- sr-users - kamailio.org

[Serusers] SCTP support in SER?
by Magnus Sörman 14 Jun '08

14 Jun '08

Hi, Does SER 2.0 support SIP over SCTP? If so, how do I configure it? Thanks in advance, /Magnus

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[OpenSER-Users] How to detect if both UAs are from the same LAN
by Pete Kay 14 Jun '08

14 Jun '08

Hi, I am using .nat_uac_test to detect if the UA is from behind a NAT and if so use rtpproxy. However, if both UA are behind the "same" NAT, is there anyway to tell? If both UAs are from the same LAN then I don't need to apply rtpproxy. What would be a good way to do that? Any suggestion will be greatly appreciated. Regards, Pete

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[OpenSER-Users] LCR not finding a route?
by David Villasmil 14 Jun '08

14 Jun '08

Hello all, I'm trying to configure the LCR module, but I'm running into a little problem, I don't get any errors but it is not routing the call properly, this is my lcr info: openserctl lcr show database engine 'MYSQL' loaded Control engine 'FIFO' loaded lcr routes +----+--------+----------+--------+----------+ | id | prefix | from_uri | grp_id | priority | +----+--------+----------+--------+----------+ | 1 | 001 | | 1 | 1 | +----+--------+----------+--------+----------+ lcr gateway groups +--------+----------+ | grp_id | grp_name | +--------+----------+ | 1 | GWs | +--------+----------+ lcr gateways +---------+---------------+------+------------+-----------+--------+-------+---------------+ | gw_name | ip_addr | port | uri_scheme | transport | grp_id | strip | prefix | +---------+---------------+------+------------+-----------+--------+-------+---------------+ | USA | 1.2.3.4 | 5060 | 1 | 1 | 1 | 2 | 99 | +---------+---------------+------+------------+-----------+--------+-------+---------------+ I'm sending the following call: Jun 14 07:49:31 rgi-sip /sbin/openser[9458]: DBG:core:parse_msg: SIP Request: Jun 14 07:49:31 rgi-sip /sbin/openser[9458]: DBG:core:parse_msg: method: <INVITE> Jun 14 07:49:31 rgi-sip /sbin/openser[9458]: DBG:core:parse_msg: uri: < sip:0013058883456@9.8.7.6:5060> Jun 14 07:49:31 rgi-sip /sbin/openser[9458]: DBG:core:parse_msg: version: <SIP/2.0> my script says: # routing to the public network if (!load_gws()) { xlog("L_INFO","LCR: Unable to load gateways"); sl_send_reply("503", "Unable to load gateways"); exit; }else{ xlog("L_INFO","LCR: Sending to $td\n"); } if(!next_gw()){ sl_send_reply("503", "Unable to find a gateway"); exit; } t_on_failure("2"); if (!t_relay()) { sl_reply_error(); }; exit; On the syslog i get the message: LCR: Sending to 9.8.7.6 <-- This, although the IP is wrong, I KNOW it didn't fail loading the gateways... where 9.8.7.6 if my sever's IP. Why is it not matching? Thanks a lot David

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[OpenSER-Users] OPENSER: 3 years
by Daniel-Constantin Mierla 14 Jun '08

14 Jun '08

Hello everybody, today is the third anniversary of the OPENSER project. There were three years of development, community events and fun. And it is more to come... I want to thank to all that invested time and money into the project, writing code, offering support on mailing lists and forum, traveling around the world to promote the project, developing documentation and tools, disclosing their usage of OPENSER in production environments setting trust to this SIP server. It will be a huge list to detail everything, openser.org and google reveal very good the OPENSER environment for the past three years. Looking forward to next year! Cheers, Daniel -- http://www.asipto.com

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[OpenSER-Users] Deploying Openser in HA
by mark morreny 14 Jun '08

14 Jun '08

Hi, What is the best way to deploy Openser in an HA environment? If I want to use ldirector, what is the best solution for determining whether the openser is still alive? Could someone share with me how it can be done? Thanks, Mark

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[OpenSER-Users] Loose route problem or misunderstanding
by Watkins, Bradley 13 Jun '08

13 Jun '08

I'm having a problem using loose_route, and I'm not sure if it's a bug/feature/misunderstanding. I have a B2BUA (Asterisk) running on the same system as OpenSER, and in my failure_route for voicemail I direct calls there which actually sends them on to the voicemail server. If someone 'zeroes out' of voicemail, to be forwarded to the operator, the voicemail server issues a re-INVITE to redirect media. This is where the problem comes in. If I try to loose_route this re-INVITE, it fails. However, putting the B2BUA on a different system works fine. This is running OpenSER 1.2.3 currently, though I'm not particularly averse to upgrading to the latest 1.3.x if for some reason this is something that's been resolved there. Any suggestions? Thanks in advance, - Brad

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[OpenSER-Users] Multiple login problem
by Chaitanya V.N.S 13 Jun '08

13 Jun '08

Hi, I have a problem with openser. When multiple clients are registered with the same userid [ example A] When another user [ say B] tried to call A then the invite message goes to last logined client but not to both clients! Is that a problem with my configuration or a behaviour of Openser Thanks in advance chaitu

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Re: [OpenSER-Users] Radius accounting and Dialog
by David Villasmil 13 Jun '08

13 Jun '08

SORRY TO HAVE REPLIED TO YOUR EMAIL!!! I hate it when gmail does that! Thanks, It rather very easy, I need all calls to go through the media-proxy. I'm testing behind nat, server is on the internet. If I make a call from one UAC natted to another UAC natted, it works fine, rtps go to the proxy. If I call from behind NAT to echo test, I don't get any audio. rtps go directly to the echo test server's IP, but i don't get anything back. I'm doing something very wrong, I've modifed the script like a thousand time, and still... On Fri, Jun 13, 2008 at 7:25 PM, David Villasmil < david.villasmil.work(a)gmail.com> wrote: > Thanks, > > It rather very easy, I need all calls to go through the media-proxy. > I'm testing behind nat, server is on the internet. If I make a call from one > UAC natted to another UAC natted, it works fine, rtps go to the proxy. If I > call from behind NAT to echo test, I don't get any audio. rtps go directly > to the echo test server's IP, but i don't get anything back. I'm doing > something very wrong, I've modifed the script like a thousand time, and > still... > > > On Fri, Jun 13, 2008 at 7:11 PM, Raúl Alexis Betancor Santana < > rabs(a)dimension-virtual.com> wrote: > >> El Viernes, 13 de Junio de 2008 17:47, David Villasmil escribió: >> > That was not my intention at all and I'm sorry you got it that way. I >> did >> > take a look at my script and I thought I should do was eliminate all >> tests >> > for nats and simply call use_media_proxy when an INVITE comes in, and an >> > on_reply was hit. Also of course end_media_session() n a BYE or a >> CANCEL... >> > But it is still not working, well it instruct my UAC to send rtps >> directly >> > to the echo test ip... and this wasn't working. >> > >> > And because I did not see anything work (I don't have much experience >> with >> > media-proxy) I thought I'd ask here... my bad. ;-) >> > >> > Of course I will keeo looking at this, I don't want anyone to do my >> job... >> > ;-) >> > >> >> Ok, so from that point on ... what it's your test scenario, paint it and >> we >> could guide you to the right point. >> >> We do not have a crystal ball ;-) >> >> -- >> Saludos. >> >> Raúl Alexis Betancor Santana >> Dimensión Virtual S.L. >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users >> > >

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[OpenSER-Users] Carrierroute error
by Pablo Bertuzzi 13 Jun '08

13 Jun '08

Hello everyone, i am testing the carrierroute module to do the following: -If destination starts with 1760, the call is sent to PSTN GW. -If destination starts with 1305, the call is blocked. The lab works as desired, however i am getting this error: ERROR:carrierroute:carrier_rewrite_msg: during rewrite_uri_recursor, uri 13054941522 , carrier 0, domain 0 These are my tables: mysql> select * from carrierroute; +----+---------+-------------+--------+------+-------+---------------+----------------+----------------+---------+ | id | carrier | scan_prefix | domain | prob | strip | rewrite_host | rewrite_prefix | rewrite_suffix | comment | +----+---------+-------------+--------+------+-------+---------------+----------------+----------------+---------+ | 1 | 0 | 1760 | 0 | 1 | 0 | 70.165.183.36 | | | NULL | | 2 | 0 | 1305 | 0 | 1 | 0 | | | | NULL | +----+---------+-------------+--------+------+-------+---------------+----------------+----------------+---------+ 2 rows in set (0.00 sec) mysql> select * from route_tree; +----+---------+ | id | carrier | +----+---------+ | 0 | default | +----+---------+ 1 row in set (0.00 sec) And this is my config: if (is_method("INVITE")) { if(!cr_rewrite_uri("0", "call_id")){ sl_send_reply("403", "Not allowed"); exit(); } } any ideas? best regards, Pablo.

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[Serusers] Problems running ser 0.9.7
by John G. Rose 13 Jun '08

13 Jun '08

<=========================================================================== = I'm having a problema trying to run ser-0.9.7. The compilation was fine.. but i'm not able to start SER due to this error : Mar 25 00:42:53 edgelcr ser: loading module /usr/local/lib/ser/modules/acc.so Mar 25 00:42:53 edgelcr ser: ERROR: load_module: could not open module </usr/local/lib/ser/modules/acc.so>: /usr/local/lib/ser/modules/acc.so: undefined symbol: parse_orig_uri Mar 25 00:42:53 edgelcr ser: parse error (88,13-14): failed to load module <=========================================================================== = I have the same problem on OpenSUSE 10.3. SER 0.9.7 compiles fine but when you try to run it you get that error. Was this ever resolved? John

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Re: [OpenSER-Users] Radius accounting and Dialog
by Raúl Alexis Betancor Santana 13 Jun '08

13 Jun '08

El Viernes, 13 de Junio de 2008 12:34, escribió: > Yes, I have audio. Attahced is the ngrep It's not possible that you have audio, because on media proxy stats you see a 0 packet relayed, so that why you have a timeout. What it's your test scenario ? EyeBeam 1 <-> NAT router <-> ^ v OpenSer + MediaProxy ^ v EyeBeam 2 <-> NAT router <-> This one?, are both EyeBeam behind the same NAT? Sip and nat problems are very hard to solve headache P.D: Just reply only to the list, no to me, that will be easier to follow the thread. -- Saludos. Raúl Alexis Betancor Santana Dimensión Virtual S.L.

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[OpenSER-Users] OPenser+Media-Proxy
by David Villasmil 13 Jun '08

13 Jun '08

Hello all, I'm trying to force all calls through mediaproxy, but can't seem to be able to do it... eyebeam doesn't do it while sjphone does... What am I doing wrong? Calls between eyebam and sjphone work fine but call to echo test, failes, rtps are going to the echo test ip directly... not going through mediaproxy... This is really killing me, any help would be greatly appreciated!! here's my config # # $Id: openser.cfg 1676 2007-02-21 13:16:34Z bogdan_iancu $ # # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=10 port=5060 listen=MY.OPENSER.IP.COM disable_dns_blacklist=true # ------------------ module loading ---------------------------------- #set module path mpath="//lib/openser/modules/" # Uncomment this if you want to use SQL database #loadmodule "mysql.so" loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "uri.so" loadmodule "uri_db.so" loadmodule "domain.so" loadmodule "permissions.so" #loadmodule "group.so" loadmodule "mi_fifo.so" loadmodule "lcr.so" loadmodule "avpops.so" loadmodule "xlog.so" loadmodule "mediaproxy.so" loadmodule "acc.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "auth.so" loadmodule "auth_db.so" # ----------------- setting module-specific parameters --------------- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("registrar", "received_avp", "$avp(i:42)") modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 4) modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "password_column", "password") modparam("rr", "enable_full_lr", 1) #modparam("auth_db|permissions|uri_db|usrloc","db_url","mysql://openser:openserrw@localhost /openser") modparam("auth_db|permissions|usrloc|uri_db|domain|lcr|avpops|avp_url","db_url","mysql://openser:password@localhost /openser") modparam("permissions", "db_mode", 1) modparam("permissions", "trusted_table", "trusted") modparam("avpops", "avp_table", "usr_preferences") modparam("mediaproxy","natping_interval",20) modparam("mediaproxy","mediaproxy_socket", "/var/run/mediaproxy/mediaproxy.sock") modparam("mediaproxy","sip_asymmetrics","/etc/openser/sip-clients") modparam("mediaproxy","rtp_asymmetrics","/ect/openser/rtp-clients") modparam("lcr", "contact_avp", "$avp(i:711)") modparam("lcr|tm", "fr_inv_timer_avp", "$avp(i:704)") modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("^auth$|lcr", "rpid_avp", "$avp(i:302)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "dm_flag", 25) modparam("acc", "radius_config", "/etc/radiusclient-ng/radiusclient.conf") modparam("acc", "radius_flag", 2) modparam("acc", "radius_missed_flag", 3) modparam("acc", "radius_extra", "User-Name=$Au;Calling-Station-Id=$from;Called-Station-Id=$to;Sip-Translated-Request-URI=$ruri;Sip-RPid=$avp(s:rpid);Source-IP=$si;Source-Port=$sp;Canonical-URI=$avp(s:can_uri);Billing-Party=$avp(s:billing_party);Divert-Reason=$avp(s:divert_reason);X-RTP-Stat=$hdr(X-RTP-Stat);Contact=$hdr(contact);Event=$hdr(event);SIP-Proxy-IP=$avp(s:sip_proxy_ip);ENUM-TLD=$avp(s:enum_tld)") # ------------------------- request routing logic ------------------- # main routing logic route{ # # -- 1 -- Request Validation # if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; # # -- 2 -- Routing Preprocessing # ## Record-route all except Register ## Mark packets with nat=yes ## This mark will be used to identify the request in the loose ## route section if(!is_method("REGISTER")){ record_route(";nat=yes"); }; ##Loose_route packets if (has_totag()) { #sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if(method=="BYE") { #Account BYE transactions setflag(2); }; #Check authentication of re-invites if(method=="INVITE" && (!allow_trusted())) { if (!proxy_authorize("","subscriber")) { proxy_challenge("","1"); exit; } else if (!check_from()) { sl_send_reply("403", "Forbidden, use From=ID"); exit; }; }; if(method=="BYE" || method=="CANCEL") { end_media_session(); }; ##Detect requests in the dialog behind NAT and flag with 6 route(1); } else { sl_send_reply("404","Not here"); } exit; } #CANCEL processing if (is_method("CANCEL")) { if (t_check_trans()) { end_media_session(); t_relay(); }; exit; } t_check_trans(); # # -- 3 -- Determine Request Target # if (method=="REGISTER") { route(2); } else { route(3); }; } route[1] { # # -- 4 -- Forward request to target # # Forward statefully t_on_reply("1"); t_on_failure("1"); if (!t_relay()) { sl_reply_error(); }; exit; } route[2] { ## Register request handler if (is_uri_host_local()) { if (!www_authorize("", "subscriber")) { www_challenge("", "1"); exit; }; if (!check_to()) { sl_send_reply("403", "Forbidden"); exit; }; setbflag(6); force_rport(); save("location"); exit; } else if { sl_send_reply("403", "Forbidden"); }; } route[3] { ## Requests handler if (method=="INVITE") { # Account INVITE packets setflag(2); # Account Missed calls setflag(3); # Radius Extra $avp(s:sip_proxy_ip)="MY.OPENSER.IP.COM"; }; if (is_from_local()){ # From an internal domain -> check the credentials and the FROM if(!allow_trusted()){ if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); exit; } else if(!check_from()) { sl_send_reply("403", "Forbidden, use From=ID"); exit; }; }; #unconditional call forward if(avp_db_load("$ru/username","$avp(s:callfwd)")) { avp_pushto("$ru", "$avp(s:callfwd)"); route(1); exit; } consume_credentials(); #verify aliases, if found replace R-URI. lookup("aliases"); if (is_uri_host_local()) { # -- Inbound to Inbound xlog("L_INFO","OPENSER: INBOUND -> INBOUND ($tU)"); route(10); } else { # -- Inbound to outbound xlog("L_INFO","OPENSER: INBOUND -> OUTBOUND ($tU)"); route(11); }; } else { #From an external domain ->do not check credentials #Verify aliases, if found replace R-URI. lookup("aliases"); if (is_uri_host_local()) { xlog("L_INFO","OPENSER: OUTBOUND -> INBOUND ($tU)"); #-- Outbound to inbound route(12); } else { xlog("L_INFO","OPENSER: OUTBOUND -> OUTBOUND ($tU)"); # -- Outbound to outbound route(13); }; }; } route[4] { # routing to the public network if (!load_gws()) { sl_send_reply("503", "Unable to load gateways"); exit; } if(!next_gw()){ sl_send_reply("503", "Unable to find a gateway"); exit; } t_on_failure("2"); if (!t_relay()) { sl_reply_error(); }; exit; } route[6] { # # -- NAT handling -- # append_hf("P-hint: Route[6]: mediaproxy \r\n"); use_media_proxy(); } route[10] { #from an internal domain -> inbound #Native SIP destinations are handled using the location table #Gateway destinations are handled by regular expressions append_hf("P-hint: inbound->inbound \r\n"); xlog("L_INFO","LOOKING FOR: $tU"); if (uri=~"^sip:55613") { # ECHO TEST xlog("L_INFO","IN ROUTE 613"); route(6); rewritehostport("fwdnat2.pulver.com:5060"); if (!t_relay()) { sl_reply_error(); }; exit; }; if (uri=~"^sip:301") { # ECHO TEST route(6); rewritehostport("blueface.ie"); if (!t_relay()) { sl_reply_error(); }; exit; }; # # if (uri=~"^sip:1[2-9][0-9]{9}@") { # if (is_user_in("credentials","ld")) { # strip(1); # prefix("+1"); # route(6); # route(4); # exit; # } else { # sl_send_reply("403", "No permissions for long distance"); # exit; # }; # }; # # if (uri=~"^sip:011[0-9]*@") { # if (is_user_in("credentials","int")) { # strip(3); # prefix("+"); # route(6); # route(4); # exit; # } else { # sl_send_reply("403", "No permissions for international calls"); # }; # }; if (!lookup("location")) { if (does_uri_exist()) { ## User not registered at this time. ## Use the IP Address of your e-mail server revert_uri(); prefix("u"); rewritehostport("192.168.1.171"); #Use the IP address of your voicemail server route(6); route(1); } else { sl_send_reply("404", "Not Found"); exit; } sl_send_reply("404", "Not Found"); exit; }; route(6); route(1); } route[11] { # from an internal domain -> outbound # Simply route the call outbound using DNS search append_hf("P-hint: inbound->outbound \r\n"); route(1); } route[12] { # From an external domain -> inbound # Verify aliases, if found replace R-URI. lookup("aliases"); if (!lookup("location")) { sl_send_reply("404", "Not Found"); exit; }; route(1); } route[13] { #From an external domain outbound #we are not accepting these calls append_hf("P-hint: outbound->inbound \r\n"); sl_send_reply("403", "Forbidden"); exit; } failure_route[1] { ##-- ##-- If cancelled, exit. ##-- if (t_was_cancelled()) { exit; }; ##-- ##-- If busy send to the e-mail server, prefix the "b" ##-- character to indicate busy. ##-- if (t_check_status("486")) { revert_uri(); prefix("b"); rewritehostport("192.168.1.171"); append_branch(); route(1); exit; }; ##-- ##-- If timeout (408) or unavailable temporarily (480), ##-- prefix the uri with the "u"character to indicate ##-- unanswered and send to the e-mail ##-- sever ##-- if (t_check_status("408") || t_check_status("480")) { revert_uri(); prefix("u"); rewritehostport("192.168.1.171"); append_branch(); route(1); exit; }; } failure_route[2] { if(!next_gw()) { t_reply("503", "Service not available, no more gateways"); exit; }; t_on_failure("2"); t_relay(); } onreply_route[1] { # #-- On-replay block routing -- # append_hf("P-hint: onreply_route|usemediaproxy \r\n"); use_media_proxy(); exit; }

1 0

[OpenSER-Users] Incoming call routing from a SIP Provider
by Stagg Shelton 13 Jun '08

13 Jun '08

Hello, I have just started investigating OpenSER, and am looking for some guidance. I have several Asterisk PBX's registered to OpenSER and each can make outgoing calls to my sip provider without any issue. I have not figured out how to route incoming calls based on DID to the appropriate PBX. Should I be looking at aliases, or carrierroute, or something else entirely. I would appreciate any help you can give for this newbie question. Thank You Stagg Shelton www.sheltonjohns.com

2 1

[OpenSER-Users] ACC Error
by Ali Jawad 13 Jun '08

13 Jun '08

Hi All I am using the ACC module and I only have this parameter set modparam("acc", "failed_transaction_flag", 1) and I get this error Jun 13 09:09:27 [23981] INFO:acc:mod_init: initializing... Jun 13 09:09:27 [23981] DBG:core:find_cmd_export_t: found <load_tm>(0) in module tm [/usr/local/lib/openser/modules/] Jun 13 09:09:27 [23981] ERROR:acc:init_acc_rad: acc: can't get code for the Sip-Response-Code attribute Jun 13 09:09:28 [23981] ERROR:acc:mod_init: failed to init radius Jun 13 09:09:28 [23981] ERROR:core:init_mod: failed to initialize module acc Jun 13 09:09:28 [23981] ERROR:core:main: error while initializing modules Jun 13 09:09:28 [23981] DBG:jabber:destroy: unloading module ...

2 1

Re: [OpenSER-Users] Dialog module, timeout_avp and record_route()
by Henning Westerholt 13 Jun '08

13 Jun '08

On Thursday 12 June 2008, kionez wrote: > > is used. Some logs (from trunk): > > [cut] > > > Jun 11 14:37:52 ca ../openser[23446]: INFO:dialog:get_dlg_timeout: > > > invalid AVP value, use default timeout > > I try to set my debug level to 9, but i can't reproduce this behaviour, > i never seen "invalid AVP value".. what version are you using? Hi Kionez, i just commited this additional log messages yesterday to the trunk version, rev 4373. I thought this could be useful in debugging this problem. :-) Henning

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[OpenSER-Users] Disable the safety checks in the perl module
by Yazid Hadj Said 13 Jun '08

13 Jun '08

Hi, I am trying to disable the safety checks in the perl module. I use modparam("perl","unsafemodfnc", 1), this parmeter is not documented, I find it in source code. But I have message indicating that this parameter does not exist. Regards, Yazid

2 2

[OpenSER-Users] Out Of Memory in OPENSER
by dilip.m＠renovau.com 13 Jun '08

13 Jun '08

Hello Everyone, I have instlled openser-1.1-notls and radiusclient-ng.I have 4 GB memory in my pc. In the openser, generally Shared memory size is 32MB and private meory is 1 MB. For the call firing i am using sipp testing tool and my call rete is 100(-r is 100 ). I got out of pkg memory error in the Radius module during call firing. When i incraese the private memory in the openser,the no of calls will become higher,but i increase the shared memory,there is not any effect in the call.The no of calls remain same. But everytime i got the "no more free memory" and "out of memory" Error in the auth_radius module and everytime i haveto restart the openser. Plz tell me what should i have to change in the auth_radius module,so that Out Of Memory problem is solve. Regards, Dilip

2 1

[Serusers] fake RBT
by Ivo Kirilov 13 Jun '08

13 Jun '08

1 0

[OpenSER-Users] load balance based by weight
by Gang Liu (VSC) 13 Jun '08

13 Jun '08

Hi, I have 3 trunk gateways, two are cisco 5300 with 4E1, the other is 16E1 audio codes. Could I use dispatch module to load balance those gateways based weight I set. I successfully setup dispatcher module to load 50%, 50% for two gateway. regards, Danile

3 2

[OpenSER-Users] Trouble with avp in openser 1.3.2
by Robert Dyck 13 Jun '08

13 Jun '08

With the following in cfg -- exec_avp("uname -s","$avp(s:test)"); xlog("L_NOTICE","test result is <$avp(s:test)>\n"); 1.3.1 yields "test result is <linux>" 1.3.2 yields "test result is <>" I also noticed that 1.3.2 handles an uninitialized $var differently than 1.3.1. with the following in cfg -- if !$var(test) == "foo" {xlog("L_NOTICE", "test not equal to foo\n");}; 1.3.1 yields "test not equal to foo" 1.3.2 yields "CRITICAL:core:comp_scriptvar: cannot get left var value" "WARNING:core:do_action: error in expression" Is this behaviour intentional Thanks RD

1 1

[Serusers] registration to ser server fails
by Shrirang Yardi 13 Jun '08

13 Jun '08

Hello, I am a complete novice to VoIP and am trying to setup SER for the first time...so some of my questions may sound naive. I have installed ser-0.96 on x86 machine running fedora core 9. I followed the getting started guide to start ser using the hello-world.cfg config file. The service starts succesfully: Listening on udp: <ser_ip> [ser_ip]:5060 tcp: <ser_ip> [ser_ip]:5060 where ser_ip is the IP address of the server. I verified that the ser processes are actually running. I then used a UA from ekiga.net to register with the server, but the registration fails - reason: timeout. I dont have any problems pinging the ser_ip. What am I missing? Any help will be appreciated. Thanks, Shri

1 0

[OpenSER-Users] Lost BYEs and Carriers
by David Villasmil 12 Jun '08

12 Jun '08

Hello all, Now I got a question: I understand that sometimes, when a UAC droppes out of the internet, a BYE message will be lost. But, if I were to use Openser as a Carrier switch, this is NOT supposed to happen, right? as all VoIP Carriers MUST be on the internet permanently. Any thoughts? David

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[OpenSER-Users] concurrent call limit per user / customer
by Patrick Miccio 12 Jun '08

12 Jun '08

Hi @ all, I was wondering if anyone of you guys already implemented a system where one could limit the maximum number of concurrent calls? With ISDN-terminaladapters or analog-telefon-adapters it is no problem because the hardware itself is the limiting device, but with the growing amount of IP PBXs the need becomes obvious. I am playing with the thought of writing something myself, but I am more a copy&paste programmer, so it will take quite some time :D Maybe someone here likes the idea? Here are my thoughts of how a module could provide a simple functionality: INVITE: check 200 OK: create Re-INVITE: update BYE: delete must provide: create_call() update_call() delete call() check_limit() create must save uuid,from,to,callid,timestamp in DB, from/to just for human readability ;) update must update timestamp with uuid/callid match ( we enforce 300sec re-invite call refresh ) delete must delete the DB entry with uuid/callid match check must count DB entries with uuid match and return amount of active calls. cheers, Patrick.

3 3

[OpenSER-Users] Problem with OPENXCAP
by ADALA Asma 12 Jun '08

12 Jun '08

Hello everybody; I've installed *openser v1.3.2* and it works perfectly. Then, i''ve installed *openxcap v0.9.9. i'm using Uctimsclient as ims client. * When i try to upload or download an xml document using uctimsclient, i have this error in log messages : *error: XCAP root not found for request URI: http://127.0.0.1/xcap-root/ *My xcap root is: *127.0.0.1:8000/xcap-root* and i set the same in my client. *Can anyone help me please. thx *

1 0

[OpenSER-Users] multi-leg accounting
by Ruchir Lists 12 Jun '08

12 Jun '08

Hi All, I'm trying to configure OpenSER with multi-leg accounting. I'm using OpenSER 1.2 & radius. I'm using acc_radius table for writing cdr records. I've searched through several articles and mailing list posts about configuring multi-leg accounting and everywhere they talk about setting up multi-leg-info parameter of acc module to configure leg source & destination and setting up accounting flag. But this is not working for me. I get only BYE record in table if I use this way to write multiple records for one call in call forwarding scenario. I managed to get multiple records by calling acc_rad_request on INVITE, ACK, BYE & CANCEL. For instance, I have 4 users; 90001, 90002, 90003, 90004. The call forwarding is setup as below: 90004 -> 90001 -> 90003. I dial 90004 from 90002 user and it forwards the call to 90001 & then 90003 and they're connected properlry. However I don't get leg source/destination properly and also I get 7-8 bye records for this call. Can anyone guide me in right direction if I'm doing anything wrong. Regards, Ruchir

3 11

[OpenSER-Users] call functions with Perl (moduleFunction())
by Yazid Hadj Said 12 Jun '08

12 Jun '08

Hi, Please, i want to know why when i try to use remove_hf() and append_hf() functions with Perl (moduleFunction(func,string1,string2)), I have this message: Jun 12 11:03:53 [12838] ERROR:core:moduleFunc: Module function 'remove_hf' is unsafe. Call is refused. Jun 12 11:03:53 [12838] ERROR:core:XS_OpenSER__Message_moduleFunction: calling module function 'remove_hf' failed. Missing loadmodule? Jun 12 11:03:53 [12838] ERROR:core:moduleFunc: Module function 'append_hf' is unsafe. Call is refused. Jun 12 11:03:53 [12838] ERROR:core:XS_OpenSER__Message_moduleFunction: calling module function 'append_hf' failed. Missing loadmodule? Thanks, Yazid

2 2

[OpenSER-Users] Dialog module, timeout_avp and record_route()
by kionez 12 Jun '08

12 Jun '08

Hi all, i'm trying to debug a strange behaviour of dialog mode. If I use "record_route" in my script, to force any message to toward the server, dialog module stops to check "timeout_avp", if I comment it out, when timeout_avp is reached, the dialog module notify me in sylog (as shown in log.txt) It seems that dialog module has some problems to work if i use record_route()... I try to use version 1.3.1 and 1.3.2, but i see the same behaviour. So, where i'm wrong? ;) Here's relevant part of config, and in openser.txt there's my routing script ---- loadmodule "rr.so" modparam("rr", "enable_full_lr", 1) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) loadmodule "dialog.so" modparam("dialog", "enable_stats", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "timeout_avp", "$avp(s:dlgtimeout)") modparam("dialog", "dlg_extra_hdrs", "Hint: credit expired\r\n") modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_url", "mysql://dbuser:dbpass@localhost/openser") modparam("dialog", "db_mode", 1) ---- thanx in advance k. # With record_route() Jun 11 09:36:07 New request - M=INVITE RURI=sip:alice@192.168.27.74 F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.82 ID=2112176925 Jun 11 09:36:07 BRANCH - S=<null> R=sip:alice@192.168.1.201:5060;line=2c542c23af8e7c7 D=<null> F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.82 ID=2112176925 Jun 11 09:36:07 Reply - S=101 R=<null> D=Dialog Establishement F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.201 ID=2112176925 Jun 11 09:36:07 Reply - S=100 R=<null> D=Trying F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.201 ID=2112176925 Jun 11 09:36:07 Reply - S=180 R=<null> D=Ringing F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.201 ID=2112176925 Jun 11 09:36:09 Reply - S=200 R=<null> D=OK F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.201 ID=2112176925 Jun 11 09:36:09 New request - M=ACK RURI=sip:alice@192.168.1.201:5060 F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.82 ID=2112176925 Jun 11 09:36:26 New request - M=BYE RURI=sip:alice@192.168.1.201:5060 F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.82 ID=2112176925 Jun 11 09:36:26 Reply - S=200 R=<null> D=OK F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.201 ID=2112176925 # Without record_route() Jun 11 09:36:49 New request - M=INVITE RURI=sip:alice@192.168.27.74 F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.82 ID=1675712119 Jun 11 09:36:49 BRANCH - S=<null> R=sip:alice@192.168.1.201:5060;line=2c542c23af8e7c7 D=<null> F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.82 ID=1675712119 Jun 11 09:36:49 Reply - S=100 R=<null> D=Trying F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.201 ID=1675712119 Jun 11 09:36:49 Reply - S=101 R=<null> D=Dialog Establishement F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.201 ID=1675712119 Jun 11 09:36:49 Reply - S=180 R=<null> D=Ringing F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.201 ID=1675712119 Jun 11 09:36:51 Reply - S=200 R=<null> D=OK F=sip:bob@192.168.27.74 T=sip:alice@192.168.27.74 IP=192.168.27.201 ID=1675712119 Jun 11 09:36:56 WARNING:dialog:dlg_ontimeout: timeout for dlg with CallID '1675712119' and tags '53233056' '453709088' ####### Routing Logic ######## route[0] { xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(loose_route()) { if(!has_totag()) { xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } t_on_reply("1"); t_on_failure("1"); } if(!is_method("REGISTER")) { #record_route(); } if(is_method("CANCEL") || is_method("ACK")) { t_on_reply("1"); if(t_check_trans()) { if(!t_relay()) { sl_reply_error(); } } else { xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } exit; } t_check_trans(); if(is_method("REGISTER")) { sl_send_reply("100", "Trying"); if(!save("location")) { xlog("L_ERR", "Register FAIL - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Register OK - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } if(is_method("INVITE")) { $avp(s:dlgtimeout)=5; $avp(s:ringtimeout)=10; setflag(4); sl_send_reply("100", "Trying"); } if(is_method("BYE")) { route(1); } if (!lookup("location")) { switch ($retcode) { case -1: case -3: t_newtran(); t_reply("404", "Not Found"); exit; case -2: sl_send_reply("405", "Method Not Allowed"); exit; } } route(1); } route[1] { if (is_method("INVITE")) { t_on_branch("1"); t_on_reply("1"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); }; exit; } branch_route[1] { xlog("L_INFO", "BRANCH - S=$rs R=$ru D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); } onreply_route[1] { xlog("L_INFO", "Reply - S=$rs R=$ru D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); } failure_route[1] { if (t_was_cancelled()) { exit; } }

2 3

[OpenSER-Users] stun and symmetric
by Pezhman Lali 12 Jun '08

12 Jun '08

Dear, I find that stun and symmetric nat can not work together , but some big sip providers like sipgate or fwd, added stun settings to their ata as default without, any question about customer's nat type. where is the key ?

2 1

Re: [Serusers] stun and s
by Pezhman Lali 12 Jun '08

12 Jun '08

> well, I would hope, if it's their ATA, they probably > know that it > doesn't do symmetric NAT.... How ??? can ATAs behind router or dsl modem, select nat type ? --- On Thu, 6/12/08, Harry Behrens <hb(a)4snewcom.de> wrote: > From: Harry Behrens <hb(a)4snewcom.de> > Subject: Re: [Serusers] stun and s > To: pezhman_lali(a)yahoo.com > Cc: serusers(a)lists.iptel.org > Date: Thursday, June 12, 2008, 11:49 AM > Pezhman Lali schrieb: > > Dear, > > I find that stun and symmetric nat can not work > together , > > but some big sip providers like sipgate or fwd, added > stun settings to their ata as default without, any question > about customer's nat type. > > > > > > well, I would hope, if it's their ATA, they probably > know that it > doesn't do symmetric NAT.... How ??? > > > where is the key ? > > > > > > > open? > > -h > > > > > _______________________________________________ > > Serusers mailing list > > Serusers(a)lists.iptel.org > > http://lists.iptel.org/mailman/listinfo/serusers > > > > > >

1 0

[Serusers] stun and s
by Pezhman Lali 12 Jun '08

12 Jun '08

Dear, I find that stun and symmetric nat can not work together , but some big sip providers like sipgate or fwd, added stun settings to their ata as default without, any question about customer's nat type. where is the key ?

1 0

[OpenSER-Users] First route in Route header not being removed
by Robert Dyck 11 Jun '08

11 Jun '08

Using 1.3.2 notls I found that when an Ekiga softphone is configured with an outgoing proxy, it will include a Route header in an initial INVITE. The Route header contains a single URI pointing to the outgoing proxy. This allowed according to RFC 3261 section 8.1.1.1 "When a provider wishes to configure a UA with an outbound proxy, it is RECOMMENDED that this be done by providing it with a pre-existing route set with a single URI, that of the outbound proxy. " Since the first route in a Route header over rides a request URI, this Route URI must be removed prior to routing or the request will loop. Openser will loop when Ekiga is configured with Openser as the outgoing proxy. Of course the header could be removed using the openser.cfg but the operation should be built in.

3 8

[OpenSER-Users] pua_xmpp Notify2Xmpp error
by Gaoqiang Qian 11 Jun '08

11 Jun '08

Hello everyone, I have setup my openser 1.3.2 with presence, pua_xmpp, xmpp modules according to the openser.org dokuwiki about pua_xmpp configuration. From SIP client(eyebeam) and XMPP client(pigdin), IM is now working fine between two sides with below destination address: sip: "sip:username<delim>jabber_server@gateway_domain" xmpp: "sip_username<delim>openser_domain@xmpp_domain" But the presence only works in one side, in xmpp client, it shows the availabity of sip contacts, but on sip client, the xmpp contacts are always offline. I got below line error continously loged: Jun 11 20:44:33 ims /home/openser/openser1.3.2/sbin/openser[16077]: ERROR:pua_xmpp:Notify2Xmpp: 'To' header ALREADY PARSED:<sip:test*ovi.com@xmpp.rcs.demo> Could anyone tell me what might be the reason. Thanks a lot in advance.

1 0

[OpenSER-Users] Why a $var cannot be 'null' ?
by Iñaki Baz Castillo 11 Jun '08

11 Jun '08

Hi, if I set: $var(kk) = null and later printf it (or compare with =~) then I see '0': xlog("L_INFO", "var(kk) = $var(kk)\n"); => var(kk) = 0 But AVP's do allow 'null' value and store 'null' instead of '0'. Is there any reason for it? -- Iñaki Baz Castillo

3 4

[OpenSER-Users] compiling ldap module on freebsd 7.0-release
by Dian Dwi Nugraha 11 Jun '08

11 Jun '08

Hi, I've been having problem with compiling OpenSER with LDAP module on FreeBSD 7.0-RELEASE. I have 'ldap.so' and put it under /usr/local/lib/ openser/modules/. The problem occured when I tried to run openser. --- Error Message --- Jun 11 19:49:53 [16442] ERROR:core:sr_load_module: could not open module </usr/local/lib/openser/modules/ldap.so>: /usr/local/lib/ openser/modules/ldap.so: Undefined symbol "ber_sockbuf_io_debug" Any advice will help. Thanks. Best regards, --- Dian Dwi Nugraha > Computer Engineering Department > School of Informatics and Electrical Engineering > Institut Teknologi Bandung > dian132(a)students.ee.itb.ac.id

2 5

Re: [OpenSER-Users] Error Compiling
by Ali Jawad 11 Jun '08

11 Jun '08

Did a chmod 777.. -----Original Message----- From: Henning Westerholt [mailto:henning.westerholt@1und1.de] Sent: Wednesday, June 11, 2008 6:05 PM To: Ali Jawad Subject: Re: [OpenSER-Users] Error Compiling On Wednesday 11 June 2008, Ali Jawad wrote: > Hi Henning > I was able to fix it thanks to your help. I have one last hurdle. When I > try to start openser now I get this error > ERROR:acc:init_acc_rad: failed to open radius config file: > /etc/raddb/clients.conf > > However /etc/raddb/clients.conf exists Hi Ali, perhaps this is a permission problem? Henning -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

1 0

[OpenSER-Users] Error Compiling
by Ali Jawad 11 Jun '08

11 Jun '08

Hi All I did remove all the radius mods from the exclude section of the Makefile. And I did enable radius in the Makefile of the acc module. All compiled well howver when I try to load the acc module I get the following error ERROR:core:sr_load_module: could not open module </usr/local/lib/openser/modules/acc.so>: libradiusclient-ng.so.2: cannot open shared object file: No such file or directory However the file in question can be found at /usr/local/lib/libradiusclient-ng.so.2 With Regards

2 5

[OpenSER-Users] OpenSER-1.1.1 crashes
by dilip.m＠renovau.com 11 Jun '08

11 Jun '08

Hello Everyone, I have inlcude auth_diameter module in the openser.cfg. It works fine with DISC. I can authorize the user with same. When i increase the call rate using sipp tool at that time the openser going to crash. It disconnect the connection. The openser shows me this output. 1(19698) SIP Request: 1(19698) method: <REGISTER> 1(19698) uri: <sip:192.168.1.66> 1(19698) version: <SIP/2.0> 1(19698) parse_headers: flags=2 1(19698) end of header reached, state=5 1(19698) parse_headers: Via found, flags=2 1(19698) parse_headers: this is the first via 1(19698) After parse_msg... 1(19698) preparing to run routing scripts... 1(19698) parse_headers: flags=100 1(19698) DEBUG:parse_to:end of header reached, state=10 1(19698) DBUG:parse_to: display={1113}, ruri={sip:1113@192.168.1.66:5060} 1(19698) DEBUG: get_hdr_field: <To> [35]; uri=[sip:1113@192.168.1.66:5060] 1(19698) DEBUG: to body [1113 <sip:1113@192.168.1.66:5060> ] 1(19698) get_hdr_field: cseq <CSeq>: <1> <REGISTER> 1(19698) DEBUG:maxfwd:is_maxfwd_present: value = 70 1(19698) parse_headers: flags=200 1(19698) DEBUG: get_hdr_body : content_length=0 1(19698) found end of header 1(19698) find_first_route: No Route headers found 1(19698) loose_route: There is no Route HF 1(19698) grep_sock_info - checking if host==us: 12==12 && [192.168.1.66] == [192.168.1.66] 1(19698) grep_sock_info - checking if port 5060 matches port 5060 1(19698) grep_sock_info - checking if host==us: 12==12 && [192.168.1.66] == [192.168.1.66] 1(19698) grep_sock_info - checking if port 5060 matches port 5060 1(19698) parse_headers: flags=4000 1(19698) auth_diameter:pre_auth(): Credentials with given realm not found 1(19698) ******* m_id=3 1(19698) DEBUG:do_read (sock=8): whole message read (len=304)! 1(19698) ######## m_id=3 1(19698) parse_headers: flags=ffffffffffffffff 1(19698) check_via_address(192.168.1.1, 192.168.1.1, 0) 1(19698) build_auth_hf(): 'WWW-Authenticate: Digest realm="192.168.1.66", nonce="484e1d9a81643c688e09493c7b615dacbc004e81", qop="auth" ' 1(19698) parse_headers: flags=ffffffffffffffff 1(19698) check_via_address(192.168.1.1, 192.168.1.1, 0) 1(19698) DEBUG:destroy_avp_list: destroying list (nil) 1(19698) receive_msg: cleaning up 4(19703) SIP Request: 4(19703) method: <ACK> 4(19703) uri: <sip:192.168.1.66> 4(19703) version: <SIP/2.0> 4(19703) parse_headers: flags=2 4(19703) end of header reached, state=5 4(19703) parse_headers: Via found, flags=2 4(19703) parse_headers: this is the first via 4(19703) After parse_msg... 4(19703) preparing to run routing scripts... 4(19703) parse_headers: flags=8 4(19703) DEBUG: add_param: tag=71db33f84c952ca903e2273ad59f1848.3074 4(19703) DEBUG:parse_to:end of header reached, state=29 4(19703) DBUG:parse_to: display={sut}, ruri={sip:1113@192.168.1.66:5060} 4(19703) DEBUG: get_hdr_field: <To> [76]; uri=[sip:1113@192.168.1.66:5060] 4(19703) DEBUG: to body [sut <sip:1113@192.168.1.66:5060>] 4(19703) DEBUG: sl_filter_ACK : local ACK found -> dropping it! 4(19703) DEBUG:destroy_avp_list: destroying list (nil) 4(19703) receive_msg: cleaning up 2(19699) SIP Request: 2(19699) method: <REGISTER> 2(19699) uri: <sip:192.168.1.66> 2(19699) version: <SIP/2.0> 2(19699) parse_headers: flags=2 2(19699) end of header reached, state=5 2(19699) parse_headers: Via found, flags=2 2(19699) parse_headers: this is the first via 2(19699) After parse_msg... 2(19699) preparing to run routing scripts... 2(19699) parse_headers: flags=100 2(19699) DEBUG:parse_to:end of header reached, state=10 2(19699) DBUG:parse_to: display={1113}, ruri={sip:1113@192.168.1.66:5060} 2(19699) DEBUG: get_hdr_field: <To> [35]; uri=[sip:1113@192.168.1.66:5060] 2(19699) DEBUG: to body [1113 <sip:1113@192.168.1.66:5060> ] 2(19699) get_hdr_field: cseq <CSeq>: <2> <REGISTER> 2(19699) DEBUG:maxfwd:is_maxfwd_present: value = 70 2(19699) parse_headers: flags=200 2(19699) DEBUG: get_hdr_body : content_length=0 2(19699) found end of header 2(19699) find_first_route: No Route headers found 2(19699) loose_route: There is no Route HF 2(19699) grep_sock_info - checking if host==us: 12==12 && [192.168.1.66] == [192.168.1.66] 2(19699) grep_sock_info - checking if port 5060 matches port 5060 2(19699) grep_sock_info - checking if host==us: 12==12 && [192.168.1.66] == [192.168.1.66] 2(19699) grep_sock_info - checking if port 5060 matches port 5060 2(19699) ******* m_id=3 2(19699) DEBUG:do_read (sock=9): whole message read (len=208)! 2(19699) ######## m_id=3 2(19699) parse_headers: flags=ffffffffffffffff 2(19699) parse_headers: flags=8000000 2(19699) parse_headers: flags=ffffffffffffffff 2(19699) build_contact(): Created Contact HF: Contact: <sip:1113@192.168.1.5:5060>;expires=27, <sip:1113@192.168.1.1:5060>;expires=3600 2(19699) parse_headers: flags=ffffffffffffffff 2(19699) check_via_address(192.168.1.1, 192.168.1.1, 0) 2(19699) DEBUG:destroy_avp_list: destroying list (nil) 2(19699) receive_msg: cleaning up 4(19703) SIP Request: 4(19703) method: <ACK> 4(19703) uri: <sip:192.168.1.66> 4(19703) version: <SIP/2.0> 4(19703) parse_headers: flags=2 4(19703) end of header reached, state=5 4(19703) parse_headers: Via found, flags=2 4(19703) parse_headers: this is the first via 4(19703) After parse_msg... 4(19703) preparing to run routing scripts... 4(19703) parse_headers: flags=8 4(19703) DEBUG: add_param: tag=71db33f84c952ca903e2273ad59f1848.3074 4(19703) DEBUG:parse_to:end of header reached, state=29 4(19703) DBUG:parse_to: display={sut}, ruri={sip:1113@192.168.1.66:5060} 4(19703) DEBUG: get_hdr_field: <To> [76]; uri=[sip:1113@192.168.1.66:5060] 4(19703) DEBUG: to body [sut <sip:1113@192.168.1.66:5060>] 4(19703) DEBUG: sl_filter_ACK : local ACK found -> dropping it! 4(19703) DEBUG:destroy_avp_list: destroying list (nil) 4(19703) receive_msg: cleaning up 10(19717) ERROR: receive_fd: EOF on 8 10(19717) DBG: handle_ser_child: dead child 3, pid 19701 (shutting down?) 10(19717) DBG: io_watch_del (0x6d31a0, 8, -1, 0x0) fd_no=14 called 0(19695) child process 19701 exited by a signal 11 0(19695) core was generated 0(19695) INFO: terminating due to SIGCHLD 8(19712) INFO: signal 15 received 8(19712) 4(19703) 7(19710) Memory status (pkg): 2(19699) INFO: signal 15 received 2(19699) Memory status (pkg): 2(19699) fm_status (0x705700): 2(19699) heap size= 1048576 2(19699) used= 18682, used+overhead=56810, free=991766 2(19699) max used (+overhead)= 61502 2(19699) dumping free list: 2(19699) hash = 1 fragments no.: 2, unused: 0 bucket size: 8 - 8 (first 8) 2(19699) hash = 2 fragments no.: 3, unused: 0 bucket size: 16 - 16 (first 16) 2(19699) INFO: signal 15 received INFO: signal 15 received 8(19712) fm_status (0x705700): 8(19712) heap size= 1048576 8(19712) used= 75736, used+overhead=113464, free=935112 8(19712) 6(19708) hash = 3 fragments no.: 1, unused: 0 bucket size: 24 - 24 (first 30) 2(19699) hash = 6 fragments no.: 1, unused: 0 bucket size: 48 - 48 (first 48) 2(19699) hash = 8 fragments no.: 5, unused: 0 bucket size: 64 - 64 (first 64) 2(19699) 4(19703) 7(19710) INFO: signal 15 received 6(19708) Memory status (pkg): 6(19708) fm_status (0x705700): 6(19708) heap size= 1048576 6(19708) used= 75736, used+overhead=113464, free=935112 6(19708) max used (+overhead)= 113464 6(19708) dumping free list: 6(19708) hash = 27 fragments no.: 1, unused: 0 bucket size: 216 - 216 (first 216) 6(19708) hash = 39 fragments no.: 11, unused: 0 bucket size: 312 - 312 (first 312) 6(19708) hash = 73 fragments no.: 1, unused: 0 bucket size: 584 - 584 (first 584) 6(19708) hash = 192 fragments no.: 1, unused: 0 bucket size: 1536 - 1536 (first 1536) 6(19708) hash = 2054 fragments no.: 1, unused: 0 bucket size: 524288 - 1048576 (first 929344) 6(19708) TOTAL: 15 free fragments = 935112 free bytes 6(19708) ----------------------------- 1(19698) INFO: signal 15 received 1(19698) Memory status (pkg): 1(19698) fm_status (0x705700): 1(19698) heap size= 1048576 1(19698) used= 20040, used+overhead=58424, free=990152 1(19698) max used (+overhead)= 62264 1(19698) dumping free list: 1(19698) hash = 1 fragments no.: 2, unused: 0 bucket size: 8 - 8 (first 8) 1(19698) hash = 2 fragments no.: 3, unused: 0 bucket size: 16 - 16 (first 16) 1(19698) hash = 3 fragments no.: 1, unused: 0 bucket size: 24 - 24 (first 30) 1(19698) hash = 6 fragments no.: 1, unused: 0 bucket size: 48 - 48 (first 48) 1(19698) hash = 8 fragments no.: 5, unused: 0 bucket size: 64 - 64 (first 64) 1(19698) hash = 9 fragments no.: 9, unused: 0 bucket size: 72 - 72 (first 72) 1(19698) hash = 10 fragments no.: 1, unused: 0 bucket size: 80 - 80 (first 80) 1(19698) hash = 11 fragments no.: 1, unused: 0 bucket size: 88 - 88 (first 88) 1(19698) hash = 15 fragments no.: 1, unused: 0 bucket size: 120 - 120 (first 120) 10(19717) INFO: signal 15 received 10(19717) Memory status (pkg): 10(19717) fm_status (0x705700): I have also try with to solve it with gdb,but cant help. The gdb output as follows, (gdb) Loaded symbols for /usr/local/lib64/openser/modules/auth_diameter.so Reading symbols from /usr/local/lib64/openser/modules/domain.so...done. Loaded symbols for /usr/local/lib64/openser/modules/domain.so Reading symbols from /lib64/libnss_files.so.2...done. Loaded symbols for /lib64/libnss_files.so.2 Core was generated by `openser'. Program terminated with signal 11, Segmentation fault. #0 0x00002aaaabfa2b72 in AAAAddAVPToMessage (msg=0x713cd8, avp=0x713330, position=0x713b08) at avp.c:178 178 for(avp_t=msg->avpList.head;avp_t&&avp_t!=position;avp_t=avp_t->next); (gdb) What was the problem ?? Please explain me. Thanks in advance. Regards, Dilip Modi

3 2

Re: [OpenSER-Users] Why can't I "append_branch()" in FAILURE_ROUTE if reply is 6XX ??
by Igor Sidorov 11 Jun '08

11 Jun '08

Hi! Yes, it is very important and for us! LCR module. If we get 6XX response from the first gateway, it is impossible to redirect request to any others gateways . We have very strange situation! If the first gateway (with the best priority for some distinations) due his reasons answer with 6XX it destroy lcr logic at all !!! (and we do not want to change this gateway because we have no problems with others destinations(and some others reasons)) please, change this tm behavior asap. Or if you can suggest something else (i mean some rows of config code) ? thank you. >Why? doesn't it make sense to forward to some destination even if the negative >reply is a 4XX or 6XX? -- View this message in context: http://www.nabble.com/Why-can%27t-I-%22append_branch%28%29%22-in-FAILURE_RO… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

3 4

[OpenSER-Users] stun
by Pezhman Lali 10 Jun '08

10 Jun '08

--- On Mon, 6/9/08, Pezhman Lali <pezhman_lali(a)yahoo.com> wrote: > From: Pezhman Lali <pezhman_lali(a)yahoo.com> > Subject: stun > To: pezhman.lali(a)gmail.com > Date: Monday, June 9, 2008, 6:02 PM > Dear, > does setting the ip of stun server in the sip-phones, > behind the symmetric nat, make the problem ? > my experience with stund 0.96, said yes. > the stun server, can detects the type of nats properly, but > the sip-phones behind the un-symmetric nat, can not > register, or one-way calling. > > ????

6 9

Re: [OpenSER-Users] stun
by Aymeric Moizard 10 Jun '08

10 Jun '08

On Tue, 10 Jun 2008, David Villasmil wrote: > True (about getting the IP from the register), but that's putting more > processing on the server, I'd rather let ths UAC do the ip resolution when > we're talking thousands of clients (Which I am) The behavior I described is made on the client/endpoint which look at the received and rport in the answer to a request sent to the server. (and this is not a DNS resolution: this is just "looking at"...) The server in this case don't have any processing to do anymore: he always receive the correct contact header. No need for STUN to have a correct IP/port for SIP even behind symmetric NAT. ;) tks, Aymeric MOIZARD / ANTISIP amsip - http://www.antisip.com osip2 - http://www.osip.org eXosip2 - http://savannah.nongnu.org/projects/exosip/ > Cheers > > On Tue, Jun 10, 2008 at 1:59 PM, Aymeric Moizard <jack(a)atosc.org> wrote: > >> >> Stun can work even behind symmetric NAT if the stun server was >> running on the same socket the SIP server is running... I hope >> this feature will come soon! >> >> I can help with this. >> >> Also, SIP don't need to use STUN to work: you can discover your >> IP address and port very easily by looking at the REGISTER answer >> or OPTIONS answer which contains the "received" and "rport" >> parameter. >> >> Of course, symmetric NAT are a nightmare for RTP, but not for SIP... >> >> tks, >> Aymeric MOIZARD / ANTISIP >> amsip - http://www.antisip.com >> osip2 - http://www.osip.org >> eXosip2 - http://savannah.nongnu.org/projects/exosip/ >> >> >> >> On Tue, 10 Jun 2008, Iñaki Baz Castillo wrote: >> >> El Tuesday 10 June 2008 10:44:51 Ali Jawad escribió: >>> >>>> Hi All >>>> >>>> I had a lot of problems with STUN and symmetric NAT. The thing is that >>>> it does not always work with symmetric NAT. That would be mostly routers >>>> based on Linux IPTABLES. >>>> >>> >>> STUN **CANNOT** work with symetric NAT. The NAT router maps the internal >>> port >>> with a different public port *depending* on the destination IP:port, so >>> STUN >>> can't work. >>> >>> -- >>> Iñaki Baz Castillo >>> ibc(a)in.ilimit.es >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.openser.org >>> http://lists.openser.org/cgi-bin/mailman/listinfo/users >>> >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users >> >> >

1 0

[OpenSER-Users] loose_route() and t_relay(proto:host:port, [flags])
by Anatoly Pidruchny 10 Jun '08

10 Jun '08

Hi All, I have something like this in my script: ... if (loose_route()) { ... t_relay("udp:mydomain2.com"); ... } ... The idea is that the initial INVITE message can be relayed to one destination (mydomain1.com). But later, for the BYE message, if that destination becomes unavailable and we know about that, then we send the message to mydomain2.com. The problem is that it does not work that way. Even if mydomain2.com is passed to t_relay, the message is still sent to mydomain1.com. It looks like the parameter is ignored when loose routing is done. Is it a bug or a feature? If it is a feature, then the documentation for the t_relay function should mention that the parameter is ignored when loose routing is done. Another question is, how to make it send the message to mydomain2.com? Do I have to modify the message (like, modify the Route header)? Thanks, -------- Anatoly.

3 4

[OpenSER-Users] SIP max session timer
by Kaeman Chris 10 Jun '08

10 Jun '08

Please let me know how can i configure this sip max session timer on the openser? (SIP max timer) - If the openser doesn’t receive any BYE message from the origination or termination equipment than the SIP max timer will forcefully kill the call by generating the BYE on its own. Thanks Kchris --------------------------------- Explore your hobbies and interests. Click here to begin.

4 6

[OpenSER-Users] Dialog module... again!
by David Villasmil 10 Jun '08

10 Jun '08

Hello all, Got a question for those well versed on the Dialog Module. I have it working great against a db, but it seems to me that -although there's nothing on the module's documentations to indicate it- Dialogs are inserted on the table once the initial transaction is finished, i.e. 200 OK. Here's my script... # # $Id: openser.cfg 1827 2007-03-12 15:22:53Z bogdan_iancu $ # # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=4 check_via=no dns=no rev_dns=no listen=A.B.C.D disable_dns_blacklist=true port=5060 #set module path mpath="/lib/openser/modules/" # ------------------ module loading ---------------------------------- loadmodule "mysql.so" loadmodule "maxfwd.so" loadmodule "sl.so" loadmodule "dispatcher.so" loadmodule "tm.so" loadmodule "mi_fifo.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "rr.so" loadmodule "avpops.so" loadmodule "dialog.so" modparam("mi_fifo","fifo_name", "/tmp/openser_fifo") #Timer which hits if no final reply for a request #or ACK for a negative INVITE reply arrives modparam("tm", "fr_timer", 5) modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("avpops","avp_url","mysql://user:pass@localhost/openser") modparam("avpops", "avp_table", "dialog") modparam("rr","enable_full_lr", 1) modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_mode", 1) modparam("dialog", "dlg_match_mode", 1) modparam("dialog", "db_url", "mysql://user:pass@localhost/openser") 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if(loose_route()) { append_hf("P-hint: rr-enforced\r\n"); route(1); }; if (!method=="REGISTER") record_route(); if (method=="INVITE") { setflag(4); xlog("L_ERR","OPENSER: NEW INVITE $ru -> $ci \n"); }; if (method=="BYE"||method=="CANCEL") { setflag(4); xlog("L_ERR","OPENSER: DISCONNECT $ru -> $ci \n"); }; if ( method=="INVITE" || method=="ACK" || method=="BYE" || method=="OPTIONS" || method=="CANCEL" ) { ds_select_dst("1","2"); t_on_reply("1"); t_on_failure("1"); t_relay(); exit; }; } route[1] { # Forward Statefully t_on_reply("1"); t_on_failure("1"); if(!t_relay()){ sl_reply_error(); }; exit; } failure_route[1] { if (t_check_status("408")) { xlog("L_INFO","Marking GW as failed...\n"); ds_mark_dst("p"); ds_next_dst(); t_on_failure("1"); t_relay(); } else{ t_reply("501","Not Implemented"); xlog("L_ERR","OPENSER: FAILED $ru -> $ci \n"); } } thanks to all David

2 1

Re: [OpenSER-Users] multi-leg accounting
by Ruchir Lists 10 Jun '08

10 Jun '08

I've tried setting flags on INVITE, ACK & BYE all. On Mon, Jun 9, 2008 at 4:01 PM, David Villasmil < david.villasmil.work(a)gmail.com> wrote: > have you set the flag on INVITES as well? > > On Mon, Jun 9, 2008 at 11:01 AM, Ruchir Lists <ruchir.lists(a)gmail.com> > wrote: > >> Hi All, >> >> I'm trying to configure OpenSER with multi-leg accounting. I'm using >> OpenSER 1.2 & radius. I'm using acc_radius table for writing cdr records. >> I've searched through several articles and mailing list posts about >> configuring multi-leg accounting and everywhere they talk about setting up >> multi-leg-info parameter of acc module to configure leg source & destination >> and setting up accounting flag. But this is not working for me. I get only >> BYE record in table if I use this way to write multiple records for one call >> in call forwarding scenario. I managed to get multiple records by calling >> acc_rad_request on INVITE, ACK, BYE & CANCEL. For instance, I have 4 users; >> 90001, 90002, 90003, 90004. The call forwarding is setup as below: >> 90004 -> 90001 -> 90003. >> >> I dial 90004 from 90002 user and it forwards the call to 90001 & then >> 90003 and they're connected properlry. However I don't get leg >> source/destination properly and also I get 7-8 bye records for this call. >> Can anyone guide me in right direction if I'm doing anything wrong. >> >> Regards, >> Ruchir >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users >> >> >

1 0

[OpenSER-Users] Linux variant recommendation?
by Bill Brownrigg 10 Jun '08

10 Jun '08

We have done some basic development using OpenSER, now it is time to build a production system. The first application will be an alias_db application (using mySQL) to do call forwarding. We do expect to develop other applications in the future. We have some brand new Dell server class boxes for the project so we have plenty of hardware capacity. We do not have a lot of UNIX knowledge, so we would like to take advantage of a package based environment. What variant of Linux would you recommend for this system? E-MAIL CONFIDENTIALITY NOTICE: The contents of this e-mail message and any attachments are intended solely for the addressee(s) and may contain confidential and/or legally privileged information. If you are not the intended recipient of this message or if this message has been addressed to you in error, please alert the sender immediately by reply e-mail and then delete this message and any attachments. If you are not the intended recipient, you are notified that any use, dissemination, distribution, copying, or storage of this message or any attachment is strictly prohibited. Disclaimer added by CodeTwo Exchange Rules http://www.codetwo.com

4 3

[OpenSER-Users] need help on FreeRadius & CDRTools
by Do Nguyen Ha 10 Jun '08

10 Jun '08

hi i manage the Cisco BTS 10200 system, and now i am assigned a task to work with the Radius i choose the FreeRadius to work with Cisco BTS system modify some config in radiusd.conf & clients.conf first, BTS send message Access Request to FreeRadius and then Freeradius send message Access Accept, they working properly when BTS send message Account Start to Radius, there are a lot attribute and freeradius save them in the radacct table the messages flow: BTS FreeRadius -----Service Authorize------> <-------Access Accept-------- -----Service Start-----------> <-----Account Respone ---- ----Service ReAuthorize----> <-------Account Accept------> -----Service Stop-------------> <-------Account Respone---- but in radacct , there some columm doesn't match with the message Account Start, i want to change the name of some columm to match the message Should i need to edit the sql.conf for this issue On the CDR Tool, should i need to edit to working properly Attribute Vendo Sub Status Attribute Name Type Value ID ID Attribut M- User-Name String ANI 1 ID M- NAS-IP-Address IP ddres BTS IP address 4 M- Service-Type Integer Service Type = Outbound 6 C+ Class String CLASS Attribute 25 M- h323-incoming-conf-id=value String A unique number for identifying a calling session on a gateway, where a session is closed when the calling party hangs up. 26 9 1 M- h323-conf-id String Unique call identifier generated by the BTS. Used to distinguish the separate billable events (calls) within a single calling session. 26 9 24 M- h323-setup-time String hh:mm:ss.mmm ZON DDD MMM ## YYYY Setup time in NTP format. 26 9 25 M+ h323-connect-time String hh:mm:ss.mmm ZON DDD MMM ## YYYY Connect time in NTP format. 26 9 28 C- h323-call-type String Telephony,VOIP 30 27 C+ Called-Station-ID String DNIS 31 M+ Calling-Station-ID Integer ANI 40 M+ Acct-Status-Type String Acct-Status-Type = Start (1) 44 M+ Acct-Session-ID Integer Accounting Session ID 55 M- Event-Timestamp Integer Event Time Stamp 61 M- NAS-Port-Type Integer Ethernet (15) Thanks Ha`

2 1

[OpenSER-Users] 答复: how to configure to support PRACK
by Steven.Wu＠teleca.com 10 Jun '08

10 Jun '08

Hi I?aki, At first, I repied the PRACK without Route header but contian the to tag. I added the Route header after receive your opinion. But also failed. I attached my SIP message, could you please take a look them and give me some suggestion? Thank you! Best regards, Steven Wu UDP Data send to: 10.57.39.120:5060 INVITE sip:1141@10.57.39.120 SIP/2.0 Via: SIP/2.0/UDP 10.57.39.33:5060 From: sip:6846@10.57.39.120;tag=032baj647 To: sip:1141@10.57.39.120 Call-ID: asdbasdb3-asdb552 CSeq: 899 INVITE Contact: sip:6846@10.57.39.33 Max-Forwards: 70 Content-Type: application/sdp Expires: 180 Accept-Contact: *;+mckoppa Supported: precondition, 100Rel Require: precondition Content-Length: 392 v=0 o=Inviter2007 63241204263093750 132223800 IN IP4 10.57.39.33 s=- c=IN IP4 10.57.39.33 t=0 0 m=audio 5000 RTP/AVP 106 8 0 a=sendrecv a=rtpmap:106 AMR/8000 a=ptime:160 a=maxptime:200 a=fmtp:106 octet-align=1; mode-set=7 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=curr:qos local none a=curr:qos remote none a=des:qos mandatory local sendrecv a=des:qos none remote sendrecv UDP Data received from: 10.57.39.120:5060 SIP/2.0 183 Session Progress Record-Route: <sip:10.57.39.120;lr=on> Via: SIP/2.0/UDP 10.57.39.33:5060 Require: 100Rel To: <sip:1141@10.57.39.120>;tag=53m9f85odqq1uld83vs6 Contact: <sip:0y9N9S0dFpYtlSrLGwwF@10.57.39.114> From: <sip:6846@10.57.39.120>;tag=032baj647 Supported: 100Rel,precondition RSeq: 2956138 Call-ID: asdbasdb3-asdb552 CSeq: 899 INVITE Allow: UPDATE,PRACK,SUBSCRIBE,REFER,NOTIFY,INVITE,ACK,CANCEL,OPTIONS,BYE Content-Type: application/sdp Content-Length: 465 v=0 o=1141 63380817057795000 63380817057795000 IN IP4 10.57.39.114 s=- c=IN IP4 10.57.39.114 t=0 0 m=audio 5000 RTP/AVP 106 8 0 a=sendrecv a=curr:qos local sendrecv a=curr:qos remote none a=des:qos mandatory local sendrecv a=des:qos mandatory remote sendrecv a=conf:qos remote sendrecv a=rtcp:5001 IN IP4 10.57.39.114 a=rtpmap:106 AMR/8000 a=ptime:160 a=maxptime:200 a=fmtp:106 octet-align=1; mode-set=7 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 UDP Data send to: 10.57.39.120:5060 PRACK sip:1141@10.57.39.120 SIP/2.0 Via: SIP/2.0/UDP 10.57.39.33:5060 From: sip:6846@10.57.39.120;tag=032baj647 To: sip:1141@10.57.39.120;tag=53m9f85odqq1uld83vs6 Call-ID: asdbasdb3-asdb552 CSeq: 89951 PRACK Contact: sip:6846@10.57.39.33 Max-Forwards: 70 Route: <sip:10.57.39.120;lr=on> Content-Type: application/sdp Expires: 180 Accept-Contact: *;+mckoppa Supported: precondition, 100Rel Require: precondition RAck:2956138 899 INVITE Content-Length: 278 v=0 o=Inviter2007 63241204263093750 132223801 IN IP4 10.57.39.33 s=- c=IN IP4 10.57.39.33 t=0 0 m=audio 5000 RTP/AVP 106 a=rtpmap:106 AMR/8000 a=curr:qos local sendrecv a=curr:qos remote sendrecv a=des:qos mandatory local sendrecv a=des:qos mandatory remote sendrecv If added a route rule 'if ( is_method("PRACK") ) { t_relay(); exit;} ' , I'll receive the 483 response. Otherwise, receive 404 Not here UDP Data received from: 10.57.39.120:5060 SIP/2.0 483 Too Many Hops Via: SIP/2.0/UDP 10.57.39.33:5060 From: sip:6846@10.57.39.120;tag=032baj647 To: sip:1141@10.57.39.120;tag=53m9f85odqq1uld83vs6 Call-ID: asdbasdb3-asdb552 CSeq: 89951 PRACK Server: OpenSER (1.3.1-notls (i386/linux)) Content-Length: 0 ________________________________ From: users-bounces(a)lists.openser.org 代表 I?aki Baz Castillo Sent: 2008-6-3 (星期二) 17:26 To: users(a)lists.openser.org Subject: Re: [OpenSER-Users] how to configure to support PRACK El Tuesday 03 June 2008 11:02:39 Steven Wu escribi: > Hi All, > > Could someone tell me how to config openser 1.3.x route to support PRACK? OpenSer doesn't need to do nothing to support PRACK (they are the endpoints who support it or not). PRACK is an in-dialog request, just it. I suppose that the 183 OpenSer forwards to UAC contains a Record-Route added by OpenSer in the INVITE, so the PRACK must contain a Route header and also the To tag received in the 183. Then "has_totag" and "loose_route" return true so a simple t_relay must work. Nothing special must be done, I sure it by experience. Regards. -- Iaki Baz Castillo ibc(a)in.ilimit.es _______________________________________________ Users mailing list Users(a)lists.openser.org http://lists.openser.org/cgi-bin/mailman/listinfo/users

2 1

[OpenSER-Users] Blacklisted Destinations using Database or Text File
by Pablo Bertuzzi 09 Jun '08

09 Jun '08

Hello Everyone, I was using openser for some time as a proxy registrar only with excellent results, now I need to do a more complex application. The scenario is: PSTN--DS3 Connection--CISCO AS5400 ----Openser We have a Cisco AS5400 connected with a DS3 to the PSTN and need to terminate some traffic, however we need to block about 5,000 out of 120,000 possible destinations based in the first 6 digits of destination dialed. For example, destination dialed is 13054941678. First, the leading 1 is removed, then if 305494 is in blacklist some message is sent back to sender (503 Service Unavailable). If 305494 is not in blacklist the call is routed to Cisco AS5400. How do you think I can accomplish this without adding appreciable post-dial-delay and having 1 call per second? Thank you for reading my post. JP

5 6

[OpenSER-Users] pua_bla question
by Steven C. Blair 09 Jun '08

09 Jun '08

Hello: I am considering using OpenSER to implement Bridged Line Appearance (BLA) with Busy Lamp Field (BLF) using Polycom phones. The challenge is that I have other phones which I would like to include in a BLA configuration. Does anyone on this list have documentation showing how the "pua_bla" capability works including methods used, protocol flow, phones supported, etc? Thanks, Steve Senior Network Engineer, Information Systems and Computing Networking and Telecommunications , Suite 221A /6228 University of Pennsylvania Voice:215-573-8396 FAX:215-898-9348

1 0

[OpenSER-Users] Clustering with openser
by Stanisław Pitucha 09 Jun '08

09 Jun '08

Hey I've got some openser servers running in db-only mode on the same database. Load for them is balanced using srv records, and they can handle exactly the same traffic -> standard clustering. I wonder about procedure of shutting them down if something fails. Assuming that a new openser will be started with the same ip on another host if old one fails, should I allow old openser to shutdown as cleanly as it can, or should I cut the communication as soon as possible? What I worry about most, is -> does shutting down openser which handles a dialog at the moment cause it to send any packets that I wouldn't like to be sent in this scenario? I think it works this way: - If I cut the communication and kill all openser processes (kill, not term), I can assign the new ip faster because old host is in unknown/failed state anyways. But it may leave some unused stuff in database. - If I allow it to close properly, it may cleanup it's state it database. But it may remove too much dialog data (which could be used by the backup host). Which one is more likely? Thanks, Stan

3 3

[OpenSER-Users] About t_replicate
by Ali Jawad 09 Jun '08

09 Jun '08

Hi All As per previous Q "mentioned below" I did try using t_replicate however once the client A logons on to server A it just keeps sending subscribe requests to server B Hi All I have the following scenario Openser Server A --> Uses Localhost MySQL Server Openser Server B --> Uses ServerA MySQL server Each server is in a different location. Both servers share the same database. I.e. in other words a user can successfully logon to either server. Both servers have a similar configuration file. My problem at hand is the following User A and User C logon to Server A User B logs on to Server B User A and User C can call each other. How can User A and C communicate with user B ? Can anyone provide me with a working example. My Config ####### Modules Section ######## #set module path mpath="/usr/local/lib/openser/modules/" /* uncomment next line for MySQL DB support */ loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "rr.so" loadmodule "maxfwd.so" loadmodule "usrloc.so" loadmodule "registrar.so" loadmodule "textops.so" loadmodule "mi_fifo.so" loadmodule "uri_db.so" loadmodule "uri.so" loadmodule "xlog.so" loadmodule "auth.so" loadmodule "auth_db.so" # ----- rr params ----- # add value to ;lr param to cope with most of the UAs modparam("rr", "enable_full_lr", 1) # do not append from tag to the RR (no need for this script) modparam("rr", "append_fromtag", 0) # ----- rr params ----- modparam("registrar", "method_filtering", 1) /* uncomment the next line to disable parallel forking via location */ # modparam("registrar", "append_branches", 0) /* uncomment the next line not to allow more than 10 contacts per AOR */ #modparam("registrar", "max_contacts", 10) # ----- uri_db params ----- modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "db_url", "mysql://db_string") # ----- usrloc params ----- modparam("usrloc", "db_mode", 3) modparam("usrloc", "db_url","mysql://db_string") # ----- auth_db params ----- modparam("auth_db", "calculate_ha1", yes) modparam("auth_db", "password_column", "password") modparam("auth_db", "use_domain", 0) modparam("auth_db", "db_url","mysql://db_string") modparam("usrloc", "db_url","mysql://db_string") # ----------------- setting module-specific parameters --------------- # -- mi_fifo params -- modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo_xmpp") # -- usrloc params -- # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) ###### Routing Logic ######## route { if(!mf_process_maxfwd_header("10")){ sl_send_reply("486", "To Many Hops"); return; } if(msg:len >= max_len){ sl_send_reply("513", "Message To Big"); return; } #########End of XMPP Section ################### if(!is_method("REGISTER")){ record_route(); } if(loose_route()){ route(1); return; } if(uri == myself){ if(is_method("REGISTER")){ if(!www_authorize("mydomain.net", "subscriber")){ www_challenge("mydomain.net", "1"); return; } save("location"); if(!src_ip==xx.yy.zz.12) { t_replicate("xx.yy.zz.12", "5060"); }; return; } if(does_uri_exist()){ # local uri does exist, is probably a user. # lookup location if(lookup("location")){ route(1); return; } } else { # probably a call to pstn.... route(2); return; } } } route[1] { # for INVITEs enable some additional helper routes if (is_method("INVITE")) { t_on_branch("2"); t_on_reply("2"); t_on_failure("1"); } if (!t_relay()) { sl_reply_error(); }; exit; } route[2] { # pstn handling, simply route out to pstn. xlog("L_INFO","rewritehostport to Splendor_VOIP_GW:5065"); sl_send_reply("100","Call Going To PSTN"); sethostport("xx.yy.zz.39"); route(1); } /* route[2] { # for INVITEs enable some additional helper routes rewritehostport ("xx.yy.zz.39:5060"); xlog("L_INFO","rewritehostport to Splendor_VOIP_GW:5065"); route(1); } */ # Presence route /* uncomment the whole following route for enabling presence NOTE: do not forget to enable the call of this route from the main route */ branch_route[2] { xlog("new branch at $ru\n"); } onreply_route[2] { xlog("incoming reply\n"); }

2 1

[OpenSER-Users] Dialog module
by David Villasmil 09 Jun '08

09 Jun '08

Hello List, I have a simple load balancer with dispatcher. I'm using the DIALOG module, it is supposed to store in db all info about ongoing dialogs. But my setup must be wrong, because a lot of dialogs are not removed when the call is hung up. Here's my cfg: # # $Id: openser.cfg 1827 2007-03-12 15:22:53Z bogdan_iancu $ # # simple quick-start config script # Please refer to the Core CookBook at http://www.openser.org/dokuwiki/doku.php # for a explanation of possible statements, functions and parameters. # # ----------- global configuration parameters ------------------------ debug=3 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=no # (cmd line: -E) children=4 check_via=no dns=no rev_dns=no listen=a.b.c.d disable_dns_blacklist=true port=5060 #set module path mpath="/lib/openser/modules/" # ------------------ module loading ---------------------------------- loadmodule "mysql.so" loadmodule "maxfwd.so" loadmodule "sl.so" loadmodule "dispatcher.so" loadmodule "tm.so" loadmodule "mi_fifo.so" loadmodule "textops.so" loadmodule "xlog.so" loadmodule "rr.so" loadmodule "dialog.so" loadmodule "avpops.so" modparam("mi_fifo","fifo_name", "/tmp/openser_fifo") modparam("tm", "fr_timer", 5) modparam("dispatcher", "list_file", "/etc/openser/dispatcher.list") modparam("dispatcher", "flags", 2) modparam("dispatcher", "force_dst", 1) modparam("dispatcher", "dst_avp", "$avp(i:271)") modparam("dispatcher", "grp_avp", "$avp(i:272)") modparam("dispatcher", "cnt_avp", "$avp(i:273)") modparam("dialog", "dlg_flag", 4) modparam("dialog", "db_mode", 1) modparam("dialog", "table_name", "dialog") modparam("dialog", "db_url", "mysql://user:pass@localhost/openser") modparam("avpops","avp_url","mysql://user:pass@localhost/openser") modparam("avpops", "avp_table", "dialog") 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"); exit; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); exit; }; if (!method=="REGISTER") record_route(); if (method=="INVITE") { setflag(4); } if ( method=="INVITE" || method=="ACK" || method=="BYE" || method=="OPTIONS" || method=="CANCEL" ) { ds_select_dst("1","2"); t_on_failure("1"); t_relay(); exit; }; } failure_route[1] { if (t_check_status("408")) { xlog("L_INFO","Marking GW as failed...\n"); ds_mark_dst("p"); ds_next_dst(); t_on_failure("1"); t_relay(); } else{ t_reply("501","Not Implemented"); } } As far as I know, using the record_route assures that all messages are passed through the proxy so that dialogs must be removed... but they'r not Any ideas? Thanks a lot. David Villasmil

4 5

[Serusers] stun
by Pezhman Lali 09 Jun '08

09 Jun '08

--- On Mon, 6/9/08, Pezhman Lali <pezhman_lali(a)yahoo.com> wrote: > From: Pezhman Lali <pezhman_lali(a)yahoo.com> > Subject: stun > To: pezhman.lali(a)gmail.com > Date: Monday, June 9, 2008, 6:02 PM > Dear, > does setting the ip of stun server in the sip-phones, > behind the symmetric nat, make the problem ? > my experience with stund 0.96, said yes. > the stun server, can detects the type of nats properly, but > the sip-phones behind the un-symmetric nat, can not > register, or one-way calling. > > ????

1 0

[OpenSER-Users] AVPOPS Issue
by Ali Jawad 09 Jun '08

09 Jun '08

Hi All I have added a column called status to the subscriber table. I want to check that column in the route() function and act according to it's value. In the global settings I put # ----- AVP params -------- modparam("avpops", "avp_url", "mysql://openser:ccccc@cccccc/openser") modparam("avpops","avp_table","subscriber") modparam("avpops","db_scheme", "scheme1:username_col=username;value_col=status;value_type=string;table= subscriber") And In the route statement I put if (avp_db_load("$ru","$avp(s:disabled)/$scheme1")) { xlog("L_INFO", "This did match"); } I am getting this error in the log file Jun 9 09:12:51 [15789] ERROR:avpops:ops_dbload_avps: incomplet uri <sip:xx.xx.xx.xx> "Where xx.xx.xx.xx is my correct IP" And the IF statement is matching everytime. Thanks for any suggestions.

2 9

Re: [OpenSER-Users] Two Openser Servers On DB ..Route Calls
by Daniel-Constantin Mierla 09 Jun '08

09 Jun '08

Hello, On 06/09/08 12:52, David Villasmil wrote: > Isn't it easier to setup ONE mysql server and connect both openser > server to the same db? you would have only one location... > > Just a thought it is a possibility as well. Depending if you want to run with one or two database servers, where you can have also replication options. It is also a possibility to use t_replicate to replicate registrations, but it getx more complex with NAT environments. Cheers, Daniel > > David. > > > > On Mon, Jun 9, 2008 at 10:52 AM, Daniel-Constantin Mierla > <miconda(a)gmail.com <mailto:miconda@gmail.com>> wrote: > > Hello, > > > On 06/03/08 14:20, Ali Jawad wrote: > > > > Hi All > > > > I have the following scenario > > Openser Server A --> Uses Localhost MySQL Server > > Openser Server B --> Uses ServerA MySQL server > > > > Each server is in a different location. Both servers share the same > > database. I.e. in other words a user can successfully logon to > either > > server. Both servers have a similar configuration file. My > problem at > > hand is the following > > > > User A and User C logon to Server A > > User B logs on to Server B > > > > User A and User C can call each other. > > > > How can User A and C communicate with user B ? > > > after you do the lookup on server A and the user is not found there, > then forward the request to server B, do there location lookup as > well, > then if not found, send the negative reply. Otherwise, send it to > user. > > Do same on the server B, forwarding to A... > > Cheers, > Daniel > > > > > > Can anyone provide me with a working example. > > > > > > > > Thanks > > > -- > > http://www.asipto.com > > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > -- http://www.asipto.com

1 0

[OpenSER-Users] Two Openser Servers On DB ..Route Calls
by Ali Jawad 09 Jun '08

09 Jun '08

Hi All I have the following scenario Openser Server A --> Uses Localhost MySQL Server Openser Server B --> Uses ServerA MySQL server Each server is in a different location. Both servers share the same database. I.e. in other words a user can successfully logon to either server. Both servers have a similar configuration file. My problem at hand is the following User A and User C logon to Server A User B logs on to Server B User A and User C can call each other. How can User A and C communicate with user B ? Can anyone provide me with a working example. Thanks

2 1

[OpenSER-Users] OpenSER and MySQL cpu utilization question
by John Barry 09 Jun '08

09 Jun '08

Hello, I am using OpenSER 1.3.0 with digest authentication (MySQL database) in a multiprocessor machine (4 CPU). When using OpenSER configured with one child. I have noticed that OpenSER process can reach a maximum CPU utilization of 70% while the MySQL DB reaches around 30% CPU utilization (100% in total). Tipically the OpenSER process and the MySQL process will be using the same processor. However, this behavior happens also if I forced each processor to use different processors (using processor affinity). My question is why the OpenSER and the DB processes can't use more CPU time if they are running in independent processors (they behave as if they were running in a single processor). What is the dependency/relationship between the OpenSER and MySQL processes? Thanks, JB _________________________________________________________________ Express yourself instantly with MSN Messenger! Download today it's FREE! http://messenger.msn.click-url.com/go/onm00200471ave/direct/01/

2 1

[OpenSER-Users] serMyAdmin on www.sermyadmin.org
by Zen Kato 09 Jun '08

09 Jun '08

I upgraded from tomcat5.5.15 to tomcat6.0.16 on fc5. After I entered username: admin(a)192.168.0.3 Password:xxxx on the screen of http://192.168.0.3:8080/serMyAdmin/auth/login, [Login] click doesn't work. Only [Register] click goes to http://192.168.0.3:8080/serMyAdmin/newUser/create Is this a correct behavior of serMyAdmin? I followed up inside of openser.sql of Step 17: Create the administrator user in the database. on Last Updated ( Friday, 02 May 2008 16:55 ) web infomation, but I don't know the reason of the following procedure(1),(2). (1) ALTER TABLE `domain` ALTER `last_modified` SET DEFAULT '2008-01-01 00:00:00'; UPDATE domain set `last_modified`='2008-01-01 00:00:00'; INSERT INTO `domain` (`id`, `domain`, `last_modified`, `version`) VALUES (2, 'setup', '2008-04-09 16:13:12', 0); (2) INSERT INTO `jsec_user_role_rel` (`id`, `version`, `role_id`, `user_id`) VALUES (1, 0, 1, 4), (3, 0, 1, 7), (4, 0, 1, 3), (5, 0, 3, 11); Do I miss somthing to run serMyAdmin? -- Zen

1 0

[OpenSER-Users] mirror proxy/upstream Registrar
by sylver_b 09 Jun '08

09 Jun '08

Hi Thanks for your reply. Yes in fact, the registrar is some sort of a billing application that enforce AAA. The only problem is , it does not handle NAT traversal. That's why we need OpenSer to act as an SBC. El Domingo, 8 de Junio de 2008, sylver_b escribió: > Hello, > > We would like to evaluate OpenSER to test interoperability with our > sip registrar platform - as our registrar does not handle NAT Traversal > we would like to use OpenSER as a mirror proxy . Basically we would like > OpenSER to act as an Upper Registrar proxy where all the sip messages would > be forwarded to our registrar, based on the domain name used in the > Register. ie: REGISTER 111 at domain1.com -> forward to -> registrar_ip1 > REGISTER 222 at domain2.com -> forward to -> registrar_ip2 > and so on... Take a look to "t_replicate" function of "tm" module. > If a call is initiated, OpenSER should transparently handle NAT traversal > functionalities while forwarding the INVITE request to the proper registrar > server. Do you mean that an INVITE should be forwarded to a registrar server? maybe the registrar is also a proxy/PBX? Please, explain a little more your escenario. Regards. -- Iñaki Baz Castillo _____________________________________________________________________________ Envoyez avec Yahoo! Mail. Une boite mail plus intelligente http://mail.yahoo.fr

1 0

[OpenSER-Users] mirror proxy/upstream Registrar
by sylver_b 08 Jun '08

08 Jun '08

Hello, We would like to evaluate OpenSER to test interoperability with our sip registrar platform - as our registrar does not handle NAT Traversal we would like to use OpenSER as a mirror proxy . Basically we would like OpenSER to act as an Upper Registrar proxy where all the sip messages would be forwarded to our registrar, based on the domain name used in the Register. ie: REGISTER 111(a)domain1.com -> forward to -> registrar_ip1 REGISTER 222(a)domain2.com -> forward to -> registrar_ip2 and so on... If a call is initiated, OpenSER should transparently handle NAT traversal functionalities while forwarding the INVITE request to the proper registrar server. Below some detailed requirements: Upper Registration is the capability of a SBC to proxy Registrations towards an upstream Registrar. While it is necessary that all SIP requests traverse the SBC for NAT continuity, Registrations should be allowed to be relayed towards the upstream Registrar while the SBC retains a copy of the AOR and masquerade on behalf of the UA that send the registration request. Please let us know the best way to configure OpenSER to achieve this type of configuration. Thanks SB _____________________________________________________________________________ Envoyez avec Yahoo! Mail. Une boite mail plus intelligente http://mail.yahoo.fr

2 1

[OpenSER-Users] newbie question on NAT
by mark morreny 08 Jun '08

08 Jun '08

Hi, I am working on configuring RTP Proxy. The problem I am having is that Openser will use rtp proxy to serve the rtp session even when both UAs and Openser are withing the same LAN. Here is the openser.cfg I used: if (nat_uac_test("19")) { if (method=="REGISTER") { fix_nated_register(); } else { fix_nated_contact(); }; setflag(5); }; Hre is a INVITE SIP packet from one of the UA: U 192.168.1.101:5062 -> 192.168.1.104:5060 INVITE sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104> SIP/2.0..Via: SIP/2.0/UDP 192.168.1.101:5062;branch=z9hG4b K-220120f9..From: "1005" <sip:1005@192.168.1.104<sip%3A1005(a)192.168.1.104>>;tag=490c5d623d5acbf5o0..To: "1006" <si p:1006@192.168.1.104 <p%3A1006(a)192.168.1.104>>..Call-ID: 3e5720e1-bf774681@192.168.1.101..CSeq: 102 INVITE..Max-F orwards: 70..Proxy-Authorization: Digest username="1005",realm=" 192.168.1.104",nonce="48 4b3b51ea052f45482ef4fd26f00a14239c3465",uri="sip:1006@192.168.1.104<sip%3A1006(a)192.168.1.104> ",algorithm=MD5,respo nse="d46ea7cd9781b9a1493e6eaf7b978086",qop=auth,nc=00000001,cnonce="676aa6d8"..Contact: "1005" <sip:1005@192.168.1.101:5062>..Expires: 240..User-Agent: Sipura/SPA921-4.1.10(b). .Content-Length: 401..Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER..Con tent-Type: application/sdp....v=0..o=- 26093424 26093424 IN IP4 192.168.1.101..s=-..c=IN IP4 192.168.1.101..t=0 0..m=audio 16474 RTP/AVP 0 2 4 8 18 96 97 98 101..a=rtpmap:0 PCM U/8000..a=rtpmap:2 G726-32/8000..a=rtpmap:4 G723/8000..a=rtpmap:8 PCMA/8000..a=rtpmap:18 G729a/8000..a=rtpmap:96 G726-40/8000..a=rtpmap:97 G726-24/8000..a=rtpmap:98 G726-16/800 0..a=rtpmap:101 telephone-event/8000..a=fmtp:101 0-15..a=ptime:30..a=sendrecv.. # I even tested with nat_uac_test("1") and nat_uac_test("2") and nat_uac_test("16") seperately and found that nat_uac_test("1") is return true, meaning that Openser detects a RFC1918 address in the Contact field. If Openser is in the same LAN as the UA, then no rtp proxy should be utilized. Could someone please help me out to configure rtp proxy correctly? I would appreciate any help in solving this problem. Regards, Mark

2 4

Re: [OpenSER-Users] Dialog table "state" value
by Stanisław Pitucha 08 Jun '08

08 Jun '08

----- "David Villasmil" <david.villasmil.work(a)gmail.com> napisał: > Although I've looked and looked for any info on the different values > of the "state" column of the dialog table, I can't find it anywhere, > What does it mean? >From branches/1.3/modules/dialog/dlg_hash.h: #define DLG_STATE_UNCONFIRMED 1 #define DLG_STATE_EARLY 2 #define DLG_STATE_CONFIRMED_NA 3 #define DLG_STATE_CONFIRMED 4 #define DLG_STATE_DELETED 5 Unconfirmed means that only INVITE was sent. Early - after any 1XX response Confirmed_na - after 2XX Confirmed - after ACK response after errors (or BYE) state switches to deleted. At least that's what I understood from source and observing database... Stan

1 0

[OpenSER-Users] CDRTool: How to avoid invoices for answered calls of duration 0
by Adrian Georgescu 07 Jun '08

07 Jun '08

If the call has 0 seconds duration, CDRTool does not calculate a cost, it is zero. The connection fee applies only if the call has a positive duration. Adrian >>>>>>>>>>>> Hi, RFC3261 says that a UAC should send inmediately a BYE after a 2XX response if the offer in the 2xx response is not acceptable (when the INVITE has no body). So this will generate an answered call of duration 0 (probably). But if the CDRTool rate has "connect" fixed amount then this invalid call will have cost. I think this shouldn't have cost since it's a SIP protocol issue (accepting a call and sending a BYE inmediately) but not an human decision. Opinions?

2 1

[OpenSER-Users] Dialog table "state" value
by David Villasmil 07 Jun '08

07 Jun '08

Hello all, Although I've looked and looked for any info on the different values of the "state" column of the dialog table, I can't find it anywhere, What does it mean? Thanks David

1 0

[OpenSER-Users] storing extra info in Dialog table
by David Villasmil 06 Jun '08

06 Jun '08

Hello all Is it possible to store extra info in the dialog table? For instance, I want to add a column, named time_in, and I want the dialog module to add to the fields inserted a "now()" Is this possible? Thanks David

2 1

[OpenSER-Users] AVP Setup
by Ali Jawad 06 Jun '08

06 Jun '08

Dear All I am trying to do the following: I want to enable/disable users, to do that I added a column named status to the subscribers table. It contains a string saying enabled or disabled. I did add the following info to the config file. # ----- AVP params -------- modparam("avpops", "avp_url", "mysql://openser:0o0o0o0o0@xx.xx.xx.xx/openser") modparam("avpops","avp_table","subscriber") modparam("avpops","db_scheme", "scheme1:username_col=username;value_col=status;value_type=string;table= subscriber") And in the route statement I added "This is a sample provided by MPWizard..thanks for that" however it apparently needs some changes to work for me. if (avp_db_load("$tU", "$avp(s:disabled)")) { xlog("L_INFO", "User is local but disabled, denying registration") } The error I get in the log file is as follows: Jun 6 11:20:38 [21919] ERROR:mysql:db_mysql_submit_query: driver error: Unknown column 'value' in 'field list' Jun 6 11:20:38 [21919] ERROR:mysql:db_mysql_query: error while submitting query Jun 6 11:20:38 [21919] ERROR:avpops:ops_dbload_avps: db_load failed Any suggestions please. Thanks to all of you guys. For all the support now and before ..you rock.

3 2

Re: [OpenSER-Users] Openser Multi-home
by Bogdan-Andrei Iancu 06 Jun '08

06 Jun '08

Note that there are two way to configure openser in multi-home envs. : - either enable mhome param and openser will autodetect the outbound interface based on the destination - quite slow - either use force_send_socket() function if the msg flow is clear and you know from the script what interface should be used - very fast Regards, Bogdan jvelev(a)avangardsolutions.com wrote: > That fixed the problem. 10x > > Regards, > J.Velev > > On Thu, 2008-06-05 at 20:11 +0300, Bogdan-Andrei Iancu wrote: > >> Hi, >> >> have you enabled the mhomed param? >> http://www.openser.org/dokuwiki/doku.php/core-cookbook:1.3.x#mhomed >> >> Regards, >> Bogdan >> >> jvelev(a)avangardsolutions.com wrote: >> >>> Hello all >>> >>> I'm trying to run OpenSER on a multi-home environment. OpenSER listens >>> on the real IP and on the private one. The main idea is to proxy traffic >>> coming from real IPs to the internal network. >>> >>> OpenSER tries to reach the internal IPs talking to them with it's REAL >>> IP despite it's directly connected to the internal network and can use >>> it's private IP. >>> >>> Comments & Ideas appreciated. >>> >>> >>> >>> >>> >>> >> > > >

1 0

[OpenSER-Users] CDRTool: How to avoid invoices for answered calls of duration 0
by Iñaki Baz Castillo 06 Jun '08

06 Jun '08

Hi, RFC3261 says that a UAC should send inmediately a BYE after a 2XX response if the offer in the 2xx response is not acceptable (when the INVITE has no body). So this will generate an answered call of duration 0 (probably). But if the CDRTool rate has "connect" fixed amount then this invalid call will have cost. I think this shouldn't have cost since it's a SIP protocol issue (accepting a call and sending a BYE inmediately) but not an human decision. Opinions? 13.2.2.4 2xx Responses -- Iñaki Baz Castillo ibc(a)in.ilimit.es

1 0

Re: [OpenSER-Users] [OpenSER-Devel] Memory dump for OpenSER 1.3.1 which slowly leaks memory
by Julian Yap 06 Jun '08

06 Jun '08

Bogdan, I'll need to check out the shmem statistics. I mean, this is the memory output for the OpenSER processes having run for about 24 hours. Does it look right?: # ps -C openser -o vsz,rss VSZ RSS 234124 12636 369940 159392 369940 158840 369940 158860 370340 159220 369804 158696 370068 158932 370204 159048 370464 159344 234124 20904 234124 9484 234128 7952 234124 7948 234124 7880 234124 7880 234124 7880 234124 7880 234124 7880 234124 7880 234124 7880 234124 8332 Here are the modules I'm using. Not sure if any have a dependency on libxml: loadmodule "mysql.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "usrloc.so" loadmodule "auth.so" loadmodule "auth_db.so" loadmodule "maxfwd.so" loadmodule "mi_fifo.so" loadmodule "nathelper.so" loadmodule "perl.so" loadmodule "registrar.so" loadmodule "rr.so" loadmodule "textops.so" loadmodule "uri.so" Thanks, Julian On Wed, May 7, 2008 at 5:22 AM, Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> wrote: > Hi Julian, > > Have you tried to use the the shmem statistics to monitor the usage of SHM > memory? > > Both for PKG and SHM memory, openser allocs at startup the whole memory > (depending of how much you configured). Any leak will not be vizible for any > external tools. > > Also, are you using any module that have dependency of libxml library? > > The dump you posted shows more or less nothing more than init phase - > before sending the signal, be sure you select a worker process (use fifo > command "ps" to get the list and types of processes). > > Regards, > Bogdan > > Julian Yap wrote: > > > > > > > > > On Mon, May 5, 2008 at 10:25 PM, Henning Westerholt > > <henning.westerholt(a)1und1.de> wrote: > > > > > > > On Saturday 03 May 2008, Julian Yap wrote: > > > > I'm currently running OpenSER 1.3.1. > > > > > > > > It seems to slowly leaks memory. > > > > > > > > I've recompiled it as per for debug support in the memory manager: > > > > http://www.openser.org/dokuwiki/doku.php/troubleshooting:memory > > > > > > > > Here is the memory manager dump of the memory status: > > > > http://julianyap.com/download/openser1.3.1_memory_dump20080502.txt > > > > > > > > Any suggestions would be very welcome. > > > > > > Hi Julian, > > > > > > can you give a little more details about your problem? Why do you think > it > > > leaks memory? Do you get memory allocation errors in the logs? > > > > > > Cheers, > > > > > > Henning > > > > > > > > > > Henning, > > > > I don't get any memory allocation errors. > > > > I've recently updated to OpenSER 1.3 branch SVN r4107 but the 'leak' > > still remains. > > > > On a 4GB box, free memory just seems to deplete. > > > > I have a cron job which restarts OpenSER every night but it seems like > > not all the memory is free'd up. Eventually the box needs to be > > rebooted. In the past, running out of memory has led to oom-killer > > killing OpenSER. > > > > For example, I have this cron job: > > */5 * * * * root echo "`date` `free | grep 'buffers/cache'`" >> > > /var/log/free_memory.log > > > > Memory just depletes. Here's where OpenSER was restarted around midnight: > > Sun May 4 23:10:01 HST 2008 -/+ buffers/cache: 2014972 2095204 > > Sun May 4 23:15:01 HST 2008 -/+ buffers/cache: 2027184 2082992 > > Sun May 4 23:20:01 HST 2008 -/+ buffers/cache: 2027604 2082572 > > Sun May 4 23:25:01 HST 2008 -/+ buffers/cache: 2038064 2072112 > > Sun May 4 23:30:01 HST 2008 -/+ buffers/cache: 2037868 2072308 > > Sun May 4 23:35:01 HST 2008 -/+ buffers/cache: 2053184 2056992 > > Sun May 4 23:40:01 HST 2008 -/+ buffers/cache: 2055216 2054960 > > Sun May 4 23:45:01 HST 2008 -/+ buffers/cache: 2075612 2034564 > > Sun May 4 23:50:01 HST 2008 -/+ buffers/cache: 2103232 2006944 > > Sun May 4 23:55:01 HST 2008 -/+ buffers/cache: 2114384 1995792 > > Mon May 5 00:00:01 HST 2008 -/+ buffers/cache: 2117320 1992856 > > Mon May 5 00:05:01 HST 2008 -/+ buffers/cache: 860304 3249872 > > Mon May 5 00:10:01 HST 2008 -/+ buffers/cache: 861208 3248968 > > Mon May 5 00:15:01 HST 2008 -/+ buffers/cache: 868968 3241208 > > > > Here's where I just restarted OpenSER: > > Mon May 5 23:30:01 HST 2008 -/+ buffers/cache: 2041740 2068436 > > Mon May 5 23:35:01 HST 2008 -/+ buffers/cache: 2049572 2060604 > > Mon May 5 23:40:01 HST 2008 -/+ buffers/cache: 2054856 2055320 > > Mon May 5 23:45:01 HST 2008 -/+ buffers/cache: 2065316 2044860 > > Mon May 5 23:50:01 HST 2008 -/+ buffers/cache: 2062248 2047928 > > Mon May 5 23:55:01 HST 2008 -/+ buffers/cache: 2076148 2034028 > > Tue May 6 00:00:01 HST 2008 -/+ buffers/cache: 2087256 2022920 > > Tue May 6 00:05:01 HST 2008 -/+ buffers/cache: 2091300 2018876 > > Tue May 6 00:10:01 HST 2008 -/+ buffers/cache: 2093244 2016932 > > Tue May 6 00:15:01 HST 2008 -/+ buffers/cache: 1382632 2727544 > > > > Thanks, > > Julian > > > > _______________________________________________ > > Devel mailing list > > Devel(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/devel > > > > > > > >

4 6

Re: [OpenSER-Users] Terminate a call from script
by Bogdan-Andrei Iancu 06 Jun '08

06 Jun '08

Hi David, But this is not from script. Is is from outside openser. Regards, Bogdan David Villasmil wrote: > I've done that... > > Using the dialog module, and getting the hash_id and hash_entry, you can: > > :dlg_end_dlg:_reply_fifo_file_ > 342 > 56 > _empty_line_ > > and that ends the call. > > On Thu, Jun 5, 2008 at 3:17 PM, Bogdan-Andrei Iancu > <bogdan(a)voice-system.ro <mailto:bogdan@voice-system.ro>> wrote: > > Hi, > > No, there is no way to terminate the call directly from the script > - at > least not so far :). > > Regards, > Bogdan > > kionez wrote: > > Hi all, > > I'm trying to add a simple priority mechanism in my openser 1.3 > script. > > > > Reading docs i found that dialog module gives a MI command to > terminate > > a call, dlg_end_dlg. Is there any way to terminate a call within a > > script (or a perl script)? I can't find anything on the web, the > only > > way i found is to exec something like "openserctl fifo > dlg_end_dlg $hash > > $id", but it's slower than a direct function inside the main > script.. > > > > thanks in advance > > > > k. > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org <mailto:Users@lists.openser.org> > http://lists.openser.org/cgi-bin/mailman/listinfo/users > >

2 2

Re: [OpenSER-Users] Openser-Perl module in onreply_route
by Mik Cheez 06 Jun '08

06 Jun '08

The 'perl' module has been a godsend for me, but the issue I'm having still has to do with the memory leak in the perl DBI module; my question is, has anyone considered using the 'perl' framework for other scripting languages? Python and PHP would be terrific, in my opinion. As I have things now, I can lose up to 1.5MB/minute using the DBI module (during heavy usage). This is in reference to a posting made back in October 2007: http://www.mail-archive.com/users@openser.org/msg13387.html Best regards, Mik

2 1

[OpenSER-Users] New OpenSER-mi-proxy release 1.0.0
by Adrian Georgescu 06 Jun '08

06 Jun '08

Hello, There is a new component useful for implementing the presence server solution using OpenSER and OpenXCAP. OpenSER MI Proxy ---------------- OpenSER MI Proxy is designed to allow remote XML-RPC clients to issue MI commands to OpenSER. This proxy replaces the need for OpenSER built-in xml-rpc server, which has no access list and is prone to crash the server. One of its purposes is to allow a remote OpenXCAP server to safely call the refreshWatchers() command on the management interface of the SIP Presence server when the presence policy document has changed. The proxy can listen on custom IP addresses and provides an access list for the clients allowed to connect. The software can be downloaded from: http://download.ag-projects.com/OpenXCAP/ Kind regards, Adrian Georgescu

4 7

[OpenSER-Users] how to reload openser.cfg
by Qi Fan 06 Jun '08

06 Jun '08

Hello OperSER users, This is my first post here. ;-) Can anyone tell me know how to load the modified openser.cfg on the fly? Is there any command like "extensions reload" in Asterisk? Many thanks for your help. Regards, Qi Fan

4 7

[OpenSER-Users] Openser Multi-home
by jvelev＠avangardsolutions.com 05 Jun '08

05 Jun '08

Hello all I'm trying to run OpenSER on a multi-home environment. OpenSER listens on the real IP and on the private one. The main idea is to proxy traffic coming from real IPs to the internal network. OpenSER tries to reach the internal IPs talking to them with it's REAL IP despite it's directly connected to the internal network and can use it's private IP. Comments & Ideas appreciated. -- This message has been scanned for viruses and dangerous content by MailScanner, and is believed to be clean.

2 1

[OpenSER-Users] OT: TLS mandatory for proxies?
by Klaus Darilion 05 Jun '08

05 Jun '08

Hi! The TLS docs say that TLS is mandatory for SIP proxies. I tried to find the relevant text in the RFC but could not find it. Does somebody knows if it is really mandatory and where it is described? thanks klaus

2 2

[OpenSER-Users] Doesn't "proxy_authorize()" allow credentials if "From" == "Anonymous" <sip:anonymous@anonymous.invalid> ?
by Iñaki Baz Castillo 05 Jun '08

05 Jun '08

Hi, when I do a call with hiddn callerid from my Twinkle it sends an INVITE with: ----------- INVITE sip:1414@my_domain.net SIP/2.0 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=emirn Privacy: id P-Preferred-Identity: "IBC" <sip:ibc@my_domain.net> ----------- and of course my Openser asks Twinkle for authentication: ----------- SIP/2.0 407 Proxy Authentication Required Proxy-Authenticate: Digest realm="my_domain.net", nonce="4838005d57e5ee0a1f96430d63696fe1407ef93e", qop="auth" ----------- and I see the $rc code of "proxy-authentication()" is -4 (no credentials). So my Twinkle sends a correct INVITE (CSeq + 1 and "Proxy-Authorization" header): ----------- INVITE sip:1414@my_domain.net SIP/2.0 From: "Anonymous" <sip:anonymous@anonymous.invalid>;tag=emirn Privacy: id P-Preferred-Identity: "IBC" <sip:ibc@my_domain.net> Proxy-Authorization: Digest username="ibc", realm="mt_domain.net", nonce="4838005d57e5ee0a1f96430d63696fe1407ef93e", uri="sip:1414@my_domain.net", response="061d82967778669a01e53d168f2da93b", algorithm=MD5, cnonce="9n7ln49l46", qop=auth, nc=00000001 ----------- But "proxy-authentication()" replies again with -4 (no credentials). ----------- SIP/2.0 407 Proxy Authentication Required Proxy-Authenticate: Digest realm="my_domain.net", nonce="4838005d57e5ee0a1f96430d63696fe1407ef93e", qop="auth" ----------- Why? Obviously the INVITE **does include** Proxy-Authorization. Maybe RC3261 says that a server must not challenge for authentication a user with From "Anonymous" <sip:anonymous@anonymous.invalid> ? Of course authentication works properly when not hidding callerid. Thanks for any explanation. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

3 6

Re: [OpenSER-Users] t_replicate function`s doubt??
by Marcio 05 Jun '08

05 Jun '08

Hi Zahid, Thanks for help, maybe you can help me....I have an environment of test with more than 2 proxies....A UAc of a domain visiting another domain.....wanted him to succeed in registering domain in place and had home domain pointing to the local domain(visited), the case reached a "request" to this UAc....would be a distributed a sketch of records that follow where the UAc is. The idea to use t_replicate() function, i did think that could resolve this question, but now i think not.....because the home proxy of the UAc not know the location of your UAc, would have to be informed at the time of replication, such as by the validation credentials into table "subscriber", could insert the location into table "location"....??? It would have some idea that could help me? best regards, Marcio 2008/6/5, Zahid Mehmood <zm23(a)columbia.edu>: > > Marcio, > the way we use t_replicate is as follows. > > UAC -- register ---> proxy1 > > proxy1 code > > { > > if (src_ip == ip_of_proxy_2 ) { > # do not challenge because we assume that proxy2 did the > authenticate already > # save location and any other thing you want to do > } else { > # do challenge / authenticate here > # save location > # t_replicate to proxy2 > } > > proxy2 code > > { > > if (src_ip == ip_of_proxy_1 ) { > # do not challenge because we assume that proxy2 did the > authenticate already > # save location and any other thing you want to do > } else { > # do challenge / authenticate here > # save location > # t_replicate to proxy2 > } > > > Hope this helps. > > -- > Zahid > > > > On Jun 4, 2008, at 4:10 PM, Marcio wrote: > > Hi, >> >> Someone would know if t_replicate() function....perform a t_relay() of the >> REGISTER....and carry the "credentials" of the UAc to slave or remote UAs? >> >> And the reply of the UAs(remote or slvae) is "200, ok" to UAs (fork`s >> origin) or not a reply?? >> >> best regards, >> Marcio >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users >> > >

1 0

[OpenSER-Users] smart stund
by Pezhman Lali 05 Jun '08

05 Jun '08

Dear, some sip-phones behind the symmetric nat, can not be registered, if the ip of stun server was filled ,in the setting of sip-phone. I am looking for a way,that all of sip-phones, with the same setting , work properly. may be a solution, the stun server doesn't reply to the request behind the symmetric request, what is your idea? best Mani

2 1

[OpenSER-Users] 6xx handling - new options
by Bogdan-Andrei Iancu 05 Jun '08

05 Jun '08

Hi, Following a several discussions on devel + users and sip-implementers mailing lists, I just added a new module parameter in TM to control how the 6xx replies are handled: -(0) the 6xx replies will block any further serial forking (adding new branches). This is the RFC3261 behaviour. -(1) the 6xx replies will be handled as any other negative reply - serial forking will be allowed. Logically, you need to break RFC3261 if you want to do redirects to announcement and voicemail services. Again, this is more a logical debate (about the 6xx replies).See: https://lists.cs.columbia.edu/pipermail/sip-implementors/2008-June/019413.h… and this new param was added to let you choose between: - following RFC, but not being able to implement some scenarios - breaking RFC, but being more practical ;) Best regards, Bogdan

1 0

[OpenSER-Users] Accept-Contact not in NOTIFY request ?
by KevinKinnan 05 Jun '08

05 Jun '08

I just modified the relevnt source files such as notify.c and subscribe.c. It works now :) Kevin _________________________________________________________________ 用手机MSN聊天写邮件看空间，无限沟通，分享精彩！ http://mobile.msn.com.cn/

1 0

[OpenSER-Users] Terminate a call from script
by kionez 05 Jun '08

05 Jun '08

Hi all, I'm trying to add a simple priority mechanism in my openser 1.3 script. Reading docs i found that dialog module gives a MI command to terminate a call, dlg_end_dlg. Is there any way to terminate a call within a script (or a perl script)? I can't find anything on the web, the only way i found is to exec something like "openserctl fifo dlg_end_dlg $hash $id", but it's slower than a direct function inside the main script.. thanks in advance k.

2 1

Re: [OpenSER-Users] srv load-balancing
by Stanisław Pitucha 05 Jun '08

05 Jun '08

----- "Sajith T S" <sajith(a)gmail.com> wrote: > stanis(a)zimbra-1.gradwell.net wrote: > > > and got 6 calls on asterisk-2 and none on asterisk-1. If I change > > the order, all calls go to the first host on the list. Shouldn't > > openser do load-balancing in this scenario? I don't use any options > > in t_relay(). > > You should use openser dispatcher module for this. Yes... I've just noticed in http://www.openser.org/pipermail/users/2007-October/013660.html that openser basically decided to go against SRV RFC and ignores weights. (RFC says that clients should sort and select from list of servers, not servers as B.A.I. suggested if I understood that correctly) Anyways - about dispatcher: I couldn't find any description about it's behavior with call transfers / taking over. I understand that while I'm using callid-based hash there will be no problems with REFER, because call-id stays the same and each leg will go to the same server. But what about INVITE + Replaces: ? I don't really see a way for it to work just with hashing. Has anyone tested this / has solution? PS. sorry Sajith, I replied to you before by mistake Thanks, Stan

3 2

[OpenSER-Users] openser 1.3.0 config
by Mark Quitoriano 05 Jun '08

05 Jun '08

HI All, Im new to openser and trying to config it following this howto http://www.openser.org/dokuwiki/doku.php/install:openser-from-svn i get errors when running the database setup script. # /usr/local/sbin/openserdbctl create ERROR: could not load the script in /usr/local/lib64/openser/openserctl/openserdbctl.mysql for database engine MYSQL ERROR: database engine not loaded - tried 'MYSQL' I'm usng Centos 5 btw.

2 2

[OpenSER-Users] need help on Free Radius & CDR Tool
by Do Nguyen Ha 05 Jun '08

05 Jun '08

hi i manage the Cisco BTS 10200 system, and now i am assigned a task to work with the Radius i choose the FreeRadius to work with Cisco BTS system the attached file are message flow & message's structure of account start modify some config in radiusd.conf & clients.conf first, BTS send message Access Request to FreeRadius and then Freeradius send message Access Accept, they working properly when BTS send message Account Start to Radius, there are a lot attribute and freeradius save them in the radacct table but in radacct , there some columm doesn't match with the message Account Start, i want to change the name of some columm to match the message Should i need to edit the sql.conf for this issue On the CDR Tool, should i need to edit to working properly Thanks Ha`

1 0

[OpenSER-Users] need help on Free Radius & CDR Tool
by Do Nguyen Ha 05 Jun '08

05 Jun '08

hi i manage the Cisco BTS 10200 system, and now i am assigned a task to work with the Radius i choose the FreeRadius to work with Cisco BTS system AReq : Radius Access-Request AAcc: Radius Access-Accept AcctStart: Radius Accounting-Start AcctStop: Radius Accounting-Stop AcctRes: Radius Accounting-Response modify some config in radiusd.conf & clients.conf first, BTS send message Access Request to FreeRadius and then Freeradius send message Access Accept, they working properly when BTS send message Account Start to Radius, there are a lot attribute and freeradius save them in the radacct table but in radacct , there some columm doesn't match with the message Account Start, i want to change the name of some columm to match the message this is the account start message structure: Should i need to edit the sql.conf for this issue On the CDR Tool, should i need to edit to working properly Thanks Ha`

1 0

[OpenSER-Users] Openser working with REFER (Transfer)
by Thiago Maluf 05 Jun '08

05 Jun '08

Hi List, I am research in Brazil University and in our system we have several User Agents in SIP. We have the openser working as Sip Proxy and Asterisk or Cisco Routers as Gateway with PSTN or PBX. We have too Hardphone Cisco IP Phone and Polycom IP Pphone and Softphone X-Lite, Eyebeam and Ekiga. Just now our system is working to all users, but I found one problem. When the user, work with BlindTransfer* the call is transfered no problem. But when the user (no matter if is Hardphone or Softphone) try transfer** the call, it don't work. In my OpenSER, when I receive one REFER without LOOSE_ROUTE I just end the request. While if this request have LOOSE_ROUTE I send this request to route defined in dialog. I would like if you can I help me, if I can insert one logical in OpenSER to do it work. is it possible? Very thanks for everybody, Thiago! (!) * BLINDTRANSFER: UA1 call to UA2 and when the call is established, UA1 for example send one request REFER to UA2 to this user call UA3. Then UA2 hangup call with UA1, and make one call to UA3. ** TRANSFER: UA1 call to UA2 and when the call is established, UA1 for example put this call in hold on, and make one call to UA3. When this last call is established, UA1 send one request REFER to UA2 to get the call with UA3. Then UA2 will talk with UA3, and UA1 is hangup its calls. -- ---------------------------------------------------------------- THIAGO MALUF RESENDE Consultor Voip e Programador WEB (Voip Developer and Web Developer) Tel: +55 21 86042100 e-mail: malufrj(a)gmail.com

2 1

[OpenSER-Users] t_replicate function`s doubt??
by Marcio 04 Jun '08

04 Jun '08

Hi, Someone would know if t_replicate() function....perform a t_relay() of the REGISTER....and carry the "credentials" of the UAc to slave or remote UAs? And the reply of the UAs(remote or slvae) is "200, ok" to UAs (fork`s origin) or not a reply?? best regards, Marcio

1 0

[Serusers] SER 0.10.x AVP syntax documentation
by Steven C. Blair 04 Jun '08

04 Jun '08

I'm trying to convert my SER 0.9.x config file to a 0.10.x file. I need some help converting AVPs. I use to use rules such as " if (avp_db_load("$ruri/username", "s:callfwdall"))..." however this now generates a syntax error. Is there a how-to which describes how to convert these types of statements to the new format? Thanks,steve Senior Network Engineer, Information Systems and Computing Networking and Telecommunications , Suite 221A /6228 University of Pennsylvania Voice:215-573-8396 FAX:215-898-9348

2 1

[Serusers] SER 0.10 AVPs
by Steven C. Blair 04 Jun '08

04 Jun '08

Hello: I saw your post on the iptel.org web site about SER 0.10 and AVPs. I'm trying to convert a SER 0.9.x system to SER 0.10.x and the AVPOPS are giving me the most trouble. Did you find any documentation on how to implement the new avpops to do MySQL queries ? Thanks,Steve Senior Network Engineer, Information Systems and Computing Networking and Telecommunications , Suite 221A /6228 University of Pennsylvania Voice:215-573-8396 FAX:215-898-9348

2 3

[OpenSER-Users] Accept-Contact not in NOTIFY request ?
by KevinKinnan 04 Jun '08

04 Jun '08

I am using N95 to test SUBSCRIBE request with openser that dedicates as presence server. According to nokia notes, in nokia java sip stack, the accept-contact header is mandatory in all requests. Any request without this header will not be recognized by sip stack in N95. The NOTIFY request from openser seems not add this header in itself. Or maybe I omit some configurations. Please give me some clue, thank you. Regards, Kevin _________________________________________________________________ 新年换新颜，快来妆扮自己的MSN给心仪的TA一个惊喜！ http://im.live.cn/emoticons/?ID=18

4 3

[Serusers] Gentilini, Paul is out of the office.
by PGentilini＠intellicare.com 04 Jun '08

04 Jun '08

I will be out of the office starting Mon 06/02/2008 and will not return until Mon 06/09/2008. I will have very limited access to mail, so please contact the CSC at 888-899-4227 if you need immediate assistance.

1 0

[OpenSER-Users] Accept-Contact not in NOTIFY request ?
by KevinKinnan 04 Jun '08

04 Jun '08

It is not the matter of rfc3261, the nokia sip stack force any request adding this header. Without it, the test will not be going on. Regards, Kevin _________________________________________________________________ 新年换新颜，快来妆扮自己的MSN给心仪的TA一个惊喜！ http://im.live.cn/emoticons/?ID=18

1 0

Re: [OpenSER-Users] [OpenSER-Devel] OpenSER usrloc timeouts
by Henning Westerholt 04 Jun '08

04 Jun '08

On Wednesday 04 June 2008, Jason Penton wrote: > Hi Henning > > would you mind elaborating a little? Hi Jason, please always CC to the user list. First of all you need to define a failure_route in your config. After the lookup() to your location database you arm this with t_on_failure. In the failure route you check with t_check_status("408") the error code and forward then to your voicemail. Cheers, Henning

2 1

Re: [OpenSER-Users] [OpenSER-Devel] OpenSER usrloc timeouts
by Henning Westerholt 04 Jun '08

04 Jun '08

On Wednesday 04 June 2008, Jason Penton wrote: > yes, but maybe instead of deregistering the user, it could foeard to > voicemail or similar instead of the callee just hearing 'nothing' and not > knowing what is going on????? (moved to the user list) Hi Jason, normally you just setup a prober INVITE timeout, and forward on the 408 that is generated to voicemail. Cheers, Henning

1 0

[OpenSER-Users] T_replicate Bad Request Issue
by Ali Jawad 04 Jun '08

04 Jun '08

Hi All I have the following setup Openser Server A --> Uses Localhost MySQL Server Openser Server B --> Uses ServerA MySQL server Each server is in a different location. Both servers share the same database. I.e. in other words a user can successfully logon to either server. Both servers have a similar configuration file. My problem at hand is the following User A and User C logon to Server A User B logs on to Server B I need user A or user C to be able to call user B. I am using t_replicate for this setup. The relevant config section is found below: if(src_ip == OTHERSERVER) { xlog("L_INFO","PACKET FROM 9 PEER SERVER"); save("location"); }else{ xlog("L_INFO","NORMAL Packet"); if(is_method("REGISTER")){ if(!www_authorize("sero.test.net", "subscriber")){ www_challenge("sero.ahwar.net", "1"); return; } save("location"); t_replicate("sip:OTHERSERVER:5060"); return; } I have looked at traces and it seems to me that if a user logs on to a server t_replcicate forwards those register packets to the second server. However the second server is not reponding to those register packets it is replying with Bad Request replies U OriginatingServer:5060 -> Target_of_Treplicate:5060 REGISTER sip:OriginatingServer SIP/2.0. Via: SIP/2.0/UDP OriginatingServer;branch=z9hG4bKe47f.4bd99ca4.0. Via: SIP/2.0/UDP 192.168.0.176:45374;received=193.227.186.146;branch=z9hG4bK-d87543-00509 f6c6f4a047f-1--d87543-;rport=45374. Max-Forwards: 69. Contact: <sip:9613041705@193.227.186.146:45374;rinstance=1367153d6d22ca20>;expire s=0. To: "9613041705"<sip:9613041705@OriginatingServer>. From: "9613041705"<sip:9613041705@OriginatingServer>;tag=7d242230. Call-ID: MmE3MTlmZDhiYWI0NjNmZjI5NWU5Njg1MmNhNDUwMDU.. CSeq: 6 REGISTER. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: X-Lite release 1006e stamp 34025. Authorization: Digest username="9613041705",realm="sero.ahwar.net",nonce="484653b8b642971aef30 31dfa8d894fa90a4faf3",uri="sip:OriginatingServer",response="fd7943d66556 832aff9abcba071a9d85",cnonce="4f6aa73e9cba374f",nc=00000001,qop=auth,alg orithm=MD5. Content-Length: 0. . U Target_of_Treplicate:5060 -> OriginatingServer:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP OriginatingServer;branch=z9hG4bKe47f.4bd99ca4.0. Via: SIP/2.0/UDP 192.168.0.176:45374;received=193.227.186.146;branch=z9hG4bK-d87543-00509 f6c6f4a047f-1--d87543-;rport=45374. To: "9613041705"<sip:9613041705@OriginatingServer>;tag=991a35e6ab26c1f4ea62d e8176a0818c.502f. From: "9613041705"<sip:9613041705@OriginatingServer>;tag=7d242230. Call-ID: MmE3MTlmZDhiYWI0NjNmZjI5NWU5Njg1MmNhNDUwMDU.. CSeq: 6 REGISTER. Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . U OriginatingServer:5060 -> Target_of_Treplicate:5060 REGISTER sip:OriginatingServer SIP/2.0. Via: SIP/2.0/UDP OriginatingServer;branch=z9hG4bK4cef.3d1bfc.0. Via: SIP/2.0/UDP 192.168.0.176:7988;received=193.227.186.146;branch=z9hG4bK-d87543-221288 3e6a0dc710-1--d87543-;rport=7988. Max-Forwards: 69. Contact: <sip:9613041705@192.168.0.176:7988;rinstance=c1ac0eda25c39951>. To: "9613041705"<sip:9613041705@OriginatingServer>. From: "9613041705"<sip:9613041705@OriginatingServer>;tag=7f0b7450. Call-ID: MGJhNmM1ZDdlZmNmNWQ2ODZkOWE4YWEwNzkzNmEzMzA.. CSeq: 2 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: X-Lite release 1006e stamp 34025. Authorization: Digest username="9613041705",realm="sero.ahwar.net",nonce="484653c5b066afdc9b4d 3bc20faa9e549e1bbe93",uri="sip:OriginatingServer",response="fc7ee7ed4e3c 68f85de2a298f7579421",cnonce="42f3c2c5b71b4623",nc=00000001,qop=auth,alg orithm=MD5. Content-Length: 0. . U Target_of_Treplicate:5060 -> OriginatingServer:5060 SIP/2.0 400 Bad Request. Via: SIP/2.0/UDP OriginatingServer;branch=z9hG4bK4cef.3d1bfc.0. Via: SIP/2.0/UDP 192.168.0.176:7988;received=193.227.186.146;branch=z9hG4bK-d87543-221288 3e6a0dc710-1--d87543-;rport=7988. To: "9613041705"<sip:9613041705@OriginatingServer>;tag=991a35e6ab26c1f4ea62d e8176a0818c.7860. From: "9613041705"<sip:9613041705@OriginatingServer>;tag=7f0b7450. Call-ID: MGJhNmM1ZDdlZmNmNWQ2ODZkOWE4YWEwNzkzNmEzMzA.. CSeq: 2 REGISTER. Contact: <sip:9613041705@192.168.0.176:7988;rinstance=c1ac0eda25c39951>;expires=5 67. P-Registrar-Error: Invalid CSeq number. Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . U OriginatingServer:5060 -> Target_of_Treplicate:5060 REGISTER sip:OriginatingServer SIP/2.0. Via: SIP/2.0/UDP OriginatingServer;branch=z9hG4bK5cef.dfe76626.0. Via: SIP/2.0/UDP 192.168.0.176:7988;received=193.227.186.146;branch=z9hG4bK-d87543-022b94 01307f967e-1--d87543-;rport=7988. Max-Forwards: 69. Contact: <sip:9613041705@192.168.0.176:7988;rinstance=c1ac0eda25c39951>;expires=0 . To: "9613041705"<sip:9613041705@OriginatingServer>. From: "9613041705"<sip:9613041705@OriginatingServer>;tag=7f0b7450. Call-ID: MGJhNmM1ZDdlZmNmNWQ2ODZkOWE4YWEwNzkzNmEzMzA.. CSeq: 3 REGISTER. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: X-Lite release 1006e stamp 34025. Authorization: Digest username="9613041705",realm="sero.ahwar.net",nonce="484653c5b066afdc9b4d 3bc20faa9e549e1bbe93",uri="sip:OriginatingServer",response="9ded822006fa 51735838d3fdea96800b",cnonce="68a12fcb5c00aeaf",nc=00000002,qop=auth,alg orithm=MD5. Content-Length: 0. . U Target_of_Treplicate:5060 -> OriginatingServer:5060 SIP/2.0 200 OK. Via: SIP/2.0/UDP OriginatingServer;branch=z9hG4bK5cef.dfe76626.0. Via: SIP/2.0/UDP 192.168.0.176:7988;received=193.227.186.146;branch=z9hG4bK-d87543-022b94 01307f967e-1--d87543-;rport=7988. To: "9613041705"<sip:9613041705@OriginatingServer>;tag=991a35e6ab26c1f4ea62d e8176a0818c.93c8. From: "9613041705"<sip:9613041705@OriginatingServer>;tag=7f0b7450. Call-ID: MGJhNmM1ZDdlZmNmNWQ2ODZkOWE4YWEwNzkzNmEzMzA.. CSeq: 3 REGISTER. Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . U OriginatingServer:5060 -> Target_of_Treplicate:5060 REGISTER sip:OriginatingServer SIP/2.0. Via: SIP/2.0/UDP OriginatingServer;branch=z9hG4bK2cef.bec40741.0. Via: SIP/2.0/UDP 192.168.0.176:7988;received=193.227.186.146;branch=z9hG4bK-d87543-ea14ff 1ac524e10c-1--d87543-;rport=7988. Max-Forwards: 69. Contact: <sip:9613041705@193.227.186.146:7988;rinstance=7d4ba8842a9e4a7e>. To: "9613041705"<sip:9613041705@OriginatingServer>. From: "9613041705"<sip:9613041705@OriginatingServer>;tag=7f0b7450. Call-ID: MGJhNmM1ZDdlZmNmNWQ2ODZkOWE4YWEwNzkzNmEzMzA.. CSeq: 4 REGISTER. Expires: 3600. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. User-Agent: X-Lite release 1006e stamp 34025. Authorization: Digest username="9613041705",realm="sero.ahwar.net",nonce="484653c5b066afdc9b4d 3bc20faa9e549e1bbe93",uri="sip:OriginatingServer",response="f126d934be4b 12c32a8ec783097a2ff6",cnonce="e3d03c6f36863764",nc=00000003,qop=auth,alg orithm=MD5. Content-Length: 0. . U Target_of_Treplicate:5060 -> OriginatingServer:5060 SIP/2.0 400 Bad Request. Via: SIP/2.0/UDP OriginatingServer;branch=z9hG4bK2cef.bec40741.0. Via: SIP/2.0/UDP 192.168.0.176:7988;received=193.227.186.146;branch=z9hG4bK-d87543-ea14ff 1ac524e10c-1--d87543-;rport=7988. To: "9613041705"<sip:9613041705@OriginatingServer>;tag=991a35e6ab26c1f4ea62d e8176a0818c.5d32. From: "9613041705"<sip:9613041705@OriginatingServer>;tag=7f0b7450. Call-ID: MGJhNmM1ZDdlZmNmNWQ2ODZkOWE4YWEwNzkzNmEzMzA.. CSeq: 4 REGISTER. Contact: <sip:9613041705@193.227.186.146:7988;rinstance=7d4ba8842a9e4a7e>;expires =566. P-Registrar-Error: Invalid CSeq number. Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. .

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Re: [OpenSER-Users] compilation on solaris 10
by Bogdan-Andrei Iancu 04 Jun '08

04 Jun '08

Hi Eric, To be honest, no, I haven;t, because I do not have access to such a machine. Could you post the errors you get now? Regards, Bogdan wliangy(a)gmail.com wrote: > Thanks! > But there is still problem. > May I ask whether you have compiled successfully on Solaris 8 or 10? > > 2008/5/26 Bogdan-Andrei Iancu <bogdan(a)voice-system.ro>: > >> Hi Eric, >> >> use "gmake" and not "make" on Solaris. >> >> Regards, >> Bogdan >> >> wliangy(a)gmail.com wrote: >> >>> Hi experts, >>> I am start user of openser and need use openser to support sctp. >>> now I have download the source codes of openser-1.3.1-tls. >>> then I compiled it, there have many errors that need your supports. >>> >>> e.g. >>> openser-1.3.1-tls> make all >>> Makefile.defs:717: *** missing separator. Stop. >>> >>> then I commented Makefile.defs line717 and 718. >>> # $(warning Old gcc detected ($(CC_SHORTVER)), use >>> gcc >= 3.1 \ >>> # for better results) >>> >>> Errors were still there. >>> >>> openser-1.3.1-tls> make all >>> Makefile:402: warning: overriding commands for target >>> `/opt/CiscoMGC$(DESTDIR)' >>> Makefile:393: warning: ignoring old commands for target >>> `/opt/CiscoMGC$(DESTDIR)' >>> yacc -d -b cfg cfg.y >>> "cfg.y", line 391: warning: redeclaration of precedence of SLASH. >>> >>> conflicts: 1 shift/reduce >>> flex cfg.lex >>> statistics.c:58: warning: #warning STATISTICS: Architecture with no >>> support for atomic operations. Using Locks!! >>> Makefile:402: warning: overriding commands for target >>> `/opt/CiscoMGC$(DESTDIR)' >>> Makefile:393: warning: ignoring old commands for target >>> `/opt/CiscoMGC$(DESTDIR)' >>> Compiling action.c >>> gcc -g -O9 -funroll-loops -Wall -mv8 -DNAME='"openser"' >>> -DVERSION='"1.3.1-notls"' -DARCH='"sparc64"' -DOS='"solaris"' >>> -DCOMPILER='"gcc 2.91.66"' -D__CPU_sparc64 -D__OS_solaris -D__SMP_no >>> -DCFG_DIR='"$(DESTDIR)/etc/openser/"' -DPKG_MALLOC -DSHM_MEM >>> -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE >>> -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC >>> -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 >>> -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD >>> -DHAVE_ALLOCA_H -DUSE_SIGACTION -D_POSIX_PTHREAD_SEMANTICS >>> -DHAVE_DEVPOLL -DHAVE_SELECT -c action.c -o action.o >>> parser/msg_parser.h: In function `char_msg_val': >>> In file included from action.h:27, >>> from action.c:44: >>> parser/msg_parser.h:300: parse error before `)' >>> forward.h: In function `msg_send': >>> In file included from action.c:49: >>> forward.h:95: parse error before `)' >>> forward.h:103: parse error before `)' >>> forward.h:107: parse error before `)' >>> mem/../lock_alloc.h: In function `lock_set_alloc': >>> In file included from mem/../locking.h:66, >>> from mem/../statistics.h:115, >>> from mem/shm_mem.h:33, >>> from ut.h:52, >>> from action.c:54: >>> mem/../lock_alloc.h:68: parse error before `)' >>> ut.h: In function `int2bstr': >>> In file included from action.c:54: >>> ut.h:172: parse error before `)' >>> ut.h: In function `un_escape': >>> ut.h:365: parse error before `)' >>> ut.h: In function `shm_str_dup': >>> ut.h:498: parse error before `)' >>> ut.h: In function `pkg_str_dup': >>> ut.h:515: parse error before `)' >>> ut.h: In function `str_strcmp': >>> ut.h:532: parse error before `)' >>> ut.h:535: parse error before `int' >>> ut.h:539: `minlen' undeclared (first use in this function) >>> ut.h:539: (Each undeclared identifier is reported only once >>> ut.h:539: for each function it appears in.) >>> ut.h:547: `alen' undeclared (first use in this function) >>> ut.h:547: `blen' undeclared (first use in this function) >>> ut.h: In function `str_strcasecmp': >>> ut.h:563: parse error before `)' >>> ut.h:566: parse error before `int' >>> ut.h:570: `minlen' undeclared (first use in this function) >>> ut.h:578: `alen' undeclared (first use in this function) >>> ut.h:578: `blen' undeclared (first use in this function) >>> action.c: In function `run_action_list': >>> action.c:140: parse error before `)' >>> action.c: In function `do_assign': >>> action.c:193: parse error before `)' >>> action.c:202: parse error before `)' >>> action.c:222: parse error before `)' >>> action.c:228: parse error before `)' >>> action.c: In function `do_action': >>> action.c:302: parse error before `)' >>> action.c:309: parse error before `)' >>> action.c:335: parse error before `)' >>> action.c:498: parse error before `)' >>> action.c:524: parse error before `)' >>> action.c:682: parse error before `)' >>> action.c:710: parse error before `)' >>> action.c:723: parse error before `)' >>> action.c:757: parse error before `)' >>> action.c:763: parse error before `)' >>> action.c:807: parse error before `)' >>> action.c:819: parse error before `)' >>> action.c:870: parse error before `)' >>> action.c:896: parse error before `)' >>> action.c:904: parse error before `)' >>> action.c:930: parse error before `)' >>> action.c:939: parse error before `)' >>> action.c:945: parse error before `)' >>> action.c:966: parse error before `)' >>> make: *** [action.o] Error 1 >>> >>> Your any help is much appreciated! >>> >>> BR, >>> Eric >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)lists.openser.org >>> http://lists.openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >> > >

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[OpenSER-Users] Why can't I "append_branch()" in FAILURE_ROUTE if reply is 6XX ??
by Zappasodi Daniele 03 Jun '08

03 Jun '08

Hello, If I want to have the ability to add a new branch for every negative response, included the 6xx (and despite what is written in the RFC3261), where should I change? It is enough to remove the following code in t_should_relay_response function? 783 if (new_code>=600) { 784 /* this is a winner and close all branches */ 785 which_cancel( Trans, cancel_bitmap ); 786 picked_branch=branch; 787 /* no more new branches should be added to this transaction */ 788 Trans->flags |= T_NO_NEW_BRANCHES_FLAG; ********************************************************************** The information in this message is confidential and may be legally privileged. It is intended solely for the addressee. Access to this message by anyone else is unauthorized. If you are not the intended recipient, any disclosure, copying, or distribution of the message, or any action or omission taken by you in reliance on it, is prohibited and may be unlawful. Please immediately contact the sender if you have received this message inerror. **********************************************************************

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[OpenSER-Users] Problem using path module
by Thomas Mangin 03 Jun '08

03 Jun '08

Hello, I am trying to configure a number of openser servers. I have the following testbed: Phone1 + Proxy - Registrar - out proxy Phone2/ The Proxy acts as outbound proxy for customers and uses ienum to route messages to the right registrar (using user@domain), the registrar then figures out where the destination is and sends the packet out to ENUM/PSTN or in my case back to the same proxy. The proxy is running nathelper (and redirects rtp to an rtp proxy) and performs route-recording. The proxy configuration has: modparam("path", "use_received", 0) and the registrar configuration contains: modparam("registrar", "use_path", 1) modparam("registrar", "path_mode", 0) modparam("registrar", "path_use_received", 1) When registering all is fine and well and the packet flows like: Phone<X> -> Proxy -> Registrar (storing in DB) -> Proxy -> Phone<X> and the DB contains in path: <sip:ProxyIP;lr;received=sip:PhoneIP:PhonePort> However, when performing an invite the ACK packet (INVITE/OK/ACK) can not be routed properly as it is not part of the transaction, which causes the call to be cancelled after a few seconds (as it should be given the constance). The ACK packet starts as URI: <sip:user@<Registrar IP>:5060;rinstance=xxxxxx> Contact: <sip:user@PhoneIP:PhonePort> with the IP and Port correctly translated from the internal to the firewall external. It then travels like this: Phone1 -> Proxy -> Registrar > Proxy <stop here> When leaving Registrar the ACK URI has been changed to the one stored in the DB path field ie: <sip:ProxyIP;lr;ftag=yyyyyy> at that point the Proxy is not able to figure out where to forward the packet even if the Contact field of the packet is still correctly set to the IP:Port of the phone to be reached. I was wondering if I am missing some understanding of how RFC3327 is working. I can see a few possible solutions: - generate a first Path header with the Phone information on the proxy before the current one. - try to catch those 'out of transaction ACK' and set the URI to the contact But before doing anything I would appreciate if someone could let me know if I am attempting something idiotic from a design point of view and tell me why what I expected to 'just work' does not behave as expected. Thank you very much your your time and help. Thomas

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[OpenSER-Users] NEW FUNCTIONALITY: secure nonce in auth module
by Bogdan-Andrei Iancu 03 Jun '08

03 Jun '08

Hi, Thanks to Anca Vamanu < anca at voice-system dot ro> , the auth module has a new mechanism for protecting the nonce against third party attacks. So far, the auth module provided only nonce lifetime limitation to prevent re-usage of the nonce. The new mechanism offers protection against sniffing intrusion. The module generates and verifies the nonces so that they can be used only once (in an auth response). This is done by having a lifetime value and an index associated with every nonce. Using only an expiration value is not good enough because,as this value has to be of few tens of seconds, it is possible for someone to sniff on the network, get the credentials and then reuse them in another packet with which to register a different contact or make calls using the other's account. The index ensures that this will never be possible since it is generated as unique through the lifetime of the nonce. What problem is solved: ======================== Typical intrusion scenario is: 1) UAC sends INVITE with no credentials 2) openser sends challenge to UAC 3) UAC sends INVITE with credentials (lifetime 1 minute) 4) the credentials (auth response) are exposed on the net, so anybody can grab them (lets assume by person X) 5) openser accepts the credentials and call is establish 6) person X injects (by sending to openser) an re-INVITE (to change the media destination, for example) by re-using the credentials that he grabbed from the net. 7) credentials are accepted by openser if the lifetime is not expired. So, third parties can modify (or even initiate) dialogs by re-using the credentials they captures from a previous authentication. The only limitation is the nonce lifetime which is about 30 - 60 seconds, long enough for allowing such an atack. The new mechanism, by blocking the re-usage of the same nonce (even if the lifetime is still valid), will block step 7) as the same nonce was used in step 5) (by UAC). Technical details: ================== The auth module keeps state for each nonce - to validate it only on the first usage. A binary array (which can by default accomodate 100K nonces) is used to keep the state. An index in this array is allocated when the challenge is generated; this index in kept for the whole life duration of the nonce. After the first auth result (for the nonce), the following auth results for that nonce are discarded and re-challenged. As nonce can be generated and later checked from different processes, the nonce state (binary array) is kept in shm memory and the access to it is synchronized via locking. The see what is the overhead added by the computation time (keeping the nonce state) and locking (sharing between processes), Anca run several of tests : - openser single process - openser multiple processes, single processor machine - openser multiple processes, multi processor machine For tests we were using authentication via PV (for password and username) to avoid using any slow data backend (radius or DB) that might eclipse the overhead. The measured overhead is 4% - 7% (from the original version). Regards, Bogdan

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[OpenSER-Users] how to configure to support PRACK
by Steven Wu 03 Jun '08

03 Jun '08

Hi All, Could someone tell me how to config openser 1.3.x route to support PRACK? I had installed openser 1.3.x and didn't change any route configuration. My scenario is: A openser B | --> INVITE (with 100rel) -> | --> INVITE (with 100rel) --> | | | | | <--183 <-- | <-- 183 <-- | | | --> PRACK --> | | <-- 404 Not here <--- | For the default the openser reply the "404 Not here" to A. I found the '404' was replied here: if (has_totag()) { # sequential request withing a dialog should # take the path determined by record-routing if (loose_route()) { if (is_method("BYE")) { setflag(1); # do accouting ... setflag(3); # ... even if the transaction fails } route(1); } else { /* !loose_route */ /* uncomment the following lines if you want to enable presence */ ##if (is_method("SUBSCRIBE") && $rd == "your.server.ip.address") { ## # in-dialog subscribe requests ## route(2); ## exit; ##} if ( is_method("ACK") ) { if ( t_check_trans() ) { # non loose-route, but stateful ACK; must be an ACK after a 487 or e.g. 404 from upstream server t_relay(); exit; } else { # ACK without matching transaction ... ignore and discard.\n"); exit; } ## These code will cause openser reply 483 response to PRACK if ( is_method("PRACK") ) { t_relay(); exit; } } ## Here reply me the 404 sl_send_reply("404","Not here"); } exit; } Than I added the green code to route the PRACK, But I received "483","Too Many Hops" from openser. And I found the Max-Forward is 70 not a wrong value. Does anyone can help me? Best regards, Steven Wu

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Re: [OpenSER-Users] Trouble with RADIUS Accounting
by Dmitry Lyubimkov 03 Jun '08

03 Jun '08

But we did it!!! I don't need to remove this attributes. We need User-Name, Called-Station-Id, Calling-Station-Id, etc for billing purpose. But our RADIUS server does not receive them from OpenSER! Look at our Start/Stop messages. There are not such important attributes as User-Name, etc. Is this bug of OpenSER or bug of our script? If it is bug of script, then what commands, what modules, what modparam can lead to such result? Dmitry -----Original Message----- From: Mike O'Connor [mailto:mike@pineview.net] Sent: Monday, June 02, 2008 1:07 PM To: Dmitry Lyubimkov Subject: Re: [OpenSER-Users] Trouble with RADIUS Accounting Hi Dmitry You not can remove them from OpenSER with out adjusting the code, instead you need to put the radius packets thought something like a radius proxy which removes them. Mike Dmitry Lyubimkov wrote: > Hi. > We have new installation of OpenSER V.1.3.1 with new script. All work fine > except RADIUS accounting. Accounting Start and Stop records come to RADIUS > server, but there are no some attributes in them - User-Name, > Called-Station-Id, Calling-Station-Id,... > Example of Start and Stop records (FreeRADIUS log): > Acct-Status-Type = Start > Service-Type = IAPP-Register > Sip-Response-Code = 200 > Sip-Method = 1 > Event-Timestamp = "Jun 2 2008 10:34:05 MSD" > Sip-From-Tag = "5d12c92c" > Sip-To-Tag = "6845627C-1EF5" > Acct-Session-Id = "ODhmMjE4MmUxNzc3ZmZjN2U3MjJlM2I4NzQ0MjMzNTU." > NAS-Port = 5060 > Acct-Delay-Time = 0 > NAS-IP-Address = 172.16.2.10 > Client-IP-Address = 172.16.2.10 > Timestamp = 1212388445 > > Acct-Status-Type = Stop > Service-Type = IAPP-Register > Sip-Response-Code = 200 > Sip-Method = 8 > Event-Timestamp = "Jun 2 2008 10:34:12 MSD" > Sip-From-Tag = "5d12c92c" > Sip-To-Tag = "6845627C-1EF5" > Acct-Session-Id = "ODhmMjE4MmUxNzc3ZmZjN2U3MjJlM2I4NzQ0MjMzNTU." > NAS-Port = 5060 > Acct-Delay-Time = 0 > NAS-IP-Address = 172.16.2.10 > Client-IP-Address = 172.16.2.10 > Timestamp = 1212388452 > > Where is trouble? Required attributes are described in dictionary file. If I > remove it then OpenSER says > Jun 2 11:02:23 [4064] ERROR:auth_radius:mod_init: auth_radius: can't get > code for the User-Name attribute > Jun 2 11:02:23 [4064] ERROR:core:init_mod: failed to initialize module > auth_radius > Jun 2 11:02:23 [4064] ERROR:core:main: error while initializing modules > > How and where in script we can prohibit send of some attributes? > > Dmitry > > > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users >

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[OpenSER-Users] Can't get sipping working in nathelper module
by Pablo Guijarro Enríquez 03 Jun '08

03 Jun '08

Hi, Im using OpenSER v1.3.1 and sipping does not work with my configuration. Here is a sample: loadmodule "usrloc.so" loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 10) #modparam("nathelper", "ping_nated_only", 1) modparam("nathelper", "sipping_bflag", 7) modparam("nathelper", "sipping_from", "sip:pinger@cmovil.net") modparam("nathelper", "received_avp", "$avp(i:801)") modparam("registrar", "received_avp", "$avp(i:801)") modparam("usrloc", "db_mode", 2) modparam("usrloc", "nat_bflag", 6) route { if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); exit; } # for clients behind nats if ( nat_uac_test("1") ) { if (method=="REGISTER") { fix_nated_register(); setbflag(6); setbflag(7); } else if ( method=="SUBSCRIBE" || method=="PUBLISH ) { fix_nated_contact(); setbflag(6); setbflag(7); }; }; ... } ... TCP clients register without problems and NAT users are identified properly. Routing for them works fine and so on, but I cannot see any ping packet leaving OpenSER to keep the TCP connection alive. Is there something missing? Thank in advance. Pablo

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[OpenSER-Users] Register Distributed
by Marcio 02 Jun '08

02 Jun '08

Hi, Someone could have tips on how to register in local domain a UAc mobility of another domain (federation) using the OpenSER's config....when one of the INVITE's UAc have to be verified as valid in the home domain and the answer would be allowed its domain registration in place, with a pointer/indicator in the home domain...where he was as that moment?? UAs(local) (accept register after validation on UAs(home) \ \ (Invite/register) \ UAc (mobility)-- -- -- -- -- UAs(home Thx for any help... Marcio

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[OpenSER-Users] setting up radius accounting
by Pete Kay 02 Jun '08

02 Jun '08

Hi, I am trying to set up using Radius for accounting. As a newbie, I am simply following the standard configuration from the menu. Here is where I put my flags in openser.cfg: if(method=="BYE") { setflag(2); }; if (method=="INVITE") { # Account INVITE packets setflag(2); # Account Missed calls setflag(3); }; The cdr is successfully logged to the database via radius, but the starttime and endtime are incorrect. The starting time and ending time being recorded is the same. Also, I noticed that Openser only sends the message (both INSERT and UPDATE cdr) to radius when the BYE method is received. I was expecting Openser to instruct Radius to create a new record ( INSERT ) during the INVITE method and update the stoptime during the BYE method. Here is the SQL Query log i extracted from mysql: 80603 4:38:06 6481 Query UPDATE radacct SET acctstoptime = '2008-06-03 04:38:06', acctsessiontime = '', acctinputoctets = '0' << 32 | '0', acctoutputoctets = '0' << 32 | '0', acctterminatecause = '', acctstopdelay = '0', connectinfo_stop = '' WHERE acctsessionid = 'NWQyZGI1ZTkzNGM5ZjMzZTg5ZGQ2OGVmZDY5MjYxMGE.' AND username = '1005' AND nasipaddress = '127.0.0.1' 6481 Query INSERT INTO radacct (acctsessionid, acctuniqueid, username, realm, nasipaddress, nasportid, nasporttype, acctstarttime, acctstoptime, acctsessiontime, acctauthentic, connectinfo_start, connectinfo_stop, acctinputoctets, acctoutputoctets, calledstationid, callingstationid, acctterminatecause, servicetype, framedprotocol, framedipaddress, acctstartdelay, acctstopdelay) VALUES ('NWQyZGI1ZTkzNGM5ZjMzZTg5ZGQ2OGVmZDY5MjYxMGE.', '1412cd5bf3dd4a31', '1005', '192.168.1.104', ' 127.0.0.1', '5060', '', DATE_SUB('2008-06-03 04:38:06', INTERVAL (0 + 0) SECOND), '2008-06-03 04:38:06', '', '', '', '', '0' << 32 | '0', '0' << 32 | '0', 'sip:1006@192.168.1.104<sip%3A1006(a)192.168.1.104>', 'sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104>', '', 'IAPP-Register', '', '', '0', '0') I would appreciate any help on getting the radius accounting to work. Thanks alot in advance. Pete

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[OpenSER-Users] Question about proxy_authorization
by mark morreny 02 Jun '08

02 Jun '08

Hi, I am not able to proxy_authorize a INVITE request from UA if the US connects to Opnerser using Openser's public IP. I am wonering if there is any additional setting I need to enable that? From the log, Openser does recognize '88,251.97.47' as a local domain. I did not hard-code the private ip anywhere in openser.cfg. Tue Jun 3 03:20:05 2008-58.251.97.17::method <INVITE> r-uri < sip:1005@58.251.97.17 <sip%3A1005(a)58.251.97.17>> to-tag <<null>> 2nd via <<null>> to-header <sip:1005@58.251.97.17 <sip%3A1005(a)58.251.97.17>> from-header <sip:1006@58.251.97.17 <sip%3A1006(a)58.251.97.17>> auth-user <1006> Jun 3 03:20:05 [27566] DBG:maxfwd:is_maxfwd_present: value = 70 before fix nated contact Jun 3 03:20:05 [27566] DBG:core:parse_headers: flags=80 checking to see if setflag should do or not Jun 3 03:20:05 [27566] DBG:uri:has_totag: no totag Jun 3 03:20:05 [27566] DBG:core:parse_headers: flags=78 Jun 3 03:20:05 [27566] DBG:tm:t_lookup_request: start searching: hash=29544, isACK=0 Jun 3 03:20:05 [27566] DBG:tm:matching_3261: RFC3261 transaction matching failed Jun 3 03:20:05 [27566] DBG:tm:t_lookup_request: no transaction found it is not a REGISTER, so going to route 3 Jun 3 03:20:05 [27566] DBG:domain:is_domain_local: Realm '88,251.97.47' is local Jun 3 03:20:05 [27566] DBG:core:db_free_rows: freeing 1 rows Jun 3 03:20:05 [27566] DBG:core:db_free_rows: row[0]=0x81a1e18 Jun 3 03:20:05 [27566] DBG:core:db_free_rows: 0x81a1e18=pkg_free() RES_ROWS request is from a non-trusted domain Jun 3 03:20:05 [27566] DBG:auth:check_nonce: comparing [4844491107980b084e3d5d3a283446ad02324d49] and [4844491107980b084e3d5d3a283446ad02324d49] Jun 3 03:20:05 [27566] DBG:auth:check_response: our result = '578a6ec1af28eed79329ed5414313f43' Jun 3 03:20:05 [27566] DBG:auth:check_response: authorization failed Thanks in advance for your help. Regards, Mark

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[OpenSER-Users] Mediaproxy + XLite RTCP timeout
by Dmitry Lyubimkov 02 Jun '08

02 Jun '08

OpenSER v.1.3.1 We use Mediaproxy for NAT traversal purpose. All work fine except one feature. XLite has option to hangup call after RTCP inactivity. When call is not proxied this option work right. But when call is proxied by Nediaproxy it cannot be longer than 54 second + RTCP timeout in XLite. Xlite cannot see RTCP activity and hangup the call. Does anybody meet with this trouble? Dmitry

1 0

[OpenSER-Users] Problem authenticating Xlite or Zoiper in Openser
by voipanywhere 02 Jun '08

02 Jun '08

Hi All, After doing ngrep teste(a)xxx.xxx.xxx.xxx -lqtt -W byline port 5060 xlite somehow is sending wrong information. my subscriber is teste(a)xxx.xxx.xxx.xxx, i put it on xlite but on sip messages i get teste%40xxx.xxx.xxx.xxx and then of course the 401 error. something is changing the @ in the username to %40. i've been trying to change locales in debian, still didn't figure it out what the problem really is. i had the console with wrong code support for portuguese language, and could be it (i tought) now it's ok. tried to check /etc/environment and /etc/locales.gen to see if everything is okay, and it is now. Can be some strange stuff because of utf, unicode etc? Thanks Regards from Portugal, Helder -- View this message in context: http://www.nabble.com/Problem-authenticating-Xlite-or-Zoiper-in-Openser-tp1… Sent from the OpenSER Users Mailing List mailing list archive at Nabble.com.

1 0

[OpenSER-Users] Trouble with RADIUS Accounting
by Dmitry Lyubimkov 02 Jun '08

02 Jun '08

Hi. We have new installation of OpenSER V.1.3.1 with new script. All work fine except RADIUS accounting. Accounting Start and Stop records come to RADIUS server, but there are no some attributes in them - User-Name, Called-Station-Id, Calling-Station-Id,... Example of Start and Stop records (FreeRADIUS log): Acct-Status-Type = Start Service-Type = IAPP-Register Sip-Response-Code = 200 Sip-Method = 1 Event-Timestamp = "Jun 2 2008 10:34:05 MSD" Sip-From-Tag = "5d12c92c" Sip-To-Tag = "6845627C-1EF5" Acct-Session-Id = "ODhmMjE4MmUxNzc3ZmZjN2U3MjJlM2I4NzQ0MjMzNTU." NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 172.16.2.10 Client-IP-Address = 172.16.2.10 Timestamp = 1212388445 Acct-Status-Type = Stop Service-Type = IAPP-Register Sip-Response-Code = 200 Sip-Method = 8 Event-Timestamp = "Jun 2 2008 10:34:12 MSD" Sip-From-Tag = "5d12c92c" Sip-To-Tag = "6845627C-1EF5" Acct-Session-Id = "ODhmMjE4MmUxNzc3ZmZjN2U3MjJlM2I4NzQ0MjMzNTU." NAS-Port = 5060 Acct-Delay-Time = 0 NAS-IP-Address = 172.16.2.10 Client-IP-Address = 172.16.2.10 Timestamp = 1212388452 Where is trouble? Required attributes are described in dictionary file. If I remove it then OpenSER says Jun 2 11:02:23 [4064] ERROR:auth_radius:mod_init: auth_radius: can't get code for the User-Name attribute Jun 2 11:02:23 [4064] ERROR:core:init_mod: failed to initialize module auth_radius Jun 2 11:02:23 [4064] ERROR:core:main: error while initializing modules How and where in script we can prohibit send of some attributes? Dmitry

1 0

[OpenSER-Users] rtpproxy
by Pezhman Lali 02 Jun '08

02 Jun '08

Dear, I found using rtpproxy on different machine is really more useful, for solving nat problems, but I don't know the reseon. good help for setting rtpproxy on different mach : http://www.iptel.org/rtpproxy_on_different_machine Best Mani

1 0

[OpenSER-Users] how to drop sip call when i receive ring
by abo 02 Jun '08

02 Jun '08

Dear sir: i am trying openser 1.2.0 version.and now i want to drop call when i got 180 or 183.this is my openser.cfg.can you help me. thanks a lot! # ----------- global configuration parameters ------------------------ debug=9 # 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" listen=192.168.1.254 alias=openser.org # Uncomment this to prevent the blacklisting of temporary not available destinations disable_dns_blacklist=yes # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database # - MySQL loaded for accounting as well loadmodule "/usr/local/lib/openser/modules/mysql.so" loadmodule "/usr/local/lib/openser/modules/xlog.so" loadmodule "/usr/local/lib/openser/modules/uri.so" loadmodule "/usr/local/lib/openser/modules/uri_db.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/acc.so" loadmodule "/usr/local/lib/openser/modules/avpops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! 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) # 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) # -- acc params -- modparam("acc", "db_url", "mysql://openser:openserrw@localhost/openser") modparam("acc", "db_flag", 7) modparam("acc", "db_missed_flag", 8) modparam("acc", "log_level", 3) modparam("acc", "log_flag", 2) modparam("acc", "log_missed_flag", 8) modparam("acc", "report_ack", 1) modparam("acc", "report_cancels", 1) modparam("acc", "failed_transaction_flag", 1) modparam("acc", "early_media", 1) modparam("acc", "db_extra", "src_ip=$si;dst_user=$tU;dst_domain=$du;src_user=$hdr(User-Agent)") # -- avpops params -- # avp aliases - define c_uri (canonical uri) as AVP ID 888 and s_ip as ID 999 avp_aliases="c_uri=i:888;s_ip=i:999" #tm timeout for voicemail params modparam("tm", "fr_timer", 2) modparam("tm", "fr_inv_timer", 35) modparam("tm", "noisy_ctimer", 1) # ------------------------- request routing logic ------------------- # main routing logic #Routing Script 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, >= 12048 bytes\n"); sl_send_reply("513", "Message too large, exceeded limit\n"); return; }; #record everything besides registers and acks if(method!="REGISTER" && method!="ACK") { setflag(1); }; #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(7); setflag(8); }; #grant route if route headers already present if (loose_route()) { route(1); return; }; #Always require authentication, which could result in a PSTN, ie $$$ if (method=="REGISTER") { # if(!www_authorize("openser.org", "subscriber")) # { # www_challenge("openser.org", "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 Authenticated Correctly\n"); save("location"); return; # }; }; if (method=="INVITE") { setflag(2); setflag(7); setflag(8); route(1); } } route[1] { #send the call outward if(uri=~"sip:[0-9]+@.*") { rewritehostport("192.168.1.100:5070"); #record_route(); t_on_reply("1"); # t_relay(); } if (!t_relay()) { xlog("L_WARN", "ERROR: t_relay failed"); sl_reply_error(); }; } onreply_route[1] { xlog("----------aaaaaaaaaa-------\n"); if (t_check_status("183")||t_check_status("180")){ exit; xlog("------------bbbbbbb------- \n"); } }

1 0

[OpenSER-Users] Please help with "auth:pre_auth: credentials with given realm not found" error
by mark morreny 31 May '08

31 May '08

Hi, I have found a few posts on this error from Google but none of the solutions works. I am able to use radius_proxy_authorize for REGISTER request without problem. But when I use radius_proxy_authorize for INVITE, Opensers can't authroize it. I have tried: radius_proxy_authorize("192.168.1.104")) proxy_authorize("192.168.1.104","subscriber") radius_proxy_authorize("")) proxy_authorize("","subscriber") I have been stuck with this for days already and I truely hope someone can tell me what could be wrong with my setup. I am pretty much in a stage of desperation right now. Thanks in advance for all your help. Here are the relevant lines in openser.cfg: #if (!radius_proxy_authorize("192.168.1.104")) { if (!proxy_authorize("192.168.1.104","subscriber")) { www_challenge("192.168.1.104","1"); exit; } Here is the openser debug log, I have highlighted the part that proxy_authorize throws error: Jun 1 05:37:16 [29028] DBG:core:parse_msg: SIP Request: Jun 1 05:37:16 [29028] DBG:core:parse_msg: method: <INVITE> Jun 1 05:37:16 [29028] DBG:core:parse_msg: uri: < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> Jun 1 05:37:16 [29028] DBG:core:parse_msg: version: <SIP/2.0> Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=2 Jun 1 05:37:16 [29028] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-f072f593>; state=16 Jun 1 05:37:16 [29028] DBG:core:parse_via: end of header reached, state=5 Jun 1 05:37:16 [29028] DBG:core:parse_headers: via found, flags=2 Jun 1 05:37:16 [29028] DBG:core:parse_headers: this is the first via Jun 1 05:37:16 [29028] DBG:core:receive_msg: After parse_msg... Jun 1 05:37:16 [29028] DBG:core:receive_msg: preparing to run routing scripts... Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=8 Jun 1 05:37:16 [29028] DBG:core:parse_to: end of header reached, state=10 Jun 1 05:37:16 [29028] DBG:core:parse_to: display={"1006"}, ruri={ sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>} Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: <To> [33]; uri=[ sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>] Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: to body ["1006" < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> ] Jun 1 05:37:16 [29028] DBG:core:pv_get_to_attr: no To tag Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=ffffffffffffffff Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: cseq <CSeq>: <101> <INVITE> Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: content_length=393 Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: found end of header Jun 1 05:37:16 [29028] DBG:core:parse_to_param: tag=39a9124ca59076d9o0 Jun 1 05:37:16 [29028] DBG:core:parse_to: end of header reached, state=29 Jun 1 05:37:16 [29028] DBG:core:parse_to: display={"1005"}, ruri={ sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104>} Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=14000 Jun 1 05:37:16 [29028] DBG:core:pv_get_authattr: no [Proxy-]Authorization header Sun Jun 1 05:37:16 2008-192.168.1.101::method <INVITE> r-uri < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> to-tag <<null>> 2nd via <<null>> to-header <sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> from-header <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104>> auth-user <<null>> Jun 1 05:37:16 [29028] DBG:maxfwd:is_maxfwd_present: value = 70 Jun 1 05:37:16 [29028] DBG:core:check_via_address: params 192.168.1.101, 192.168.1.101, 0 Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=80 before fix nated contact Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=80 Jun 1 05:37:16 [29028] DBG:uri:has_totag: no totag Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=78 Jun 1 05:37:16 [29028] DBG:tm:t_lookup_request: start searching: hash=59293, isACK=0 Jun 1 05:37:16 [29028] DBG:tm:matching_3261: RFC3261 transaction matching failed Jun 1 05:37:16 [29028] DBG:tm:t_lookup_request: no transaction found Jun 1 05:37:16 [29028] DBG:domain:is_domain_local: Realm '192.168.1.104' is local Jun 1 05:37:16 [29028] DBG:core:db_free_rows: freeing 1 rows Jun 1 05:37:16 [29028] DBG:core:db_free_rows: row[0]=0x81a1c28 Jun 1 05:37:16 [29028] DBG:core:db_free_rows: 0x81a1c28=pkg_free() RES_ROWS Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=10000 Jun 1 05:37:16 [29028] DBG:auth:pre_auth: credentials with given realm not found Jun 1 05:37:16 [29028] DBG:auth:build_auth_hf: 'WWW-Authenticate: Digest realm="192.168.1.104", nonce="4841c6380fa8573a4e9794b1a0a2df9f3e53cb49", qop="auth" ' Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=ffffffffffffffff Jun 1 05:37:16 [29028] DBG:core:check_via_address: params 192.168.1.101, 192.168.1.101, 0 Jun 1 05:37:16 [29028] DBG:core:destroy_avp_list: destroying list 0xb5bf9be0 Jun 1 05:37:16 [29028] DBG:core:receive_msg: cleaning up Jun 1 05:37:16 [29028] DBG:core:parse_msg: SIP Request: Jun 1 05:37:16 [29028] DBG:core:parse_msg: method: <ACK> Jun 1 05:37:16 [29028] DBG:core:parse_msg: uri: < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> Jun 1 05:37:16 [29028] DBG:core:parse_msg: version: <SIP/2.0> Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=2 Jun 1 05:37:16 [29028] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-f072f593>; state=16 Jun 1 05:37:16 [29028] DBG:core:parse_via: end of header reached, state=5 Jun 1 05:37:16 [29028] DBG:core:parse_headers: via found, flags=2 Jun 1 05:37:16 [29028] DBG:core:parse_headers: this is the first via Jun 1 05:37:16 [29028] DBG:core:receive_msg: After parse_msg... Jun 1 05:37:16 [29028] DBG:core:receive_msg: preparing to run routing scripts... Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=8 Jun 1 05:37:16 [29028] DBG:core:parse_to_param: tag=026ab2c49274b4e0186dd72188f7d50a.939a Jun 1 05:37:16 [29028] DBG:core:parse_to: end of header reached, state=29 Jun 1 05:37:16 [29028] DBG:core:parse_to: display={"1006"}, ruri={ sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>} Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: <To> [75]; uri=[ sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>] Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: to body ["1006" < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>>] Jun 1 05:37:16 [29028] DBG:sl:sl_filter_ACK: local ACK found -> dropping it! Jun 1 05:37:16 [29028] DBG:core:destroy_avp_list: destroying list (nil) Jun 1 05:37:16 [29028] DBG:core:receive_msg: cleaning up Jun 1 05:37:16 [29028] DBG:core:parse_msg: SIP Request: Jun 1 05:37:16 [29028] DBG:core:parse_msg: method: <INVITE> Jun 1 05:37:16 [29028] DBG:core:parse_msg: uri: < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> Jun 1 05:37:16 [29028] DBG:core:parse_msg: version: <SIP/2.0> Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=2 Jun 1 05:37:16 [29028] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-dc12d56c>; state=16 Jun 1 05:37:16 [29028] DBG:core:parse_via: end of header reached, state=5 Jun 1 05:37:16 [29028] DBG:core:parse_headers: via found, flags=2 Jun 1 05:37:16 [29028] DBG:core:parse_headers: this is the first via Jun 1 05:37:16 [29028] DBG:core:receive_msg: After parse_msg... Jun 1 05:37:16 [29028] DBG:core:receive_msg: preparing to run routing scripts... Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=8 Jun 1 05:37:16 [29028] DBG:core:parse_to: end of header reached, state=10 Jun 1 05:37:16 [29028] DBG:core:parse_to: display={"1006"}, ruri={ sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>} Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: <To> [33]; uri=[ sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>] Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: to body ["1006" < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> ] Jun 1 05:37:16 [29028] DBG:core:pv_get_to_attr: no To tag Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=ffffffffffffffff Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: cseq <CSeq>: <102> <INVITE> Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: content_length=393 Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: found end of header Jun 1 05:37:16 [29028] DBG:core:parse_to_param: tag=39a9124ca59076d9o0 Jun 1 05:37:16 [29028] DBG:core:parse_to: end of header reached, state=29 Jun 1 05:37:16 [29028] DBG:core:parse_to: display={"1005"}, ruri={ sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104>} Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=14000 Sun Jun 1 05:37:16 2008-192.168.1.101::method <INVITE> r-uri < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> to-tag <<null>> 2nd via <<null>> to-header <sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> from-header <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104>> auth-user <1005> Jun 1 05:37:16 [29028] DBG:maxfwd:is_maxfwd_present: value = 70 Jun 1 05:37:16 [29028] DBG:core:check_via_address: params 192.168.1.101, 192.168.1.101, 0 Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=80 Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=80 Jun 1 05:37:16 [29028] DBG:uri:has_totag: no totag Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=78 Jun 1 05:37:16 [29028] DBG:tm:t_lookup_request: start searching: hash=59290, isACK=0 Jun 1 05:37:16 [29028] DBG:tm:matching_3261: RFC3261 transaction matching failed Jun 1 05:37:16 [29028] DBG:tm:t_lookup_request: no transaction found Jun 1 05:37:16 [29028] DBG:domain:is_domain_local: Realm '192.168.1.104' is local Jun 1 05:37:16 [29028] DBG:core:db_free_rows: freeing 1 rows Jun 1 05:37:16 [29028] DBG:core:db_free_rows: row[0]=0x81a1c28 Jun 1 05:37:16 [29028] DBG:core:db_free_rows: 0x81a1c28=pkg_free() RES_ROWS Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=10000 Jun 1 05:37:16 [29028] DBG:auth:pre_auth: credentials with given realm not found Jun 1 05:37:16 [29028] DBG:auth:build_auth_hf: 'WWW-Authenticate: Digest realm="192.168.1.104", nonce="4841c6380fa8573a4e9794b1a0a2df9f3e53cb49", qop="auth" ' Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=ffffffffffffffff Jun 1 05:37:16 [29028] DBG:core:check_via_address: params 192.168.1.101, 192.168.1.101, 0 Jun 1 05:37:16 [29028] DBG:core:destroy_avp_list: destroying list 0xb5bf9be0 Jun 1 05:37:16 [29028] DBG:core:receive_msg: cleaning up Jun 1 05:37:16 [29028] DBG:core:parse_msg: SIP Request: Jun 1 05:37:16 [29028] DBG:core:parse_msg: method: <ACK> Jun 1 05:37:16 [29028] DBG:core:parse_msg: uri: < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>> Jun 1 05:37:16 [29028] DBG:core:parse_msg: version: <SIP/2.0> Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=2 Jun 1 05:37:16 [29028] DBG:core:parse_via_param: found param type 232, <branch> = <z9hG4bK-dc12d56c>; state=16 Jun 1 05:37:16 [29028] DBG:core:parse_via: end of header reached, state=5 Jun 1 05:37:16 [29028] DBG:core:parse_headers: via found, flags=2 Jun 1 05:37:16 [29028] DBG:core:parse_headers: this is the first via Jun 1 05:37:16 [29028] DBG:core:receive_msg: After parse_msg... Jun 1 05:37:16 [29028] DBG:core:receive_msg: preparing to run routing scripts... Jun 1 05:37:16 [29028] DBG:core:parse_headers: flags=8 Jun 1 05:37:16 [29028] DBG:core:parse_to_param: tag=026ab2c49274b4e0186dd72188f7d50a.6388 Jun 1 05:37:16 [29028] DBG:core:parse_to: end of header reached, state=29 Jun 1 05:37:16 [29028] DBG:core:parse_to: display={"1006"}, ruri={ sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>} Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: <To> [75]; uri=[ sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>] Jun 1 05:37:16 [29028] DBG:core:get_hdr_field: to body ["1006" < sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104>>] Jun 1 05:37:16 [29028] DBG:sl:sl_filter_ACK: local ACK found -> dropping it! Jun 1 05:37:16 [29028] DBG:core:destroy_avp_list: destroying list (nil) Jun 1 05:37:16 [29028] DBG:core:receive_msg: cleaning up Here is the SIP trace: U 192.168.1.101:5062 -> 192.168.1.104:5060 INVITE sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.101:5062;branch=z9hG4bK-d392c9a5. From: "1005" <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=131fa1047c30f2a4o0. To: <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104>>. Call-ID: a3888884-92d13124(a)192.168.1.101. CSeq: 101 INVITE. Max-Forwards: 70. Contact: "1005" <sip:1005@192.168.1.101:5062>. Expires: 240. User-Agent: Sipura/SPA921-4.1.10(b). Content-Length: 395. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Content-Type: application/sdp. . v=0. o=- 36185 36185 IN IP4 192.168.1.101. s=-. c=IN IP4 192.168.1.101. t=0 0. m=audio 16424 RTP/AVP 0 2 4 8 18 96 97 98 101. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 192.168.1.104:5060 -> 192.168.1.101:5062 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.101:5062;branch=z9hG4bK-d392c9a5;rport=5062. From: "1005" <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=131fa1047c30f2a4o0. To: <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=026ab2c49274b4e0186dd72188f7d50a.f608. Call-ID: a3888884-92d13124(a)192.168.1.101. CSeq: 101 INVITE. WWW-Authenticate: Digest realm="192.168.1.104", nonce="4841caae665a98cec71734a5cf84c70d10ac8f63", qop="auth". Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . U 192.168.1.101:5062 -> 192.168.1.104:5060 ACK sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.101:5062;branch=z9hG4bK-d392c9a5. From: "1005" <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=131fa1047c30f2a4o0. To: <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=026ab2c49274b4e0186dd72188f7d50a.f608. Call-ID: a3888884-92d13124(a)192.168.1.101. CSeq: 101 ACK. Max-Forwards: 70. Contact: "1005" <sip:1005@192.168.1.101:5062>. User-Agent: Sipura/SPA921-4.1.10(b). Content-Length: 0. . U 192.168.1.101:5062 -> 192.168.1.104:5060 INVITE sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.101:5062;branch=z9hG4bK-4cf05fb6. From: "1005" <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=131fa1047c30f2a4o0. To: <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104>>. Call-ID: a3888884-92d13124(a)192.168.1.101. CSeq: 102 INVITE. Max-Forwards: 70. Authorization: Digest username="1005",realm="192.168.1.104 ",nonce="4841caae665a98cec71734a5cf84c70d10ac8f63",uri=" sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> ",algorithm=MD5,response="e39bd9dc46a72ea191fca87e98cf9136",qop=auth,nc=00000001,cnonce="9d027a79". Contact: "1005" <sip:1005@192.168.1.101:5062>. Expires: 240. User-Agent: Sipura/SPA921-4.1.10(b). Content-Length: 395. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Content-Type: application/sdp. . v=0. o=- 36185 36185 IN IP4 192.168.1.101. s=-. c=IN IP4 192.168.1.101. t=0 0. m=audio 16424 RTP/AVP 0 2 4 8 18 96 97 98 101. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 192.168.1.104:5060 -> 192.168.1.101:5062 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.101:5062;branch=z9hG4bK-4cf05fb6;rport=5062. From: "1005" <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=131fa1047c30f2a4o0. To: <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=026ab2c49274b4e0186dd72188f7d50a.218e. Call-ID: a3888884-92d13124(a)192.168.1.101. CSeq: 102 INVITE. WWW-Authenticate: Digest realm="192.168.1.104", nonce="4841caae665a98cec71734a5cf84c70d10ac8f63", qop="auth". Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . U 192.168.1.101:5062 -> 192.168.1.104:5060 ACK sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.101:5062;branch=z9hG4bK-4cf05fb6. From: "1005" <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=131fa1047c30f2a4o0. To: <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=026ab2c49274b4e0186dd72188f7d50a.218e. Call-ID: a3888884-92d13124(a)192.168.1.101. CSeq: 102 ACK. Max-Forwards: 70. Authorization: Digest username="1005",realm="192.168.1.104 ",nonce="4841caae665a98cec71734a5cf84c70d10ac8f63",uri=" sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> ",algorithm=MD5,response="dc66b568ed8ec48f0f405aa063156eaa",qop=auth,nc=00000001,cnonce="9d027a79". Contact: "1005" <sip:1005@192.168.1.101:5062>. User-Agent: Sipura/SPA921-4.1.10(b). Content-Length: 0. . U 192.168.1.101:5062 -> 192.168.1.104:5060 INVITE sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.101:5062;branch=z9hG4bK-e2c260be. From: "1005" <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=131fa1047c30f2a4o0. To: <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104>>. Call-ID: a3888884-92d13124(a)192.168.1.101. CSeq: 103 INVITE. Max-Forwards: 70. Authorization: Digest username="1005",realm="192.168.1.104 ",nonce="4841caae665a98cec71734a5cf84c70d10ac8f63",uri=" sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> ",algorithm=MD5,response="8d2b52278a72326a57361ec1ca2601c3",qop=auth,nc=00000001,cnonce="f4b6f3a0". Contact: "1005" <sip:1005@192.168.1.101:5062>. Expires: 240. User-Agent: Sipura/SPA921-4.1.10(b). Content-Length: 395. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Content-Type: application/sdp. . v=0. o=- 36185 36185 IN IP4 192.168.1.101. s=-. c=IN IP4 192.168.1.101. t=0 0. m=audio 16424 RTP/AVP 0 2 4 8 18 96 97 98 101. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:18 G729a/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. U 192.168.1.104:5060 -> 192.168.1.101:5062 SIP/2.0 401 Unauthorized. Via: SIP/2.0/UDP 192.168.1.101:5062;branch=z9hG4bK-e2c260be;rport=5062. From: "1005" <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=131fa1047c30f2a4o0. To: <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=026ab2c49274b4e0186dd72188f7d50a.8d8a. Call-ID: a3888884-92d13124(a)192.168.1.101. CSeq: 103 INVITE. WWW-Authenticate: Digest realm="192.168.1.104", nonce="4841caae665a98cec71734a5cf84c70d10ac8f63", qop="auth". Server: OpenSER (1.3.2-notls (i386/linux)). Content-Length: 0. . U 192.168.1.101:5062 -> 192.168.1.104:5060 ACK sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.101:5062;branch=z9hG4bK-e2c260be. From: "1005" <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=131fa1047c30f2a4o0. To: <sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> >;tag=026ab2c49274b4e0186dd72188f7d50a.8d8a. Call-ID: a3888884-92d13124(a)192.168.1.101. CSeq: 103 ACK. Max-Forwards: 70. Authorization: Digest username="1005",realm="192.168.1.104 ",nonce="4841caae665a98cec71734a5cf84c70d10ac8f63",uri=" sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> ",algorithm=MD5,response="8ebde837f7c7c49f721d9f2c44b3797d",qop=auth,nc=00000001,cnonce="f4b6f3a0". Contact: "1005" <sip:1005@192.168.1.101:5062>. User-Agent: Sipura/SPA921-4.1.10(b). Content-Length: 0.

1 0

[Serusers] (no subject)
by Bin Zhang 31 May '08

31 May '08

-- Bin Zhang

1 0

[OpenSER-Users] Hints about to capture codes negative www_authorize??
by Marcio 30 May '08

30 May '08

Hi, Who would have tips on how to capture the return codes negative of the function "www_authorize" ??? I read somewhere that said to use $retval........but ...How can I to use it ("$retval") ?? -1 - Non existent user; -2 - invalid passwd; -3 - stale nonce; -4 - no credentials; -5 - error; Any help will be greatly appreciated. thx, Marcio

3 5

[OpenSER-Users] Fwd: Re: Installing perl module
by Henning Westerholt 30 May '08

30 May '08

---------- Forward message ---------- Subject: Re: [OpenSER-Users] Installing perl module Date: Freitag, 30. Mai 2008 From: "Yazid Hadj Said" <yazid.hadjsaid(a)gmail.com> Tp: "Henning Westerholt" <henning.westerholt(a)1und1.de> Hi Henning, I think that the problem is in openser ebuild, so I do this modifications: # diff openser-1.3.1.ebuild openser-1.3.1.ebuild.old 7c7 < IUSE="ipv6 mysql radius postgres jabber ssl odbc perl perlvdb" --- 58,63d57 < use perl && < modules="${modules} perl" < < use perlvdb && < modules="${modules} perlvdb" < --------------------------------------- Hi Yazid, please always CC at the list, so that others can read the resolution too. Great that it works now for you. I'm not sure if the ebuild that is included in the openser distribution is also used in gentoo. Perhaps you can file a bug against the gentoo package (on gentoo.org), and attach your patch, so that the maintainer could fix this? Cheers, Henning

2 1

[OpenSER-Users] Installing perl module
by Yazid Hadj Said 30 May '08

30 May '08

Hi, I use 1.3.1 openser version, and i don't find perl module (I don't know Why). My question is if I can add perl module without recompiling openser. (I use Gentoo distribution) Regards Yazid

2 1

[OpenSER-Users] variables in regexp
by Yazid Hadj Said 30 May '08

30 May '08

Hi, please I want to know: If and How I can use variables in "regexp" . example: avp_subst("avp(s:var1)/$avp(s:var2)","/(.*)$avp(s:var3)(.*)/\1/") In this example, simply the "$avp(s:var3)" string is matched, not the $avp(s:var3)'s content. I am very need this function. Thanks in advance, Yazid

3 2

[OpenSER-Users] Correction in documentation of tm module(version 1.2.X)
by Ruchir Brahmbhatt 30 May '08

30 May '08

Hi, The following sentence in section 1.4.8. t_flush_flags() of tm module documentation(version 1.2.X) has several spelling mistakes. Incorrect Flush the flags from current request into the already created transaction. It make sens only in routing block if the trnasaction was created via t_newtran() and the flags have been altered since. Correct Flush the flags from current request into the already created transaction. It make sense only in routing block if the transaction was created via t_newtran() and the flags have been altered since. Thanks, Ruchir

3 3

[OpenSER-Users] Openser not processing requests at rate being sent to it
by Brian Peters 30 May '08

30 May '08

I am doing what I thought was a simple volume test that OpenSER should be able to handle, but the results I am getting so far show it is not handling the load I'm sending it. Here is my configuration: - OpenSER 1.3.1 - Simple script that uses OpenSER as a proxy to forward requests to a carrier. All it does is rewrite the host and port of the R-URI and relays to the carrier. - children=16 - shared memory set to 1 G. (-m 1000) - using transaction module to handle the requests. - Sending 10 INVITE requests per second to OpenSER using Asterisk. When I do a "tcpdump" trace of the network traffic to/from OpenSER, it is clear that a large percentage of the INVITE requests sent to OpenSER were not handled. Asterisk needed to resend the INVITE requests several times before OpenSER would handle them, and in about 20% of the cases Asterisk terminated the INVITES because it reached the timeout of 30 seconds without any response. If I go directly from asterisk to the carrier, this test works without any issues. I'm not sure what the limitations are of OpenSER, but this seems like something it should be able to handle. Any help or information on how to configure OpenSER so it will work would be appreciated. Thanks, Brian.

6 17

[OpenSER-Users] openser trunk performance
by jh＠tutpro.com 30 May '08

30 May '08

also i did some performance tests on openser trunk in order to clear doubt that has been expressed recently. with only 4 child processes and several radius/mysql queries per invite, i was able to achieve 50 new calls per second using sipp including radius accounting to database of each call. this was on my two years old laptop and everything was running on the same host. i didn't try yet how many calls i could have achieved before retransmissions. i'm satisfied with the result and don't think that there are any performance issues with openser trunk. -- juha

3 2

[OpenSER-Users] Corrections in rr module documentation version 1.2.X
by Ruchir Brahmbhatt 30 May '08

30 May '08

Hi, I've noticed some mistakes in documentation of rr module in version 1.2.X. 1.5.6. is_direction(dir): Incorrect sentence: As for checking it's used the "ftag" Route header parameter, the append_fromtag (see Section 1.4.2 module parameter must be enables. Also this must be call only after loose_route() (see Section 1.5.1). Correct sentence: As for checking it's used the "ftag" Route header parameter, the append_fromtag (see Section 1.4.2 module parameter must be enabled. Also this must be called only after loose_route() (see Section 1.5.1). Thanks, Ruchir Brahmbhatt Director Ecosmob Technologies Pvt. Ltd.

2 1

[OpenSER-Users] Scrolling through resultset
by mark morreny 30 May '08

30 May '08

Hi, If I want to walk through a resultset of numbers, how can I do that? I have tried: $var(index) = 1 while (is_avp_set($var(i:$index) ) $var(index) = $var(index) +1; However $var(i:$index) won't work. I am wondering if someone can tell me how I can correct it. The other question I have is about psuedo-variable. Is there any psuedo-variable can give me the IP of the Openser server? Thanks, Mark

2 1

[OpenSER-Users] Explanation of 0x02 in "t_relay()" ?
by Iñaki Baz Castillo 30 May '08

30 May '08

Hi. I read in "t_relay" doc: -------- 0x02 - do not internally send a negative reply in case of failure. It applies only when the transaction is created. By default one is sent. Useful if you want to implement a serial forking in case of failure. -------- Could you please explain it a little more? I use failure_route with append_branch and before it OpenSer doesn't send back a negative reply to caller (in fct it would end the transaction). What the purpose of this bit option? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

3 3

[OpenSER-Users] Need help with proxy_authorize for NATed UA
by mark morreny 30 May '08

30 May '08

Hi, I am using the same UA and Openser to do two tests. For the first case, UA is pointing to the external IP of openser.; the other test points the UA to openser's local ip. Registering is ok for both cases, but when I try to do an INVITE, the UA pointing to the local IP works whereas the one pointing to the external IP fails during proxy_authorize. 88.251.45.4 -> external ip of the entire network 192.168.1.104 -> openser's local ip 192.168.1.100 -> UA's local ip Here is the INVITE with authentication digest from the UA pointing to the external IP address of openser: U 88.251.45.4:59864 -> 192.168.1.104:5060 INVITE sip:1005@88.251.45.4 <sip%3A1005(a)88.251.45.4> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.100:17480 ;branch=z9hG4bK-d87543-ea31ca6af52bd14c-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:1006@88.251.45.4:59864>. To: "1005"<sip:1005@88.251.45.4 <sip%3A1005(a)88.251.45.4>>. From: "1006"<sip:1006@88.251.45.4 <sip%3A1006(a)88.251.45.4>>;tag=d21e4745. Call-ID: NGFlOTRmNTg3ZDFlYzJlYzdhNTlhNzZmMDRlOTJmOGY.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: Digest username="1006",realm="88.251.45.4 ",nonce="483edcaf82beda80a0c91fc75f486b8fd3f7bbc3",uri="sip:1005@88.251.45.4<sip%3A1005(a)88.251.45.4> ",response="4423a99e44bb7266912de3a4af752b3f",cnonce="ece582bf4b1708b67bd1e86952313a63",nc=00000001,qop=auth,algorithm=MD5. User-Agent: X-Lite release 1011s stamp 41150. Content-Length: 423. . v=0. o=- 1 2 IN IP4 192.168.1.100. s=CounterPath X-Lite 3.0. c=IN IP4 192.168.1.100. t=0 0. m=audio 34842 RTP/AVP 107 119 100 106 0 105 98 8 101. a=alt:1 1 : bIn/n4cp vm9zXwFh 192.168.1.100 34842. a=fmtp:101 0-15. a=rtpmap:107 BV32/16000. a=rtpmap:119 BV32-FEC/16000. a=rtpmap:100 SPEEX/16000. a=rtpmap:106 SPEEX-FEC/16000. a=rtpmap:105 SPEEX-FEC/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. Here is the INVITE resend from the UA pointing to openser's local IP: U 192.168.1.100:50314 -> 192.168.1.104:5060 INVITE sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> SIP/2.0. Via: SIP/2.0/UDP 192.168.1.100:50314 ;branch=z9hG4bK-d87543-c34be44e3704d673-1--d87543-;rport. Max-Forwards: 70. Contact: <sip:1006@192.168.1.100:50314>. To: "1005"<sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104>>. From: "1006"<sip:1006@192.168.1.104 <sip%3A1006(a)192.168.1.104> >;tag=d4048226. Call-ID: YTc4OTE2NmIyMWRkMjk3ZGNmYmY4OTg3NTQ5NWRhZDI.. CSeq: 2 INVITE. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO. Content-Type: application/sdp. Proxy-Authorization: Digest username="1006",realm="192.168.1.104 ",nonce="483ede954fef336afd24141e99d516b0b7d9091a",uri=" sip:1005@192.168.1.104 <sip%3A1005(a)192.168.1.104> ",response="1b77bf191fef631049bd6d6b7dcc9094",cnonce="e96a7e7d9677c4c41952bce7ac2a36e2",nc=00000001,qop=auth,algorithm=MD5. User-Agent: X-Lite release 1011s stamp 41150. Content-Length: 423. . v=0. o=- 7 2 IN IP4 192.168.1.100. s=CounterPath X-Lite 3.0. c=IN IP4 192.168.1.100. t=0 0. m=audio 14816 RTP/AVP 107 119 100 106 0 105 98 8 101. a=alt:1 1 : SP8QJ5e9 96jlnBkg 192.168.1.100 14816. a=fmtp:101 0-15. a=rtpmap:107 BV32/16000. a=rtpmap:119 BV32-FEC/16000. a=rtpmap:100 SPEEX/16000. a=rtpmap:106 SPEEX-FEC/16000. a=rtpmap:105 SPEEX-FEC/8000. a=rtpmap:98 iLBC/8000. a=rtpmap:101 telephone-event/8000. a=sendrecv. The openser script associated wit proxy_authorize is : if (!proxy_authorize("","subscriber")) { xlog("not authorize need to do proxy challenge\n"); proxy_challenge("","1"); exit; } What could be wrong with my setup? Thanks in advance for your help. Thanks, Mark

2 1

[OpenSER-Users] Problems with (RE-)REGISTER
by Lucio Maciel 30 May '08

30 May '08

Hi, I am having a situation where the client sends each RE-REGISTER with a new Call-ID, causing it to create a new entry in the location table. The problem is that this client sends the Re-REGISTER in a Expires/2 time, and as the RFC says, the 200ok is sent with the 2 contacts, with each expire, then the client receive this 2 contacts with 2 expires and "thinks" that it should renew the REGISTER for the old contact and sends a new REGISTER, with a new Call-ID, and adds another entry in the location table, which causes the response returns 3 contacts with 3 expires, and so on... There is some way to handle this issue? Thanks Lucio

2 1

[OpenSER-Users] how to use AVP variables in regular expressions
by Yazid Hadj Said 30 May '08

30 May '08

Hi, Please, I want to know how to use AVP variables in regular expressions with avp_subst? Thanks Yazid

2 1

[OpenSER-Users] Question about proxy_authenticate
by mark morreny 30 May '08

30 May '08

Hi, I am wondering if someone could help me to understand a bit more about the internal mechanism of proxy_challenge? If the UA is connecting to the server using the external ip, proxy_challenge fails, but if the UA is connecting to the server using the lan ip, proxy_challenge succeeds. What is the reason why this is happening? How does proxy_challenge work exactly? Thanks in advance for your help. Regards, Mark

2 1

[OpenSER-Users] Unable to t_relay (ERROR:tm:update_uac_dst)
by Pete Kay 30 May '08

30 May '08

Hi, After next_gw(), openser appends "64.231.23.234:5060;transport=tcp" as the domain of th ruri. When doing a t_relay with that ruri, I am getting ERROR:tm:update_uac_dst. before relay ruri = sip:86143455048@64.231.23.234:5060;transport=tcp starting to relay May 29 21:18:04 [12107] ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 2 (no corresponding listening socket) May 29 21:18:04 [12107] ERROR:tm:t_forward_nonack: failure to add branches However, if I do a t_relay with ruri= "sip:86143455048@64.231.23.234<sip%3A86143455048(a)64.231.23.234>", then the call can go through. Could anyone tell me why? A workaround that I can think of is to reconstruct the ruri to get rid of the port and the "transport=tcp" with something like: rewriteuri("sip:$rU@$rd"); But, the above line gives "sip:$rU@$rd" instead of the desired " sip:86143455048@64.231.23.234 <sip%3A86143455048(a)64.231.23.234>". Is there anyway of using psuedo variable in rewriteuri? Thanks alot in advance for your help. Regards, Pete

5 4

[OpenSER-Users] Managing MSILO responses
by Saúl Ibarra 30 May '08

30 May '08

Hi all, I'm having a little trouble setting up msilo: When I send a MESSAGE to an offline user, MSILO does it's job and stores it, and when the other user gets registered, he sees the MESSAGES. But if I enable onification, by putting: modparam("msilo", "registrar", "sip:softswitch@OPENSER_IP") MSILO repplies me sayng the user is offline, but hten I can't continue sending MESSAGEs, because they will be directed to the URI indicated by the registyrar parameter instead of the original user I wanted to send the MESSAGE. I'm using Twinkle 1.2. Thanks in advance!! -- Saúl -- "Nunca subestimes el ancho de banda de un camión lleno de disketes." ---------------------------------------------------------------- http://www.saghul.net/

2 2

[OpenSER-Users] BYE not relaying properly
by Ruchir Brahmbhatt 30 May '08

30 May '08

Hi All, I have OpenSER 1.2.3(previously it was 1.2.1 but installed latest version to make sure that it is not an OpenSER bug) configured and running for following features: * Registrar * Proxy * Media proxy for NAT traversal * LCR module for PSTN termination * SEMS for voicemail * Multi-leg accounting with Radiator radius * Call forwarding using user_preferences I'm having one really strange issue. My OpenSER is configured to listen on particular IP address say 1.2.3.4. I've added alias for the domain name say voip.domain.com. User agent register to OpenSER using voip.domain.com as a proxy. They can call each other & pstn. I'm using Media Proxy for handling media in case of clients behind NAT. Audio & signaling seems to be working fine in case of internal and pstn calling. I have configured call forwarding unconditional, no answer & busy. In case of unconditional call forwarding, audio and signaling works fine. But when I test call forwarding on no answer or busy, audio is proper but signaling doesn't work properly. Specially BYE. Here is my setup: * 3 users configured; 1000, 2000 & 3000. * 2000 has no answer forwarding to pstn number. * 3000 has busy forwarding to 2000 If 1000 calls 2000 and 2000 doesn't pickup the phone in 30 seconds(configured using fr_inv_timer param of tm module), the pstn number is pushed in request uri and request is sent to gateway for pstn termination. Audio works fine. If caller(1000) hangs up, the callee is hung up properly. However if callee (pstn number) hangs up the call, the caller doesn't receive the BYE message and hence stays connected. Media Proxy session is closed properly but in OpenSER, BYE request is not sent to caller but keeps looping in OpenSER instead. I have several doubts about this issue. One thing is that the pstn gateway is sending BYE to OpenSER IP address instead of domain name. I know this kind of problem can occur if we haven't configured alias parameter properly. Reference: http://www.openser.org/mos/view/-OpenSER-Installation-Notes/ (Section 4. Troubleshooting) I have added ip and domain in domain table as well but still no change. I've been facing this issue since several days and still not able to solve. I've done many OpenSER setups but having such issue for the first time. I've compared the OpenSER config file to my older setups but it seems proper. Can anyone please guide me in correct direction? Thanks, Ruchir Brahmbhatt

1 2

[OpenSER-Users] Can openser auto-retry if a call fails?
by Pete Kay 29 May '08

29 May '08

Hi, I know this kind of function exists in Asterisk, but I want to know if Openser can do it as well. Let's say there are two fail-over numbers(DID-b,DID-c) that associates with a DID(DID-a), if the original number(DID-a) that get dailed can't rearch the destination due to BUSY, NO RESPONSE, etc, is there anyway of having Openser to auto-retry with the next DID(DID-b)? If the next one fails, try the next next one(DID-c). One way I can think of is to do a lookup from DB for the alias in the fail_on_route section, but if I have multiple aliases, then I will have no way of *remembering* which alias has been tried unless I store it in the DB which is not a good solution. I am wondering if there any other way of doing it? Thanks in advance for all your kind suggestion. Regards, Pete

4 6

[OpenSER-Users] OPENSER Dinner Berlin, May 28
by Daniel-Constantin Mierla 29 May '08

29 May '08

Hello, bound to Asterisk Tag (http://www.asterisk-tag.org) and Linux Tag (http://www.linuxtag.org) events, Henning Westerholt and I will take care of an OPENSER group meeting in Berlin, Germany, May 28 at 20:00. The place is a small brewery next to Chartlottenburg Castle in Berlin, named Lusinen Brau, the website: http://www.luisenbraeu.com/ Organized in same idea of casual community meeting, everybody is welcome to join at any time. It is recommended to contact me or Henning to announce your participation so you got your seat reserved. Cheers, Daniel -- http://www.asipto.com

2 1

[OpenSER-Users] t_relay() issue when bad host name
by Iñaki Baz Castillo 29 May '08

29 May '08

Hi, perhaps I missing something but AFAIK if I do: rewritehostport("kk.kkkkkkkkkkaaa.bad"); append_branch(); xlog("L_INFO", "1111111111\n"); if (!t_relay()) { xlog("L_ERROR","222222222222\n"); } xlog("L_ERROR","3333333333\n"); Then, since "kk.kkkkkkkkkkaaa.bad" is a non existing domain I expect to see "222222222" because t_relay should reply a negative value: -4 bad destination (unresolvable address) But the fact is that I don't see that "2222222222", neither "3333333333" but this error: ---------- May 29 10:38:32 [16795] CRITICAL:core:mk_proxy: could not resolve hostname: "kk.kkkkkkkkkkaaa.bad" May 29 10:38:32 [16795] ERROR:tm:uri2proxy: bad host name in URI <sip:number_doesnt_exist@kk.kkkkkkkkkkaaa.new;transport=UDP;user=phone> May 29 10:38:32 [16795] ERROR:tm:t_forward_nonack: failure to add branches ----------- Why? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

2 2

[OpenSER-Users] Variables in regular expression
by Yazid Hadj Said 29 May '08

29 May '08

Hi Can I use AVP variables in regular expression? regards Yazid.

2 1

[OpenSER-Users] presence module - force subscription termination
by Sigrid Thijs 29 May '08

29 May '08

Hi, is it possible to terminate all presence subscriptions for a certain presentity through the current config script (1.3.x branch)? Or is there another easy way to accomplish this, without modifying the authorization rules? kind regards, Sigrid

2 1

[Serusers] Presence-rule handling in SER2.0
by pravin.swami＠wipro.com 29 May '08

29 May '08

Dear All, We are having problems with presence-rules handling of SER 2.0 Presence Server. +------+ +----------+ +------+ | | | Presence | | | | UE1 |--------------| Server |--------------| UE2 | | | | (PS) | | | +------+ +----------+ +------+ 1. UE1 & UE2 subscribe to "watcher-info". 2. UE1 SUBSCRIBEs to UE2's "presence" information. 3. PS sends info winfo-NOTIFY to UE2 (with UE1 status as pending). 4. UE2 fetches its presence-rules.xml document(from path /var/simulated-xcap-root/pres-rules/users/UE2/presence-rules.xml) 5. UE2 updates presence-rules.xml by adding UE1 into ALLOWLIST section & PUTs the document back to PS. 6. User changes presence status at UE2. UE2 sends PUBLISH request to PS. At this stage, PS should trigger presence NOTIFY to UE1. This isn't happening. (Another thing we observe is: Everytime watcher subscription is refreshed, UE2 receives NOTIFY with UE1 status as pending) Are we missing something in the above callflow? Should UE2 send any SIP/HTTP request after HTTP PUT presence-rules.xml ? Are we missing something when building presence-rules.xml ? We have modified the ser.cfg for presence authorization as per ser presence Handbook. modparam("pa", "auth", "xcap") modparam("xcap", "xcap_root", "http://localhost/xcap-root") Should I make any more changes to ser.cfg We have attached the ser.cfg & presence-rules.xml file that we are using. Please help us in solving this problem. Regards, -Pravin Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

[OpenSER-Users] how to force free-radius to use my own database
by Do Nguyen Ha 29 May '08

29 May '08

hi i have question on freeRadius i create my own database and now i want to use my own database instead of Openser database if yes, please show me which file i should edit Thanks Ha`

2 1

[OpenSER-Users] multiples rows in avp_db_query
by Yazid Hadj Said 29 May '08

29 May '08

Hi, When I use avp_db_query(), I only can access to to the first row by using simply the name of my variable ($avp(s:example)), but if I try to use for example ($avp(s:example[1],$avp(s:example[2]) , it indicate me that this variables isn't set. Regards Yazid

2 1

Re: [OpenSER-Users] Problem with Openser-freeradius communication [SOLVED]
by Pete Kay 29 May '08

29 May '08

Hi, I just figured it out. In the radgroupcheck, I had Local as my Value and then I changed it to "Digest", it is working fine now. Thanks a million for all of your help. Regards, Pete

2 1

Re: [OpenSER-Users] Problem with Openser-freeradius communication
by Dan-Cristian Bogos 29 May '08

29 May '08

Hi Pete, the Service-Type does matter only if u use it in your auth part. The reason you see "IAPP-Register" is due to radius dictionaries, responsible for translating between numbers and attributes (radius only transports numbers). Regarding the problem of FreeRADIUS not authorizing the requests I think it is a matter of which module do u want to perform that and configure that properly. I would recomed you should running freeradius in debug mode (-X ) and post more info here. Cheers, DanB On Wed, May 28, 2008 at 10:02 AM, Pete Kay <petedao(a)gmail.com> wrote: > Hi, > I am having problem with configuring FreeRadius to authenticate suername > and password. I have tried to test freeradius with a test client and it > works by sending the following message: > Ready to process requests. > User-Name = "1005(a)192.168.1.104" > User-Password = "1234" > Service-Type = Authenticate-Only > NAS-Port = 0 > NAS-IP-Address = 127.0.0.1 > > The message that is sent by Openser is as follows: > Waking up in 1.9 seconds. > User-Name = "1005(a)192.168.1.104" > Digest-Attributes = 0x0a0631303036 > Digest-Attributes = 0x010f3139322e3136382e312e313034 > Digest-Attributes = > 0x022a34383364376265396630623738663734363935343562386366373865303561316161623032633366 > Digest-Attributes = 0x04137369703a3139322e3136382e312e313034 > Digest-Attributes = 0x030a5245474953544552 > Digest-Response = "eb5f24a5563e9c1096855ef8b2c72d2b" > Service-Type = IAPP-Register > X-Ascend-PW-Lifetime = 825241654 > NAS-Port = 5060 > NAS-IP-Address = 127.0.0.1 > > Freeradius can't find the password and is reject users as a result of > that. Also, is the Service-Type supposed to be Authenticate-Only instead? > Is this something to do with my openser.cfg setup that is causing the > problem? > > Thanks in advance for your help. > > Regards, > Pete > > > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > >

3 9

[OpenSER-Users] Problem with t_relay after next_gw()
by Pete Kay 29 May '08

29 May '08

Hi, After running next_gw(), I can see that the ruri is being modified with the gateway ip as the domain. However, when doing a t_relay, I am getting the following error. From ngrep, I don't see any going package to the gateway ip address, so I am suspecting that the error is internal. before relay ruri = sip:86143455048@64.231.23.234:5060;transport=tcp starting to relay May 29 21:18:04 [12107] ERROR:tm:update_uac_dst: failed to fwd to af 2, proto 2 (no corresponding listening socket) May 29 21:18:04 [12107] ERROR:tm:t_forward_nonack: failure to add branches Could someone tell me what could be the problem? Thenks in advance for all your help. Regards, Pete

1 0

[OpenSER-Users] How to differentiate auth level for gateway
by mark morreny 28 May '08

28 May '08

Hi, Assuming I have 3 gateways, GW-a, GW-b, and GW-c. All of them can REGISTER without username and password by having an entry in the trusted table. Is it possible to config Opernser so that GW-a and GW-b can route to GW-c (using INVITE) without username and password, but any INVITEs from GW-c will not be accepted? Thanks in advance for your suggestion. Regards, Mark

2 1

[OpenSER-Users] CDRTool question - ShowPrice command changes
by Adrian Georgescu 28 May '08

28 May '08

Dan, you are right and I found the problem. The fix will be available with the next CDRTool release due next week together with Mediaproxy 2.0 Adrian If you need a fix now for your 6.4.1: --- rating_lib-6.4.1.phtml 2008-05-28 18:25:18.000000000 +0200 +++ rating_lib.phtml 2008-05-28 18:27:00.000000000 +0200 @@ -6078,26 +6078,28 @@ $timestamp=time(); } - $application="audio"; - if ($NetFields['application']) $application=$NetFields ['application']; + $CDRStructure=array ( + $this->CDRS->CDRFields ['callId'] => $NetFields['callid'], + $this->CDRS->CDRFields ['aNumber'] => $NetFields['from'], + $this->CDRS->CDRFields ['CanonicalURI'] => $NetFields['to'], + $this->CDRS->CDRFields ['duration'] => $NetFields['duration'], + $this->CDRS->CDRFields ['timestamp'] => time() + ); + + $CDR = new $this->CDRS->CDR_class(&$this->CDRS, & $CDRStructure); + $CDR->normalize(); $Rate = new Rate($this->settings, $this->db); $RateDictionary=array( - 'callId' => $NetFields ['callid'], - 'timestamp' => $timestamp, - 'duration' => $NetFields ['duration'], - 'inputTraffic' => $NetFields ['inputtraffic'], - 'outputTraffic' => $NetFields ['outputtraffic'], - 'DestinationId' => $NetFields ['destinationid'], - 'BillingPartyId' => $NetFields ['from'], - 'domain' => $NetFields ['domain'], - 'gateway' => $NetFields ['gateway'], - 'aNumber' => $NetFields ['from'], - 'cNumber' => $NetFields ['to'], - 'ENUMtld' => $NetFields ['enumtld'], - 'applicationType' => $application, - 'RatingTables' => &$this->CDRS- >RatingTables, + 'duration' => $CDR->duration, + 'callId' => $CDR->callId, + 'timestamp' => $CDR->timestamp, + 'DestinationId' => $CDR- >DestinationId, + 'domain' => $CDR->domain, + 'BillingPartyId' => $CDR- >BillingPartyId, + 'ENUMtld' => $CDR->ENUMtld, + 'RatingTables' => &$this->CDRS- >RatingTables ); $Rate->calculate($RateDictionary);

2 1

[OpenSER-Users] CDRTool question - ShowPrice command changes
by Adrian Georgescu 28 May '08

28 May '08

Dan I have no changed anything related to this. Usually I write in the changelog about such dramatic changes. What did you notice and between which versions? Adrian >>>>>>>>> I have noticed that recently ShowPrice behavior has changed, and since did not find it in Changelog I decided to try my shoot over here . There are no more queries based on destination number but DestinationId. Was the support for destination number lost for good, or only temporarily? If that was done, why is "To=" parameter still mandatory? The presence or absence of it does not make any different in queries, as far as I can see. My version: version CDRTool version 6.4.1 Ta, DanB

2 1

[Serusers] ser_ctl question
by Steven C. Blair 28 May '08

28 May '08

Hello: I'm just starting to make a serious attempt to build a SER v2 system. SER is built, ser_mysql has been executed and the MySQL tables defined. With mysql -user -p I can access the database with ser_ctl add domain I cannot. I get the following error. DB_URI in serctl.conf was changed to reflect the new password for the ser user. Yet I still get the following error. Any ideas why? ser_ctl domain add myschool.edu ser_ctl: (1045, "Access denied for user 'ser'@'localhost' (using password: YES)"): OperationalError Thanks,steve Senior Network Engineer, Information Systems and Computing Networking and Telecommunications , Suite 221A /6228 University of Pennsylvania Voice:215-573-8396 FAX:215-898-9348

1 1

[OpenSER-Users] Save Contact List in Server
by Habib, Yeasin 28 May '08

28 May '08

Hi Everybody, Is there any plug-in available for SIP Communicator to save contact list in the server side. I mean when the SIP Communicator register itself to server the server will save to contact list to its own persistence storage. So the client doesn't need a local copy of contact list. I am using OpenIMSCore and OpenSER. I am just wondering whether this functionality is already available in SIP Communicator or not? If not then how much work has to be done to implement in properly? I am just looking for some idea about this. So that I can make an estimation about this implementation. I am looking forward to hearing from all of you. Best regards Yeasin Habib

2 1

[OpenSER-Users] CDRTool question - ShowPrice command changes
by Dan-Cristian Bogos 28 May '08

28 May '08

Guys, I have noticed that recently ShowPrice behavior has changed, and since did not find it in Changelog I decided to try my shoot over here . There are no more queries based on destination number but DestinationId. Was the support for destination number lost for good, or only temporarily? If that was done, why is "To=" parameter still mandatory? The presence or absence of it does not make any different in queries, as far as I can see. My version: version CDRTool version 6.4.1 Ta, DanB

2 1

[OpenSER-Users] New tables in openser DB
by Yazid Hadj Said 28 May '08

28 May '08

Hi, If I add tables to openser database with an other tool (e.g phpMyAdmin), is it possible to create me some problems? Regards, Yazid

2 1

[Serusers] presence status - issue when Authorization is enabled in SER 2.0.
by pravin.swami＠wipro.com 28 May '08

28 May '08

Hi... All , I am facing problem with SER presence status, when authorization is enabled. B elow is the scenario of my test. 1) PUA1 & PUA2 are two user agents subscribed to SER (2.0) presence server. 2) PUA1 adds PUA2 into its buddly list . PUA2 gets Watcher-info notification from PUA1 with subscription - state pending. 3) PUA2 accepts PUA1 and presence-rules.xml file is updated for PUA2 . In whitelist (Allow List)," <id PUA1(a)ser.org <mailto:PUA1@ser.org> <id> " is added successfully. problem statement : PUA2 status is offline in PUA1 's buddy list. I f PUA2 changes its status (sends PUBLISH with away/on phone/busy activity ) , SER presence sends 200 OK success response , but no notifcation will be generated from server to watcher PUA1. - also when PUA2 logs out and logs in again Wathcer-info subscription will be generated with pending state though presence-rules.xml document was already updated for this. I have modified the ser.cfg for presence authorization below as per ser presence Handbook . modparam("pa", "auth", "xcap") modparam("xcap", "xcap_root", "http://localhost/xcap-root <http://localhost/xcap-root> ") I have attached the ser.cfg file. which i tried ... but doesn't work out. let me know any of your comments. regards, Pravin . Please do not print this email unless it is absolutely necessary. The information contained in this electronic message and any attachments to this message are intended for the exclusive use of the addressee(s) and may contain proprietary, confidential or privileged information. If you are not the intended recipient, you should not disseminate, distribute or copy this e-mail. Please notify the sender immediately and destroy all copies of this message and any attachments. WARNING: Computer viruses can be transmitted via email. The recipient should check this email and any attachments for the presence of viruses. The company accepts no liability for any damage caused by any virus transmitted by this email. www.wipro.com

1 0

[OpenSER-Users] Problem with Openser-freeradius communication
by Pete Kay 28 May '08

28 May '08

Hi, I am having problem with configuring FreeRadius to authenticate suername and password. I have tried to test freeradius with a test client and it works by sending the following message: Ready to process requests. User-Name = "1005(a)192.168.1.104" User-Password = "1234" Service-Type = Authenticate-Only NAS-Port = 0 NAS-IP-Address = 127.0.0.1 The message that is sent by Openser is as follows: Waking up in 1.9 seconds. User-Name = "1005(a)192.168.1.104" Digest-Attributes = 0x0a0631303036 Digest-Attributes = 0x010f3139322e3136382e312e313034 Digest-Attributes = 0x022a34383364376265396630623738663734363935343562386366373865303561316161623032633366 Digest-Attributes = 0x04137369703a3139322e3136382e312e313034 Digest-Attributes = 0x030a5245474953544552 Digest-Response = "eb5f24a5563e9c1096855ef8b2c72d2b" Service-Type = IAPP-Register X-Ascend-PW-Lifetime = 825241654 NAS-Port = 5060 NAS-IP-Address = 127.0.0.1 Freeradius can't find the password and is reject users as a result of that. Also, is the Service-Type supposed to be Authenticate-Only instead? Is this something to do with my openser.cfg setup that is causing the problem? Thanks in advance for your help. Regards, Pete

1 0

[OpenSER-Users] Question on modification A-number & B-number
by Do Nguyen Ha 28 May '08

28 May '08

hi all i read several docs of Openser, i didnt find the information to modify the A-number & B-number does openser have the ability to modify A-number & B-number if yes, please send me the link to this Thanks Ha`

2 1

[OpenSER-Users] SIP Text Message to SMSC
by Dome Charoenyost 27 May '08

27 May '08

Dear All, I use Openser for test SIP Text Message from IP phone to IP phone. it's work perfect :) Look like only openser support this feature light now. Asterisk not support and Freeswitch fail also. (I use yealink ip phone). So now my customer want to relay SIP text Message to PSTN via SMSC Can someone tell me is posible to use openser SMS module send SIP text Message to SMSC ? Best Regards. Dome C.

1 0

[OpenSER-Users] Me too: OpenSER 1.3.2 is unusually slow
by Peter Lemenkov 27 May '08

27 May '08

Hello All! After upgrading to 1.3.2 from 1.3.1 I'm encountered (similar to described in that list already) issue, e.g. openser answers unreliably slow to all messages (~40 seconds delay). DNS lookup related fix doesn't helps. I'll try to find what exactly caused this delay (if it's openser-related issue and not something else). -- With best regards!

3 4

[OpenSER-Users] OpenSER corrects the "Contact" field only one way.
by Höskuldur 27 May '08

27 May '08

Hi I have two user clients behind separate NAT networks. My OpenSER is directly connected to the Internet and has it's own external IP address. Both clients are using STUN to get their external IP address and correct port for SIP messages. They successfully register with Openser, which is using Nathelper module and Mediaproxy. In this case, since the client is behind NAT, it puts the wrong port in the Contact field of a SIP message. As a result, SOME SIP messages from OpenSER are never received by the client. But with the help of Nathelper, OpenSER should correct the port in Contact field, by replacing the NAT port with the src port of a SIP message (UDP pack). The problem is, that it looks like OpenSER only corrects the port in Contact field for client A (The one who initiates the call) but not Client B(The one who's called). For a better explaination, please check out my logs from ngrep. CLIENT A to OPENSER U 2008/05/26 16:53:26.285586 157.157.78.173:54182 -> 212.50.206.33:5060 INVITE sip:1001@212.50.206.33 <sip%3A1001(a)212.50.206.33> SIP/2.0. From: 1006<sip:1006@212.50.206.33 <sip%3A1006(a)212.50.206.33> >;tag=10007500-26260f5b. To: <sip:1001@212.50.206.33 <sip%3A1001(a)212.50.206.33>>. Via: SIP/2.0/UDP 157.157.78.173:62356 ;branch=z9hG4bKc0a8150a10007e0026261023. Contact: 1006<sip:1006@157.157.78.173:62356>. !!!NB!!! <-- This is the WRONG reply port. OPENSER RELAYS TO CLIENT B U 2008/05/26 16:53:26.301864 212.50.206.33:5060 -> 130.208.183.251:49211 INVITE sip:1001@130.208.183.251:49208 SIP/2.0. Record-Route: <sip:212.50.206.33;lr;ftag=10007500-26260f5b;nat=yes>. From: 1006<sip:1006@212.50.206.33 <sip%3A1006(a)212.50.206.33> >;tag=10007500-26260f5b. To: <sip:1001@212.50.206.33 <sip%3A1001(a)212.50.206.33>>. Via: SIP/2.0/UDP 212.50.206.33;branch=z9hG4bK5db1.ce9aea2.0;rport. Via: SIP/2.0/UDP 157.157.78.173:62356 ;rport=54182;branch=z9hG4bKc0a8150a10007e0026261023. Contact: 1006<sip:1006@157.157.78.173:54182>. !!!NB!!! <-- OpenSER detects it and changes it to correct reply port. Works great. CLIENT B ANSWERS TO OPENSER U 2008/05/26 16:53:27.301368 130.208.183.251:49211 -> 212.50.206.33:5060 SIP/2.0 200 OK. From: 1006<sip:1006@212.50.206.33 <sip%3A1006(a)212.50.206.33> >;tag=10007500-26260f5b. To: <sip:1001@212.50.206.33 <sip%3A1001(a)212.50.206.33> >;tag=10007900-26261046. Via: SIP/2.0/UDP 212.50.206.33;branch=z9hG4bK5db1.ce9aea2.0. Via: SIP/2.0/UDP 157.157.78.173:62356 ;branch=z9hG4bKc0a8150a10007e0026261023. Contact: 1001<sip:1001@130.208.183.251:49208>. !!!NB!!! <-- Now lets try Client B. Wrong reply port. OPENSER TO CLIENT A U 2008/05/26 16:53:27.305241 212.50.206.33:5060 -> 157.157.78.173:54182 SIP/2.0 200 OK. From: 1006<sip:1006@212.50.206.33 <sip%3A1006(a)212.50.206.33> >;tag=10007500-26260f5b. To: <sip:1001@212.50.206.33 <sip%3A1001(a)212.50.206.33> >;tag=10007900-26261046. Via: SIP/2.0/UDP 157.157.78.173:62356 ;branch=z9hG4bKc0a8150a10007e0026261023. Contact: 1001<sip:1001@130.208.183.251:49208>. !!!NB!!! <-- OpenSER didn't change the port like before, so now Client B never receives ACK from Client A. CLIENT A TO OPENSER U 2008/05/26 16:53:27.518621 157.157.78.173:54182 -> 212.50.206.33:5060 ACK sip:1001@130.208.183.251:49208 SIP/2.0. From: 1006<sip:1006@212.50.206.33 <sip%3A1006(a)212.50.206.33> >;tag=10007500-26260f5b. To: <sip:1001@212.50.206.33 <sip%3A1001(a)212.50.206.33> >;tag=10007900-26261046. Via: SIP/2.0/UDP 157.157.78.173:62356 ;branch=z9hG4bKc0a8150a10007e0026261505. Contact: 1006<sip:1006@157.157.78.173:62356>. Route: <sip:212.50.206.33;lr;ftag=10007500-26260f5b;nat=yes>. OPENSER TO CLIENT B U 2008/05/26 16:53:27.519192 212.50.206.33:5060 -> 130.208.183.251:49208 !!!NB!!! <-- OpenSER sends to the WRONG port! ACK sip:1001@130.208.183.251:49208 SIP/2.0. Record-Route: <sip:212.50.206.33;lr;ftag=10007500-26260f5b;nat=yes>. From: 1006<sip:1006@212.50.206.33 <sip%3A1006(a)212.50.206.33> >;tag=10007500-26260f5b. To: <sip:1001@212.50.206.33 <sip%3A1001(a)212.50.206.33> >;tag=10007900-26261046. CSeq: 101 ACK. Via: SIP/2.0/UDP 212.50.206.33;branch=z9hG4bK5db1.ce9aea2.2;rport. Via: SIP/2.0/UDP 157.157.78.173:62356 ;rport=54182;branch=z9hG4bKc0a8150a10007e0026261505. This last ACK message is never received by Client B because the client isn't listening on port 49208. If I manually use rewriteport("49211") it works, because obviously this last package is then forced to port 49211 which the client listening on. Also, the first line in my main route is force_rport(); but that still doesn't do it. I've been struggling with this problem for a few weeks now so all comments and suggestions are very welcome. My OpenSER config file can be found here: http://hoski.public.is/openser.cfg Best regards, Höskuldur

2 4

[OpenSER-Users] need help on Radius --
by Do Nguyen Ha 27 May '08

27 May '08

Hi all i am intent to use Radius for prepaid but i need more information of RADIUS, how to limit the call, how the RADIUS work together with Openser could you suggest where i got it Thanks Ha

3 2

[OpenSER-Users] How to set the "radius.seq" file (posible bug) ?
by Iñaki Baz Castillo 27 May '08

27 May '08

Hi, I use FRadius accounting. OpenSer tries to use a file /var/run/radius.seq but it fails when restarting computer since OpenSer init script creates this file as: /var/run/openser/openser_radius.seq create_radius_seqfile () { # Create a radius sequence file to be used by the radius client if # radius accounting is enabled. This is needed to avoid any issue # with the file not being writable if openser first starts as user # root because DUMP_CORE is enabled and creates this file as user # root and then later it switches back to user openser and cannot # write to the file. If the file exists before openser starts, it # won't change it's ownership and will be writable for both root # and openser, no matter what options are chosen at install time RADIUS_SEQ_FILE=/var/run/openser/openser_radius.seq I don't find where to change this file in OpenSer script. Isn't a bug if default value is: /var/run/radius.seq but OpenSer init script (Debian Etch) creates it in: /var/run/openser/openser_radius.seq ? -- Iñaki Baz Castillo ibc(a)in.ilimit.es

1 1

[OpenSER-Users] bridging two existing channels
by mark morreny 27 May '08

27 May '08

Hi, In my scenario, I have UA1 -> Openser -> B2BUA -> Openser -> UA2. The reason why I need B2BUA to originate the channel to UA2 is because it is going to play a wav file when UA2 answers the call. After the call has been answered by UA2 and dialog is in progress, is it possible for B2BUA to notify Openser that it wants to be out of the look, so the scenario can before: UA1 -> Openser -> UA2 without UA2 hanging up? How would I be able to config that? Any help will be greatly appreciated. Thanks, Mark

2 1

[OpenSER-Users] obtain content of header dynamically
by Yazid Hadj Said 27 May '08

27 May '08

Hi, I try to obtain content of header dynamically, but with $hdr(...) i can't do this because it accept only static parameters. For example I try to use: $avp(s:body)=$mb; $avp(s:HdName)="To"; avp_subst("$avp(s:corps)/$avp(s:HdContent)","/^$avp(s:HdName):(.*)\r\n(.*)/\1/"); With this manner, i can change header name ($avp(s:HdName) )dynamically from data source (e.g database) and use his content in my script. But until now i don't obtain what i want. Regards, Yazid

2 1

[OpenSER-Users] Fixing wrong SDP with "textutils" module (LF -> CRLF)
by Iñaki Baz Castillo 27 May '08

27 May '08

Hi, I'm handling with a Nortel CS2K softswitch that sends me incorrect SDP's since it uses LF at the end of each line in SDP instead of CRLF (as it's mandatory according to the SDP grammar). AFAIK some phones can fail when receiving this incorrect SDP. Unfortunatelly I can't try a workaround now but I'm thinkink about using "textmodules" in this way: if ($si == NORTEL_CS2K_IP) replace_body("\n$", "\r\n"); By your experience, should it work? I assume that "Content-Length" is automatically updated before the message leaves OpenSer. So just it, should it work? Thanks a lot. -- Iñaki Baz Castillo ibc(a)in.ilimit.es

3 6

[OpenSER-Users] lcr problem - openser 1.3
by Ion Minzu 27 May '08

27 May '08

hello i use openser 1.3 But I have trouble NOTICE:core:main: version: openser 1.3.2-notls (i386/freebsd) ERROR:lcr:reload_gws: Unknown or unsupported transport <0> CRITICAL:lcr:mod_init: Failed to reload gateways and routes ERROR:core:init_mod: failed to initialize module lcr ERROR:core:main: error while initializing modules ####config openser.cfg##### loadmodule "lcr.so" modparam("lcr", "db_url", "mysql://sqluser:ser@localhost/openser") modparam("lcr", "gw_table", "gw") modparam("lcr", "gw_name_column", "gw_name") modparam("lcr", "ip_addr_column", "ip_addr") modparam("lcr", "port_column", "port") modparam("lcr", "uri_scheme_column", "uri_scheme") modparam("lcr", "transport_column", "transport") modparam("lcr", "grp_id_column", "grp_id") modparam("lcr", "lcr_table", "lcr") modparam("lcr", "strip_column", "strip") modparam("lcr", "prefix_column", "prefix") modparam("lcr", "from_uri_column", "from_uri") modparam("lcr", "priority_column", "priority") modparam("lcr", "gw_uri_avp", "$avp(i:709)") modparam("lcr", "ruri_user_avp", "$avp(i:500)") modparam("lcr", "contact_avp", "$avp(i:711)") modparam("lcr", "fr_inv_timer_avp", "$avp(i:704)") modparam("lcr", "fr_inv_timer", 90) modparam("lcr", "fr_inv_timer_next", 30) modparam("lcr", "rpid_avp", "$avp(i:302)") modparam("lcr", "dm_flag", 25) ################## What is problem? Thanks

2 1

[OpenSER-Users] problem nat, openser 1.3 + mediaproxy +gateway
by Ion Minzu 27 May '08

27 May '08

Hello, I use openser 1.3 + mediaproxy and all work ok. But I have trouble calling to PSTN gateway, where is no voice both ways. route[7] is the route to gateway Thanks ############################################## listen = udp:my.openser.server:5060 mpath = "/usr/local/lib/openser/modules" children = 8 debug = 3 fork = yes #group = "openser" #user = "openser" disable_tcp = no log_facility = LOG_DAEMON log_stderror = no tcp_children = 4 mhomed = no server_signature = yes #sock_group = "openser" #sock_mode = 0600 #sock_user = "openser" #unix_sock = "/tmp/openser.sock" #unix_sock_children = 1 #reply_to_via = no sip_warning = no check_via = no dns = no rev_dns = no disable_core_dump = no dns_try_ipv6 = no dns_use_search_list = yes loadmodule "usrloc.so" modparam("usrloc", "user_column", "username") modparam("usrloc", "domain_column", "domain") modparam("usrloc", "contact_column", "contact") modparam("usrloc", "expires_column", "expires") modparam("usrloc", "q_column", "q") modparam("usrloc", "callid_column", "callid") modparam("usrloc", "cseq_column", "cseq") modparam("usrloc", "methods_column", "methods") modparam("usrloc", "flags_column", "flags") modparam("usrloc", "user_agent_column", "user_agent") modparam("usrloc", "received_column", "received") modparam("usrloc", "socket_column", "socket") modparam("usrloc", "use_domain", 0) modparam("usrloc", "desc_time_order", 0) modparam("usrloc", "timer_interval", 60) modparam("usrloc", "db_url", "mysql://sqluser:ser@localhost/openser") modparam("usrloc", "db_mode", 1) modparam("usrloc", "matching_mode", 0) modparam("usrloc", "cseq_delay", 20) modparam("usrloc", "nat_bflag", 6) loadmodule "textops.so" loadmodule "rr.so" modparam("rr", "enable_full_lr", 0) modparam("rr", "append_fromtag", 1) modparam("rr", "enable_double_rr", 1) modparam("rr", "add_username", 0) loadmodule "tm.so" modparam("tm", "fr_timer", 30) modparam("tm", "fr_inv_timer", 120) modparam("tm", "wt_timer", 5) modparam("tm", "delete_timer", 2) modparam("tm", "noisy_ctimer", 0) modparam("tm", "ruri_matching", 1) modparam("tm", "via1_matching", 1) modparam("tm", "unix_tx_timeout", 2) modparam("tm", "restart_fr_on_each_reply", 1) modparam("tm", "pass_provisional_replies", 0) loadmodule "xlog.so" modparam("xlog", "buf_size", 4096) modparam("xlog", "force_color", 0) loadmodule "mi_fifo.so" modparam("mi_fifo", "fifo_name", "/tmp/openser_fifo") modparam("mi_fifo", "fifo_mode", 0660) #modparam("mi_fifo", "fifo_group", "openser") #modparam("mi_fifo", "fifo_user", "openser") modparam("mi_fifo", "reply_dir", "/tmp/") modparam("mi_fifo", "reply_indent", "\t") loadmodule "domain.so" modparam("domain", "db_url", "mysql://sqluser:ser@localhost/openser") modparam("domain", "db_mode", 1) modparam("domain", "domain_table", "domain") modparam("domain", "domain_col", "domain") loadmodule "nathelper.so" modparam("nathelper", "natping_interval", 30) #modparam("nathelper", "ping_nated_only", 1) #modparam("nathelper", "rtpproxy_sock", "unix:/var/run/rtpproxy.sock") #modparam("nathelper", "rtpproxy_disable", 1) #modparam("nathelper", "rtpproxy_disable_tout", 60) #modparam("nathelper", "rtpproxy_tout", 1) #modparam("nathelper", "rtpproxy_retr", 5) #modparam("nathelper", "sipping_method", "OPTIONS") modparam("nathelper", "received_avp", "$avp(i:801)") loadmodule "sl.so" modparam("sl", "enable_stats", 1) loadmodule "mediaproxy.so" modparam("mediaproxy", "mediaproxy_socket", "/var/run/proxydispatcher.sock") #modparam("mediaproxy", "sip_asymmetrics", "/etc/openser/sip-asymmetric-clients") #modparam("mediaproxy", "rtp_asymmetrics", "/etc/openser/rtp-asymmetric-clients") modparam("mediaproxy", "natping_interval", 60) loadmodule "uri.so" loadmodule "registrar.so" modparam("registrar", "default_expires", 3600) modparam("registrar", "min_expires", 60) modparam("registrar", "max_expires", 0) modparam("registrar", "default_q", 0) modparam("registrar", "append_branches", 1) modparam("registrar", "case_sensitive", 0) modparam("registrar", "received_param", "received") modparam("registrar", "max_contacts", 0) modparam("registrar", "retry_after", 0) modparam("registrar", "method_filtering", 0) modparam("registrar", "path_mode", 2) modparam("registrar", "path_use_received", 0) modparam("registrar", "received_avp", "$avp(i:801)") loadmodule "maxfwd.so" modparam("maxfwd", "max_limit", 256) loadmodule "mysql.so" modparam("mysql", "ping_interval", 300) modparam("mysql", "auto_reconnect", 1) loadmodule "auth.so" modparam("auth", "nonce_expire", 300) modparam("auth", "rpid_suffix", ";party=calling;id-type=subscriber;screen=yes") modparam("auth", "rpid_avp", "$avp(s:rpid)") loadmodule "auth_db.so" modparam("auth_db", "db_url", "mysql://sqluser:ser@localhost/openser") modparam("auth_db", "user_column", "username") modparam("auth_db", "domain_column", "domain") modparam("auth_db", "password_column", "password") modparam("auth_db", "password_column_2", "ha1b") modparam("auth_db", "calculate_ha1", 1) modparam("auth_db", "use_domain", 0) modparam("auth_db", "load_credentials", "rpid") loadmodule "uri_db.so" modparam("uri_db", "db_url", "mysql://sqluser:ser@localhost/openser") modparam("uri_db", "uri_table", "uri") modparam("uri_db", "uri_user_column", "username") modparam("uri_db", "uri_domain_column", "domain") modparam("uri_db", "uri_uriuser_column", "uri_user") modparam("uri_db", "subscriber_table", "subscriber") modparam("uri_db", "subscriber_user_column", "username") modparam("uri_db", "subscriber_domain_column", "domain") modparam("uri_db", "use_uri_table", 0) modparam("uri_db", "use_domain", 0) #for accounting loadmodule "acc.so" modparam("acc", "log_flag", 1) modparam("acc", "log_level", 2) modparam("acc", "db_flag", 1) modparam("acc", "db_missed_flag", 3) modparam("acc", "log_missed_flag", 3) modparam("acc", "db_url", "mysql://sqluser:ser@localhost/openser") modparam("acc", "report_ack", 0) ######################################################################## # Request route 'main' ######################################################################## route[0] { xlog("L_INFO", "New request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); force_rport(); if(msg:len > max_len) { xlog("L_INFO", "Message too big - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("513", "Message Too Big"); exit; } if (!mf_process_maxfwd_header("10")) { xlog("L_INFO", "Too many hops - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("483", "Too Many Hops"); exit; } if(!is_method("REGISTER")) { if(nat_uac_test("19")) { record_route(";nat=yes"); } else { record_route(); } } if(is_method("CANCEL") || is_method("BYE")) { end_media_session(); } if(loose_route()) { if(!has_totag()) { xlog("L_INFO", "Initial loose-routing rejected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Initial Loose-Routing Rejected"); exit; } if(nat_uac_test("19") || search("^Route:.*;nat=yes")) { fix_nated_contact(); setbflag(6); } route(3); } if(is_method("REGISTER")) { route(2); } if(is_method("INVITE")) { route(4); } if(is_method("CANCEL") || is_method("ACK")) { route(8); } route(9); } ######################################################################## # Request route 'stop-media-proxy' ######################################################################## route[1] { if(isflagset(22)) { end_media_session(); } } ######################################################################## # Request route 'base-route-register' ######################################################################## route[2] { sl_send_reply("100", "Trying"); if(!www_authorize("", "subscriber")) { xlog("L_INFO", "Register authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); www_challenge("", "0"); exit; } if(!check_to()) { xlog("L_INFO", "Spoofed To-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed To-URI Detected"); exit; } consume_credentials(); if(!search("^Contact:[ ]*\*") && nat_uac_test("19")) { fix_nated_register(); setbflag(6); } if(!save("location")) { xlog("L_ERR", "Saving contact failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_reply_error(); exit; } xlog("L_INFO", "Registration successful - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Request route 'base-outbound' ######################################################################## route[3] { if(isbflagset(6)) { if(!isflagset(22) && !search("^Content-Length:[ ]*0")) { setflag(22); use_media_proxy(); } t_on_reply("2"); } else { t_on_reply("1"); } if(!isflagset(21)) { t_on_failure("1"); } if(isflagset(29)) { append_branch(); } if(is_present_hf("Proxy-Authorization")) { consume_credentials(); } xlog("L_INFO", "Request leaving server, D-URI='$du' - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); # no 100 (we already sent it) and no DNS blacklisting if(!t_relay("0x05")) { sl_reply_error(); if(is_method("INVITE") && isbflagset(6)) { end_media_session(); } } exit; } ######################################################################## # Request route 'base-route-invite' ######################################################################## route[4] { sl_send_reply("100", "Trying"); if(!proxy_authorize("", "subscriber")) { xlog("L_INFO", "Proxy authentication failed - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); proxy_challenge("", "0"); exit; } if(!check_from()) { xlog("L_INFO", "Spoofed From-URI detected - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("403", "Spoofed From-URI Detected"); exit; } if(nat_uac_test("19")) { fix_nated_contact(); setbflag(6); } route(5); } ######################################################################## # Request route 'invite-find-callee' ######################################################################## route[5] { if(!is_domain_local("$rd")) { setflag(20); route(7); } if(does_uri_exist()) { xlog("L_INFO", "Callee is local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(6); } else { xlog("L_INFO", "Callee is not local - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(7); } exit; } ######################################################################## # Request route 'invite-to-internal' ######################################################################## route[6] { if(!lookup("location")) { xlog("L_INFO", "Local user offline - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Offline"); } else { xlog("L_INFO", "Local user online - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); } exit; } ######################################################################## # Request route 'invite-to-external' ######################################################################## route[7] { if(isflagset(20)) { xlog("L_INFO", "Call to foreign domain - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(3); exit; } #go to pstn if (uri=~"sip:[2-8][0-9][0-9][0-9][0-9][0-9]+@.*") { prefix("400"); t_relay("my.pstn.gateway:5060"); end_media_session(); exit; } #finish xlog("L_INFO", "Call to unknown user - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("404", "User Not Found"); exit; } ######################################################################## # Request route 'base-route-local' ######################################################################## route[8] { t_on_reply("1"); if(t_check_trans()) { xlog("L_INFO", "Request leaving server - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); if(!t_relay()) { sl_reply_error(); } } else { xlog("L_INFO", "Dropping mis-routed request - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); } exit; } ######################################################################## # Request route 'base-route-generic' ######################################################################## route[9] { xlog("L_INFO", "Method not supported - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); sl_send_reply("501", "Method Not Supported Here"); exit; } ######################################################################## # Request route 'base-filter-failover' ######################################################################## route[10] { if(!t_check_status("408|500|503")) { xlog("L_INFO", "No failover routing needed for this response code - M=$rm RURI=$ru F=$fu T=$tu IP=$si ID=$ci\n"); route(1); exit; } } ######################################################################## # Reply route 'base-standard-reply' ######################################################################## onreply_route[1] { xlog("L_INFO", "Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); exit; } ######################################################################## # Reply route 'base-nat-reply' ######################################################################## onreply_route[2] { xlog("L_INFO", "NAT-Reply - S=$rs D=$rr F=$fu T=$tu IP=$si ID=$ci\n"); if(nat_uac_test("1")) { fix_nated_contact(); } if(isbflagset(6) && status=~"(180)|(183)|2[0-9][0-9]") { if(!search("^Content-Length:[ ]*0")) { use_media_proxy(); } } exit; } ######################################################################## # Failure route 'base-standard-failure' ######################################################################## failure_route[1] { route(10); route(1); }

1 0

[OpenSER-Users] Res: Hiding the config code
by Bruno Machado 26 May '08

26 May '08

Hi Andel Yes, it would be possible. But I'd like to hide the code even the root user. Like Bogdan said, I want "way to run OpenSER with script that it is not plain text or accessible for humans". Your suggest works as well when the another users do not have the root password. Anyway, thanks for your comment, Andel. ----- Mensagem original ---- De: Ladislav Andel <ladia6(a)centrum.cz> Para: Bruno Machado <bsmufu(a)yahoo.com.br> Cc: users(a)lists.openser.org Enviadas: Segunda-feira, 26 de Maio de 2008 7:18:00 Assunto: Re: [OpenSER-Users] Hiding the config code How about setting the user rights in your unix-based system ? Lada Bruno Machado wrote: > Hi all > > I have been wondering if is possible to hide the Openser's config file > or do anything else to avoid reading by another people. Possible? > > Tks > > Bruno > > ------------------------------------------------------------------------ > Abra sua conta no Yahoo! Mail > <http://br.rd.yahoo.com/mail/taglines/mail/*http://br.mail.yahoo.com/>, > o único sem limite de espaço para armazenamento! > ------------------------------------------------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/

1 0

[OpenSER-Users] Res: Hiding the config code
by Bruno Machado 26 May '08

26 May '08

Exactly, Bogdan. It would be a good feature for OpenSER considering security purposes. Sometimes, you need to hide the script from some people, even from the same company. I think that it would be a good feature. Thanks Bogdan ----- Mensagem original ---- De: Bogdan-Andrei Iancu <bogdan(a)voice-system.ro> Para: Ladislav Andel <ladia6(a)centrum.cz> Cc: Bruno Machado <bsmufu(a)yahoo.com.br>; users(a)lists.openser.org Enviadas: Segunda-feira, 26 de Maio de 2008 7:48:41 Assunto: Re: [OpenSER-Users] Hiding the config code Well, that's a limited hiding - I understood the question as "if it a way to run OpenSER with script that it is not plain text or accessible for humans. Regards, Bogdan Ladislav Andel wrote: > How about setting the user rights in your unix-based system ? > Lada > > Bruno Machado wrote: > >> Hi all >> >> I have been wondering if is possible to hide the Openser's config file >> or do anything else to avoid reading by another people. Possible? >> >> Tks >> >> Bruno >> >> ------------------------------------------------------------------------ >> Abra sua conta no Yahoo! Mail >> <http://br.rd.yahoo.com/mail/taglines/mail/*http://br.mail.yahoo.com/>, >> o único sem limite de espaço para armazenamento! >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Users mailing list >> Users(a)lists.openser.org >> http://lists.openser.org/cgi-bin/mailman/listinfo/users >> >> > > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/

1 0

Re: [OpenSER-Users] Possible problem with script presence-create.sql
by Florent Le Saout 26 May '08

26 May '08

Hello all, I also found the reason reading precisely the documentation. I use the line modparam("presence_xml", "xcap_table", "xcaps") in my configuration as Anca Vamanu said. All is clear now. Thank for your help. Florent. > Message du 26/05/08 12:20 > De : "Henning Westerholt" <henning.westerholt(a)1und1.de> > A : "Anca Vamanu" <anca(a)voice-system.ro> > Copie à : "Florent Le Saout" <le-saout.florent(a)laposte.net>, users(a)lists.openser.org > Objet : Re: [OpenSER-Users] Possible problem with script presence-create.sql > > On Monday 26 May 2008, Anca Vamanu wrote: > > Indeed the name of the script was wrong in the documentation and I have > > fixed it. > > The name of the xcap table was not misspelled as 'xcaps'. > > /The module requires 1 table in OpenSER database: xcap. / > > You probably refer to the line with the example of how to set the xcap > > table name module parameter: > > / 'modparam("presence_xml", "xcap_table", "xcaps")'. / > > This should be used if you what the table to have a different name, and > > of course, you should use the same when creating the database table. > > Thank you Anca for the clarification, and the fix. > > Henning > >

1 0

[OpenSER-Users] openSER with freeRADIUS
by Patrick Pichler 26 May '08

26 May '08

Hi everybody, does anyone has got a tested documention on setting up freeRADIUS to work together with openSER. I'm currently having problems with the authentication via a MySQL DB. I didn't get it how to tell freeRADIUS to take the users credentials from the "subscribers" table. Could anybody help me dealing eith this problem? My environment: openSER with radiusclient-ng, mysql and freeradius on a single machine, all are up and running so far. Without RADIUS it's prossible for a UAC to register...but with RADIUS authentication via MySQL the UAC just gets a "401 unauthorized". Thanks in advance. Patrick

2 3

[OpenSER-Users] [OT] Why the project name in SourceForge is "Scalable SIP server"?
by Iñaki Baz Castillo 26 May '08

26 May '08

Hi, if you get into: https://sourceforge.net/projects/openser/ you won't read the word "OpenSER" but "Scalable SIP server". Also, in the "location" line we read: SF.net > Projects > Scalable SIP server > Summary Scalable SIP server I wonder: why not name it as OpenSER? -- Iñaki Baz Castillo ibc(a)in.ilimit.es

3 4

[OpenSER-Users] Hiding the config code
by Bruno Machado 26 May '08

26 May '08

Hi all I have been wondering if is possible to hide the Openser's config file or do anything else to avoid reading by another people. Possible? Tks Bruno Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento! http://br.mail.yahoo.com/

3 3

[OpenSER-Users] serweb-0.9.6 on openser-1.3.2
by Zen Kato 26 May '08

26 May '08

I installed serweb-0.9.6 on openser-1.3.2, but when I enter username and password, the following Warning comes out. How should I solve this problem? -------------------------------------------- http://localhost/openserweb/user/index.php username:0103@192.168.0.3 password:xxxxx --------------------------------------------- ----------------------------------------------------------------------------- http://localhost/openserweb/user/my_account.php?kvrk=483a8e78876b5 ( ! ) Warning: fopen(/tmp/openser_fifo) [function.fopen]: failed to open stream: Permission denied in /var/www/openserweb/serweb-0.9.6/html/functions.php on line 345 ----------------------------------------------------------------------------- /var/www/openserweb/serweb-0.9.6/html/functions.php (L:344-348) /* open fifo now */ $fifo_handle=fopen( $config->fifo_server, "w" ); if (!$fifo_handle) { $errors[]="sorry -- cannot open write fifo"; return false; } -- Zen Kato

1 0

[OpenSER-Users] Possible problem with script presence-create.sql
by Florent Le Saout 26 May '08

26 May '08

Hello, I'm a new user of openser. (I use openser 1.3.1-3+b1 from package of debian testing) I want to use the module of presence. If I refer to documentation I read that I have to use presence_xml-create.sql to create the tables. But I can't find it. I use the script presence-create.sql which seems to replace it. I create a table named xcap but the module need xcaps, if I create the same table renamed xcaps that work well. I think there is a mistake of naming of the table xcap(s). I take a look to 1.3.x of your repository and I see the same thing in the sql script. Thank you for your job. Best regards. Florent le Saout. Créez votre adresse électronique prénom.nom(a)laposte.net 1 Go d'espace de stockage, anti-spam et anti-virus intégrés.

3 3

[OpenSER-Users] Acceptor peer failed establishing state[3003], closing
by bhargav.b＠renovau.com 26 May '08

26 May '08

actually i am authenticating a user using openser and opendiameters NASREQ application as the client and server.at present the openser is sending the all needed avp's to opendiameter client.i can see all the avp code and avp data on the client side. my problem is that the on the diameter server side the avp,s are not transferred by the client. the client is giving something like (6372|1698151312) Acceptor peer failed establishing state [3003], closing i am confused about it.by reading some drafts i came to know that the error is due to realm mismatch.but i am sending the same realm as the opendiameter's realm.in both the cases it is [bhargav.s]. before i sent the avp's from openser side there was no such problem showing on the client side.if anybody know the solution please help me regards bhargav.b(a)renovau.com

2 1

[OpenSER-Users] How to determine that a reply is a retransmission?
by Jon Steer 26 May '08

26 May '08

Environment: Openser 1.3.1-TLS Fedora 7 I have a module that performs a heavyweight operation on Invites and their respective Replys. I would prefer not to do the operation again if there is a retranmission of the Reply. How can I determine in a REPLY-ROUTE if a reply has already been seen? $T_reply_code would work, if on entry to the first reply route it was zero, but I suspect it is the value of the current reply. thanks, jon

2 1

[OpenSER-Users] Centos 5.1 or Debian4.0.r0 for Openser server
by fatmeh fatah 26 May '08

26 May '08

Hello: May you comparison Centos 5.1 and Debian4.0.r0 for Openser server? Thank you.

3 2

[OpenSER-Users] Click to Call functionality
by Darren Sessions 26 May '08

26 May '08

Quick question . . >From what I've been able to gather, the fifo db functionality has been removed since 1.2.x. What is another easy way to do click-to-call functionality now that it is gone? Thanks, - Darren

2 1

Re: [OpenSER-Users] Problems with sl module in openser1.3.1
by Arturo Ochoa 26 May '08

26 May '08

On Friday 23 May 2008, arturo arturo wrote: > Could some help me with this problem, I'm trying to run openser 1.3.1, but > it dies complaining about the sl module. I issued just 'make all' command > before this > > root@zeus:/home/aochoa/openser/openser-1.3.1-notls > # ./openser > May 22 15:57:16 [4482] DBG:core:yyparse: loading module > /usr/local/lib/openser/modules/mysql.so > May 22 15:57:16 [4482] DBG:core:yyparse: loading module > /usr/local/lib/openser/modules/sl.so > May 22 15:57:16 [4482] ERROR:core:version_control: *module compile flags > mismatch for* /usr/local/lib/openser/modules/sl.so > core: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, > SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > module: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, > USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT Hi Arturo, the version of the sl module installed on your machine don't match the one of your server in the home directory. Do a make distclean; make all in the source directory, and install again. Cheers, Henning Thanks for your response Henning, I've already do what you mention, but the problem persists. I tried also with openser 1.3.2, and the same thing. Also I noted that there are different linked libs on the same module but compiled on different machines: Fedora: [root@apolo] /usr/local/lib/openser/modules # ldd sl.so linux-gate.so.1 => (0x0028b000) libc.so.6 => /lib/libc.so.6 (0x00111000) /lib/ld-linux.so.2 (0x0028c000) Centos: (this is the one with the problem) root@zeus:/home/aochoa/openser # ldd /usr/local/lib/openser/modules/sl.so libc.so.6 => /lib/tls/libc.so.6 (0x00149000) /lib/ld-linux.so.2 (0x003f7000) Any comments?

2 1

[OpenSER-Users] CANCEL not always beeing relayed
by Tobias Lindgren 26 May '08

26 May '08

Hi all, I have some problems with CANCELs not always beeing relayed from A to B via an OpenSER 1.3.1. When I try to make a call and CANCEL it with my own equipment, everything works just fine. OpenSER receives CANCEL and passes it on to the PBX. However, some customer calls that are beeing CANCELed are not always relayed by OpenSER and I cannot find out why. They are received by OpenSER but never sent forward. I have sip-logs and OpenSER debugs, are there someone with more understanding of the OpenSER-code that can take a look at them? Br, /Tobias

2 1

Re: [OpenSER-Users] call failed 403 forbiden
by luzango mfupe 26 May '08

26 May '08

Hi mates, I still need your pointers regarding my problem in this post, today i have attached the routes suspected to be involved in this saga. From my config file plz see below. # ----------------------------------------------------------------- # Unauthorized relay # ----------------------------------------------------------------- if (!is_uri_host_local()) { if (is_from_local()) { # We prevent unauthorised relays "clever guys we got u" append_hf("P-hint: outbound\r\n"); route(10); # need to be authenticated } else { sl_send_reply("403", "Forbidden"); }; return; }; route[10] { #------------------------------------------------------------- # Default Message Handler with Proxy Authentication # ----------------------------------------------------------------- if(method=="ACK") { #these you never proxy authenticate route(1); return; }; if(method=="BYE" || method=="CANCEL") { #these you never proxy authenticate route(1); return; }; xlog("L_INFO", "Proxy auth $fd $dP destination:$du $dd $ds");# if (!route(7)) { #verify the user return(0); }; if (!is_user_in("From", "noauth")) { #no authentication required if (!proxy_authorize("","subscriber")) { proxy_challenge("","0"); return; } else if (!check_from()) { sl_send_reply("403", "Use From=ID"); return; }; # consume_credentials(); }; # if (is_user_in("Credentials", "local")) { # Uncomment to use the group options route(1); # }else{ # sl_send_reply("403", "Busted!!!, you are not allowed this route"); # }; return; } With Best Regards, LU. > > Message: 3 > Date: Tue, 20 May 2008 17:38:50 +0200 > From: "luzango mfupe" <luzango.mfupe(a)gmail.com> > Subject: [OpenSER-Users] Call failed 403 Forbiden > To: users(a)lists.openser.org > Message-ID: > <9cdd611a0805200838oc11cfedg9762b7451bf543c4(a)mail.gmail.com> > Content-Type: text/plain; charset="iso-8859-1" > > Hi mates, > Everytime i do try to make my 2 xlite clients talk (which i correctly added > them into the database), i encountered with the same problm, Openser > perfoms > authentication and return call failed 403 forbiden signal. > > My setup comprise of the first box with openser 1.3 and mediaproxy the > second box with Mysql and two Xlite clients in two other boxes all are in > an > internal network, as far as am concerned my NetAdmin have already opened > ports 5060 and 3306 for me. I need your right direction on this probm. > below is my ngrep snapshot > > mzee:/# ngrep -d eth1 -W byline port 5060 > interface: eth1 (168.172.200.0/255.255.255.0) > filter: (ip or ip6) and ( port 5060 ) > # > U 168.172.200.70:1824 -> 168.172.200.87:5060 > INVITE sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87> < > sip%3Amusketeerm(a)168.172.200.87 <sip%253Amusketeerm(a)168.172.200.87> > >SIP/2.0. > Via: SIP/2.0/UDP 168.172.200.70:1824 > ;branch=z9hG4bK-d87543-ef08fb62b30f1a54-1--d87543-. > Max-Forwards: 70. > Contact: <sip:dreamteam@168.172.200.70:1824>. > To: "musk"<sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87>< > sip%3Amusketeerm(a)168.172.200.87 <sip%253Amusketeerm(a)168.172.200.87>>>. > From: "dream"<sip:dreamteam@168.172.200.87<sip%3Adreamteam(a)168.172.200.87>< > sip%3Adreamteam(a)168.172.200.87 <sip%253Adreamteam(a)168.172.200.87>> > >;tag=af4bd714. > Call-ID: OTAyNWJhOTdiZjBlNmUwNDYyZWI0YzdkM2JhZTUxMTU.. > CSeq: 1 INVITE. > Session-Expires: 95. > Min-SE: 90. > Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, > SUBSCRIBE, > INFO. > Content-Type: application/sdp. > Supported: timer. > User-Agent: X-Lite release 1011s stamp 41150. > Content-Length: 426. > . > v=0. > o=- 9 2 IN IP4 168.172.200.70. > s=CounterPath X-Lite 3.0. > c=IN IP4 168.172.200.70. > t=0 0. > m=audio 52166 RTP/AVP 107 119 100 106 0 105 98 8 101. > a=alt:1 1 : uZB2dYm+ NKBRK8Ep 168.172.200.70 52166. > a=fmtp:101 0-15. > a=rtpmap:107 BV32/16000. > a=rtpmap:119 BV32-FEC/16000. > a=rtpmap:100 SPEEX/16000. > a=rtpmap:106 SPEEX-FEC/16000. > a=rtpmap:105 SPEEX-FEC/8000. > a=rtpmap:98 iLBC/8000. > a=rtpmap:101 telephone-event/8000. > a=sendrecv. > # > U 168.172.200.87:5060 -> 168.172.200.70:1824 > SIP/2.0 403 Forbidden. > Via: SIP/2.0/UDP 168.172.200.70:1824 > ;branch=z9hG4bK-d87543-ef08fb62b30f1a54-1--d87543-. > To: "musk"<sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87>< > sip%3Amusketeerm(a)168.172.200.87 <sip%253Amusketeerm(a)168.172.200.87>> > >;tag=9a17bd4180f96d7136f8b30b25c6947e.d7e9. > From: "dream"<sip:dreamteam@168.172.200.87<sip%3Adreamteam(a)168.172.200.87>< > sip%3Adreamteam(a)168.172.200.87 <sip%253Adreamteam(a)168.172.200.87>> > >;tag=af4bd714. > Call-ID: OTAyNWJhOTdiZjBlNmUwNDYyZWI0YzdkM2JhZTUxMTU.. > CSeq: 1 INVITE. > Server: OpenSER (1.3.0-notls (i386/linux)). > Content-Length: 0. > . > # > U 168.172.200.70:1824 -> 168.172.200.87:5060 > ACK sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87> < > sip%3Amusketeerm(a)168.172.200.87 <sip%253Amusketeerm(a)168.172.200.87>> > SIP/2.0. > Via: SIP/2.0/UDP 168.172.200.70:1824 > ;branch=z9hG4bK-d87543-ef08fb62b30f1a54-1--d87543-. > To: "musk"<sip:musketeerm@168.172.200.87 <sip%3Amusketeerm(a)168.172.200.87>< > sip%3Amusketeerm(a)168.172.200.87 <sip%253Amusketeerm(a)168.172.200.87>> > >;tag=9a17bd4180f96d7136f8b30b25c6947e.d7e9. > From: "dream"<sip:dreamteam@168.172.200.87<sip%3Adreamteam(a)168.172.200.87>< > sip%3Adreamteam(a)168.172.200.87 <sip%253Adreamteam(a)168.172.200.87>> > >;tag=af4bd714. > Call-ID: OTAyNWJhOTdiZjBlNmUwNDYyZWI0YzdkM2JhZTUxMTU.. > CSeq: 1 ACK. > Content-Length: 0. > > WBR, > LU. > >

2 1

[OpenSER-Users] Problem with BYE signaling during no answer call forward to PSTN
by Krunal Patel 26 May '08

26 May '08

Hi, Here is the scenario of the call. 12345 ---> 56565656 ----> PSTN I have registered user 12345 using sjphone. Dialing 56565656 which is registered to the same openser. Now i let the call ring & after timeout call is transffered to PSTN number. Which is terminated by PSTN gateway. I am using mediaproxy for audio signaling & getting audio properly both side. Now the problem is when PSTN number hangsup the call,it should send BYE signal to the domain name. But instead of sending BYE signal to domain name it sends it to the IP. & because of that the UA can't acknowledge the BYE signal & stay connected. & BYE keeps looping in openser. As the pstn number sends BYE to IP , I have tried by setting alias=xxx.xxx.xxx.xxx . Eventhough it was not solved. Thanks in advance -- Regards, Krunal Patel

2 1

[OpenSER-Users] compilation on solaris 10
by wliangy＠gmail.com 26 May '08

26 May '08

Hi experts, I am start user of openser and need use openser to support sctp. now I have download the source codes of openser-1.3.1-tls. then I compiled it, there have many errors that need your supports. e.g. openser-1.3.1-tls> make all Makefile.defs:717: *** missing separator. Stop. then I commented Makefile.defs line717 and 718. # $(warning Old gcc detected ($(CC_SHORTVER)), use gcc >= 3.1 \ # for better results) Errors were still there. openser-1.3.1-tls> make all Makefile:402: warning: overriding commands for target `/opt/CiscoMGC$(DESTDIR)' Makefile:393: warning: ignoring old commands for target `/opt/CiscoMGC$(DESTDIR)' yacc -d -b cfg cfg.y "cfg.y", line 391: warning: redeclaration of precedence of SLASH. conflicts: 1 shift/reduce flex cfg.lex statistics.c:58: warning: #warning STATISTICS: Architecture with no support for atomic operations. Using Locks!! Makefile:402: warning: overriding commands for target `/opt/CiscoMGC$(DESTDIR)' Makefile:393: warning: ignoring old commands for target `/opt/CiscoMGC$(DESTDIR)' Compiling action.c gcc -g -O9 -funroll-loops -Wall -mv8 -DNAME='"openser"' -DVERSION='"1.3.1-notls"' -DARCH='"sparc64"' -DOS='"solaris"' -DCOMPILER='"gcc 2.91.66"' -D__CPU_sparc64 -D__OS_solaris -D__SMP_no -DCFG_DIR='"$(DESTDIR)/etc/openser/"' -DPKG_MALLOC -DSHM_MEM -DSHM_MMAP -DUSE_IPV6 -DUSE_MCAST -DUSE_TCP -DDISABLE_NAGLE -DHAVE_RESOLV_RES -DSTATISTICS -DCHANGEABLE_DEBUG_LEVEL -DF_MALLOC -DFAST_LOCK -DADAPTIVE_WAIT -DADAPTIVE_WAIT_LOOPS=1024 -DHAVE_GETIPNODEBYNAME -DHAVE_SYS_SOCKIO_H -DHAVE_SCHED_YIELD -DHAVE_ALLOCA_H -DUSE_SIGACTION -D_POSIX_PTHREAD_SEMANTICS -DHAVE_DEVPOLL -DHAVE_SELECT -c action.c -o action.o parser/msg_parser.h: In function `char_msg_val': In file included from action.h:27, from action.c:44: parser/msg_parser.h:300: parse error before `)' forward.h: In function `msg_send': In file included from action.c:49: forward.h:95: parse error before `)' forward.h:103: parse error before `)' forward.h:107: parse error before `)' mem/../lock_alloc.h: In function `lock_set_alloc': In file included from mem/../locking.h:66, from mem/../statistics.h:115, from mem/shm_mem.h:33, from ut.h:52, from action.c:54: mem/../lock_alloc.h:68: parse error before `)' ut.h: In function `int2bstr': In file included from action.c:54: ut.h:172: parse error before `)' ut.h: In function `un_escape': ut.h:365: parse error before `)' ut.h: In function `shm_str_dup': ut.h:498: parse error before `)' ut.h: In function `pkg_str_dup': ut.h:515: parse error before `)' ut.h: In function `str_strcmp': ut.h:532: parse error before `)' ut.h:535: parse error before `int' ut.h:539: `minlen' undeclared (first use in this function) ut.h:539: (Each undeclared identifier is reported only once ut.h:539: for each function it appears in.) ut.h:547: `alen' undeclared (first use in this function) ut.h:547: `blen' undeclared (first use in this function) ut.h: In function `str_strcasecmp': ut.h:563: parse error before `)' ut.h:566: parse error before `int' ut.h:570: `minlen' undeclared (first use in this function) ut.h:578: `alen' undeclared (first use in this function) ut.h:578: `blen' undeclared (first use in this function) action.c: In function `run_action_list': action.c:140: parse error before `)' action.c: In function `do_assign': action.c:193: parse error before `)' action.c:202: parse error before `)' action.c:222: parse error before `)' action.c:228: parse error before `)' action.c: In function `do_action': action.c:302: parse error before `)' action.c:309: parse error before `)' action.c:335: parse error before `)' action.c:498: parse error before `)' action.c:524: parse error before `)' action.c:682: parse error before `)' action.c:710: parse error before `)' action.c:723: parse error before `)' action.c:757: parse error before `)' action.c:763: parse error before `)' action.c:807: parse error before `)' action.c:819: parse error before `)' action.c:870: parse error before `)' action.c:896: parse error before `)' action.c:904: parse error before `)' action.c:930: parse error before `)' action.c:939: parse error before `)' action.c:945: parse error before `)' action.c:966: parse error before `)' make: *** [action.o] Error 1 Your any help is much appreciated! BR, Eric

2 1

[OpenSER-Users] Need help to solve BYE signalling problem
by Krunal Patel 26 May '08

26 May '08

Hi, ++ I have registered 3 UACs to the same openser server which are behind NAT. UAC 1 -- 111111 UAC 2 -- 222222 UAC 3 -- 333333 ++ So in location table , I am getting : ------------------------------------------------+-----------------------+-------------------------- Contact | Received | ------------------------------------------------+-----------------------+-------------------------- 111111@<private-IP>:5060 | sip:public-IP : 5060| 222222@<private-IP>:5060 | sip:public-IP : 5060| 333333@<private-IP>:5060 | sip:public-IP : 5060| ++ I have set user preferences : +------+------------------+-------------------------+-------------+------+------------------------------------------------------+ | uuid | username | domain | attribute | type | value | +------+------------------+-------------------------+-------------+------+------------------------------------------------------+ | | 222222 | domainname.com | fwdbusy | 0 | fwdbusy:sip:333333@domainname.com <fwdbusy%3Asip%3A333333(a)domainname.com> | | 222222 | domainname.com | noanscfwd | 0 | noanscfwd:sip:333333@domainname.com<noanscfwd%3Asip%3A333333(a)domainname.com> +------+------------------+-------------------------+--------------+------+------------------------------------------------------+ ++ I am using mediaproxy for audio signaling. ++ Now I am dialing 222222 using 111111 & keeps 222222 ringing until it stops automatically. So call is being forwarded to 333333. 111111 ---------------> 222222---------------> 333333 ++ I pick up the call . So 111111 & 333333 is connected. ++ Now I hangup 333333 , so it sends BYE request to 111111. + Now the problem is 111111 do not accept the BYE request. So it stays connected. & the BYE request keeps looping infinitly in openser. Here is the BYE request : Message Body: BYE sip:111111@<server-IP>:5060 SIP/2.0 Via: SIP/2.0/UDP <111111's Private IP>:5060;branch=z9hG4bK-ed33673a;rport From: <sip:222222@domainname.com <sip%3A222222(a)domainname.com> >;tag=632a4773b79ebfa8i1 To: Anonymous <sip:111111@domainname.com <sip%3A111111(a)domainname.com> >;tag=26190b20d13c8197o0 Call-ID: 4d753c94-f366deb3@localhost CSeq: 101 BYE Max-Forwards: 70 Route: <sip:[SERVER'sIP]:5060;nat=yes;ftag=26190b20d13c8197o0;lr=on>, <sip:[SERVER's IP]:5060;nat=yes;ftag=26190b20d13c8197o0;lr=on> User-Agent: Linksys/PAP2T-3.1.15(LS) Content-Length: 0 0(5877) DEBUG:maxfwd:is_maxfwd_present: value = 70 0(5877) #############################record_route############################### 0(5877) #######################After record_route########################3 0(5877) ------ ------ bye ---call tear down --- 0(5877) parse_headers: flags=78 0(5877) ##################################Before loose_route########################## 0(5877) parse_headers: flags=200 0(5877) is_preloaded: No 0(5877) grep_sock_info - checking if host==us: 13==13 && [SERVER IP] == [SERVER IP] 0(5877) grep_sock_info - checking if port 5060 matches port 5060 0(5877) after_strict: Next hop: 'sip:SERVER IP:5060;nat=yes;ftag=26190b20d13c8197o0;lr=on' is loose router 0(5877) parse_headers: flags=ffffffffffffffff 0(5877) rewrite_uri: Rewriting Request-URI with 'sip:SERVER IP:5060;nat=yes;ftag=26190b20d13c8197o0;lr=on' 0(5877) after_strict: The last route URI: 'sip:SERVER IP:5060;nat=yes;ftag=26190b20d13c8197o0;lr=on' 0(5877) ##################################into loose_route########################## 0(5877) DEBUG: has_totag: totag found 0(5877) DEBUG: has_totag: totag found 0(5877) check_via_address(PUBLIC IP of 111111, PRIVATE IP of 111111, 0) 0(5877) ##############################client_nat_test or search #########BYE############################## 0(5877) ###########################setbflag6################################# 0(5877) ------ ------ bye - loose route ----- 0(5877) DEBUG: t_newtran: T on entrance=0xffffffff 0(5877) parse_headers: flags=ffffffffffffffff 0(5877) parse_headers: flags=78 0(5877) t_lookup_request: start searching: hash=16425, isACK=0 0(5877) DEBUG: RFC3261 transaction matching failed 0(5877) DEBUG: t_lookup_request: no transaction found 0(5877) DBG: trans=0xb612d7f8, callback type 1, id 0 entered 0(5877) DEBUG: mk_proxy: doing DNS lookup... 0(5877) check_via_address(PUBLIC IP of 111111, PRIVATE IP of 111111, 0) 0(5877) DBG:check_against_rule_list: using list dns 0(5877) DEBUG:tm:set_timer: relative timeout is 500000 0(5877) DEBUG: add_to_tail_of_timer[4]: 0xb612d944 (401700000) 0(5877) DEBUG:tm:set_timer: relative timeout is 20 0(5877) DEBUG: add_to_tail_of_timer[0]: 0xb612d960 (421) 0(5877) DEBUG:tm:t_relay_to: new transaction fwd'ed 0(5877) -------------- cheked t reply --- 0(5877) DEBUG:tm:UNREF_UNSAFE: after is 0 0(5877) DEBUG:destroy_avp_list: destroying list (nil) 0(5877) receive_msg: cleaning up ++ From the parsing of BYE request , I found evenif I am doing loose route in cfg , It is doing strict route. Can you please help me to solve this. Thanks in advance. -- Regards, Krunal Patel

1 1

[OpenSER-Users] Need help with buffer overflow error
by Pete Kay 25 May '08

25 May '08

Hi, When I run a *long* sql query in openser.cfg using avp_db_query, I am getting the following error: May 26 02:52:15 [13784] ERROR:core:pv_printf: no more space for text [584] May 26 02:52:15 [13784] ERROR:core:pv_printf: buffer overflow -- increase the buffer size... May 26 02:52:15 [13784] ERROR:avpops:ops_dbquery_avps: cannot print the query How can I increase the buffer size? Thanks in advance for your help. Regards, Pete

2 1

[OpenSER-Users] DISC
by Patrick Pichler 25 May '08

25 May '08

Hi everybody, does anyone have a tested documentation (on installing, configuring or setting up DISC to work together with openSER) ... the DISC (DIameter Server Client) ? On http://developer.berlios.de/projects/disc/ and after searching the web I couldn't find anything except the tarball... thanks in advance regards Patrick

1 0

[OpenSER-Users] MediaProxy or RTPProxy
by MagoDaViola 25 May '08

25 May '08

Untitled Document Hi... What´s the best system in performance question, to use NAT MediaProxy or RTPProxy? -------------------------------------------------------------------------- Alcindo Schleder Sistema Processa Acessoria e Comunicações Ltda. a.. MSN: a_schleder(a)hotmail.com b.. Skipe: alcindo_schleder Demétrio P. dos Santos, 705 - 95670-000 - Gramado-RS-Brasil Celular + 55 54 9966 7591 alcindo(a)processa.org Fone +55 54 3286-1738 comercial(a)processa.org

4 4

[OpenSER-Users] serMyAdmin-0.5-BETA.war on FC5.tomcat5
by Zen Kato 24 May '08

24 May '08

Hi, I tried to install serMyAdmin-0.5-BETA.war on FC5, tomcat5-5.5.15-1jpp_6fc. I did the following, but http://localhost:8080/serMyAdmin does not work. Please let me know how to install sermyadmin on fedora. (1)# cp mysql-connector-java-5.1.6-bin.jar /usr/share/tomcat5/common/lib (2)/usr/share/tomcat5/conf/context.xml <?xml version="1.0" encoding="UTF-8"?> <Context path="/serMyAdmin"> <Resource auth="Container" driverClassName="com.mysql.jdbc.Driver" maxActive="20" maxIdle="10" maxWait="-1" name="jdbc/openser_MySQL" type="javax.sql.DataSource" url="jdbc:mysql://localhost:3306/openser" username="sermyadmin" password="secret"/> </Context> (3)# mysql>grant all privileges on openser.* to sermyadmin@'%' identified by 'sercret'; (4)# cp serMyAdmin-0.5.war /usr/share/tomcat5/webapps/serMyAdmin.war /var/log/tomcat5/catalina.out ....(snip)......... 2008/05/24 7:10:39 org.apache.catalina.startup.HostConfig deployWAR 情報: Webアプリケーションアーカイブ serMyAdmin.war を配備します log4j:ERROR setFile(null,true) call failed. java.io.FileNotFoundException: stacktrace.log (Permission denied) at java.io.FileOutputStream.openAppend(Native Method) at java.io.FileOutputStream.<init>(FileOutputStream.java:177) at java.io.FileOutputStream.<init>(FileOutputStream.java:102) at org.apache.log4j.FileAppender.setFile(FileAppender.java:290) at org.apache.log4j.FileAppender.activateOptions(FileAppender.java:164) at org.apache.log4j.config.PropertySetter.activate(PropertySetter.java:257) at org.apache.log4j.config.PropertySetter.setProperties(PropertySetter.java:133) at org.apache.log4j.config.PropertySetter.setProperties(PropertySetter.java:97) at org.apache.log4j.PropertyConfigurator.parseAppender(PropertyConfigurator.java:689) at org.apache.log4j.PropertyConfigurator.parseCategory(PropertyConfigurator.java:647) at org.apache.log4j.PropertyConfigurator.parseCatsAndRenderers(PropertyConfigurator.java:568) at org.apache.log4j.PropertyConfigurator.doConfigure(PropertyConfigurator.java:442) .......(snip)........ -- Zen Kato

1 0

[OpenSER-Users] sermyadmin: problems with tomcat6 automatic initialization
by Robson Souza 23 May '08

23 May '08

Hello, I got a openser 1.3.2 working with sermyadmin 0.5 installed in a Debian 4.0 OS. After some tests, everything is working fine except for one problem. By the time I start or restart the server, the tomcat spends approximately 15 min to start as follows (catalina.out file): May 23, 2008 4:10:47 PM org.apache.catalina.core.AprLifecycleListener init INFO: The APR based Apache Tomcat Native library which allows optimal performance in production environments was not found on the java.library.path: /usr/lib/jvm/java-1.5.0-sun-1.5.0.14/jre/lib/i386/client:/usr/lib/jvm/java-1.5.0-sun-1.5.0.14/jre/lib/i386:/usr/lib/jvm/java-1.5.0-sun-1.5.0.14/jre/../lib/i386 May 23, 2008 4:10:48 PM org.apache.coyote.http11.Http11Protocol init INFO: Initializing Coyote HTTP/1.1 on http-8080 May 23, 2008 4:10:48 PM org.apache.catalina.startup.Catalina load INFO: Initialization processed in 3111 ms May 23, 2008 4:10:48 PM org.apache.catalina.core.StandardService start INFO: Starting service Catalina May 23, 2008 4:10:48 PM org.apache.catalina.core.StandardEngine start INFO: Starting Servlet Engine: Apache Tomcat/6.0.16 May 23, 2008 4:10:48 PM org.apache.catalina.startup.HostConfig deployWAR INFO: Deploying web application archive serMyAdmin.war May 23, 2008 4:10:54 PM org.apache.catalina.core.StandardContext addApplicationListener INFO: The listener "org.springframework.web.util.Log4jConfigListener" is already configured for this context. The duplicate definition has been ignored. May 23, 2008 4:10:54 PM org.apache.catalina.core.StandardContext addApplicationListener INFO: The listener "org.codehaus.groovy.grails.web.context.GrailsContextLoaderListener" is already configured for this context. The duplicate definition has been ignored. [0] spring.GrailsWebApplicationContext Refreshing org.codehaus.groovy.grails.commons.spring.GrailsWebApplicationContext@1d6c336: display name [org.codehaus.groovy.grails.commons.spring.GrailsWebApplicationContext@1d6c336]; startup date [Fri May 23 16:26:43 BRT 2008]; parent: org.springframework.web.context.support.XmlWebApplicationContext@e61a35 [0] spring.GrailsWebApplicationContext Bean factory for application context [org.codehaus.groovy.grails.commons.spring.GrailsWebApplicationContext@1d6c336]: org.springframework.beans.factory.support.DefaultListableBeanFactory@baa573 May 23, 2008 4:26:49 PM org.apache.catalina.core.StandardContext addApplicationListener INFO: The listener "listeners.ContextListener" is already configured for this context. The duplicate definition has been ignored. May 23, 2008 4:26:49 PM org.apache.catalina.core.StandardContext addApplicationListener INFO: The listener "listeners.SessionListener" is already configured for this context. The duplicate definition has been ignored. May 23, 2008 4:26:49 PM org.apache.coyote.http11.Http11Protocol start INFO: Starting Coyote HTTP/1.1 on http-8080 May 23, 2008 4:26:49 PM org.apache.jk.common.ChannelSocket init INFO: JK: ajp13 listening on /0.0.0.0:8009 May 23, 2008 4:26:49 PM org.apache.jk.server.JkMain start INFO: Jk running ID=0 time=0/40 config=null May 23, 2008 4:26:49 PM org.apache.catalina.startup.Catalina start INFO: Server startup in 960673 ms In order to start faster (10 sec) after the server restart, I have to kill the java process (in CPU 99%) and start tomcat manually by performing "/etc/init.d/tomcat6 start". I'm using the tomcat6 init file provided in www.sermyadmin.org such as its instructions. How can I solve that? Robson --------------------------------- Abra sua conta no Yahoo! Mail, o único sem limite de espaço para armazenamento!

2 2

Re: [OpenSER-Users] Problems with sl module in openser1.3.1
by Arturo Ochoa 23 May '08

23 May '08

-----Mensaje original----- De: Henning Westerholt [mailto:henning.westerholt@1und1.de] Enviado el: Friday, May 23, 2008 8:52 AM Para: users(a)lists.openser.org CC: arturo arturo Asunto: Re: [OpenSER-Users] Problems with sl module in openser1.3.1 On Friday 23 May 2008, arturo arturo wrote: > Could some help me with this problem, I'm trying to run openser 1.3.1, but > it dies complaining about the sl module. I issued just 'make all' command > before this > > root@zeus:/home/aochoa/openser/openser-1.3.1-notls > # ./openser > May 22 15:57:16 [4482] DBG:core:yyparse: loading module > /usr/local/lib/openser/modules/mysql.so > May 22 15:57:16 [4482] DBG:core:yyparse: loading module > /usr/local/lib/openser/modules/sl.so > May 22 15:57:16 [4482] ERROR:core:version_control: *module compile flags > mismatch for* /usr/local/lib/openser/modules/sl.so > core: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, > SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT > module: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, > USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT Hi Arturo, the version of the sl module installed on your machine don't match the one of your server in the home directory. Do a make distclean; make all in the source directory, and install again. Cheers, Henning Thanks for your response Henning, I've already do what you mention, but the problem persists. I tried also with openser 1.3.2, and the same thing. Also I noted that there are different linked libs on the same module but compiled on different machines: Fedora: [root@apolo] /usr/local/lib/openser/modules # ldd sl.so linux-gate.so.1 => (0x0028b000) libc.so.6 => /lib/libc.so.6 (0x00111000) /lib/ld-linux.so.2 (0x0028c000) Centos: (this is the one with the problem) root@zeus:/home/aochoa/openser # ldd /usr/local/lib/openser/modules/sl.so libc.so.6 => /lib/tls/libc.so.6 (0x00149000) /lib/ld-linux.so.2 (0x003f7000) Any comments?

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[OpenSER-Users] Help - As I picked up codes negative www_authorize()
by Marcio 23 May '08

23 May '08

Hi... Anybody could me help, I'm doing some tests and would like to pick up negative codes of the return www_authorize.... I trying with ($retcode) ($?) I've not had success..... Does anyone know how...? Any hints.....?? thx...

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[OpenSER-Users] Problems with sl module in openser1.3.1
by arturo arturo 23 May '08

23 May '08

Hi guys, Could some help me with this problem, I'm trying to run openser 1.3.1, but it dies complaining about the sl module. I issued just 'make all' command before this root@zeus:/home/aochoa/openser/openser-1.3.1-notls # ./openser May 22 15:57:16 [4482] DBG:core:yyparse: loading module /usr/local/lib/openser/modules/mysql.so May 22 15:57:16 [4482] DBG:core:yyparse: loading module /usr/local/lib/openser/modules/sl.so May 22 15:57:16 [4482] ERROR:core:version_control: *module compile flags mismatch for* /usr/local/lib/openser/modules/sl.so core: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT module: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT Ing Arturo Ochoa. Electrosystems

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[OpenSER-Users] Open Source VoIP in Porto, Portugal
by Johansson Olle E 23 May '08

23 May '08

Friends, I will be spending a few days in Porto, Portugal in the beginning of June. Any Asterisk and/or OpenSER users there that wants to go out and have dinner and Open Source Voip talk? Respond off list, and we'll see if we can meet. Have a nice weekend! /Olle --- * Olle E. Johansson - oej(a)edvina.net * Asterisk Training http://edvina.net/training/

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[OpenSER-Users] [OT] NGREP showing \r and \n symbols
by Iñaki Baz Castillo 23 May '08

23 May '08

Hi, does somebody know how to use NGREP command to show \r and \n symbols instead of a dot "." replacing them? By default NGREP prints a dot "." relacing any non-printable caracter. This symbol can be changed with "-p" option but waht I want is to show \r and \n literally. Thanks for any suggestion. PD: Any other recomendation (program) to get SIP traces showing \r and \n? -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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Re: [OpenSER-Users] Authentication problem in openSER (Henning Westerholt)
by Ahmed Huraimel 23 May '08

23 May '08

hello Henning, i would like to thank you very much. you save alot of work because i decided to change my database from LDAP to MySQL. just to make it clear to anyone who may face this problem. the solution to keep you passwords hashed in the LDAP server is to save it as HA1 = MD5(username:realm:password). after you build the HA1 save as clear in the LDAP server (i am using openLADP). if you save it as MD5 you will retrieve it as {MD5} blablabla which may cause a problem. so hash it then save it as clear. thanks for every one helps me in this issue regards, Ahmed Huraimel ALALI On Fri, May 23, 2008 at 12:00 PM, <users-request(a)lists.openser.org> wrote: > Send Users mailing list submissions to > users(a)lists.openser.org > > To subscribe or unsubscribe via the World Wide Web, visit > http://lists.openser.org/cgi-bin/mailman/listinfo/users > or, via email, send a message with subject or body 'help' to > users-request(a)lists.openser.org > > You can reach the person managing the list at > users-owner(a)lists.openser.org > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of Users digest..." > > > Today's Topics: > > 1. Re: [OT] NGREP showing \r and \n symbols (Andreas Heise) > 2. Re: [OT] NGREP showing \r and \n symbols (I?aki Baz Castillo) > 3. Re: Problem with audio when either or both UAs are not behind > NAT (I?aki Baz Castillo) > 4. Re: Authentication problem in openSER (Henning Westerholt) > > > ---------------------------------------------------------------------- > > Message: 1 > Date: Thu, 22 May 2008 23:55:56 +0200 > From: Andreas Heise <aheise(a)gmx.de> > Subject: Re: [OpenSER-Users] [OT] NGREP showing \r and \n symbols > To: Ovidiu Sas <osas(a)voipembedded.com> > Cc: users(a)lists.openser.org > Message-ID: <4835EBEC.7070800(a)gmx.de> > Content-Type: text/plain; charset=ISO-8859-1; format=flowed > > Hi I?aki, > > I had the same issue with a Nortel CS1k switch, the easiest way to check > this > problem is to use the Wireshark feature "Display raw text for SIP message" > which can be enabled under preferences / protocols / SIP > > regards, > Andreas > > > Ovidiu Sas schrieb: > > Hello Inaki, > > > > Perform a raw capture using tcpdump and open the file with wireshark. > > Check the lower window in wireshark where you have the message > > displayed in hex format. > > There you will be able to spot '\n' and '\r'. To find the hex codes, > > perform a 'man ascii' on your linux box. > > > > > > Regards, > > Ovidiu Sas > > > > On Tue, May 20, 2008 at 10:53 AM, I?aki Baz Castillo <ibc(a)in.ilimit.es> > wrote: > > > >> El Tuesday 20 May 2008 16:37:03 Bogdan-Andrei Iancu escribi?: > >> > >>> Hi I?aki, > >>> > >>> have you tried the "-W byline" ? - it will format you the lines. > >>> > >> Hi Bogdan. Yes, in fact I use this option "-W byline", but this option > just > >> splits the trace into lines (macthing "\n"). This is not enough for me > since > >> \r is not showed. > >> > >> I need it since I send and receive traffic from a carrier using a > painful > >> Nortel CS2K softswitch that separes SDP lines with \r instead of the > >> mandatory \r\n adn they are applying dirty and not working patches > during > >> production state so I need a way to monitorize it. > >> > >> Thanks a lot. > >> > >> > >> > >> -- > >> I?aki Baz Castillo > >> ibc(a)in.ilimit.es > >> > >> _______________________________________________ > >> Users mailing list > >> Users(a)lists.openser.org > >> http://lists.openser.org/cgi-bin/mailman/listinfo/users > >> > >> > > > > _______________________________________________ > > Users mailing list > > Users(a)lists.openser.org > > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > > ------------------------------ > > Message: 2 > Date: Fri, 23 May 2008 09:31:31 +0200 > From: I?aki Baz Castillo <ibc(a)in.ilimit.es> > Subject: Re: [OpenSER-Users] [OT] NGREP showing \r and \n symbols > To: users(a)lists.openser.org > Message-ID: <200805230931.31959.ibc(a)in.ilimit.es> > Content-Type: text/plain; charset="iso-8859-1" > > El Thursday 22 May 2008 23:55:56 Andreas Heise escribi?: > > Hi I?aki, > > > > I had the same issue with a Nortel CS1k switch > > It's nice to know that I'm not theonly sending/receiving traffic from a > Nortel > CS[12]K. ;) > > -- > I?aki Baz Castillo > ibc(a)in.ilimit.es > > > > ------------------------------ > > Message: 3 > Date: Fri, 23 May 2008 09:32:21 +0200 > From: I?aki Baz Castillo <ibc(a)in.ilimit.es> > Subject: Re: [OpenSER-Users] Problem with audio when either or both > UAs are not behind NAT > To: users(a)lists.openser.org > Message-ID: <200805230932.21444.ibc(a)in.ilimit.es> > Content-Type: text/plain; charset="ISO-8859-1" > > El Thursday 22 May 2008 18:54:53 Krunal Patel escribi?: > > Hi, > > > > I have registered an UA 12345 , which is public. > > In location table, I found NULL value for received. > > I dial any UA , irrespective to their NAT status. > > Call has been established but can't get audio both side. > > It does not use mediaproxy for audio signaling. > > > > How to resolve this problem? > > Do a full sip flow capture and analyze trhe address and port of media in > SDP > of INVITE and 200 OK. > > > -- > I?aki Baz Castillo > ibc(a)in.ilimit.es > > > > ------------------------------ > > Message: 4 > Date: Fri, 23 May 2008 10:57:56 +0200 > From: Henning Westerholt <henning.westerholt(a)1und1.de> > Subject: Re: [OpenSER-Users] Authentication problem in openSER > To: users(a)lists.openser.org > Cc: Ahmed Huraimel <huraimel(a)gmail.com> > Message-ID: <200805231057.56974.henning.westerholt(a)1und1.de> > Content-Type: text/plain; charset="utf-8" > > On Thursday 22 May 2008, Ahmed Huraimel wrote: > > i am investigating the authentication on openSER. I search for a proper > > explanations but unfortunately i did not find how it is exactly done so i > > did some experiments. i assumed that the response is generated as the > > following: note that i set the username and password with the same string > > "alali" > > > > Response = MD5( username + MD5(password) + realm + nonce) > > [..] > > could anyone tell me how exactly the authentication is done in openSER? > is > > the response generated is like the one i assumed? what + means in the > > response? does is mean concatenation or exoring? > > Hi Ahmed, > > the authentification in OpenSER/ SIP is based on HTTP auth. You find a > detailed explanation for the construction of the response for example at: > http://en.wikipedia.org/wiki/Digest_access_authentication , some further > informations at: http://www.voip-info.org/wiki/view/SIP+Authentication > > For the exact logic inside OpenSER just take a look at the auth module > source > code, it should be not that hard to understand. ;-) > > Cheers, > > Henning > > > > ------------------------------ > > _______________________________________________ > Users mailing list > Users(a)lists.openser.org > http://lists.openser.org/cgi-bin/mailman/listinfo/users > > > End of Users Digest, Vol 36, Issue 48 > ************************************* >

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[OpenSER-Users] Authentication problem in openSER
by Ahmed Huraimel 23 May '08

23 May '08

hello all, i am investigating the authentication on openSER. I search for a proper explanations but unfortunately i did not find how it is exactly done so i did some experiments. i assumed that the response is generated as the following: note that i set the username and password with the same string "alali" Response = MD5( username + MD5(password) + realm + nonce) and in the openser.cfg i tell the server to take the username from the header and set the password from the username since the username and the password are the same (only for testing). TEST1: setting modparam("auth", "calculate_ha1", 1) and telling the server to expect a clear password $avp(s:username) = $fU; $avp(s:password) = $fU; IT WORKS TEST2:setting modparam("auth", "calculate_ha1", 0) and telling the server to expect a hashed password $avp(s:username) = $fU; $avp(s:password) = $(fU{s.md5}); IT DOES NOT WORK could anyone tell me how exactly the authentication is done in openSER? is the response generated is like the one i assumed? what + means in the response? does is mean concatenation or exoring? if(!pv_www_authorize("")) { www_challenge(""/*realm*/,"0"/*qop*/); exit; } if (!save("location")) sl_reply_error(); sl_send_reply("200", "ok"); exit;

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[OpenSER-Users] Problem with audio when either or both UAs are not behind NAT
by Krunal Patel 23 May '08

23 May '08

Hi, I have registered an UA 12345 , which is public. In location table, I found NULL value for received. I dial any UA , irrespective to their NAT status. Call has been established but can't get audio both side. It does not use mediaproxy for audio signaling. How to resolve this problem? Thanks in advance. -- Regards, Krunal Patel

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[OpenSER-Users] Problems with sl module in openser1.3.1
by aochoa aochoa 22 May '08

22 May '08

Hi guys, Could some help me with this problem, I'm trying to run openser 1.3.1, but it dies complaining about the sl module. I issued just 'make all' command before this root@zeus:/home/aochoa/openser/openser-1.3.1-notls # ./openser May 22 15:57:16 [4482] DBG:core:yyparse: loading module /usr/local/lib/openser/modules/mysql.so May 22 15:57:16 [4482] DBG:core:yyparse: loading module /usr/local/lib/openser/modules/sl.so May 22 15:57:16 [4482] ERROR:core:version_control: *module compile flags mismatch for* /usr/local/lib/openser/modules/sl.so core: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT module: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT Ing Arturo Ochoa. Electrosystems

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[OpenSER-Users] Problems with sl module in openser1.3.1
by aochoa aochoa 22 May '08

22 May '08

Hi guys, Could some help me with this problem, I'm trying to run openser 1.3.1, but it dies complaining about the sl module. I issued just 'make all' command before this root@zeus:/home/aochoa/openser/openser-1.3.1-notls # ./openser May 22 15:57:16 [4482] DBG:core:yyparse: loading module /usr/local/lib/openser/modules/mysql.so May 22 15:57:16 [4482] DBG:core:yyparse: loading module /usr/local/lib/openser/modules/sl.so May 22 15:57:16 [4482] ERROR:core:version_control: *module compile flags mismatch for* /usr/local/lib/openser/modules/sl.so core: STATS: Off, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT module: STATS: Off, EXTRA_DEBUG, USE_IPV6, USE_TCP, DISABLE_NAGLE, USE_MCAST, SHM_MEM, SHM_MMAP, PKG_MALLOC, F_MALLOC, FAST_LOCK-ADAPTIVE_WAIT Ing Arturo Ochoa. Electrosystems

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[Serusers] Mysql Problem
by vijay tiwari 22 May '08

22 May '08

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[OpenSER-Users] OPENSER Dinner, Vienna, Austria, May 24
by Daniel-Constantin Mierla 22 May '08

22 May '08

Hello, just to announce the next OPENSER Social Networking Event in Vienna, Austria, May 24, 2008, 19:30 at Plutzer Brau: http://www.plutzerbraeu.at/plutzerbraeu.html <http://www.voipuser.org/ship_to.php?url=http://www.plutzerbraeu.at/plutzerb…> Among the participants that confirmed the presence already: - Elena-Ramona Modroiu, co-founder OpenSER, COO at http://www.asipto.com <http://www.voipuser.org/ship_to.php?url=http://www.asipto.com> - Andreas Granig, developer OpenSER, CEO at http://www.sipwise.com <http://www.voipuser.org/ship_to.php?url=http://www.sipwise.com> - Daniel-Constantin Mierla, co-founder OpenSER, CEO at http://www.asipto.com <http://www.voipuser.org/ship_to.php?url=http://www.asipto.com> The primary goal of the event is to strengthen the relations within community and give opportunity to network for collaboration and business. Participation to the event is free, everybody pays for its expenses. Anyone interested in OpenSER, SIP or VoIP in general is welcome to join. You should contact us to reserve your seat and get notifications in case of changes of time or place. Previous similar event was in Barcelona, May 6 -- remarks and photos: http://www.asipto.com/photos/BARCELONA2008-ALBUM/ http://openser.blogspot.com/2008/05/openser-dinner-barcelona-remarks.html Next one is in Berlin, May 28 (more details soon). Cheers, Daniel -- http://www.asipto.com

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[OpenSER-Users] [Fwd: Re: facing problem]
by Amit Vijayvargiya 22 May '08

22 May '08

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Re: [Serusers] ser/sems question
by Stefan Sayer 22 May '08

22 May '08

o Steven C. Blair [05/22/08 13:52]: > Thanks Stefan. Is the README for sems-1.0.0 correct? I'm questioning step #4. If I am running this version of SEMS on a Linux box without a proxy server, our proxy resides elsewhere, do I need to perform step #4? no, that is optional, and you only need to do that if you want SER as frontend to SEMS. That's why it says 'Optional' in in http://ftp.iptel.org/pub/sems/doc/current/Readme.html Stefan > > Thanks,Steve > >> -----Original Message----- >> From: Stefan Sayer [mailto:stefan.sayer@iptego.com] >> Sent: Wednesday, May 21, 2008 4:45 PM >> To: Steven C. Blair >> Cc: serusers(a)iptel.org >> Subject: Re: [Serusers] ser/sems question >> >> Hello, >> >> o Steven C. Blair [05/20/08 20:13]: >>> I've been away from SEMS for a few years. II hear the new version of >>> SEMS makes an excellent conference server. Does anyone happen to know >> if >>> I can install SEMS on a standalone Linux machine and interface it, >> via >>> SIP, to a SER 0.9.7-pre3 proxy server? >> yes, you can use it now with any SIP proxy thanks to its own SIP stack. ... -- Stefan Sayer VoIP Services stefan.sayer(a)iptego.com www.iptego.com iptego GmbH Am Borsigturm 40 13507 Berlin Germany Amtsgericht Charlottenburg, HRB 101010 Geschaeftsfuehrer: Alexander Hoffmann

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[OpenSER-Users] Problem PUA_USRLOC, Presence acc. REGISTER/INVITE
by Schumann Sebastian 22 May '08

22 May '08

Dear all I have a problem, which I thought I can solve with PUA_USRLOC. I have a OpenSER SIP Proxy and want him to create presence information about the messages that pass him. Some examples: - REGISTER: Publish as online/available - REGISTER+Expires=0: Publish as offline - INVITE/200 OK: Publish as busy - BYE: Publish as available again. PUA_USRLOC seems to me can handle only simple on/offline according registration state, that is saved on the server. As I have only a proxy server and messages just pass by, this solution seems not the best. What would you recommend me to solve the problem? Change the module, add a module? As I don't have that much of experience in that area I was thinking about using perl module and trigger some perl functions that parse the messages that pass the proxy and make perl send the PUBLISHes to the presence server. Maybe some of you have other ideas or suggestions about that.... I would be happy if you share them with me. Best regards Sebastian

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[OpenSER-Users] Does somebody use $(rU{s.unescape.user}) to be RFC3261 compliant?
by Iñaki Baz Castillo 22 May '08

22 May '08

Hi, to be RFC3261 compliant a SIP proxy should accept hex encoded username in any URI, this is: sip:alice@domain == sip:%61l%69ce@domain For allowing this we must use, explicitely, the transformation $(rU{s.unescape.user}). Do OpenSer administrators allow this hex encoding? -- Iñaki Baz Castillo ibc(a)in.ilimit.es

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[OpenSER-Users] Small documentation mistake pua_usrloc
by Schumann Sebastian 22 May '08

22 May '08

Dear all I noticed a small mistake in the documentation: I had some errors in config file using the pua_usrloc modparam parameters with copy & paste. The problem were the semicolons ; behind the modparam definitions. As I had the error some time ago already I easily figured that out. Although, it might be good to remove them from the doc for other users in the future to not have the same problem as well. Best regards Sebastian

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