sr-users March 2006

sr-users@lists.kamailio.org

223 participants
506 discussions

[Serusers] Speed Dial logging
by sip 28 Mar '06

28 Mar '06

Is there a trick to logging with the SpeedDial module? Right now, acc is only logging the dialed number, not the number that sd_lookup() resolves to. Since the URI rewrite is inside the module, is there an easy way to change that? Is there a hard way to change that? That sort of breaks some functionality. N.

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[Users] Accounting of branched request - value of o_uri
by Arek Bekiersz 28 Mar '06

28 Mar '06

Hello, When using failure_route[] with db accounting enabled, the o_uri stored in accounting report in SER "acc" table contains o_uri of original request. Is there any way to store the real o_uri, which was after branching? Is there any other way to store in DB the IP request was forwarded to, without using AVPs? Consider example: modparam("acc", "db_flag", 3) route{ # if we do not get a positive reply, continue at route1 setflag(3); t_on_failure("1"); rewritehost( "A"); t_relay_to_udp("A", "5060"); break; } failure_route[1] { # forwarding failed -- try again at another destination t_on_failure("2"); rewritehost( "B"); append_branch(); t_relay_to_udp("B", "5060"); } failure_route[2] { # try out the last resort destination rewritehost( "C"); append_branch(); t_relay_to_udp("C", "5060"); # we no more call t_on_negative here; if this destination # fails too, transaction will complete } When INVITE will be answered in failure_route[1] or 2, it will contain o_uri=A. I would like this to contain B or C, according to real situation. -- Regards, Arek Bekiersz

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[Users] Message to SMS config- help
by Andreas M. 28 Mar '06

28 Mar '06

Hello, has anyone a working config to send SMS via SIP MESSAGES ? I´ve already installed a SMS Modem, that works with smstools, but i´m not able to send a MESSAGE via SMS through openser, in my debug log there are always a lot of failures regarding the tm module, so i think i missconfigured something. Also the param "domain_str" is not acceptet in my config. In the docs there are some examples, but where to place in existing configs (old ser configs i guess)? If anyone could leave me a sample, it would be very helpfull. best regards, Andreas Matzel

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[Users] 2 OpenSERs with TLS
by Joao Pereira 28 Mar '06

28 Mar '06

Hello to all I have 2 OpenSERs working with TLS. They both have registered clients (X-Lite). How can I be sure that they are exchanging the certificates when one call goes from serverA to serverB ? Is there any debug where we can see it happening? Thanks Joao

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[Users] export SIP_Domain
by D'Addelfio Davide 28 Mar '06

28 Mar '06

Hi to all, i'm totally new in openser and i'd like to enjoy it... I've installed 1.0.1 version on my debian and seems it work fine with mysql database. Just a stupid question: I miss to export SIP_DOMAIN cause I can't understand what name I have to give...the default is openser.org...and I've seen on install file that I have to change it...could someone explain me what I have to do, please? Thanks in advance Davide

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[Serusers] ANNOUNCE: WIST - Web Interface for SIP Trace
by Rodrigo P. Telles 28 Mar '06

28 Mar '06

Hi Folks, I'm glad to announce WIST for SIP debug/trace dialogs. This software born as a prof concept of the idea to capture SIP traffic from a remote host (SIP Proxy, Gateway, etc) and show up alive SIP messages about an specific dialog (filtered by From SIP user) to help our tech support team to debug SIP transactions in a friendly way. http://www.devel-it.org/index.php?modulo=projetos&lang=en_US We hope you enjoy our work. Telles

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[Users] registering problems when using TLS
by Joao Pereira 28 Mar '06

28 Mar '06

Hello to all Im using openser with TLS (1.0.1-tls) but when I try to register a client (X-Lite), the answer is always: "Too Many Hops" because of this: if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; Whats the problem? Without TLS, this error doesnt happen. Thanks Joao Pereira

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[Serusers] SER PSTN Failover
by Darren Nay 28 Mar '06

28 Mar '06

Hello All, I am having a problem with the t_on_failure failure route using SER. I have been using SER for 2+ years now and am servicing 5000+ voip customers but now we are moving to a PSTN termination partner who does not have redundant gateways. So I would like to setup SER so that I have call route advance to a 2nd PSTN carrier if the 1st fails. However, I am running into some snags... With my configuration the call is processed correctly if the 1st PSTN carrier takes the call. However, when the failover happens then I get some odd behavior. The call is initially setup correctly with the failover carrier, but then it hangs up after about 5-7 seconds and a new call comes in to the called number, and a loop begins where this same call keeps coming in one after another until SER is restarted. I do get the following errors on the SER console. 0(17304) forward_req: ERROR: cannot forward to af 2, proto 2 no corresponding listening socket 0(17304) Warning: sl_send_reply: I won't send a reply for ACK!! 0(17304) ERROR: sl_reply_error used: I'm terribly sorry, server error occurred (7/SL) It almost seems as if the the 1st route PSTN carrier is not accepting my ACK when I send it in response to the 403 Forbidden that they send to me. Then they just keep sending the 403 Forbidden over and over again. Each time that they do that then SER processes it as a failed call and starts a new call to the secondary PSTN carrier. Does this sound possible? Or am I way off base here? I have attached my basic lab test SER cfg that I have been using to test with. Perhaps someone can tell me what I might be doing incorrectly? Or - Is there a better way to do this? ie. With AVP-Pops? If so, does anyone have a sample configuration that I can look at? Thanks so much for any help provided!! Darren Nay Ionosphere, Inc. dnay(a)ionosphere.net +1 864-678-3158 # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) memlog=0 #uncomment for debug mode debug=3 fork=no log_stderror=yes #Bind IP listen=XX.XX.XX.XX check_via=no # (cmd. line: -v) dns=yes # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=5 mhomed=no # multi-homed? fifo="/tmp/ser_fifo" # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/nathelper.so" loadmodule "/usr/local/lib/ser/modules/textops.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/xlog.so" # ----------------- setting module-specific parameters --------------- # -- timers -- modparam("tm", "wt_timer", 10) modparam("tm", "retr_timer1p1", 2) # -- usrloc params -- modparam("usrloc", "db_mode", 0) # -- rr params -- modparam("rr", "enable_full_lr", 1) # -- Nathelper -- modparam("registrar", "nat_flag", 6) modparam("nathelper", "natping_interval", 10) modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "rtpproxy_disable", 1) alias=proxy.ionosphere.net # ------------------------- request routing logic ------------------- # main routing logic route { # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len > max_len) { sl_send_reply("513", "Message too big"); break; }; # Nathelper if (method == "REGISTER" || !search("^Record-Route:.*XX.XX.XX.XX")) { fix_nated_contact(); # Rewrite contact with source IP of signaling force_rport(); setflag(6); }; # 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(); # loose-route processing if (loose_route()) { route(1); break; }; lookup("aliases"); if (!uri==myself) { 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") { fix_nated_register(); save("location"); break; }; }; ### Process URI routing # Now we determine if called number is on # or off net and route accordingly if (!lookup("location")) { route(2); # Nothing found - Route Off-Net } else { route(1); # Direct match - Route On-Net }; } # # No direct route via location or location_novm (Off-Net Route) # route[2] { t_on_failure("2"); #Send the call to Level3 rewritehostport("XX.XX.XX.XX:5060"); if (!t_relay()) { sl_reply_error(); }; break; } # # Direct route found via location or location_novm (On-Net Route) # route[1] { # !! Nathelper if (uri=~"[@:](192\.168\.|10\.|172\.(1[6-9]|2[0-9]|3[0-1])\.)" && !search("^Route:")) { sl_send_reply("479", "We don't forward to private IP addresses"); break; }; # if client or server know to be behind a NAT, enable relay if (isflagset(6)) { # NAT processing of replies; apply to all transactions (for example, # re-INVITEs from public to private UA are hard to identify as # NATed at the moment of request processing); look at replies t_on_reply("1"); append_hf("P-Behind-NAT: Yes\r\n"); }; # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } onreply_route[1] { if (status =~ "(183)|(180)|2[0-9][0-9]") { fix_nated_contact(); }; if (nat_uac_test("1")) { fix_nated_contact(); }; break; } failure_route[2] { # forwarding failed try again at another destination rewritehostport("XX.XX.XX.XX:5060"); append_branch(); t_relay(); break; }

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[Users] what responses can drop() drop
by Dr. Rodney G. McDuff 28 Mar '06

28 Mar '06

I am trying to drop a 302 redirect response from a invite and branch a new invite off somewhere else. In my onreply_route[1] I have a if (condition) { xlog(various stuff); drop(); } Using ngrep I see that the response is not drop. Can I have some confirmation that drop only drops 1XX provisional responses. The documentations is a bit fuzzy about this. -- 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 |

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[Users] Issue w/ OpenSER + Asterisk on same box
by Corey S. McFadden 28 Mar '06

28 Mar '06

Hi, I understand this is a fairly typical setup for Voicemail and media but we're running OpenSER and Asterisk on the same box. (OpenSER on 5060 and Asterisk on 5061.) We also use rtpproxy. Here are the scenarios: UA -> OpenSER -> UA (no problems) UA -> OpenSER -> External Asterisk PSTN Gateway (no problems) UA -> OpenSER -> External Asterisk Voicemail (no problems) UA -> OpenSER -> Same box Asterisk Voicemail - NO AUDIO I need to get this running on the same box and am not sure what to do next. This is probably something silly / obvious, so if anyone can point me in the right direction it would be greatly appreciated! Thanks in advance, -Corey ********************************************* This message has been scanned for viruses and dangerous content, and is believed to be clean.

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sr-users March 2006