sr-users September 2006

sr-users@lists.kamailio.org

184 participants
436 discussions

by senthil.nachimuthu＠bt.com

Hi, I want to configure SER <ser/about> 0.9.6 in such a way that It can proxy presence messages(SUBSCRIBE,PUBLISH) to presence server. Can you let me know how can I do that? Thanks and Regards, Senthil Kumar, Presence Capability, BT-Service Execution Platform, B62-MH,Orion Building,Adastral Park,Ipswich Work:+44 (0)1473 605 668 Mobile:+44 (0)7772287447

17 years, 9 months

[Serusers] USE XLOG

by Legos Engineering

Hello, I am a student and i do a workplacement. So i must work on the VOIP, but ihave no experience in. I would like know thank to a log, when an user connect/disconnect his user ugent. I know that i must use xlog, but i'm lost for many reasons. The first : Idon't know variables i must use to do what i want. The second : I don't know where i must write the line in ser.cfg to have this log. The third : Where is the log generated by xlog, and if there isn't one, how i can create this log. The fourth : If i must write something in xlog.so, how open it. Because when i open it with VI or NANO there is only encrypted data. THX to help me. If you can answer on my mail : sylvain.nigen(a)legos.fr, because it's easier ton know if i have an answer than to search the answer on SER USER. THANKS

17 years, 9 months

[Serusers] ipv6 support

by P Jayasree

hi, I;ve installed ser 0.9.x version on linux ipv6 enabled PC.Now the ser is running.But it says it is listening on ipv4 address on udp and tcp.How do i know whether it is listening on ipv6 ?do i need to configure for anything ,for it to listen on ipv6 adress also! Kindly help in this issue. Regards, Jayasree The information contained in this e-mail message and/or attachments to it may contain confidential or privileged information. If you are not the intended recipient, any dissemination, use, review, distribution, printing or copying of the information contained in this e-mail message and/or attachments to it are strictly prohibited. If you have received this communication in error, please notify us by reply e-mail immediately and permanently delete the message and any attachments.

17 years, 9 months

[Users] NAT: minimise media proxying whilst maximising usability

by kjcsb

I am trying to reach a suitable trade-off between minimising the amount of proxied media and minimising one-way/no audio for all clients including NATed ones. When a UA that has been configured to use STUN sends a register message to SER the behaviour differs depending on the device. Some, such as a Sipura device, send the private address in the Via and the public address in the Contact: U 60.234.nnn.nnn:5060 -> 147.202.nnn.nnn:5060 REGISTER sip:domain.com SIP/2.0..Via: SIP/2.0/UDP 192.168.0.14:5060;branch=z9hG4bK-64d76b15..From: <sip:6435488773@domain.com >;tag=4cdc1208fa650916o0..To: <sip:6435488773@domain.com>..Call-ID: 1143f50-45857c6e@192.168.0.14..CSeq: 1 REGISTER..Max-Forwards: 70..Contact: <sip:6435488773@60.234.nnn.nnn:5060>;expires=60..User-Agent: Sipura/SPA3000-2.0.13(GWg)..Content-Length: 0..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Supported: x-sipura.... Others, such as a Grandstream device, send the public address in both the Via and Contact: U 60.234.nnn.nnn:32483 -> 147.202.nnn.nnn:5060 REGISTER sip:domain.com SIP/2.0..Via: SIP/2.0/UDP 60.234.nnn.nnn:32483;branch=z9hG4bK113cfc09c6acd189..From: "User" <sip: 5551234(a)domain.com>;tag=dede17c9405d4e0e..To: <sip:5551234@domain.com>..Contact: <sip:5551234@60.234.nnn.nnn:32483>.. Call-ID: fc200867c9a03268@192.168.0.10..CSeq: 100 REGISTER..Expires: 60..User-Agent: Grandstream BT100 1.0.6.7..Max-Forwards: 70..Allow: INVITE,ACK,CANCEL,BYE,NOTIFY,REFER,OPTIONS,INFO,SUBSCRIBE..Content-Length: 0.... I am using the opsip.org configuration file together with mediaproxy. In the case of the Sipura, flag(6) is set in route[2] since client_nat_test("7") evaluates to true and so any calls with this device will be proxied. From ready various posts on the topic I understand this is a very safe configuration since it will deal with virtually every scenario. However it is also very expensive since it will proxy audio in situations where it is not really required. For example a call between two Sipura devices will be proxied when, due to STUN, it is unecessary in most cases. As I understand it the cases when it would be necessary are: - both UAs behind the same NAT with the SER proxy outside the NAT and router doesn't support hairpinning - devices behind symmetric NATs On the other hand, flag(6) will not be set with the Grandstream and so audio will not be proxied when this device calls another device that is either not behind NAT or is another NATed Grandstream. This is great in most cases but audio problems could arise in the cases described above. What I am therefore thinking is that by default I will not proxy media for devices where a public IP address has been determined. But I will create an avp that specifies by user whether to enable the standard onsip.org config. Also there could even be a dial prefix (e.g. *666) that turns proxying on. route[2] #Do this by default if (!search("^Contact: [ ]*\*") && client_nat_test("1") { etc #Do this if avp set or prefix dialled if (!search("^Contact: [ ]*\*") && client_nat_test("7") { etc I would appreciate any comments on the above. Have I understood the issues correctly? Will this type of configuration work? What are the disadvantages? Regards Cameron

17 years, 9 months

[Serusers] Problem in testing MSILO module. (Please response..Really need help!!)

by Meidiana Ten

Hi there, Good afternoon.. I'd like to ask some questions depending on my last project problems.. maybe I had ever asked it in this forum, but I quite don't understand the solution you guys have told me. Maybe it's because I don't give the details about my OS and many more, and I'm a newbie too. But I really thanx for the reply on my last question. I'm using Red Hat 9 Linux to build my SIP Server. I use SER 0.9.7 as the server. The problem I found here is to test the MSILO module which I add to the ser.cfg Below is the description of my ser.cfg: ------------------------------------------------------- debug=9 fork=no log_stderror=yes listen=202.95.149.2 # put your server IP address here port=5060 children=4 dns=no rev_dns=no fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:heslo@localhost/ser" fifo_mode=0666 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/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/uri_db.so" loadmodule "/usr/local/lib/ser/modules/uri.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" loadmodule "/usr/local/lib/ser/modules/domain.so" loadmodule "/usr/local/lib/ser/modules/permissions.so" loadmodule "/usr/local/lib/ser/modules/msilo.so" modparam("auth_db|permissions|uri_db|usrloc","db_url", "mysql://ser:heslo@localhost/ser") 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", "/var/run/rtpproxy.sock") modparam("usrloc", "db_mode", 2) modparam("registrar", "nat_flag", 6) modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_inv_timer", 27) modparam("tm", "fr_inv_timer_avp", "inv_timeout") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "wt_timer", 10 ) modparam("msilo", "db_url", "mysql://ser:heslo@localhost/ser") modparam("msilo", "db_table", "silo") modparam("msilo","registrar","sip:registrar@pcr.ac.id") alias="pcr.ac.id" 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; }; # ----------------------------------------------------------------- # Offline Message Store Section # ----------------------------------------------------------------- if (is_from_local()) { if (method=="REGISTER") { save("location"); log("REGISTER received -> dumping messages with MSILO\n"); # MSILO - dumping user's offline messages if (m_dump()) { log("MSILO: offline messages dumped - if they were\n"); }else{ log("MSILO: no offline messages dumped\n"); }; break; }; # domestic SIP destinations are handled using our USRLOC DB if(!lookup("location")) { if (! t_newtran()) { sl_reply_error(); break; }; # we do not care about anything else but MESSAGEs if (!method=="MESSAGE") { if (!t_reply("404", "Not found")) { sl_reply_error(); }; break; }; log("MESSAGE received -> storing using MSILO\n"); # MSILO - storing as offline message if (m_store("0")) { log("MSILO: offline message stored\n"); if (!t_reply("202", "Accepted")) { sl_reply_error(); }; }else{ log("MSILO: offline message NOT stored\n"); if (!t_reply("503", "Service Unavailable")) { sl_reply_error(); }; }; break; }; # if the downstream UA does not support MESSAGE requests # go to failure_route[1] t_on_failure("1"); t_relay(); break; }; # forward to current uri now; use stateful forwarding that # works reliably even if we forward from TCP to UDP if (!t_relay()) { sl_reply_error(); }; # ----------------------------------------------------------------- # Call Type Processing Section # ----------------------------------------------------------------- if (uri!=myself) { route(5); route(1); break; }; if (uri==myself) { if (method=="ACK") { route(6); break; } else if (method=="CANCEL") { route(3); break; } else if (method=="INVITE") { route(3); break; } else if (method=="REGISTER") { route(2); break; }; lookup("aliases"); if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { sl_send_reply("404", "User Not Found"); break; }; }; route(1); } failure_route[1] { # forwarding failed -- check if the request was a MESSAGE if (!method=="MESSAGE") { break; }; log(1,"MSILO:the downstream UA doesn't support MESSAGEs\n"); # we have changed the R-URI with the contact address, ignore it now if (m_store("1")) { log("MSILO: offline message stored\n"); t_reply("202", "Accepted"); }else{ log("MSILO: offline message NOT stored\n"); t_reply("503", "Service Unavailable"); }; } 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("pcr.ac.id","subscriber")) { www_challenge("pcr.ac.id","0"); break; }; if (!check_to()) { sl_send_reply("401", "Unauthorized"); break; }; consume_credentials(); if (!save("location")) { sl_reply_error(); }; } route[3] { # ----------------------------------------------------------------- # CANCEL and INVITE Message Handler # ----------------------------------------------------------------- if (!allow_trusted() && nat_uac_test("19")) { setflag(6); } lookup("aliases"); if (method=="INVITE" && !allow_trusted()) { if (!proxy_authorize("pcr.ac.id","subscriber")) { proxy_challenge("pcr.ac.id","0"); break; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); break; }; consume_credentials(); }; if (uri=~"^sip:9[0-9]*@") { route(4); break; }; if (uri!=myself) { route(5); route(1); break; }; if (!lookup("location")) { if (uri=~"^sip:[0-9]{10}@") { route(4); break; }; sl_send_reply("404", "User Not Found"); break; }; if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; t_on_reply("1"); if (!t_relay()) { if(isflagset(6)) { unforce_rtp_proxy(); } sl_reply_error(); }; } route[4] { # ----------------------------------------------------------------- # PSTN Handler # ----------------------------------------------------------------- rewritehostport("202.95.149.2:5060"); # INSERT YOUR PSTN GATEWAY IP ADDRESS avp_write("i:45", "inv_timeout"); if (isflagset(6)) { force_rport(); fix_nated_contact(); force_rtp_proxy(); }; route(1); } 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(); }; } ------------------------------------------------------- I don't know whether my configuration below is right or still wrong..I've test it using 2 clients. I made one of them online, and the other off. I try to call the offline one, but it said user not found. In my thought, if I've activated MSILO module, it will automatically ask the online client if he wants to store offline voice message or not... Please tell me what's wrong with the configuration. In addition, the server can be use to make a call between online clients. Second question is about my serweb. I have seen my config.php in /var/www/html/serweb directory. It contains as shown below: ------------------------------------------------------- /* Unless you used brute-force to change SER table names */ $this->table_subscriber="subscriber"; $this->table_pending="pending"; $this->table_grp="grp"; $this->table_aliases="aliases"; $this->table_location="location"; $this->table_missed_calls="missed_calls"; $this->table_accounting="acc"; $this->table_phonebook="phonebook"; $this->table_event="event"; $this->table_netgeo_cache="netgeo_cache"; $this->table_ser_mon="server_monitoring"; $this->table_ser_mon_agg="server_monitoring_agg"; $this->table_message_silo="silo"; / below is the problem / $this->table_voice_silo="voice_silo"; ------------------------------------------------------- And below is my SER tables: +-----------------------+ | Tables_in_ser | +-----------------------+ | acc | | active_sessions | | admin_privileges | | aliases | | calls_forwarding | | config | | domain | | event | | grp | | location | | missed_calls | | pending | | phonebook | | preferences_types | | reserved | | server_monitoring | | server_monitoring_agg | | silo | | speed_dial | | subscriber | | trusted | | uri | | usr_preferences | | version | +-----------------------+ 24 rows in set (0.00 sec) ------------------------------------------------------- You can see that there is no table named "voice_silo" in my SER tables. It cause error on one of my web page named message_store.php. It requires voice_silo table. I'd like to ask you what is the different between table named "silo" and "voice_silo"? Can both of them store offline voice message? If I have to add "voice_silo" in my SER table, please give me the description of entities needed to build voice_silo table. Last, I'd like to thank you guys for the willingness to read my questions and problems. Of course I'm expecting the answers too. Thanx again. Regards, Meidiana Ten __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

17 years, 9 months

Fwd: [Users] mediaproxy working, but not if asterisk is involved

by Arne Van Theemsche

I had forgotten to cc the list ---------- Forwarded message ---------- From: Arne Van Theemsche <arnevt(a)gmail.com> Date: 22-sep-2006 9:06 Subject: Re: [Users] mediaproxy working, but not if asterisk is involved To: daniel(a)voice-system.ro the problem is that I don't even see the "reply received"... So for some reason the asterisk reply isn't passed through to the onreply_route. My theory is that asterisk doesn't respect the reply parameters somewhere, but it isn't clear to me where arne 2006/9/22, Daniel-Constantin Mierla <daniel(a)voice-system.ro>: > > Do you get "using mediaproxy" message in the logs? If not, that the > search() matches, I cannot sot right now what is wrong with the > expression. But you can move t_on_reply("1") into if*method=="INVITE") > statement and replace the search condition with if (status =~ > "(183)|(2[0-9][0-9])"). > > See: > http://voip-info.org/wiki/view/OpenSER+And+Mediaproxy > > Cheers, > Daniel > > > On 09/21/06 21:46, Arne Van Theemsche wrote: > > below is the transaction of the failed mediaproxy invite. I allready > > could tell that replies go through openser, but I don't see the reason > > why ser doesn't see them as replies (and use the mediaproxy function). > > > > as you can see, the invite from <ip client> to <ip asterisk> (through > > <ip OPENSER>, which is also ip of mediaproxy) goes in one direction > > good (the ip in the SDP is changed from <ip client> to <ip openser>, > > but the return path en the OK (with it's SDP) is not changed > > > > I did a tcpdump with a call between 2 clients, where the proxy works, > > and the only difference I see is that in the reply of asterisk, there > > is no rinstance field in the contact header > > > > thanks > > arne > > > > U <ip client>:5060 -> <ip OPENSER>:5060 > > INVITE sip:701@<sip domain>;transport=UDP SIP/2.0..From: "arne" > > <sip:1002@<sip domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To: > > "701"< sip:701@sipgat > > e.evonet.be <http://e.evonet.be>>..Call-ID: > > 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip > > <mailto: 1064dc44-514a90c3-13c4-7a70a-1de331be-529@%3Cip> > > client>..CSeq: 1 INVITE..Via: SIP/2.0/UDP <ip > > client>:5060;rport;branch=z9hG4bK-7a70a-1d > > e331c2-69dc..Max-Forwards: 70..Supported: > > replaces,100rel,timer..Allow: INVITE, ACK, BYE, REFER, NOTIFY, CANCEL, > > OPTIONS, INFO, PRACK..User-Agent: Swissvoice IP1 > > 0 SP v1.0.1 (Build 3) 3.0.5.1..Allow-Events: talk, hold, > > conference..Contact: "arne" <sip:1002@<ip > > client>:5060;transport=UDP>..Session-Expires: 1800..Content- > > Type: application/sdp..Content-Length: 246....v=0..o=rtp/1 501514 > > 501514 IN IP4 <ip client>..s=-..c=IN IP4 <ip client>..t=0 0..m=audio > > 50000 RTP/AVP 18 0 8.. > > a=fmtp:18 annexb=yes..a=ptime:40..a=SilenceSupp:on..a=rtpmap:18 > > g729/8000..a=rtpmap:0 pcmu/8000..a=rtpmap:8 pcma/8000..a=sendrecv.. > > # > > > > U <ip OPENSER>:5060 -> <ip asterisk>:5060 > > INVITE sip:701@<sip domain>;transport=UDP SIP/2.0..Record-Route: > > <sip:<ip OPENSER>;lr=on;ftag=514a90c3-13c4-7a70a-1de331c0-5e4f>..From: > > "arne" < sip:1002@si > > pgate.evonet.be > > <http://pgate.evonet.be>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To: > > "701"<sip:701@<sip domain>>..Call-ID: > > 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip > > <mailto:1064dc44-514a90c3-13c4-7a70a-1de331be-529@%3Cip> client>..C > > Seq: 1 INVITE..Via: SIP/2.0/UDP <ip OPENSER>;branch=0..Via: > > SIP/2.0/UDP <ip > > > client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc..Max-Forwards: > > 69..Supp > > orted: replaces,100rel,timer..Allow: INVITE, ACK, BYE, REFER, > > NOTIFY, CANCEL, OPTIONS, INFO, PRACK..User-Agent: Swissvoice IP10 SP > > v1.0.1 (Build 3) 3.0.5.1..Allo > > w-Events: talk, hold, conference..Contact: "arne" <sip:1002@<ip > > client>:5060;transport=UDP>..Session-Expires: 1800..Content-Type: > > application/sdp..Content-Leng > > th: 246....v=0..o=rtp/1 501514 501514 IN IP4 <ip client>..s=-..c=IN > > IP4 <ip OPENSER>..t=0 0..m=audio 60106 RTP/AVP 18 0 8..a=fmtp:18 > > annexb=yes..a=ptime:40..a > > =SilenceSupp:on..a=rtpmap:18 g729/8000..a=rtpmap:0 > > pcmu/8000..a=rtpmap:8 pcma/8000..a=sendrecv.. > > # > > > > U <ip asterisk>:5060 -> <ip OPENSER>:5060 > > SIP/2.0 100 Trying..Via: SIP/2.0/UDP <ip > > OPENSER>;branch=0;received=<ip OPENSER>..Via: SIP/2.0/UDP <ip > > client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2- > > 69dc..From: "arne" <sip:1002@<sip > > domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To: "701"<sip:701@<sip > > domain>>..Call-ID: 1064dc44-514a90c3-13c4-7a70 > > a-1de331be-529@<ip <mailto:a-1de331be-529@%3Cip> client>..CSeq: 1 > > INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, > > BYE, REFER, SUBSCRIBE, NOTIFY..Contact: <sip:701@ > > <ip asterisk>>..Content-Length: 0.... > > # > > > > U <ip OPENSER>:5060 -> <ip client>:5060 > > SIP/2.0 100 Trying..Via: SIP/2.0/UDP <ip > > client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc..From: > > "arne" <sip:1002@<sip domain>>;tag=514a90c3-13c > > 4-7a70a-1de331c0-5e4f..To: "701"<sip:701@<sip domain>>..Call-ID: > > 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip > > <mailto:1064dc44-514a90c3-13c4-7a70a-1de331be-529@%3Cip> > > client>..CSeq: 1 INVITE..User-Agent: Asteri > > sk PBX..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, > > NOTIFY..Contact: <sip:701@<ip asterisk>>..Content-Length: 0.... > > # > > > > U <ip asterisk>:5060 -> <ip OPENSER>:5060 > > SIP/2.0 200 OK..Via: SIP/2.0/UDP <ip OPENSER>;branch=0;received=<ip > > OPENSER>..Via: SIP/2.0/UDP <ip > > client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc > > ..Record-Route: <sip:<ip > > OPENSER>;lr=on;ftag=514a90c3-13c4-7a70a-1de331c0-5e4f>..From: "arne" > > <sip:1002@<sip domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f > > ..To: "701"<sip:701@<sip domain>>;tag=as60ebd3fc..Call-ID: > > 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip > > <mailto:1064dc44-514a90c3-13c4-7a70a-1de331be-529@%3Cip> > > client>..CSeq: 1 INVITE..User-Agent: Asterisk PBX > > ..Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, > > NOTIFY..Contact: <sip:701@<ip asterisk>>..Content-Type: > > application/sdp..Content-Length: 188....v= > > 0..o=root 26276 26276 IN IP4 <ip asterisk>..s=session..c=IN IP4 <ip > > asterisk>..t=0 0..m=audio 13434 RTP/AVP 0 8..a=rtpmap:0 > > PCMU/8000..a=rtpmap:8 PCMA/8000..a= > > silenceSupp:off - - - -.. > > # > > > > U <ip OPENSER>:5060 -> <ip client>:5060 > > SIP/2.0 200 OK..Via: SIP/2.0/UDP <ip > > > client>:5060;rport=5060;branch=z9hG4bK-7a70a-1de331c2-69dc..Record-Route: > > <sip:<ip OPENSER>;lr=on;ftag=514a90c3-13c4-7a70 > > a-1de331c0-5e4f>..From: "arne" <sip:1002@<sip > > domain>>;tag=514a90c3-13c4-7a70a-1de331c0-5e4f..To: "701"<sip:701@<sip > > domain>>;tag=as60ebd3fc..Call-ID: > > 1064dc44-514a90c3-13c4-7a70a-1de331be-529@<ip > > <mailto: 1064dc44-514a90c3-13c4-7a70a-1de331be-529@%3Cip> > > client>..CSeq: 1 INVITE..User-Agent: Asterisk PBX..Allow: INVITE, ACK, > > CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NO > > TIFY..Contact: <sip:701@<ip asterisk>>..Content-Type: > > application/sdp..Content-Length: 188....v=0..o=root 26276 26276 IN IP4 > > <ip asterisk>..s=session..c=IN IP4 > > <ip asterisk>..t=0 0..m=audio 13434 RTP/AVP 0 8..a=rtpmap:0 > > PCMU/8000..a=rtpmap:8 PCMA/8000..a=silenceSupp:off - - - -.. > > # > > > > > > > > > > > > 2006/9/21, Daniel-Constantin Mierla <daniel(a)voice-system.ro > > <mailto:daniel@voice-system.ro>>: > > > > Hello, > > > > watch the network traffic with ngrep on your sip server. You can > > see the > > call flow which may help to identify the issue. You can paste it > > to the > > list and someone may give you hints. > > > > Cheers, > > Daniel > > > > > > On 09/21/06 12:28, Arne Van Theemsche wrote: > > > hi > > > > > > my users subscribe with openser, en asterisk is used as > connectivity > > > to pstn > > > > > > i am now installing a mediaproxy, for all users, so every call > goes > > > via a mediaproxy. > > > > > > I'm doing this as follows (relevant statements only) > > > > > > in route > > > > > > #I installed the t_on_reply here to be sure that every > reply > > > gets parsed, but normally in the INVITE section should be enough? > > > t_on_reply("1"); > > > > > > if (method==INVITE) { > > > use_media_proxy(); > > > } > > > > > > > > > onreply_route[1] { > > > log(-3,"reply received"); > > > if (!search("^Content-Length:[ ]*0")) { > > > log(-3,"using mediaproxy"); > > > use_media_proxy(); > > > }; > > > } > > > > > > > > > the weird is, for all local users, this works fine, but as soon as > > > > asterisk is involved, the reply doesn't get triggered (not > > seeing the > > > "reply received" either, only when disconnecting the call). The > call > > > get's established fine, asterisk is sending media to the > > mediaproxy, > > > but the SDP towards the calling phone is not modified (since the > > > onreply isn't triggered) > > > > > > am I missing something here? > > > > > > thanks > > > Arne > > > > > > > > > ------------------------------------------------------------------------ > > > > > > > > _______________________________________________ > > > Users mailing list > > > Users(a)openser.org <mailto:Users@openser.org> > > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > >

17 years, 9 months

[Users] How do I see who is registered?

by Ronald Wiplinger

I am watching openserctl moni to see what happens, but more than that 5 people are registered I cannot see. How do I see who is registered? bye Ronald

17 years, 9 months

[serusers] Openser Admin Login Access

by John Shellhouse

Hi All, I am sure I must be missing something just not sure what it is. I granted the permission in mysql: GRANT ALL ON openser.* TO 'openser'@'dbhost' IDENTIFIED BY 'openserrw'; as stated in the Install guide and then launched the server from terminal with ./script/server and in the browser typed: http://myomain:3000 which displayed the login page. I then typed into the login:password area openser:openserrw But I was unsuccessful each time. Does anyone have any idea what I am missing or doing incorrectly? Thank you. jwshel

17 years, 9 months

[serusers] Openser Admin Login Access

by John Shellhouse

Hi All, I accessed the login/password area but I am erroring but don't know how to fix. I granted all: GRANT ALL ON openser.* TO 'openser'@'dbhost' IDENTIFIED BY 'openserrw'; and changed the database.yml to dbhost as well which allowed me to login when I left it the same as localhost I was not able to login at all with default password. Now that I am in the login zone, I get this error: Mysql::Error in LoginController#login Unknown MySQL server host 'dbhost' (1) RAILS_ROOT: script/../config/.. Application Trace | Framework Trace | Full Trace /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.4/lib/active_record/connection_adapters/mysql_adapter.rb:330:in `real_connect' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.4/lib/active_record/connection_adapters/mysql_adapter.rb:330:in `connect' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.4/lib/active_record/connection_adapters/mysql_adapter.rb:87:in `initialize' /usr/local/lib/ruby/gems/1.8/gems/activerecord-1.14.4/lib/active_record/connection_adapters/mysql_adapter.rb:36:in `mysql_connection' Plus more, but I did not want to paste it all. Any help appreciated. jwshel

17 years, 9 months

[Users] Trusted IP examples

by ram

Hi iam trying to test trusted IP kind of setup I have openser server connect to my ITSP another Office i have OpenSER, i want to connect to main Server and send the calls out could some one give me examples or docs where i can find similar kind of real world examples Ram

17 years, 9 months

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sr-users September 2006