sr-users October 2006

sr-users@lists.kamailio.org

189 participants
345 discussions

by Mark Price

What is the best way to view the sgml documentation included in the ser-0.9.6 tarball? Preferrably a website or such where it is already rendered? Thanks Mark Price

18 years, 1 month

[serusers] serctl ping response - 477

by Kamal.Mann＠t-systems.com

Hi All I am trying to implement blind call-forwarding. But from ser logs m getting "Oct 19 14:48:52 localhost /usr/sbin/ser[3971]: CRITICAL: invalid sendtoparameters one possible reason is the server is bound to localhost and attempts to send to the net". And when I am trying 'serctl ping sip:sony@imsfw.de' its giving me 477!!! Why it is so?? Even when I enable listen=IP_address into my ser.cfg my clients unable to register with ser but if I comment it out then it works fine & starts listening on my IP. Please give me your valuable suggestions. Thanks in anticipation Kamal Mann

18 years, 1 month

[Serusers] issue w/ permissions module

by Corey

Hello! I'm using postgresql for the db backend for my ser instance, after creating and populating a few obvious tables by handI seem to have it "mostly working". ( using ser-0.9.4 ) However, I'm stuck on the following: Oct 18 21:09:36 server /usr/sbin/ser[23595]: ERROR: permissions: trusted_reload(): Database problem Oct 18 21:09:36 server /usr/sbin/ser[23595]: init_trusted(): Reload of trusted table failed Oct 18 21:09:36 server /usr/sbin/ser[23595]: Error while initializing allow_trusted function Scratching my head, trying to figure out what might be causing that, but with no success - any clues? What I imagine is the relevant info: server ser # psql -U ser -W ser Password for user ser: Welcome to psql 8.1.4, the PostgreSQL interactive terminal. ser=# \dt List of relations Schema | Name | Type | Owner --------+------------+-------+------- public | subscriber | table | ser public | trusted | table | ser public | version | table | ser (3 rows) ser=# \d trusted Table "public.trusted" Column | Type | Modifiers --------------+-----------------------+----------- src_ip | character varying(39) | not null proto | character varying(4) | not null from_pattern | character varying(64) | not null Indexes: "trusted_pkey" PRIMARY KEY, btree (src_ip, proto, from_pattern) Am I missing a table, or a column somewhere? Is this a ser issue, or a db/sys-admin/permissions issue? Should I also provide my ser.cfg? Many thanks for any help/advice! ( also, is it possible to use the ser-0.10.x ser_postgres.sh/ser_dbtext.sh scripts for ser-0.9.4, rather than doing it all by hand? )

18 years, 1 month

[Users] flags and pseudo variables problem

by Fatih Cerit

Hi all When the save function called the value of FLAG is 1 and when we want to get this from pseudo $mf the value is 64. I only set flag to 6. Is there any conversation at here or something wrong?? Thanks Fatih Cerit

18 years, 1 month

[Users] Problem with Call Forwarding

by Darryl Ross

Hey All, I have tried to set up forwarding using a document for SER which I found at http://siprouter.onsip.org/doc/gettingstarted/ch10.html. In trying to get it to work though I am having a couple of issues: 1) Call forward immediate and not available works, but the billing party is set to the caller and it needs to be rewritten to be the account doing the forward. 2) Call forward busy and no answer don't work at all. The caller's phone drops the call and the forwarding destination phone does not ring at all. I believe the solution to both of those problems is the same: find out how branching works and set it up properly, but I am struggling to find some documentation to help. Does anyone have a sample config they can send me. I'm sure it would help enormously! I'm using OpenSER 1.1.0 with mediaproxy 1.7.2 on debian sarge. Regards Darryl

18 years, 1 month

[Users] Mediaproxy newbie

by Marnus van Niekerk

Hi, I am trying to set up openser with mediaproxy (at xx.xx.xx.133) to route calls from UA behind NAT to asterisk as voicemail (at xx.xx.xx.134) and PSTN gateways (at xx.xx.xx.32) I can see in the SDP payload that the RTP is being sent from asterisk to mediaproxy, but in sessions.py it shows the private ip not the public one and I have one way audio. Can anybody help please. opnser.cfg below. Marnus -- debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) log_facility=LOG_LOCAL6 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 "/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/textops.so" loadmodule "/usr/local/lib/openser/modules/uri_db.so" loadmodule "/usr/local/lib/openser/modules/domain.so" loadmodule "/usr/local/lib/openser/modules/mediaproxy.so" loadmodule "/usr/local/lib/openser/modules/nathelper.so" # Logging loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/auth.so" loadmodule "/usr/local/lib/openser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- modparam("usrloc", "db_mode", 0) modparam("usrloc", "db_mode", 2) modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) #tm timeout for voicemail params modparam("tm", "fr_timer", 3) modparam("tm", "fr_inv_timer", 35) modparam("tm", "noisy_ctimer", 1) # parms for NAT/mediaproxy 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/openser/sip-clients") modparam("mediaproxy", "rtp_asymmetrics", "/usr/local/etc/openser/rtp-clients") modparam("registrar", "nat_flag", 6) # ------------------------- request routing logic ------------------- # main routing logic route { #check for old messages: could mean a problem with the DNS entries or some other loop-causer... if (!mf_process_maxfwd_header("10")) { xlog("L_WARN", "WARNING: Too many hops\n"); sl_send_reply("483", "Too many hops, forward count exceeded limit\n"); return; }; #check for extremely large messages; we don't need a sip dos attack if (msg:len >= 2048) { xlog("L_WARN", "WARNING: Message too large, &>= 2048 bytes\n"); sl_send_reply("513", "Message too large, exceeded limit\n"); return; }; # Track what is happening xlog("L_INFO", "SIP Request: method [$rm] from [$fu] to [$tu]\n"); #record everything besides registers and acks if(method!="REGISTER" && method!="ACK") { setflag(1); }; # Record Route Section if (method=="INVITE" && client_nat_test("3")) { record_route_preset("xx.xx.xx.133:5060;nat=yes"); } else if (method!="REGISTER") { record_route(); } # Call tear down section if (method=="BYE" || method=="CANCEL") { end_media_session(); } #do not send to voicemail if BYE or CANCEL #is used to end call before user pickup or timeout if(method=="CANCEL" || method=="BYE") { setflag(10); }; #grant route if route headers already present if (loose_route()) { # May need client_nat_test & use_media_proxy here... route(1); return; }; #Always require authentication, which could result in a PSTN if (method=="REGISTER") { if (!search("^Contact:[ ]*\*") && client_nat_test("7")) { setflag(6); fix_nated_register(); force_rport(); }; if(!www_authorize("domain.tld", "subscriber")) { www_challenge("domain.tld", "0"); return; } else { if (!check_to()) { sl_send_reply("401", "Unauthorized"); return; }; #Save into user database, used below when checking if user is available xlog("L_INFO", "REGISTER: User $fu Authenticated Correctly\n"); save("location"); return; }; }; if (method=="INVITE") { if (client_nat_test("3")) { setflag(7); force_rport(); fix_nated_contact(); }; if(uri=~"sip:\*86@.*") { #authorize if a call is going to VM if(!proxy_authorize("domain.tld", "subscriber")) { proxy_challenge("domain.tld", "0"); return; }; xlog("L_INFO", "CALL: Call from $fu to check voicemail\n"); rewritehostport("vm.domain.tld:5060"); } else { if (does_uri_exist()) { #Call is to sip client, so do nothing but route xlog("L_INFO", "CALL: Sip client\n"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); xlog("L_ERROR", "ERROR: User $tu Not Found\n"); return; }; } else { #authorize if a call is going to PSTN if(!proxy_authorize("domain.tld", "subscriber")) { proxy_challenge("domain.tld", "0"); return; }; #Call destination is PSTN, so send it to the gateway xlog("L_INFO", "CALL: PSTN $tu from $fu \n"); rewritehostport("ast1.domain.tld:5060"); }; }; #Make sure that all subsequent requests go through us; #done at the top already #record_route(); } else { if (does_uri_exist()) { #Call is to sip client, so do nothing but route xlog("L_INFO", "CALL: Sip client\n"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); xlog("L_ERROR", "ERROR: User $tu Not Found\n"); return; }; } else { #Call destination is PSTN, so send it to the gateway xlog("L_INFO", "CALL: PSTN $tu from $fu \n"); rewritehostport("ast1.domain.tld:5060"); }; #record_route(); }; #ALL PROCESSING IS DONE, SO ROUTE route(4); route(1); } route[1] { #send the call outward if(method=="INVITE" && !isflagset(10)) { t_on_failure("2"); # voicemail if failure }; if (!t_relay()) { xlog("L_WARN", "ERROR: t_relay failed"); sl_reply_error(); }; } # ----------------------------------------------------------------- # NAT Traversal Section # ----------------------------------------------------------------- route[4] { if (isflagset(6) || isflagset(7)) { if (!isflagset(8)) { setflag(8); use_media_proxy(); }; }; } failure_route[2] { if(!t_was_cancelled() && !t_check_status("407")) { revert_uri(); rewritehostport("vm.domain.tld:5060"); append_branch(); #PREVENT SOME CRAZY VOICEMAIL LOOP xlog("L_INFO", "INFO: CALL TO VOICEMAIL"); setflag(10); route(1); } } onreply_route[1] { if ((isflagset(6) || isflagset(7)) && (status=~"(180)|(183)|2[0-9][0-9]")) { if (!search("^Content-Length:[ ]*0")) { use_media_proxy(); }; }; if (client_nat_test("1")) { fix_nated_contact(); }; } -- "Opportunity is missed by most people because it is dressed in overalls and looks like work." Thomas Alva Edison - Inventor of 1093 patents, including the light bulb, phonogram and motion pictures.

18 years, 1 month

Re: [Users] OpenSER not working after running for several days

by Mark Kent

>> Locking it to 10/full was what the MCI technician recommended after >> I opened a ticket because the line was dropping to 500Kbs >> up/down. This was a 4Mbit up/down line burstable to 10. They said >> it was not possible to increase the capacity of this line, so I >> guess they are using their old 10Mbit equipment for this. Since >> locking it like they asked speed has been fine, OK, it wasn't clear that you were talking about a specialized circuit delivered from a telco. BTW, this phrase "the line was dropping to 500Kbs up/down" implies that there was some protocol synching taking place that negotiated the speed at that level (like a dsl line). It would be more accurate to type "my observed throughput was only 500Kbs up/down." >> and the cable is a 2 meter factory made one plugging into the >> datacenter patch-panel, so I don't think that's a problem... My reference to the cable was for a situation where we're talking about a real local area network, where you might have a central switching fabric and machines distributed perhaps throughout a building and was meant to apply to the cable run from the main switches to the openser box. So, now I don't get your set-up... when you typed "I had to lock-down the network card to 10mbit full-duplex" and then the bit above about the MCI technician and the 4Mbit circuit etc. then this sounds like you have a box running openser that is directly plugged into a metro-lan-style connection that is hardcoded at the provider end to 10/full. And yet, when you type "datacenter patch-panel" this implies that there is a local area network which implies some sort of central switching fabric and then when I consider "old 10Mbit equipment" together with the phrase "datacenter" my jaw hits the ground... Where is your server? Thanks, -mark

18 years, 1 month

[Users] Difference between Alias and Listen

by Mark Kent

>> Could someone please explain to me the actual difference between *Alias=*and >> *Listen=*. I have read the documentation ( >> [snip] >> listen=udp:sip1.foobar.com:5060 /* Domain 1 */ Ahh, this links back to your previous message and adds a little light. IMO, it is not good practice to put domain names for listen. I would only list IP addresses, and that list should be relatively small because you don't need the IP address to distinguish between clients. For example, I just have one listen command in my config. For a critical service like this, either hard-coding in the IP addresses to listen on, or be willing to listen on all attached interfaces, should be Standard Operating Procedure (I vote for the former). >> alias="sip1.foobar.com:5060" /* Aliases for Domain 1 */ Don't put the port in there. Just list the domains that the server is responsible for. >> Now if I understand correctly then *if (uri==myself) { }* should >> now be true for each of these domains. I think this should be explained more fully by someone in the know... For example, if you are using the "domain" module then would we expect to never have to use "uri==myself" and not have to list aliases? (using is_uri_host_local() instead) An earlier hint from someone here to me said something like "openser doesn't know that the message is directed at itself" and I wonder how the use of the "domain" module affects things *inside* the openser core. And then there is the environment variable SIP_DOMAIN... does that have any bearing on the function of openser or is it just used for the supporting scripts (openserctl). What is the role of this variable in a multi-domain setting? -mark

18 years, 1 month

[Serusers] Serweb: Forbidden, You don't have permission to access /serweb/admin/index.php

by lin9k＠yahoo.com

Hi, After followed the INSTALL file set up the serweb. Have configure Alias /serweb "/var/serweb/html" in httpd.conf But while trying to browse http://192.168.0.200/serweb/admin/index.php the error message showed up on the broswer: Forbidden You don't have permission to access /serweb/admin/index.php on this server If I just browse form the local machine as: /var/serweb/html/admin/index.php the content of index.php would show up What could be wrong of my setting? I have: httpd-2.2.3 php-5.1.6 serweb-0.9.4 installed in my redhat 9.0 machine where ser-0.9.6 has been working fine. Thanks a lot for help. Ling __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

18 years, 1 month

Re: [Users] Difference between Alias and Listen

by Mark Kent

>> Thus, openser internally has a list of aliases it is responsible of. >> This list includes: >> [snip] >> - all the domain names found out by reverse lookups of the listening IPs I see that Richard Bennett had discovered that as he earlier typed: == it seems something was trying to look-up the aliases == that that server itself was listening on where he likely meant IP addresses and not aliases. I have to say that this is a big mistake. Doing the lookups is a bad idea and basing any sort of operational aspects of openser on the domains culled from such reverse lookups is even worse. The list of supported domains should be purely under the explicit configuration control of the person running the openser box... either with explicit "alias" statements, or with the use of the "domain" module. I could support this claim with a lengthy, experience supported, argument. But I'll save time with just this: 70.22.4.4.in-addr.arpa name = wbar1.sjo1-4-4-022-070.sjo1.dsl-verizon.net. -mark

18 years, 1 month

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sr-users October 2006