- sr-users - kamailio.org

[Serusers] avp_radius
by samuel 21 Feb '06

21 Feb '06

Hi all!! I'm testing stable ser version (0.9.6) with RADIUS authentication. I have properly setup the auth and next step was to get some extra attributes from the auth server using the avp_radius module. The thing is that I am a newby in the RADIUS world and don't know exactly what do I have to setup in both dictionaries. The main problem is that the attribute I have to read from ser script is a cisco-h323-whatever and don't even know if it is possible to read it as a caller_cisco-h323-… [View More]

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Re: [Users] Beer at wednesday 22 at 18:00 ?
by Helge Waastad 21 Feb '06

21 Feb '06

Hi, yet again a brilliant initiative, Atle. As much as I would like to join this time, I just cant. I hope we can have another go again soon. br hw -- Helge Waastad Senior Konsulent Systemavdelingen Smartnet

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[Users] how to use OpenSER
by Brijesh 21 Feb '06

21 Feb '06

hi all i am explaining my problem i have installed openser on one of machine. i have assigned SIP_DOMAIN in that machine as "test.com". then i also added users to the openser using *openserctl add user1(a)test.com test user1(a)test.com* command. so that user is also added to the myslql database. till now it is ok. now i am using Minisip as a sip end application. in that i have given account name as user1(a)test.com, sip uri as sip:user1@test.com, sip proxy address as machine's ip … [View More]

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[Serusers] Help! DNS?
by ryan nalupa 21 Feb '06

21 Feb '06

hello! i was just wondering, since i'm still new here, where will i config this? i'm using debian sarge amd64 and ser 0.9.4. DNS SVR Resource Records It is important that your SIP clients can connect to your server for purposes of registration and call control. You might even want to have a redundant server to handle calls if your primary server is unavailable. These requirements can be meet by using DNS SVR Resource Records, available in BIND 8.X and up releases. The format for a … [View More]SVR RR is this: _service._protocol SVR Priority Weight Port hostname In this case we want to establish an entry for our primary SIP server, gateway.mydomain.com, that will listen on UDP port 5060. The entry will look like this: _sip._udp SRV 0 0 5060 gateway.mydomain.com Placement of the new resource record is important. Here is a sample zone file: ; zone 'mydomain.com' last serial 1998071308 $ORIGIN com. mydomain 86400 IN SOA gateway.mydomain.com. postmaster.mydomain.com. ( 1998111908 ; Serial 36000 ; Refresh 900 ; Retry 36000 ; Expire 28800 ); Minimum IN NS gateway.mydomain.com. IN NS ns3.backupdomain.com. IN MX 1 gateway.mydomain.com. IN A 192.168.0.1 ;If we place the SRV record above the next line it fails to load $ORIGIN fitawi.com. _sip._udp SRV 0 0 5060 gateway.mydomain.com. gateway IN A 192.168.0.1 www IN CNAME gateway.mydomain.com. After reloading your zone file you can verify that the entry is working by using dig. dig -t SRV _sip._udp.mydomain.com The results should look something like this: ; <<>> DiG 9.1.0 <<>> -t SRV _sip._udp.mydomain.com ;; global options: printcmd ;; Got answer: ;; ->>HEADER<<- opcode: QUERY, status: NOERROR, id: 32654 ;; flags: qr aa rd ra; QUERY: 1, ANSWER: 1, AUTHORITY: 2, ADDITIONAL: 1 ;; QUESTION SECTION: ;_sip._udp.mydomain.com. IN SRV ;; ANSWER SECTION: _sip._udp.mydomain.com. 86400 IN SRV 0 0 5060 gateway.mydomain.com. ;; AUTHORITY SECTION: mydomain.com. 86400 IN NS ns3.elsewhere.com. mydomain..com 86400 IN NS gateway. mydomain.com. ;; ADDITIONAL SECTION: gateway. mydomain.com. 86400 IN A 192.168.0.150 ;; Query time: 6 msec ;; SERVER: 192.168.0.150#53(192.168.0.150) ;; WHEN: Tue Dec 3 08:34:17 2002 ;; MSG SIZE rcvd: 132 thanks in advance! ryan --------------------------------- Do you Yahoo!? Try the new Yahoo! Philippines Front Page! [View Less]

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[Serusers] loose_route bug?!
by Ramin Nikaeen 21 Feb '06

21 Feb '06

Hi Everyone, I am using SER version Server: Sip EXpress router (0.9.2 (i386/linux)). I am initiating a call from a caller sip UA to a callee sip UA and I am receiving the ACK response to the INVITE from the callee on SER. Before loose_route processing it looks like this: ACK sip:2224440133@10.1.10.65:5060 SIP/2.0 Via: SIP/2.0/UDP 172.16.15.52:5060;rport;branch=z9hG4bK83e54f590401584628d473053d1ef8a8 From: <sip:2224440132@xxx.yyy.zzz:5060>;tag=1cac2190 To: <sip:… [View More]

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RE: [Users] next_gw(): No ruri_user AVP
by Raymond Chen 21 Feb '06

21 Feb '06

Hi Bogdan, I solved problem by forcing avp_write("$ruri/username", "i:1402"). Thanks Ray -----Original Message----- From: users-bounces(a)openser.org [mailto:users-bounces@openser.org] On Behalf Of Raymond Chen Sent: Thursday, February 16, 2006 12:27 AM To: 'Bogdan-Andrei Iancu' Cc: Users(a)openser.org Subject: RE: [Users] next_gw(): No ruri_user AVP Hi Bogdan, Both, after load_gw and before next_gw avps are visible. But not after next_gw. There is no where in the script delete any … [View More]avp. Ray -----Original Message----- From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] Sent: Wednesday, February 15, 2006 8:18 AM To: Raymond Chen Cc: daniel(a)voice-system.ro; Users(a)openser.org Subject: Re: [Users] next_gw(): No ruri_user AVP Ray, I think there is a problem in the script - if you see the gw_uri avps, it's quite impossible not to see them also before and after the next_gw() call in request route. Disregarding the ruri_avp, the gw_uri avps should be visible. are you sure you do not delete all the avps in your script. Try placing avp_print before and after each lcr function call. regards, bogdan Raymond Chen wrote: >Hi bogdan, > >I put avp_print after next_gw , no avp output. > > >1(1334) load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.138:5060> > 1(1334) load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.139:5060> > 1(1334) does_uri_exit(): User in request uri does not exist > 1(1334) is_user_in(): User is in group 'local' > 1(1334) db_flags=3, flags=12 > 1(1334) DEBUG:avpops:load_avps: loaded avps = 1 > 1(1334) parse_headers: flags=ffffffffffffffff > 1(1334) DEBUG:avpops:pushto_avps: 1 avps were processed >1(1334) next_gw(): No ruri_user AVP > > > > > > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Wednesday, February 15, 2006 4:39 AM >To: Raymond Chen >Cc: daniel(a)voice-system.ro; Users(a)openser.org >Subject: Re: [Users] next_gw(): No ruri_user AVP > >Ray, > >the ruri_avp is added by next_gw after its usage from the REQUEST route. >You may check this by placing an avp_print after you did next_gw() in >request route (after calling route 3, for example). Check if there is >any avp with ID 1402. > >regards, >bogdan > >Raymond Chen wrote: > > > >>Hi bogdan, >> >>We called Load_gw and next-gw from request route. We have no problem with >>next_gw if the if (avp_pushto("$ruri", "s:fwdnoanswer")) happens in the >> >> >main > > >>route. But when it does in the failure_route the next_gw can't find the >>ruri_user avp. >> >>Raymond >> >> >> >>Route { >> >> .......... >> >> Route(3); >> >> ........... >> >>} >> >>failure_route[1] { >> >> if (t_check_status("(480)|(408)")) { >> if (avp_pushto("$ruri", "s:fwdnoanswer")) { >> avp_delete("s:fwdnoanswer"); >> route(3); >> }; >> }; >> >>} >> >>Route[3] { >> >> if (!load_gws()) { >> sl_send_reply("500", "Server Internal Error - Cannot load >>gateways"); >> return; >> }; >> >> ............... >> >> Route(5); >> >>} >> >>Route[5] { >> >> if (!next_gw()) { >> rewriteuri("sip:userbusy@211.102.91.134:443"); >> t_relay(); >> return; >> }; >> >> .............. >> >>} >> >> >> >>-----Original Message----- >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >>Sent: Wednesday, February 15, 2006 2:25 AM >>To: Raymond Chen >>Cc: daniel(a)voice-system.ro; Users(a)openser.org >>Subject: Re: [Users] next_gw(): No ruri_user AVP >> >>Hi Ray, >> >>do you call load_gws() from failure route? if so, not that this is not >>supported. >>For LCR to work properly, you need to call load_gws() and next_gw() from >>the request route and later next_gw() from failure routes. >> >>regards, >>bogdan >> >>Raymond Chen wrote: >> >> >> >> >> >>>Hi Bogdan, >>> >>>Here is the debug >>> >>>1(2584) load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.138:5060> >>>1(2584) load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.139:5060> >>>1(2584) DEBUG:avpops:print_avp: p=0xf4f167c8, flags=2 >>>1(2584) DEBUG: id=<1400> >>>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.139:5060> >>>1(2584) DEBUG:avpops:print_avp: p=0xf4f16790, flags=2 >>>1(2584) DEBUG: id=<1400> >>>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.138:5060> >>>1(2584) does_uri_exit(): User in request uri does not exist >>>1(2584) is_user_in(): User is in group 'local' >>>1(2584) db_flags=3, flags=12 >>>1(2584) DEBUG:avpops:print_avp: p=0xf4f167f8, flags=B >>>1(2584) DEBUG: id=<1400> >>>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.139:5060> >>>1(2584) DEBUG:avpops:print_avp: p=0xf4f16790, flags=2 >>>1(2584) DEBUG: id=<1400> >>>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.xxx:5060> >>>1(2584) next_gw(): No ruri_user AVP >>> >>> >>> >>>Raymond >>> >>> >>>-----Original Message----- >>>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >>>Sent: Tuesday, February 14, 2006 10:55 AM >>>To: Raymond Chen >>>Cc: daniel(a)voice-system.ro; Users(a)openser.org >>>Subject: Re: [Users] next_gw(): No ruri_user AVP >>> >>>Hi Ray, >>> >>>use avp_print() after the load_gw() to see what avps were loaded and >>>again just before next_gw() to see the available avp. >>>this will help to see if it's a problem at the load or search part. >>> >>>regards, >>>bogdan >>> >>> _______________________________________________ Users mailing list Users(a)openser.org http://openser.org/cgi-bin/mailman/listinfo/users [View Less]

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[Serusers] Errors...
by ryan nalupa 21 Feb '06

21 Feb '06

hello! i'm following the ser howto that i got at iptel.org and i'm having errors when following these... 3.2. Adding an admin for your realm Now that we have a working database and ser is configured to use it, we need to add some users and at least one of them should have administrator privileges. The administrator role becomes important if you want to use a web management tool such as serweb. Basic account manipulation can be performed with the serctl script, located in /usr/sbin. … [View More]

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Re: [Serusers] Problem with time of call invite in Mysql
by mkumar＠mantragroup.com 21 Feb '06

21 Feb '06

Hi, Thanks for the reply. I can chage my TimeZone but many Users will be having many timezones and if calls of each and every user records in Mysql DB according to his timezone then how can we have a common timezone for all atleast for getting time records in a particular period for billing. Manoj. > Quoting Andrey Kouprianov <andrey.kouprianov(a)gmail.com>: > >> Change your timezone >> >> On 2/20/06, mkumar(a)mantragroup.com <mkumar(a)mantragroup.com> … [View More]

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[Serusers] modifying SDP of 180
by Ever Zalazar 21 Feb '06

21 Feb '06

Hi, I have a question. I need to modify the SDP of code 180 when my clients use my mediaproxy. How is the process between a SER and the media proxy? Best REgards Ever

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[Serusers] SER billing softwares
by mkumar＠mantragroup.com 20 Feb '06

20 Feb '06

Hi All, Are there any open source SER billing softwares? If there are any, please point me to that. Thanks, Manoj.

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[Users] Beer at wednesday 22 at 18:00 ?
by Atle Samuelsen 20 Feb '06

20 Feb '06

Hi guys, girls and Geeks :-) Have you booked your Plain-ticket's yet? No? Remember, it's beer meeting on Wednesday the 22 february 2006 at 18:00. The beer will be consumed at Beer Palace at Akerbrygge Oslo, Norway. If somebody wanna join in on IP-Conferance, We can try to arrange that. (pleace send a mail saying if you are coming, since if we are more then a few ppl we need to book a table) Best Regards Greger Teigre and Atle Samuelsen

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[Serusers] SER load balancer for asterisk
by Script Head 20 Feb '06

20 Feb '06

Hello, I am trying to build a setup that's probably a very generic. I have a number of asterisk servers that I use to handle transcoding. What I'd like to use SER for is to "spread" calls among these asterisk servers. Where should I start looking? ScriptHead

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[Serusers] permissions module & allow_routing
by Kostas Marneris 20 Feb '06

20 Feb '06

Hello all, I have some trouble with permissions module and allow_routing(). I want to allow_routing ONLY from IP 1.2.3.4 (which is my PSTN GW) and all the rest will be denied (actually all the other will be send for authentication with "radius_proxy_authorize"). What is the syntax of the permissions.* files ? The following doesn't work.. permissions.deny: ALL : ALL permissions.allow: "^sip:*@1.2.3.4" : "^sip:*@10.10.10.1" where 10.10.10.1 is my SER's IP. thanks for any help, … [View More]

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[Users] Mysql DB Auth
by Sam Lee 20 Feb '06

20 Feb '06

Hi all, I've just finish installing the OPENSER 1.0 Trying to make it work on the mysql part. Just can't get it to connect to my mysql server properly with the correct authentication. It seems like it tries to go in as the default userid/password (openserro,openser) even though i've specified an alternate userid password. modparam("auth_db|uri_db|usrloc|alias_db", "db_url", "mysql://ser:heslo@localhost/ser") Refer to above , this line is already included in the openser.cfg. As u might … [View More]

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[Users] Multiple Interfaces rtpproxy
by Fernando Rodriguez 20 Feb '06

20 Feb '06

Hello, I have a running openser on a machine with multiple interfaces im using mediaproxy but I was only able to set it up using 2 interfaces .. How can I set it up to have more working interfaces 1 lan 1 dmz 1 wan Right now it only mediaproxies the wan and dmz interfaces But I want to be able to mediaproxy betrwen the lan and wan and dmz to lan Does any one has a set up like this and will be able to guide me on seting this up .. Thanks a lot for your help Fernando Rodriguez V. frod(a)aitelecom.net

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[Users] new version of "OpenSER Database Structure" document
by Daniel-Constantin Mierla 20 Feb '06

20 Feb '06

Hello, a new version of "OpenSER Database Structure" document submitted by Norman Brandinger is posted at: http://openser.org/docs/OpenSER_Database_Structure.html It describe the meaning of the columns in the tables used by openser. Once again, many thanks to Norman! Cheers, Daniel

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[Serusers] Is it possible to run SerWeb and SER on separated machines? (what about FIFO?)
by benaudi＠gmx.de 20 Feb '06

20 Feb '06

Hello, is it at the moment possible to run SER and SerWeb on two different pc's? In SerWeg config-file you have to define the path to "ser_fifo". But i cant access the ser_fifo which is running on another PC. Is there any example or discussion already? Thank you in Advance! Alex -- Lust, ein paar Euro nebenbei zu verdienen? Ohne Kosten, ohne Risiko! Satte Provisionen f�r GMX Partner: http://www.gmx.net/de/go/partner

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[Serusers] error querying table version
by Yi Zheng 20 Feb '06

20 Feb '06

Hi, While trying to setup auth_db() with mysql, I got the following error messages in /var/log/messages. table_version(): Invalid number of rows received: 2, location Feb 16 17:55:17 localhost ./ser[13709]: register_udomain(): Error while querying table version Feb 16 17:55:17 localhost ./ser[13709]: domain_fixup(): Error while registering domain I used ser_mysql.sh create to generate the ser database. It looks like that ser has two location entries in its version table. I tried … [View More]

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[Serusers] Multiple domains via a DB
by Nick Hoffman 20 Feb '06

20 Feb '06

How do you setup a multi-domain SER using a database? I've seen examples where you hard code into ser.cfg a regular expression for each domain. Is there a call that can do something similar as in compare the URI with returned values from the database? I tried this: if(is_from_local()) { sl_send_reply("600", "is local"); break; }; but it didn't fire for existing or non-existing domains. Does anyone have suggestions as to what should be done instead, or what needs … [View More]

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[Serusers] UA Header Field & Mediaproxy
by George Lambson 20 Feb '06

20 Feb '06

Does anyone know how to insert a User-Agent field into the header BEFORE Mediaproxy gets it? I am trying to set up mediaproxy to work with a UA that is asymmetric, but it doesn't recognize the UA because this header field is missing. Thanks, George

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[Serusers] Postgres + ser + problem
by Andrew Nowrot 20 Feb '06

20 Feb '06

Hi I am new to ser issues. I installed it last Friday. I managed to set up it correctly with Asterisk, but now I have a slightly problem with authorization, frankly with postgres. part of my ser.cfg file looks like this: # Uncomment this if you want to use SQL database #loadmodule "/usr/lib/ser/modules/mysql.so" loadmodule "/usr/lib/ser/modules/postgres.so" loadmodule "/usr/lib/ser/modules/sl.so" loadmodule "/usr/lib/ser/modules/tm.so" loadmodule "/usr/lib/ser/modules/rr.so" loadmodule "/usr/… [View More]lib/ser/modules/maxfwd.so" loadmodule "/usr/lib/ser/modules/usrloc.so" loadmodule "/usr/lib/ser/modules/registrar.so" loadmodule "/usr/lib/ser/modules/textops.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/lib/ser/modules/auth.so" loadmodule "/usr/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # -- usrloc params -- #modparam("usrloc", "db_mode", 0) modparam("usrloc", "user_column", "username") modparam("usrloc","db_url", "postgres://ser:ser1234@localhost:5432/ser") # 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") modparam("auth_db","db_url", "postgres://ser:ser1234@localhost:5432/ser") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication if (!www_authorize("ser.leon.pl", "subscriber")) { www_challenge("ser.leon.pl", "0"); break; }; save("location"); break; }; } *** my postgres table subscriber look like this: username | domain | password ------------ + ------------- + ---------- 2222 | ser.xxx.org | serxxx this is my only postgres table connected with ser. I did my configuration according to http://www.voip-info.org/wiki-SER+module+postgres. When I restarting ser I getting this error in syslog and my phones are not able to register to ser: Feb 18 16:31:39 asterix /usr/sbin/ser[6164]: Maxfwd module- initializing Feb 18 16:31:39 asterix /usr/sbin/ser[6164]: PG[364] submit_query query 'select table_version from version where table_name='subscriber'', result 'ERROR: relation "version" does not exist ' Feb 18 16:31:39 asterix /usr/sbin/ser[6164]: db_query(): Error while submitting query Feb 18 16:31:39 asterix /usr/sbin/ser[6164]: table_version(): Error in db_query Feb 18 16:31:39 asterix /usr/sbin/ser[6164]: auth_db:str_fixup(): Error while querying table version Feb 18 16:31:39 asterix /usr/sbin/ser[6164]: ERROR: fix_expr : fix_actions error Does anyone can help me with this because I stuck and I can't go any further with my configuration. Please help Andrew Cheers [View Less]

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[Serusers] Problem with time of call invite in Mysql
by mkumar＠mantragroup.com 20 Feb '06

20 Feb '06

Hi All, I have SER and mysql DB in USA and I am calling from India. For all the calls I am dialling from India the time of call invite is getting recorded 5.5 hours ahead the local time in USA. Suppose if time in USA is 6:30 and if I call from India, in mysql DB it is getting stored that I called at 12:00. So for billing if I calculate calls between NOW() and NOW()-30 days, all the calls in this 5.5 hours are missing. Please tell me where I have to change my configuration. I have ser -0.9.4 … [View More]

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[Serusers] ENUM
by Anderson Alves Albuquerque 20 Feb '06

20 Feb '06

Does someone has a example with ENUM in SER? I need to use SER with ENUM, but I think that the manaul in IPTEL is very sample and it is not complete.

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[Users] Partition / Heteling feature
by Girish Nayak 20 Feb '06

20 Feb '06

Hi, Does OpenSER have partition / hoteling feature? Thanks NG

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[Serusers] Partition / Hoteling feature
by Girish Nayak 20 Feb '06

20 Feb '06

Hi Does SER have partition / heteling feature? Thanks NG --------------------------------- Yahoo! Autos. Looking for a sweet ride? Get pricing, reviews, & more on new and used cars.

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Re: [Users] how to use openSER
by Pjothi 20 Feb '06

20 Feb '06

Hello Brijesh, The question could have been more specific. What are you looking at ? Installion instructions or uses of openser ? be more specific and am sure experts would point you in the right direction OpenSER is a SIP proxy. Very basically it can be used for SIP client registration with TLS transport layer security. Have you used SER before ? IF yes, its almost the same as SER with added TLS support IF not, Compile OpenSER with Mysql and with or without TLS support. Go through the … [View More]

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[Users] how to use openSER
by Brijesh 20 Feb '06

20 Feb '06

Hi all Can anyone tell me how to use openser ? I am not getting it from the site. Thanx in advance Brijesh

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[Serusers] Re: Calling External Script from Ser.cfg
by mkumar＠mantragroup.com 20 Feb '06

20 Feb '06

Hi, I am facing a problem returning a value from shell script for ser.cfg. I could not even get working with exec_msg. I added a line like this if(exec_msg('x=1; if [ $x == 0 ];then exit 1 ;fi')) ERROR: exec_msg: cmd x=1; if [ $x == 0 ];then exit 1 ;fi failed. exit_status=-1, errno=32: Broken pipe I am getting this error but If I include this if(exec_msg('x=0; if [ $x == 0 ];then exit 1 ;fi')) it works. Please tell me what can I do? As described below my plan is to call java program from … [View More]

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[Serusers] Presence Module with Ser
by Warren McCoubrey 20 Feb '06

20 Feb '06

Hello, I received the following errors when attempting to compile the source related to ser-0.10.99-dev30-tm-timers-pa-3. Any ideas on what is causing the problem? I was able to successfully compile and run ser-0.9.6 and want to add the presence patch. In file included from route_struct.h:44, from route_struct.c:41: /usr/include/regex.h:362: parse error before "size_t" /usr/include/regex.h:369: parse error before ':' token /usr/include/regex.h:378: parse error … [View More]

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[Serusers] Assymetric RTP and Symmetric RTP
by George Lambson 20 Feb '06

20 Feb '06

I guess this is the proper terminology I am looking for. I have an Asymmetric UA and an Symmetric UA that I amtrying to connect through SER and Mediaproxy. The Asymmetric UA does not register, rather it is on the LAN and is completely accessible to SER. The Symmetric UA however is on the internet and would likely be going through a NAT--hence the need for SER. The problem that I am having is that SER and Mediaproxy are ignoring the SDP info from the Asymmetric UA and sending the RTP stream … [View More]

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Fwd: [Serusers] Invalid table version AGAIN
by Yi Zheng 20 Feb '06

20 Feb '06

Hi, I found the problem is that the new ser_mysql.sh did not create a subscriber entry in its version table. Now I am trying to use admin/heslo to login and got 500 internal server error. Here is the log: Feb 17 17:31:46 localhost ./ser[9700]: parsing accept header: application/watcherinfo+xml Feb 17 17:31:46 localhost ./ser[9700]: winfo_implicit_auth(sip:admin@10.1.104.254): enabled for sip:admin@10.1.104.254 Feb 17 17:31:46 localhost ./ser[9700]: handle_subscription about to … [View More]return 1: w->event_package=2 w->accept=196618 p->flags=4 w->flags=1 w=0xb61c33e4 Feb 17 17:31:47 localhost pam_timestamp_check: pam_timestamp: `/' owner UID != 0 Feb 17 17:31:52 localhost ./ser[9708]: ! unregistering callback to admin, 0xb61c3048 Feb 17 17:31:52 localhost ./ser[9708]: ! unregistered callback to admin, 0xb61c3048 Feb 17 17:31:52 localhost pam_timestamp_check: pam_timestamp: `/' owner UID != 0 Feb 17 17:32:27 localhost last message repeated 7 times Feb 17 17:32:31 localhost ./ser[9701]: new_presentity_no_wb=0xb61c3048 for uri=sip:admin@10.1.104.254 uuid=admin Feb 17 17:32:31 localhost ./ser[9701]: ! registering callback to admin, 0xb61c3048 Feb 17 17:32:31 localhost ./ser[9701]: read_presentity(ns=urn:ietf:params:xml:ns:pidf)Feb 17 17:32:31 localhost ./ser[9701]: read_tuple() Feb 17 17:32:31 localhost ./ser[9694]: parsing accept header: application/watcherinfo+xml Feb 17 17:32:31 localhost ./ser[9694]: winfo_implicit_auth(sip:admin@10.1.104.254): enabled for sip:admin@10.1.104.254 Feb 17 17:32:31 localhost ./ser[9694]: handle_subscription about to return 1: w->event_package=2 w->accept=196618 p->flags=5 w->flags=1 w=0xb61c33e4 Feb 17 17:32:32 localhost pam_timestamp_check: pam_timestamp: `/' owner UID != 0 Feb 17 17:32:33 localhost ./ser[9707]: submit_query: Unknown column 'flags' in 'field list' Feb 17 17:32:33 localhost ./ser[9707]: db_query: Error while submitting query Feb 17 17:32:33 localhost ./ser[9707]: auth_db:get_ha1: Error while querying database Feb 17 17:32:37 localhost pam_timestamp_check: pam_timestamp: `/' owner UID != 0 Can someone please shed light on this? - ming Yi Zheng <yizheng(a)sbcglobal.net> wrote: Date: Fri, 17 Feb 2006 15:28:18 -0800 (PST) From: Yi Zheng <yizheng(a)sbcglobal.net> To: ser sip <serusers(a)lists.iptel.org> Subject: [Serusers] Invalid table version AGAIN I was able to have SER working with MySql database. I can see the location table updated with the registered users. However if I add the following line to ser.cfg to use digest authentication and restart ser, then I saw the following errors in /var/log/messages and ser failed to start. ser.cfg ---------- # Uncomment this if you want to use digest authentication if (!www_authenticate("10.1.104.254", "subscriber")) { www_challenge("10.1.104.254", "0"); break; }; /var/log/messages: -------------------------- Feb 17 16:25:39 localhost ./ser[32533]: authdb_fixup: Invalid table version (use ser_mysql.sh reinstall) Feb 17 16:25:39 localhost ./ser[32533]: ERROR: fix_expr : fix_actions error I am using the presence server snapshot. There is no reinstall for this version of ser_mysql.sh. And I made no change to the database or version table. So what is it complaining? I found this strange. - ming _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers [View Less]

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[Serusers] Re: Calling External Script from Ser.cfg
by mkumar＠mantragroup.com 20 Feb '06

20 Feb '06

Hi Group, After some search I found exec_msg return false, if return value of script does not equals zero, so from shell script I must collect java program output and based on that I must return some value. Thanks, Manoj. > > Quoting mkumar(a)mantragroup.com: > >> Hi All, >> >> I am trying to call a external script from Ser.cfg using exec_msg. >> It calls a >> shell script and from shell script I am running a java program which >> connects >&… [View More]

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[Serusers] Sems Thread Memory
by Nicky 20 Feb '06

20 Feb '06

Dear All, I am having a serious problem, my thread memory size is growing to up to 2GB which applies to SEMS, SER and Mysqld thread. Please help. I am testing with about 10 people calling in at the same time. What shall I do? :( Nicky ----- Original Message ----- From: Nicky To: serusers(a)lists.iptel.org ; sems(a)lists.iptel.org Sent: Saturday, October 29, 2005 11:44 AM Subject: pthread create failed with code 11 Hi Support, I have experienced this error, please advice … [View More]

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[Serusers] Calling External Script from Ser.cfg
by mkumar＠mantragroup.com 19 Feb '06

19 Feb '06

Hi All, I am trying to call a external script from Ser.cfg using exec_msg. It calls a shell script and from shell script I am running a java program which connects to Database and makes some queries for getting number of minutes dialled out by a customer. My problem is that how can I get the output of the java programme value to ser.cfg? I am doing like this exec_msg("sh -a /usr/local/ser/mobi.sh $SIP_USER;Some more commands required") I tried one approach, if the user crossed number of … [View More]

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RE: [Serusers] UAC UDP to Proxy TCP... Problem
by Christian Thomas 18 Feb '06

18 Feb '06

Hi Greg, I'm not sure. I have tried : dig -t SRV _sip._udp.sip.carrierTCP.net.br and dig -t SRV _sip._tcp.sip.carrierTCP.net.br without success. then dig -t SRV _sip._udp.carrierTCP.net.br and dig -t SRV _sip._tcp.carrierTCP.net.br. The same. In this case it could be the problem... I'm verifying with him. If not, I could use T_relay_to_tcp or t_relay_to_udp depending on the request line. In this case my problem is to put the right uri and port in the function : as his GW could … [View More]change regarding the destination, even the transport could change for the same GW. And the function seems to work so with ("123.456.123","5060") as parameters and not with variables. Am I wrong? I hope because if I could put variables in the t_relay_to_xxx functions, I will be able to work with any other Sip proxy, even if not well configured with DNS SRV records. Anyway, thank Greg...I know you will give me the trick. Regards, Christian Thomas -----Original Message----- From: Greger V. Teigre [mailto:greger@teigre.com] Sent: quinta-feira, 16 de fevereiro de 2006 05:22 To: cthomas(a)canalwest.com; serusers(a)lists.iptel.org Subject: Re: [Serusers] UAC UDP to Proxy TCP... Problem Does sip.carrierTCP.net.br have DNS SRV records?! g-) ----- Original Message ----- From: Christian Thomas To: Greger V. Teigre ; serusers(a)lists.iptel.org Sent: Wednesday, February 15, 2006 11:28 PM Subject: RE: [Serusers] UAC UDP to Proxy TCP... Problem Greger, Thank you for your response. See the attached ethereal trace. NOTA: 66.xxx.xxx.xx is the aterisk UAC 200.xxx.xxx.xx is my proxy server 20.196.xxx.xx is the TCP Carrier GW The ACK my proxy send to the TCP GW - after have received 200 OK - have the request line including the transport=tcp indication The contact headers are ok in each case. The Bye Sent by my proxy does not have the request line transport=tcp. But we have already a one way communication before. if the asterisk does not send a bye, the call is terminated by the TCP GW. I think it is because of a time out from the TCP GW. Christian -----Original Message----- From: Greger V. Teigre [mailto:greger@teigre.com] Sent: quarta-feira, 15 de fevereiro de 2006 04:35 To: cthomas(a)canalwest.com; serusers(a)lists.iptel.org Subject: Re: [Serusers] UAC UDP to Proxy TCP... Problem Seems like you have done your research and what you write seems reasonable. It's difficult to give you an answer without having a complete ngrep trace. However, you should also look at the Record-Route and Route headers... g-) ----- Original Message ----- From: Christian Thomas To: serusers(a)lists.iptel.org Sent: Tuesday, February 14, 2006 6:33 PM Subject: [Serusers] UAC UDP to Proxy TCP... Problem Hi everybody, I have a UDP2TCP problem with Ser. (0.9-4) The UACs connected to my SER Proxy are all UDP. When I use some carriers to connect them to PSTN The Carrier Proxy / GW could talk UDP or TCP. It depends on the destination. Each time the Carrier GW want to talk TCP, the contact field is correct and contains the 'transport=tcp' mention. all seems to be correct and, after a complete reading about this subject in the list, I trust that using t_relay() must do the right translation and send response via TCP to the Carrier GW when this one is transport=tcp declared. If I Check the message for an Ack or a BYE sent to the TCP GW the uri is correct : the transport=tcp is added. Request-Line: ACK sip:50033299848400@200.1xx.xx.xx:5060;transport=tcp SIP/2.0 I have red that this is sufficient to let SER understand that it must be sent via TCP. unfortunately, this message is ent thru UDP. May I force with a t_relay_to_tcp ? As this case depends on the carrier dynamic routing it could be complicated because of the need to know the destination to script t_relay_to_tco(uri, port)... If anybody has the trick or any advice... Other thing related, when TCP is the transport mode from the Carrier GW and I forward staefully to the UAC (Asterisk) the call failed between Asterisk as UAC and the carrier GW. I wondering it was because the Carrier Gw didn't receive an ACK TCP from my server.. Well I'm completely lost... as you can feel with my message... I need Help.. But don't cal me Harry ;) Regards, Christian Thomas ---------------------------------------------------------------------- _______________________________________________ Serusers mailing list serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers [View Less]

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RE: [Serusers] Does SER need a public IP interface?
by Bart J. Smit 18 Feb '06

18 Feb '06

No, rtpproxy. Is there a difference in NAT support? Bart... -----Original Message----- From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] On Behalf Of George Lambson Sent: 18 February 2006 15:54 To: serusers(a)lists.iptel.org Subject: Re: [Serusers] Does SER need a public IP interface? Are you using mediaproxy? >>> "Bart J. Smit" <bart(a)smits.co.uk> 2/18/2006 7:16 AM >>> I have based my attached SER config file on the very informative '… [View More]

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Re: [Serusers] Does SER need a public IP interface?
by George Lambson 18 Feb '06

18 Feb '06

Obviously the easiest way to set it up is with a public interface, but I think you can do what you described if you can modify your NAT box and the SER setup. try this site: http://openser.org/dokuwiki/doku.php?id=openser_core_cookbook It has core functions for SER that might get you going in the right direction. >>> "Bart J. Smit" <bart(a)smits.co.uk> 2/18/2006 7:16 AM >>> I have based my attached SER config file on the very informative 'getting started 5' … [View More]

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Re: [Serusers] Does SER need a public IP interface?
by George Lambson 18 Feb '06

18 Feb '06

Are you using mediaproxy? >>> "Bart J. Smit" <bart(a)smits.co.uk> 2/18/2006 7:16 AM >>> I have based my attached SER config file on the very informative 'getting started 5' instructions/sample from http://www.onsip.org/. I have SER on a DMZ with Asterisk on the same box, listening on 5065. I can register two internal phones with SER and talk. I can register a phone to the Asterisk and talk to the rest of the Asterisk infrastructure through IAX2. I entered a log … [View More]

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[Serusers] Does SER need a public IP interface?
by Bart J. Smit 18 Feb '06

18 Feb '06

I have based my attached SER config file on the very informative 'getting started 5' instructions/sample from http://www.onsip.org/. I have SER on a DMZ with Asterisk on the same box, listening on 5065. I can register two internal phones with SER and talk. I can register a phone to the Asterisk and talk to the rest of the Asterisk infrastructure through IAX2. I entered a log statement for my forward to asterisk and I can see that it is being executed. What I can't get to work is external … [View More]

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[Users] Permissions and UnixODBC incompatability
by Lenir 18 Feb '06

18 Feb '06

Hey guys, The permissions module doesn't work with unixodbc module, but it works fine with mysql or postgres. Here's the debut output: 0(0) DEBUG: init_mod: permissions 0(0) permissions - initializing 0(0) WARNING: File not found: /usr/local/etc/openser/permissions.allow 0(0) Default allow file (/usr/local/etc/openser/permissions.allow) not found => empty rule set 0(0) WARNING: File not found: /usr/local/etc/openser/permissions.deny 0(0) Default deny file (/usr/local/etc/… [View More]openser/permissions.deny) not found => empty rule set 0(0) find_mod_export: found <db_use_table> in module unixodbc [/usr/local/lib/openser/modules/unixodbc.so] 0(0) find_mod_export: found <db_init> in module unixodbc [/usr/local/lib/openser/modules/unixodbc.so] 0(0) find_mod_export: found <db_close> in module unixodbc [/usr/local/lib/openser/modules/unixodbc.so] 0(0) find_mod_export: found <db_query> in module unixodbc [/usr/local/lib/openser/modules/unixodbc.so] 0(0) find_mod_export: found <db_raw_query> in module unixodbc [/usr/local/lib/openser/modules/unixodbc.so] 0(0) find_mod_export: found <db_free_result> in module unixodbc [/usr/local/lib/openser/modules/unixodbc.so] 0(0) find_mod_export: found <db_insert> in module unixodbc [/usr/local/lib/openser/modules/unixodbc.so] 0(0) find_mod_export: found <db_delete> in module unixodbc [/usr/local/lib/openser/modules/unixodbc.so] 0(0) find_mod_export: found <db_update> in module unixodbc [/usr/local/lib/openser/modules/unixodbc.so] 0(0) find_mod_export: found <db_replace> in module unixodbc [/usr/local/lib/openser/modules/unixodbc.so] 0(0) db_init: Connection 'unixodbc://username:password@xx.xx.xx.xx/sybasetds' not found in pool 0(0) DEBUG:unixodbc:build_conn_str: connection string is <DSN=sybasetds;UID=username;PWD=password;> 0(0) DEBUG:unixodbc:new_connection: connection succeeded with reply <DSN=sybasetds;UID=username;PWD=password;> 0(0) DEBUG: register_fifo_cmd: new command (trusted_reload) registered 0(0) DEBUG: register_fifo_cmd: new command (trusted_dump) registered 0(0) unixsock_register_cmd: New command (trusted_reload) registered 0(0) unixsock_register_cmd: New command (trusted_dump) registered 0(0) Number of rows in trusted table: 3 0(0) ERROR: permissions: trusted_reload(): Database problem 0(0) pool_remove: Removing connection from the pool 0(0) init_trusted(): Reload of trusted table failed 0(0) Error while initializing allow_trusted function Here's the trusted table, all columns are varchar src_ip proto from_pattern -------------------------------- 216.82.124.34 any NULL 216.82.124.35 any NULL 216.82.124.62 any NULL Regards, Lenir [View Less]

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[Users] question about avp_subst
by unplug 18 Feb '06

18 Feb '06

If I have a avp variable s:contact= <sip:871966806561@10.0.0.46:5060> How can I use avp_subst to get 871966806561?

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[Serusers] Full Duplex Audio
by George Lambson 18 Feb '06

18 Feb '06

Does anyone know how to acheive full duplex audio using SER and Mediaproxy? Turns out my call center server does full duplex audio by sending RTP out one port and recieving it on another. The problem is that I want Mediaproxy to send the audio down the right stream, but it doesn't use the info in the SDP that it gets, instead it sends it back on the port it recieved it from Any help would be great. Thanks.

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[Serusers] Invalid table version AGAIN
by Yi Zheng 17 Feb '06

17 Feb '06

I was able to have SER working with MySql database. I can see the location table updated with the registered users. However if I add the following line to ser.cfg to use digest authentication and restart ser, then I saw the following errors in /var/log/messages and ser failed to start. ser.cfg ---------- # Uncomment this if you want to use digest authentication if (!www_authenticate("10.1.104.254", "subscriber")) { … [View More]

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Fwd: [Serusers] error querying table version
by Yi Zheng 17 Feb '06

17 Feb '06

I deleted all rows in the version table and manually created the table based on the ser_mysql.sh. But now I am seeing a different set of error messages. I searched ser code but can't find where it comes from. What could be wrong? Feb 17 12:07:46 localhost ./ser[24061]: submit_query: Unknown column 'uid' in 'field list' Feb 17 12:07:46 localhost ./ser[24061]: db_query: Error while submitting query Feb 17 12:07:46 localhost ./ser[24061]: preload_udomain(): Error while doing db_query Feb … [View More]

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Re: AW: [Users] question about db_schemes and avp functionality
by Daniel-Constantin Mierla 17 Feb '06

17 Feb '06

Hello, On 02/17/06 17:03, Broch, Andreas wrote: > Hi > We installed todays latest openserdev-cvs version on the system and tested the avp_db_query() function again, but its still not working. > > Here an example of the current syntax > avp_db_query("select column_name1,column_name2 from mytable where username='$fu/username'","$avp(s:value1);$avp(s:value2)") > "/username" is a flag for parameter of avp_db_load() or other functions. The pseudo-variable referring to From … [View More]username is $fU. Try: avp_db_query("select column_name1,column_name2 from mytable where username='$fU'","$avp(s:value1);$avp(s:value2)") Cheers, Daniel > => Problem: No debug Errors, but no Action > The only difference is that, openser now knows the Function( produces no Errors on startup). > > But if I load the values from my self defined scheme(s), the data is loaded correctly into the avps (memory). > > For example > avp_db_load("$fu/username","$avp(s:anyvalue)/$scheme1") > > => Problem new avp's cant be stored to database ! > > -----Ursprüngliche Nachricht----- > Von: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] > Gesendet: Donnerstag, 16. Februar 2006 10:50 > An: Broch, Andreas > Cc: users(a)openser.org > Betreff: Re: [Users] question about db_schemes and avp functionality > > Hello, > > > On 02/15/06 17:53, Broch, Andreas wrote: > >> Hi >> >> We have a running IMS test - system without HSS functionality at the >> moment. >> >> Next step to realize are the Initial Filter Criteria. >> >> -> >> >> I want to create my own db_scheme, where I can store the required >> values of the user profile. >> >> My questions are: >> >> - How can use the db_scheme beside the default avp_table ( >> usr_preferences ) ? >> >> > db_scheme definition allows to specify a custom table name. See: > http://www.voice-system.ro/docs/avpops/ar01s05.html#db_scheme > > If the table has the same structure as usr_preferences, then you can > give the name in the second parameter of avp_db_load(). > http://www.voice-system.ro/docs/avpops/ar01s06.html#avp_db_load > >> - Avp's only have 1 value column. How can I load different attributes >> , if the parameters are stored in different columns of the self >> defined table >> >> > If they are in the subscriber table, then you can load them upon > authentication: > http://openser.org/docs/modules/1.1.x/auth_db.html#AEN147 > > Otherwise, with the latest cvs version (it was committed yesterday), you > can use avp_db_query() > http://openser.org/docs/modules/1.1.x/avpops.html#AEN239 > > Cheers, > Daniel > > >> Mfg >> >> Andreas Broch >> >> FTW >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> Users mailing list >> Users(a)openser.org >> http://openser.org/cgi-bin/mailman/listinfo/users >> >> > > [View Less]

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Re: [Serusers] Re: SerWeb Displays blank page
by mkumar＠mantragroup.com 17 Feb '06

17 Feb '06

Finally my problem is solved, php code in page.php under html dir caused problems. After calls to virtual() nothing happens, php dies and displays blank page. Now I need to fix this php problem. > Quoting mkumar(a)mantragroup.com: > >> Hello Serusers, >> >> Now I fixed all my php,apache errors and again I have the same blank page >> problem. Previously we had serweb on a different machine and even >> there we had >> problems. So especially for serweb … [View More]we took a new unix box and >> installed serweb >> there and even here we had another problem. In our previous machine >> page use to >> display after 5 to 10 attempts but here it always displays the title page as >> "SIP Express Router Web interface" but I could never find the login >> screen. My >> phpinfo page is http://serweb.mantragroup.com/ser/html/admin/phpinfo.php I >> eneabled display errors and log errors for php >> >> My admin login page is >> >> http://serweb.mantragroup.com/ser/html/admin/index.php >> >> Please check page source of admin index.php and I have learnt from >> my previous >> installation that it should have some java script etc for displaying >> username, >> password etc but this page do not have. I also checked paths for >> style.css and >> login_completion.js.php both are correct. What else should I do? I posted in >> some php forums also but could not fix. Please help me understand and fix my >> problem. I will send if any config files are needed. >> >> Thanks for your help and time, >> Manoj. >> >> >> Quoting Karel Kozlik <karel(a)iptel.org>: >> >>>> Sorry for that, now I changed some paths. I got this error >>>> >>>> Warning: main(): open_basedir restriction in effect. >>>> File(/usr/share/pear/DB.php) is not within the allowed path(s): >>>> >>>> So I used /tmp dir, anyway I changed to /usr/shae/pear/. I am >>>> searching for >>>> fixing this errors. I changed /etc/php.ini open_basedir global >>>> value but unable >>>> to change local value so I still have same problem? >>> >>> open_basedir should not be set. And as I see it is not. So it is correct. >>> >>> The error messages I see on your serweb now say you have not >>> installed xml_rpc package from PEAR. Strange - it should be >>> installed by default with PEAR. >>> >>> You may install it by type: >>> >>> pear install xml_rpc >>> >>> >>> Karel >>> >>> >>> >>> >>> >>>> >>>> Quoting Karel Kozlik <karel(a)iptel.org>: >>>> >>>>>> I enabled logging and also restarted Apache but even now I could >>>>>> not find any >>>>>> error message and still it displays blank page. Please check my >>>>>> phpinfo at >>>>> >>>>> To me it displays this errors: >>>>> >>>>> Warning: main(./PEAR.php): failed to open stream: No such file or >>>>> directory in /tmp/DB.php on line 26 >>>>> >>>>> Fatal error: main(): Failed opening required './PEAR.php' >>>>> (include_path='.:/usr/share/pear') in /tmp/DB.php on line 26 >>>>> >>>>> >>>>> It signify there is something wrong in your PEAR library. Do you >>>>> have installed it correctly? The path /tmp/... is strange for me. >>>>> >>>>> Karel >>>>> >>>>>> http://serweb.mantragroup.com/serweb/html/admin/test.php Is >>>>>> there anything >>>>>> wrong with this related to serweb. >>>>>> >>>>>> Quoting Karel Kozlik <karel(a)iptel.org>: >>>>>> >>>>>>> Hi, >>>>>>> if no error message is logged it means logging is disabled.. :-) >>>>>>> >>>>>>> Set this values in your php.ini: >>>>>>> >>>>>>> error_reporting = E_ALL >>>>>>> log_errors = On >>>>>>> (for write errors to log file) >>>>>>> >>>>>>> and/or >>>>>>> >>>>>>> display_errors = On >>>>>>> (for display error in browser) >>>>>>> >>>>>>> >>>>>>> Karel >>>>>>> >>>>>>> >>>>>>> mkumar(a)mantragroup.com napsal(a): >>>>>>>> Hi All, >>>>>>>> >>>>>>>> I am using serweb stable 0.9.2 version and I followed all the >>>>>>>> steps in INSTALL >>>>>>>> file that comes with the installation. After my configuration >>>>>>>> when I try to >>>>>>>> load the url >>>>>>>> >>>>>>>> http://serweb.mantragroup.com/serweb/html/admin/index.php >>>>>>>> >>>>>>>> strangly nothing is getting displayed and no error messages >>>>>>>> are logged in any >>>>>>>> log. I am using >>>>>>>> >>>>>>>> Apache/2.0.51 >>>>>>>> PHP 4.4.1 >>>>>>>> Mysql 4.0.25 >>>>>>>> >>>>>>>> Did anyone experience this problem before? Please tell me what >>>>>>>> should I do to >>>>>>>> configure serweb properly? >>>>>>>> >>>>>>>> Thanks, >>>>>>>> Kumar. >>>>>>>> >>>>>>> >>>>>> >>>>>> >>>>>> >>>>> >>>> >>>> >>>> >>> >> >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> > > > > [View Less]

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[Users] RE: TLS Cipher error?
by Tóth Péter 17 Feb '06

17 Feb '06

Helo! The connection failed between openser and avaya, as you can see in sshdump, because the first handshake does not succeed. The reason of the failure, i think in the cipher, sent by avaya: TLS_DHE_RSA_WITH_DES_CBC_SHA After this package send a FIN, and close the connection! This is just 1 cipher, but in openssl must be send a dozen of ciphers, or not? Openser can not choose from one... I tried with an another openssl connection to an other machine (with s_client),and the connection was … [View More]

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[Serusers] Verify sip number in ser.cfg failed...
by jimmy 17 Feb '06

17 Feb '06

Hi All, I am using ser 0.9.6, and refer the doc (http://www.iptel.org/ser/doc/seruser/seruser.html#CONDITIONS) and I want to use number 11 to 19 for pstn number, but in the first step to verify the pstn number failed. my code in ser.cfg, ******************************************* case 1: test sip client register number is 11 Sip uri is 11(a)192.168.5.1 if(uri=~"^sip:1[1-9]@"){ log(1,"LOG: number is 11-19\n") }else{ log(1,"LOG: NOT 11-19\n") } … [View More]

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[Users] TLS Cipher error?
by Tóth Péter 17 Feb '06

17 Feb '06

Helo! Thans a lot for last help, I started succesfully the openser 1.0.0.0 tls version, but I can not connent to an avaya pbx. The ssldump's result is the following: New TCP connection #1: avaya_IP(14463) <-> debianom(5061) 1 1 0.0060 (0.0060) C>S Handshake ClientHello Version 3.1 cipher suites TLS_DHE_RSA_WITH_DES_CBC_SHA compression methods NULL 1 2 0.0572 (0.0512) S>C Alert level fatal … [View More]

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Re: [Users] multiple 487 created
by unplug 17 Feb '06

17 Feb '06

Hi, I still got the problem without any idea how to fix it. Before stating the problem, I would like to point out the configuration. I have only one sip phone registered using mediaproxy. In the location table, I have the following field. contact: sip:871966760539@10.0.0.56:5060 received: sip:210.184.23.31:15060 socket: 203.193.26.234_5060 >From the call flow below, I found that there is some strange behaviour or mismatch. http://meerkat.no-ip.com/openser/calldrop_index.html I wonder … [View More]

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[Serusers] What's the point of domains?
by Nick Hoffman 17 Feb '06

17 Feb '06

Hi guys, just a quick question. This is the only 'listen' in my ser.cfg: listen=10.0.100.12 When I add a user like so: serctl add nick(a)foo.bar password nick(a)domain.tld and dial 'nick' or 'sip:nick@10.0.100.12' on a softphone, the call connects and SER's debug messages show that this is being dialled: sip:nick@10.0.100.12 If I try dialing sip:nick@foo.bar a 404 is returned. If the domain that's specified when adding a user isn't actually used in the user's SIP address, what … [View More]

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Re: [Serusers] features-callfwd.5.0.cfg to Asterisk VM
by zhangshuai 17 Feb '06

17 Feb '06

Dear Ali Zaidi, I'm testing the forwarding now, too. I think you could be more aware after read SER-GettingStarted doc at ONsip.org. And you may want to try the onr.cfg, or redirect.cfg, which are much more concise than features-callfwd.5.0.cfg, in the directory of examples of SER src package, first. >Hi, > >I'm having some problem forwarding to Asterisk VM. What i want is when ever >486, 404, 408 or 480 error triger then ser should forward INVITE to >Asterisk. >Some one … [View More]

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[Serusers] Re: SerWeb Displays blank page
by Karel Kozlik 16 Feb '06

16 Feb '06

> Sorry for that, now I changed some paths. I got this error > > Warning: main(): open_basedir restriction in effect. > File(/usr/share/pear/DB.php) is not within the allowed path(s): > > So I used /tmp dir, anyway I changed to /usr/shae/pear/. I am searching for > fixing this errors. I changed /etc/php.ini open_basedir global value but > unable > to change local value so I still have same problem? open_basedir should not be set. And as I see it is not. So it is … [View More]correct. The error messages I see on your serweb now say you have not installed xml_rpc package from PEAR. Strange - it should be installed by default with PEAR. You may install it by type: pear install xml_rpc Karel > > Quoting Karel Kozlik <karel(a)iptel.org>: > >>> I enabled logging and also restarted Apache but even now I could not >>> find any >>> error message and still it displays blank page. Please check my >>> phpinfo at >> >> To me it displays this errors: >> >> Warning: main(./PEAR.php): failed to open stream: No such file or >> directory in /tmp/DB.php on line 26 >> >> Fatal error: main(): Failed opening required './PEAR.php' >> (include_path='.:/usr/share/pear') in /tmp/DB.php on line 26 >> >> >> It signify there is something wrong in your PEAR library. Do you have >> installed it correctly? The path /tmp/... is strange for me. >> >> Karel >> >>> http://serweb.mantragroup.com/serweb/html/admin/test.php Is there >>> anything >>> wrong with this related to serweb. >>> >>> Quoting Karel Kozlik <karel(a)iptel.org>: >>> >>>> Hi, >>>> if no error message is logged it means logging is disabled.. :-) >>>> >>>> Set this values in your php.ini: >>>> >>>> error_reporting = E_ALL >>>> log_errors = On >>>> (for write errors to log file) >>>> >>>> and/or >>>> >>>> display_errors = On >>>> (for display error in browser) >>>> >>>> >>>> Karel >>>> >>>> >>>> mkumar(a)mantragroup.com napsal(a): >>>>> Hi All, >>>>> >>>>> I am using serweb stable 0.9.2 version and I followed all the steps >>>>> in INSTALL >>>>> file that comes with the installation. After my configuration when >>>>> I try to >>>>> load the url >>>>> >>>>> http://serweb.mantragroup.com/serweb/html/admin/index.php >>>>> >>>>> strangly nothing is getting displayed and no error messages are >>>>> logged in any >>>>> log. I am using >>>>> >>>>> Apache/2.0.51 >>>>> PHP 4.4.1 >>>>> Mysql 4.0.25 >>>>> >>>>> Did anyone experience this problem before? Please tell me what >>>>> should I do to >>>>> configure serweb properly? >>>>> >>>>> Thanks, >>>>> Kumar. >>>>> >>>> >>> >>> >>> >> > > > [View Less]

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[Serusers] Clients connected - can call, but no voice
by René Nielsen 16 Feb '06

16 Feb '06

The Server (ser 0.9.3) is behind NAT - together with the one client, both clients can call the other part, Part a (behind nat) - cannot hear what part B says, but part B can hear what part A says.. Part b is connected directly to internet, throug a Sipura 2100.. Someone with a Idea what could be wrong? René (ports open in NAT router are from 5055 - 5075 and 8100 - 65000) those ports point to the SER server

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[Serusers] "db error: connect failed" in serweb
by Marco Passarinho 16 Feb '06

16 Feb '06

I have this error anytime I try to access serweb. Im sure the username and password are correct. I use the psql to acess the postgres db and the bd is running Please help me :( username: postgres@localhost pass: pass

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[Users] prefix routing for one account
by Peter Hudec 16 Feb '06

16 Feb '06

hi all, I need to route prefix to te one subscriber. It's something between PDT and ALIAS. ALIAS The R-URI shoult not be replaced by one in the alias table PDT The voip domain is rewrited, but I need to leave the old one. WHY ??? I want to connect some SIP PBX (asterisk) to our VoIP network and give them more numbers / prefix. I dont want to put it into routing logic eq (if sip:0123XX@domain; then). The best case is to call some function as PREFIX, which will lookup into PREFIX, SUBCRIBER … [View More]

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[Users] openser.init script Wiki Added
by Glenn Dalgliesh 16 Feb '06

16 Feb '06

I have added some notes to the wiki about getting the sip-server/etc/openser.ini script to work. It by default requires monit a program that I had never come across but decided to give a try. Anyway I am sure a lot more could be added but this is at least a start to get the script operational. Under Installation | Make openser.init scipt work http://www.openser.org/dokuwiki/doku.php?id=make_openser.init_scipt_work Enjoy!

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[Serusers] reviving the FreeBSD port
by Hendrik Scholz 16 Feb '06

16 Feb '06

Hi! FreeBSD carries SER 0.9.3 in net/ser and I wonder if somebody is interested in having an up-to-date version. I had a look at it before but the amount of patches included and work that has to be done only makes sense if people actually use the port. If you either use the port or are interested in using it please drop me a private mail. If you like to contribute to the cause feel free to do so ;) Thanks for your time, Hendrik -- Hendrik Scholz - <hscholz(a)raisdorf.net> - http://… [View More]

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Re: [Users] multiple 487 created
by unplug 16 Feb '06

16 Feb '06

Hi, I am using the features-callfwd.5.0.cfg from onsip (configs.GettingStarted.tar.gz). I have taken a callflow for the message as below. http://meerkat.no-ip.com/openser/feature-forward_index.html In frame 17, a cancel is created. From frame 22 to 29, a serier of 487 and ACK is created. After a few seconds, 487 is generated in frame 50. And then you can still see 487 later. Just want to ask it is the normal situation for openser to generate so much 487? As I am using mediaproxy for NAT.… [View More] Does it cause that problem? unplug On 2/15/06, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > Hi! > > I'm not sure, but the ACK has a different branch parameter in the Via > header, than the CANCEL and INVITE. I guess this will cause that openser > wont find the corresponding transaction > > klaus > > On Wed, February 15, 2006 11:16, unplug said: > > Thanks! But I don't really understand why there is a mismatch between > > 487and ACK. Below is a part of sip message pair of 487 and ACK. > > > > Below is the complete log from ngrep. (From line 290, openser creates > > 487 and line 304 UA1 ACK. After that, multiple 487 are created by > > openser) > > http://meerkat.no-ip.com/openser/cancel_normal.log > > > > Below is my openser.cfg for reference > > http://meerkat.no-ip.com/openser/openser.cfg > > > > 290 U 203.193.26.226:5060 -> 210.184.23.31:5060 > > 291 SIP/2.0 487 Request Cancelled. > > 292 Via: SIP/2.0/UDP > > 10.0.0.46:5060;rport=5060;received=210.184.23.31;branch=z9hG4bKdUyOnG9Zel82UECI. > > 293 From: > > "sip:36418488@o01.ol.com"<sip:871966806561@o01.ol.com>;tag=xLTdFs7yGkeXq3wn. > > 294 To: "34163634" <sip:34163634@o01.ol.com>;tag=A81AFED4-21D7. > > 295 Date: Tue, 15 Feb 2005 01:52:57 GMT. > > 296 Call-ID: Z0wcK4WwdTU6mH00(a)10.0.0.46. > > 297 Server: Cisco-SIPGateway/IOS-12.x. > > 298 CSeq: 1 INVITE. > > 299 Allow-Events: telephone-event. > > 300 Content-Length: 0. > > 301 . > > 302 > > 303 # > > 304 U 210.184.23.31:5060 -> 203.193.26.226:5060 > > 305 ACK sip:34163634@o01.ol.com SIP/2.0. > > 306 Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bKdUyOnG9Zel82UECI. > > 307 Route: > > <sip:34163634@203.193.26.226:5060;nat=yes;ftag=xLTdFs7yGkeXq3wn;lr=on>. > > 308 Max-Forwards: 70. > > 309 User-Agent: Koncept KE10XX v4.32.10 00-09-45-0a-fc-2d. > > 310 From: "871966806561" > > <sip:871966806561@o01.ol.com>;tag=xLTdFs7yGkeXq3wn. > > 311 To: "34163634" <sip:34163634@o01.ol.com>;tag=A81AFED4-21D7. > > 312 Call-ID: Z0wcK4WwdTU6mH00(a)10.0.0.46. > > 313 Contact: <sip:871966806561@10.0.0.46:5060>. > > 314 CSeq: 1 ACK. > > 315 Content-Length: 0. > > > > In my openser.cfg line 184-187, it is remarked. Do you mean it should > > un-remarked for handling ACK according to your point 2? > > > > On 2/15/06, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > >> This means that openser does not receive the ACK or is not able to match > >> the ACK to the canceled transaction. Thus it is re-transmitting the 487. > >> > >> Increas the debug level and take a look at the syslog to find out if: > >> 1. the ACK to the 487 is received by openser > >> 2. if it is received, check if there is a problem during handling of the > >> ACK. (for these kind of ACKs just t_relay()). > >> > >> klaus > >> > >> On Wed, February 15, 2006 5:13, unplug said: > >> > For the normal cancel situation. > >> > > >> > UA1 OPENSER UA2 > >> > --------INVITE-----------------> > >> > <-----------100----------------- > >> > --------------------INVITE----------> > >> > <---------------100------------------- > >> > ----------CANCEL----------> > >> > --------------CANCEL-------------> > >> > <--------------200------------------ > >> > <------------487------------------- > >> > --------------ACK-----------------> > >> > <---------487--------------- > >> > -------------ACK------------> > >> > > >> > But in my case, I found that there are many 487 created from openser > >> > to UA1. The multiple 487 may cause a call drop in the following call. > >> > Why does openser create so many 487 for UA1? How can I prevent it? > >> > > >> > UA1 OPENSER UA2 > >> > --------INVITE-----------------> > >> > <-----------100----------------- > >> > --------------------INVITE----------> > >> > <---------------100------------------- > >> > ----------CANCEL----------> > >> > --------------CANCEL-------------> > >> > <--------------200------------------ > >> > <------------487------------------- > >> > --------------ACK-----------------> > >> > <---------487--------------- > >> > -------------ACK------------> > >> > <---------487------------- > >> > -------------ACK------------> > >> > <---------487--------------- > >> > -------------ACK------------> > >> > <---------487------------- > >> > -------------ACK------------> > >> > <---------487------------- > >> > -------------ACK------------> > >> > <---------487------------- > >> > -------------ACK------------> > >> > <---------487------------- > >> > -------------ACK------------> > >> > ... > >> > > >> > _______________________________________________ > >> > Users mailing list > >> > Users(a)openser.org > >> > http://openser.org/cgi-bin/mailman/listinfo/users > >> > > >> > >> > >> > > > > > [View Less]

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[Serusers] SerWeb Displays blank page
by mkumar＠mantragroup.com 16 Feb '06

16 Feb '06

Hi All, I am using serweb stable 0.9.2 version and I followed all the steps in INSTALL file that comes with the installation. After my configuration when I try to load the url http://serweb.mantragroup.com/serweb/html/admin/index.php strangly nothing is getting displayed and no error messages are logged in any log. I am using Apache/2.0.51 PHP 4.4.1 Mysql 4.0.25 Did anyone experience this problem before? Please tell me what should I do to configure serweb properly? Thanks, Kumar.

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[Users] question about db_schemes and avp functionality
by Broch, Andreas 16 Feb '06

16 Feb '06

Hi We have a running IMS test - system without HSS functionality at the moment. Next step to realize are the Initial Filter Criteria. -> I want to create my own db_scheme, where I can store the required values of the user profile. My questions are: - How can use the db_scheme beside the default avp_table ( usr_preferences ) ? - Avp's only have 1 value column. How can I load different attributes , if the parameters are stored in different columns of the self … [View More]

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[Users] distributed setup
by Anders Brownworth 16 Feb '06

16 Feb '06

Hi, I have an OpenSER DISPATCHER instance successfully forwarding requests to one of four OpenSER "worker" instances. I need to authorize by originating IP on the worker machines but of course requests are only coming from the dispatcher machine. I had thought of having the dispatcher drop the originating IP in the headers but haven't been able to figure out how to do that. Something like append_hf( "origin: $si\r\n" );? I had also thought about having the dispatcher send back a 302 … [View More]

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[Serusers] Bad config file [3 errors]
by ryan nalupa 16 Feb '06

16 Feb '06

hi everyone! i've just installed ser_0.9.6 on my debian pc, included also ser-mysql and ser-jabber and ser-acc...etc. i've included my ser.cfg and restarted ser and it displays an error at the end saying Segmentation fault.. i don't know what to do with it since i'm still new to ser... here's my config file...can anyone help me work this out? ryan __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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Re: Re: Re: Re: Re: [Serusers] a question about "cancel" inforwardingscenario
by zhangshuai 16 Feb '06

16 Feb '06

Dear Greger V. Teigre, Take it easy. I really appreciate your spending time to help me figure it out. Of course I have tested the onr.cfg, a non-altered version of the onsip.org script, for many times. As I said before, the following lines: #when caller cancel, terminate the call if (t_check_status("487")) { log("cancel received\n"); t_reply("487", "Request Terminated"); break; }; in failure_route[1] never took effect in my … [View More]test, because there is no corresponding log in debug info at all! Those lines only work under this circumstance: when ua1 calls ua2, ua1 cancels the call before timer hits. So, please check the debug info, I sent out last time, again to confirm my thought. Thanks a million. >Your INVITE to seturi() to 200@ times out: >0(20167) receive_msg: cleaning up > 1(20168) DEBUG: timer routine:4,tl=0x422b9e50 next=(nil) > 1(20168) DEBUG: retransmission_handler : request resending (t=0x422b9d38, >INVITE si ... ) > 1(20168) DEBUG: add_to_tail_of_timer[5]: 0x422b9e50 > 1(20168) DEBUG: retransmission_handler : done > 1(20168) DEBUG: timer routine:5,tl=0x422b9e50 next=(nil) > 1(20168) DEBUG: retransmission_handler : request resending (t=0x422b9d38, >INVITE si ... ) > 1(20168) DEBUG: add_to_tail_of_timer[6]: 0x422b9e50 > 1(20168) DEBUG: retransmission_handler : done > 1(20168) DEBUG: timer routine:6,tl=0x422b9e50 next=(nil) > 1(20168) DEBUG: retransmission_handler : request resending (t=0x422b9d38, >INVITE si ... ) > 1(20168) DEBUG: add_to_tail_of_timer[7]: 0x422b9e50 > 1(20168) DEBUG: retransmission_handler : done > 1(20168) DEBUG: timer routine:0,tl=0x422b9e60 next=(nil) > 1(20168) DEBUG: final_response_handler:stop retr. and send CANCEL >(0x422b9d38) > 1(20168) ->>>>>>>>> T_code=100, new_code=408 > 1(20168) DEBUG: t_check: msg id=0 global id=0 T start=0x422b9d38 > 1(20168) DEBUG: t_check: T already found! > 1(20168) DEBUG:t_check_status: checked status is <408> > >then it's forwarded in t_on_failure to 300@ > >The CANCEL is received ok, forwarded ok and the 487 should NOT be met with a >t_reply, but just break in t_on_failure!! > > >As you have effectively dealt with the 487, SER only has a 408 left to send >back. > >:-( I told you to use the callfwd script from onsip.org!! But you used your >own f... up test script instead, and stupid me spent time on trying to >figure it out. Have you tried a non-altered version of the onsip.org script >at all?! > >g-( > >----- Original Message ----- >From: "zhangshuai" <zhangshuai(a)goldentek.biz> >To: "serusers" <serusers(a)lists.iptel.org> >Cc: "wuyuan" <wuyuan(a)goldentek.biz> >Sent: Wednesday, February 15, 2006 10:18 AM >Subject: Re: Re: Re: Re: [Serusers] a question about "cancel" >inforwardingscenario > > >> Dear Greger V. Teigre, >> >> >> Thanks for your suggestion. >> >> I've reset the fr_inv_timer and debug level, and put some log functions >> into ser.cfg: >> >> # ----------- global configuration parameters ------------------------ >> >> debug=7 # debug level (cmd line: -dddddddddd) >> fork=no >> log_stderror=yes # (cmd line: -E) >> >> listen=218.97.252.39 >> check_via=no # (cmd. line: -v) >> dns=no # (cmd. line: -r) >> rev_dns=no # (cmd. line: -R) >> #port=5060 >> #children=4 >> alias=localhost.localdomain >> alias=218.97.252.39 >> >> # ------------------ module loading ---------------------------------- >> loadmodule "/usr/local/lib/ser/modules/sl.so" >> loadmodule "/usr/local/lib/ser/modules/tm.so" >> # ----------------- setting module-specific parameters --------------- >> >> # -- tm params -- >> # set time for which ser will be waiting for a final response; >> # fr_inv_timer sets value for INVITE transactions, fr_timer >> # for all others >> modparam("tm", "fr_inv_timer", 120 ) >> modparam("tm", "fr_timer", 10 ) >> >> # ------------------------- request routing logic ------------------- >> >> # main routing logic >> >> >> route{ >> # for testing purposes, simply okay all REGISTERs >> if (method=="REGISTER") { >> log("REGISTER\n"); >> sl_send_reply("200", "ok"); >> break; >> }; >> seturi("sip:200@218.97.252.35:5060"); >> # if we do not get a positive reply, continue at reply_route[1] >> log("for test\n"); >> t_on_failure("1"); >> # forward the request to all destinations in destination set now >> t_relay(); >> } >> >> failure_route[1] { >> #when caller cancel, terminate the call >> if (t_check_status("487")) { >> log("cancel received\n"); >> t_reply("487", "Request Terminated"); >> break; >> }; >> # forwarding failed -- try again at another destination >> append_branch("sip:300@218.97.252.44:5060"); >> log(1,"first redirection\n"); >> # if this alternative destination fails too, proceed to >> reply_route[2] >> #t_on_failure("2"); >> t_relay(); >> } >> >> Here is my test: >> >> When ua1(sip:100@localhost.localdomain:5060) call >> <sip:400@localhost.localdomain:5060>, SER, at once, rewrites uri and sends >> invite to ua2(sip:200@218.97.252.35:5060). But because ua2 is offline, SER >> forwards the invite to ua3(sip:300@218.97.252.44:5060) and then ua3 begins >> ringing. Now, ua1 sends cancel to SER, SER sends cancel to ua3. ua3 >> returns "487" to SER and SER sends "408" to ua1. I don't know whether the >> timers were reset properly before forwarding. >> >> And with the attachment is the ser debug info. Please have a look. >> >> Thanks again. >> >>>I think you need to make an ngrep SIP trace of the two scenarios. SER will >>>not generate a 408 unless the timer hits. Increase the timer (it will be >>>easier to see what happens). You use a very low timer value >>>(fr_inv_timer=15). Set it to 120. Then look for the CANCEL from ua3. Does >>>SER receive the CANCEL and process it (use debug mode 7)? >>>I'm not sure if this is the same as mentioned in the bug referred to in >>>another reply, but let's try to find out. >>>Please post the ngrep trace for the two scenarios. >>>g-) >>> >>>----- Original Message ----- >>>From: "zhangshuai" <zhangshuai(a)goldentek.biz> >>>To: "serusers" <serusers(a)lists.iptel.org> >>>Cc: "wuyuan" <wuyuan(a)goldentek.biz> >>>Sent: Tuesday, February 14, 2006 9:41 AM >>>Subject: Re: Re: Re: [Serusers] a question about "cancel" in >>>forwardingscenario >>> >>> >>>> Dear Greger V. Teigre, >>>> >>>> >>>> I've tried t_on_failure("2"). >>>> >>>> failure_route[2] { >>>> #when caller cancel, terminate the call >>>> if (t_check_status("487")) { >>>> log(1,"cancel received\n"); >>>> t_reply("487", "Request Terminated"); >>>> break; >>>> } >>>> } >>>> >>>> There are similar lines in failure_route[1]. When ua2 is ringing and ua1 >>>> cancel the call, the value of "t_check_status("487")" in >>>> failure_route[1] >>>> will be true. However, When ua3 is ringing and ua1 cancel the call, the >>>> value of "t_check_status("487")" in failure_route[2] will be false. This >>>> time, t_check_status() returns "408". I don't know why SER generates >>>> "408" >>>> under this circumstance. My ser's version is 0.9.3 and I just run >>>> update_from_cvs. And I found similar description here: >>>> http://bugs.sip-router.org/browse/SER-63 >>>> Is it an unsolved problem? >>>> >>>> >>>>>Do some logging, what happens in the failure route? To me, it looks like >>>>>your config appends a branch. BTW, run update_from_cvs in your SER >>>>>source >>>>>dir top make sure you get the latest bug fixes! >>>>>g-) >>>>> >>>>>----- Original Message ----- >>>>>From: "zhangshuai" <zhangshuai(a)goldentek.biz> >>>>>To: "serusers" <serusers(a)lists.iptel.org> >>>>>Cc: "wuyuan" <wuyuan(a)goldentek.biz> >>>>>Sent: Tuesday, February 14, 2006 3:49 AM >>>>>Subject: Re: Re: [Serusers] a question about "cancel" in forwarding >>>>>scenario >>>>> >>>>> >>>>>> Dear Greger V. Teigre, >>>>>> >>>>>> >>>>>> >>>>>> Thanks for your quick reply. >>>>>> >>>>>> I have read the sergettingstarted05.pdf (Based upon SER version 0.9.3) >>>>>> and >>>>>> tested the onr.cfg in >>>>>> ser-0.9.3.GettingStarted.1.3.tar.gz/ser-0.9.3/examples many times. SER >>>>>> sent out "408" to caller immediately after received "486" from ua3. I >>>>>> don't know how does ser's failure route deal with this kind of >>>>>> "cancel" >>>>>> when the invite has been forwarded. Any ideas will help me a lot. >>>>>> Thanks! >>>>>> >>>>>> Here is the onr.cfg: >>>>>> >>>>>> # ------------------ module loading ---------------------------------- >>>>>> loadmodule "/usr/local/lib/ser/modules/sl.so" >>>>>> loadmodule "/usr/local/lib/ser/modules/tm.so" >>>>>> # ----------------- setting module-specific parameters --------------- >>>>>> >>>>>> # -- tm params -- >>>>>> # set time for which ser will be waiting for a final response; >>>>>> # fr_inv_timer sets value for INVITE transactions, fr_timer >>>>>> # for all others >>>>>> modparam("tm", "fr_inv_timer", 15 ) >>>>>> modparam("tm", "fr_timer", 10 ) >>>>>> >>>>>> # ------------------------- request routing logic ------------------- >>>>>> >>>>>> # main routing logic >>>>>> >>>>>> route{ >>>>>> # for testing purposes, simply okay all REGISTERs >>>>>> if (method=="REGISTER") { >>>>>> log("REGISTER"); >>>>>> sl_send_reply("200", "ok"); >>>>>> break; >>>>>> }; >>>>>> # try these two destinations first in parallel; the second >>>>>> # destination is targeted to sink port -- that will make ser >>>>>> # wait until timer hits >>>>>> seturi("sip:200@218.97.252.35:5060"); >>>>>> #append_branch("sip:300@218.97.252.36:5060"); >>>>>> # if we do not get a positive reply, continue at reply_route[1] >>>>>> t_on_failure("1"); >>>>>> # forward the request to all destinations in destination set >>>>>> now >>>>>> t_relay(); >>>>>> } >>>>>> >>>>>> failure_route[1] { >>>>>> #when caller cancel, terminate the call >>>>>> if (t_check_status("487")) { >>>>>> t_reply("487", "Request Terminated"); >>>>>> break; >>>>>> }; >>>>>> # forwarding failed -- try again at another destination >>>>>> append_branch("sip:300@218.97.252.44:5060"); >>>>>> log(1,"first redirection\n"); >>>>>> # if this alternative destination fails too, proceed to >>>>>> reply_route[2] >>>>>> #t_on_failure("2"); >>>>>> t_relay(); >>>>>> } >>>>>> >>>>>> >>>>>> Thanks in advance. >>>>>> >>>>>> Have a good day! >>>>>> >>>>>> >>>>>> >>>>>> Shuai >>>>>> >>>>>> >>>>>>>The internal SER timer hits and the request times out (the request is >>>>>>>not >>>>>>>terminated). >>>>>>>See SER-GettingStarted doc at ONsip.org for a full example of call >>>>>>>features >>>>>>>like forwarding. >>>>>>>g-) >>>>>>>----- Original Message ----- >>>>>>>From: "zhangshuai" <zhangshuai(a)goldentek.biz> >>>>>>>To: "serusers" <serusers(a)lists.iptel.org> >>>>>>>Cc: "wuyuan" <wuyuan(a)goldentek.biz> >>>>>>>Sent: Friday, February 10, 2006 11:18 AM >>>>>>>Subject: [Serusers] a question about "cancel" in forwarding scenario >>>>>>> >>>>>>> >>>>>>>> Dear All, >>>>>>>> >>>>>>>> >>>>>>>> I want to make use of avpops module to achieve forward on-no-reply. >>>>>>>> When >>>>>>>> ua1 calls ua2 and ua2 is off-line, or busy, or doesn't answer the >>>>>>>> phone, >>>>>>>> the invite should be forwarded to ua3 and ua3 begins ringing. >>>>>>>> >>>>>>>> In my tests, first, ua1 sends "cancel" when ua2 is ringing. Ua1 >>>>>>>> receives >>>>>>>> "487 Request Terminated" from SER. Second, ua1 sends "cancel" when >>>>>>>> ua3 >>>>>>>> is >>>>>>>> ringing. So, theoretically, ua1 should receive "487 Request >>>>>>>> Terminated" >>>>>>>> after "200 canceling" from SER, right? But, ua1 received "408 >>>>>>>> Request >>>>>>>> Timeout" instead. So strange. How come? >>>>>>>> >>>>>>>> I think something is wrong with my ser.cfg. Could anyone help me >>>>>>>> about >>>>>>>> this? Thanks in advance. >>>>>>>> >>>>>>>> Here is my ser.cfg: >>>>>>>> >>>>>>>> # ------------------ module >>>>>>>> loading ---------------------------------- >>>>>>>> >>>>>>>> loadmodule "/usr/local/lib/ser/modules/sl.so" >>>>>>>> loadmodule "/usr/local/lib/ser/modules/tm.so" >>>>>>>> loadmodule "/usr/local/lib/ser/modules/rr.so" >>>>>>>> loadmodule "/usr/local/lib/ser/modules/maxfwd.so" >>>>>>>> loadmodule "/usr/local/lib/ser/modules/usrloc.so" >>>>>>>> loadmodule "/usr/local/lib/ser/modules/registrar.so" >>>>>>>> loadmodule "/usr/local/lib/ser/modules/textops.so" >>>>>>>> loadmodule "/usr/local/lib/ser/modules/avp.so" >>>>>>>> loadmodule "/usr/local/lib/ser/modules/acc.so" >>>>>>>> loadmodule "/usr/local/lib/ser/modules/mysql.so" >>>>>>>> loadmodule "/usr/local/lib/ser/modules/dbtext.so" >>>>>>>> loadmodule "/usr/local/lib/ser/modules/avpops.so" >>>>>>>> #loadmodule "/usr/local/lib/ser/modules/postgres.so" >>>>>>>> #loadmodule "/usr/local/lib/ser/modules/flatstore.so" >>>>>>>> >>>>>>>> # Uncomment this if you want digest authentication >>>>>>>> # mysql.so must be loaded ! >>>>>>>> loadmodule "/usr/local/lib/ser/modules/auth.so" >>>>>>>> #loadmodule "/usr/local/lib/ser/modules/auth_db.so" >>>>>>>> >>>>>>>> loadmodule "/usr/local/lib/ser/modules/auth_radius.so" >>>>>>>> loadmodule "/usr/local/lib/ser/modules/group_radius.so" >>>>>>>> loadmodule "/usr/local/lib/ser/modules/uri_radius.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", 1) >>>>>>>> >>>>>>>> # -- auth params -- >>>>>>>> # Uncomment if you are using auth module >>>>>>>> # >>>>>>>> #modparam("auth_db", "calculate_ha1", yes) >>>>>>>> # >>>>>>>> # If you set "calculate_ha1" parameter to yes (which true in this >>>>>>>> config), >>>>>>>> # uncomment also the following parameter) >>>>>>>> # >>>>>>>> #modparam("auth_db", "password_column", "password") >>>>>>>> >>>>>>>> # -- rr params -- >>>>>>>> # add value to ;lr param to make some broken UAs happy >>>>>>>> >>>>>>>> modparam("rr", "enable_full_lr", 1) >>>>>>>> modparam("tm", "fr_timer", 10 ) >>>>>>>> modparam("tm", "fr_inv_timer", 20) >>>>>>>> modparam("tm", "noisy_ctimer", 1) >>>>>>>> >>>>>>>> modparam("auth_radius|uri_radius|group_radius", "radius_config", >>>>>>>> "/usr/local/etc/radiusclient-ng/radiusclient.conf") >>>>>>>> >>>>>>>> modparam("avpops","avp_url","mysql://ser:heslo@localhost/ser") >>>>>>>> modparam("avpops","avp_table","usr_preferences") >>>>>>>> # ------------------------- request routing >>>>>>>> logic ------------------- >>>>>>>> >>>>>>>> # main routing logic >>>>>>>> >>>>>>>> route{ >>>>>>>> # initial sanity checks -- messages with >>>>>>>> # max_forwards==0, or excessively long requests >>>>>>>> if (!mf_process_maxfwd_header("10")) { >>>>>>>> sl_send_reply("483","Too Many Hops"); >>>>>>>> break; >>>>>>>> }; >>>>>>>> if (msg:len >= 2048 ) { >>>>>>>> sl_send_reply("513", "Message too big"); >>>>>>>> break; >>>>>>>> }; >>>>>>>> >>>>>>>> if (!method=="REGISTER") record_route(); >>>>>>>> >>>>>>>> if (loose_route()) { >>>>>>>> t_relay(); >>>>>>>> break; >>>>>>>> }; >>>>>>>> >>>>>>>> if (method=="INVITE") { >>>>>>>> lookup("location"); >>>>>>>> #if (!lookup("location")) { >>>>>>>> # sl_send_reply("404", "Not Found"); >>>>>>>> # break; >>>>>>>> #}; >>>>>>>> } >>>>>>>> >>>>>>>> if (method=="REGISTER") { >>>>>>>> log("REGISTER\n"); >>>>>>>> if (!radius_www_authorize("localhost.localdomain")) { >>>>>>>> www_challenge("localhost.localdomain", "0"); >>>>>>>> break; >>>>>>>> }; >>>>>>>> save("location"); >>>>>>>> break; >>>>>>>> }; >>>>>>>> >>>>>>>> if (method=="INVITE") { >>>>>>>> if (!radius_proxy_authorize("localhost.localdomain")) >>>>>>>> { >>>>>>>> proxy_challenge("localhost.localdomain", >>>>>>>> "1"); >>>>>>>> break; >>>>>>>> }; >>>>>>>> } >>>>>>>> >>>>>>>> # if we do not get a positive reply, continue at >>>>>>>> failure_route[1] >>>>>>>> t_on_failure("1"); >>>>>>>> # forward the request to all destinations in destination set now >>>>>>>> t_relay(); >>>>>>>> >>>>>>>> #if (!t_relay()) { >>>>>>>> # sl_reply_error(); >>>>>>>> #}; >>>>>>>> } >>>>>>>> >>>>>>>> failure_route[1] { >>>>>>>> >>>>>>>> #when caller cancel, terminate the call >>>>>>>> if (t_check_status("487")) { >>>>>>>> t_reply("487", "Request Terminated"); >>>>>>>> break; >>>>>>>> }; >>>>>>>> >>>>>>>> if ( ( avp_db_load("$ruri","s:redirect_on_failure") && >>>>>>>> avp_check("redirect_on_failure","eq/i:1"))) { >>>>>>>> # User need to forward >>>>>>>> log(1, "User wants redirection.\n"); >>>>>>>> if ( ( avp_db_load("$ruri","s:redirectnumber") && >>>>>>>> !avp_check("redirectnumber","re/^$"))) { >>>>>>>> log(1,"first redirect\n"); >>>>>>>> #avp_print(); >>>>>>>> attr2uri("redirect number"); >>>>>>>> lookup("location"); >>>>>>>> append_branch(); >>>>>>>> #t_on_failure("2"); >>>>>>>> t_relay(); >>>>>>>> } >>>>>>>> else { >>>>>>>> t_reply("408", "TimeOut"); >>>>>>>> break; >>>>>>>> }; >>>>>>>> } >>>>>>>> else { >>>>>>>> t_reply("404", "Not Found"); >>>>>>>> break; >>>>>>>> }; >>>>>>>> } >>>>>>>> >>>>>>>> >>>>>>>> _______________________________________________ >>>>>>>> Serusers mailing list >>>>>>>> serusers(a)lists.iptel.org >>>>>>>> http://lists.iptel.org/mailman/listinfo/serusers >>>>>>>> >>>>>>> >>>>>>> >>>>>> >>>>>> = = = = = = = = = = = = = = = = = = = = >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Serusers mailing list >>>>>> serusers(a)lists.iptel.org >>>>>> http://lists.iptel.org/mailman/listinfo/serusers >>>>>> >>>>> >>>>> >>>> >>>> = = = = = = = = = = = = = = = = = = = = >>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Serusers mailing list >>>> serusers(a)lists.iptel.org >>>> http://lists.iptel.org/mailman/listinfo/serusers >>>> >>> >>> >> >> = = = = = = = = = = = = = = = = = = = = >> >> >> > > >-------------------------------------------------------------------------------- > > >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> > > = = = = = = = = = = = = = = = = = = = = [View Less]

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Re: [Users] next_gw(): No ruri_user AVP
by Bogdan-Andrei Iancu 16 Feb '06

16 Feb '06

Ray, I think there is a problem in the script - if you see the gw_uri avps, it's quite impossible not to see them also before and after the next_gw() call in request route. Disregarding the ruri_avp, the gw_uri avps should be visible. are you sure you do not delete all the avps in your script. Try placing avp_print before and after each lcr function call. regards, bogdan Raymond Chen wrote: >Hi bogdan, > >I put avp_print after next_gw , no avp output. > > >1(1334) … [View More]load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.138:5060> > 1(1334) load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.139:5060> > 1(1334) does_uri_exit(): User in request uri does not exist > 1(1334) is_user_in(): User is in group 'local' > 1(1334) db_flags=3, flags=12 > 1(1334) DEBUG:avpops:load_avps: loaded avps = 1 > 1(1334) parse_headers: flags=ffffffffffffffff > 1(1334) DEBUG:avpops:pushto_avps: 1 avps were processed >1(1334) next_gw(): No ruri_user AVP > > > > > > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Wednesday, February 15, 2006 4:39 AM >To: Raymond Chen >Cc: daniel(a)voice-system.ro; Users(a)openser.org >Subject: Re: [Users] next_gw(): No ruri_user AVP > >Ray, > >the ruri_avp is added by next_gw after its usage from the REQUEST route. >You may check this by placing an avp_print after you did next_gw() in >request route (after calling route 3, for example). Check if there is >any avp with ID 1402. > >regards, >bogdan > >Raymond Chen wrote: > > > >>Hi bogdan, >> >>We called Load_gw and next-gw from request route. We have no problem with >>next_gw if the if (avp_pushto("$ruri", "s:fwdnoanswer")) happens in the >> >> >main > > >>route. But when it does in the failure_route the next_gw can't find the >>ruri_user avp. >> >>Raymond >> >> >> >>Route { >> >> .......... >> >> Route(3); >> >> ........... >> >>} >> >>failure_route[1] { >> >> if (t_check_status("(480)|(408)")) { >> if (avp_pushto("$ruri", "s:fwdnoanswer")) { >> avp_delete("s:fwdnoanswer"); >> route(3); >> }; >> }; >> >>} >> >>Route[3] { >> >> if (!load_gws()) { >> sl_send_reply("500", "Server Internal Error - Cannot load >>gateways"); >> return; >> }; >> >> ............... >> >> Route(5); >> >>} >> >>Route[5] { >> >> if (!next_gw()) { >> rewriteuri("sip:userbusy@211.102.91.134:443"); >> t_relay(); >> return; >> }; >> >> .............. >> >>} >> >> >> >>-----Original Message----- >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >>Sent: Wednesday, February 15, 2006 2:25 AM >>To: Raymond Chen >>Cc: daniel(a)voice-system.ro; Users(a)openser.org >>Subject: Re: [Users] next_gw(): No ruri_user AVP >> >>Hi Ray, >> >>do you call load_gws() from failure route? if so, not that this is not >>supported. >>For LCR to work properly, you need to call load_gws() and next_gw() from >>the request route and later next_gw() from failure routes. >> >>regards, >>bogdan >> >>Raymond Chen wrote: >> >> >> >> >> >>>Hi Bogdan, >>> >>>Here is the debug >>> >>>1(2584) load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.138:5060> >>>1(2584) load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.139:5060> >>>1(2584) DEBUG:avpops:print_avp: p=0xf4f167c8, flags=2 >>>1(2584) DEBUG: id=<1400> >>>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.139:5060> >>>1(2584) DEBUG:avpops:print_avp: p=0xf4f16790, flags=2 >>>1(2584) DEBUG: id=<1400> >>>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.138:5060> >>>1(2584) does_uri_exit(): User in request uri does not exist >>>1(2584) is_user_in(): User is in group 'local' >>>1(2584) db_flags=3, flags=12 >>>1(2584) DEBUG:avpops:print_avp: p=0xf4f167f8, flags=B >>>1(2584) DEBUG: id=<1400> >>>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.139:5060> >>>1(2584) DEBUG:avpops:print_avp: p=0xf4f16790, flags=2 >>>1(2584) DEBUG: id=<1400> >>>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.xxx:5060> >>>1(2584) next_gw(): No ruri_user AVP >>> >>> >>> >>>Raymond >>> >>> >>>-----Original Message----- >>>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >>>Sent: Tuesday, February 14, 2006 10:55 AM >>>To: Raymond Chen >>>Cc: daniel(a)voice-system.ro; Users(a)openser.org >>>Subject: Re: [Users] next_gw(): No ruri_user AVP >>> >>>Hi Ray, >>> >>>use avp_print() after the load_gw() to see what avps were loaded and >>>again just before next_gw() to see the available avp. >>>this will help to see if it's a problem at the load or search part. >>> >>>regards, >>>bogdan >>> >>> [View Less]

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Re: [Users] next_gw(): No ruri_user AVP
by Bogdan-Andrei Iancu 16 Feb '06

16 Feb '06

Ray, the ruri_avp is added by next_gw after its usage from the REQUEST route. You may check this by placing an avp_print after you did next_gw() in request route (after calling route 3, for example). Check if there is any avp with ID 1402. regards, bogdan Raymond Chen wrote: >Hi bogdan, > >We called Load_gw and next-gw from request route. We have no problem with >next_gw if the if (avp_pushto("$ruri", "s:fwdnoanswer")) happens in the main >route. But when it does in the … [View More]failure_route the next_gw can't find the >ruri_user avp. > >Raymond > > > >Route { > > .......... > > Route(3); > > ........... > >} > >failure_route[1] { > > if (t_check_status("(480)|(408)")) { > if (avp_pushto("$ruri", "s:fwdnoanswer")) { > avp_delete("s:fwdnoanswer"); > route(3); > }; > }; > >} > >Route[3] { > > if (!load_gws()) { > sl_send_reply("500", "Server Internal Error - Cannot load >gateways"); > return; > }; > > ............... > > Route(5); > >} > >Route[5] { > > if (!next_gw()) { > rewriteuri("sip:userbusy@211.102.91.134:443"); > t_relay(); > return; > }; > > .............. > >} > > > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Wednesday, February 15, 2006 2:25 AM >To: Raymond Chen >Cc: daniel(a)voice-system.ro; Users(a)openser.org >Subject: Re: [Users] next_gw(): No ruri_user AVP > >Hi Ray, > >do you call load_gws() from failure route? if so, not that this is not >supported. >For LCR to work properly, you need to call load_gws() and next_gw() from >the request route and later next_gw() from failure routes. > >regards, >bogdan > >Raymond Chen wrote: > > > >>Hi Bogdan, >> >>Here is the debug >> >>1(2584) load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.138:5060> >>1(2584) load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.139:5060> >>1(2584) DEBUG:avpops:print_avp: p=0xf4f167c8, flags=2 >>1(2584) DEBUG: id=<1400> >>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.139:5060> >>1(2584) DEBUG:avpops:print_avp: p=0xf4f16790, flags=2 >>1(2584) DEBUG: id=<1400> >>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.138:5060> >>1(2584) does_uri_exit(): User in request uri does not exist >>1(2584) is_user_in(): User is in group 'local' >>1(2584) db_flags=3, flags=12 >>1(2584) DEBUG:avpops:print_avp: p=0xf4f167f8, flags=B >>1(2584) DEBUG: id=<1400> >>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.139:5060> >>1(2584) DEBUG:avpops:print_avp: p=0xf4f16790, flags=2 >>1(2584) DEBUG: id=<1400> >>1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.xxx:5060> >>1(2584) next_gw(): No ruri_user AVP >> >> >> >>Raymond >> >> >>-----Original Message----- >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >>Sent: Tuesday, February 14, 2006 10:55 AM >>To: Raymond Chen >>Cc: daniel(a)voice-system.ro; Users(a)openser.org >>Subject: Re: [Users] next_gw(): No ruri_user AVP >> >>Hi Ray, >> >>use avp_print() after the load_gw() to see what avps were loaded and >>again just before next_gw() to see the available avp. >>this will help to see if it's a problem at the load or search part. >> >>regards, >>bogdan >> >>Raymond Chen wrote: >> >> >> >> >> >>>Hi Daniel, >>> >>>The problem we have is not about next_gw forward to >>>userbusy(a)xxx.xxx.xxx.xxx. We have no problem with load_gw, it loads two >>>gateways against ruri_user correctly. but when next_gw called, it saids >>> >>>next_gw(): No ruri_user AVP >>> >>>looks like ruri_user avp is not saved when load_gw was called. So next_gw >>>can't find the ruri_user avp. >>> >>>Thanks >>> >>>Ray >>> >>> >>>-----Original Message----- >>>From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] >>>Sent: Tuesday, February 14, 2006 1:44 AM >>>To: Raymond Chen >>>Cc: Users(a)openser.org >>>Subject: Re: [Users] next_gw(): No ruri_user AVP >>> >>>Hello, >>> >>>On 02/14/06 13:43, Raymond Chen wrote: >>> >>> >>> >>> >>> >>> >>>>[...] >>>> if (!next_gw()) { >>>> rewriteuri("sip:userbusy@211.102.91.134:443"); >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>here you must call append_branch() so the rewriteuri() has the desired >>>effect. >>> >>>Cheers, >>>Daniel >>> >>> >>> >>> >>> >>> >>> >>>> t_relay(); >>>> return; >>>> }; >>>> >>>> ........... >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>> >>>_______________________________________________ >>>Users mailing list >>>Users(a)openser.org >>>http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> >>> >>> >>> >>> >> >> >> >> >> >> > > > > > > [View Less]

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[Serusers] error running SER: segmentation fault
by ryan nalupa 16 Feb '06

16 Feb '06

hi! i've installed SER using apt-get in debian and having some errors when running it while when i installed it from source using centos it accepts my cfg file but i've seen that it's easy to install it on debian systems only that i'm having this kind of problem, can anyone direct me to some link wherein i can read about having this error "segmentation fault" at the end when running SER? thanks in advance! :-) ryanalupa __________________________________________________ Do You Yahoo!? … [View More]

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[Serusers] features-callfwd.5.0.cfg to Asterisk VM
by Ali Zaidi 16 Feb '06

16 Feb '06

Hi, I'm having some problem forwarding to Asterisk VM. What i want is when ever 486, 404, 408 or 480 error triger then ser should forward INVITE to Asterisk. Some one please tell me where and what i write in my features-callfwd.5.0.cfg so it will forward to my asterisk box. Thanks, Ali... :lol:

1 0

Re: [Users] next_gw(): No ruri_user AVP
by Bogdan-Andrei Iancu 16 Feb '06

16 Feb '06

Hi Ray, do you call load_gws() from failure route? if so, not that this is not supported. For LCR to work properly, you need to call load_gws() and next_gw() from the request route and later next_gw() from failure routes. regards, bogdan Raymond Chen wrote: >Hi Bogdan, > >Here is the debug > >1(2584) load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.138:5060> > 1(2584) load_gws(): DEBUG: Added gw_uri_avp <sip:@xxx.xxx.xxx.139:5060> > 1(2584) DEBUG:… [View More]avpops:print_avp: p=0xf4f167c8, flags=2 > 1(2584) DEBUG: id=<1400> > 1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.139:5060> > 1(2584) DEBUG:avpops:print_avp: p=0xf4f16790, flags=2 > 1(2584) DEBUG: id=<1400> > 1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.138:5060> > 1(2584) does_uri_exit(): User in request uri does not exist > 1(2584) is_user_in(): User is in group 'local' > 1(2584) db_flags=3, flags=12 >1(2584) DEBUG:avpops:print_avp: p=0xf4f167f8, flags=B >1(2584) DEBUG: id=<1400> > 1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.139:5060> > 1(2584) DEBUG:avpops:print_avp: p=0xf4f16790, flags=2 > 1(2584) DEBUG: id=<1400> > 1(2584) DEBUG: val_str=<sip:@xxx.xxx.xxx.xxx:5060> > 1(2584) next_gw(): No ruri_user AVP > > > >Raymond > > >-----Original Message----- >From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] >Sent: Tuesday, February 14, 2006 10:55 AM >To: Raymond Chen >Cc: daniel(a)voice-system.ro; Users(a)openser.org >Subject: Re: [Users] next_gw(): No ruri_user AVP > >Hi Ray, > >use avp_print() after the load_gw() to see what avps were loaded and >again just before next_gw() to see the available avp. >this will help to see if it's a problem at the load or search part. > >regards, >bogdan > >Raymond Chen wrote: > > > >>Hi Daniel, >> >>The problem we have is not about next_gw forward to >>userbusy(a)xxx.xxx.xxx.xxx. We have no problem with load_gw, it loads two >>gateways against ruri_user correctly. but when next_gw called, it saids >> >>next_gw(): No ruri_user AVP >> >>looks like ruri_user avp is not saved when load_gw was called. So next_gw >>can't find the ruri_user avp. >> >>Thanks >> >>Ray >> >> >>-----Original Message----- >>From: Daniel-Constantin Mierla [mailto:daniel@voice-system.ro] >>Sent: Tuesday, February 14, 2006 1:44 AM >>To: Raymond Chen >>Cc: Users(a)openser.org >>Subject: Re: [Users] next_gw(): No ruri_user AVP >> >>Hello, >> >>On 02/14/06 13:43, Raymond Chen wrote: >> >> >> >> >>>[...] >>> if (!next_gw()) { >>> rewriteuri("sip:userbusy@211.102.91.134:443"); >>> >>> >>> >>> >>> >>here you must call append_branch() so the rewriteuri() has the desired >>effect. >> >>Cheers, >>Daniel >> >> >> >> >> >>> t_relay(); >>> return; >>> }; >>> >>> ........... >>> >>> >>> >>> >>> >>> >>> >> >> >>_______________________________________________ >>Users mailing list >>Users(a)openser.org >>http://openser.org/cgi-bin/mailman/listinfo/users >> >> >> >> >> > > > > > > [View Less]

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[Serusers] UAC UDP to Proxy TCP... Problem
by Christian Thomas 15 Feb '06

15 Feb '06

Hi everybody, I have a UDP2TCP problem with Ser. (0.9-4) The UACs connected to my SER Proxy are all UDP. When I use some carriers to connect them to PSTN The Carrier Proxy / GW could talk UDP or TCP. It depends on the destination. Each time the Carrier GW want to talk TCP, the contact field is correct and contains the 'transport=tcp' mention. all seems to be correct and, after a complete reading about this subject in the list, I trust that using t_relay() must do the right translation and send … [View More]

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Re: Re: Re: Re: [Serusers] a question about "cancel" in forwardingscenario
by zhangshuai 15 Feb '06

15 Feb '06

Dear Greger V. Teigre, Thanks for your suggestion. I've reset the fr_inv_timer and debug level, and put some log functions into ser.cfg: # ----------- global configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) fork=no log_stderror=yes # (cmd line: -E) listen=218.97.252.39 check_via=no # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) #port=5060 #children=4 alias=localhost.localdomain alias=… [View More]218.97.252.39 # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" # ----------------- setting module-specific parameters --------------- # -- tm params -- # set time for which ser will be waiting for a final response; # fr_inv_timer sets value for INVITE transactions, fr_timer # for all others modparam("tm", "fr_inv_timer", 120 ) modparam("tm", "fr_timer", 10 ) # ------------------------- request routing logic ------------------- # main routing logic route{ # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { log("REGISTER\n"); sl_send_reply("200", "ok"); break; }; seturi("sip:200@218.97.252.35:5060"); # if we do not get a positive reply, continue at reply_route[1] log("for test\n"); t_on_failure("1"); # forward the request to all destinations in destination set now t_relay(); } failure_route[1] { #when caller cancel, terminate the call if (t_check_status("487")) { log("cancel received\n"); t_reply("487", "Request Terminated"); break; }; # forwarding failed -- try again at another destination append_branch("sip:300@218.97.252.44:5060"); log(1,"first redirection\n"); # if this alternative destination fails too, proceed to reply_route[2] #t_on_failure("2"); t_relay(); } Here is my test: When ua1(sip:100@localhost.localdomain:5060) call <sip:400@localhost.localdomain:5060>, SER, at once, rewrites uri and sends invite to ua2(sip:200@218.97.252.35:5060). But because ua2 is offline, SER forwards the invite to ua3(sip:300@218.97.252.44:5060) and then ua3 begins ringing. Now, ua1 sends cancel to SER, SER sends cancel to ua3. ua3 returns "487" to SER and SER sends "408" to ua1. I don't know whether the timers were reset properly before forwarding. And with the attachment is the ser debug info. Please have a look. Thanks again. >I think you need to make an ngrep SIP trace of the two scenarios. SER will >not generate a 408 unless the timer hits. Increase the timer (it will be >easier to see what happens). You use a very low timer value >(fr_inv_timer=15). Set it to 120. Then look for the CANCEL from ua3. Does >SER receive the CANCEL and process it (use debug mode 7)? >I'm not sure if this is the same as mentioned in the bug referred to in >another reply, but let's try to find out. >Please post the ngrep trace for the two scenarios. >g-) > >----- Original Message ----- >From: "zhangshuai" <zhangshuai(a)goldentek.biz> >To: "serusers" <serusers(a)lists.iptel.org> >Cc: "wuyuan" <wuyuan(a)goldentek.biz> >Sent: Tuesday, February 14, 2006 9:41 AM >Subject: Re: Re: Re: [Serusers] a question about "cancel" in >forwardingscenario > > >> Dear Greger V. Teigre, >> >> >> I've tried t_on_failure("2"). >> >> failure_route[2] { >> #when caller cancel, terminate the call >> if (t_check_status("487")) { >> log(1,"cancel received\n"); >> t_reply("487", "Request Terminated"); >> break; >> } >> } >> >> There are similar lines in failure_route[1]. When ua2 is ringing and ua1 >> cancel the call, the value of "t_check_status("487")" in failure_route[1] >> will be true. However, When ua3 is ringing and ua1 cancel the call, the >> value of "t_check_status("487")" in failure_route[2] will be false. This >> time, t_check_status() returns "408". I don't know why SER generates "408" >> under this circumstance. My ser's version is 0.9.3 and I just run >> update_from_cvs. And I found similar description here: >> http://bugs.sip-router.org/browse/SER-63 >> Is it an unsolved problem? >> >> >>>Do some logging, what happens in the failure route? To me, it looks like >>>your config appends a branch. BTW, run update_from_cvs in your SER source >>>dir top make sure you get the latest bug fixes! >>>g-) >>> >>>----- Original Message ----- >>>From: "zhangshuai" <zhangshuai(a)goldentek.biz> >>>To: "serusers" <serusers(a)lists.iptel.org> >>>Cc: "wuyuan" <wuyuan(a)goldentek.biz> >>>Sent: Tuesday, February 14, 2006 3:49 AM >>>Subject: Re: Re: [Serusers] a question about "cancel" in forwarding >>>scenario >>> >>> >>>> Dear Greger V. Teigre, >>>> >>>> >>>> >>>> Thanks for your quick reply. >>>> >>>> I have read the sergettingstarted05.pdf (Based upon SER version 0.9.3) >>>> and >>>> tested the onr.cfg in >>>> ser-0.9.3.GettingStarted.1.3.tar.gz/ser-0.9.3/examples many times. SER >>>> sent out "408" to caller immediately after received "486" from ua3. I >>>> don't know how does ser's failure route deal with this kind of "cancel" >>>> when the invite has been forwarded. Any ideas will help me a lot. >>>> Thanks! >>>> >>>> Here is the onr.cfg: >>>> >>>> # ------------------ module loading ---------------------------------- >>>> loadmodule "/usr/local/lib/ser/modules/sl.so" >>>> loadmodule "/usr/local/lib/ser/modules/tm.so" >>>> # ----------------- setting module-specific parameters --------------- >>>> >>>> # -- tm params -- >>>> # set time for which ser will be waiting for a final response; >>>> # fr_inv_timer sets value for INVITE transactions, fr_timer >>>> # for all others >>>> modparam("tm", "fr_inv_timer", 15 ) >>>> modparam("tm", "fr_timer", 10 ) >>>> >>>> # ------------------------- request routing logic ------------------- >>>> >>>> # main routing logic >>>> >>>> route{ >>>> # for testing purposes, simply okay all REGISTERs >>>> if (method=="REGISTER") { >>>> log("REGISTER"); >>>> sl_send_reply("200", "ok"); >>>> break; >>>> }; >>>> # try these two destinations first in parallel; the second >>>> # destination is targeted to sink port -- that will make ser >>>> # wait until timer hits >>>> seturi("sip:200@218.97.252.35:5060"); >>>> #append_branch("sip:300@218.97.252.36:5060"); >>>> # if we do not get a positive reply, continue at reply_route[1] >>>> t_on_failure("1"); >>>> # forward the request to all destinations in destination set now >>>> t_relay(); >>>> } >>>> >>>> failure_route[1] { >>>> #when caller cancel, terminate the call >>>> if (t_check_status("487")) { >>>> t_reply("487", "Request Terminated"); >>>> break; >>>> }; >>>> # forwarding failed -- try again at another destination >>>> append_branch("sip:300@218.97.252.44:5060"); >>>> log(1,"first redirection\n"); >>>> # if this alternative destination fails too, proceed to >>>> reply_route[2] >>>> #t_on_failure("2"); >>>> t_relay(); >>>> } >>>> >>>> >>>> Thanks in advance. >>>> >>>> Have a good day! >>>> >>>> >>>> >>>> Shuai >>>> >>>> >>>>>The internal SER timer hits and the request times out (the request is >>>>>not >>>>>terminated). >>>>>See SER-GettingStarted doc at ONsip.org for a full example of call >>>>>features >>>>>like forwarding. >>>>>g-) >>>>>----- Original Message ----- >>>>>From: "zhangshuai" <zhangshuai(a)goldentek.biz> >>>>>To: "serusers" <serusers(a)lists.iptel.org> >>>>>Cc: "wuyuan" <wuyuan(a)goldentek.biz> >>>>>Sent: Friday, February 10, 2006 11:18 AM >>>>>Subject: [Serusers] a question about "cancel" in forwarding scenario >>>>> >>>>> >>>>>> Dear All, >>>>>> >>>>>> >>>>>> I want to make use of avpops module to achieve forward on-no-reply. >>>>>> When >>>>>> ua1 calls ua2 and ua2 is off-line, or busy, or doesn't answer the >>>>>> phone, >>>>>> the invite should be forwarded to ua3 and ua3 begins ringing. >>>>>> >>>>>> In my tests, first, ua1 sends "cancel" when ua2 is ringing. Ua1 >>>>>> receives >>>>>> "487 Request Terminated" from SER. Second, ua1 sends "cancel" when ua3 >>>>>> is >>>>>> ringing. So, theoretically, ua1 should receive "487 Request >>>>>> Terminated" >>>>>> after "200 canceling" from SER, right? But, ua1 received "408 Request >>>>>> Timeout" instead. So strange. How come? >>>>>> >>>>>> I think something is wrong with my ser.cfg. Could anyone help me about >>>>>> this? Thanks in advance. >>>>>> >>>>>> Here is my ser.cfg: >>>>>> >>>>>> # ------------------ module loading ---------------------------------- >>>>>> >>>>>> loadmodule "/usr/local/lib/ser/modules/sl.so" >>>>>> loadmodule "/usr/local/lib/ser/modules/tm.so" >>>>>> loadmodule "/usr/local/lib/ser/modules/rr.so" >>>>>> loadmodule "/usr/local/lib/ser/modules/maxfwd.so" >>>>>> loadmodule "/usr/local/lib/ser/modules/usrloc.so" >>>>>> loadmodule "/usr/local/lib/ser/modules/registrar.so" >>>>>> loadmodule "/usr/local/lib/ser/modules/textops.so" >>>>>> loadmodule "/usr/local/lib/ser/modules/avp.so" >>>>>> loadmodule "/usr/local/lib/ser/modules/acc.so" >>>>>> loadmodule "/usr/local/lib/ser/modules/mysql.so" >>>>>> loadmodule "/usr/local/lib/ser/modules/dbtext.so" >>>>>> loadmodule "/usr/local/lib/ser/modules/avpops.so" >>>>>> #loadmodule "/usr/local/lib/ser/modules/postgres.so" >>>>>> #loadmodule "/usr/local/lib/ser/modules/flatstore.so" >>>>>> >>>>>> # Uncomment this if you want digest authentication >>>>>> # mysql.so must be loaded ! >>>>>> loadmodule "/usr/local/lib/ser/modules/auth.so" >>>>>> #loadmodule "/usr/local/lib/ser/modules/auth_db.so" >>>>>> >>>>>> loadmodule "/usr/local/lib/ser/modules/auth_radius.so" >>>>>> loadmodule "/usr/local/lib/ser/modules/group_radius.so" >>>>>> loadmodule "/usr/local/lib/ser/modules/uri_radius.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", 1) >>>>>> >>>>>> # -- auth params -- >>>>>> # Uncomment if you are using auth module >>>>>> # >>>>>> #modparam("auth_db", "calculate_ha1", yes) >>>>>> # >>>>>> # If you set "calculate_ha1" parameter to yes (which true in this >>>>>> config), >>>>>> # uncomment also the following parameter) >>>>>> # >>>>>> #modparam("auth_db", "password_column", "password") >>>>>> >>>>>> # -- rr params -- >>>>>> # add value to ;lr param to make some broken UAs happy >>>>>> >>>>>> modparam("rr", "enable_full_lr", 1) >>>>>> modparam("tm", "fr_timer", 10 ) >>>>>> modparam("tm", "fr_inv_timer", 20) >>>>>> modparam("tm", "noisy_ctimer", 1) >>>>>> >>>>>> modparam("auth_radius|uri_radius|group_radius", "radius_config", >>>>>> "/usr/local/etc/radiusclient-ng/radiusclient.conf") >>>>>> >>>>>> modparam("avpops","avp_url","mysql://ser:heslo@localhost/ser") >>>>>> modparam("avpops","avp_table","usr_preferences") >>>>>> # ------------------------- request routing logic ------------------- >>>>>> >>>>>> # main routing logic >>>>>> >>>>>> route{ >>>>>> # initial sanity checks -- messages with >>>>>> # max_forwards==0, or excessively long requests >>>>>> if (!mf_process_maxfwd_header("10")) { >>>>>> sl_send_reply("483","Too Many Hops"); >>>>>> break; >>>>>> }; >>>>>> if (msg:len >= 2048 ) { >>>>>> sl_send_reply("513", "Message too big"); >>>>>> break; >>>>>> }; >>>>>> >>>>>> if (!method=="REGISTER") record_route(); >>>>>> >>>>>> if (loose_route()) { >>>>>> t_relay(); >>>>>> break; >>>>>> }; >>>>>> >>>>>> if (method=="INVITE") { >>>>>> lookup("location"); >>>>>> #if (!lookup("location")) { >>>>>> # sl_send_reply("404", "Not Found"); >>>>>> # break; >>>>>> #}; >>>>>> } >>>>>> >>>>>> if (method=="REGISTER") { >>>>>> log("REGISTER\n"); >>>>>> if (!radius_www_authorize("localhost.localdomain")) { >>>>>> www_challenge("localhost.localdomain", "0"); >>>>>> break; >>>>>> }; >>>>>> save("location"); >>>>>> break; >>>>>> }; >>>>>> >>>>>> if (method=="INVITE") { >>>>>> if (!radius_proxy_authorize("localhost.localdomain")) { >>>>>> proxy_challenge("localhost.localdomain", "1"); >>>>>> break; >>>>>> }; >>>>>> } >>>>>> >>>>>> # if we do not get a positive reply, continue at failure_route[1] >>>>>> t_on_failure("1"); >>>>>> # forward the request to all destinations in destination set now >>>>>> t_relay(); >>>>>> >>>>>> #if (!t_relay()) { >>>>>> # sl_reply_error(); >>>>>> #}; >>>>>> } >>>>>> >>>>>> failure_route[1] { >>>>>> >>>>>> #when caller cancel, terminate the call >>>>>> if (t_check_status("487")) { >>>>>> t_reply("487", "Request Terminated"); >>>>>> break; >>>>>> }; >>>>>> >>>>>> if ( ( avp_db_load("$ruri","s:redirect_on_failure") && >>>>>> avp_check("redirect_on_failure","eq/i:1"))) { >>>>>> # User need to forward >>>>>> log(1, "User wants redirection.\n"); >>>>>> if ( ( avp_db_load("$ruri","s:redirectnumber") && >>>>>> !avp_check("redirectnumber","re/^$"))) { >>>>>> log(1,"first redirect\n"); >>>>>> #avp_print(); >>>>>> attr2uri("redirect number"); >>>>>> lookup("location"); >>>>>> append_branch(); >>>>>> #t_on_failure("2"); >>>>>> t_relay(); >>>>>> } >>>>>> else { >>>>>> t_reply("408", "TimeOut"); >>>>>> break; >>>>>> }; >>>>>> } >>>>>> else { >>>>>> t_reply("404", "Not Found"); >>>>>> break; >>>>>> }; >>>>>> } >>>>>> >>>>>> >>>>>> _______________________________________________ >>>>>> Serusers mailing list >>>>>> serusers(a)lists.iptel.org >>>>>> http://lists.iptel.org/mailman/listinfo/serusers >>>>>> >>>>> >>>>> >>>> >>>> = = = = = = = = = = = = = = = = = = = = >>>> >>>> >>>> >>>> >>>> >>>> _______________________________________________ >>>> Serusers mailing list >>>> serusers(a)lists.iptel.org >>>> http://lists.iptel.org/mailman/listinfo/serusers >>>> >>> >>> >> >> = = = = = = = = = = = = = = = = = = = = >> >> >> >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> > > = = = = = = = = = = = = = = = = = = = = [View Less]

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[Serusers] SER authentication through FreeRadius with an LDAP backend
by Jon Steer 15 Feb '06

15 Feb '06

Requirement Authenticating SIP Clients through a SIP proxy via RADIUS with an LDAP Backend using digest mode. The path is SIP client X -> Ser SIP Proxy -> FreeRadius -> Fedora Directory Server Issue Cannot authenticate a SIP client using Freeradius digest mode and LDAP. After reading through a number of the newslists of all of the projects being used, There is contradictory information about whether this is possible. One solution described in the FreeRadius mailing list … [View More]

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[Users] Requests resent by proxy?
by Walter Schober 15 Feb '06

15 Feb '06

Hello! Using OpenSER 1.0.0 in my path I encounter it rather often, that (exactly the same) requests (no append_branch()) are resent by the proxy (!) after 2 to 40 ms. Any idea what triggers that? /walter

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[Serusers] Error 481 and CANCEL
by Benito Caracuel 15 Feb '06

15 Feb '06

Hello all, I am trying to cancel an INVITE but I can't. UAC ----------------- SER ----------------- UAS INVITE ------> CANCEL -----> <---------481 When UAS receive the CANCEL, it sends a "481 Call/Transaction does not exist". UAS must send 200 ok and 487 instead of 481. How can I solve this problem? Help me please. Thank you.

1 0

[Users] install TLS package
by Tóth Péter 15 Feb '06

15 Feb '06

Helo! I'm very new on the list, and I have a basic problem. After installing openser deb packages (1.0.0 tls version), i got the following error at the start: #/etc/init.d/openser start Starting openser: openserbad user name/uid number: -u openser # When I installed the 1.0.0 simple package, a user started up automatically with name "openser", but at the tls install there is no user entry in /etc/passwd file. I meant this is the error, but I don't know exactly. What should I do? … [View More]

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[Users] fifo get_statistics
by Helge Waastad 15 Feb '06

15 Feb '06

Hi, not really a critical observation this time, but it seems I can only run openserctl fifo get_statistics one time. The next time it's hanging...after a ctrl-c, I can run it successfully again. (running get_statistics on a module, ex openserctl fifo get_statistics tm:, works fine) br hw -- Helge Waastad Senior Konsulent Systemavdelingen Smartnet

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[Serusers] avpops prefix adding for areacode
by Ibrahim TUNALI 15 Feb '06

15 Feb '06

Hi, If user call 7 digits number, I want to add caller prefix from avpops record to TO uri. I know prefix() function uses for adding prefix but it doesn't work with avpops module. I couldn't use avp_write() function because it's only work with avp_values or avp_alias. I want to mix of avp_alias and TO uri. ie sip:2304862@domain.com -> sip:9252304862@domain.com like traditional PSTN dialing (you don't dial your areacode if you want to call your area) My configuration part for avpops … [View More]

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[Users] nathelper natping OPTIONS packets formated to not get reply?
by Glenn Dalgliesh 15 Feb '06

15 Feb '06

I was looking at packet traces of the OPTIONS packets generated by natping and it appears that at least in my implementation of OpenSer 1.0.0 the "To: sip" line has no username which causes many UA's require in order to respond to the OPTIONS packet. I was wondering if this was intentional or if it would be possible to change this behavior or at least make it an configurable option. I think a lot could be done/determined based on the results of the reply; including determining if the packet is … [View More]

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[Users] Caller ID
by Martin Kjeldsen 15 Feb '06

15 Feb '06

Hi I'm having a problem with users whose Caller-ID I would like to hide. I'm aware that the FROM header has to carry the value anonymous(a)anonymous.invalid, for a user to be shown as anonymous in the callees display. But I'm not sure how to accomplish this without violating core rules, concerning the FROM headerss tag! My accounting is done using the FROM header aswell, but I can imagine that I could just place the flag prior to altering the FROM header, if that is the way to go about … [View More]

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Re: [Users] next_gw(): No ruri_user AVP
by Bogdan-Andrei Iancu 15 Feb '06

15 Feb '06

Hi Ray, use avp_print() after the load_gw() to see what avps were loaded and again just before next_gw() to see the available avp. this will help to see if it's a problem at the load or search part. regards, bogdan Raymond Chen wrote: >Hi Daniel, > >The problem we have is not about next_gw forward to >userbusy(a)xxx.xxx.xxx.xxx. We have no problem with load_gw, it loads two >gateways against ruri_user correctly. but when next_gw called, it saids > >next_gw(): No … [View More]

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Re: [Users] multiple 487 created
by unplug 15 Feb '06

15 Feb '06

Thanks! But I don't really understand why there is a mismatch between 487and ACK. Below is a part of sip message pair of 487 and ACK. Below is the complete log from ngrep. (From line 290, openser creates 487 and line 304 UA1 ACK. After that, multiple 487 are created by openser) http://meerkat.no-ip.com/openser/cancel_normal.log Below is my openser.cfg for reference http://meerkat.no-ip.com/openser/openser.cfg 290 U 203.193.26.226:5060 -> 210.184.23.31:5060 291 SIP/2.0 487 Request … [View More]Cancelled. 292 Via: SIP/2.0/UDP 10.0.0.46:5060;rport=5060;received=210.184.23.31;branch=z9hG4bKdUyOnG9Zel82UECI. 293 From: "sip:36418488@o01.ol.com"<sip:871966806561@o01.ol.com>;tag=xLTdFs7yGkeXq3wn. 294 To: "34163634" <sip:34163634@o01.ol.com>;tag=A81AFED4-21D7. 295 Date: Tue, 15 Feb 2005 01:52:57 GMT. 296 Call-ID: Z0wcK4WwdTU6mH00(a)10.0.0.46. 297 Server: Cisco-SIPGateway/IOS-12.x. 298 CSeq: 1 INVITE. 299 Allow-Events: telephone-event. 300 Content-Length: 0. 301 . 302 303 # 304 U 210.184.23.31:5060 -> 203.193.26.226:5060 305 ACK sip:34163634@o01.ol.com SIP/2.0. 306 Via: SIP/2.0/UDP 10.0.0.46:5060;branch=z9hG4bKdUyOnG9Zel82UECI. 307 Route: <sip:34163634@203.193.26.226:5060;nat=yes;ftag=xLTdFs7yGkeXq3wn;lr=on>. 308 Max-Forwards: 70. 309 User-Agent: Koncept KE10XX v4.32.10 00-09-45-0a-fc-2d. 310 From: "871966806561" <sip:871966806561@o01.ol.com>;tag=xLTdFs7yGkeXq3wn. 311 To: "34163634" <sip:34163634@o01.ol.com>;tag=A81AFED4-21D7. 312 Call-ID: Z0wcK4WwdTU6mH00(a)10.0.0.46. 313 Contact: <sip:871966806561@10.0.0.46:5060>. 314 CSeq: 1 ACK. 315 Content-Length: 0. In my openser.cfg line 184-187, it is remarked. Do you mean it should un-remarked for handling ACK according to your point 2? On 2/15/06, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: > This means that openser does not receive the ACK or is not able to match > the ACK to the canceled transaction. Thus it is re-transmitting the 487. > > Increas the debug level and take a look at the syslog to find out if: > 1. the ACK to the 487 is received by openser > 2. if it is received, check if there is a problem during handling of the > ACK. (for these kind of ACKs just t_relay()). > > klaus > > On Wed, February 15, 2006 5:13, unplug said: > > For the normal cancel situation. > > > > UA1 OPENSER UA2 > > --------INVITE-----------------> > > <-----------100----------------- > > --------------------INVITE----------> > > <---------------100------------------- > > ----------CANCEL----------> > > --------------CANCEL-------------> > > <--------------200------------------ > > <------------487------------------- > > --------------ACK-----------------> > > <---------487--------------- > > -------------ACK------------> > > > > But in my case, I found that there are many 487 created from openser > > to UA1. The multiple 487 may cause a call drop in the following call. > > Why does openser create so many 487 for UA1? How can I prevent it? > > > > UA1 OPENSER UA2 > > --------INVITE-----------------> > > <-----------100----------------- > > --------------------INVITE----------> > > <---------------100------------------- > > ----------CANCEL----------> > > --------------CANCEL-------------> > > <--------------200------------------ > > <------------487------------------- > > --------------ACK-----------------> > > <---------487--------------- > > -------------ACK------------> > > <---------487------------- > > -------------ACK------------> > > <---------487--------------- > > -------------ACK------------> > > <---------487------------- > > -------------ACK------------> > > <---------487------------- > > -------------ACK------------> > > <---------487------------- > > -------------ACK------------> > > <---------487------------- > > -------------ACK------------> > > ... > > > > _______________________________________________ > > Users mailing list > > Users(a)openser.org > > http://openser.org/cgi-bin/mailman/listinfo/users > > > > > [View Less]

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[Serusers] Bye and error 481
by Benito Caracuel 15 Feb '06

15 Feb '06

Hi All, Sometimes when I send a BYE, the other user sends "481not dialog found". What is the problem? How can I solve this? Thank you very much.

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Re: [Users] refer message handling?
by Bogdan-Andrei Iancu 15 Feb '06

15 Feb '06

Hi , that's not correct - you shouldn't route sequential request based on user location, but based on Route headers. Roughly : if (is_method("REFER")) { loose_route(); t_relay(); exit; } regards, bogdan Frogger wrote: >Thanks for the quick response, > >So basically its as simple as: > >if (method=="REFER") { > > #routing code here or > > if (lookup("location")) { > > # or routing code here > > } > >} > >Does that … [View More]

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Re: [Users] multiple 487 created
by Klaus Darilion 15 Feb '06

15 Feb '06

This means that openser does not receive the ACK or is not able to match the ACK to the canceled transaction. Thus it is re-transmitting the 487. Increas the debug level and take a look at the syslog to find out if: 1. the ACK to the 487 is received by openser 2. if it is received, check if there is a problem during handling of the ACK. (for these kind of ACKs just t_relay()). klaus On Wed, February 15, 2006 5:13, unplug said: > For the normal cancel situation. > > UA1 OPENSER UA2 … [View More]

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[Users] [Fwd: Re: [Devel] Psuedo variables / AVPOPS enhancement requests]
by Klaus Darilion 15 Feb '06

15 Feb '06

You also have to make sure that the Content-Length: header will be adopted when changing the size of the SDP. regards klaus On Tue, February 14, 2006 21:08, S G said: > Hi, > > Can psuedo variables and avpops be enhanced to include the headers in the SDP message body? > > v=0. > o=2139869454 25762964 25762964 IN IP4 192.168.1.133. > s=SDP Session For C&S MoIP. > c=IN IP4 64.xx.xxx.xxx > t=0 0. > m=audio 40000 RTP/AVP 4 0 97. > a=rtpmap:4 G723/8000. > a=… [View More]

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RE: [Serusers] Problem with SER Accounting
by mkumar＠mantragroup.com 15 Feb '06

15 Feb '06

Thanks for the reply. I added setflag(1) statement also but even then "acc" table of ser is empty. I have mysql 3.23 version, is that creating the problem? I am attaching my ser.cfg again, please check it and help me know where I am going wrong? Quoting Ramin NIkaeen <Rnikaeen(a)goldline.net>: > I think you should have a "setflag(1)" for the accounting to work! > > ramin > > > -----Original Message----- > From: mkumar(a)mantragroup.com [mailto:mkumar@mantragroup.… [View More]

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Re: Re: Re: [Serusers] a question about "cancel" in forwarding scenario
by zhangshuai 15 Feb '06

15 Feb '06

Dear Greger V. Teigre, I've tried t_on_failure("2"). failure_route[2] { #when caller cancel, terminate the call if (t_check_status("487")) { log(1,"cancel received\n"); t_reply("487", "Request Terminated"); break; } } There are similar lines in failure_route[1]. When ua2 is ringing and ua1 cancel the call, the value of "t_check_status("487")" in failure_route[1] will be true. However, When ua3 is ringing and ua1 cancel the call, the value of "… [View More]t_check_status("487")" in failure_route[2] will be false. This time, t_check_status() returns "408". I don't know why SER generates "408" under this circumstance. My ser's version is 0.9.3 and I just run update_from_cvs. And I found similar description here: http://bugs.sip-router.org/browse/SER-63 Is it an unsolved problem? >Do some logging, what happens in the failure route? To me, it looks like >your config appends a branch. BTW, run update_from_cvs in your SER source >dir top make sure you get the latest bug fixes! >g-) > >----- Original Message ----- >From: "zhangshuai" <zhangshuai(a)goldentek.biz> >To: "serusers" <serusers(a)lists.iptel.org> >Cc: "wuyuan" <wuyuan(a)goldentek.biz> >Sent: Tuesday, February 14, 2006 3:49 AM >Subject: Re: Re: [Serusers] a question about "cancel" in forwarding scenario > > >> Dear Greger V. Teigre, >> >> >> >> Thanks for your quick reply. >> >> I have read the sergettingstarted05.pdf (Based upon SER version 0.9.3) and >> tested the onr.cfg in >> ser-0.9.3.GettingStarted.1.3.tar.gz/ser-0.9.3/examples many times. SER >> sent out "408" to caller immediately after received "486" from ua3. I >> don't know how does ser's failure route deal with this kind of "cancel" >> when the invite has been forwarded. Any ideas will help me a lot. Thanks! >> >> Here is the onr.cfg: >> >> # ------------------ module loading ---------------------------------- >> loadmodule "/usr/local/lib/ser/modules/sl.so" >> loadmodule "/usr/local/lib/ser/modules/tm.so" >> # ----------------- setting module-specific parameters --------------- >> >> # -- tm params -- >> # set time for which ser will be waiting for a final response; >> # fr_inv_timer sets value for INVITE transactions, fr_timer >> # for all others >> modparam("tm", "fr_inv_timer", 15 ) >> modparam("tm", "fr_timer", 10 ) >> >> # ------------------------- request routing logic ------------------- >> >> # main routing logic >> >> route{ >> # for testing purposes, simply okay all REGISTERs >> if (method=="REGISTER") { >> log("REGISTER"); >> sl_send_reply("200", "ok"); >> break; >> }; >> # try these two destinations first in parallel; the second >> # destination is targeted to sink port -- that will make ser >> # wait until timer hits >> seturi("sip:200@218.97.252.35:5060"); >> #append_branch("sip:300@218.97.252.36:5060"); >> # if we do not get a positive reply, continue at reply_route[1] >> t_on_failure("1"); >> # forward the request to all destinations in destination set now >> t_relay(); >> } >> >> failure_route[1] { >> #when caller cancel, terminate the call >> if (t_check_status("487")) { >> t_reply("487", "Request Terminated"); >> break; >> }; >> # forwarding failed -- try again at another destination >> append_branch("sip:300@218.97.252.44:5060"); >> log(1,"first redirection\n"); >> # if this alternative destination fails too, proceed to >> reply_route[2] >> #t_on_failure("2"); >> t_relay(); >> } >> >> >> Thanks in advance. >> >> Have a good day! >> >> >> >> Shuai >> >> >>>The internal SER timer hits and the request times out (the request is not >>>terminated). >>>See SER-GettingStarted doc at ONsip.org for a full example of call >>>features >>>like forwarding. >>>g-) >>>----- Original Message ----- >>>From: "zhangshuai" <zhangshuai(a)goldentek.biz> >>>To: "serusers" <serusers(a)lists.iptel.org> >>>Cc: "wuyuan" <wuyuan(a)goldentek.biz> >>>Sent: Friday, February 10, 2006 11:18 AM >>>Subject: [Serusers] a question about "cancel" in forwarding scenario >>> >>> >>>> Dear All, >>>> >>>> >>>> I want to make use of avpops module to achieve forward on-no-reply. When >>>> ua1 calls ua2 and ua2 is off-line, or busy, or doesn't answer the phone, >>>> the invite should be forwarded to ua3 and ua3 begins ringing. >>>> >>>> In my tests, first, ua1 sends "cancel" when ua2 is ringing. Ua1 receives >>>> "487 Request Terminated" from SER. Second, ua1 sends "cancel" when ua3 >>>> is >>>> ringing. So, theoretically, ua1 should receive "487 Request Terminated" >>>> after "200 canceling" from SER, right? But, ua1 received "408 Request >>>> Timeout" instead. So strange. How come? >>>> >>>> I think something is wrong with my ser.cfg. Could anyone help me about >>>> this? Thanks in advance. >>>> >>>> Here is my ser.cfg: >>>> >>>> # ------------------ module loading ---------------------------------- >>>> >>>> loadmodule "/usr/local/lib/ser/modules/sl.so" >>>> loadmodule "/usr/local/lib/ser/modules/tm.so" >>>> loadmodule "/usr/local/lib/ser/modules/rr.so" >>>> loadmodule "/usr/local/lib/ser/modules/maxfwd.so" >>>> loadmodule "/usr/local/lib/ser/modules/usrloc.so" >>>> loadmodule "/usr/local/lib/ser/modules/registrar.so" >>>> loadmodule "/usr/local/lib/ser/modules/textops.so" >>>> loadmodule "/usr/local/lib/ser/modules/avp.so" >>>> loadmodule "/usr/local/lib/ser/modules/acc.so" >>>> loadmodule "/usr/local/lib/ser/modules/mysql.so" >>>> loadmodule "/usr/local/lib/ser/modules/dbtext.so" >>>> loadmodule "/usr/local/lib/ser/modules/avpops.so" >>>> #loadmodule "/usr/local/lib/ser/modules/postgres.so" >>>> #loadmodule "/usr/local/lib/ser/modules/flatstore.so" >>>> >>>> # Uncomment this if you want digest authentication >>>> # mysql.so must be loaded ! >>>> loadmodule "/usr/local/lib/ser/modules/auth.so" >>>> #loadmodule "/usr/local/lib/ser/modules/auth_db.so" >>>> >>>> loadmodule "/usr/local/lib/ser/modules/auth_radius.so" >>>> loadmodule "/usr/local/lib/ser/modules/group_radius.so" >>>> loadmodule "/usr/local/lib/ser/modules/uri_radius.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", 1) >>>> >>>> # -- auth params -- >>>> # Uncomment if you are using auth module >>>> # >>>> #modparam("auth_db", "calculate_ha1", yes) >>>> # >>>> # If you set "calculate_ha1" parameter to yes (which true in this >>>> config), >>>> # uncomment also the following parameter) >>>> # >>>> #modparam("auth_db", "password_column", "password") >>>> >>>> # -- rr params -- >>>> # add value to ;lr param to make some broken UAs happy >>>> >>>> modparam("rr", "enable_full_lr", 1) >>>> modparam("tm", "fr_timer", 10 ) >>>> modparam("tm", "fr_inv_timer", 20) >>>> modparam("tm", "noisy_ctimer", 1) >>>> >>>> modparam("auth_radius|uri_radius|group_radius", "radius_config", >>>> "/usr/local/etc/radiusclient-ng/radiusclient.conf") >>>> >>>> modparam("avpops","avp_url","mysql://ser:heslo@localhost/ser") >>>> modparam("avpops","avp_table","usr_preferences") >>>> # ------------------------- request routing logic ------------------- >>>> >>>> # main routing logic >>>> >>>> route{ >>>> # initial sanity checks -- messages with >>>> # max_forwards==0, or excessively long requests >>>> if (!mf_process_maxfwd_header("10")) { >>>> sl_send_reply("483","Too Many Hops"); >>>> break; >>>> }; >>>> if (msg:len >= 2048 ) { >>>> sl_send_reply("513", "Message too big"); >>>> break; >>>> }; >>>> >>>> if (!method=="REGISTER") record_route(); >>>> >>>> if (loose_route()) { >>>> t_relay(); >>>> break; >>>> }; >>>> >>>> if (method=="INVITE") { >>>> lookup("location"); >>>> #if (!lookup("location")) { >>>> # sl_send_reply("404", "Not Found"); >>>> # break; >>>> #}; >>>> } >>>> >>>> if (method=="REGISTER") { >>>> log("REGISTER\n"); >>>> if (!radius_www_authorize("localhost.localdomain")) { >>>> www_challenge("localhost.localdomain", "0"); >>>> break; >>>> }; >>>> save("location"); >>>> break; >>>> }; >>>> >>>> if (method=="INVITE") { >>>> if (!radius_proxy_authorize("localhost.localdomain")) { >>>> proxy_challenge("localhost.localdomain", "1"); >>>> break; >>>> }; >>>> } >>>> >>>> # if we do not get a positive reply, continue at failure_route[1] >>>> t_on_failure("1"); >>>> # forward the request to all destinations in destination set now >>>> t_relay(); >>>> >>>> #if (!t_relay()) { >>>> # sl_reply_error(); >>>> #}; >>>> } >>>> >>>> failure_route[1] { >>>> >>>> #when caller cancel, terminate the call >>>> if (t_check_status("487")) { >>>> t_reply("487", "Request Terminated"); >>>> break; >>>> }; >>>> >>>> if ( ( avp_db_load("$ruri","s:redirect_on_failure") && >>>> avp_check("redirect_on_failure","eq/i:1"))) { >>>> # User need to forward >>>> log(1, "User wants redirection.\n"); >>>> if ( ( avp_db_load("$ruri","s:redirectnumber") && >>>> !avp_check("redirectnumber","re/^$"))) { >>>> log(1,"first redirect\n"); >>>> #avp_print(); >>>> attr2uri("redirect number"); >>>> lookup("location"); >>>> append_branch(); >>>> #t_on_failure("2"); >>>> t_relay(); >>>> } >>>> else { >>>> t_reply("408", "TimeOut"); >>>> break; >>>> }; >>>> } >>>> else { >>>> t_reply("404", "Not Found"); >>>> break; >>>> }; >>>> } >>>> >>>> >>>> _______________________________________________ >>>> Serusers mailing list >>>> serusers(a)lists.iptel.org >>>> http://lists.iptel.org/mailman/listinfo/serusers >>>> >>> >>> >> >> = = = = = = = = = = = = = = = = = = = = >> >> >> >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> > > = = = = = = = = = = = = = = = = = = = = [View Less]

3 3

[Serusers] aliases
by Olivier.taylor 15 Feb '06

15 Feb '06

hi all is there a way to have the aliases table updated without using serctl or restart ser? Thanks, Olivier

2 1

Re: [Users] next_gw(): No ruri_user AVP
by Daniel-Constantin Mierla 15 Feb '06

15 Feb '06

Hello, On 02/14/06 13:43, Raymond Chen wrote: > [...] > if (!next_gw()) { > rewriteuri("sip:userbusy@211.102.91.134:443"); > here you must call append_branch() so the rewriteuri() has the desired effect. Cheers, Daniel > t_relay(); > return; > }; > > ........... > > >

2 1

[Users] multiple 487 created
by unplug 15 Feb '06

15 Feb '06

For the normal cancel situation. UA1 OPENSER UA2 --------INVITE-----------------> <-----------100----------------- --------------------INVITE----------> <---------------100------------------- ----------CANCEL----------> --------------CANCEL-------------> <--------------200------------------ <------------487------------------- --------------ACK-----------------> <---------487--------------- -------------ACK------------> But in … [View More]

1 0

[Users] call drop in a few seconds
by unplug 15 Feb '06

15 Feb '06

When UA1 calls UA2, UA1 drop the call before UA2 take the call. After a few second, UA1 makes another call to UA3. The call established and it will drop after a few second. I expect the situation below. UA1 ------INVITE ----> UA2 UA2 (ringing) UA1 ------CANCEL----> UA2 UA2 (ringing) UA1 -----INVITE--------> UA3 (in few seconds) UA1 ------call established----UA3 UA1 <-------BYE------- UA2 UA1 -----call drop--------UA3 What can I do to reduce the situation above?

2 6

[Users] add/rewrite a tag/parameter in SDP
by Lenir 14 Feb '06

14 Feb '06

Hey guys, What is the best way to modify the SDP message to always include (if not replace) the a= tags in a SDP message? My problem is specifically with g729, according to the RFC, if the parameter: "a=fmtp:18 annexb=yes" is present or missing then it is assumed that g729b (g729 with VAD) and if "a=fmtp:18 annexb=no" is present then the devices will use g729a (g729 with VAD Disabled). My problem is that some cisco gateways and ATAs behave differently (of course Cisco always doing it … [View More]

1 0

[Users] NEW FUNCTIONALITY: add_rr_param() in BRANCH / FAILURE routes
by Bogdan-Andrei Iancu 14 Feb '06

14 Feb '06

Hi, the add_rr_param() can be new safely called from BRANCH / FAILURE routes. The behaviour will be: - in BRANCH route - the RR parameters are set only for the current branch, so you can have different RR params for each branch - in FAILURE route - the RR parameters will be set only for the outgoing branch, so during serial forking you may add additional RR param function of the new destination. Additional improvements in the RR module usage: -record_route() may take as … [View More]

1 0

[Serusers] Ser and Fax over IP and Real-Time Fax configurations
by Elias Kaaria 14 Feb '06

14 Feb '06

Dear Tobias, Thanks you for your assistance on As5350 configurations and sip. Now am about to configure Fax over IP and Real-Time Fax in my ser sip server. I will appreciate your input on the same, Thanking you in advance. Yours sincerely, Elias Gitonga kaaria On 1/12/06, Tobias Lindgren <tobias.lindgren(a)ip-only.se> wrote: > > Hello, > > Just an example. > > dial-peer voice 1234567 voip > huntstop > application session > destination-pattern 01234567. > … [View More]

2 2

Re: RE : [Serusers] SER amd Minute Provider
by mkumar＠mantragroup.com 14 Feb '06

14 Feb '06

Olivier.taylor,I did not exactly get how this will solve my requirement? I just need two lines for my ser.cfg after invite, I am diverting call to my minute provider but here I do not know how to give my account details supplied by my minute provider here if (uri=~"^sip:6[0-8][0-9]*@mantragroup.com"){ strip(1); log(1, "Trying to forward to Teliax.\n"); rewriteuri("username(a)VOIP-CO2.TELIAX.COM"); rewriteuserpass("username:password"); rewritehost("VOIP-CO2.TELIAX.COM"); … [View More] forward("username:password@VOIP-CO2.TELIAX.COM"); } Can someone please tell me how can I give uri for whom call has to be sent and username,password supplied by my minute provider. I have been trying from two days mailing this list and searching net but sadly I could not find the solution. Thanks for your time and help, Manoj. Quoting "Olivier.taylor" <Olivier.taylor(a)gmail.com>: > Have a try with us if you want... > U can register your ser as a single user, every calls will be billed to > the same account. > > I really don't use to use the mailing list for my marketing but... > > I can open an test account if needed. > > Kind regards, > > Olivier > > -----Message d'origine----- > De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De > la part de mkumar(a)mantragroup.com > Envoyé : vendredi 10 février 2006 14:52 > À : Waldo Rubinstein > Cc : serusers(a)lists.iptel.org > Objet : Re: [Serusers] SER amd Minute Provider > > > Waldo Rubinstein, even I am shocked with their resonse. > > Can someone please tell me how can I do this or point me to related > links. > > Quoting Waldo Rubinstein <waldo(a)trianet.net>: > >> I am simply in shock to see/hear/read that Teliax is not >> able/capable/ willing to help you. >> >> - Waldo >> >> On Feb 10, 2006, at 2:16 AM, mkumar(a)mantragroup.com wrote: >> >>> Hi All, >>> >>> I added these lines >>> >>> if (uri=~"^sip:5[0-8][0-9]*@mantragroup.com"){ >>> if (!is_user_in("credentials", "ld")) { >>> sl_send_reply("403", "No permission"); >>> break; >>> }; >>> strip(1); >>> log(1, "Trying to forward to Teliax.\n"); > rewriteuri >>> ("sip:username@VOIP-CO2.TELIAX.COM"); > rewriteuserpass >>> ("username:password"); >>> rewritehost("VOIP-CO2.TELIAX.COM"); >>> } >>> >>> I get a respnse error returned not found from ser and If comment >>> these two lines >>> #rewriteuri("sip:username@VOIP-CO2.TELIAX.COM"); > >>> #rewriteuserpass("username:password"); >>> >>> keeping third line it says error number. >>> >>> Please help me and tell me how can I do this correctly? >>> >>> Thanks, >>> Manoj. >>> >>> >>> Quoting "Greger V. Teigre" <greger(a)teigre.com>: >>> >>>> Teliax is not a DID (Direct Inward Dialing) provider? But just a >>>> regular provider of VoIP service? >>>> So all calls will have the same caller-id? >>>> If that is right, you need to use the uac module for >>>> authenticating. It think it still has an issue with a CSeq not >>>> being incremented correctly, so if their GW is strict, it will not >>>> work. I'm not sure if this has been fixed in openser, but I >>>> thought I saw something on it. >>>> Good luck, >>>> g-) >>>> >>>> ----- Original Message ----- From: <mkumar(a)mantragroup.com> >>>> To: "Greger V. Teigre" <greger(a)teigre.com> >>>> Cc: <serusers(a)lists.iptel.org> >>>> Sent: Friday, February 10, 2006 7:11 AM >>>> Subject: Re: [Serusers] SER amd Minute Provider >>>> >>>> >>>>> I contacted Teliax support they have no idea on how to do this so >>>>> I started >>>>> searching myself and I need GSM codec support and I found only >>>>> Teliax for that >>>>> so I have to stick with them even if they do not help me. >>>>> >>>>> Coming to my problem, >>>>> >>>>> I am asking how to send both uri to whom call must be delivered >>>>> and username and >>>>> password provided by minute provider. Using rewrite I can change >>>>> uri for whom >>>>> call must be sent and forward will forward calls to particular > port and >>>>> destination then where can I send username and password provided >>>>> by minute >>>>> provider so that he can validate and deliver my call to that uri. >>>>> >>>>> Thanks, >>>>> Manoj. >>>>> >>>>> Quoting "Greger V. Teigre" <greger(a)teigre.com>: >>>>> >>>>>> That's highly non-standard if teliax requires you to rewrite the >>>>>> ruri with your credentials. If that is correct, they probably >>>>>> have some equally non-standar way to pick up the callee (maybe >>>>>> From?) and you should ask them. >>>>>> g-) >>>>>> ----- Original Message ----- From: <mkumar(a)mantragroup.com> >>>>>> To: "Voipers Portugal" <voipers(a)gmail.com> >>>>>> Cc: <serusers(a)lists.iptel.org> >>>>>> Sent: Friday, February 10, 2006 6:42 AM >>>>>> Subject: Re: [Serusers] SER amd Minute Provider >>>>>> >>>>>> >>>>>>> Hi, >>>>>>> >>>>>>> If I rewriteuri with my account credentials provided by minute >>>>>>> provider and >>>>>>> forward it to him then where I am passing the actual PSTN or >>>>>>> URI for whom the >>>>>>> call has to be sent. I have following lines in ser.cfg >>>>>>> >>>>>>> if (uri=~"^sip:4[0-8][0-9]*@mantragroup.com"){ >>>>>>> if (!is_user_in("credentials", "ld")) { sl_send_reply("403", "No >>>>>>> permission"); break; >>>>>>> }; >>>>>>> strip(1); >>>>>>> log(1, "Trying to forward to Teliax.\n"); >>>>>>> rewriteuri("sip:username:password@VOIP-CO2.TELIAX.COM"); >>>>>>> forward("VOIP-CO2.TELIAX.COM", 5060); >>>>>>> } >>>>>>> >>>>>>> Please tell me what must I do here, I am not finding any related > >>>>>>> info. >>>>>>> >>>>>>> Thanks, >>>>>>> Manoj. >>>>>>> >>>>>>> Quoting Voipers Portugal <voipers(a)gmail.com>: >>>>>>> >>>>>>>> Normally you just have to forward the invite message to the >>>>>>>> provider (after >>>>>>>> rewrite the URI of course). >>>>>>>> >>>>>>>> Jose Simoes >>>>>>>> >>>>>>>> >>>>>>>> On 2/9/06, mkumar(a)mantragroup.com <mkumar(a)mantragroup.com> >>>>>>>> wrote: >>>>>>>>> >>>>>>>>> Hi All, >>>>>>>>> >>>>>>>>> I purchased some minutes from a minute provider and looking >>>>>>>>> to configure >>>>>>>>> that >>>>>>>>> with SER. But I did not find any documentation or info on how >>>>>>>>> to do that? >>>>>>>>> Please tell me how can I do this or lead me to related >>>>>>>>> documentaion or >>>>>>>>> links. >>>>>>>>> >>>>>>>>> Thanks, >>>>>>>>> Manoj. >>>>>>>>> >>>>>>>>> _______________________________________________ >>>>>>>>> Serusers mailing list >>>>>>>>> serusers(a)lists.iptel.org >>>>>>>>> http://lists.iptel.org/mailman/listinfo/serusers >>>>>>>>> >>>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Serusers mailing list >>>>>>> serusers(a)lists.iptel.org >>>>>>> http://lists.iptel.org/mailman/listinfo/serusers >>>>>>> >>>>>> >>>>>> >>>>> >>>>> >>>>> >>>>> >>>> >>>> >>> >>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers >> >> > > > > _______________________________________________ > Serusers mailing list > serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers > > [View Less]

3 3

[Users] next_gw(): No ruri_user AVP
by Raymond Chen 14 Feb '06

14 Feb '06

Dear all, We have setup a noanswer call forwarding, below is our configuration much like the example from onsip.org if (t_check_status("(480)|(408)")) { if (avp_pushto("$ruri", "s:fwdnoanswer")) { avp_delete("s:fwdnoanswer"); route(6); break; }; }; if (!load_gws()) { sl_send_reply("500", "Server Internal Error - Cannot load gateways"); return; }; ........... if (!next_gw()) { … [View More]

1 0

Re: Re: Re: Re: [Serusers] a question about "cancel" in forwarding scenario
by zhangshuai 14 Feb '06

14 Feb '06

Dear Antonio Rabena, You mean parallel forking. Not forwarding, right? I receive the correct signal, "486", in parallel forking scenario. Only after the invite was forwarded, I experience the "cancel" problem, which is bothering so many days. Have you tested serial forking or forwarding? >Hi, > >Im also experiencing the same problem. The only diff. is ua2 and >ua3 are using same sip account so when ua1 sends invite to that sip >account, both ua2 and ua3 ring, and when … [View More]

1 0

[Users] refer or hold/transfer
by Frogger 14 Feb '06

14 Feb '06

I have a successful implementation of openser 0.9.5 and its been working very nicely. I am faced with transferring of calls from pstn to userC by way of userB and I am struggling. path: PSTN=>GW=>SBC=>openser=>uB uB transfers call to uC Call is failing to transfer as "refer" message is not properly handled by my openser script. Any guidance would be appreciated. Transfer works when all legs are from registered clients. Only when the initial call leg is from the PSTN does it … [View More]

2 2

Re: Re: Re: [Serusers] a question about "cancel" in forwarding scenario
by zhangshuai 14 Feb '06

14 Feb '06

Dear Greger V. Teigre, With the attachment is the signal trace. >Do some logging, what happens in the failure route? To me, it looks like >your config appends a branch. BTW, run update_from_cvs in your SER source >dir top make sure you get the latest bug fixes! >g-) > >----- Original Message ----- >From: "zhangshuai" <zhangshuai(a)goldentek.biz> >To: "serusers" <serusers(a)lists.iptel.org> >Cc: "wuyuan" <wuyuan(a)goldentek.biz> >Sent: … [View More]Tuesday, February 14, 2006 3:49 AM >Subject: Re: Re: [Serusers] a question about "cancel" in forwarding scenario > > >> Dear Greger V. Teigre, >> >> >> >> Thanks for your quick reply. >> >> I have read the sergettingstarted05.pdf (Based upon SER version 0.9.3) and >> tested the onr.cfg in >> ser-0.9.3.GettingStarted.1.3.tar.gz/ser-0.9.3/examples many times. SER >> sent out "408" to caller immediately after received "486" from ua3. I >> don't know how does ser's failure route deal with this kind of "cancel" >> when the invite has been forwarded. Any ideas will help me a lot. Thanks! >> >> Here is the onr.cfg: >> >> # ------------------ module loading ---------------------------------- >> loadmodule "/usr/local/lib/ser/modules/sl.so" >> loadmodule "/usr/local/lib/ser/modules/tm.so" >> # ----------------- setting module-specific parameters --------------- >> >> # -- tm params -- >> # set time for which ser will be waiting for a final response; >> # fr_inv_timer sets value for INVITE transactions, fr_timer >> # for all others >> modparam("tm", "fr_inv_timer", 15 ) >> modparam("tm", "fr_timer", 10 ) >> >> # ------------------------- request routing logic ------------------- >> >> # main routing logic >> >> route{ >> # for testing purposes, simply okay all REGISTERs >> if (method=="REGISTER") { >> log("REGISTER"); >> sl_send_reply("200", "ok"); >> break; >> }; >> # try these two destinations first in parallel; the second >> # destination is targeted to sink port -- that will make ser >> # wait until timer hits >> seturi("sip:200@218.97.252.35:5060"); >> #append_branch("sip:300@218.97.252.36:5060"); >> # if we do not get a positive reply, continue at reply_route[1] >> t_on_failure("1"); >> # forward the request to all destinations in destination set now >> t_relay(); >> } >> >> failure_route[1] { >> #when caller cancel, terminate the call >> if (t_check_status("487")) { >> t_reply("487", "Request Terminated"); >> break; >> }; >> # forwarding failed -- try again at another destination >> append_branch("sip:300@218.97.252.44:5060"); >> log(1,"first redirection\n"); >> # if this alternative destination fails too, proceed to >> reply_route[2] >> #t_on_failure("2"); >> t_relay(); >> } >> >> >> Thanks in advance. >> >> Have a good day! >> >> >> >> Shuai >> >> >>>The internal SER timer hits and the request times out (the request is not >>>terminated). >>>See SER-GettingStarted doc at ONsip.org for a full example of call >>>features >>>like forwarding. >>>g-) >>>----- Original Message ----- >>>From: "zhangshuai" <zhangshuai(a)goldentek.biz> >>>To: "serusers" <serusers(a)lists.iptel.org> >>>Cc: "wuyuan" <wuyuan(a)goldentek.biz> >>>Sent: Friday, February 10, 2006 11:18 AM >>>Subject: [Serusers] a question about "cancel" in forwarding scenario >>> >>> >>>> Dear All, >>>> >>>> >>>> I want to make use of avpops module to achieve forward on-no-reply. When >>>> ua1 calls ua2 and ua2 is off-line, or busy, or doesn't answer the phone, >>>> the invite should be forwarded to ua3 and ua3 begins ringing. >>>> >>>> In my tests, first, ua1 sends "cancel" when ua2 is ringing. Ua1 receives >>>> "487 Request Terminated" from SER. Second, ua1 sends "cancel" when ua3 >>>> is >>>> ringing. So, theoretically, ua1 should receive "487 Request Terminated" >>>> after "200 canceling" from SER, right? But, ua1 received "408 Request >>>> Timeout" instead. So strange. How come? >>>> >>>> I think something is wrong with my ser.cfg. Could anyone help me about >>>> this? Thanks in advance. >>>> >>>> Here is my ser.cfg: >>>> >>>> # ------------------ module loading ---------------------------------- >>>> >>>> loadmodule "/usr/local/lib/ser/modules/sl.so" >>>> loadmodule "/usr/local/lib/ser/modules/tm.so" >>>> loadmodule "/usr/local/lib/ser/modules/rr.so" >>>> loadmodule "/usr/local/lib/ser/modules/maxfwd.so" >>>> loadmodule "/usr/local/lib/ser/modules/usrloc.so" >>>> loadmodule "/usr/local/lib/ser/modules/registrar.so" >>>> loadmodule "/usr/local/lib/ser/modules/textops.so" >>>> loadmodule "/usr/local/lib/ser/modules/avp.so" >>>> loadmodule "/usr/local/lib/ser/modules/acc.so" >>>> loadmodule "/usr/local/lib/ser/modules/mysql.so" >>>> loadmodule "/usr/local/lib/ser/modules/dbtext.so" >>>> loadmodule "/usr/local/lib/ser/modules/avpops.so" >>>> #loadmodule "/usr/local/lib/ser/modules/postgres.so" >>>> #loadmodule "/usr/local/lib/ser/modules/flatstore.so" >>>> >>>> # Uncomment this if you want digest authentication >>>> # mysql.so must be loaded ! >>>> loadmodule "/usr/local/lib/ser/modules/auth.so" >>>> #loadmodule "/usr/local/lib/ser/modules/auth_db.so" >>>> >>>> loadmodule "/usr/local/lib/ser/modules/auth_radius.so" >>>> loadmodule "/usr/local/lib/ser/modules/group_radius.so" >>>> loadmodule "/usr/local/lib/ser/modules/uri_radius.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", 1) >>>> >>>> # -- auth params -- >>>> # Uncomment if you are using auth module >>>> # >>>> #modparam("auth_db", "calculate_ha1", yes) >>>> # >>>> # If you set "calculate_ha1" parameter to yes (which true in this >>>> config), >>>> # uncomment also the following parameter) >>>> # >>>> #modparam("auth_db", "password_column", "password") >>>> >>>> # -- rr params -- >>>> # add value to ;lr param to make some broken UAs happy >>>> >>>> modparam("rr", "enable_full_lr", 1) >>>> modparam("tm", "fr_timer", 10 ) >>>> modparam("tm", "fr_inv_timer", 20) >>>> modparam("tm", "noisy_ctimer", 1) >>>> >>>> modparam("auth_radius|uri_radius|group_radius", "radius_config", >>>> "/usr/local/etc/radiusclient-ng/radiusclient.conf") >>>> >>>> modparam("avpops","avp_url","mysql://ser:heslo@localhost/ser") >>>> modparam("avpops","avp_table","usr_preferences") >>>> # ------------------------- request routing logic ------------------- >>>> >>>> # main routing logic >>>> >>>> route{ >>>> # initial sanity checks -- messages with >>>> # max_forwards==0, or excessively long requests >>>> if (!mf_process_maxfwd_header("10")) { >>>> sl_send_reply("483","Too Many Hops"); >>>> break; >>>> }; >>>> if (msg:len >= 2048 ) { >>>> sl_send_reply("513", "Message too big"); >>>> break; >>>> }; >>>> >>>> if (!method=="REGISTER") record_route(); >>>> >>>> if (loose_route()) { >>>> t_relay(); >>>> break; >>>> }; >>>> >>>> if (method=="INVITE") { >>>> lookup("location"); >>>> #if (!lookup("location")) { >>>> # sl_send_reply("404", "Not Found"); >>>> # break; >>>> #}; >>>> } >>>> >>>> if (method=="REGISTER") { >>>> log("REGISTER\n"); >>>> if (!radius_www_authorize("localhost.localdomain")) { >>>> www_challenge("localhost.localdomain", "0"); >>>> break; >>>> }; >>>> save("location"); >>>> break; >>>> }; >>>> >>>> if (method=="INVITE") { >>>> if (!radius_proxy_authorize("localhost.localdomain")) { >>>> proxy_challenge("localhost.localdomain", "1"); >>>> break; >>>> }; >>>> } >>>> >>>> # if we do not get a positive reply, continue at failure_route[1] >>>> t_on_failure("1"); >>>> # forward the request to all destinations in destination set now >>>> t_relay(); >>>> >>>> #if (!t_relay()) { >>>> # sl_reply_error(); >>>> #}; >>>> } >>>> >>>> failure_route[1] { >>>> >>>> #when caller cancel, terminate the call >>>> if (t_check_status("487")) { >>>> t_reply("487", "Request Terminated"); >>>> break; >>>> }; >>>> >>>> if ( ( avp_db_load("$ruri","s:redirect_on_failure") && >>>> avp_check("redirect_on_failure","eq/i:1"))) { >>>> # User need to forward >>>> log(1, "User wants redirection.\n"); >>>> if ( ( avp_db_load("$ruri","s:redirectnumber") && >>>> !avp_check("redirectnumber","re/^$"))) { >>>> log(1,"first redirect\n"); >>>> #avp_print(); >>>> attr2uri("redirect number"); >>>> lookup("location"); >>>> append_branch(); >>>> #t_on_failure("2"); >>>> t_relay(); >>>> } >>>> else { >>>> t_reply("408", "TimeOut"); >>>> break; >>>> }; >>>> } >>>> else { >>>> t_reply("404", "Not Found"); >>>> break; >>>> }; >>>> } >>>> >>>> >>>> _______________________________________________ >>>> Serusers mailing list >>>> serusers(a)lists.iptel.org >>>> http://lists.iptel.org/mailman/listinfo/serusers >>>> >>> >>> >> >> = = = = = = = = = = = = = = = = = = = = >> >> >> >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> > > = = = = = = = = = = = = = = = = = = = = [View Less]

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RE: [Serusers] SER accounting
by Kapil Dhawan 14 Feb '06

14 Feb '06

No Binu, i never got this issue. I will suggest to grep the logs using ethreal and check timestamps. >From: Binu Nair <binu(a)staff.ownmail.com> >Reply-To: Binu Nair <binu(a)staff.ownmail.com> >To: "Kapil Dhawan" <sersavvy(a)hotmail.com> >Subject: RE: [Serusers] SER accounting >Date: Tue, 14 Feb 2006 13:18:55 +0530 > >Hello Kapil, >Thanx for the reply. > >In fact the timestamps from acc and syslog are the same. But I am comparing >with my … [View More]KDE clock which configured to show the seconds also. I note down >the time when I actually disconnect the call. And this time seems to vary >from acc time. The difference is 3 to 6 seconds. > >But have you observed this? > >Syslog is in the same server. > > > >Thanks & Regards, > >"Research is what I'm doing when I don't know what I'm doing." > >Binu V B >Trikon Electronics Pvt Ltd > > > >Kapil Dhawan wrote: > > >Kindly check with the timestamps you are getting in acc and match it >with > >syslog. Is your syslog a different server. > > > > > > > >>From: Binu Nair <binu(a)staff.ownmail.com> > >>Reply-To: Binu Nair <binu(a)staff.ownmail.com> > >>To: "Kapil Dhawan" <sersavvy(a)hotmail.com> > >>Subject: RE: [Serusers] SER accounting > >>Date: Mon, 13 Feb 2006 14:18:20 +0530 > >> > >>Hello, > >> > >>The difference is not 5:30. In fact, it is some seconds. Mostly, the > >>timestamp that i get in the syslog is 2 to 5 seconds after the actual > >>disconnect time > >> > >> > >>Thanks & Regards, > >> > >>"Research is what I'm doing when I don't know what I'm doing." > >> > >>Binu V B > >>Trikon Electronics Pvt Ltd > >> > >> > >> > >>Kapil Dhawan wrote: > >> > > > >> >It works as per GMT...Add 5:30 hours while you draw them on screen. > >> > > >> >>From: Binu Nair <binu(a)staff.ownmail.com> > >> >>Reply-To: Binu Nair <binu(a)staff.ownmail.com> > >> >>To: serusers(a)lists.iptel.org > >> >>Subject: [Serusers] SER accounting > >> >>Date: Mon, 13 Feb 2006 12:05:03 +0530 > >> >> > >> >>Hello, > >> >> > >> >>I have installed ser-0.9.6 and I was able to make calls. But when I >make > >> >>the call logs using log() and xlog() functions, value of the >timestamp > >>that > >> >>comes in the log is found to be different from the actual timestamp. > >> >> > >> >> I used (%Ts,%Tf) parameters in xlog() function to create the log. >Is > >> >>there any way I can log the call disconnect time? > >> >> > >> >> > >> >>Thanks & Regards, > >> >> > >> >>"Research is what I'm doing when I don't know what I'm doing." > >> >> > >> >>Binu V B > >> >>Trikon Electronics Pvt Ltd > >> >> > >> >> > >> >> > >> > > >> > > >> >>_______________________________________________ > >> >>Serusers mailing list > >> >>serusers(a)lists.iptel.org > >> >>http://lists.iptel.org/mailman/listinfo/serusers > >> > > >> >_________________________________________________________________ > >> >Spice up your IM conversations. New, colorful and animated emoticons. >Get > >> >chatting! http://server1.msn.co.in/SP05/emoticons/ > >> > >> > > > >_________________________________________________________________ > >Shah Rukh fan? Know all about the Baadshah of Bollywood. On MSN Search > >http://server1.msn.co.in/profile/shahrukh.asp > > _________________________________________________________________ How good are you in a Formula One car? Play now http://server1.msn.co.in/sp05/tataracing/onlinegame.asp [View Less]

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RE: [Serusers] SER accounting
by Kapil Dhawan 14 Feb '06

14 Feb '06

Kindly check with the timestamps you are getting in acc and match it with syslog. Is your syslog a different server. >From: Binu Nair <binu(a)staff.ownmail.com> >Reply-To: Binu Nair <binu(a)staff.ownmail.com> >To: "Kapil Dhawan" <sersavvy(a)hotmail.com> >Subject: RE: [Serusers] SER accounting >Date: Mon, 13 Feb 2006 14:18:20 +0530 > >Hello, > >The difference is not 5:30. In fact, it is some seconds. Mostly, the >timestamp that i get in the … [View More]

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Re: Re: [Serusers] a question about "cancel" in forwarding scenario
by zhangshuai 14 Feb '06

14 Feb '06

Dear Greger V. Teigre, Thanks for your quick reply. I have read the sergettingstarted05.pdf (Based upon SER version 0.9.3) and tested the onr.cfg in ser-0.9.3.GettingStarted.1.3.tar.gz/ser-0.9.3/examples many times. SER sent out "408" to caller immediately after received "486" from ua3. I don't know how does ser's failure route deal with this kind of "cancel" when the invite has been forwarded. Any ideas will help me a lot. Thanks! Here is the onr.cfg: # ------------------ module loading … [View More]---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" # ----------------- setting module-specific parameters --------------- # -- tm params -- # set time for which ser will be waiting for a final response; # fr_inv_timer sets value for INVITE transactions, fr_timer # for all others modparam("tm", "fr_inv_timer", 15 ) modparam("tm", "fr_timer", 10 ) # ------------------------- request routing logic ------------------- # main routing logic route{ # for testing purposes, simply okay all REGISTERs if (method=="REGISTER") { log("REGISTER"); sl_send_reply("200", "ok"); break; }; # try these two destinations first in parallel; the second # destination is targeted to sink port -- that will make ser # wait until timer hits seturi("sip:200@218.97.252.35:5060"); #append_branch("sip:300@218.97.252.36:5060"); # if we do not get a positive reply, continue at reply_route[1] t_on_failure("1"); # forward the request to all destinations in destination set now t_relay(); } failure_route[1] { #when caller cancel, terminate the call if (t_check_status("487")) { t_reply("487", "Request Terminated"); break; }; # forwarding failed -- try again at another destination append_branch("sip:300@218.97.252.44:5060"); log(1,"first redirection\n"); # if this alternative destination fails too, proceed to reply_route[2] #t_on_failure("2"); t_relay(); } Thanks in advance. Have a good day! Shuai >The internal SER timer hits and the request times out (the request is not >terminated). >See SER-GettingStarted doc at ONsip.org for a full example of call features >like forwarding. >g-) >----- Original Message ----- >From: "zhangshuai" <zhangshuai(a)goldentek.biz> >To: "serusers" <serusers(a)lists.iptel.org> >Cc: "wuyuan" <wuyuan(a)goldentek.biz> >Sent: Friday, February 10, 2006 11:18 AM >Subject: [Serusers] a question about "cancel" in forwarding scenario > > >> Dear All, >> >> >> I want to make use of avpops module to achieve forward on-no-reply. When >> ua1 calls ua2 and ua2 is off-line, or busy, or doesn't answer the phone, >> the invite should be forwarded to ua3 and ua3 begins ringing. >> >> In my tests, first, ua1 sends "cancel" when ua2 is ringing. Ua1 receives >> "487 Request Terminated" from SER. Second, ua1 sends "cancel" when ua3 is >> ringing. So, theoretically, ua1 should receive "487 Request Terminated" >> after "200 canceling" from SER, right? But, ua1 received "408 Request >> Timeout" instead. So strange. How come? >> >> I think something is wrong with my ser.cfg. Could anyone help me about >> this? Thanks in advance. >> >> Here is my ser.cfg: >> >> # ------------------ module loading ---------------------------------- >> >> loadmodule "/usr/local/lib/ser/modules/sl.so" >> loadmodule "/usr/local/lib/ser/modules/tm.so" >> loadmodule "/usr/local/lib/ser/modules/rr.so" >> loadmodule "/usr/local/lib/ser/modules/maxfwd.so" >> loadmodule "/usr/local/lib/ser/modules/usrloc.so" >> loadmodule "/usr/local/lib/ser/modules/registrar.so" >> loadmodule "/usr/local/lib/ser/modules/textops.so" >> loadmodule "/usr/local/lib/ser/modules/avp.so" >> loadmodule "/usr/local/lib/ser/modules/acc.so" >> loadmodule "/usr/local/lib/ser/modules/mysql.so" >> loadmodule "/usr/local/lib/ser/modules/dbtext.so" >> loadmodule "/usr/local/lib/ser/modules/avpops.so" >> #loadmodule "/usr/local/lib/ser/modules/postgres.so" >> #loadmodule "/usr/local/lib/ser/modules/flatstore.so" >> >> # Uncomment this if you want digest authentication >> # mysql.so must be loaded ! >> loadmodule "/usr/local/lib/ser/modules/auth.so" >> #loadmodule "/usr/local/lib/ser/modules/auth_db.so" >> >> loadmodule "/usr/local/lib/ser/modules/auth_radius.so" >> loadmodule "/usr/local/lib/ser/modules/group_radius.so" >> loadmodule "/usr/local/lib/ser/modules/uri_radius.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", 1) >> >> # -- auth params -- >> # Uncomment if you are using auth module >> # >> #modparam("auth_db", "calculate_ha1", yes) >> # >> # If you set "calculate_ha1" parameter to yes (which true in this config), >> # uncomment also the following parameter) >> # >> #modparam("auth_db", "password_column", "password") >> >> # -- rr params -- >> # add value to ;lr param to make some broken UAs happy >> >> modparam("rr", "enable_full_lr", 1) >> modparam("tm", "fr_timer", 10 ) >> modparam("tm", "fr_inv_timer", 20) >> modparam("tm", "noisy_ctimer", 1) >> >> modparam("auth_radius|uri_radius|group_radius", "radius_config", >> "/usr/local/etc/radiusclient-ng/radiusclient.conf") >> >> modparam("avpops","avp_url","mysql://ser:heslo@localhost/ser") >> modparam("avpops","avp_table","usr_preferences") >> # ------------------------- request routing logic ------------------- >> >> # main routing logic >> >> route{ >> # initial sanity checks -- messages with >> # max_forwards==0, or excessively long requests >> if (!mf_process_maxfwd_header("10")) { >> sl_send_reply("483","Too Many Hops"); >> break; >> }; >> if (msg:len >= 2048 ) { >> sl_send_reply("513", "Message too big"); >> break; >> }; >> >> if (!method=="REGISTER") record_route(); >> >> if (loose_route()) { >> t_relay(); >> break; >> }; >> >> if (method=="INVITE") { >> lookup("location"); >> #if (!lookup("location")) { >> # sl_send_reply("404", "Not Found"); >> # break; >> #}; >> } >> >> if (method=="REGISTER") { >> log("REGISTER\n"); >> if (!radius_www_authorize("localhost.localdomain")) { >> www_challenge("localhost.localdomain", "0"); >> break; >> }; >> save("location"); >> break; >> }; >> >> if (method=="INVITE") { >> if (!radius_proxy_authorize("localhost.localdomain")) { >> proxy_challenge("localhost.localdomain", "1"); >> break; >> }; >> } >> >> # if we do not get a positive reply, continue at failure_route[1] >> t_on_failure("1"); >> # forward the request to all destinations in destination set now >> t_relay(); >> >> #if (!t_relay()) { >> # sl_reply_error(); >> #}; >> } >> >> failure_route[1] { >> >> #when caller cancel, terminate the call >> if (t_check_status("487")) { >> t_reply("487", "Request Terminated"); >> break; >> }; >> >> if ( ( avp_db_load("$ruri","s:redirect_on_failure") && >> avp_check("redirect_on_failure","eq/i:1"))) { >> # User need to forward >> log(1, "User wants redirection.\n"); >> if ( ( avp_db_load("$ruri","s:redirectnumber") && >> !avp_check("redirectnumber","re/^$"))) { >> log(1,"first redirect\n"); >> #avp_print(); >> attr2uri("redirect number"); >> lookup("location"); >> append_branch(); >> #t_on_failure("2"); >> t_relay(); >> } >> else { >> t_reply("408", "TimeOut"); >> break; >> }; >> } >> else { >> t_reply("404", "Not Found"); >> break; >> }; >> } >> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers >> > > = = = = = = = = = = = = = = = = = = = = [View Less]

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[Serusers] Problem with CANCEL and INVITE (2)
by Benito Caracuel 14 Feb '06

14 Feb '06

Hello all, UAC -------------- SER ----------------- UAS INVITE ---------->INVITE------------>INVITE <----------TRYING CANCEL------------> <-----------TRYING If UAC sends an INVITE and then sends a CANCEL, this CANCEL doesn't stop SER. SER is still processing the INVITE to UAS, why? This is my ser.cfg: route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("16")) { … [View More]sl_send_reply("483","Too Many Hops"); }; if (msg:len > max_len) { sl_send_reply("513", "Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) # if (uri==myself) { # if (uri=~"^sip:(.+@)?(10\.22\.1\.[0-9]+|(.+@)?(10\.22\.2\.[0-9]+|(.+\.)?T\.local)([:;\?].*)?$"){ # if (uri=~"^sip:(.+@)?([0-9]+\.[0-9]+\.[0-9]+\.[0-9]+|(.+\.)?T\.local)([:;\?].*)?$"){ if (uri=~"^sip:(.+@)?(.+)([:;\?].*)?$"){ if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if ((!www_authorize("10.22.1.249", "subscriber"))) { # www_challenge("10.22.1.249", "0"); # break; # }; save("location"); break; }; if (method=="BYE") { t_relay(); sl_send_reply("200", "OK"); break; }; if (method=="ACK") { t_relay(); break; }; if (method=="CANCEL") { t_relay(); sl_send_reply("200", "OK"); break; }; if ((method=="INVITE")&&(!lookup("location"))) { exec_msg("printenv SRCIP; /bin/bash /usr/local/ser/sbin/sms"); sl_send_reply("100", "Trying"); exec_msg("sleep 10"); if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("404", "Not on-line"); break; } else { t_relay(); break; } }; if (method=="SUBSCRIBE") { if (!t_newtran()) { # log(1, "newtran error\n"); sl_reply_error(); }; handle_subscription("registrar"); break; }; if (method=="PUBLISH") { if (!t_newtran()) { # log(1, "newtran error\n"); sl_reply_error(); }; handle_publish("registrar"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; } append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } Help me please. Thank you. > >_______________________________________________ >Serusers mailing list >serusers(a)lists.iptel.org >http://lists.iptel.org/mailman/listinfo/serusers [View Less]

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[Serusers] CDR generation - ser-0.9.6
by Binu Nair 14 Feb '06

14 Feb '06

Hello, I have seen the archives of ser users. It says setflag() can be used to log the call. I used that and I am getting a call log. But my problem is the timestamp . The disconnect time , that is, the timestamp when the method is BYE, seems to differ from the actual time by some seconds. This is not always true. Still , I noticed this many a time. Has anyone experienced this? Please help Thanks & Regards, "Research is what I'm doing when I don't know what I'm doing." … [View More]

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[Users] Converting from SER to OpenSER
by Brian McCrary 13 Feb '06

13 Feb '06

Hello, I am currently using SER 0.9.4 in a production environment. I've patched it with various modules that OpenSER has, but would like to convert over to the OpenSER project to be able to use a lot of the new functionality. I've comiled the latest CVS source (the main release has Solaris compile problems that have been fixed in CVS, hence the CVS source). The process doesn't look too difficult, a few things to be changed in the config file, namely break and return. However, I'm not sure … [View More]

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[Users] new avpops
by Helge Waastad 13 Feb '06

13 Feb '06

Hi, earlier I had: modparam("avpops","avp_aliases","ouri=i:704") I did (before lcr processing): avp_write("$ruri/username","$ouri"); [LCR processing] (put back the original) avp_pushto("$ruri/username","$ouri"); now I'm using: avp_write("$oU","$avp(ouri)"); [LCR prcessing] (try to write back the old username) avp_pushto("$ru/username","$avp(ouri)"); This does not work... I'm not quite sure if I'm using the new format correct. br hw -- Helge Waastad Senior Konsulent Systemavdelingen Smartnet

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[Serusers] changing caller number
by miliczo 13 Feb '06

13 Feb '06

Hi, How can I change a caller number? I tried with avp but no success. There must be a simpler way! Is there a core command for something like that. I tried rewritehost & rewriteuser in the REGISTER part of the cfg, but nothing. best regards, Zoran

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[Serusers] Problem with CANCEL and INVITE
by Benito Caracuel 13 Feb '06

13 Feb '06

Hello all, UAC -------------- SER ----------------- UAS INVITE ---------->INVITE------------>INVITE <----------TRYING CANCEL------------> <-----------TRYING If UAC sends an INVITE and then sends a CANCEL, this CANCEL doesn't stop SER. SER is still processing the INVITE to UAS, why? Help me please. Thank you.

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[Serusers] Changing From_uri ...
by Carlos Loarca 13 Feb '06

13 Feb '06

Hi everyone: I want to change the "From" field on the SIP header to any value. For instance, I dont want to show the caller ID on the end-point for security reasons. Then I just want to show "restricted number" instead. Which is the module that would alow me to do so??? Cheers, /Carlos Loarca __________________________________________________ Do You Yahoo!? Tired of spam? Yahoo! Mail has the best spam protection around http://mail.yahoo.com

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[Serusers] Problem with ACK and INVITE
by Benito Caracuel 13 Feb '06

13 Feb '06

Hello all, I have: UAC ---------- SER ------------ UAS I have a problem with ACK. When I send an INVITE, UAS sends a 200 OK and then UAC sends ACK to finalize the INVITE. The problem is that SER doesn't reply this ACK from UAC to UAS. This is my ser.cfg: route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("16")) { sl_send_reply("483","Too Many Hops"); }; if (msg:len > max_len) { sl_send_reply("513", "… [View More]Message too big"); break; }; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) # if (uri==myself) { # if (uri=~"^sip:(.+@)?(10\.22\.1\.[0-9]+|(.+@)?(10\.22\.2\.[0-9]+|(.+\.)?T\.local)([:;\?].*)?$"){ # if (uri=~"^sip:(.+@)?([0-9]+\.[0-9]+\.[0-9]+\.[0-9]+|(.+\.)?T\.local)([:;\?].*)?$"){ if (uri=~"^sip:(.+@)?(.+)([:;\?].*)?$"){ if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if ((!www_authorize("10.22.1.249", "subscriber"))) { # www_challenge("10.22.1.249", "0"); # break; # }; save("location"); break; }; if (method=="BYE") { t_relay(); sl_send_reply("200", "OK"); break; }; if (method=="ACK") { t_relay(); break; }; if (method=="CANCEL") { t_relay(); sl_send_reply("200", "OK"); break; }; if (method=="INVITE") { record_route(); }; if ((method=="INVITE")&&(!lookup("location"))) { record_route(); exec_msg("printenv SRCIP; /bin/bash /usr/local/ser/sbin/script"); sl_send_reply("100", "Trying"); exec_msg("sleep 10"); if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("100", "Trying"); exec_msg("sleep 10"); } else { t_relay(); break; } if (!lookup("location")){ sl_send_reply("404", "Not on-line"); break; } else { t_relay(); break; } }; if (method=="SUBSCRIBE") { if (!t_newtran()) { # log(1, "newtran error\n"); sl_reply_error(); }; handle_subscription("registrar"); break; }; if (method=="PUBLISH") { if (!t_newtran()) { # log(1, "newtran error\n"); sl_reply_error(); }; handle_publish("registrar"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; } append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (!t_relay()) { sl_reply_error(); }; } Could someone help me? Thank you. [View Less]

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[Users] APPs for call center
by Jolly M. Recto 13 Feb '06

13 Feb '06

Hi, I am currently using the Openser debian version with mysql. I am quite impress on the performance of the server but wondering if anyone is using the system for a call center with different Application running on different server. Right now my setup is a astguiclient with two server which depends more on the power of it (mem and processor)... //jollyr

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[Serusers] [Auto-Reply] Re: Serusers Digest, Vol 34, Issue 15
by Binu Nair 13 Feb '06

13 Feb '06

Hello, This is an auto reply. Will contact u

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[Users] avpops updates
by Daniel-Constantin Mierla 13 Feb '06

13 Feb '06

Hello, the avpops module was refurbished to benefit better of pseudo-variables. In addition, the avp and hdr pseudo-variables can have dynamic names. Here are some details: - all parameters being avp names in avpops functions MUST be specified in the same format as pseudo-variables: - $avp(i:23) - refers to the avp having the integer id 23 - $avp(s:test) - refers to the avp having the name 'test' - $avp($aliasid) - refers to the avp defined by avp alias 'aliasid' - example:… [View More] avp_write("i:10", "$avp(i:30)") - write integer value 10 into the avp having the id 30 - avps and hdrs pseudo-variables can have dynamic names: -$avp($pvar) refers to the avp having the name the value of $pvar -$hdr($pvar) refers to the hdr having the name the value of $pvar - please note that the pseudo-variables specifiers have priority over avp aliases (if you define an avp alias $ru, $avp(ru) will refer to the avp having the name the value of R-URI) - in all parameters of avpops module can be used pseudo-variables where are expected values - avp_check("$fd", "eq/$td") returns true if the domain of From uri is the same with domain part of To uri - avp_write("$tU", "$avp($fU)") - writes the username in To header in the avp having the name the username from From header - avp_pushto() - cannot add anymore headers - same functionality is provided by the textops module via append_hf() and append_to_reply() functions - the format of the first parameter of avp_pushto() follows the pseudo-variables names as well. - avp_pushto("$ru[/username|domain]", "...") to rewrite parts of R-URI - avp_pushto("$du", "...") to rewrite dst_uri - for developers, internal structure int_str has changed from (int, str*) to (int,str) to avoid extra reference - the documentation will be updated soon (module readme and avpops tutorial) Still to come: - support for global avps: avps shared between all processes during the openser lifetime - support for script avps: avps specific per process with life time only during the script execution (temporary avps, which do not need to last too much, using private memory) - support for local avps: avps specific per process kept during openser lifetime (avps which need to be kept for each process, without synchronized access, using private memory) - avp_db_query() - ability to send a raw query (simple, joins, unions ...) to database and load the result in avps (e.g., avp_db_query("select first_name, last_name from subscriber where username='$tU' and domain='$td'", "$avp(first_name);$avp(last_name)")) Testing and feedback are more than welcome. Cheers, Daniel [View Less]

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[Users] refer message handling?
by Frogger 13 Feb '06

13 Feb '06

I have a successful implementation of openser 0.9.5 and its been working very nicely. I am faced with transferring of calls from pstn to userC by way of userB and I am struggling. path: PSTN=>GW=>SBC=>openser=>uB uB transfers call to uC Call is failing to transfer as "refer" message is not properly handled by my openser script. Any guidance would be appreciated. Transfer works when all legs are from registered clients. Only when the initial call leg is from the PSTN does it … [View More]

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[Serusers] Associate a # with a SIP client
by Nick Hoffman 13 Feb '06

13 Feb '06

Hey guys. How can you configure SER to associate a particular phone number, say 12341234, with a particular SIP client/user agent? The aim is to be able to call them with either john(a)foo.bar or 12341234, but ensure that SER associates 12341234 with john(a)foo.bar rather than john(a)foo.bar telling SER "I can be reached at 12341234". Cheers! -- Nick e: nick.hoffman(a)altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make … [View More]

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[Serusers] SER accounting
by Binu Nair 13 Feb '06

13 Feb '06

Hello, I have installed ser-0.9.6 and I was able to make calls. But when I make the call logs using log() and xlog() functions, value of the timestamp that comes in the log is found to be different from the actual timestamp. I used (%Ts,%Tf) parameters in xlog() function to create the log. Is there any way I can log the call disconnect time? Thanks & Regards, "Research is what I'm doing when I don't know what I'm doing." Binu V B Trikon Electronics Pvt Ltd

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[Serusers] CDRs/accounting
by Nick Hoffman 13 Feb '06

13 Feb '06

Hey guys, so I've been doing lots of reading up on how SER works, but I haven't found all that much information how SER deals with CDRs. [How] can CDRs be stored in a database? [How] can CDRs be sent to another running process in realtime? Thanks! -- Nick e: nick.hoffman(a)altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not waive any privilege, confidentiality or copyright associated with it.

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[Users] RE: [Openswan Users] Openswan VPN MOnitor
by Fernando Rodriguez 12 Feb '06

12 Feb '06

I get this response /usr/libexec/ipsec/eroute: NETKEY does not support eroute table. Im using kernel 2.6.15 Thanks _____ De: users-bounces(a)openswan.org [mailto:users-bounces@openswan.org] En nombre de Christophe Ngo Van Duc Enviado el: Viernes, 10 de Febrero de 2006 02:19 p.m. Para: users(a)openswan.org Asunto: Re: [Openswan Users] Openswan VPN MOnitor Wouldn't "ipsec eroute" gives you this information ? On 2/10/06, Fernando Rodriguez <frod(a)aitelecom.net> wrote: … [View More]

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[Serusers] SER doubt
by Manoj Kumar Patel 12 Feb '06

12 Feb '06

hi ser users.... any one can help me .... I am ser user. My requirement is to farward rtp streams to multiple user agents which are interested to stablish media session (i.e sending 200 OK ). with regards manoj

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[Serusers] parse_msg() SIP parser test program
by Hendrik Scholz 12 Feb '06

12 Feb '06

Hi! Every once in a while I came across certain messages that apparently could not be parsed properly. To track down these I've written a small helper program. Put this into the tests/ subdirectory and follow the compilation instructions! Until now I only found STUN messages that used port 5060 so I might need to increase debugging inside the SER code. Here's the README: This program feeds SIP messages into the SER SIP parser to analyze its behaviour. Valid data sources are live traffic … [View More]

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[Serusers] Help me for simple ser installation.
by Friend 12 Feb '06

12 Feb '06

Hi Friend, I am from India. Just following the Dan Austin SER HowTo and trying to install and configure ser server for the past one week no result. Feel helpless, can any one help me to make it successful. Am using ser-0.8.12-0.rpm ser-mysql-0.8.12-0.rpm RedHat Enterprise Linux 3 bind 9 MySQL 4.0.20-0 Machine: Intel x86 Can you provide me any sample files - ser.cnf, serctl, ser_mysql.sh from a successful installation. Awaiting your valuable … [View More]

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[Serusers] a question about "cancel" in forwarding scenario
by zhangshuai 11 Feb '06

11 Feb '06

Dear All, I want to make use of avpops module to achieve forward on-no-reply. When ua1 calls ua2 and ua2 is off-line, or busy, or doesn't answer the phone, the invite should be forwarded to ua3 and ua3 begins ringing. In my tests, first, ua1 sends "cancel" when ua2 is ringing. Ua1 receives "487 Request Terminated" from SER. Second, ua1 sends "cancel" when ua3 is ringing. So, theoretically, ua1 should receive "487 Request Terminated" after "200 canceling" from SER, right? But, ua1 received "… [View More]408 Request Timeout" instead. So strange. How come? I think something is wrong with my ser.cfg. Could anyone help me about this? Thanks in advance. Here is my ser.cfg: # ------------------ module loading ---------------------------------- loadmodule "/usr/local/lib/ser/modules/sl.so" loadmodule "/usr/local/lib/ser/modules/tm.so" loadmodule "/usr/local/lib/ser/modules/rr.so" loadmodule "/usr/local/lib/ser/modules/maxfwd.so" loadmodule "/usr/local/lib/ser/modules/usrloc.so" loadmodule "/usr/local/lib/ser/modules/registrar.so" loadmodule "/usr/local/lib/ser/modules/textops.so" loadmodule "/usr/local/lib/ser/modules/avp.so" loadmodule "/usr/local/lib/ser/modules/acc.so" loadmodule "/usr/local/lib/ser/modules/mysql.so" loadmodule "/usr/local/lib/ser/modules/dbtext.so" loadmodule "/usr/local/lib/ser/modules/avpops.so" #loadmodule "/usr/local/lib/ser/modules/postgres.so" #loadmodule "/usr/local/lib/ser/modules/flatstore.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! loadmodule "/usr/local/lib/ser/modules/auth.so" #loadmodule "/usr/local/lib/ser/modules/auth_db.so" loadmodule "/usr/local/lib/ser/modules/auth_radius.so" loadmodule "/usr/local/lib/ser/modules/group_radius.so" loadmodule "/usr/local/lib/ser/modules/uri_radius.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", 1) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth_db", "calculate_ha1", yes) # # If you set "calculate_ha1" parameter to yes (which true in this config), # uncomment also the following parameter) # #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) modparam("tm", "fr_timer", 10 ) modparam("tm", "fr_inv_timer", 20) modparam("tm", "noisy_ctimer", 1) modparam("auth_radius|uri_radius|group_radius", "radius_config", "/usr/local/etc/radiusclient-ng/radiusclient.conf") modparam("avpops","avp_url","mysql://ser:heslo@localhost/ser") modparam("avpops","avp_table","usr_preferences") # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests if (!mf_process_maxfwd_header("10")) { sl_send_reply("483","Too Many Hops"); break; }; if (msg:len >= 2048 ) { sl_send_reply("513", "Message too big"); break; }; if (!method=="REGISTER") record_route(); if (loose_route()) { t_relay(); break; }; if (method=="INVITE") { lookup("location"); #if (!lookup("location")) { # sl_send_reply("404", "Not Found"); # break; #}; } if (method=="REGISTER") { log("REGISTER\n"); if (!radius_www_authorize("localhost.localdomain")) { www_challenge("localhost.localdomain", "0"); break; }; save("location"); break; }; if (method=="INVITE") { if (!radius_proxy_authorize("localhost.localdomain")) { proxy_challenge("localhost.localdomain", "1"); break; }; } # if we do not get a positive reply, continue at failure_route[1] t_on_failure("1"); # forward the request to all destinations in destination set now t_relay(); #if (!t_relay()) { # sl_reply_error(); #}; } failure_route[1] { #when caller cancel, terminate the call if (t_check_status("487")) { t_reply("487", "Request Terminated"); break; }; if ( ( avp_db_load("$ruri","s:redirect_on_failure") && avp_check("redirect_on_failure","eq/i:1"))) { # User need to forward log(1, "User wants redirection.\n"); if ( ( avp_db_load("$ruri","s:redirectnumber") && !avp_check("redirectnumber","re/^$"))) { log(1,"first redirect\n"); #avp_print(); attr2uri("redirect number"); lookup("location"); append_branch(); #t_on_failure("2"); t_relay(); } else { t_reply("408", "TimeOut"); break; }; } else { t_reply("404", "Not Found"); break; }; } [View Less]

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[Serusers] Problem with SER Accounting
by mkumar＠mantragroup.com 11 Feb '06

11 Feb '06

Hi All, I was able to install SER and call out successfully. I configured ser.cfg after seeing some examples in voip-info.org, almost all worked except accounting. I am using stable ser 0.9.4 version and followed instructions in http://www.voip-info.org/tiki-index.php?page=SER+example+accounting. When I check the database for records related tables are empty. I did not install radius, is this required to enable accounting? I am attaching my ser.cfg, please check it and guide me. Any help will … [View More]

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[Serusers] changing domain
by postmaster＠orisonweb.com 10 Feb '06

10 Feb '06

hello, we had configured the mysql database and html site on hard disk.everything was working fine with domain alise sip.abcd.com for abcd.com . ip address was . later we wated to change it from sip.abcd.com to simply abcd.com . so we just removed child name sip from the domain now everything is working fine on web end but when we try to connect or log via x10 like software it prompts a error login failure contact network admin message while login in via X10. we have completely changed mysql … [View More]

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[Serusers] SER amd Minute Provider
by mkumar＠mantragroup.com 10 Feb '06

10 Feb '06

Hi All, I purchased some minutes from a minute provider and looking to configure that with SER. But I did not find any documentation or info on how to do that? Please tell me how can I do this or lead me to related documentaion or links. Thanks, Manoj.

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[Serusers] Aliases
by Nick Hoffman 10 Feb '06

10 Feb '06

Just a couple of quick questions relating to aliases :) If the environment variable SIP_DOMAIN is not set, why does adding an alias with serctl fail?: nick@customer-dev3:~> sudo serctl alias add 123456 sip:nick@altcall.com domain unknown: use usernames with domain or set default domain in SIP_DOMAIN Once I set SIP_DOMAIN like so: export SIP_DOMAIN=altcall.com Adding an alias as described above works, but why is the alias not listed in the MySQL database?: mysql> … [View More]

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[Users] Question on SIP authentication with users from OpenSER
by Barry Flanagan 10 Feb '06

10 Feb '06

Hi, This is more an Asterisk question, I know, but I am sure a good many of you have similar setups.... We are using Asterisk 1.2.3 with RealTime for PSTN and Voicemail where users register with an OpenSER cluster (2 nodes currently). When they request PSTN they are forwarded to * where they have entries in SIP realtime database. This ensures that they get their correct CallerID and context, etc. This is working fine at present, where I have the SIP users set up with the following relevant … [View More]

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[Users] REMINDER: TM functions
by Bogdan-Andrei Iancu 10 Feb '06

10 Feb '06

Hi, this is reminder for the TM API clean-up. Friday 10th, the old TM functions like t_relay_to_xxx(), t_replicate_xxx(), t_forward_nonack_udp(), etc will be definitely removed. Here is the original email describing the changes: http://www.openser.org/pipermail/devel/2006-January/001659.html this changes will affect *only* the devel branch. For its users, please migrate your scripts to new function prototypes. thanks and regards, Bogdan

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[Users] sorry wrong address
by Pjothi 10 Feb '06

10 Feb '06

Sorry for the last email regarding IPSec. Inadvertently sent to the wrong group

1 0

[Users] UDP Encapsulation in Suse 10 with OpenSwan?
by Pjothi 10 Feb '06

10 Feb '06

Hello all, I am successful in setting up IPSec with preshared keys (both transport and tunnel ) using racoon between two Suse Linux 10 machines which uses openswan and comes pre-configured. I would like to know if there is already UDP encapsualation capability integrated with Freeswan implementation on Suse Linux ( kernel release 2.6.13-15). Because the README files for NAT traversal says, I need to patch up the kernel. Do I have to do it and if yes, where to get the patches from, kindly … [View More]

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[Serusers] SER Newbie Request
by mkumar＠mantragroup.com 10 Feb '06

10 Feb '06

Hi All, Can anyone please send me the ser.cfg file or lines of code which will allow me to configure SER with a minute provider. Thanks, Kumar.

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[Serusers] Up-to-date documentation
by nick.hoffman＠altcall.com 10 Feb '06

10 Feb '06

Hi guys. As I'm new to using SER and wanting to do my own research before posting to the mailing list, I've scoured around for documentation, how-tos, guides, etc on SER. However, just about all of the information I've found is severely out of date. For instance, the SER Admin's Guide (iptel.org/ser/doc/seruser/seruser.html) was last updated in 2002, and the SER How-To (iptel.org/ser/doc/ser-howto/ser-Howto.html) suggests this: mysql> update subscriber set perms=?admin? where USER_ID=?… [View More]

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RE:[Serusers] SER amd Minute Provider
by Robert Volta 10 Feb '06

10 Feb '06

I was hunting around for an answer to this one myself. My minutes provider happens to require a (user:password@myprovider.com) format to be sent for authentication. Can this also be accomplished in the rewrite?? Any examples would be greatly appreciated. Thanks in advance, Rob _________________________________________________________________ Dont just search. Find. Check out the new MSN Search! http://search.msn.click-url.com/go/onm00200636ave/direct/01/

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[Serusers] domain_dump and domain_reload
by Nick Hoffman 10 Feb '06

10 Feb '06

Hi folks. I have several domains in the SER database's domain table. When I run sudo serctl domain show this error occurs: 500 command 'domain_dump' not available And when I run sudo serctl domain add foo.bar and give the MySQL password, this error occurs: 500 command 'domain_reload' not available Any suggestions as to what I might be doing wrong? Cheers! -- Nick e: nick.hoffman(a)altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please … [View More]

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[Serusers] Asterisk + SER
by Nick Hoffman 09 Feb '06

09 Feb '06

When using Asterisk and SER together, should SER place calls to the PSTN, and Asterisk only deal with "special features" such as voicemail, queues, autoattendants, etc? Or should SER be used ONLY as a proxy/registrar, and all calls be routed to Asterisk so that Asterisk places the calls to the PSTN? Cheers! -- Nick e: nick.hoffman(a)altcall.com p: +61 7 5591 3588 f: +61 7 5591 6588 If you receive this email by mistake, please notify us and do not make any use of the email. We do not … [View More]

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[Serusers] SIP network engineers wanted
by Adrian Georgescu 09 Feb '06

09 Feb '06

AG Projects is hiring skilled engineers with experience in operations and troubleshooting of SER/OpenSER/MediaProxy/Radius environments. The role is responsible for software rollout and third-level customer support. Both permanent (the Netherlands) or tele-working options are available. If you wish to apply, please send your resume to ag(a)ag-projects.com Regards, Adrian Georgescu

1 0

[Serusers] Precence Module
by Piergiorgio Venuti 09 Feb '06

09 Feb '06

Hi all, I have a problem with pa module, when I log in ser with my Windows Messanger 4.7 there isn't problem, when another user log in ser he can see my presence but I don't see him, if I log out and log in now I can see his status but he don't see me. I suppose that it's related to a function that update my mysql. Someone have any idea? I have attached my ser.cfg # # $Id: ser.cfg,v 1.25.2.1 2005/02/18 14:30:44 andrei Exp $ # # simple quick-start config script # # ----------- global … [View More]configuration parameters ------------------------ debug=7 # debug level (cmd line: -dddddddddd) fork=yes log_stderror=yes # (cmd line: -E) listen=ip_interno listen=ip_esterno # hostname matching an alias will satisfy the condition uri==myself". alias=demiurgo.mediaservice.pri #alias=ip_interno alias=demiurgo.mediaservice.net #alias=ip_esterno /* Uncomment these lines to enter debugging mode log_stderror=yes */ check_via=yes # (cmd. line: -v) dns=no # (cmd. line: -r) rev_dns=no # (cmd. line: -R) port=5060 children=7 # fifo - FIFO special file pathname fifo="/tmp/ser_fifo" fifo_db_url="mysql://ser:password@127.0.0.1/ser" # syn_branch - Shall the server use stateful synonym branches? It is # faster but not reboot-safe. Default is yes. syn_branch=yes # memlog - Debugging level for final memory statistics report. Default # is L_DBG -- memory statistics are dumped only if debug is set high. memlog=3 # sip_warning - Should replies include extensive warnings? By default # yes, it is good for trouble-shooting. sip_warning=yes # reply_to_via - A hint to reply modules whether they should send reply # to IP advertised in Via. Turned off by default, which means that # replies are sent to IP address from which requests came. reply_to_via=no # ------------------ module loading ---------------------------------- # Uncomment this if you want to use SQL database loadmodule "/opt/ser/lib/ser/modules/mysql.so" loadmodule "/opt/ser/lib/ser/modules/domain.so" loadmodule "/opt/ser/lib/ser/modules/uri_db.so" loadmodule "/opt/ser/lib/ser/modules/sl.so" loadmodule "/opt/ser/lib/ser/modules/tm.so" loadmodule "/opt/ser/lib/ser/modules/rr.so" loadmodule "/opt/ser/lib/ser/modules/maxfwd.so" loadmodule "/opt/ser/lib/ser/modules/usrloc.so" loadmodule "/opt/ser/lib/ser/modules/registrar.so" loadmodule "/opt/ser/lib/ser/modules/textops.so" loadmodule "/opt/ser/lib/ser/modules/mediaproxy.so" loadmodule "/opt/ser/lib/ser/modules/nathelper.so" loadmodule "/opt/ser/lib/ser/modules/pa.so" # Uncomment this if you want digest authentication # mysql.so must be loaded ! #loadmodule "/opt/ser/lib/ser/modules/auth.so" #loadmodule "/opt/ser/lib/ser/modules/auth_db.so" # ----------------- setting module-specific parameters --------------- # param settings modparam("pa", "use_db",1) modparam("pa","db_url", "mysql://ser:password@127.0.0.1/ser") modparam("pa", "default_expires", 3) modparam("pa","pa_domain", "demiurgo.mediaservice.pri") #modparam("domain", "db_mode", 1) modparam("domain", "db_url", "mysql://ser:password@127.0.0.1/ser") #modparam("domain", "domain_table", "new_name") #modparam("domain", "domain_col", "domain_col") 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("nathelper", "rtpproxy_disable", 1) modparam("nathelper", "natping_interval", 0) modparam("mediaproxy", "natping_interval", 40) modparam("mediaproxy", "mediaproxy_socket", "/var/run/mediaproxy.sock") modparam("mediaproxy", "sip_asymmetrics", "/etc/ser/sip-clients") modparam("mediaproxy", "rtp_asymmetrics", "/etc/ser/rtp-clients") modparam("registrar", "nat_flag", 6) # -- usrloc params -- #modparam("usrloc", "db_mode", 0) # 2 enables write-back to persistent mysql storage for speed # disable=0, write-through=1 # Uncomment this if you want to use SQL database # for persistent storage and comment the previous line modparam("usrloc", "db_mode", 2) # database location modparam("usrloc", "db_url", "mysql://ser:password@127.0.0.1/ser") # minimize write back window - default is 60 seconds modparam("usrloc", "timer_interval", 10) # -- auth params -- # Uncomment if you are using auth module # #modparam("auth", "secret", "johndoessecretphrase") #modparam("auth", "nonce_expire", 600) #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", "db_url", "mysql://ser:password@127.0.0.1/ser") #modparam("auth_db", "user_column", "user") #modparam("auth_db", "domain_column", "domain") #modparam("auth_db", "password_column", "password") # -- rr params -- # add value to ;lr param to make some broken UAs happy modparam("rr", "enable_full_lr", 1) # ------------------------- request routing logic ------------------- # main routing logic route{ # initial sanity checks -- messages with # max_forwards==0, or excessively long requests #if (!mf_process_maxfwd_header("10")) { # sl_send_reply("483","Too Many Hops"); # break; #}; #if (msg:len >= 2048 ) { # sl_send_reply("513", "Message too big"); # break; #}; # we record-route all messages -- to make sure that # subsequent messages will go through our proxy; that's # particularly good if upstream and downstream entities # use different transport protocol if (!method=="REGISTER") record_route(); # subsequent messages withing a dialog should take the # path determined by record-routing if (loose_route()) { # mark routing logic in request append_hf("P-hint: rr-enforced\r\n"); setflag(6); use_media_proxy(); route(1); break; }; if (!uri==myself) { # mark routing logic in request append_hf("P-hint: outbound\r\n"); route(1); break; }; if (method=="BYE" || method=="CANCEL" ) { end_media_session(); }; # if the request is for other domain use UsrLoc # (in case, it does not work, use the following command # with proper names and addresses in it) if (uri==myself) { if (method=="REGISTER") { # Uncomment this if you want to use digest authentication # if (!www_authorize("iptel.org", "subscriber")) { # www_challenge("iptel.org", "0"); # break; # }; save("location"); # pa_handle_registration("demiurgo.mediaservice.pri"); break; }; lookup("aliases"); if (!uri==myself) { append_hf("P-hint: outbound alias\r\n"); route(1); break; }; # native SIP destinations are handled using our USRLOC DB if (!lookup("location")) { sl_send_reply("404", "Not Found"); break; }; }; append_hf("P-hint: usrloc applied\r\n"); route(1); } route[1] { # send it out now; use stateful forwarding as it works reliably # even for UDP2TCP if (isflagset(6) || isflagset(7)) { use_media_proxy(); }; if (!t_relay()) { sl_reply_error(); }; } [View Less]

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[Serusers] limiting no. of concurrent calls ?
by Mike W 09 Feb '06

09 Feb '06

I have a requirement, where i need to configure to set up a maximum no. of allowable concurrent calls . If the number exceeds this, a busy signal is sent back .. Any insights/ suggesions appreciated .. Thanks Mike --------------------------------- Yahoo! Mail - Helps protect you from nasty viruses.

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[Serusers] Multi-Domains / RADIUS Multi-Realms
by Waldo Rubinstein 09 Feb '06

09 Feb '06

I'm trying to setup something like virtual PBX for a few of my client's offices. I was reccommended on this list to look into multi- domains and multi-realms in order to allow, for example, two offices to have the same numbering sequence for their telephone extensions (that is, two distinct clients could have extensions 101, 102, etc) so that when they call the extension number, SER will route the call to the extension within that office. I haven't been able to find much info on … [View More]

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[Users] Question on realm_prefix with multiple prefixes
by Barry Flanagan 09 Feb '06

09 Feb '06

Hi, I seem to recall that realm_prefix for the various modules could take multiple prefixes like: modparam("auth", "realm_prefix", "sip.|sip2.|sip3.") ...but this does not appear to work. Does this capability exist, and if so, what is the syntax? All the docs examples only show a single prefix. Thanks. -Barry Flanagan

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[Users] newbie-nat
by Mark Anthony C. Delfin 09 Feb '06

09 Feb '06

Hello openser users I just installed openser 1.0 , rtpproxy, openser=mysql on centos. calls can be made between UAs with no NAT between them but if one UA is behind nat calls are unsuccessful. does anyone have a sample config for nat calls using openser + rtpproxy. Thanks in advance mark

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[Users] New Feature - Global Statistics Management support
by Bogdan-Andrei Iancu 09 Feb '06

09 Feb '06

Hi everybody, A new feature is present on the development version of openser - Global Statistics Management support: the core includes a Statistics Manager which roll is to collect statistics variables from all over openser and to offer management support (to create and operate them); it also provides a single, centralized point of access to allow external apps or internal modules to get any statistics. How it works? the core and the module exports different statistics variables (via … [View More]the module interface). The Statistics Manager (SM) offers support for auto-initialization, for updating ans resetting variables. The statistics variables are kept into shared mem to be accessible to all processes. Synchronization support is offered by SM. Statistic variable capabilities ------------------------------- be defined with: NO_RESET (it cannot be reset -ex. used mem, register subscribers, etc) NO_SYNC (it will not be synchronized during ops; by their nature, some variables are already synchronized) be updated (added with new val) be reset (set to zero if allowed). Accessing the statistics ------------------------- The SM export two FIFO functions for offering access to the statistics: "get_statistics all" - gets all statistics "get_statistics stat_name" - gets the value of the stat_name variable "get_statistics module_name:" - get all statistics exported by module "module name"; to get statistics from core, use "core" as module_name "reset_statistics stat_name" - sets to 0 the value of the stat_name variable The SM internally offers an API to be used by other modules interested in accessing the statistics. Ex: a future snmp module Current statistics ------------------ For the moment only a few core statistics are defined (we can easily add them in time): received requests : "rcv_reqs" received replies : "rcv_rpls" dropped requests (by scripts or post-script callbacks) : "drp_reqs" dropped replies (by scripts or post-script callbacks) : "drp_rpls" error requests (basic parsing failed) : "err_reqs" error replies (bassic parsing failed) : "err_rpls" How to use it -------------- first to need to enable its compiling - enable the -DSTATISTICS flag in Makefile.def (by default is disabled). Recompile and run openser. Use fifo to get the statistics. Ex: openserctl fifo get_statistics all openserctl fifo get_statistics core: openserctl fifo get_statistics rcv_reqs Next steps ----------- optimization of the SM (better locking for sync, usage of atomic ops if possible) replace old statistics (see stats.c) - the code is not compiling since it depends on some snmp stuff... replace local module statistics (sl,tm,usrloc) with sttistics via SM add more statistics...ideas are welcomed new module to implement dynamic statistics variables to be used from script. regards, bogdan [View Less]

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[Serusers] grandstream budgetone not able to make/take calls
by ryan nalupa 09 Feb '06

09 Feb '06

hello all, i'm ryan and still a SER noob. copied some cfg files from onsip.org and from voip-info.org. basically, i'm using two softphones, at first they're all ok. can make a call, can accept...when i try it on my budgetone hardphone, my desktop phone registers but cannot make a call, can't take calls either. as of now i still don't know what to do with it yet, can anyone help/guide me on how to configure this little problem? i'm using sarge and ser-0.9.4.... is this an issue with NAT? i … [View More]

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[Users] Avoid loop on call forwarding
by Jens Carl 09 Feb '06

09 Feb '06

Hay list, I've got following scenario: I have call forwarding with the help of OpenSER. There are no problems with forwarding within the SIP network. Also the forwarding to a PSTN destination is possible if the caller is from the SIP network. And also the forwarding to a SIP destination if the call comes from a PSTN destination. The only problem is when the caller is from the PSTN network and the callee tries to forward this call to another PSTN destination. one server one … [View More]

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[Users] Natping in "dispatcher scenario"
by Helge Waastad 09 Feb '06

09 Feb '06

Hi, I just wonder: I have a demo scenario in which I have a dispatcher (and using path header) ahead of the registrar(s) and other fun stuff. I am currently implementing the NAT ping function in the registrar(s), but the nathelper function just sends the ping to the contact stored in location. This, of course, does not work in fw scenarios since the relation between the UA and the SIP server is the dispatchers IP:port. Does anyone else find it interesting to have the ping also relayed via the … [View More]

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[Serusers] SER: Measuring ASR & PDD
by Jean-Michel Hiver 08 Feb '06

08 Feb '06

Hi List, At the moment, I log 'INVITE', 'ACK' and 'BYE' with this: # ----------------------------------------------------------------- # Sets the acc accounting flag 4 billing # ----------------------------------------------------------------- if( (method=="INVITE") || (method=="ACK") || (method=="BYE")) { setflag(1); }; To measure (approximately) the ASR, I think it is sort of reasonable to use the formula (number of INVITES) / (number of BYEs + number of INVITES) The … [View More]

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[Users] DISPATCHER not sending outgoing INVITES 100% of the time
by Anders Brownworth 08 Feb '06

08 Feb '06

Hi, I haven't been able to get an outgoing INVITE for every incoming INVITE my dispatcher setup receives. What I'm expecting: When the first INVITE comes from 10.1.50.30 (Asterisk) to 10.1.50.31, (OpenSER 1.0.0 with dispatcher) I want to see an outgoing INVITE from 10.1.50.31 to one of the listed addresses. (10.1.50.[34-37]) What I'm seeing: A tcpdump shows that most of the time (90%) all I see is the incoming INVITE. Seemingly random INVITES produce an outgoing INVITE. Presumably the … [View More]only substantive difference being the CallID. Interestingly, all the outgoing INVITEs are addressed to the same IP. (10.1.50.36) Am I not expecting the right thing? Can anyone shed some light on what I'm doing wrong here? Thanks. openser.cfg: ----------------------------------------------------------------------------- #debug=4 fork=yes log_stderror=no children=4 check_via=no dns=off rev_dns=off port=5060 mpath="/usr/local/lib/openser/modules" loadmodule "maxfwd.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "dispatcher.so" modparam( "dispatcher", "list_file", "/usr/local/etc/openser/dispatcher.list" ) route { if ( ! mf_process_maxfwd_header( "10" ) ) { sl_send_reply( "483", "To Many Hops" ); drop( ); }; ds_select_domain( "1", "0" ); forward( uri:host, uri:port ); } ----------------------------------------------------------------------- dispatcher.list: ----------------------------------------------------------------------- # gateways 1 sip:10.1.50.34:5060 1 sip:10.1.50.35:5060 1 sip:10.1.50.36:5060 1 sip:10.1.50.37:5060 ----------------------------------------------------------------------- tcpdump shows: ----------------------------------------------------------------------- 07:43:37.216888 IP 10.1.50.30.5060 > 10.1.50.31.5060: UDP, length: 685 . at . <http://openser.org/cgi-bin/mailman/listinfo/users>@... .2. .2.........INVITE sip:+18666775910 at 10.1.50.31 <http://openser.org/cgi-bin/mailman/listinfo/users> SIP/2.0 Via: SIP/2.0/UDP 10.1.50.30:5060;branch=z9hG4bK434da07b From: "+19195551212" <sip:+19195551212 at 10.1.50.30 <http://openser.org/cgi-bin/mailman/listinfo/users>>;tag=as3a379ead To: <sip:+18666775910 at 10.1.50.31 <http://openser.org/cgi-bin/mailman/listinfo/users>> Contact: <sip:+19195551212 at 10.1.50.30 <http://openser.org/cgi-bin/mailman/listinfo/users>> Call-ID: 7dd7978262a876832fc69a4364f89f22 at 10.1.50.30 <http://openser.org/cgi-bin/mailman/listinfo/users> CSeq: 102 INVITE User-Agent: BandwidthVoice Date: Fri, 03 Feb 2006 12:41:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 202 v=0 o=root 11014 11014 IN IP4 10.1.50.30 s=session c=IN IP4 10.1.50.30 t=0 0 m=audio 8360 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - ----------------------------------------------------------------------- with no outgoing INVITE. Things seem to be dead. Then I try the same call a few seconds later and I get the outgoing INVITE, but always to the same IP (10.1.50.36) ----------------------------------------------------------------------- 07:43:48.695958 IP 10.1.50.30.5060 > 10.1.50.31.5060: UDP, length: 686 . at . <http://openser.org/cgi-bin/mailman/listinfo/users>@... .2. .2.........INVITE sip:+18666775910 at 10.1.50.31 <http://openser.org/cgi-bin/mailman/listinfo/users> SIP/2.0 Via: SIP/2.0/UDP 10.1.50.30:5060;branch=z9hG4bK15490a4e From: "+19195551212" <sip:+19195551212 at 10.1.50.30 <http://openser.org/cgi-bin/mailman/listinfo/users>>;tag=as302ac772 To: <sip:+18666775910 at 10.1.50.31 <http://openser.org/cgi-bin/mailman/listinfo/users>> Contact: <sip:+19195551212 at 10.1.50.30 <http://openser.org/cgi-bin/mailman/listinfo/users>> Call-ID: 531db3695f15035620baae890d555216 at 10.1.50.30 <http://openser.org/cgi-bin/mailman/listinfo/users> CSeq: 102 INVITE User-Agent: BandwidthVoice Date: Fri, 03 Feb 2006 12:41:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 203 v=0 o=root 11015 11015 IN IP4 10.1.50.30 s=session c=IN IP4 10.1.50.30 t=0 0 m=audio 28484 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - 07:43:48.696191 IP 10.1.50.31.5060 > 10.1.50.36.5060: UDP, length: 747 E....*@. at ..g <http://openser.org/cgi-bin/mailman/listinfo/users> .2. .2$......S.INVITE sip:+18666775910 at 10.1.50.36 <http://openser.org/cgi-bin/mailman/listinfo/users>:5060 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.1.50.31;branch=0 Via: SIP/2.0/UDP 10.1.50.30:5060;branch=z9hG4bK15490a4e From: "+19195551212" <sip:+19195551212 at 10.1.50.30 <http://openser.org/cgi-bin/mailman/listinfo/users>>;tag=as302ac772 To: <sip:+18666775910 at 10.1.50.31 <http://openser.org/cgi-bin/mailman/listinfo/users>> Contact: <sip:+19195551212 at 10.1.50.30 <http://openser.org/cgi-bin/mailman/listinfo/users>> Call-ID: 531db3695f15035620baae890d555216 at 10.1.50.30 <http://openser.org/cgi-bin/mailman/listinfo/users> CSeq: 102 INVITE User-Agent: BandwidthVoice Date: Fri, 03 Feb 2006 12:41:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 203 v=0 o=root 11015 11015 IN IP4 10.1.50.30 s=session c=IN IP4 10.1.50.30 t=0 0 m=audio 28484 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - [View Less]

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[Users] CVS troubles...yet again
by Helge Waastad 08 Feb '06

08 Feb '06

Hi, just wonder if something is to be done regarding the cvs? I am experiencing problems updating the source. To day, I got can't create temporary directory /tmp/cvs-serv28859 No space left on device ....and I'm dying to get a bug-fix this morning :-) br hw -- Helge Waastad Senior Konsulent Systemavdelingen Smartnet

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[Users] postgres and long AVPs values BUG?
by JF 08 Feb '06

08 Feb '06

Hi, I increased the length of the value column in the usr_preferences table from 128 to 1024, but I got a core dump after these logs: 0(24138) val2str(): converting record_route, 12 0(24138) PG[224] str2valp got string <sip:10.112.64.59;ftag=605ff764c617d3cd28dbbdd72be8f9a2;lr=on>,<sip:10.112.64.12;ftag=605ff764c617d3cd28dbbdd72be8f9a2;lr=on>,<sip:10.112.64.12;ftag=605ff764c617d3cd28dbbdd72be8f9a2;lr=on>,<sip:10.112.64.59;ftag=605ff764c617d3cd28dbbdd72be8f9a2;lr=on> … [View More]

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RE: [Users] Receiving different VIA in INVITE and CANCEL
by Papadopoulos Georgios 08 Feb '06

08 Feb '06

Aaah, I see... So how does TM work? Does it try first the pre-RFC3261 and then the RFC3261? Thank you George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Wednesday, February 08, 2006 4:05 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] Receiving different VIA in INVITE and CANCEL > > George, > > there are two algs for transaction matching: > - the pre-RFC3261 one - a … [View More]set of hdrs and RURI are used > to match the CANCEL to the original INVITE > - the RFC3261 alg - the cookies and ID from the branch > VIA param is used. > > via1_matching and ruri_matching are options only for the > pre-RFC3261 alg. > > regards, > bogdan > > Papadopoulos Georgios wrote: > > >Hi Bogdan, > > > >I will try the force_rport(). > > > >Meanwhile I am confused by what you wrote. Can you please clarify? > > > > > >>The problem is the different port in via. The "via1_matching" > >>param has no effect on RFC3261 transaction matching algorithm since > >>this is entirely based on via :). > >> > >> > > > >thanks a lot > > > >George > > > > > > > >>-----Original Message----- > >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > >>Sent: Wednesday, February 08, 2006 11:43 AM > >>To: Papadopoulos Georgios > >>Cc: users(a)openser.org > >>Subject: Re: [Users] Receiving different VIA in INVITE and CANCEL > >> > >>Hi Geroge, > >> > >>it looks like your device uses STUN to perform nat traversal (the > >>callid has a private IP). The improper nat traversal via > STUN is the > >>reason for the different port in VIA; to fix this, call > >>"force_rport()" > >>in the beginning of your script for all requests. > >> > >>now, going back to the CANCEL matching....SIPURA uses the > >>RFC3261 transaction matching algorithm (based on some > cookie and ids > >>stored in branch VIA param. These are matching in your case, but > >>openser uses as additional checkings the via host, port and proto > >>(just to be sure that the originator matches too to avoid confusion > >>with different senders generating the same cookie/id). > >>The problem is the different port in via. The "via1_matching" > >>param has no effect on RFC3261 transaction matching algorithm since > >>this is entirely based on via :). > >> > >>my suggestion is to fix the nat traversal part. > >> > >>regards, > >>bogdan > >> > >>Papadopoulos Georgios wrote: > >> > >> > >> > >>>Hi Bogdan, > >>> > >>>Thank you for your reply. > >>> > >>>I used > >>>modparam("tm", "via1_matching", 0) > >>>modparam("tm", "ruri_matching", 0) > >>>and I got the following trace. You can see that the ACK to > the Proxy > >>>Authentication Required and the CANCEL have different Via than the > >>>INVITE. I know that this breaks the specs but I thought that > >>> > >>> > >>these two > >> > >> > >>>modparams should take care of these broken UAs. > >>> > >>>Also, any ideas what makes the UA switch ports? Is it a NAT > >>> > >>> > >>issue? Only > >> > >> > >>>a very small percentage of the same UA behave this way. > >>> > >>>thank you for any help > >>> > >>>George > >>> > >>> > >>># > >>>U 2006/02/07 12:05:29.756686 195.167.60.21:1231 -> > 213.5.43.134:5060 > >>>INVITE sip:2116872933@sip.i-call.gr SIP/2.0. > >>>Via: SIP/2.0/UDP 195.167.60.21:1327;branch=z9hG4bK-cc55e25. > >>>From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >>>To: <sip:2116872933@sip.i-call.gr>. > >>>Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >>>CSeq: 101 INVITE. > >>>Contact: tester <sip:tester@195.167.60.21:1327>. > >>>Remote-Party-ID: tester > >>><sip:tester@sip.i-call.gr>;screen=yes;party=calling. > >>>Max-Forwards: 70. > >>>Expires: 240. > >>>User-Agent: Sipura/SPA3000-3.1.7(GWg). > >>>Supported: x-sipura. > >>>Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. > >>>Content-Type: application/sdp. > >>>Content-Length: 423 . > >>>. > >>>v=0. > >>>o=- 33757 33757 IN IP4 195.167.60.21. > >>>s=-.. > >>>c=IN IP4 195.167.60.21. > >>>t=0 0. > >>>m=audio 10200 RTP/AVP 18 0 2 4 8 96 97 98 100 101. > >>>a=rtpmap:18 G729a/8000. > >>>a=rtpmap:0 PCMU/8000. > >>>a=rtpmap:2 G726-32/8000. > >>>a=rtpmap:4 G723/8000. > >>>a=rtpmap:8 PCMA/8000. > >>>a=rtpmap:96 G726-40/8000. > >>>a=rtpmap:97 G726-24/8000. > >>>a=rtpmap:98 G726-16/8000. > >>>a=rtpmap:100 NSE/8000. > >>>a=rtpmap:101 telephone-event/8000. > >>>a=fmtp:101 0-15. > >>>a=ptime:30. > >>>a=sendrecv. > >>> > >>># > >>>U 2006/02/07 12:05:29.757360 213.5.43.134:5060 -> > 195.167.60.21:1327 > >>>SIP/2.0 407 Proxy Authentication Required. > >>>Via: SIP/2.0/UDP 195.167.60.21:1327;branch=z9hG4bK-cc55e25. > >>>From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >>>To: > >>><sip:2116872933@sip.i-call.gr>;tag=97e44c910458bd56beb40ca402 > >>> > >>> > >>8d7cc8.0f0 > >> > >> > >>>e > >>>. > >>>Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >>>CSeq: 101 INVITE. > >>>Proxy-Authenticate: Digest realm="i-call.gr", > >>>nonce="43e87215b69f85cf82f231965df4ad938a65d43f". > >>>Server: OpenSer (1.0.0-tls (x86_64/linux)). > >>>Content-Length: 0. > >>>Warning: 392 213.5.43.134:5060 "Noisy feedback tells: pid=10474 > >>>req_src_ip=195.167.60.21 req_src_port=1231 > >>>in_uri=sip:2116872933@sip.i-call.gr > >>>out_uri=sip:2116872933@sip.i-call.gr > >>>via_cnt==1". > >>>. > >>> > >>># > >>>U 2006/02/07 12:05:29.865886 195.167.60.21:1327 -> > 213.5.43.134:5060 > >>>ACK sip:2116872933@sip.i-call.gr SIP/2.0. > >>>Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-cc55e25. > >>>From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >>>To: > >>><sip:2116872933@sip.i-call.gr>;tag=97e44c910458bd56beb40ca402 > >>> > >>> > >>8d7cc8.0f0 > >> > >> > >>>e > >>>. > >>>Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >>>CSeq: 101 ACK. > >>>Contact: tester <sip:tester@195.167.60.21:1328>. > >>>Max-Forwards: 70. > >>>User-Agent: Sipura/SPA3000-3.1.7(GWg). > >>>Content-Length: 0. > >>>. > >>>************************************************************* > >>> > >>> > >>********** > >> > >> > >>>* > >>>************** > >>>*** The Via in the ACK is different than in the INVITE and Openser > >>>sends it > >>>*** to the wrong server > >>>************************************************************* > >>> > >>> > >>********** > >> > >> > >>>* > >>>************** > >>># > >>>U 2006/02/07 12:05:29.866660 213.5.43.134:5060 -> > 213.5.xxx.xxx:5060 > >>>ACK sip:401#2116872933@213.5.xxx.xxx:5060 SIP/2.0. > >>>Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. > >>>Via: SIP/2.0/UDP 213.5.43.134;branch=0. > >>>Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-cc55e25. > >>>From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >>>To: > >>><sip:2116872933@sip.i-call.gr>;tag=97e44c910458bd56beb40ca402 > >>> > >>> > >>8d7cc8.0f0 > >> > >> > >>>e > >>>. > >>>Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >>>CSeq: 101 ACK. > >>>Contact: tester <sip:tester@195.167.60.21:1328>. > >>>Max-Forwards: 69. > >>>User-Agent: Sipura/SPA3000-3.1.7(GWg). > >>>Content-Length: 0. > >>>. > >>> > >>># > >>>U 2006/02/07 12:05:29.959041 195.167.60.21:1327 -> > 213.5.43.134:5060 > >>>INVITE sip:2116872933@sip.i-call.gr SIP/2.0. > >>>Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. > >>>From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >>>To: <sip:2116872933@sip.i-call.gr>. > >>>Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >>>CSeq: 102 INVITE. > >>>Contact: tester <sip:tester@195.167.60.21:1328>. > >>>Proxy-Authorization: Digest username="tester", realm="i-call.gr", > >>>nonce="43e87215b69f85cf82f231965df4ad938a65d43f", > >>>uri="sip:2116872933@sip.i-call.gr", > >>>response="da850791f85d3e882d384d96e4aff30e", algorithm=MD5. > >>>Remote-Party-ID: tester > >>><sip:tester@sip.i-call.gr>;screen=yes;party=calling. > >>>Max-Forwards: 70. > >>>Expires: 240. > >>>User-Agent: Sipura/SPA3000-3.1.7(GWg). > >>>Supported: x-sipura. > >>>Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. > >>>Content-Type: application/sdp. > >>>Content-Length: 423 . > >>>. > >>>v=0. > >>>o=- 33757 33757 IN IP4 195.167.60.21. > >>>s=-.. > >>>c=IN IP4 195.167.60.21. > >>>t=0 0. > >>>m=audio 10200 RTP/AVP 18 0 2 4 8 96 97 98 100 101. > >>>a=rtpmap:18 G729a/8000. > >>>a=rtpmap:0 PCMU/8000. > >>>a=rtpmap:2 G726-32/8000. > >>>a=rtpmap:4 G723/8000. > >>>a=rtpmap:8 PCMA/8000. > >>>a=rtpmap:96 G726-40/8000. > >>>a=rtpmap:97 G726-24/8000. > >>>a=rtpmap:98 G726-16/8000. > >>>a=rtpmap:100 NSE/8000. > >>>a=rtpmap:101 telephone-event/8000. > >>>a=fmtp:101 0-15. > >>>a=ptime:30. > >>>a=sendrecv. > >>> > >>># > >>>U 2006/02/07 12:05:29.960454 213.5.43.134:5060 -> > 195.167.60.21:1328 > >>>SIP/2.0 100 trying -- your call is important to us. > >>>Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. > >>>From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >>>To: <sip:2116872933@sip.i-call.gr>. > >>>Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >>>CSeq: 102 INVITE. > >>>Server: OpenSer (1.0.0-tls (x86_64/linux)). > >>>Content-Length: 0. > >>>Warning: 392 213.5.43.134:5060 "Noisy feedback tells: pid=10473 > >>>req_src_ip=195.167.60.21 req_src_port=1327 > >>>in_uri=sip:2116872933@sip.i-call.gr > >>>out_uri=sip:400#2116872933@213.5.yyy.yyy:5060 via_cnt==1". > >>>. > >>> > >>>************************************************************* > >>> > >>> > >>********** > >> > >> > >>>* > >>>*** Openser forwards the request to PSTN gateway > >>>************************************************************* > >>> > >>> > >>********** > >> > >> > >>>* > >>> > >>># > >>>U 2006/02/07 12:05:29.960516 213.5.43.134:5060 -> > 213.5.yyy.yyy:5060 > >>>INVITE sip:400#2116872933@213.5.yyy.yyy:5060 SIP/2.0. > >>>Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. > >>>Via: SIP/2.0/UDP 213.5.43.134;branch=z9hG4bK53cd.4e18cee6.0. > >>>Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. > >>>From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >>>To: <sip:2116872933@sip.i-call.gr>. > >>>Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >>>CSeq: 102 INVITE. > >>>Contact: tester <sip:tester@195.167.60.21:1328>. > >>>Max-Forwards: 69. > >>>Expires: 240. > >>>User-Agent: Sipura/SPA3000-3.1.7(GWg). > >>>Supported: x-sipura. > >>>Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. > >>>Content-Type: application/sdp. > >>>Content-Length: 423 . > >>>. > >>>v=0. > >>>o=- 33757 33757 IN IP4 195.167.60.21. > >>>s=-.. > >>>c=IN IP4 195.167.60.21. > >>>t=0 0. > >>>m=audio 10200 RTP/AVP 18 0 2 4 8 96 97 98 100 101. > >>>a=rtpmap:18 G729a/8000. > >>>a=rtpmap:0 PCMU/8000. > >>>a=rtpmap:2 G726-32/8000. > >>>a=rtpmap:4 G723/8000. > >>>a=rtpmap:8 PCMA/8000. > >>>a=rtpmap:96 G726-40/8000. > >>>a=rtpmap:97 G726-24/8000. > >>>a=rtpmap:98 G726-16/8000. > >>>a=rtpmap:100 NSE/8000. > >>>a=rtpmap:101 telephone-event/8000. > >>>a=fmtp:101 0-15. > >>>a=ptime:30. > >>>a=sendrecv. > >>> > >>># > >>>U 2006/02/07 12:05:29.975762 213.5.yyy.yyy:57918 -> > >>> > >>> > >>213.5.43.134:5060 > >> > >> > >>>SIP/2.0 100 Trying. > >>>Via: SIP/2.0/UDP > >>> > >>> > >>213.5.43.134;branch=z9hG4bK53cd.4e18cee6.0,SIP/2.0/UDP > >> > >> > >>>195.167.60.21:1328;branch=z9hG4bK-bf20b945. > >>>From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >>>To: <sip:2116872933@sip.i-call.gr>;tag=329C938C-7DF. > >>>Date: Tue, 07 Feb 2006 10:06:34 GMT. > >>>Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >>>Server: Cisco-SIPGateway/IOS-12.x. > >>>CSeq: 102 INVITE. > >>>Allow-Events: telephone-event. > >>>Content-Length: 0. > >>>. > >>> > >>> > >>># > >>>U 2006/02/07 12:05:30.149177 213.5.yyy.yyy:57918 -> > >>> > >>> > >>213.5.43.134:5060 > >> > >> > >>>SIP/2.0 183 Session Progress. > >>>Via: SIP/2.0/UDP > >>> > >>> > >>213.5.43.134;branch=z9hG4bK53cd.4e18cee6.0,SIP/2.0/UDP > >> > >> > >>>195.167.60.21:1328;branch=z9hG4bK-bf20b945. > >>>From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >>>To: <sip:2116872933@sip.i-call.gr>;tag=329C938C-7DF. > >>>Date: Tue, 07 Feb 2006 10:06:34 GMT. > >>>Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >>>Server: Cisco-SIPGateway/IOS-12.x. > >>>CSeq: 102 INVITE. > >>>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, > >>>SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. > >>>Allow-Events: telephone-event. > >>>Contact: <sip:400#2116872933@213.5.yyy.yyy:5060>. > >>>Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. > >>>Content-Disposition: session;handling=required. > >>>Content-Type: application/sdp. > >>>Content-Length: 288. > >>>. > >>>v=0. > >>>o=CiscoSystemsSIP-GW-UserAgent 9560 6985 IN IP4 213.5.yyy.yyy. > >>>s=SIP Call. > >>>c=IN IP4 213.5.yyy.yyy. > >>>t=0 0. > >>>m=audio 17868 RTP/AVP 18 101. > >>>c=IN IP4 213.5.yyy.yyy. > >>>a=rtpmap:18 G729/8000. > >>>a=fmtp:18 annexb=no. > >>>a=rtpmap:101 telephone-event/8000. > >>>a=fmtp:101 0-15. > >>>a=ptime:20. > >>>a=silenceSupp:off - - - -. > >>> > >>># > >>>U 2006/02/07 12:05:30.149263 213.5.43.134:5060 -> > 195.167.60.21:1328 > >>>SIP/2.0 183 Session Progress. > >>>Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. > >>>From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >>>To: <sip:2116872933@sip.i-call.gr>;tag=329C938C-7DF. > >>>Date: Tue, 07 Feb 2006 10:06:34 GMT. > >>>Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >>>Server: Cisco-SIPGateway/IOS-12.x. > >>>CSeq: 102 INVITE. > >>>Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, > >>>SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. > >>>Allow-Events: telephone-event. > >>>Contact: <sip:400#2116872933@213.5.yyy.yyy:5060>. > >>>Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. > >>>Content-Disposition: session;handling=required. > >>>Content-Type: application/sdp. > >>>Content-Length: 288. > >>>. > >>>v=0. > >>>o=CiscoSystemsSIP-GW-UserAgent 9560 6985 IN IP4 213.5.yyy.yyy. > >>>s=SIP Call. > >>>c=IN IP4 213.5.yyy.yyy. > >>>t=0 0. > >>>m=audio 17868 RTP/AVP 18 101. > >>>c=IN IP4 213.5.yyy.yyy. > >>>a=rtpmap:18 G729/8000. > >>>a=fmtp:18 annexb=no. > >>>a=rtpmap:101 telephone-event/8000. > >>>a=fmtp:101 0-15. > >>>a=ptime:20. > >>>a=silenceSupp:off - - - -. > >>> > >>>************************************************************* > >>> > >>> > >>********** > >> > >> > >>>* > >>>**** > >>>*** User Agents sends CANCEL with different Via than in INVITE > >>>*** and Openser send it to the wrong server > >>>************************************************************* > >>> > >>> > >>********** > >> > >> > >>>* > >>>**** > >>> > >>># > >>>U 2006/02/07 12:05:37.859125 195.167.60.21:1328 -> > 213.5.43.134:5060 > >>>CANCEL sip:2116872933@sip.i-call.gr SIP/2.0. > >>>Via: SIP/2.0/UDP 195.167.60.21:1375;branch=z9hG4bK-bf20b945. > >>>From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >>>To: <sip:2116872933@sip.i-call.gr>. > >>>Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >>>CSeq: 102 CANCEL. > >>>Proxy-Authorization: Digest username="tester", realm="i-call.gr", > >>>nonce="43e87215b69f85cf82f231965df4ad938a65d43f", > >>>uri="sip:2116872933@sip.i-call.gr", > >>>response="0c4bcac79619588b61993a5ea407f520", algorithm=MD5. > >>>Max-Forwards: 70. > >>>User-Agent: Sipura/SPA3000-3.1.7(GWg). > >>>Content-Length: 0. > >>>. > >>> > >>># > >>>U 2006/02/07 12:05:37.859999 213.5.43.134:5060 -> > 213.5.xxx.xxx:5060 > >>>CANCEL sip:401#2116872933@213.5.xxx.xxx:5060 SIP/2.0. > >>>Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. > >>>Via: SIP/2.0/UDP 213.5.43.134;branch=z9hG4bK53cd.5e18cee6.0. > >>>Via: SIP/2.0/UDP 195.167.60.21:1375;branch=z9hG4bK-bf20b945. > >>>From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >>>To: <sip:2116872933@sip.i-call.gr>. > >>>Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >>>CSeq: 102 CANCEL. > >>>Proxy-Authorization: Digest username="tester", realm="i-call.gr", > >>>nonce="43e87215b69f85cf82f231965df4ad938a65d43f", > >>>uri="sip:2116872933@sip.i-call.gr", > >>>response="0c4bcac79619588b61993a5ea407f520", algorithm=MD5. > >>>Max-Forwards: 69. > >>>User-Agent: Sipura/SPA3000-3.1.7(GWg). > >>>Content-Length: 0. > >>>. > >>> > >>># > >>>U 2006/02/07 12:05:37.860901 213.5.xxx.xxx:5060 -> > 213.5.43.134:5060 > >>>SIP/2.0 481 Call Leg Does Not Exist. > >>>Via: SIP/2.0/UDP > >>>213.5.43.134;branch=z9hG4bK53cd.5e18cee6.0;received=213.5.43.134. > >>>Via: SIP/2.0/UDP 195.167.60.21:1375;branch=z9hG4bK-bf20b945. > >>>From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >>>To: <sip:2116872933@sip.i-call.gr>;tag=as3061dfb6. > >>>Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >>>CSeq: 102 CANCEL. > >>>User-Agent: i-Call Service. > >>>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > >>>Max-Forwards: 70. > >>>Content-Length: 0. > >>>. > >>> > >>># > >>>U 2006/02/07 12:05:37.860998 213.5.43.134:5060 -> > 195.167.60.21:1375 > >>>SIP/2.0 481 Call Leg Does Not Exist. > >>>Via: SIP/2.0/UDP 195.167.60.21:1375;branch=z9hG4bK-bf20b945. > >>>From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >>>To: <sip:2116872933@sip.i-call.gr>;tag=as3061dfb6. > >>>Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >>>CSeq: 102 CANCEL. > >>>User-Agent: i-Call Service. > >>>Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > >>>Max-Forwards: 70. > >>>Content-Length: 0. > >>>. > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>> > >>>>-----Original Message----- > >>>>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > >>>>Sent: Friday, February 03, 2006 10:43 PM > >>>>To: Papadopoulos Georgios > >>>>Cc: users(a)openser.org > >>>>Subject: Re: [Users] Receiving different VIA in INVITE and CANCEL > >>>> > >>>>Hi George, > >>>> > >>>>different VIA in INVITE and CANCEL brakes the specs of RFC > >>>> > >>>> > >>3261. The > >> > >> > >>>>"via1_matching" is a kind of a trick to make un-compliant > >>>> > >>>> > >>clients to > >> > >> > >>>>work. The trick is functional. I suspect that maybe not > >>>> > >>>> > >>only the VIA > >> > >> > >>>>is the problem. Can you please post the INVITE and CANCEL > to have a > >>>>look on them? > >>>> > >>>>regards, > >>>>bogdan > >>>> > >>>>Papadopoulos Georgios wrote: > >>>> > >>>> > >>>> > >>>> > >>>> > >>>>>Hello, > >>>>> > >>>>>I have an issue with some user agents, (Linksys PAP2 and > >>>>> > >>>>> > >>>>> > >>>>> > >>>>Sipura 3000) > >>>> > >>>> > >>>> > >>>> > >>>>>sending different Via in CANCEL than in INVITE. > >>>>>Of course the result is that the caller hangs up and the > >>>>> > >>>>> > >>>>> > >>>>> > >>>>callee keeps > >>>> > >>>> > >>>> > >>>> > >>>>>ringing. > >>>>>I tried to use > >>>>> modparam("tm", "via1_matching", 0) but it did not solve the > >>>>>problem. > >>>>> > >>>>>I have two questions: > >>>>>1. Is the via1_matching in TM working as described or is > it broken? > >>>>> > >>>>>2. What can be causing this behavior of the user agent? > >>>>> > >>>>> > >>Not all our > >> > >> > >>>>>users who have these user agents report this problem > >>>>> > >>>>> > >>>>> > >>>>> > >>>>(actually very few > >>>> > >>>> > >>>> > >>>> > >>>>>have the problem). In our lab we tried same user agent, same > >>>>> > >>>>> > >>>>> > >>>>> > >>>>firmware > >>>> > >>>> > >>>> > >>>> > >>>>>but could never reproduce this behavior no matter how bad we > >>>>> > >>>>> > >>>>> > >>>>> > >>>>tried to > >>>> > >>>> > >>>> > >>>> > >>>>>mess up their configuration. > >>>>> > >>>>>Any help will be greatly appreciated. Thanks. > >>>>> > >>>>>George > >>>>> > >>>>> > >>>>> > >>>>> > > > >Disclaimer > >The information in this e-mail and any attachments is > confidential. It is intended solely for the attention and use > of the named addressee(s). If you are not the intended > recipient, or person responsible for delivering this > information to the intended recipient, please notify the > sender immediately. Unless you are the intended recipient or > his/her representative you are not authorized to, and must > not, read, copy, distribute, use or retain this message or > any part of it. E-mail transmission cannot be guaranteed to > be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or > contain viruses. > > > > > > > > > > Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. [View Less]

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[Users] NEW FEATURE: NAPTR lookup
by Bogdan-Andrei Iancu 08 Feb '06

08 Feb '06

Hi, A new feature is available in openser, devel version : NAPTR lookup. Overview ========= OpenSER is now RFC3263 (Locating SIP Server) compliant. See http://www.ietf.org/rfc/rfc3263.txt NAPTR queries are used to discovered the transport protocols for SIP that are supported (and a preference order) by a SIP server. Combining the protocols that are supported by the remote server and the protocols that re locally implemented, the local server will choose the protocol to be used (if an … [View More]explicit one is not provided). One the protocol is selected, SRV queries are used to discover the port to be used; after that A record query is made to get the IP address. This will make possible for a proxy to advertise its preferred protocols. Like : if supported, use TLS, otherwise try TCP and if not UDP. Establishing relationships/connections via TLS will be much simpler to configure and it would be dynamically done - if both proxies supports TLS they will use TLS, but this will not exclude UDP and TCP to be used as alternatives.... How it works ============ If NAPTR / SRV lookup is used depends if the proto and port are known. Here is a small diagram: if port specify then -> do just A record lookup and use defaults protos (UDP for SIP and TLS for SIPS) if not specified; done # no port if no proto then -> do NAPTR to get proto; if no results, use defaults protos (UDP for SIP and TLS for SIPS); continue # we have proto now do SRV to get port; if no results, use defaults ports (5060 for SIP and 5061 for SIPS); continue # we have proto and port do A record lookup to get IP The resolver will sequentially try all NAPTR and SRV results until it successfully gets an IP address. NAPTR / SRV records which cannot be completely followed (resolved) will be skipped. If the TLS / TCP support is compiled and enabled, the proxy will consider this protos to use; otherwise it will skip them. How to use ========== NAPTR will be used only if proto is not specify neither via the relaying function nor via the RURI. Ex: t_relay("openser.org"); # send to openser.org and use NAPTR and SRV to resolve the address t_relay(); # RURI=sip:user@domain.org forward(uri:host,uri:port); # RURI=sip:user@domain.org NOTE: these will not use NAPTR t_relay("tcp:openser.org"); t_relay(); # RURI=sip:user@domain.org;transport=udp NOTE: these will not use NAPTR/SRV t_relay("openser.org:5060"); t_relay(); # RURI=sip:user@domain.org:5060 Still under discussion ====================== According to the RFC, for a SIPS uri, the only protocol to be used is TLS. My question is : if SIPS is found force TLS as proto and do SRV lookup or perform NAPTR lookup and look for TLS entries. The difference may be: one extra query (from NAPTR) versus the fact that the NAPTR query may lead to a SRV into another domain. Ex: for sips:user@domainA If forcing TLS -> do SRV for "_sips._tcp.domainA." -> port server1.domainA:5061 If using first NAPTR -> do NAPTR on "domainA" and get "_sips._tcp.domainB." which will lead somewhere else than using directly SRV. According the RFC this is possible. What I'm not sure about is if this cross domain NAPTR records are allowed for TLS: see chapter 4.1 : For NAPTR records with SIPS protocol fields, (if the server is using a site certificate), the domain name in the query and the domain name in the replacement field MUST both be valid based on the site certificate handed out by the server in the TLS exchange. Similarly, the domain name in the SRV query and the domain name in the target in the SRV record MUST both be valid based on the same site certificate. Otherwise, an attacker could modify the DNS records to contain replacement values in a different domain, and the client could not validate that this was the desired behavior or the result of an attack. for the moment, NAPTR and SRV are used for SIPS....but any other opinions on this TLS matter are welcomed... regards, Bogdan PS: feedback is welcomed :D [View Less]

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RE: [Users] Receiving different VIA in INVITE and CANCEL
by Papadopoulos Georgios 08 Feb '06

08 Feb '06

Hi Bogdan, I will try the force_rport(). Meanwhile I am confused by what you wrote. Can you please clarify? > The problem is the different port in via. The "via1_matching" > param has no effect on RFC3261 transaction matching algorithm > since this is entirely based on via :). thanks a lot George > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Wednesday, February 08, 2006 11:43 AM > To: Papadopoulos Georgios > Cc: … [View More]users(a)openser.org > Subject: Re: [Users] Receiving different VIA in INVITE and CANCEL > > Hi Geroge, > > it looks like your device uses STUN to perform nat traversal > (the callid has a private IP). The improper nat traversal via > STUN is the reason for the different port in VIA; to fix > this, call "force_rport()" > in the beginning of your script for all requests. > > now, going back to the CANCEL matching....SIPURA uses the > RFC3261 transaction matching algorithm (based on some cookie > and ids stored in branch VIA param. These are matching in > your case, but openser uses as additional checkings the via > host, port and proto (just to be sure that the originator > matches too to avoid confusion with different senders > generating the same cookie/id). > The problem is the different port in via. The "via1_matching" > param has no effect on RFC3261 transaction matching algorithm > since this is entirely based on via :). > > my suggestion is to fix the nat traversal part. > > regards, > bogdan > > Papadopoulos Georgios wrote: > > >Hi Bogdan, > > > >Thank you for your reply. > > > >I used > >modparam("tm", "via1_matching", 0) > >modparam("tm", "ruri_matching", 0) > >and I got the following trace. You can see that the ACK to the Proxy > >Authentication Required and the CANCEL have different Via than the > >INVITE. I know that this breaks the specs but I thought that > these two > >modparams should take care of these broken UAs. > > > >Also, any ideas what makes the UA switch ports? Is it a NAT > issue? Only > >a very small percentage of the same UA behave this way. > > > >thank you for any help > > > >George > > > > > ># > >U 2006/02/07 12:05:29.756686 195.167.60.21:1231 -> 213.5.43.134:5060 > >INVITE sip:2116872933@sip.i-call.gr SIP/2.0. > >Via: SIP/2.0/UDP 195.167.60.21:1327;branch=z9hG4bK-cc55e25. > >From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >To: <sip:2116872933@sip.i-call.gr>. > >Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >CSeq: 101 INVITE. > >Contact: tester <sip:tester@195.167.60.21:1327>. > >Remote-Party-ID: tester > ><sip:tester@sip.i-call.gr>;screen=yes;party=calling. > >Max-Forwards: 70. > >Expires: 240. > >User-Agent: Sipura/SPA3000-3.1.7(GWg). > >Supported: x-sipura. > >Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. > >Content-Type: application/sdp. > >Content-Length: 423 . > >. > >v=0. > >o=- 33757 33757 IN IP4 195.167.60.21. > >s=-.. > >c=IN IP4 195.167.60.21. > >t=0 0. > >m=audio 10200 RTP/AVP 18 0 2 4 8 96 97 98 100 101. > >a=rtpmap:18 G729a/8000. > >a=rtpmap:0 PCMU/8000. > >a=rtpmap:2 G726-32/8000. > >a=rtpmap:4 G723/8000. > >a=rtpmap:8 PCMA/8000. > >a=rtpmap:96 G726-40/8000. > >a=rtpmap:97 G726-24/8000. > >a=rtpmap:98 G726-16/8000. > >a=rtpmap:100 NSE/8000. > >a=rtpmap:101 telephone-event/8000. > >a=fmtp:101 0-15. > >a=ptime:30. > >a=sendrecv. > > > ># > >U 2006/02/07 12:05:29.757360 213.5.43.134:5060 -> 195.167.60.21:1327 > >SIP/2.0 407 Proxy Authentication Required. > >Via: SIP/2.0/UDP 195.167.60.21:1327;branch=z9hG4bK-cc55e25. > >From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >To: > ><sip:2116872933@sip.i-call.gr>;tag=97e44c910458bd56beb40ca402 > 8d7cc8.0f0 > >e > >. > >Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >CSeq: 101 INVITE. > >Proxy-Authenticate: Digest realm="i-call.gr", > >nonce="43e87215b69f85cf82f231965df4ad938a65d43f". > >Server: OpenSer (1.0.0-tls (x86_64/linux)). > >Content-Length: 0. > >Warning: 392 213.5.43.134:5060 "Noisy feedback tells: pid=10474 > >req_src_ip=195.167.60.21 req_src_port=1231 > >in_uri=sip:2116872933@sip.i-call.gr > >out_uri=sip:2116872933@sip.i-call.gr > >via_cnt==1". > >. > > > ># > >U 2006/02/07 12:05:29.865886 195.167.60.21:1327 -> 213.5.43.134:5060 > >ACK sip:2116872933@sip.i-call.gr SIP/2.0. > >Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-cc55e25. > >From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >To: > ><sip:2116872933@sip.i-call.gr>;tag=97e44c910458bd56beb40ca402 > 8d7cc8.0f0 > >e > >. > >Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >CSeq: 101 ACK. > >Contact: tester <sip:tester@195.167.60.21:1328>. > >Max-Forwards: 70. > >User-Agent: Sipura/SPA3000-3.1.7(GWg). > >Content-Length: 0. > >. > >************************************************************* > ********** > >* > >************** > >*** The Via in the ACK is different than in the INVITE and Openser > >sends it > >*** to the wrong server > >************************************************************* > ********** > >* > >************** > ># > >U 2006/02/07 12:05:29.866660 213.5.43.134:5060 -> 213.5.xxx.xxx:5060 > >ACK sip:401#2116872933@213.5.xxx.xxx:5060 SIP/2.0. > >Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. > >Via: SIP/2.0/UDP 213.5.43.134;branch=0. > >Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-cc55e25. > >From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >To: > ><sip:2116872933@sip.i-call.gr>;tag=97e44c910458bd56beb40ca402 > 8d7cc8.0f0 > >e > >. > >Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >CSeq: 101 ACK. > >Contact: tester <sip:tester@195.167.60.21:1328>. > >Max-Forwards: 69. > >User-Agent: Sipura/SPA3000-3.1.7(GWg). > >Content-Length: 0. > >. > > > ># > >U 2006/02/07 12:05:29.959041 195.167.60.21:1327 -> 213.5.43.134:5060 > >INVITE sip:2116872933@sip.i-call.gr SIP/2.0. > >Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. > >From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >To: <sip:2116872933@sip.i-call.gr>. > >Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >CSeq: 102 INVITE. > >Contact: tester <sip:tester@195.167.60.21:1328>. > >Proxy-Authorization: Digest username="tester", realm="i-call.gr", > >nonce="43e87215b69f85cf82f231965df4ad938a65d43f", > >uri="sip:2116872933@sip.i-call.gr", > >response="da850791f85d3e882d384d96e4aff30e", algorithm=MD5. > >Remote-Party-ID: tester > ><sip:tester@sip.i-call.gr>;screen=yes;party=calling. > >Max-Forwards: 70. > >Expires: 240. > >User-Agent: Sipura/SPA3000-3.1.7(GWg). > >Supported: x-sipura. > >Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. > >Content-Type: application/sdp. > >Content-Length: 423 . > >. > >v=0. > >o=- 33757 33757 IN IP4 195.167.60.21. > >s=-.. > >c=IN IP4 195.167.60.21. > >t=0 0. > >m=audio 10200 RTP/AVP 18 0 2 4 8 96 97 98 100 101. > >a=rtpmap:18 G729a/8000. > >a=rtpmap:0 PCMU/8000. > >a=rtpmap:2 G726-32/8000. > >a=rtpmap:4 G723/8000. > >a=rtpmap:8 PCMA/8000. > >a=rtpmap:96 G726-40/8000. > >a=rtpmap:97 G726-24/8000. > >a=rtpmap:98 G726-16/8000. > >a=rtpmap:100 NSE/8000. > >a=rtpmap:101 telephone-event/8000. > >a=fmtp:101 0-15. > >a=ptime:30. > >a=sendrecv. > > > ># > >U 2006/02/07 12:05:29.960454 213.5.43.134:5060 -> 195.167.60.21:1328 > >SIP/2.0 100 trying -- your call is important to us. > >Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. > >From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >To: <sip:2116872933@sip.i-call.gr>. > >Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >CSeq: 102 INVITE. > >Server: OpenSer (1.0.0-tls (x86_64/linux)). > >Content-Length: 0. > >Warning: 392 213.5.43.134:5060 "Noisy feedback tells: pid=10473 > >req_src_ip=195.167.60.21 req_src_port=1327 > >in_uri=sip:2116872933@sip.i-call.gr > >out_uri=sip:400#2116872933@213.5.yyy.yyy:5060 via_cnt==1". > >. > > > >************************************************************* > ********** > >* > >*** Openser forwards the request to PSTN gateway > >************************************************************* > ********** > >* > > > ># > >U 2006/02/07 12:05:29.960516 213.5.43.134:5060 -> 213.5.yyy.yyy:5060 > >INVITE sip:400#2116872933@213.5.yyy.yyy:5060 SIP/2.0. > >Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. > >Via: SIP/2.0/UDP 213.5.43.134;branch=z9hG4bK53cd.4e18cee6.0. > >Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. > >From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >To: <sip:2116872933@sip.i-call.gr>. > >Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >CSeq: 102 INVITE. > >Contact: tester <sip:tester@195.167.60.21:1328>. > >Max-Forwards: 69. > >Expires: 240. > >User-Agent: Sipura/SPA3000-3.1.7(GWg). > >Supported: x-sipura. > >Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. > >Content-Type: application/sdp. > >Content-Length: 423 . > >. > >v=0. > >o=- 33757 33757 IN IP4 195.167.60.21. > >s=-.. > >c=IN IP4 195.167.60.21. > >t=0 0. > >m=audio 10200 RTP/AVP 18 0 2 4 8 96 97 98 100 101. > >a=rtpmap:18 G729a/8000. > >a=rtpmap:0 PCMU/8000. > >a=rtpmap:2 G726-32/8000. > >a=rtpmap:4 G723/8000. > >a=rtpmap:8 PCMA/8000. > >a=rtpmap:96 G726-40/8000. > >a=rtpmap:97 G726-24/8000. > >a=rtpmap:98 G726-16/8000. > >a=rtpmap:100 NSE/8000. > >a=rtpmap:101 telephone-event/8000. > >a=fmtp:101 0-15. > >a=ptime:30. > >a=sendrecv. > > > ># > >U 2006/02/07 12:05:29.975762 213.5.yyy.yyy:57918 -> > 213.5.43.134:5060 > >SIP/2.0 100 Trying. > >Via: SIP/2.0/UDP > 213.5.43.134;branch=z9hG4bK53cd.4e18cee6.0,SIP/2.0/UDP > >195.167.60.21:1328;branch=z9hG4bK-bf20b945. > >From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >To: <sip:2116872933@sip.i-call.gr>;tag=329C938C-7DF. > >Date: Tue, 07 Feb 2006 10:06:34 GMT. > >Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >Server: Cisco-SIPGateway/IOS-12.x. > >CSeq: 102 INVITE. > >Allow-Events: telephone-event. > >Content-Length: 0. > >. > > > > > ># > >U 2006/02/07 12:05:30.149177 213.5.yyy.yyy:57918 -> > 213.5.43.134:5060 > >SIP/2.0 183 Session Progress. > >Via: SIP/2.0/UDP > 213.5.43.134;branch=z9hG4bK53cd.4e18cee6.0,SIP/2.0/UDP > >195.167.60.21:1328;branch=z9hG4bK-bf20b945. > >From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >To: <sip:2116872933@sip.i-call.gr>;tag=329C938C-7DF. > >Date: Tue, 07 Feb 2006 10:06:34 GMT. > >Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >Server: Cisco-SIPGateway/IOS-12.x. > >CSeq: 102 INVITE. > >Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, > >SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. > >Allow-Events: telephone-event. > >Contact: <sip:400#2116872933@213.5.yyy.yyy:5060>. > >Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. > >Content-Disposition: session;handling=required. > >Content-Type: application/sdp. > >Content-Length: 288. > >. > >v=0. > >o=CiscoSystemsSIP-GW-UserAgent 9560 6985 IN IP4 213.5.yyy.yyy. > >s=SIP Call. > >c=IN IP4 213.5.yyy.yyy. > >t=0 0. > >m=audio 17868 RTP/AVP 18 101. > >c=IN IP4 213.5.yyy.yyy. > >a=rtpmap:18 G729/8000. > >a=fmtp:18 annexb=no. > >a=rtpmap:101 telephone-event/8000. > >a=fmtp:101 0-15. > >a=ptime:20. > >a=silenceSupp:off - - - -. > > > ># > >U 2006/02/07 12:05:30.149263 213.5.43.134:5060 -> 195.167.60.21:1328 > >SIP/2.0 183 Session Progress. > >Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. > >From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >To: <sip:2116872933@sip.i-call.gr>;tag=329C938C-7DF. > >Date: Tue, 07 Feb 2006 10:06:34 GMT. > >Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >Server: Cisco-SIPGateway/IOS-12.x. > >CSeq: 102 INVITE. > >Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, > >SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. > >Allow-Events: telephone-event. > >Contact: <sip:400#2116872933@213.5.yyy.yyy:5060>. > >Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. > >Content-Disposition: session;handling=required. > >Content-Type: application/sdp. > >Content-Length: 288. > >. > >v=0. > >o=CiscoSystemsSIP-GW-UserAgent 9560 6985 IN IP4 213.5.yyy.yyy. > >s=SIP Call. > >c=IN IP4 213.5.yyy.yyy. > >t=0 0. > >m=audio 17868 RTP/AVP 18 101. > >c=IN IP4 213.5.yyy.yyy. > >a=rtpmap:18 G729/8000. > >a=fmtp:18 annexb=no. > >a=rtpmap:101 telephone-event/8000. > >a=fmtp:101 0-15. > >a=ptime:20. > >a=silenceSupp:off - - - -. > > > >************************************************************* > ********** > >* > >**** > >*** User Agents sends CANCEL with different Via than in INVITE > >*** and Openser send it to the wrong server > >************************************************************* > ********** > >* > >**** > > > ># > >U 2006/02/07 12:05:37.859125 195.167.60.21:1328 -> 213.5.43.134:5060 > >CANCEL sip:2116872933@sip.i-call.gr SIP/2.0. > >Via: SIP/2.0/UDP 195.167.60.21:1375;branch=z9hG4bK-bf20b945. > >From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >To: <sip:2116872933@sip.i-call.gr>. > >Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >CSeq: 102 CANCEL. > >Proxy-Authorization: Digest username="tester", realm="i-call.gr", > >nonce="43e87215b69f85cf82f231965df4ad938a65d43f", > >uri="sip:2116872933@sip.i-call.gr", > >response="0c4bcac79619588b61993a5ea407f520", algorithm=MD5. > >Max-Forwards: 70. > >User-Agent: Sipura/SPA3000-3.1.7(GWg). > >Content-Length: 0. > >. > > > ># > >U 2006/02/07 12:05:37.859999 213.5.43.134:5060 -> 213.5.xxx.xxx:5060 > >CANCEL sip:401#2116872933@213.5.xxx.xxx:5060 SIP/2.0. > >Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. > >Via: SIP/2.0/UDP 213.5.43.134;branch=z9hG4bK53cd.5e18cee6.0. > >Via: SIP/2.0/UDP 195.167.60.21:1375;branch=z9hG4bK-bf20b945. > >From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >To: <sip:2116872933@sip.i-call.gr>. > >Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >CSeq: 102 CANCEL. > >Proxy-Authorization: Digest username="tester", realm="i-call.gr", > >nonce="43e87215b69f85cf82f231965df4ad938a65d43f", > >uri="sip:2116872933@sip.i-call.gr", > >response="0c4bcac79619588b61993a5ea407f520", algorithm=MD5. > >Max-Forwards: 69. > >User-Agent: Sipura/SPA3000-3.1.7(GWg). > >Content-Length: 0. > >. > > > ># > >U 2006/02/07 12:05:37.860901 213.5.xxx.xxx:5060 -> 213.5.43.134:5060 > >SIP/2.0 481 Call Leg Does Not Exist. > >Via: SIP/2.0/UDP > >213.5.43.134;branch=z9hG4bK53cd.5e18cee6.0;received=213.5.43.134. > >Via: SIP/2.0/UDP 195.167.60.21:1375;branch=z9hG4bK-bf20b945. > >From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >To: <sip:2116872933@sip.i-call.gr>;tag=as3061dfb6. > >Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >CSeq: 102 CANCEL. > >User-Agent: i-Call Service. > >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > >Max-Forwards: 70. > >Content-Length: 0. > >. > > > ># > >U 2006/02/07 12:05:37.860998 213.5.43.134:5060 -> 195.167.60.21:1375 > >SIP/2.0 481 Call Leg Does Not Exist. > >Via: SIP/2.0/UDP 195.167.60.21:1375;branch=z9hG4bK-bf20b945. > >From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. > >To: <sip:2116872933@sip.i-call.gr>;tag=as3061dfb6. > >Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. > >CSeq: 102 CANCEL. > >User-Agent: i-Call Service. > >Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. > >Max-Forwards: 70. > >Content-Length: 0. > >. > > > > > > > > > > > > > > > >>-----Original Message----- > >>From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > >>Sent: Friday, February 03, 2006 10:43 PM > >>To: Papadopoulos Georgios > >>Cc: users(a)openser.org > >>Subject: Re: [Users] Receiving different VIA in INVITE and CANCEL > >> > >>Hi George, > >> > >>different VIA in INVITE and CANCEL brakes the specs of RFC > 3261. The > >>"via1_matching" is a kind of a trick to make un-compliant > clients to > >>work. The trick is functional. I suspect that maybe not > only the VIA > >>is the problem. Can you please post the INVITE and CANCEL to have a > >>look on them? > >> > >>regards, > >>bogdan > >> > >>Papadopoulos Georgios wrote: > >> > >> > >> > >>>Hello, > >>> > >>>I have an issue with some user agents, (Linksys PAP2 and > >>> > >>> > >>Sipura 3000) > >> > >> > >>>sending different Via in CANCEL than in INVITE. > >>>Of course the result is that the caller hangs up and the > >>> > >>> > >>callee keeps > >> > >> > >>>ringing. > >>>I tried to use > >>> modparam("tm", "via1_matching", 0) > >>>but it did not solve the problem. > >>> > >>>I have two questions: > >>>1. Is the via1_matching in TM working as described or is it broken? > >>> > >>>2. What can be causing this behavior of the user agent? > Not all our > >>>users who have these user agents report this problem > >>> > >>> > >>(actually very few > >> > >> > >>>have the problem). In our lab we tried same user agent, same > >>> > >>> > >>firmware > >> > >> > >>>but could never reproduce this behavior no matter how bad we > >>> > >>> > >>tried to > >> > >> > >>>mess up their configuration. > >>> > >>>Any help will be greatly appreciated. Thanks. > >>> > >>>George > >>> > >>> > Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. [View Less]

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[Users] Dispatcher module: another question.
by Antoine Fressancourt 08 Feb '06

08 Feb '06

Hello, I followed with great interest the discussion about the use of the Dispatcher module that occured in the last few days. Yet, I have another question. For design issues, I wish to balance the incoming SIP messages according to the method they hold : for instance, I want to treat the REGISTER, SUBSCRBE and NOTIFY messages on some machines, and the INVITE, BYE, CANCEL on another set of machines. I know this may seem strange first, but I wonder if any mecanism allow to do that. Thank you … [View More]

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[Users] avpops, duplicate key, empty username and domain
by JF 08 Feb '06

08 Feb '06

Hi, I'm trying to use avpops module, but I'm facing the following error: 0(8178) PG[367] submit_query query 'insert into usr_preferences (uuid,attribute,value,type,username,domain ) values ('test','avpname','test',0,'','')', result 'ERROR: duplicate key violates unique constraint "usr_preferences_pkey" It seems to be caused by the username and domain parameters being empty strings, since the usr_preferences table has this PK constraint: "usr_preferences_pkey" primary key, btree (attribute,… [View More]

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[Serusers] limit the cps-rate
by Thomas Balsfulland 08 Feb '06

08 Feb '06

hello list, i am searching for a way to limit the cps-rate with ser. our class-iv (cisco pgw) behind der ser has problems with too many sip-calls per second and will going to crash. is there any way to limit the cps to a rate i can define? there are no hints in the ser documentation i know. thanks for helping. regards thomas -- thomas balsfulland tbals(a)ctrl-c.de

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[Serusers] how to download the tls module
by Yi Zheng 08 Feb '06

08 Feb '06

Hi, How to download/checkout the experimental tls module? I plan to use it with the latest presence server snapshot (ser-0.10.99-dev30-tm-timers-pa-3]). Thanks, - ming

2 1

[Users] how to add custom certificates to windows messenger 5.1
by Pjothi 08 Feb '06

08 Feb '06

Hello openser users, I am trying to connect windows messenger 5.1 with openser in TLS mode. OpenSER uses the custom created certificate and predictably windows msngr returns -------------- There was a problem in verfiying the certificate from the server. Please contact your network administrator. -------------- I do have my own client certificate and ca-list created with openser scripts and I would like to add it to windows messenger to verify the server certificate. Is it possible to add a … [View More]

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[Users] acc module db_extra accounting issue
by Marcello Lupo 08 Feb '06

08 Feb '06

Hi, i have a db_extra accounting rule like: modparam("acc", "db_extra", "customer_code=$avp(i:102)") The avp value is 12 and i checked it with avp_print() . DEBUG:avpops:print_avp: p=0x2a97a866e8, flags=0 DEBUG: id=<102> DEBUG: val_int=<12> When it is accounted in the table the value become 11 !!!!! the field in the table is type int(10) . How is it possible?? Thanks, Bye, MArcello

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[Users] Clarity on the DISPATCHER module
by Anders Brownworth 08 Feb '06

08 Feb '06

Hi, I am trying to get an OpenSER instance running with the dispatcher module spreading load amongst several other OpenSER servers. I'm having trouble understanding from the docs and practical application exactly what the dispatcher module does and how to make it do what I want. I want to accept requests and use the dispatcher module to send requests to one of a number of other servers. (I want this setup to scale so I don't want to use the tm module.) Each subsequent request with the … [View More]same CallId should go back to the same server. Therefore using the hash over the CallId algorithm in the dispatcher module seems like the best choice. I'm running OpenSER 1.0.0 and working off of the configuration in the modules documentation: http://openser.org/docs/modules/1.0.x/dispatcher.html I fixed up a few things that didn't seem to work out of the box such as replacing forward( ) with forward( uri:host, uri:port ). I have several questions. 1. What exactly does ds_select_dst("1", "0") do? I understand the parameters but the module documentation says it "selects a destination from addresses set". OK, so does it return it somehow? How do I get my hands on what it picks? What is an example of a situation where this function would be useful? 2. What exactly ds_select_domain("1", "0") do? I gather it rewrites the host and port with what it gets from the addresses set (which seems more useful than ds_select_dst), but when I run this in practice, _if_ it rewrites the host / port, it always seems to pick the same destination. (in my tests I only get an outgoing INVITE about 10% of the time) It does seem like I want to be using ds_select_domain() though. Can someone point out where my misunderstanding is? I have my configuration copied below. Thanks. -Anders --------------------------------------------------------------------------------------- What I'm expecting: When the first INVITE comes from 10.1.50.30 (Asterisk) to 10.1.50.31, (OpenSER with dispatcher) I want to see an outgoing INVITE from 10.1.50.31 to one of the listed addresses. (10.1.50.[34-37]) What I'm seeing: A tcpdump shows that most of the time all I see is the incoming INVITE. Seemingly random INVITES produce an outgoing INVITE, even when calling the same number. Presumably the only substantive difference is the CallID. openser.cfg: ----------------------------------------------------------------------------- #debug=4 fork=yes log_stderror=no children=4 check_via=no dns=off rev_dns=off port=5060 mpath="/usr/local/lib/openser/modules" loadmodule "maxfwd.so" loadmodule "sl.so" loadmodule "tm.so" loadmodule "dispatcher.so" modparam( "dispatcher", "list_file", "/usr/local/etc/openser/dispatcher.list" ) route { if ( ! mf_process_maxfwd_header( "10" ) ) { sl_send_reply( "483", "To Many Hops" ); drop( ); }; ds_select_domain( "1", "0" ); forward( uri:host, uri:port ); } ----------------------------------------------------------------------- dispatcher.list: ----------------------------------------------------------------------- # gateways 1 sip:10.1.50.34:5060 1 sip:10.1.50.35:5060 1 sip:10.1.50.36:5060 1 sip:10.1.50.37:5060 ----------------------------------------------------------------------- tcpdump shows: ----------------------------------------------------------------------- 07:43:37.216888 IP 10.1.50.30.5060 > 10.1.50.31.5060: UDP, length: 685 .@.@... .2. .2.........INVITE sip:+18666775910@10.1.50.31 SIP/2.0 Via: SIP/2.0/UDP 10.1.50.30:5060;branch=z9hG4bK434da07b From: "+19195551212" <sip:+19195551212@10.1.50.30>;tag=as3a379ead To: <sip:+18666775910@10.1.50.31> Contact: <sip:+19195551212@10.1.50.30> Call-ID: 7dd7978262a876832fc69a4364f89f22(a)10.1.50.30 CSeq: 102 INVITE User-Agent: BandwidthVoice Date: Fri, 03 Feb 2006 12:41:35 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 202 v=0 o=root 11014 11014 IN IP4 10.1.50.30 s=session c=IN IP4 10.1.50.30 t=0 0 m=audio 8360 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - ----------------------------------------------------------------------- with no outgoing INVITE. Things seem to be dead. Then I try the same call a few seconds later and I get the outgoing INVITE: ----------------------------------------------------------------------- 07:43:48.695958 IP 10.1.50.30.5060 > 10.1.50.31.5060: UDP, length: 686 .@.@... .2. .2.........INVITE sip:+18666775910@10.1.50.31 SIP/2.0 Via: SIP/2.0/UDP 10.1.50.30:5060;branch=z9hG4bK15490a4e From: "+19195551212" <sip:+19195551212@10.1.50.30>;tag=as302ac772 To: <sip:+18666775910@10.1.50.31> Contact: <sip:+19195551212@10.1.50.30> Call-ID: 531db3695f15035620baae890d555216(a)10.1.50.30 CSeq: 102 INVITE User-Agent: BandwidthVoice Date: Fri, 03 Feb 2006 12:41:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 203 v=0 o=root 11015 11015 IN IP4 10.1.50.30 s=session c=IN IP4 10.1.50.30 t=0 0 m=audio 28484 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - 07:43:48.696191 IP 10.1.50.31.5060 > 10.1.50.36.5060: UDP, length: 747 E....*@.@..g .2. .2$......S.INVITE sip:+18666775910@10.1.50.36:5060 SIP/2.0 Max-Forwards: 10 Via: SIP/2.0/UDP 10.1.50.31;branch=0 Via: SIP/2.0/UDP 10.1.50.30:5060;branch=z9hG4bK15490a4e From: "+19195551212" <sip:+19195551212@10.1.50.30>;tag=as302ac772 To: <sip:+18666775910@10.1.50.31> Contact: <sip:+19195551212@10.1.50.30> Call-ID: 531db3695f15035620baae890d555216(a)10.1.50.30 CSeq: 102 INVITE User-Agent: BandwidthVoice Date: Fri, 03 Feb 2006 12:41:46 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER Content-Type: application/sdp Content-Length: 203 v=0 o=root 11015 11015 IN IP4 10.1.50.30 s=session c=IN IP4 10.1.50.30 t=0 0 m=audio 28484 RTP/AVP 0 3 8 a=rtpmap:0 PCMU/8000 a=rtpmap:3 GSM/8000 a=rtpmap:8 PCMA/8000 a=silenceSupp:off - - - - ----------------------------------------------------------------------- any help greatly appreciated. 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[Users] dispatcher forwarding troubles...
by Helge Waastad 08 Feb '06

08 Feb '06

Hi, I just wonder about one thing. I had my dispatcher proxy layer working before the last cvs update. Now I see in my logs: BUG: get_send_socket: unknown proto 0 Feb 4 16:50:04 proxy-01 /usr/sbin/openser[8192]: forward_req: ERROR: cannot forward to af 2, proto 0 no corresponding listening socket I just wonder what these messages mean. I'm forwarding messages to another OpenSER..... br hw -- Helge Waastad Senior Konsulent Systemavdelingen Smartnet

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Re: [Users] user is not found
by Bogdan-Andrei Iancu 08 Feb '06

08 Feb '06

Hi, have you subscribed the user into the DB? regards, bogdan Brijesh wrote: > Hi all > > I am new to OpenSER. > > I am using one of sip client. > > I want to connect sip clients through openSER proxy > > I have installed it and configured the clients but giving me error > that user is not found > > Can anyone tell me what is reason for it ? > >------------------------------------------------------------------------ > >… [View More]

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RE: [Users] Receiving different VIA in INVITE and CANCEL
by Papadopoulos Georgios 08 Feb '06

08 Feb '06

Hi Bogdan, Thank you for your reply. I used modparam("tm", "via1_matching", 0) modparam("tm", "ruri_matching", 0) and I got the following trace. You can see that the ACK to the Proxy Authentication Required and the CANCEL have different Via than the INVITE. I know that this breaks the specs but I thought that these two modparams should take care of these broken UAs. Also, any ideas what makes the UA switch ports? Is it a NAT issue? Only a very small percentage of the same UA behave this way.… [View More] thank you for any help George # U 2006/02/07 12:05:29.756686 195.167.60.21:1231 -> 213.5.43.134:5060 INVITE sip:2116872933@sip.i-call.gr SIP/2.0. Via: SIP/2.0/UDP 195.167.60.21:1327;branch=z9hG4bK-cc55e25. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 101 INVITE. Contact: tester <sip:tester@195.167.60.21:1327>. Remote-Party-ID: tester <sip:tester@sip.i-call.gr>;screen=yes;party=calling. Max-Forwards: 70. Expires: 240. User-Agent: Sipura/SPA3000-3.1.7(GWg). Supported: x-sipura. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Content-Type: application/sdp. Content-Length: 423 . . v=0. o=- 33757 33757 IN IP4 195.167.60.21. s=-.. c=IN IP4 195.167.60.21. t=0 0. m=audio 10200 RTP/AVP 18 0 2 4 8 96 97 98 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 2006/02/07 12:05:29.757360 213.5.43.134:5060 -> 195.167.60.21:1327 SIP/2.0 407 Proxy Authentication Required. Via: SIP/2.0/UDP 195.167.60.21:1327;branch=z9hG4bK-cc55e25. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>;tag=97e44c910458bd56beb40ca4028d7cc8.0f0e . Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 101 INVITE. Proxy-Authenticate: Digest realm="i-call.gr", nonce="43e87215b69f85cf82f231965df4ad938a65d43f". Server: OpenSer (1.0.0-tls (x86_64/linux)). Content-Length: 0. Warning: 392 213.5.43.134:5060 "Noisy feedback tells: pid=10474 req_src_ip=195.167.60.21 req_src_port=1231 in_uri=sip:2116872933@sip.i-call.gr out_uri=sip:2116872933@sip.i-call.gr via_cnt==1". . # U 2006/02/07 12:05:29.865886 195.167.60.21:1327 -> 213.5.43.134:5060 ACK sip:2116872933@sip.i-call.gr SIP/2.0. Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-cc55e25. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>;tag=97e44c910458bd56beb40ca4028d7cc8.0f0e . Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 101 ACK. Contact: tester <sip:tester@195.167.60.21:1328>. Max-Forwards: 70. User-Agent: Sipura/SPA3000-3.1.7(GWg). Content-Length: 0. . ************************************************************************ ************** *** The Via in the ACK is different than in the INVITE and Openser sends it *** to the wrong server ************************************************************************ ************** # U 2006/02/07 12:05:29.866660 213.5.43.134:5060 -> 213.5.xxx.xxx:5060 ACK sip:401#2116872933@213.5.xxx.xxx:5060 SIP/2.0. Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. Via: SIP/2.0/UDP 213.5.43.134;branch=0. Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-cc55e25. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>;tag=97e44c910458bd56beb40ca4028d7cc8.0f0e . Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 101 ACK. Contact: tester <sip:tester@195.167.60.21:1328>. Max-Forwards: 69. User-Agent: Sipura/SPA3000-3.1.7(GWg). Content-Length: 0. . # U 2006/02/07 12:05:29.959041 195.167.60.21:1327 -> 213.5.43.134:5060 INVITE sip:2116872933@sip.i-call.gr SIP/2.0. Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 102 INVITE. Contact: tester <sip:tester@195.167.60.21:1328>. Proxy-Authorization: Digest username="tester", realm="i-call.gr", nonce="43e87215b69f85cf82f231965df4ad938a65d43f", uri="sip:2116872933@sip.i-call.gr", response="da850791f85d3e882d384d96e4aff30e", algorithm=MD5. Remote-Party-ID: tester <sip:tester@sip.i-call.gr>;screen=yes;party=calling. Max-Forwards: 70. Expires: 240. User-Agent: Sipura/SPA3000-3.1.7(GWg). Supported: x-sipura. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Content-Type: application/sdp. Content-Length: 423 . . v=0. o=- 33757 33757 IN IP4 195.167.60.21. s=-.. c=IN IP4 195.167.60.21. t=0 0. m=audio 10200 RTP/AVP 18 0 2 4 8 96 97 98 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 2006/02/07 12:05:29.960454 213.5.43.134:5060 -> 195.167.60.21:1328 SIP/2.0 100 trying -- your call is important to us. Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 102 INVITE. Server: OpenSer (1.0.0-tls (x86_64/linux)). Content-Length: 0. Warning: 392 213.5.43.134:5060 "Noisy feedback tells: pid=10473 req_src_ip=195.167.60.21 req_src_port=1327 in_uri=sip:2116872933@sip.i-call.gr out_uri=sip:400#2116872933@213.5.yyy.yyy:5060 via_cnt==1". . ************************************************************************ *** Openser forwards the request to PSTN gateway ************************************************************************ # U 2006/02/07 12:05:29.960516 213.5.43.134:5060 -> 213.5.yyy.yyy:5060 INVITE sip:400#2116872933@213.5.yyy.yyy:5060 SIP/2.0. Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. Via: SIP/2.0/UDP 213.5.43.134;branch=z9hG4bK53cd.4e18cee6.0. Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 102 INVITE. Contact: tester <sip:tester@195.167.60.21:1328>. Max-Forwards: 69. Expires: 240. User-Agent: Sipura/SPA3000-3.1.7(GWg). Supported: x-sipura. Allow: ACK, BYE, CANCEL, INFO, INVITE, NOTIFY, OPTIONS, REFER. Content-Type: application/sdp. Content-Length: 423 . . v=0. o=- 33757 33757 IN IP4 195.167.60.21. s=-.. c=IN IP4 195.167.60.21. t=0 0. m=audio 10200 RTP/AVP 18 0 2 4 8 96 97 98 100 101. a=rtpmap:18 G729a/8000. a=rtpmap:0 PCMU/8000. a=rtpmap:2 G726-32/8000. a=rtpmap:4 G723/8000. a=rtpmap:8 PCMA/8000. a=rtpmap:96 G726-40/8000. a=rtpmap:97 G726-24/8000. a=rtpmap:98 G726-16/8000. a=rtpmap:100 NSE/8000. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:30. a=sendrecv. # U 2006/02/07 12:05:29.975762 213.5.yyy.yyy:57918 -> 213.5.43.134:5060 SIP/2.0 100 Trying. Via: SIP/2.0/UDP 213.5.43.134;branch=z9hG4bK53cd.4e18cee6.0,SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>;tag=329C938C-7DF. Date: Tue, 07 Feb 2006 10:06:34 GMT. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 102 INVITE. Allow-Events: telephone-event. Content-Length: 0. . # U 2006/02/07 12:05:30.149177 213.5.yyy.yyy:57918 -> 213.5.43.134:5060 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 213.5.43.134;branch=z9hG4bK53cd.4e18cee6.0,SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>;tag=329C938C-7DF. Date: Tue, 07 Feb 2006 10:06:34 GMT. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 102 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116872933@213.5.yyy.yyy:5060>. Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 288. . v=0. o=CiscoSystemsSIP-GW-UserAgent 9560 6985 IN IP4 213.5.yyy.yyy. s=SIP Call. c=IN IP4 213.5.yyy.yyy. t=0 0. m=audio 17868 RTP/AVP 18 101. c=IN IP4 213.5.yyy.yyy. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=silenceSupp:off - - - -. # U 2006/02/07 12:05:30.149263 213.5.43.134:5060 -> 195.167.60.21:1328 SIP/2.0 183 Session Progress. Via: SIP/2.0/UDP 195.167.60.21:1328;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>;tag=329C938C-7DF. Date: Tue, 07 Feb 2006 10:06:34 GMT. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. Server: Cisco-SIPGateway/IOS-12.x. CSeq: 102 INVITE. Allow: INVITE, OPTIONS, BYE, CANCEL, ACK, PRACK, COMET, REFER, SUBSCRIBE, NOTIFY, INFO, UPDATE, REGISTER. Allow-Events: telephone-event. Contact: <sip:400#2116872933@213.5.yyy.yyy:5060>. Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. Content-Disposition: session;handling=required. Content-Type: application/sdp. Content-Length: 288. . v=0. o=CiscoSystemsSIP-GW-UserAgent 9560 6985 IN IP4 213.5.yyy.yyy. s=SIP Call. c=IN IP4 213.5.yyy.yyy. t=0 0. m=audio 17868 RTP/AVP 18 101. c=IN IP4 213.5.yyy.yyy. a=rtpmap:18 G729/8000. a=fmtp:18 annexb=no. a=rtpmap:101 telephone-event/8000. a=fmtp:101 0-15. a=ptime:20. a=silenceSupp:off - - - -. ************************************************************************ **** *** User Agents sends CANCEL with different Via than in INVITE *** and Openser send it to the wrong server ************************************************************************ **** # U 2006/02/07 12:05:37.859125 195.167.60.21:1328 -> 213.5.43.134:5060 CANCEL sip:2116872933@sip.i-call.gr SIP/2.0. Via: SIP/2.0/UDP 195.167.60.21:1375;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 102 CANCEL. Proxy-Authorization: Digest username="tester", realm="i-call.gr", nonce="43e87215b69f85cf82f231965df4ad938a65d43f", uri="sip:2116872933@sip.i-call.gr", response="0c4bcac79619588b61993a5ea407f520", algorithm=MD5. Max-Forwards: 70. User-Agent: Sipura/SPA3000-3.1.7(GWg). Content-Length: 0. . # U 2006/02/07 12:05:37.859999 213.5.43.134:5060 -> 213.5.xxx.xxx:5060 CANCEL sip:401#2116872933@213.5.xxx.xxx:5060 SIP/2.0. Record-Route: <sip:213.5.43.134;ftag=a75dbec37247ce75o0;lr=on>. Via: SIP/2.0/UDP 213.5.43.134;branch=z9hG4bK53cd.5e18cee6.0. Via: SIP/2.0/UDP 195.167.60.21:1375;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 102 CANCEL. Proxy-Authorization: Digest username="tester", realm="i-call.gr", nonce="43e87215b69f85cf82f231965df4ad938a65d43f", uri="sip:2116872933@sip.i-call.gr", response="0c4bcac79619588b61993a5ea407f520", algorithm=MD5. Max-Forwards: 69. User-Agent: Sipura/SPA3000-3.1.7(GWg). Content-Length: 0. . # U 2006/02/07 12:05:37.860901 213.5.xxx.xxx:5060 -> 213.5.43.134:5060 SIP/2.0 481 Call Leg Does Not Exist. Via: SIP/2.0/UDP 213.5.43.134;branch=z9hG4bK53cd.5e18cee6.0;received=213.5.43.134. Via: SIP/2.0/UDP 195.167.60.21:1375;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>;tag=as3061dfb6. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 102 CANCEL. User-Agent: i-Call Service. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Content-Length: 0. . # U 2006/02/07 12:05:37.860998 213.5.43.134:5060 -> 195.167.60.21:1375 SIP/2.0 481 Call Leg Does Not Exist. Via: SIP/2.0/UDP 195.167.60.21:1375;branch=z9hG4bK-bf20b945. From: tester <sip:tester@sip.i-call.gr>;tag=a75dbec37247ce75o0. To: <sip:2116872933@sip.i-call.gr>;tag=as3061dfb6. Call-ID: 6d30a75b-c6343ebd(a)172.22.3.25. CSeq: 102 CANCEL. User-Agent: i-Call Service. Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY. Max-Forwards: 70. Content-Length: 0. . > -----Original Message----- > From: Bogdan-Andrei Iancu [mailto:bogdan@voice-system.ro] > Sent: Friday, February 03, 2006 10:43 PM > To: Papadopoulos Georgios > Cc: users(a)openser.org > Subject: Re: [Users] Receiving different VIA in INVITE and CANCEL > > Hi George, > > different VIA in INVITE and CANCEL brakes the specs of RFC > 3261. The "via1_matching" is a kind of a trick to make > un-compliant clients to work. The trick is functional. I > suspect that maybe not only the VIA is the problem. Can you > please post the INVITE and CANCEL to have a look on them? > > regards, > bogdan > > Papadopoulos Georgios wrote: > > >Hello, > > > >I have an issue with some user agents, (Linksys PAP2 and > Sipura 3000) > >sending different Via in CANCEL than in INVITE. > >Of course the result is that the caller hangs up and the > callee keeps > >ringing. > >I tried to use > > modparam("tm", "via1_matching", 0) > >but it did not solve the problem. > > > >I have two questions: > >1. Is the via1_matching in TM working as described or is it broken? > > > >2. What can be causing this behavior of the user agent? Not all our > >users who have these user agents report this problem > (actually very few > >have the problem). In our lab we tried same user agent, same > firmware > >but could never reproduce this behavior no matter how bad we > tried to > >mess up their configuration. > > > >Any help will be greatly appreciated. Thanks. > > > >George > > > > > > > >Disclaimer > >The information in this e-mail and any attachments is > confidential. It is intended solely for the attention and use > of the named addressee(s). If you are not the intended > recipient, or person responsible for delivering this > information to the intended recipient, please notify the > sender immediately. Unless you are the intended recipient or > his/her representative you are not authorized to, and must > not, read, copy, distribute, use or retain this message or > any part of it. E-mail transmission cannot be guaranteed to > be secure or error-free as information could be intercepted, > corrupted, lost, destroyed, arrive late or incomplete, or > contain viruses. > > > > > > > >_______________________________________________ > >Users mailing list > >Users(a)openser.org > >http://openser.org/cgi-bin/mailman/listinfo/users > > > > > > > > Disclaimer The information in this e-mail and any attachments is confidential. It is intended solely for the attention and use of the named addressee(s). If you are not the intended recipient, or person responsible for delivering this information to the intended recipient, please notify the sender immediately. Unless you are the intended recipient or his/her representative you are not authorized to, and must not, read, copy, distribute, use or retain this message or any part of it. E-mail transmission cannot be guaranteed to be secure or error-free as information could be intercepted, corrupted, lost, destroyed, arrive late or incomplete, or contain viruses. [View Less]

2 1

[Serusers] Presence changes in resource lists
by victoria beltran 08 Feb '06

08 Feb '06

Hi, I don't understand how the ser server manages the resource lists. When one contact of the resource list changes the state (via publish), how does the rls module notify this change to all subscribers of the list? Does the rls module implement the draft-ietf-simple-event-list-07 draft ("A SIP Event Notification Extension for Resource Lists")? Thanks!! --------------------------------- LLama Gratis a cualquier PC del Mundo. Llamadas a fijos y móviles desde 1 cé… [View More]

1 0

[Serusers] Problem with SER and Database connections
by mkumar＠mantragroup.com 08 Feb '06

08 Feb '06

Hi All, I installed SER 0.9.4 and I was able to make and receive calls. SER and Mysql run on the same unix box which is located at some other place and which also runs a webserver. After starting ser everything goes fine but after sometime the whole system goes down and I was not even able to connect to unix box using SSH or to any web page of that webserver. Does the Number of connections made by SER to Mysql is causing the problem or is there anything wrong with my SER configuration and … [View More]

3 6

[Serusers] scripts needed to gathering all SIP messages
by hgaillac-sip＠yahoo.fr 08 Feb '06

08 Feb '06

Hello, I have txt file with calls flow. I look for a script to Create ,for example , a 679400FC-8C211DA-9C3DBF26-F375EEC9@192.168.52.52 file, gathering all SIP messages (find the way to easily detect the start and the end of every SIP messages). Thanks for help Harry 2005/08/10-12:44:44.324 ===== Received ======================= 12:44:44 324 id1/192.168.49.1:5061 <===UDP=== /192.168.49.1:5060 SIP/2.0 200 OK To: <sip:0097143389080@192.168.52.52>;tag=A8C32828-2670 Allow-Events: … [View More]

3 2

[Serusers] SEMS G729
by Kapil Dhawan 08 Feb '06

08 Feb '06

Hi List How can i compile SEMS with g729. _________________________________________________________________ Tried the new MSN Messenger? Its cool! Download now. http://messenger.msn.com/Download/Default.aspx?mkt=en-in

6 9

[Serusers] ser installation
by Alex Wu 08 Feb '06

08 Feb '06

Hi I'm fairly new to sip express router server so the question may have been asked before. I was used debian os,.when installation ser deb package by apt-get install ser .I find this service not really start. Info follow: *********command lines*********************** Alex:/etc/ser# /etc/init.d/ser restart Restarting ser: serListening on udp: 127.0.0.1 [127.0.0.1]:5060 udp: 192.168.10.51 [192.168.10.51]:5060 tcp: 127.0.0.1 [127.0.0.1]:5060 … [View More]

1 0

[Serusers] [Help] wm not signing in
by ryan nalupa 08 Feb '06

08 Feb '06

hi all! i'm still new to ser scripting and ser itself. i just followed what i read in the admin guide and some samples that i obtained from the net. my ser box is using version 0.9.6. my problem is i can't make my wm sign in, i just followed what was said in some docs that i've read but still it won't sign in but when i'm using an xlite softphone it's fine i mean the softphone can log in and can make calls. do i still have to do some more module loading or some parameter settings? please help, … [View More]

1 0

[Serusers] SER Consultant
by Max Clark 07 Feb '06

07 Feb '06

Hello, I am looking for a SER consultant experienced with implementing SER on service provider networks. Engagement would be structured as follows: Initial conversation to judge experiance/expertise on platforms with discussion of implementation options, advise and pitfalls of SER vs. Asterisk on the following list of items. This will be followed by the definition of the scope of work to implement/integrate and document platform - and then of course the deliverables as defined by the scope of … [View More]

1 0

[Serusers] SER call waiting
by Sharon 07 Feb '06

07 Feb '06

Hello all, is it possible to do call waiting when communicating between ser and asterisk. i would appreciate users your thoughts on this. -AA

1 0

[Serusers] succesive calls rejected (radius+mediaproxy)
by miliczo 07 Feb '06

07 Feb '06

Hallo, i'm using radius+mediaproxy and since I moved my sip proxy on a public ip I have the folowing problem. I log in regulary and make my firs call with no problem. But When I hung up, and after few minutes call again, my proxy wont authorise me. I get a message that I need to authorise with proxy. During that my FRERADIUS is recieving requests for authorisation. The second i get rejected i call again and it connects me. I tried lowering nat_ping intervals but no succes. my Invite snippet … [View More]

1 0

RE : [Serusers] SER + Radius + B2Bua
by lists＠cingerr.com 07 Feb '06

07 Feb '06

Im still seeking my head how Vonage and others can do that? does Iptel have any solution for that, if yes how much may that cost? > Depends of what you want to do... > For a real time billing for prepaid, you will need a b2bua, Ser will not give you the possibility to do that(I think). > For postpaid, Ser is enough. > > Olivier > > -----Message d'origine----- > De : Waldo Rubinstein [mailto:waldo@trianet.net] > Envoyé : mardi 7 février 2006 14:15 > À : Olivier … [View More]Taylor > Cc : lists(a)cingerr.com; serusers(a)lists.iptel.org > Objet : Re: RE : RE : RE : [Serusers] SER + Radius + B2Bua > > > Would this also be a viable solution for terminating calls to other SIP gateways instead of just PSTN or would SER-only be sufficient for that? > > Thanks, > Waldo > > On Feb 7, 2006, at 7:54 AM, Olivier Taylor wrote: > >> Well we use it for pstn calls and have canreinvites=yes, then >> Asterisk is >> most of the time outside of the media path. >> The only job of this server is the billing and routing calls to pstn. Furthermore, the number of simultaneous calls Asterisk can manage depends on >> the hardware u will use, transcoding and so on. >> We had more than 50 simultaneous calls on a p4 with 2Gb ram, I >> really don't >> (yet) know what will be the limit. >> >> Olivier >> >> -----Message d'origine----- >> De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la >> part de lists(a)cingerr.com >> Envoyé : mardi 7 février 2006 13:42 >> À : serusers(a)lists.iptel.org >> Objet : Re: RE : RE : [Serusers] SER + Radius + B2Bua >> >> >> Dear Olivier, >> >> Im interesting to know how asterisk behave in heavy load? how many calls at >> a moment is your asterisk handling? >> >>> Hello again, >>> >>> Just try asterisk b2bua, it works fine for us. >>> http://developer.berlios.de/projects/b2bua/ >>> >>> The developper is very kind, subscribe to his mailing list. >>> >>> Good luck, >>> >>> Olivier >>> >>> >>> -----Message d'origine----- >>> De : serusers-bounces(a)lists.iptel.org [mailto:serusers- >>> bounces(a)iptel.org] De >>> la part de lists(a)cingerr.com Envoyé : mardi 7 février 2006 11:54 À : serusers(a)lists.iptel.org >>> Objet : Re: RE : [Serusers] SER + Radius + B2Bua >>> >>> >>> Hello Olivier, >>> >>> I have about two week spending lot of time trying to make it work but no luck, so thank you for saving my time. >>> >>> Is there any known way to make prepaid/postpaid billing with SER? >>> >>> >>> Best Regards >>> Hekuran, >>> >>> >>> >>>> Just a question, do you use Vovida b2bua? >>>> >>>> If yes, forget it, development has been stoped 2 years ago and it's very incomplete. For example, Vovida b2bua is unable to manage re-invites. >>>> >>>> Just to avoid wasted time for you >>>> >>>> Olivier >>>> >>>> -----Message d'origine----- >>>> De : serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.org] De la part de lists(a)cingerr.com Envoyé : mardi 7 février 2006 10:42 À >>>> : serusers(a)lists.iptel.org Objet : [Serusers] SER + Radius + B2Bua >>>> >>>> >>>> Hello, >>>> >>>> Im trying to make a combination of radius and b2bua to make some prepaid/postpaid billing. Till now I have managed to make SER talk to >>>> b2bua and b2bua talk to Radius, but Im not clear yet where does ser or radius stores the call detail records (radacct table is always empty). >>>> >>>> Also my question about prepaid: is there any database schema that provides tables where I store user credit or so (as it says that b2bua have bundled billing system) or should it be done using radius attributes? If so are there any extra attributes that I have to add in dictionary? >>>> >>>> Question about ser: When a user registers with radius it >>>> registers as >>>> username@domain (100(a)some-domain.com) but when I try to call, SER send the call to b2bua using only username ([100]) so the user can not be found. is there any way to register the user only by >>>> username? >>>> >>>> Best Regards >>>> Hekuran >>>> >>>> _______________________________________________ >>>> Serusers mailing list >>>> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers >>>> >>>> >>> >>> _______________________________________________ >>> Serusers mailing list >>> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers >>> >>> >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers >> >> _______________________________________________ >> Serusers mailing list >> serusers(a)lists.iptel.org >> http://lists.iptel.org/mailman/listinfo/serusers > > [View Less]

1 0

[Serusers] SER + Radius + B2Bua
by lists＠cingerr.com 07 Feb '06

07 Feb '06

Hello, Im trying to make a combination of radius and b2bua to make some prepaid/postpaid billing. Till now I have managed to make SER talk to b2bua and b2bua talk to Radius, but Im not clear yet where does ser or radius stores the call detail records (radacct table is always empty). Also my question about prepaid: is there any database schema that provides tables where I store user credit or so (as it says that b2bua have bundled billing system) or should it be done using radius attributes? … [View More]

4 9

[Users] received-Parameter in usrloc
by Andreas Granig 07 Feb '06

07 Feb '06

Hi, Just a question regarding the received-Parameter in usrloc: Is it ALWAYS in the format "sip:<ip>:<port>" or may there also be variations like "sips:<ip>:<port>"? What about TCP-connections? Could there be a transport identifier? Thanks, Andy

2 1

[Users] client could not get server certificate error
by Pjothi 07 Feb '06

07 Feb '06

Dear all, wen a sip client(minisip) tries to connect with OpenSER with TLS enabled on both sides, I get the following error ipMessageTransport: sendMessage: creating new socket Creating new SSL_CTX SSL connect: Error in system call. Could not get server certificate SipMessageTransport: sendMessage: exception thrown! SipMessageTransport: sendMessage: exception thrown! SipMessageTransport: sendMessage: exception thrown! SipMessageTransport: sendMessage: exception thrown! SipMessageTransport: … [View More]

1 0

[Serusers] budgetone not registering
by ryan nalupa 07 Feb '06

07 Feb '06

hi all! can anyone help me with my little problem? i can't make my budgetone register to my ser box? i've used xlite softphones and they're able to call to/from each other. and they're quite fast loggin in. but when i used my budgetone, to test to my ser box, it won't register though i used the same configs on my softphones with the grandstream phone. is there any issue with ser and grandstream phones? ryan __________________________________________________ Do You Yahoo!? Tired of … [View More]

1 0

[Serusers] Deploying VoIP on a WAN
by Joao Pereira 06 Feb '06

06 Feb '06

Hi, As many of you may know, we are undertaking several tests in order to test the interoperability between several PBX IP from different vendors. Until now, we were trusting that the VoIP IP PBX were good enough to be interconnected directly, however, one of the vendors have presented the "SBC" concept. The "SBC" (Session Border Controller) is not a new concept since we were using it anyway when we setup a (Asterisk+SER+SIP Proxy) Box to handle the "on-net dialout" calls. I'm now … [View More]

4 5

[Serusers] Prepaid w/ SER
by Max Clark 06 Feb '06

06 Feb '06

Hi all, I have found a number of calling card applications for Asterisk which have been modified to support a prepaid sip service for origination/termination. Does something like this exist for SER? TIA, Max -- Max Clark http://www.clarksys.com

1 0

[Serusers] SER and asterisk
by Sharon 06 Feb '06

06 Feb '06

hello all, I am able to route calls from asterisk to ser and vice versa. is it possible to make simultaneous calls from asterisk clients to ser clients without getting a 486 busy. -AA

1 0

[Serusers] "admin" user y "heslo" password not valid in webser with presence
by victoria beltran 06 Feb '06

06 Feb '06

Hi, I have installed the presence snapshot and I am trying to install the last version of the serweb from the cvs (cvs -d:pserver:anonymous@cvs.serweb.berlios.de:/cvsroot/serweb checkout .). In the point 8, when I insert the "admin" username and "heslo" password on the page http://148.83.39.57/serweb/admin/admin.php, then "bad username or password" is written. The problem may be database structure has changed in presence snapshot. Did somebody achieve to use serweb or ctl … [View More]

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[Serusers] TLS comments
by Klaus Darilion 06 Feb '06

06 Feb '06

Hi! I've tried the new TLS module: 1. It breaks compatibility with old TLS stack: Even when configured to use TLSv1, it sends an SSLv2 compatible HELLO: server2:~# ssldump New TCP connection #1: 10.10.0.41(33107) <-> 10.10.0.42(5063) 1 1 0.0088 (0.0088) C>S SSLv2 compatible client hello Version 3.1 I do not know if this is a problem with the new or the old stack. Further I do not know what other TLS enabled SIP products use. Do they accept SSL compatible HELLOs? 2. If … [View More]

4 16

[Serusers] Asterisk Registering with SER question
by Ryan Pagquil 06 Feb '06

06 Feb '06

Hi, I've been registering asterisk to ser. I'm using SER as the outbound SIP trunk for Asterisk. Users registered with Asterisk will use the SIP trunk to reach SER registered users and PSTN's. Now when I register Asterisk with SER, on my SER's location table I see these record: Username Column = asterisk Contact Column = sip:s@202.84.24.47 I have a script running that checks the accounting records and sends BYE for the username that has no credit left. I found it hard doing this because … [View More]

2 1

[Serusers] [serweb] Problem registering new user with SerWeb
by mkumar＠mantragroup.com 06 Feb '06

06 Feb '06

Hi All, I installed Serweb 0.9.4 and able to login as User and Admin. But when I hit on admin privileges it is not getting redirected to that page. Also I did not find any option to add a new user, is this a problem with my version of serweb or should I grant any new permissions to the admin. If I should give new permissions to admin then please send me what exactly the permissions are as I could not find them anywhere. Any help will be sincerely appreciated. Cheers, Manoj.

2 1

[Serusers] "Round Robin outgoing calls"
by Jean-Michel Hiver 06 Feb '06

06 Feb '06

Hi, Is there a way with SER to forward calls which should go to PSTN to different gateways in a round robin fashion? Also, is there a way to dynamically disable / enable gateways without restarting ser? Your hints would be greatly appreciated. Cheers, Jean-Michel. -- Jean-Michel Hiver - http://ykoz.net/ Découvrez la Réunion des Technologies IP & Telecom TEL: +262 (0)262 55 03 98 - RCS 434 273 330 SAINT PIERRE

3 2

Re: [Serusers] firewall
by Toygun Mavinil 06 Feb '06

06 Feb '06

After my down question, as no replies come from list, my server hacked i though, Only local sipphones can register and no wan connection is allowed as my firewall allows all connections. Yesterday sipphones from wan could register but today none of them can. I see nothing on var/log/messages ???????????What do i have to do? Which ports i have to leave opento use ser with mediaproxy and sems? Toygun _____ From: serusers-bounces(a)iptel.org [mailto:serusers-bounces@lists.iptel.… [View More]

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Re: [Users] forward_tcp
by Daniel-Constantin Mierla 06 Feb '06

06 Feb '06

In the OpenSER Core Cookbook should be, but the entry is not filled yet. http://openser.org/dokuwiki/doku.php?id=openser_core_cookbook Cheers, Daniel On 02/03/06 20:37, Ramin Dousti wrote: > Thank you so much!!! > > Is that documented somewhere that I missed? > > Ramin > > On 2/2/06, Daniel-Constantin Mierla <daniel(a)voice-system.ro> wrote: > >> Ramin Dousti wrote: >> >>> Hi Klaus, >>> >>> It seems that forward_tcp(… [View More]) requires at least one argument. Any other >>> suggestions? >>> >>> >> you can use: >> forward_tcp(uri:host); >> >> If you set the dst_uri via avpops, then it will have priority over >> request uri. >> >> Cheers, >> Daniel >> >> >>> BTW, when you said "create a new URI and URI contains a transport=tcp >>> parameter" what part of the URI contains the transport protocol? >>> >>> Ramin >>> >>> On 1/26/06, Klaus Darilion <klaus.mailinglists(a)pernau.at> wrote: >>> >>> >>>> Hi Ramin! >>>> >>>> Please always cc to the list >>>> >>>> Ramin Dousti wrote: >>>> >>>> >>>>> Thanks, Klaus. >>>>> >>>>> Here is the full picture: >>>>> >>>>> I receive a SIP URI "sip:user@dest" on a UDP socket. The only thing I need to do >>>>> is to proxy this session on a TCP socket to "dest". >>>>> >>>>> I fugured I can do: >>>>> forward_tcp(dest); >>>>> >>>>> But when I extract dest into an AVP, forwasrd_tcp does not work with an AVP as >>>>> its argument. >>>>> >>>>> avp_printf("s:dst", "$td"); >>>>> forward_tcp("s:dst"); >>>>> >>>>> it tries to resolve "s:dst" literally. >>>>> >>>>> >>>> forward_tcp does not support pseudo variables. >>>> >>>> try forward_tcp() without parameters. >>>> >>>> regards >>>> klaus >>>> >>>> >>>> >>>>>> If the destination is stored in an AVP, use avp_pushto >>>>>> (http://openser.org/docs/modules/1.0.x/avpops.html#AEN291) to push the >>>>>> new URI into the request URI. >>>>>> >>>>>> Then use t_relay (make sure the URI contains a transport=tcp parameter). >>>>>> >>>>>> If the destination is static, use t_relay_to_tcp(ip, port) >>>>>> http://openser.org/docs/modules/1.0.x/tm.html#AEN320 >>>>>> >>>>>> regards >>>>>> klaus >>>>>> >>>>>> >>>>>> >>>>>> Ramin Dousti wrote: >>>>>> >>>>>> >>>>>> >>>>>>> Hello, >>>>>>> >>>>>>> I'm trying to forward a SIP message that's been received by UDP, to >>>>>>> $td through TCP. but the following configuration doesn't work: >>>>>>> >>>>>>> send_tcp("$td", 5060 ); >>>>>>> or >>>>>>> send_tcp($td, 5060 ); >>>>>>> or >>>>>>> avp_printf("s:dst", "$td"); >>>>>>> send_tcp("s:dst", 5060 ); >>>>>>> >>>>>>> Could you please help? >>>>>>> >>>>>>> -- >>>>>>> Ramin >>>>>>> >>>>>>> _______________________________________________ >>>>>>> Users mailing list >>>>>>> Users(a)openser.org >>>>>>> http://openser.org/cgi-bin/mailman/listinfo/users >>>>>>> >>>>>>> >>>>>>> >>>>>>> >>>>> -- >>>>> Ramin >>>>> >>>>> >>>>> >>>>> >>> -- >>> Ramin >>> >>> _______________________________________________ >>> Users mailing list >>> Users(a)openser.org >>> http://openser.org/cgi-bin/mailman/listinfo/users >>> >>> >>> > > > -- > Ramin > > [View Less]

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[Users] Re: OpenSER dispatcher module
by Daniel-Constantin Mierla 06 Feb '06

06 Feb '06

ds_select_dst() sets the outbound proxy address. There is an internal attribute that is used when sending the request. ds_select_domain() rewrites the domain part of request URI. Could you watch the network (ngrep -qt port 4050 on openser box) too see what is happening with the messages? Also, do you get any error in the syslog file (/var/log/syslog or /var/log/messages)? The way you use the dispatcher is ok. Cheers, Daniel On 02/03/06 00:48, Anders Brownworth wrote: > Daniel, > … [View More]> I noticed a few posts on the OpenSER Users' list and thought I'd shoot > you a question directly as my subscription to the list doesn't seem to > be taking yet. > > I want to distribute SIP calls across a number of servers. (similar to > a load balancer) I intend to use the CallId hash algorithm to keep > subsequent messages going to the same server. > > I (like many others) am still trying to understand the dispatcher > module. Working off of the example configuration and running OpenSER > version 1.0.0, I have: > > ... > modparam( "dispatcher", "list_file", > "/usr/local/etc/openser/dispatcher.list" ) > #modparam( "dispatcher", "force_dst", 1 ) > > route { > if ( ! mf_process_maxfwd_header( "10" ) ) { > sl_send_reply( "483", "To Many Hops" ); > drop( ); > > }; > > ds_select_dst( "1", "0" ); > #ds_select_domain( "1", "0" ); > > forward( uri:host, uri:port ); > #t_relay( ); > > } > > > I have not been able to get it to do what I'm expecting. > > I have two different IPs in the dispatcher.list file and am expecting > the OpenSER instance to send an INVITE out to one of those IPs. With > debugging on a high level, I see the dispatcher module internally pick > one of those IPs but there is no attempt to send out an INVITE. > Throwing the t_relay( ) of course replies with a 100 trying, but still > no INVITE is sent out to one of the IPs in the destination set. > > Not really understanding the difference between ds_select_dst() and > ds_select_domain(), I tried ds_select_domain() with no change in > results. (both with and without t_relay()) I understand that > ds_select_destination() rewrites host and port, but then what does > ds_select_dst() do exactly? Why would you ever want to use > ds_select_dst()? It says ds_select_dst() "selects a destination from > the address set". Great, so then what does it do with that > destination? I am unclear on exactly what you are changing in the > request by running each of these functions. Can you clear that up for me? > > Then, once you get a destination, how do you tell OpenSER to send out > an INVITE to the destination server? Or does it not work this way? > > I also saw some talk about how ds_select_dst() "sets the destination > as outbound proxy" but how does that differ from the host / port > rewrite? I think I'm missing something here... > > My intent is to run a stateless distributer so I can massively scale > the front end of my setup. I don't think the tm module is an option > for what I want to do because scale is going to be key. > > I saw a mention of the lcr module but haven't taken a closer look at > that yet. I'd like to get this going as it seems like a very clean and > simple way to accomplish my goal. > > Any help you could provide would be much appreciated. > > Thanks, > > -Anders > > > [View Less]

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[Users] minisip-tls registration with openser problem - new devlopment
by Pjothi 06 Feb '06

06 Feb '06

Dear all, A new development in my attempt to register minisip with openser in tls mode. Previously I was getting an exception and after SSL_CTX but now the socket is getting created but still the client is not registered. I am fairly confident about the Certificates, since I created both client and sever certificate with the script files in OpenSER source. note: minisip (latest source from svn) and OpenSER (compiled with TLS support) both runs in a LAN environment, both on Suse Linux 10 … [View More]Below, is a post of minisip start. ---------------------------------------------------------------------------------------------------------------------- user1@linux:~> minisip Starting MiniSIP ... welcome! Initializing NetUtil Creating SipSoftPhoneConfiguration init 1/9: Creating timeout provider init 2/9: Creating GUI Creating GTK GUI (minisip:6125): gtkmm-WARNING **: gtkmm: Attempt to call Gtk::manage() on a Gtk::Window, but a Gtk::Window has no parent container to manage its lifetime. Minisip: gtk 1 Minisip: gtk 2 Setting contact db Thread 2 running - doing initParseConfig init 3/9: Parsing configuration file () Config file version checked ok! SipIdentity::SipIdentity : cretated identity id=1 SipIdentity::setSipUri: sipUsername=<user4> sipDomain=<192.168.0.4> SipIdentity::setSipProxy: autodetect is false; userUri=user4(a)192.168.0.4; transport = TLS; proxyAddr=192.168.0.4; proxyPort=5061 SipProxy:setProxy(str) : addr = 192.168.0.4 SipIdentity::setProxy: manual sipproxy success ... SipIdentity::setProxy: else ... Identities: identity=1; username=user4; domain=192.168.0.4 proxy=[proxyString=192.168.0.4; proxyString=192.168.0.4; port=5061; transport=TLS; autodetect=no; user=user4; password=user4; expires=1000]; isRegistered=0 init 4/9: Creating IP provider SimpleIPProvider: localIp = SimpleIPProvider: checking interface = lo with IP=127.0.0.1 SimpleIPProvider: checking interface = eth0 with IP=192.168.0.3 SimpleIPProvider: using localIP = 192.168.0.3 init 5/9: Creating MediaHandler Sound I/O: using Spatial Audio Mixer Adding audio codec: G.711 init 6/9: Creating MSip SIP stack init 7/9: Connecting GUI to SIP logic init 8.2/9: Starting TCP transport worker thread init 8.3/9: Starting TLS transport worker thread init 9/9: Registering Identities to registrar server Registering user user4(a)192.168.0.4 to proxy 192.168.0.4, requesting domain 192.168.0.4 SipMessageTransport: sendMessage: creating new socket Creating new SSL_CTX SipMessageTransport: sendMessage: reusing old socket TLS: Shutting down TLS Socket ------------------------------------------------------------------------------------------------------------------ The above is the message I get when I start minisip and I cannot understand the reason why its not registering with the OpenSER. When I disable TLS in OpenSer and connect minisip in udp mode, it registers perfectly with openser. would be very helpful if you can give me your thoughts.....thank you very much for your time. regards, Pjothi [View Less]

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[Serusers] postgresql.conf with Radius
by Voipers Portugal 06 Feb '06

06 Feb '06

Hi, Does anyone has an accounting mechanism working with SER and Radius? I have it all working ok, but when the start and stop request comes to Radius server, it does not write it to the postgres BD. Could anyone provide or let me know where i can find the postgresql.conf file that is supposed to be in the Radius Server machine? Regards, Jose Simoes

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[Users] SIP Session Timeout
by martin＠campus-merseburg.de 06 Feb '06

06 Feb '06

Hi, could i add a Session-Expires header in an invite-request so that the sip session has the sip session timer support? ciao, martin

2 1

Re: Re: [Serusers] radius with ser
by Kenny Yeh 06 Feb '06

06 Feb '06

Greger V. Teigre,Hi I got a new debug information below: Feb 6 16:48:14 debian3 ser: ERROR: auth_radius: can't get code for the Sip-URI-User attribute Feb 6 16:48:14 debian3 ser: init_mod(): Error while initializing module auth_radius what do you think about those? Kenny ======= 2006-02-06 17:53:27 您在来信中写道：======= >See further above in your log. You probably have an error when ser tries to >load the radius module. Probably because the … [View More]

2 1

[Users] TOS setting for SIP signalling
by Matteo Piazza 06 Feb '06

06 Feb '06

Hi, I don't understand if the Tos setting is insert in the CVS of 1.0.0 version or it is in the devel branch. Thank's Matteo

2 1

Re: [Users] Problems registering NAT clients
by Jesus Rodriguez 06 Feb '06

06 Feb '06

On Mon, 6 Feb 2006, Paco Orozco wrote: Hola Paco, I Cc: to the list. >> You can see here that the NAT device is changing the source port from >> 5060 to 10145, which is the port used by Openser to send the reply. >> Openser is doing fine while your NAT device is doing wrong because it >> keeps the external rewriten source port when sending the packet to the >> internal side of the NAT (see the second line of the above tcpdump >> trace). > > You are … [View More]

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[Serusers] radius with ser
by Kenny Yeh 06 Feb '06

06 Feb '06

Hi, I have a big problems.I followed radius howto file where I found on iptel.org,and tested freeradius.when I run ser start,I found some error in syslog below: Feb 6 03:36:39 debian ser: find_export: <radius_www_authorize> not found Feb 6 03:36:39 debian ser: find_export: <radius_www_authorize> not found Feb 6 03:36:39 debian ser: parse error (137,36-37): unknown command, missing loadmodule? I loaded auth_radius.so from ser.cfg,why ser cann't found … [View More]

2 1

[Serusers]Problem with AS5300 to SER
by 昂達 06 Feb '06

06 Feb '06

Hello : This is my AS5300 DID(direct-inward-dial) configuration *********************************************** ! dial-peer voice 101 pots incoming called-number 41000.T no digit-strip direct-inward-dial port 1:D forward-digits all ! dial-peer voice 102 voip huntstop application session destination-pattern 41000.T rtp payload-type nte 98 voice-class codec 99 session protocol sipv2 session target sip-server dtmf-relay rtp-nte ! sip-ua retry invite 3 retry response 3 retry bye 3 … [View More]

1 0

[Users] record_route_preset(): No memory left
by Bastian Schern 05 Feb '06

05 Feb '06

Hello to all, I'm using the latest CVS 1_0_0 and I got the following message in the log after a "record_route_preset( "213.191.xxx.xxx:5060;nat=yes" )": record_route_preset(): No memory left Why this comes? Regards Bastian ____________ Virus checked by G DATA AntiVirusKit Version: AVK 16.5138 from 28.01.2006 Virus news: www.antiviruslab.com

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[Serusers] rtpproxy and fax clients under linux nat
by Antanas Masevicius 04 Feb '06

04 Feb '06

Hello, latetly i got a bit confused while playing with rtpproxy. It is said that it supports nat clients with symmetric nat. As i know, linux is also symmetric by default. Now i get into situation where my natted clients work for voice, but does not for fax. I traced that my nat does not forward incomming T38 stream saying "destination unreachable (port unreachable)". After looking at ip_conntrack during audio call i see: udp 17 180 src=192.168.0.46 dst=my.rtpproxy.ip sport=4004 dport=… [View More]

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[Users] Problems registering NAT clients
by Paco Orozco 04 Feb '06

04 Feb '06

Hello, First of all, I'm spanish and my english isn't good, so please excuse my mistakes. I've been working with openser for a year. Now I'm modifying my SIP service in order to allow clients in NAT scenarios. I've read about symmetric / full cone / ... and other NAT scenarios. I've read about mediaproxy / rtpproxy and nathelper modules. I've got a openser.cfg working properly without NAT clients. And I've modified it (see attach). Client X-Lite: IP: 192.168.1.2 (public address 81.39.220.5) … [View More]Listen SIP port: 5060 Listen RTP port: 8000 Server: Openser 1.0 with Media Proxy IP: 147.83.197.39 Config in opernser.cfg If i try to register with a UAC (X-Lite or K-Phone) with symmetric NAT scenario. Opernser server doesn't answer the client to 5060 port, as i need. X-Lite Diagnostic LOG is client_messages.txt Client timed out without receiving any server messages. This is a tcpdump on clients side: 17:42:23.458980 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 448 17:42:23.558019 IP 147.83.197.39.5060 > 192.168.1.2.10145: SIP, length: 538 17:42:23.558077 IP 192.168.1.2 > 147.83.197.39: ICMP 192.168.1.2 udp port 10145 unreachable, length 556 17:42:25.085485 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 448 17:42:28.108960 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 448 17:42:32.643903 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 2 17:42:34.156151 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 448 On the server this is the tcpdump: 17:42:23.904035 IP 81.39.220.5.10145 > 147.83.197.39.5060: SIP, length: 448 17:42:23.904261 IP 147.83.197.39.5060 > 81.39.220.5.10145: SIP, length: 538 17:42:23.905028 IP 147.83.197.39.5060 > 81.39.220.5.10145: SIP, length: 680 17:42:24.004623 IP 81.39.220.5 > 147.83.197.39: ICMP 81.39.220.5 udp port 10145 unreachable, length 556 17:42:25.525470 IP 81.39.220.5.10145 > 147.83.197.39.5060: SIP, length: 448 17:42:25.525607 IP 147.83.197.39.5060 > 81.39.220.5.10145: SIP, length: 538 17:42:25.526314 IP 147.83.197.39.5060 > 81.39.220.5.10145: SIP, length: 680 17:42:28.549322 IP 81.39.220.5.10145 > 147.83.197.39.5060: SIP, length: 448 What's wrong? It's necessary to use a STUN server? Why can't register? Why openser doesn't reply to 5060 port? My client it's only reachable in this. Can anyone help me? Anyone have a OpenSer config working? Thanks in advance -- ------------------------------------- Aquel que planta árboles bajo los que sabe muy bien que nunca se sentara, ha descubierto el auténtico significado de la vida. [View Less]

2 1

[Users] Problems registering NAT clients
by Paco Orozco 04 Feb '06

04 Feb '06

Hello, The last message was mailformed.... sorry First of all, I'm spanish and my english isn't good, so please excuse my mistakes. I've been working with openser for a year. Now I'm modifying my SIP service in order to allow clients in NAT scenarios. I've read about symmetric / full cone / ... and other NAT scenarios. I've read about mediaproxy / rtpproxy and nathelper modules. I've got a openser.cfg working properly without NAT clients. And I've modified it (see attach). Client X-Lite:… [View More] IP: 192.168.1.2 (public address 81.39.220.5) Listen SIP port: 5060 Listen RTP port: 8000 Server: Openser 1.0 with Media Proxy IP: 147.83.197.39 Config in opernser.cfg If i try to register with a UAC (X-Lite or K-Phone) with symmetric NAT scenario. Opernser server doesn't answer the client to 5060 port, as i need. X-Lite Diagnostic LOG is client_messages.txt Client timed out without receiving any server messages. This is a tcpdump on clients side: 17:42:23.458980 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 448 17:42:23.558019 IP 147.83.197.39.5060 > 192.168.1.2.10145: SIP, length: 538 17:42:23.558077 IP 192.168.1.2 > 147.83.197.39: ICMP 192.168.1.2 udp port 10145 unreachable, length 556 17:42:25.085485 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 448 17:42:28.108960 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 448 17:42:32.643903 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 2 17:42:34.156151 IP 192.168.1.2.5060 > 147.83.197.39.5060: SIP, length: 448 On the server this is the tcpdump: 17:42:23.904035 IP 81.39.220.5.10145 > 147.83.197.39.5060: SIP, length: 448 17:42:23.904261 IP 147.83.197.39.5060 > 81.39.220.5.10145: SIP, length: 538 17:42:23.905028 IP 147.83.197.39.5060 > 81.39.220.5.10145: SIP, length: 680 17:42:24.004623 IP 81.39.220.5 > 147.83.197.39: ICMP 81.39.220.5 udp port 10145 unreachable, length 556 17:42:25.525470 IP 81.39.220.5.10145 > 147.83.197.39.5060: SIP, length: 448 17:42:25.525607 IP 147.83.197.39.5060 > 81.39.220.5.10145: SIP, length: 538 17:42:25.526314 IP 147.83.197.39.5060 > 81.39.220.5.10145: SIP, length: 680 17:42:28.549322 IP 81.39.220.5.10145 > 147.83.197.39.5060: SIP, length: 448 What's wrong? It's necessary to use a STUN server? Why can't register? Why openser doesn't reply to 5060 port? My client it's only reachable in this. Can anyone help me? Anyone have a OpenSer config working? Thanks in advance -- ------------------------------------- Aquel que planta árboles bajo los que sabe muy bien que nunca se sentara, ha descubierto el auténtico significado de la vida. [View Less]

1 0

[Users] user is not found
by Brijesh 04 Feb '06

04 Feb '06

Hi all I am new to OpenSER. I am using one of sip client. I want to connect sip clients through openSER proxy I have installed it and configured the clients but giving me error that user is not found Can anyone tell me what is reason for it ?

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