } onreply_route[1] { xlog("L_INFO","---[$fU-OR1]---\n"); if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } failure_route[1] { xlog("L_INFO","---[$fU-FR1]---\n"); if (t_check_status("487")) { return; }; if (isflagset(26) && t_check_status("486")) { if (avp_pushto("$ruri", "s:fwdbusy")) { xlog("L_INFO","---[$fU-failure_route1(505)]: fwdbusy---\n"); avp_delete("s:fwdbusy"); resetflag(26); route(6); return; }; }; if (isflagset(27) && t_check_status("408")) { if (avp_pushto("$ruri", "s:fwdnoanswer")) { xlog("L_INFO","---[$fU-failure_route1(515)]: fwdnoanswer---\n"); avp_delete("s:fwdnoanswer"); resetflag(27); route(6); return; }; }; end_media_session(); } -----------end of configuration----------------

1 0

[Users] How to check destination set after branch_route?
by Papadopoulos Georgios 18 Jan '06

18 Jan '06

Hello, I am trying to filter out requests going to some IPs. So, I have a branch_route block where I drop the branches that go to the IPs I want to block. After the t_on_branch command I call t_relay. This works fine if the callee has registered with at least one "good" IP. If the callee has registered with only one IP which I want to block then there is only one branch that is getting droped. The following t_relay results in a 500 Server error. Is there a way to check if all branches have been dropped? I tried using isdsturiset() but it did not work. Even if there is only one branch and this branch has been dropped, isdsturiset() returns true. Is this correct or am I wrong somewhere? thank you for any help George Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses.

2 1

[Users] Sending stored messages
by Andréa Giordanna 18 Jan '06

18 Jan '06

Hi Daniel, Thanks for all the informations. Based on those informations, Loide Mara (lmvj(a)dcc.ufam.edu.br) and I have made changes on msilo module code to make it able to send messages at a time stored in the silo table. Applying the patch patch-msilo-060117.diff the msilo module becomes able to send messages at the time stored on the table. Following the installation instructions for msilo module in http://openser.org/docs/modules/1.0.x/msilo.html#AEN199, it is just necessary to add a field in the silo table: sen_time (integer). This field will store the sending time of the stored messages. For offline messages, sending time equals incoming time. It can be useful if users are interested in store messages to be sent to one or more different users at a given time. Reminders for meetings or special dates could be sent through it. Here is a patch if someone is interested in testing it. Any comments will be appreciated. Regards, Andrea Giordanna --------------------------------- Yahoo! doce lar. Faça do Yahoo! sua homepage. diff -urN msilo_orig/msilo.c msilo/msilo.c --- msilo_orig/msilo.c 2006-01-17 19:55:34.000000000 -0400 +++ msilo/msilo.c 2006-01-17 19:57:25.000000000 -0400 @@ -71,7 +71,7 @@ #include "msfuncs.h" #define MAX_DEL_KEYS 1 -#define NR_KEYS 9 +#define NR_KEYS 10 /* LabVoIP: old value 9 */ char *sc_mid = "mid"; /* 0 */ char *sc_from = "src_addr"; /* 1 */ @@ -82,6 +82,7 @@ char *sc_ctype = "ctype"; /* 6 */ char *sc_exp_time = "exp_time"; /* 7 */ char *sc_inc_time = "inc_time"; /* 8 */ +char *sc_sen_time = "sen_time"; /* LabVoIP: 9 */ #define SET_STR_VAL(_str, _res, _r, _c) \ if (RES_ROWS(_res)[_r].values[_c].nul == 0) \ @@ -144,6 +145,7 @@ void destroy(void); void m_clean_silo(unsigned int ticks, void *); +void m_send_ready(); /* LabVoIP */ /** TM callback function */ static void m_tm_callback( struct cell *t, int type, struct tmcb_params *ps); @@ -245,6 +247,8 @@ } register_timer( m_clean_silo, 0, ms_check_time); + register_timer( m_send_ready, 0, ms_check_time); /* LabVoIP */ + reg_addr.s = ms_registrar; reg_addr.len = (ms_registrar)?strlen(ms_registrar):0; @@ -553,6 +557,13 @@ db_vals[nr_keys].val.int_val = val; nr_keys++; + /* LabVoIP: add sending time: equals incoming time */ + db_keys[nr_keys] = sc_inc_time; + db_vals[nr_keys].type = DB_INT; + db_vals[nr_keys].nul = 0; + db_vals[nr_keys].val.int_val = val; + nr_keys++; + if (msilo_dbf.use_table(db_con, ms_db_table) < 0) { LOG(L_ERR, "MSILO:m_store: Error in use_table\n"); @@ -639,9 +650,9 @@ struct to_body to, *pto = NULL; db_key_t db_keys[2]; db_val_t db_vals[2]; - db_key_t db_cols[6]; + db_key_t db_cols[7]; db_res_t* db_res = NULL; - int i, db_no_cols = 6, db_no_keys = 2, mid, n; + int i, db_no_cols = 7, db_no_keys = 2, mid, n; char hdr_buf[1024], body_buf[1024]; struct sip_uri puri; @@ -658,6 +669,7 @@ db_cols[3]=sc_body; db_cols[4]=sc_ctype; db_cols[5]=sc_inc_time; + db_cols[6]=sc_sen_time; DBG("MSILO:m_dump: ------------ start ------------\n"); @@ -759,48 +771,62 @@ DBG("MSILO:m_dump: dumping [%d] messages for <%.*s>!!!\n", RES_ROW_N(db_res), pto->uri.len, pto->uri.s); + int cur_time = (int) time(NULL); + for(i = 0; i < RES_ROW_N(db_res); i++) { mid = RES_ROWS(db_res)[i].values[0].val.int_val; - if(msg_list_check_msg(ml, mid)) - { - DBG("MSILO:m_dump: message[%d] mid=%d already sent.\n", - i, mid); - continue; - } + + if (RES_ROWS(db_res)[i].values[6].val.int_val > cur_time) { + DBG("MSILO:m_send_ready: message[%d] mid=%d not sent. Not ready!!!\nCurrent time: %d - Sending time: %d\n", i, mid, cur_time, RES_ROWS(db_res)[i].values[6].val.int_val); + } else { + + DBG("MSILO:m_send_ready: message[%d] mid=%d sent. Ready!!!\nCurrent time: %d - Sending time: %d\n", i, mid, cur_time, RES_ROWS(db_res)[i].values[6].val.int_val); + + if(msg_list_check_msg(ml, mid)) + { + DBG("MSILO:m_dump: message[%d] mid=%d already sent.\n", i, mid); + + if (msg_list_get_flag(ml, mid) & MS_MSG_ERRO) { + DBG("MSILO:m_dump: message[%d] mid=%d is a message to an offline user, or was not sent successfully. We're gonna send it again.\n", i, mid); + } else { + DBG("MSILO:m_dump: message[%d] mid=%d was sent successfully. Next...\n", i, mid); + continue; + } + } - memset(str_vals, 0, 4*sizeof(str)); - SET_STR_VAL(str_vals[0], db_res, i, 1); /* from */ - SET_STR_VAL(str_vals[1], db_res, i, 2); /* to */ - SET_STR_VAL(str_vals[2], db_res, i, 3); /* body */ - SET_STR_VAL(str_vals[3], db_res, i, 4); /* ctype */ + memset(str_vals, 0, 4*sizeof(str)); + SET_STR_VAL(str_vals[0], db_res, i, 1); /* from */ + SET_STR_VAL(str_vals[1], db_res, i, 2); /* to */ + SET_STR_VAL(str_vals[2], db_res, i, 3); /* body */ + SET_STR_VAL(str_vals[3], db_res, i, 4); /* ctype */ - hdr_str.len = 1024; - if(m_build_headers(&hdr_str, str_vals[3] /*ctype*/, + hdr_str.len = 1024; + if(m_build_headers(&hdr_str, str_vals[3] /*ctype*/, str_vals[0]/*from*/) < 0) - { - DBG("MSILO:m_dump: headers building failed!!!\n"); - if (msilo_dbf.free_result(db_con, db_res) < 0) - DBG("MSILO:m_dump: Error while freeing result of" + { + DBG("MSILO:m_dump: headers building failed!!!\n"); + if (msilo_dbf.free_result(db_con, db_res) < 0) + DBG("MSILO:m_dump: Error while freeing result of" " query\n"); - msg_list_set_flag(ml, mid, MS_MSG_ERRO); - goto error; - } + msg_list_set_flag(ml, mid, MS_MSG_ERRO); + goto error; + } - DBG("MSILO:m_dump: msg [%d-%d] for: %.*s\n", i+1, mid, + DBG("MSILO:m_dump: msg [%d-%d] for: %.*s\n", i+1, mid, pto->uri.len, pto->uri.s); - /** sending using TM function: t_uac */ - body_str.len = 1024; - rtime = - (time_t)RES_ROWS(db_res)[i].values[5/*inc time*/].val.int_val; - n = m_build_body(&body_str, rtime, str_vals[2/*body*/]); - if(n<0) - DBG("MSILO:m_dump: sending simple body\n"); - else - DBG("MSILO:m_dump: sending composed body\n"); + /** sending using TM function: t_uac */ + body_str.len = 1024; + rtime = + (time_t)RES_ROWS(db_res)[i].values[5/*inc time*/].val.int_val; + n = m_build_body(&body_str, rtime, str_vals[2/*body*/]); + if(n<0) + DBG("MSILO:m_dump: sending simple body\n"); + else + DBG("MSILO:m_dump: sending composed body\n"); - tmb.t_request(&msg_type, /* Type of the message */ + tmb.t_request(&msg_type, /* Type of the message */ &pto->uri, /* Request-URI */ &str_vals[1], /* To */ &str_vals[0], /* From */ @@ -809,6 +835,7 @@ m_tm_callback, /* Callback function */ (void*)(long)mid /* Callback parameter */ ); + } } done: @@ -900,7 +927,7 @@ */ void m_tm_callback( struct cell *t, int type, struct tmcb_params *ps) { - DBG("MSILO:m_tm_callback: completed with status %d\n", ps->code); + DBG("MSILO:m_tm_callback: message %ld completed with status %d\n", (long)*ps->param, ps->code); if(!ps->param) { DBG("MSILO m_tm_callback: message id not received\n"); @@ -914,14 +941,155 @@ if(ps->code < 200 || ps->code >= 300) { DBG("MSILO:m_tm_callback: message <%ld> was not sent successfully\n", - (long)ps->param); - msg_list_set_flag(ml, (int)(long)ps->param, MS_MSG_ERRO); + (long)*ps->param); + msg_list_set_flag(ml, (int)(long)*ps->param, MS_MSG_ERRO); goto done; } - msg_list_set_flag(ml, (int)(long)ps->param, MS_MSG_DONE); + msg_list_set_flag(ml, (int)(long)*ps->param, MS_MSG_DONE); done: return; } +/* LabVoIP: m_send_ready + * It checks on database if there are any messages in time to be sent + * - giordanna + */ + +void m_send_ready () { + + struct to_body to, *pto = NULL; + db_key_t db_keys[2]; + db_key_t db_cols[7]; + db_res_t* db_res = NULL; + int i, db_no_cols = 7, mid, n; + char hdr_buf[1024], body_buf[1024]; + + str str_vals[4], hdr_str , body_str; + time_t rtime; + + /* init */ + db_keys[0]=sc_uri_user; + db_keys[1]=sc_uri_host; + db_cols[0]=sc_mid; + + db_cols[1]=sc_from; + db_cols[2]=sc_to; + db_cols[3]=sc_body; + db_cols[4]=sc_ctype; + db_cols[5]=sc_inc_time; + db_cols[6]=sc_sen_time; + + + DBG("MSILO:m_send_ready: ------------ Checking for messages to send! ------------\n"); + hdr_str.s=hdr_buf; + hdr_str.len=1024; + body_str.s=body_buf; + body_str.len=1024; + + + if (msilo_dbf.use_table(db_con, ms_db_table) < 0) + { + LOG(L_ERR, "MSILO:m_send_ready: Error in use_table\n"); + goto error; + } + + if((msilo_dbf.query(db_con,db_keys,NULL,NULL,db_cols,0, + db_no_cols, NULL,&db_res)!=0) || (RES_ROW_N(db_res) <= 0)) + { + DBG("MSILO:m_send_ready: no messages in database!\n"); + goto done; + } + + DBG("MSILO:m_send_ready: dumping [%d] messages!!!\n", RES_ROW_N(db_res)); + + + memset( &to , 0, sizeof(to) ); + + int cur_time = (int) time(NULL); + + /* Loop to send all the messages */ + for(i = 0; i < RES_ROW_N(db_res); i++) + { + // Getting message id + mid = RES_ROWS(db_res)[i].values[0].val.int_val; + + // If it is time to send... + if (RES_ROWS(db_res)[i].values[6].val.int_val > cur_time) { + DBG("MSILO:m_send_ready: message[%d] mid=%d not sent. Not ready!!!\nCurrent time: %d - Sending time: %d\n", i, mid, cur_time, RES_ROWS(db_res)[i].values[6].val.int_val); + } else { + + DBG("MSILO:m_send_ready: message[%d] mid=%d sent. Ready!!!\nCurrent time: %d - Sending time: %d\n", i, mid, cur_time, RES_ROWS(db_res)[i].values[6].val.int_val); + + // To field + to.uri.s = (char *) RES_ROWS(db_res)[i].values[2].val.string_val; + to.uri.len = strlen(RES_ROWS(db_res)[i].values[2].val.string_val); + if (to.uri.len <= 0) /* || to.error != PARSE_OK) */ + { + DBG("MSILO:m_send_ready: 'To' erro na atribui�o!\n"); + goto error; + } + pto = &to; + + if(msg_list_check_msg(ml, mid)) + { + DBG("MSILO:m_send_ready: message[%d] mid=%d already sent.\n", i, mid); + continue; + } + + memset(str_vals, 0, 4*sizeof(str)); + SET_STR_VAL(str_vals[0], db_res, i, 1); /* from */ + SET_STR_VAL(str_vals[1], db_res, i, 2); /* to */ + SET_STR_VAL(str_vals[2], db_res, i, 3); /* body */ + SET_STR_VAL(str_vals[3], db_res, i, 4); /* ctype */ + + hdr_str.len = 1024; + if(m_build_headers(&hdr_str, str_vals[3] /*ctype*/, + str_vals[0]/*from*/) < 0) + { + DBG("MSILO:m_send_ready: headers building failed!!!\n"); + if (msilo_dbf.free_result(db_con, db_res) < 0) + DBG("MSILO:m_send_ready: Error while freeing result of" + " query\n"); + msg_list_set_flag(ml, mid, MS_MSG_ERRO); + goto error; + } + + DBG("MSILO:m_send_ready: msg [%d-%d] for: %.*s\n", + i+1, mid, pto->uri.len, pto->uri.s); + + /** sending using TM function: t_uac */ + body_str.len = 1024; + rtime = + (time_t)RES_ROWS(db_res)[i].values[5/*inc time*/].val.int_val; + n = m_build_body(&body_str, rtime, str_vals[2/*body*/]); + if(n<0) + DBG("MSILO:m_send_ready: sending simple body\n"); + else + DBG("MSILO:m_send_ready: sending composed body\n"); + + tmb.t_request(&msg_type, /* Type of the message */ + &pto->uri, /* Request-URI */ + &str_vals[1], /* To */ + &str_vals[0], /* From */ + &hdr_str, /* Optional headers including CRLF */ + (n<0)?&str_vals[2]:&body_str, /* Message body */ + m_tm_callback, /* Callback function */ + (void*)(long) mid /* Callback parameter */ + ); + } + } + +done: + /** + * Free the result because we don't need it + * anymore + */ + if (db_res!=NULL && msilo_dbf.free_result(db_con, db_res) < 0) + DBG("MSILO:m_send_ready: Error while freeing result of query\n"); + + return; +error: + return; +} diff -urN msilo_orig/ms_msg_list.c msilo/ms_msg_list.c --- msilo_orig/ms_msg_list.c 2006-01-17 19:55:34.000000000 -0400 +++ msilo/ms_msg_list.c 2006-01-17 19:57:30.000000000 -0400 @@ -244,6 +244,37 @@ } /** + * LabVoIP: + * get flag for message with mid + */ +int msg_list_get_flag(msg_list ml, int mid) +{ + msg_list_el p0; + + if(!ml || mid==0) + goto errorx; + + lock_get(&ml->sem_sent); + + p0 = ml->lsent; + while(p0) + { + if(p0->msgid==mid) + { + DBG("MSILO: msg_list_get_flag: mid:%d fl:%d\n", p0->msgid, p0->flag); + goto done; + } + p0 = p0->next; + } + +done: + lock_release(&ml->sem_sent); + return p0->flag; +errorx: + return -1; +} + +/** * check if the messages from list were sent */ int msg_list_check(msg_list ml) @@ -262,9 +293,9 @@ p0 = ml->lsent; while(p0) { - if(p0->flag & MS_MSG_DONE || p0->flag & MS_MSG_ERRO) + if(p0->flag & MS_MSG_DONE) /*|| p0->flag & MS_MSG_ERRO)*/ /* Messages with error will be maintained in sent_list*/ { - DBG("MSILO: msg_list_check: mid:%d got reply\n", p0->msgid); + DBG("MSILO: msg_list_check: mid:%d got positive reply\n", p0->msgid); if(p0->prev) (p0->prev)->next = p0->next; else diff -urN msilo_orig/ms_msg_list.h msilo/ms_msg_list.h --- msilo_orig/ms_msg_list.h 2006-01-17 19:55:34.000000000 -0400 +++ msilo/ms_msg_list.h 2006-01-17 19:57:33.000000000 -0400 @@ -70,6 +70,7 @@ void msg_list_free(msg_list); int msg_list_check_msg(msg_list, int); int msg_list_set_flag(msg_list, int, int); +int msg_list_get_flag(msg_list ml, int mid); /* LabVoIP */ int msg_list_check(msg_list); msg_list_el msg_list_reset(msg_list);

2 1

[Users] NAT Problem?
by Craig Mariner 17 Jan '06

17 Jan '06

Hi List I have a problem with OpenSER. OpenSER has a public IP and is not behind a NAT. Two clients that are behind a NAT ring each other but there is no audio either way. Any advise please? Sencond scenario two clients both behind the same NAT. . Client1 calls the Client2, Client2 UA rings, Client2 answers the call but it does not connect. Client1 call does not get answered by Client2 and Client1 just hears the ringing tone. Any ideas? Best Regards -- Craig Mariner Director Wifi Specialist Ltd www.wifispecialist.co.uk www.freewhy-fi.net Building a Free Mesh Network Tel. 08703403427 email: wifibase(a)gmail.com

2 2

[Users] infinite register loop with eyeBeam when call forking one an SER proxy
by Dr. Rodney G. McDuff 17 Jan '06

17 Jan '06

I am using SER 0.9.4 as a sip proxy. In the proxy I have setup a permanent user location pointing to our cisco sip/PSTN gateway; the idea being that when I ring my sip address it rings both my PSTN phone and my sip client. When I use X-Lite it all works perfectly. X-Lite REGISTERs a couple of times and when I INVITE someone sip URI both phones rings. First phone that asnwers get the prize all others get a CANCEL. When I use eyebeam [eyeBeam release 3010n stamp 19039] all I get is a seemingly infinite loop of REGISTERs with incrementing Cseq. Below is a ngrep at the proxy. Has anyone seen this behavior and known if its a bug with eyeBeam or the SER config? T xxx.xxx.152.78:1758 -> xxx.xxx.2.40:5060 [AP] REGISTER sip:sip.domain SIP/2.0. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-4152d5791948a902-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:username@xxx.xxx.152.78:1758>. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 375 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: eventlist. User-Agent: eyeBeam release 3010n stamp 19039. Authorization: Digest username="username",realm="sip.domain",nonce="43cc8e5a5cb47d54a7c640016623a782b1206e3b",uri="sip:sip.domain",response="cdae7a3787b5539f61c1b20087efc3e0",algorithm=MD5. Content-Length: 0. . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 100 Trying. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-4152d5791948a902-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 375 REGISTER. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-4152d5791948a902-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=b5c61af9a6130f64fdf89dc761547723.a060. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 375 REGISTER. Contact: <sip:58220@sip.pstn.gw.ip>;q=1;expires=4294965840, <sip:username@xxx.xxx.152.57:5060>;expires=503;received="sip:xxx.xxx.152.57:5060", <sip:username@xxx.xxx.152.78:5060>;expires=1356, <sip:username@xxx.xxx.152.78:1758>;expires=3600. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . ## T xxx.xxx.152.78:1758 -> xxx.xxx.2.40:5060 [AP] REGISTER sip:sip.domain SIP/2.0. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-3d42141dce2ee012-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:username@xxx.xxx.152.78:1758>. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 376 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: eventlist. User-Agent: eyeBeam release 3010n stamp 19039. Authorization: Digest username="username",realm="sip.domain",nonce="43cc8e5a5cb47d54a7c640016623a782b1206e3b",uri="sip:sip.domain",response="cdae7a3787b5539f61c1b20087efc3e0",algorithm=MD5. Content-Length: 0. . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 100 Trying. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-3d42141dce2ee012-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 376 REGISTER. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-3d42141dce2ee012-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=b5c61af9a6130f64fdf89dc761547723.80b3. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 376 REGISTER. Contact: <sip:58220@sip.pstn.gw.ip>;q=1;expires=4294965840, <sip:username@xxx.xxx.152.57:5060>;expires=503;received="sip:xxx.xxx.152.57:5060", <sip:username@xxx.xxx.152.78:5060>;expires=1356, <sip:username@xxx.xxx.152.78:1758>;expires=3600. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . ## T xxx.xxx.152.78:1758 -> xxx.xxx.2.40:5060 [AP] REGISTER sip:sip.domain SIP/2.0. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-5861a63d14707c05-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:username@xxx.xxx.152.78:1758>. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 377 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: eventlist. User-Agent: eyeBeam release 3010n stamp 19039. Authorization: Digest username="username",realm="sip.domain",nonce="43cc8e5a5cb47d54a7c640016623a782b1206e3b",uri="sip:sip.domain",response="cdae7a3787b5539f61c1b20087efc3e0",algorithm=MD5. Content-Length: 0. . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 100 Trying. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-5861a63d14707c05-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 377 REGISTER. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-5861a63d14707c05-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=b5c61af9a6130f64fdf89dc761547723.2412. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 377 REGISTER. Contact: <sip:58220@sip.pstn.gw.ip>;q=1;expires=4294965839, <sip:username@xxx.xxx.152.57:5060>;expires=502;received="sip:xxx.xxx.152.57:5060", <sip:username@xxx.xxx.152.78:5060>;expires=1355, <sip:username@xxx.xxx.152.78:1758>;expires=3600. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . ## U xxx.xxx.152.78:5060 -> xxx.xxx.2.40:5060 . # T xxx.xxx.152.78:1758 -> xxx.xxx.2.40:5060 [AP] REGISTER sip:sip.domain SIP/2.0. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-4675e1000b2e3f40-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:username@xxx.xxx.152.78:1758>. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 378 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: eventlist. User-Agent: eyeBeam release 3010n stamp 19039. Authorization: Digest username="username",realm="sip.domain",nonce="43cc8e5a5cb47d54a7c640016623a782b1206e3b",uri="sip:sip.domain",response="cdae7a3787b5539f61c1b20087efc3e0",algorithm=MD5. Content-Length: 0. . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 100 Trying. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-4675e1000b2e3f40-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 378 REGISTER. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-4675e1000b2e3f40-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=b5c61af9a6130f64fdf89dc761547723.9461. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 378 REGISTER. Contact: <sip:58220@sip.pstn.gw.ip>;q=1;expires=4294965839, <sip:username@xxx.xxx.152.57:5060>;expires=502;received="sip:xxx.xxx.152.57:5060", <sip:username@xxx.xxx.152.78:5060>;expires=1355, <sip:username@xxx.xxx.152.78:1758>;expires=3600. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . ## T xxx.xxx.152.78:1758 -> xxx.xxx.2.40:5060 [AP] REGISTER sip:sip.domain SIP/2.0. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-5f45fd45452d0a56-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:username@xxx.xxx.152.78:1758>. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 379 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: eventlist. User-Agent: eyeBeam release 3010n stamp 19039. Authorization: Digest username="username",realm="sip.domain",nonce="43cc8e5a5cb47d54a7c640016623a782b1206e3b",uri="sip:sip.domain",response="cdae7a3787b5539f61c1b20087efc3e0",algorithm=MD5. Content-Length: 0. . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 100 Trying. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-5f45fd45452d0a56-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 379 REGISTER. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-5f45fd45452d0a56-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=b5c61af9a6130f64fdf89dc761547723.7aca. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 379 REGISTER. Contact: <sip:58220@sip.pstn.gw.ip>;q=1;expires=4294965839, <sip:username@xxx.xxx.152.57:5060>;expires=502;received="sip:xxx.xxx.152.57:5060", <sip:username@xxx.xxx.152.78:5060>;expires=1355, <sip:username@xxx.xxx.152.78:1758>;expires=3600. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . ## T xxx.xxx.152.78:1758 -> xxx.xxx.2.40:5060 [AP] REGISTER sip:sip.domain SIP/2.0. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-277dc139635ed406-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:username@xxx.xxx.152.78:1758>. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 380 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: eventlist. User-Agent: eyeBeam release 3010n stamp 19039. Authorization: Digest username="username",realm="sip.domain",nonce="43cc8e5a5cb47d54a7c640016623a782b1206e3b",uri="sip:sip.domain",response="cdae7a3787b5539f61c1b20087efc3e0",algorithm=MD5. Content-Length: 0. . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 100 Trying. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-277dc139635ed406-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 380 REGISTER. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-277dc139635ed406-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=b5c61af9a6130f64fdf89dc761547723.5668. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 380 REGISTER. Contact: <sip:58220@sip.pstn.gw.ip>;q=1;expires=4294965839, <sip:username@xxx.xxx.152.57:5060>;expires=502;received="sip:xxx.xxx.152.57:5060", <sip:username@xxx.xxx.152.78:5060>;expires=1355, <sip:username@xxx.xxx.152.78:1758>;expires=3600. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . ## T xxx.xxx.152.78:1758 -> xxx.xxx.2.40:5060 [AP] REGISTER sip:sip.domain SIP/2.0. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-3e3eaf4bb77a4835-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:username@xxx.xxx.152.78:1758>. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 381 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Supported: eventlist. User-Agent: eyeBeam release 3010n stamp 19039. Authorization: Digest username="username",realm="sip.domain",nonce="43cc8e5a5cb47d54a7c640016623a782b1206e3b",uri="sip:sip.domain",response="cdae7a3787b5539f61c1b20087efc3e0",algorithm=MD5. Content-Length: 0. . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 100 Trying. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-3e3eaf4bb77a4835-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 381 REGISTER. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . # T xxx.xxx.2.40:5060 -> xxx.xxx.152.78:1758 [AP] SIP/2.0 200 OK. Via: SIP/2.0/TCP xxx.xxx.152.78:8970;branch=z9hG4bK-d87543-3e3eaf4bb77a4835-1--d87543-;rport=1758. To: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=b5c61af9a6130f64fdf89dc761547723.641b. From: "Unhappy eyeBeam Client"<sip:username@sip.domain>;tag=5b23e370. Call-ID: 480f10142f438b44@c3RnLWRlbGw.. CSeq: 381 REGISTER. Contact: <sip:58220@sip.pstn.gw.ip>;q=1;expires=4294965839, <sip:username@xxx.xxx.152.57:5060>;expires=502;received="sip:xxx.xxx.152.57:5060", <sip:username@xxx.xxx.152.78:5060>;expires=1355, <sip:username@xxx.xxx.152.78:1758>;expires=3600. Server: Sip EXpress router (0.9.4 (i386/linux)). Content-Length: 0. Warning: 392 xxx.xxx.2.40:5060 "Noisy feedback tells: pid=11879 req_src_ip=xxx.xxx.152.78 req_src_port=1758 in_uri=sip:sip.domain out_uri=sip:sip.domain via_cnt==1". . -- Dr. Rodney G. McDuff |Ex ignorantia ad sapientiam Manager, Strategic Technologies Group| Ex luce ad tenebras Information Technology Services | The University of Queensland | EMAIL: mcduff(a)its.uq.edu.au | TELEPHONE: +61 7 3365 8220 |

2 1

[Users] requirement of using mediaproxy
by unplug 17 Jan '06

17 Jan '06

I am using the features-callfwd.cfg script with mediaproxy in a testing server. I can make call directly from phone to phone and forward call. However, it is failed to forward call on no answer and forward call on busy. In my testing server, it only has a network interface. I wonder if it is the cause of the problem. Does it need to have 2 network interfaces (one for external and one for internal) for those forward features work?

2 1

[Users] openser+freeradius
by Henri Keski-Sikkilä 16 Jan '06

16 Jan '06

I installed freeradius. When I tried to register, syslog message is: Jan 14 20:39:36 localhost /usr/sbin/openser[1280]: rc_get_seqnbr: fscanf failure: /var/run/radius.seq Jan 14 20:39:38 localhost /usr/sbin/openser[1280]: ERROR:auth_radius:radius_authorize_sterman: rc_auth failed Any ideas? Henri

3 3

[Users] database description
by Daniel-Constantin Mierla 16 Jan '06

16 Jan '06

Hello, courtesy of Norman Brandinger, there is now a new tutorial describing the tables in OpenSER Database. http://openser.org/docs/OpenSER_Database_Structure.html The dokuwiki page which was create to collect users' contributions in this respect will point to this tutorial. If anyone wants to contribute with more description, please sent email to devel(a)openser.org or add the text in the dokiwiki page. http://openser.org/dokuwiki/doku.php?id=database_description Cheers, Daniel

1 0

[Users] How to use the failover support in the dispatcher module?
by Wolfgang Hottgenroth 16 Jan '06

16 Jan '06

Hi, on 18. 12. 2005 Daniel-Constantin Mierla accounted on this list that the dispatcher module now has a kind of failover support. I was really curious for this feature, since I had to config openser to support both load-balancing and failover. Failover works fine: if one PSTNGW is not available, the next one is tried, until one works. But unfortunately, for negative status reports (like 'busy'), the failure_route is also entered and thereby also the failover mechanism is triggered, with the result that a 'busy' is signalled to the A-side as 'no more gws'. (I've tried different mechanisms for ds_select_domain, also I tried with and without the append_branch in the failure_route block.) Any hints for me? Thank you very much! Wolfgang debug=3 fork=yes log_stderror=yes # (cmd line: -E) 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 ---------------------------------- loadmodule "/opt/openser/lib/openser/modules/sl.so" loadmodule "/opt/openser/lib/openser/modules/tm.so" loadmodule "/opt/openser/lib/openser/modules/rr.so" loadmodule "/opt/openser/lib/openser/modules/maxfwd.so" loadmodule "/opt/openser/lib/openser/modules/usrloc.so" loadmodule "/opt/openser/lib/openser/modules/registrar.so" loadmodule "/opt/openser/lib/openser/modules/textops.so" loadmodule "/opt/openser/lib/openser/modules/xlog.so" loadmodule "/opt/openser/lib/openser/modules/dispatcher.so" # ----------------- setting module-specific parameters --------------- modparam("rr", "enable_full_lr", 1) modparam("xlog", "buf_size", 8192) modparam("dispatcher", "list_file", "/opt/openser/etc/openser/dispatcher.list") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("tm", "fr_timer", 5) # ------------------------- request routing logic ------------------- # main routing logic route[0] { xlog("L_NOTICE", "----> before route 1 <----\n"); route(1); xlog("L_NOTICE", "----> after route 1 <----\n"); } route[1] { if (loose_route()) { xlog("L_NOTICE", "*** loose routing enforced: $rm, $ru, $si, $du\n"); t_relay(); return; } record_route(); xlog("L_NOTICE", "REQUEST: $rm, $ru, $si, $du\n"); if (method=="REGISTER") { xlog("L_NOTICE", "*** register request from $si\n"); sl_send_reply("503", "No registrar service here"); return; } if (method=="INVITE" && ! uri=~"^sip:\+[1-9][0-9]+@") { xlog("L_NOTICE", "*** illegal sip uri: $ru\n"); sl_send_reply("484", "International number required"); return; } # --- customer specific part --------------------------------------- if ( src_ip==62.191.185.122 ) { xlog("L_NOTICE", "*** $rm from laptop-who, $si\n"); append_hf("P-Hint: from laptop-who\n"); append_hf("Remote-Party-ID: <sip:+492319721231@62.191.185.47>;party=calling;screen=yes;privacy=off\n"); if (method=="INVITE") { strip(1); prefix("+0000666666"); } ds_select_domain("00", "4"); if (method=="INVITE") { t_on_failure("2"); } xlog("L_NOTICE", "----> before t_relay <----\n"); t_relay(); xlog("L_NOTICE", "----> after t_relay <----\n"); return; } if ( src_ip==62.191.185.29 ) { xlog("L_NOTICE", "*** $rm from laptop-aholt, $si\n"); append_hf("P-Hint: from laptop-aholt\n"); append_hf("Remote-Party-ID: <sip:+492319721231@62.191.185.47>;party=calling;screen=yes;privacy=off\n"); if (method=="INVITE") { strip(1); prefix("+0100555555"); } ds_select_domain("01", "4"); if (method=="INVITE") { t_on_failure("2"); } xlog("L_NOTICE", "----> before t_relay <----\n"); t_relay(); xlog("L_NOTICE", "----> after t_relay <----\n"); return; } if ( src_ip==62.191.185.53 ) { xlog("L_NOTICE", "*** $rm from ipphone-who, $si\n"); append_hf("P-Hint: from ipphone-who\n"); append_hf("Remote-Party-ID: <sip:+492319721231@62.191.185.47>;party=calling;screen=yes;privacy=off\n"); if (method=="INVITE") { strip(1); prefix("+0100555555"); } ds_select_domain("01", "4"); if (method=="INVITE") { t_on_failure("2"); } xlog("L_NOTICE", "----> before t_relay <----\n"); t_relay(); xlog("L_NOTICE", "----> after t_relay <----\n"); return; } if ( src_ip==127.0.2.1 || src_ip==127.0.2.2 ) { xlog("L_NOTICE", "*** $rm from cust3, $si\n"); append_hf("P-Hint: from cust3\n"); append_hf("Remote-Party-ID: <sip:+492319721231@62.191.185.47>;party=calling;screen=yes;privacy=off\n"); if (method=="INVITE") { strip(1); prefix("+0200000003"); } ds_select_domain("02", "4"); if (method=="INVITE") { t_on_failure("2"); } xlog("L_NOTICE", "----> before t_relay <----\n"); t_relay(); xlog("L_NOTICE", "----> after t_relay <----\n"); return; } # ------------------------------------------------------------------ if (method=="INVITE") { xlog("L_NOTICE", "*** reject invite from $si\n"); t_reply("403", "Forbidden"); return; } xlog("L_NOTICE", "*** $rm from $si\n"); t_relay(); } failure_route[2] { xlog("L_NOTICE", "*** in failure_route 1: rc: $rc, rr: $rr, rs: $rs\n"); if (! ds_next_domain()) { t_reply("503", "Service unavailable: no more gws"); return; } else { append_branch(); t_on_failure("2"); t_relay(); return; } } # $Id: main.m4,v 1.3 2006/01/12 15:27:42 whottgen.uu.net Exp $ debug=3 fork=yes log_stderror=yes # (cmd line: -E) 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 ---------------------------------- loadmodule "/opt/openser/lib/openser/modules/sl.so" loadmodule "/opt/openser/lib/openser/modules/tm.so" loadmodule "/opt/openser/lib/openser/modules/rr.so" loadmodule "/opt/openser/lib/openser/modules/maxfwd.so" loadmodule "/opt/openser/lib/openser/modules/usrloc.so" loadmodule "/opt/openser/lib/openser/modules/registrar.so" loadmodule "/opt/openser/lib/openser/modules/textops.so" loadmodule "/opt/openser/lib/openser/modules/xlog.so" loadmodule "/opt/openser/lib/openser/modules/dispatcher.so" # ----------------- setting module-specific parameters --------------- modparam("rr", "enable_full_lr", 1) modparam("xlog", "buf_size", 8192) modparam("dispatcher", "list_file", "/opt/openser/etc/openser/dispatcher.list") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("tm", "fr_timer", 5) # ------------------------- request routing logic ------------------- # main routing logic route[0] { xlog("L_NOTICE", "----> before route 1 <----\n"); route(1); xlog("L_NOTICE", "----> after route 1 <----\n"); } route[1] { if (loose_route()) { xlog("L_NOTICE", "*** loose routing enforced: $rm, $ru, $si, $du\n"); t_relay(); return; } record_route(); xlog("L_NOTICE", "REQUEST: $rm, $ru, $si, $du\n"); if (method=="REGISTER") { xlog("L_NOTICE", "*** register request from $si\n"); sl_send_reply("503", "No registrar service here"); return; } if (method=="INVITE" && ! uri=~"^sip:\+[1-9][0-9]+@") { xlog("L_NOTICE", "*** illegal sip uri: $ru\n"); sl_send_reply("484", "International number required"); return; } # --- customer specific part --------------------------------------- if ( src_ip==62.191.185.122 ) { xlog("L_NOTICE", "*** $rm from laptop-who, $si\n"); append_hf("P-Hint: from laptop-who\n"); append_hf("Remote-Party-ID: <sip:+492319721231@62.191.185.47>;party=calling;screen=yes;privacy=off\n"); if (method=="INVITE") { strip(1); prefix("+0000666666"); } ds_select_domain("00", "4"); if (method=="INVITE") { t_on_failure("2"); } xlog("L_NOTICE", "----> before t_relay <----\n"); t_relay(); xlog("L_NOTICE", "----> after t_relay <----\n"); return; } if ( src_ip==62.191.185.29 ) { xlog("L_NOTICE", "*** $rm from laptop-aholt, $si\n"); append_hf("P-Hint: from laptop-aholt\n"); append_hf("Remote-Party-ID: <sip:+492319721231@62.191.185.47>;party=calling;screen=yes;privacy=off\n"); if (method=="INVITE") { strip(1); prefix("+0100555555"); } ds_select_domain("01", "4"); if (method=="INVITE") { t_on_failure("2"); } xlog("L_NOTICE", "----> before t_relay <----\n"); t_relay(); xlog("L_NOTICE", "----> after t_relay <----\n"); return; } if ( src_ip==62.191.185.53 ) { xlog("L_NOTICE", "*** $rm from ipphone-who, $si\n"); append_hf("P-Hint: from ipphone-who\n"); append_hf("Remote-Party-ID: <sip:+492319721231@62.191.185.47>;party=calling;screen=yes;privacy=off\n"); if (method=="INVITE") { strip(1); prefix("+0100555555"); } ds_select_domain("01", "4"); if (method=="INVITE") { t_on_failure("2"); } xlog("L_NOTICE", "----> before t_relay <----\n"); t_relay(); xlog("L_NOTICE", "----> after t_relay <----\n"); return; } if ( src_ip==127.0.2.1 || src_ip==127.0.2.2 ) { xlog("L_NOTICE", "*** $rm from cust3, $si\n"); append_hf("P-Hint: from cust3\n"); append_hf("Remote-Party-ID: <sip:+492319721231@62.191.185.47>;party=calling;screen=yes;privacy=off\n"); if (method=="INVITE") { strip(1); prefix("+0200000003"); } ds_select_domain("02", "4"); if (method=="INVITE") { t_on_failure("2"); } xlog("L_NOTICE", "----> before t_relay <----\n"); t_relay(); xlog("L_NOTICE", "----> after t_relay <----\n"); return; } # ------------------------------------------------------------------ if (method=="INVITE") { xlog("L_NOTICE", "*** reject invite from $si\n"); t_reply("403", "Forbidden"); return; } xlog("L_NOTICE", "*** $rm from $si\n"); t_relay(); } failure_route[2] { xlog("L_NOTICE", "*** in failure_route 1: rc: $rc, rr: $rr, rs: $rs\n"); if (! ds_next_domain()) { t_reply("503", "Service unavailable: no more gws"); return; } else { append_branch(); t_on_failure("2"); t_relay(); return; } } main.cfg: main.mc m4/main.m4 m4/subs.m4 m4 -I ./m4 $< > $@ 00 sip:139.4.200.6:5060 01 sip:139.4.200.6:5060 include(`subs.m4') define(`_PKG_PATH_', `/opt/openser') define(`_INST_NAME_', `MAIN') define(`_DEBUG_LEVEL_', `3') CUST_BLOCK(`62.191.185.122', `laptop-who', `00', `0', `0666666') CUST_BLOCK(`62.191.185.29', `laptop-aholt', `01', `0', `0555555') CUST_BLOCK(`62.191.185.53', `ipphone-who', `01', `0', `0555555') CUST_BLOCK(`127.0.2.1,127.0.2.2', `cust3', `02', `0', `0000003') include(`main.m4') dnl dnl Main part of the SER configuration file dnl dnl dnl # $Id: main.m4,v 1.4 2006/01/13 14:29:06 whottgen.uu.net Exp $ divert(-1) ifdef(`_PKG_PATH_', `', ` errprint(`*** Package path not defined *** ')m4exit(1)') ifdef(`_INST_NAME_', `', ` errprint(`*** Instance name not defined *** ')m4exit(1)') define(`_MODULE_PATH_', _PKG_PATH_`/lib/openser/modules') ifdef(`_DEBUG_LEVEL_', `', ` define(_DEBUG_LEVEL_, `9')') divert debug=_DEBUG_LEVEL_ fork=yes log_stderror=yes # (cmd line: -E) 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 ---------------------------------- loadmodule "_MODULE_PATH_/sl.so" loadmodule "_MODULE_PATH_/tm.so" loadmodule "_MODULE_PATH_/rr.so" loadmodule "_MODULE_PATH_/maxfwd.so" loadmodule "_MODULE_PATH_/usrloc.so" loadmodule "_MODULE_PATH_/registrar.so" loadmodule "_MODULE_PATH_/textops.so" loadmodule "_MODULE_PATH_/xlog.so" loadmodule "_MODULE_PATH_/dispatcher.so" # ----------------- setting module-specific parameters --------------- modparam("rr", "enable_full_lr", 1) modparam("xlog", "buf_size", 8192) modparam("dispatcher", "list_file", "/opt/openser/etc/openser/dispatcher.list") modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "flags", 3) modparam("tm", "fr_timer", 5) # ------------------------- request routing logic ------------------- # main routing logic route[0] { xlog("L_NOTICE", "----> before route 1 <----\n"); route(1); xlog("L_NOTICE", "----> after route 1 <----\n"); } route[1] { if (loose_route()) { xlog("L_NOTICE", "*** loose routing enforced: $rm, $ru, $si, $du\n"); t_relay(); return; } record_route(); xlog("L_NOTICE", "REQUEST: $rm, $ru, $si, $du\n"); if (method=="REGISTER") { xlog("L_NOTICE", "*** register request from $si\n"); sl_send_reply("503", "No registrar service here"); return; } if (method=="INVITE" && ! uri=~"^sip:\+[1-9][0-9]+@") { xlog("L_NOTICE", "*** illegal sip uri: $ru\n"); sl_send_reply("484", "International number format required"); return; } # --- customer specific part --------------------------------------- undivert(2) # ------------------------------------------------------------------ if (method=="INVITE") { xlog("L_NOTICE", "*** reject invite from $si\n"); t_reply("403", "Forbidden"); return; } xlog("L_NOTICE", "*** $rm from $si\n"); t_relay(); } failure_route[2] { xlog("L_NOTICE", "*** in failure_route 1: rc: $rc, rr: $rr, rs: $rs\n"); if (! ds_next_domain()) { t_reply("503", "Service unavailable: no more gws"); return; } else { append_branch(); t_on_failure("2"); t_relay(); return; } } divert(-1) define(`CUST_IP_PRINT', `src_ip=='$1) define(`CUST_IP_FOREACH', `dnl ifelse($#, 0, `',dnl $#, 1, `CUST_IP_PRINT($1)',dnl `CUST_IP_PRINT($1) || CUST_IP_FOREACH(shift($@))')') define(`CUST_BLOCK', ` divert(2) if ( CUST_IP_FOREACH($1) ) { xlog("L_NOTICE", "*** $rm from '$2`, $si\n"); append_hf("P-Hint: from '$2`\n"); append_hf("Remote-Party-ID: <sip:+492319721231@62.191.185.47>;party=calling;screen=yes;privacy=off\n"); if (method=="INVITE") { strip(1); prefix("+'$3$4$5`"); } ds_select_domain("'$3`", "4"); if (method=="INVITE") { t_on_failure("2"); } xlog("L_NOTICE", "----> before t_relay <----\n"); t_relay(); xlog("L_NOTICE", "----> after t_relay <----\n"); return; } divert ') divert

2 2

[Users] Problem with IAX and Openser.
by david.castro 16 Jan '06

16 Jan '06

Thank you Daniel. The ouput is this: U 2006/01/16 11:49:22.392147 10.2.11.35:5062 -> 10.2.11.35:5060 INVITE sip:204@10.2.11.35 SIP/2.0..Via: SIP/2.0/UDP 10.2.11.35:5062;branch=z9hG4bK5c440b0c;rport..From: "David" <sip:"D avid"<sip:205@1021150@10.2.11.35:5062>;tag=as4a705794..To: <sip:204@10.2.11.35>..Contact: <sip:"David"<sip:205@1021150@ 10.2.11.35:5062>..Call-ID: 2d64d7461241acaf11db6bb21cd4830a@10.2.11.35..CSeq: 102 INVITE..User-Agent: Asterisk PBX..Max -Forwards: 70..Date: Mon, 16 Jan 2006 10:49:22 GMT..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. .Content-Type: application/sdp..Content-Length: 463....v=0..o=root 15086 15086 IN IP4 10.2.11.35..s=session..c=IN IP4 1 0.2.11.35..t=0 0..m=audio 17016 RTP/AVP 0 8 4 111 18 3 97 7 110 5 101..a=rtpmap:0 PCMU/8000..a=rtpmap:8 PCMA/8000..a=rt pmap:4 G723/8000..a=rtpmap:111 G726-32/8000..a=rtpmap:18 G729/8000..a=fmtp:18 annexb=no..a=rtpmap:3 GSM/8000..a=rtpmap: 97 iLBC/8000..a=rtpmap:7 LPC/8000..a=rtpmap:110 speex/8000..a=rtpmap:5 DVI4/8000..a=rtpmap:101 telephone-event/8000..a= fmtp:101 0-16..a=silenceSupp:off - - - -.. U 2006/01/16 11:49:22.392610 10.2.11.35:5060 -> 10.2.11.35:5062 SIP/2.0 100 Trying..Via: SIP/2.0/UDP 10.2.11.35:5062;branch=z9hG4bK5c440b0c;rport=5062..From: "David" <sip:"David"<sip: 205@1021150@10.2.11.35:5062>;tag=as4a705794..To: <sip:204@10.2.11.35>..Call-ID: 2d64d7461241acaf11db6bb21cd4830a(a)10.2.1 1.35..CSeq: 102 INVITE..Server: OpenSer (1.0.0-tls (i386/linux))..Content-Length: 0..Warning: 392 10.2.11.35:5060 "Nois y feedback tells: pid=15870 req_src_ip=10.2.11.35 req_src_port=5062 in_uri=sip:204@10.2.11.35 out_uri=sip:204@10.2.11. 35 via_cnt==1".... U 2006/01/16 11:49:22.393443 10.2.11.35:5060 -> 10.2.11.35:5062 SIP/2.0 180 Ringing..Via: SIP/2.0/UDP 10.2.11.35:5062;branch=z9hG4bK5c440b0c;rport=5062..From: "David" <sip:"David"<sip :205@1021150@10.2.11.35:5062>;tag=as4a705794..To: <sip:204@10.2.11.35>;tag=329cfeaa6ded039da25ff8cbb8668bd2.1607..Call- ID: 2d64d7461241acaf11db6bb21cd4830a@10.2.11.35..CSeq: 102 INVITE..Server: OpenSer (1.0.0-tls (i386/linux))..Content-Le ngth: 0..Warning: 392 10.2.11.35:5060 "Noisy feedback tells: pid=15870 req_src_ip=10.2.11.35 req_src_port=5062 in_uri= sip:204@10.2.11.35 out_uri=sip:204@10.2.11.64:5060;transport=udp via_cnt==1".... U 2006/01/16 11:49:22.393731 10.2.11.35:5060 -> 10.2.11.35:5062 SIP/2.0 500 I'm terribly sorry, server error occurred (1/SL)..Via: SIP/2.0/UDP 10.2.11.35:5062;branch=z9hG4bK5c440b0c;r port=5062..From: "David" <sip:"David"<sip:205@1021150@10.2.11.35:5062>;tag=as4a705794..To: <sip:204@10.2.11.35>;tag=329 cfeaa6ded039da25ff8cbb8668bd2.1607..Call-ID: 2d64d7461241acaf11db6bb21cd4830a@10.2.11.35..CSeq: 102 INVITE..Server: Ope nSer (1.0.0-tls (i386/linux))..Content-Length: 0..Warning: 392 10.2.11.35:5060 "Noisy feedback tells: pid=15870 req_sr c_ip=10.2.11.35 req_src_port=5062 in_uri=sip:204@10.2.11.35 out_uri=sip:204@10.2.11.64:5060;transport=udp via_cnt==1".. .. U 2006/01/16 11:49:22.397810 10.2.11.35:5062 -> 10.2.11.35:5060 ACK sip:204@10.2.11.35 SIP/2.0..Via: SIP/2.0/UDP 10.2.11.35:5062;branch=z9hG4bK5c440b0c;rport..From: "David" <sip:"Davi d"<sip:205@1021150@10.2.11.35:5062>;tag=as4a705794..To: <sip:204@10.2.11.35>;tag=329cfeaa6ded039da25ff8cbb8668bd2.1607. .Contact: <sip:"David"<sip:205@1021150@10.2.11.35:5062>..Call-ID: 2d64d7461241acaf11db6bb21cd4830a@10.2.11.35..CSeq: 10 2 ACK..User-Agent: Asterisk PBX..Max-Forwards: 70..Content-Length: 0.... I saw the ouput and I think it's all right except the address: <sip:"David"<sip:205@1021150@10.2.11.35:5062> What do you think about all that?

1 0

[Users] Problem with IAX and Openser.
by david.castro 16 Jan '06

16 Jan '06

I used ngrep -qt port 5060 but I don't get ouput in the screen, so I used: ngrep -qt, and the ouput was: U 2006/01/16 10:56:26.736394 10.2.11.50:4569 -> 10.2.11.35:4569 ..................204.."David"<sip:205@1021150..David..en..internal..david...................0V. U 2006/01/16 10:56:26.737313 10.2.11.35:4569 -> 10.2.11.50:4569 ..................205130695..david T 2006/01/16 10:56:26.736424 10.2.11.50:22 -> 10.2.11.35:60601 [AP] ].........I2..h.&......>,B.....X]OA.Q...@...b..a~..X.a..8...g.....)....y&.~.....@..+..<&....ly...b.{..Q...f..-..#...... ..d..\h....[.....He..&.]........._[..,.HZ|g..P.....r...f0.P.l{.....@..z........Y.&[......'..v......|.1.m.P.d.;M........ ..H$..T..<~z..Ra.. T 2006/01/16 10:56:26.736447 10.2.11.50:22 -> 10.2.11.35:60601 [AP] ..IM..D..X.Uv;.."..d...a.....q.Z:.7.l......Z..@.>-........~_..K^ U 2006/01/16 10:56:26.736538 10.2.11.50:4569 -> 10.2.11.35:4569 ............. b07125c8393cbdd4c3ca7b621bb678bc U 2006/01/16 10:56:26.736748 10.2.11.35:4569 -> 10.2.11.50:4569 .................. U 2006/01/16 10:56:26.736950 10.2.11.50:4569 -> 10.2.11.35:4569 ............ T 2006/01/16 10:56:26.737002 10.2.11.50:22 -> 10.2.11.35:60601 [AP] A.W..=..>n....S..>.....hI.PX@)..I.KHR.f2_j..5.....S.. ...h.....SH.og...zC.@...../.j..U?.Yk..%.m..........&...... T 2006/01/16 10:56:26.737065 10.2.11.50:22 -> 10.2.11.35:60601 [AP] r......]..G.n.w..H..'...rk...c..|0.`.......7.J........g....b.]Z...G.:^.f....p.[..b42..D.....t... U 2006/01/16 10:56:26.740189 10.2.11.35:4569 -> 10.2.11.50:4569 ............ U 2006/01/16 10:56:26.740263 10.2.11.35:4569 -> 10.2.11.50:4569 ............ U 2006/01/16 10:56:26.740314 10.2.11.50:4569 -> 10.2.11.35:4569 ............ U 2006/01/16 10:56:26.740381 10.2.11.50:4569 -> 10.2.11.35:4569 ............ T 2006/01/16 10:56:26.740535 10.2.11.50:22 -> 10.2.11.35:60601 [AP] ....z.O.Q,?....o...t...[..^.....0..2/X..98..M...'-...lP..I.......0..<.'.W..vDs....g...Y..G..v8.. T 2006/01/16 10:56:26.740566 10.2.11.50:22 -> 10.2.11.35:60601 [AP] ._.z...4....K^.#..J...`..l.0.)e..y.G.7.t.Y.t-W.5........41...K/.#..@..R......2T..m...J.K.:].....E.~le.....gh<-u% U 2006/01/16 10:56:26.740594 10.2.11.35:4569 -> 10.2.11.50:4569 ............*.& U 2006/01/16 10:56:26.740847 10.2.11.50:4569 -> 10.2.11.35:4569 ............ T 2006/01/16 10:56:26.741537 10.2.11.50:22 -> 10.2.11.35:60601 [AP] =.z..hym..f.z..H...8,.Uh:.....)r......j.&).".....=.Z$U... 13.....N......:.cn+L....../..:Q.....qX.."..Q\....3Z..N..1..y= ....!V....FR...j..MjpE=....j..r..k.......M..0..9.F...).M AI.-.....g..K.}S....8.z...N....3Z3.gQ?...,...#....H..5.dQ....- ...Tv.9z.....c.;..x....uq..........Ga....C... .......H}W})d....7...k..'d....LE..Kq..D!%4.yZ..?.'.......nBN.23...L)>..6. !.,......B.^.....v_.\.O.ER.J..}...7]8.bBj.. I saw this and I guess that sip packet is wrong, but I have in extexsion.conf: exten = _2XX,2,Dial(SIP/${EXTEN}@ser,100) I thought I was making a sip request. 2 notes: I have openser and asterisk in the same computer, asterisk has sip port 5062 and openser 5060. I test openser and asterisk talk one to one.

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